diff --git a/api/audio_codecs/BUILD.gn b/api/audio_codecs/BUILD.gn index b4b06fb32b..82ed31a5da 100644 --- a/api/audio_codecs/BUILD.gn +++ b/api/audio_codecs/BUILD.gn @@ -62,7 +62,6 @@ rtc_library("builtin_audio_decoder_factory") { "L16:audio_decoder_L16", "g711:audio_decoder_g711", "g722:audio_decoder_g722", - "isac:audio_decoder_isac", ] defines = [] if (rtc_include_ilbc) { @@ -95,7 +94,6 @@ rtc_library("builtin_audio_encoder_factory") { "L16:audio_encoder_L16", "g711:audio_encoder_g711", "g722:audio_encoder_g722", - "isac:audio_encoder_isac", ] defines = [] if (rtc_include_ilbc) { diff --git a/api/audio_codecs/builtin_audio_decoder_factory.cc b/api/audio_codecs/builtin_audio_decoder_factory.cc index 963cfe5cb9..881113d985 100644 --- a/api/audio_codecs/builtin_audio_decoder_factory.cc +++ b/api/audio_codecs/builtin_audio_decoder_factory.cc @@ -20,7 +20,6 @@ #if WEBRTC_USE_BUILTIN_ILBC #include "api/audio_codecs/ilbc/audio_decoder_ilbc.h" // nogncheck #endif -#include "api/audio_codecs/isac/audio_decoder_isac.h" #if WEBRTC_USE_BUILTIN_OPUS #include "api/audio_codecs/opus/audio_decoder_multi_channel_opus.h" #include "api/audio_codecs/opus/audio_decoder_opus.h" // nogncheck @@ -57,7 +56,7 @@ rtc::scoped_refptr CreateBuiltinAudioDecoderFactory() { AudioDecoderOpus, NotAdvertised, #endif - AudioDecoderIsac, AudioDecoderG722, + AudioDecoderG722, #if WEBRTC_USE_BUILTIN_ILBC AudioDecoderIlbc, diff --git a/api/audio_codecs/builtin_audio_encoder_factory.cc b/api/audio_codecs/builtin_audio_encoder_factory.cc index 530d64b2ba..4546a2eaee 100644 --- a/api/audio_codecs/builtin_audio_encoder_factory.cc +++ b/api/audio_codecs/builtin_audio_encoder_factory.cc @@ -20,7 +20,6 @@ #if WEBRTC_USE_BUILTIN_ILBC #include "api/audio_codecs/ilbc/audio_encoder_ilbc.h" // nogncheck #endif -#include "api/audio_codecs/isac/audio_encoder_isac.h" #if WEBRTC_USE_BUILTIN_OPUS #include "api/audio_codecs/opus/audio_encoder_multi_channel_opus.h" #include "api/audio_codecs/opus/audio_encoder_opus.h" // nogncheck @@ -63,7 +62,7 @@ rtc::scoped_refptr CreateBuiltinAudioEncoderFactory() { AudioEncoderOpus, NotAdvertised, #endif - AudioEncoderIsac, AudioEncoderG722, + AudioEncoderG722, #if WEBRTC_USE_BUILTIN_ILBC AudioEncoderIlbc, diff --git a/api/audio_codecs/isac/BUILD.gn b/api/audio_codecs/isac/BUILD.gn deleted file mode 100644 index 96a0ed5013..0000000000 --- a/api/audio_codecs/isac/BUILD.gn +++ /dev/null @@ -1,135 +0,0 @@ -# Copyright (c) 2017 The WebRTC project authors. All Rights Reserved. -# -# Use of this source code is governed by a BSD-style license -# that can be found in the LICENSE file in the root of the source -# tree. An additional intellectual property rights grant can be found -# in the file PATENTS. All contributing project authors may -# be found in the AUTHORS file in the root of the source tree. - -import("../../../webrtc.gni") -if (is_android) { - import("//build/config/android/config.gni") - import("//build/config/android/rules.gni") -} - -# The targets with _fix and _float suffixes unconditionally use the -# fixed-point and floating-point iSAC implementations, respectively. -# The targets without suffixes pick one of the implementations based -# on cleverly chosen criteria. - -rtc_source_set("audio_encoder_isac") { - visibility = [ "*" ] - poisonous = [ "audio_codecs" ] - public = [ "audio_encoder_isac.h" ] - public_configs = [ ":isac_config" ] - if (current_cpu == "arm") { - deps = [ ":audio_encoder_isac_fix" ] - } else { - deps = [ ":audio_encoder_isac_float" ] - } -} - -rtc_source_set("audio_decoder_isac") { - visibility = [ "*" ] - poisonous = [ "audio_codecs" ] - public = [ "audio_decoder_isac.h" ] - public_configs = [ ":isac_config" ] - if (current_cpu == "arm") { - deps = [ ":audio_decoder_isac_fix" ] - } else { - deps = [ ":audio_decoder_isac_float" ] - } -} - -config("isac_config") { - visibility = [ ":*" ] - if (current_cpu == "arm") { - defines = [ - "WEBRTC_USE_BUILTIN_ISAC_FIX=1", - "WEBRTC_USE_BUILTIN_ISAC_FLOAT=0", - ] - } else { - defines = [ - "WEBRTC_USE_BUILTIN_ISAC_FIX=0", - "WEBRTC_USE_BUILTIN_ISAC_FLOAT=1", - ] - } -} - -rtc_library("audio_encoder_isac_fix") { - visibility = [ "*" ] - poisonous = [ "audio_codecs" ] - sources = [ - "audio_encoder_isac_fix.cc", - "audio_encoder_isac_fix.h", - ] - deps = [ - "..:audio_codecs_api", - "../../../api:field_trials_view", - "../../../modules/audio_coding:isac_fix", - "../../../rtc_base:stringutils", - "../../../rtc_base/system:rtc_export", - ] - absl_deps = [ - "//third_party/abseil-cpp/absl/strings", - "//third_party/abseil-cpp/absl/types:optional", - ] -} - -rtc_library("audio_decoder_isac_fix") { - visibility = [ "*" ] - poisonous = [ "audio_codecs" ] - sources = [ - "audio_decoder_isac_fix.cc", - "audio_decoder_isac_fix.h", - ] - deps = [ - "..:audio_codecs_api", - "../../../api:field_trials_view", - "../../../modules/audio_coding:isac_fix", - "../../../rtc_base/system:rtc_export", - ] - absl_deps = [ - "//third_party/abseil-cpp/absl/strings", - "//third_party/abseil-cpp/absl/types:optional", - ] -} - -rtc_library("audio_encoder_isac_float") { - visibility = [ "*" ] - poisonous = [ "audio_codecs" ] - sources = [ - "audio_encoder_isac_float.cc", - "audio_encoder_isac_float.h", - ] - deps = [ - "..:audio_codecs_api", - "../../../api:field_trials_view", - "../../../modules/audio_coding:isac", - "../../../rtc_base:stringutils", - "../../../rtc_base/system:rtc_export", - ] - absl_deps = [ - "//third_party/abseil-cpp/absl/strings", - "//third_party/abseil-cpp/absl/types:optional", - ] -} - -rtc_library("audio_decoder_isac_float") { - visibility = [ "*" ] - poisonous = [ "audio_codecs" ] - sources = [ - "audio_decoder_isac_float.cc", - "audio_decoder_isac_float.h", - ] - deps = [ - "..:audio_codecs_api", - "../../../api:field_trials_view", - "../../../modules/audio_coding:isac", - "../../../rtc_base/system:rtc_export", - ] - absl_deps = [ - "//third_party/abseil-cpp/absl/strings", - "//third_party/abseil-cpp/absl/types:optional", - ] -} diff --git a/api/audio_codecs/isac/audio_decoder_isac.h b/api/audio_codecs/isac/audio_decoder_isac.h deleted file mode 100644 index f4e9331282..0000000000 --- a/api/audio_codecs/isac/audio_decoder_isac.h +++ /dev/null @@ -1,32 +0,0 @@ -/* - * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef API_AUDIO_CODECS_ISAC_AUDIO_DECODER_ISAC_H_ -#define API_AUDIO_CODECS_ISAC_AUDIO_DECODER_ISAC_H_ - -#if WEBRTC_USE_BUILTIN_ISAC_FIX && !WEBRTC_USE_BUILTIN_ISAC_FLOAT -#include "api/audio_codecs/isac/audio_decoder_isac_fix.h" // nogncheck -#elif WEBRTC_USE_BUILTIN_ISAC_FLOAT && !WEBRTC_USE_BUILTIN_ISAC_FIX -#include "api/audio_codecs/isac/audio_decoder_isac_float.h" // nogncheck -#else -#error "Must choose either fix or float" -#endif - -namespace webrtc { - -#if WEBRTC_USE_BUILTIN_ISAC_FIX -using AudioDecoderIsac = AudioDecoderIsacFix; -#elif WEBRTC_USE_BUILTIN_ISAC_FLOAT -using AudioDecoderIsac = AudioDecoderIsacFloat; -#endif - -} // namespace webrtc - -#endif // API_AUDIO_CODECS_ISAC_AUDIO_DECODER_ISAC_H_ diff --git a/api/audio_codecs/isac/audio_decoder_isac_fix.cc b/api/audio_codecs/isac/audio_decoder_isac_fix.cc deleted file mode 100644 index b3ab91da47..0000000000 --- a/api/audio_codecs/isac/audio_decoder_isac_fix.cc +++ /dev/null @@ -1,43 +0,0 @@ -/* - * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#include "api/audio_codecs/isac/audio_decoder_isac_fix.h" - -#include - -#include "absl/strings/match.h" -#include "modules/audio_coding/codecs/isac/fix/include/audio_decoder_isacfix.h" - -namespace webrtc { - -absl::optional AudioDecoderIsacFix::SdpToConfig( - const SdpAudioFormat& format) { - if (absl::EqualsIgnoreCase(format.name, "ISAC") && - format.clockrate_hz == 16000 && format.num_channels == 1) { - return Config(); - } - return absl::nullopt; -} - -void AudioDecoderIsacFix::AppendSupportedDecoders( - std::vector* specs) { - specs->push_back({{"ISAC", 16000, 1}, {16000, 1, 32000, 10000, 32000}}); -} - -std::unique_ptr AudioDecoderIsacFix::MakeAudioDecoder( - Config config, - absl::optional /*codec_pair_id*/, - const FieldTrialsView* field_trials) { - AudioDecoderIsacFixImpl::Config c; - c.sample_rate_hz = 16000; - return std::make_unique(c); -} - -} // namespace webrtc diff --git a/api/audio_codecs/isac/audio_decoder_isac_fix.h b/api/audio_codecs/isac/audio_decoder_isac_fix.h deleted file mode 100644 index 8f61d9ab0e..0000000000 --- a/api/audio_codecs/isac/audio_decoder_isac_fix.h +++ /dev/null @@ -1,40 +0,0 @@ -/* - * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef API_AUDIO_CODECS_ISAC_AUDIO_DECODER_ISAC_FIX_H_ -#define API_AUDIO_CODECS_ISAC_AUDIO_DECODER_ISAC_FIX_H_ - -#include -#include - -#include "absl/types/optional.h" -#include "api/audio_codecs/audio_codec_pair_id.h" -#include "api/audio_codecs/audio_decoder.h" -#include "api/audio_codecs/audio_format.h" -#include "api/field_trials_view.h" -#include "rtc_base/system/rtc_export.h" - -namespace webrtc { - -// iSAC decoder API (fixed-point implementation) for use as a template -// parameter to CreateAudioDecoderFactory<...>(). -struct RTC_EXPORT AudioDecoderIsacFix { - struct Config {}; // Empty---no config values needed! - static absl::optional SdpToConfig(const SdpAudioFormat& audio_format); - static void AppendSupportedDecoders(std::vector* specs); - static std::unique_ptr MakeAudioDecoder( - Config config, - absl::optional codec_pair_id = absl::nullopt, - const FieldTrialsView* field_trials = nullptr); -}; - -} // namespace webrtc - -#endif // API_AUDIO_CODECS_ISAC_AUDIO_DECODER_ISAC_FIX_H_ diff --git a/api/audio_codecs/isac/audio_decoder_isac_float.cc b/api/audio_codecs/isac/audio_decoder_isac_float.cc deleted file mode 100644 index 98f672b468..0000000000 --- a/api/audio_codecs/isac/audio_decoder_isac_float.cc +++ /dev/null @@ -1,56 +0,0 @@ -/* - * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#include "api/audio_codecs/isac/audio_decoder_isac_float.h" - -#include - -#include "absl/strings/match.h" -#include "modules/audio_coding/codecs/isac/main/include/audio_decoder_isac.h" - -namespace webrtc { - -absl::optional -AudioDecoderIsacFloat::SdpToConfig(const SdpAudioFormat& format) { - if (absl::EqualsIgnoreCase(format.name, "ISAC") && - (format.clockrate_hz == 16000 || format.clockrate_hz == 32000) && - format.num_channels == 1) { - Config config; - config.sample_rate_hz = format.clockrate_hz; - if (!config.IsOk()) { - RTC_DCHECK_NOTREACHED(); - return absl::nullopt; - } - return config; - } else { - return absl::nullopt; - } -} - -void AudioDecoderIsacFloat::AppendSupportedDecoders( - std::vector* specs) { - specs->push_back({{"ISAC", 16000, 1}, {16000, 1, 32000, 10000, 32000}}); - specs->push_back({{"ISAC", 32000, 1}, {32000, 1, 56000, 10000, 56000}}); -} - -std::unique_ptr AudioDecoderIsacFloat::MakeAudioDecoder( - Config config, - absl::optional /*codec_pair_id*/, - const FieldTrialsView* field_trials) { - AudioDecoderIsacFloatImpl::Config c; - c.sample_rate_hz = config.sample_rate_hz; - if (!config.IsOk()) { - RTC_DCHECK_NOTREACHED(); - return nullptr; - } - return std::make_unique(c); -} - -} // namespace webrtc diff --git a/api/audio_codecs/isac/audio_decoder_isac_float.h b/api/audio_codecs/isac/audio_decoder_isac_float.h deleted file mode 100644 index 864c6b999f..0000000000 --- a/api/audio_codecs/isac/audio_decoder_isac_float.h +++ /dev/null @@ -1,45 +0,0 @@ -/* - * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef API_AUDIO_CODECS_ISAC_AUDIO_DECODER_ISAC_FLOAT_H_ -#define API_AUDIO_CODECS_ISAC_AUDIO_DECODER_ISAC_FLOAT_H_ - -#include -#include - -#include "absl/types/optional.h" -#include "api/audio_codecs/audio_codec_pair_id.h" -#include "api/audio_codecs/audio_decoder.h" -#include "api/audio_codecs/audio_format.h" -#include "api/field_trials_view.h" -#include "rtc_base/system/rtc_export.h" - -namespace webrtc { - -// iSAC decoder API (floating-point implementation) for use as a template -// parameter to CreateAudioDecoderFactory<...>(). -struct RTC_EXPORT AudioDecoderIsacFloat { - struct Config { - bool IsOk() const { - return sample_rate_hz == 16000 || sample_rate_hz == 32000; - } - int sample_rate_hz = 16000; - }; - static absl::optional SdpToConfig(const SdpAudioFormat& audio_format); - static void AppendSupportedDecoders(std::vector* specs); - static std::unique_ptr MakeAudioDecoder( - Config config, - absl::optional codec_pair_id = absl::nullopt, - const FieldTrialsView* field_trials = nullptr); -}; - -} // namespace webrtc - -#endif // API_AUDIO_CODECS_ISAC_AUDIO_DECODER_ISAC_FLOAT_H_ diff --git a/api/audio_codecs/isac/audio_encoder_isac.h b/api/audio_codecs/isac/audio_encoder_isac.h deleted file mode 100644 index 3cb0a1f053..0000000000 --- a/api/audio_codecs/isac/audio_encoder_isac.h +++ /dev/null @@ -1,32 +0,0 @@ -/* - * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef API_AUDIO_CODECS_ISAC_AUDIO_ENCODER_ISAC_H_ -#define API_AUDIO_CODECS_ISAC_AUDIO_ENCODER_ISAC_H_ - -#if WEBRTC_USE_BUILTIN_ISAC_FIX && !WEBRTC_USE_BUILTIN_ISAC_FLOAT -#include "api/audio_codecs/isac/audio_encoder_isac_fix.h" // nogncheck -#elif WEBRTC_USE_BUILTIN_ISAC_FLOAT && !WEBRTC_USE_BUILTIN_ISAC_FIX -#include "api/audio_codecs/isac/audio_encoder_isac_float.h" // nogncheck -#else -#error "Must choose either fix or float" -#endif - -namespace webrtc { - -#if WEBRTC_USE_BUILTIN_ISAC_FIX -using AudioEncoderIsac = AudioEncoderIsacFix; -#elif WEBRTC_USE_BUILTIN_ISAC_FLOAT -using AudioEncoderIsac = AudioEncoderIsacFloat; -#endif - -} // namespace webrtc - -#endif // API_AUDIO_CODECS_ISAC_AUDIO_ENCODER_ISAC_H_ diff --git a/api/audio_codecs/isac/audio_encoder_isac_fix.cc b/api/audio_codecs/isac/audio_encoder_isac_fix.cc deleted file mode 100644 index 39603775a4..0000000000 --- a/api/audio_codecs/isac/audio_encoder_isac_fix.cc +++ /dev/null @@ -1,72 +0,0 @@ -/* - * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#include "api/audio_codecs/isac/audio_encoder_isac_fix.h" - -#include - -#include "absl/strings/match.h" -#include "modules/audio_coding/codecs/isac/fix/include/audio_encoder_isacfix.h" -#include "rtc_base/string_to_number.h" - -namespace webrtc { - -absl::optional AudioEncoderIsacFix::SdpToConfig( - const SdpAudioFormat& format) { - if (absl::EqualsIgnoreCase(format.name, "ISAC") && - format.clockrate_hz == 16000 && format.num_channels == 1) { - Config config; - const auto ptime_iter = format.parameters.find("ptime"); - if (ptime_iter != format.parameters.end()) { - const auto ptime = rtc::StringToNumber(ptime_iter->second); - if (ptime && *ptime >= 60) { - config.frame_size_ms = 60; - } - } - if (!config.IsOk()) { - RTC_DCHECK_NOTREACHED(); - return absl::nullopt; - } - return config; - } else { - return absl::nullopt; - } -} - -void AudioEncoderIsacFix::AppendSupportedEncoders( - std::vector* specs) { - const SdpAudioFormat fmt = {"ISAC", 16000, 1}; - const AudioCodecInfo info = QueryAudioEncoder(*SdpToConfig(fmt)); - specs->push_back({fmt, info}); -} - -AudioCodecInfo AudioEncoderIsacFix::QueryAudioEncoder( - AudioEncoderIsacFix::Config config) { - RTC_DCHECK(config.IsOk()); - return {16000, 1, 32000, 10000, 32000}; -} - -std::unique_ptr AudioEncoderIsacFix::MakeAudioEncoder( - AudioEncoderIsacFix::Config config, - int payload_type, - absl::optional /*codec_pair_id*/, - const FieldTrialsView* field_trials) { - AudioEncoderIsacFixImpl::Config c; - c.frame_size_ms = config.frame_size_ms; - c.bit_rate = config.bit_rate; - c.payload_type = payload_type; - if (!config.IsOk()) { - RTC_DCHECK_NOTREACHED(); - return nullptr; - } - return std::make_unique(c); -} - -} // namespace webrtc diff --git a/api/audio_codecs/isac/audio_encoder_isac_fix.h b/api/audio_codecs/isac/audio_encoder_isac_fix.h deleted file mode 100644 index de0f1d1308..0000000000 --- a/api/audio_codecs/isac/audio_encoder_isac_fix.h +++ /dev/null @@ -1,54 +0,0 @@ -/* - * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef API_AUDIO_CODECS_ISAC_AUDIO_ENCODER_ISAC_FIX_H_ -#define API_AUDIO_CODECS_ISAC_AUDIO_ENCODER_ISAC_FIX_H_ - -#include -#include - -#include "absl/types/optional.h" -#include "api/audio_codecs/audio_codec_pair_id.h" -#include "api/audio_codecs/audio_encoder.h" -#include "api/audio_codecs/audio_format.h" -#include "api/field_trials_view.h" -#include "rtc_base/system/rtc_export.h" - -namespace webrtc { - -// iSAC encoder API (fixed-point implementation) for use as a template -// parameter to CreateAudioEncoderFactory<...>(). -struct RTC_EXPORT AudioEncoderIsacFix { - struct Config { - bool IsOk() const { - if (frame_size_ms != 30 && frame_size_ms != 60) { - return false; - } - if (bit_rate < 10000 || bit_rate > 32000) { - return false; - } - return true; - } - int frame_size_ms = 30; - int bit_rate = 32000; // Limit on short-term average bit rate, in bits/s. - }; - static absl::optional SdpToConfig(const SdpAudioFormat& audio_format); - static void AppendSupportedEncoders(std::vector* specs); - static AudioCodecInfo QueryAudioEncoder(Config config); - static std::unique_ptr MakeAudioEncoder( - Config config, - int payload_type, - absl::optional codec_pair_id = absl::nullopt, - const FieldTrialsView* field_trials = nullptr); -}; - -} // namespace webrtc - -#endif // API_AUDIO_CODECS_ISAC_AUDIO_ENCODER_ISAC_FIX_H_ diff --git a/api/audio_codecs/isac/audio_encoder_isac_float.cc b/api/audio_codecs/isac/audio_encoder_isac_float.cc deleted file mode 100644 index e3e50080fa..0000000000 --- a/api/audio_codecs/isac/audio_encoder_isac_float.cc +++ /dev/null @@ -1,85 +0,0 @@ -/* - * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#include "api/audio_codecs/isac/audio_encoder_isac_float.h" - -#include - -#include "absl/strings/match.h" -#include "modules/audio_coding/codecs/isac/main/include/audio_encoder_isac.h" -#include "rtc_base/string_to_number.h" - -namespace webrtc { - -absl::optional -AudioEncoderIsacFloat::SdpToConfig(const SdpAudioFormat& format) { - if (absl::EqualsIgnoreCase(format.name, "ISAC") && - (format.clockrate_hz == 16000 || format.clockrate_hz == 32000) && - format.num_channels == 1) { - Config config; - config.sample_rate_hz = format.clockrate_hz; - config.bit_rate = format.clockrate_hz == 16000 ? 32000 : 56000; - if (config.sample_rate_hz == 16000) { - // For sample rate 16 kHz, optionally use 60 ms frames, instead of the - // default 30 ms. - const auto ptime_iter = format.parameters.find("ptime"); - if (ptime_iter != format.parameters.end()) { - const auto ptime = rtc::StringToNumber(ptime_iter->second); - if (ptime && *ptime >= 60) { - config.frame_size_ms = 60; - } - } - } - if (!config.IsOk()) { - RTC_DCHECK_NOTREACHED(); - return absl::nullopt; - } - return config; - } else { - return absl::nullopt; - } -} - -void AudioEncoderIsacFloat::AppendSupportedEncoders( - std::vector* specs) { - for (int sample_rate_hz : {16000, 32000}) { - const SdpAudioFormat fmt = {"ISAC", sample_rate_hz, 1}; - const AudioCodecInfo info = QueryAudioEncoder(*SdpToConfig(fmt)); - specs->push_back({fmt, info}); - } -} - -AudioCodecInfo AudioEncoderIsacFloat::QueryAudioEncoder( - const AudioEncoderIsacFloat::Config& config) { - RTC_DCHECK(config.IsOk()); - constexpr int min_bitrate = 10000; - const int max_bitrate = config.sample_rate_hz == 16000 ? 32000 : 56000; - const int default_bitrate = max_bitrate; - return {config.sample_rate_hz, 1, default_bitrate, min_bitrate, max_bitrate}; -} - -std::unique_ptr AudioEncoderIsacFloat::MakeAudioEncoder( - const AudioEncoderIsacFloat::Config& config, - int payload_type, - absl::optional /*codec_pair_id*/, - const FieldTrialsView* field_trials) { - AudioEncoderIsacFloatImpl::Config c; - c.payload_type = payload_type; - c.sample_rate_hz = config.sample_rate_hz; - c.frame_size_ms = config.frame_size_ms; - c.bit_rate = config.bit_rate; - if (!config.IsOk()) { - RTC_DCHECK_NOTREACHED(); - return nullptr; - } - return std::make_unique(c); -} - -} // namespace webrtc diff --git a/api/audio_codecs/isac/audio_encoder_isac_float.h b/api/audio_codecs/isac/audio_encoder_isac_float.h deleted file mode 100644 index d031d76db1..0000000000 --- a/api/audio_codecs/isac/audio_encoder_isac_float.h +++ /dev/null @@ -1,68 +0,0 @@ -/* - * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef API_AUDIO_CODECS_ISAC_AUDIO_ENCODER_ISAC_FLOAT_H_ -#define API_AUDIO_CODECS_ISAC_AUDIO_ENCODER_ISAC_FLOAT_H_ - -#include -#include - -#include "absl/types/optional.h" -#include "api/audio_codecs/audio_codec_pair_id.h" -#include "api/audio_codecs/audio_encoder.h" -#include "api/audio_codecs/audio_format.h" -#include "api/field_trials_view.h" -#include "rtc_base/system/rtc_export.h" - -namespace webrtc { - -// iSAC encoder API (floating-point implementation) for use as a template -// parameter to CreateAudioEncoderFactory<...>(). -struct RTC_EXPORT AudioEncoderIsacFloat { - struct Config { - bool IsOk() const { - switch (sample_rate_hz) { - case 16000: - if (frame_size_ms != 30 && frame_size_ms != 60) { - return false; - } - if (bit_rate < 10000 || bit_rate > 32000) { - return false; - } - return true; - case 32000: - if (frame_size_ms != 30) { - return false; - } - if (bit_rate < 10000 || bit_rate > 56000) { - return false; - } - return true; - default: - return false; - } - } - int sample_rate_hz = 16000; - int frame_size_ms = 30; - int bit_rate = 32000; // Limit on short-term average bit rate, in bits/s. - }; - static absl::optional SdpToConfig(const SdpAudioFormat& audio_format); - static void AppendSupportedEncoders(std::vector* specs); - static AudioCodecInfo QueryAudioEncoder(const Config& config); - static std::unique_ptr MakeAudioEncoder( - const Config& config, - int payload_type, - absl::optional codec_pair_id = absl::nullopt, - const FieldTrialsView* field_trials = nullptr); -}; - -} // namespace webrtc - -#endif // API_AUDIO_CODECS_ISAC_AUDIO_ENCODER_ISAC_FLOAT_H_ diff --git a/api/audio_codecs/test/BUILD.gn b/api/audio_codecs/test/BUILD.gn index 12df649feb..89f5fef1ea 100644 --- a/api/audio_codecs/test/BUILD.gn +++ b/api/audio_codecs/test/BUILD.gn @@ -32,10 +32,6 @@ if (rtc_include_tests) { "../g722:audio_encoder_g722", "../ilbc:audio_decoder_ilbc", "../ilbc:audio_encoder_ilbc", - "../isac:audio_decoder_isac_fix", - "../isac:audio_decoder_isac_float", - "../isac:audio_encoder_isac_fix", - "../isac:audio_encoder_isac_float", "../opus:audio_decoder_opus", "../opus:audio_encoder_opus", ] diff --git a/api/audio_codecs/test/audio_decoder_factory_template_unittest.cc b/api/audio_codecs/test/audio_decoder_factory_template_unittest.cc index 3662f3b76d..0b18cf934a 100644 --- a/api/audio_codecs/test/audio_decoder_factory_template_unittest.cc +++ b/api/audio_codecs/test/audio_decoder_factory_template_unittest.cc @@ -16,8 +16,6 @@ #include "api/audio_codecs/g711/audio_decoder_g711.h" #include "api/audio_codecs/g722/audio_decoder_g722.h" #include "api/audio_codecs/ilbc/audio_decoder_ilbc.h" -#include "api/audio_codecs/isac/audio_decoder_isac_fix.h" -#include "api/audio_codecs/isac/audio_decoder_isac_float.h" #include "api/audio_codecs/opus/audio_decoder_opus.h" #include "test/gmock.h" #include "test/gtest.h" @@ -182,41 +180,6 @@ TEST(AudioDecoderFactoryTemplateTest, Ilbc) { EXPECT_EQ(8000, dec->SampleRateHz()); } -TEST(AudioDecoderFactoryTemplateTest, IsacFix) { - auto factory = CreateAudioDecoderFactory(); - EXPECT_THAT(factory->GetSupportedDecoders(), - ::testing::ElementsAre(AudioCodecSpec{ - {"ISAC", 16000, 1}, {16000, 1, 32000, 10000, 32000}})); - EXPECT_FALSE(factory->IsSupportedDecoder({"isac", 16000, 2})); - EXPECT_TRUE(factory->IsSupportedDecoder({"isac", 16000, 1})); - EXPECT_FALSE(factory->IsSupportedDecoder({"isac", 32000, 1})); - EXPECT_EQ(nullptr, - factory->MakeAudioDecoder({"isac", 8000, 1}, absl::nullopt)); - auto dec = factory->MakeAudioDecoder({"isac", 16000, 1}, absl::nullopt); - ASSERT_NE(nullptr, dec); - EXPECT_EQ(16000, dec->SampleRateHz()); -} - -TEST(AudioDecoderFactoryTemplateTest, IsacFloat) { - auto factory = CreateAudioDecoderFactory(); - EXPECT_THAT( - factory->GetSupportedDecoders(), - ::testing::ElementsAre( - AudioCodecSpec{{"ISAC", 16000, 1}, {16000, 1, 32000, 10000, 32000}}, - AudioCodecSpec{{"ISAC", 32000, 1}, {32000, 1, 56000, 10000, 56000}})); - EXPECT_FALSE(factory->IsSupportedDecoder({"isac", 16000, 2})); - EXPECT_TRUE(factory->IsSupportedDecoder({"isac", 16000, 1})); - EXPECT_TRUE(factory->IsSupportedDecoder({"isac", 32000, 1})); - EXPECT_EQ(nullptr, - factory->MakeAudioDecoder({"isac", 8000, 1}, absl::nullopt)); - auto dec1 = factory->MakeAudioDecoder({"isac", 16000, 1}, absl::nullopt); - ASSERT_NE(nullptr, dec1); - EXPECT_EQ(16000, dec1->SampleRateHz()); - auto dec2 = factory->MakeAudioDecoder({"isac", 32000, 1}, absl::nullopt); - ASSERT_NE(nullptr, dec2); - EXPECT_EQ(32000, dec2->SampleRateHz()); -} - TEST(AudioDecoderFactoryTemplateTest, L16) { auto factory = CreateAudioDecoderFactory(); EXPECT_THAT( diff --git a/api/audio_codecs/test/audio_encoder_factory_template_unittest.cc b/api/audio_codecs/test/audio_encoder_factory_template_unittest.cc index 67b6883583..dbba387724 100644 --- a/api/audio_codecs/test/audio_encoder_factory_template_unittest.cc +++ b/api/audio_codecs/test/audio_encoder_factory_template_unittest.cc @@ -16,8 +16,6 @@ #include "api/audio_codecs/g711/audio_encoder_g711.h" #include "api/audio_codecs/g722/audio_encoder_g722.h" #include "api/audio_codecs/ilbc/audio_encoder_ilbc.h" -#include "api/audio_codecs/isac/audio_encoder_isac_fix.h" -#include "api/audio_codecs/isac/audio_encoder_isac_float.h" #include "api/audio_codecs/opus/audio_encoder_opus.h" #include "test/gmock.h" #include "test/gtest.h" @@ -180,49 +178,6 @@ TEST(AudioEncoderFactoryTemplateTest, Ilbc) { EXPECT_EQ(8000, enc->SampleRateHz()); } -TEST(AudioEncoderFactoryTemplateTest, IsacFix) { - auto factory = CreateAudioEncoderFactory(); - EXPECT_THAT(factory->GetSupportedEncoders(), - ::testing::ElementsAre(AudioCodecSpec{ - {"ISAC", 16000, 1}, {16000, 1, 32000, 10000, 32000}})); - EXPECT_EQ(absl::nullopt, factory->QueryAudioEncoder({"isac", 16000, 2})); - EXPECT_EQ(AudioCodecInfo(16000, 1, 32000, 10000, 32000), - factory->QueryAudioEncoder({"isac", 16000, 1})); - EXPECT_EQ(absl::nullopt, factory->QueryAudioEncoder({"isac", 32000, 1})); - EXPECT_EQ(nullptr, - factory->MakeAudioEncoder(17, {"isac", 8000, 1}, absl::nullopt)); - auto enc1 = factory->MakeAudioEncoder(17, {"isac", 16000, 1}, absl::nullopt); - ASSERT_NE(nullptr, enc1); - EXPECT_EQ(16000, enc1->SampleRateHz()); - EXPECT_EQ(3u, enc1->Num10MsFramesInNextPacket()); - auto enc2 = factory->MakeAudioEncoder( - 17, {"isac", 16000, 1, {{"ptime", "60"}}}, absl::nullopt); - ASSERT_NE(nullptr, enc2); - EXPECT_EQ(6u, enc2->Num10MsFramesInNextPacket()); -} - -TEST(AudioEncoderFactoryTemplateTest, IsacFloat) { - auto factory = CreateAudioEncoderFactory(); - EXPECT_THAT( - factory->GetSupportedEncoders(), - ::testing::ElementsAre( - AudioCodecSpec{{"ISAC", 16000, 1}, {16000, 1, 32000, 10000, 32000}}, - AudioCodecSpec{{"ISAC", 32000, 1}, {32000, 1, 56000, 10000, 56000}})); - EXPECT_EQ(absl::nullopt, factory->QueryAudioEncoder({"isac", 16000, 2})); - EXPECT_EQ(AudioCodecInfo(16000, 1, 32000, 10000, 32000), - factory->QueryAudioEncoder({"isac", 16000, 1})); - EXPECT_EQ(AudioCodecInfo(32000, 1, 56000, 10000, 56000), - factory->QueryAudioEncoder({"isac", 32000, 1})); - EXPECT_EQ(nullptr, - factory->MakeAudioEncoder(17, {"isac", 8000, 1}, absl::nullopt)); - auto enc1 = factory->MakeAudioEncoder(17, {"isac", 16000, 1}, absl::nullopt); - ASSERT_NE(nullptr, enc1); - EXPECT_EQ(16000, enc1->SampleRateHz()); - auto enc2 = factory->MakeAudioEncoder(17, {"isac", 32000, 1}, absl::nullopt); - ASSERT_NE(nullptr, enc2); - EXPECT_EQ(32000, enc2->SampleRateHz()); -} - TEST(AudioEncoderFactoryTemplateTest, L16) { auto factory = CreateAudioEncoderFactory(); EXPECT_THAT( diff --git a/media/engine/webrtc_voice_engine_unittest.cc b/media/engine/webrtc_voice_engine_unittest.cc index 0ef53bd68e..1b4445718e 100644 --- a/media/engine/webrtc_voice_engine_unittest.cc +++ b/media/engine/webrtc_voice_engine_unittest.cc @@ -53,7 +53,6 @@ using webrtc::BitrateConstraints; constexpr uint32_t kMaxUnsignaledRecvStreams = 4; const cricket::AudioCodec kPcmuCodec(0, "PCMU", 8000, 64000, 1); -const cricket::AudioCodec kIsacCodec(103, "ISAC", 16000, 32000, 1); const cricket::AudioCodec kOpusCodec(111, "opus", 48000, 32000, 2); const cricket::AudioCodec kG722CodecVoE(9, "G722", 16000, 64000, 1); const cricket::AudioCodec kG722CodecSdp(9, "G722", 8000, 64000, 1); @@ -852,7 +851,7 @@ TEST_P(WebRtcVoiceEngineTestFake, OpusSupportsTransportCc) { TEST_P(WebRtcVoiceEngineTestFake, SetRecvCodecs) { EXPECT_TRUE(SetupChannel()); cricket::AudioRecvParameters parameters; - parameters.codecs.push_back(kIsacCodec); + parameters.codecs.push_back(kOpusCodec); parameters.codecs.push_back(kPcmuCodec); parameters.codecs.push_back(kTelephoneEventCodec1); parameters.codecs.push_back(kTelephoneEventCodec2); @@ -863,7 +862,7 @@ TEST_P(WebRtcVoiceEngineTestFake, SetRecvCodecs) { EXPECT_THAT(GetRecvStreamConfig(kSsrcX).decoder_map, (ContainerEq>( {{0, {"PCMU", 8000, 1}}, - {106, {"ISAC", 16000, 1}}, + {106, {"OPUS", 48000, 2}}, {126, {"telephone-event", 8000, 1}}, {107, {"telephone-event", 32000, 1}}}))); } @@ -872,7 +871,7 @@ TEST_P(WebRtcVoiceEngineTestFake, SetRecvCodecs) { TEST_P(WebRtcVoiceEngineTestFake, SetRecvCodecsUnsupportedCodec) { EXPECT_TRUE(SetupChannel()); cricket::AudioRecvParameters parameters; - parameters.codecs.push_back(kIsacCodec); + parameters.codecs.push_back(kOpusCodec); parameters.codecs.push_back(cricket::AudioCodec(127, "XYZ", 32000, 0, 1)); EXPECT_FALSE(channel_->SetRecvParameters(parameters)); } @@ -881,9 +880,9 @@ TEST_P(WebRtcVoiceEngineTestFake, SetRecvCodecsUnsupportedCodec) { TEST_P(WebRtcVoiceEngineTestFake, SetRecvCodecsDuplicatePayloadType) { EXPECT_TRUE(SetupChannel()); cricket::AudioRecvParameters parameters; - parameters.codecs.push_back(kIsacCodec); + parameters.codecs.push_back(kOpusCodec); parameters.codecs.push_back(kCn16000Codec); - parameters.codecs[1].id = kIsacCodec.id; + parameters.codecs[1].id = kOpusCodec.id; EXPECT_FALSE(channel_->SetRecvParameters(parameters)); } @@ -891,32 +890,27 @@ TEST_P(WebRtcVoiceEngineTestFake, SetRecvCodecsDuplicatePayloadType) { TEST_P(WebRtcVoiceEngineTestFake, SetRecvCodecsWithOpusNoStereo) { EXPECT_TRUE(SetupChannel()); cricket::AudioRecvParameters parameters; - parameters.codecs.push_back(kIsacCodec); parameters.codecs.push_back(kPcmuCodec); parameters.codecs.push_back(kOpusCodec); EXPECT_TRUE(channel_->SetRecvParameters(parameters)); EXPECT_TRUE(AddRecvStream(kSsrcX)); EXPECT_THAT(GetRecvStreamConfig(kSsrcX).decoder_map, (ContainerEq>( - {{0, {"PCMU", 8000, 1}}, - {103, {"ISAC", 16000, 1}}, - {111, {"opus", 48000, 2}}}))); + {{0, {"PCMU", 8000, 1}}, {111, {"opus", 48000, 2}}}))); } // Test that we can decode OPUS with stereo = 0. TEST_P(WebRtcVoiceEngineTestFake, SetRecvCodecsWithOpus0Stereo) { EXPECT_TRUE(SetupChannel()); cricket::AudioRecvParameters parameters; - parameters.codecs.push_back(kIsacCodec); parameters.codecs.push_back(kPcmuCodec); parameters.codecs.push_back(kOpusCodec); - parameters.codecs[2].params["stereo"] = "0"; + parameters.codecs[1].params["stereo"] = "0"; EXPECT_TRUE(channel_->SetRecvParameters(parameters)); EXPECT_TRUE(AddRecvStream(kSsrcX)); EXPECT_THAT(GetRecvStreamConfig(kSsrcX).decoder_map, (ContainerEq>( {{0, {"PCMU", 8000, 1}}, - {103, {"ISAC", 16000, 1}}, {111, {"opus", 48000, 2, {{"stereo", "0"}}}}}))); } @@ -924,16 +918,14 @@ TEST_P(WebRtcVoiceEngineTestFake, SetRecvCodecsWithOpus0Stereo) { TEST_P(WebRtcVoiceEngineTestFake, SetRecvCodecsWithOpus1Stereo) { EXPECT_TRUE(SetupChannel()); cricket::AudioRecvParameters parameters; - parameters.codecs.push_back(kIsacCodec); parameters.codecs.push_back(kPcmuCodec); parameters.codecs.push_back(kOpusCodec); - parameters.codecs[2].params["stereo"] = "1"; + parameters.codecs[1].params["stereo"] = "1"; EXPECT_TRUE(channel_->SetRecvParameters(parameters)); EXPECT_TRUE(AddRecvStream(kSsrcX)); EXPECT_THAT(GetRecvStreamConfig(kSsrcX).decoder_map, (ContainerEq>( {{0, {"PCMU", 8000, 1}}, - {103, {"ISAC", 16000, 1}}, {111, {"opus", 48000, 2, {{"stereo", "1"}}}}}))); } @@ -941,7 +933,7 @@ TEST_P(WebRtcVoiceEngineTestFake, SetRecvCodecsWithOpus1Stereo) { TEST_P(WebRtcVoiceEngineTestFake, SetRecvCodecsWithMultipleStreams) { EXPECT_TRUE(SetupChannel()); cricket::AudioRecvParameters parameters; - parameters.codecs.push_back(kIsacCodec); + parameters.codecs.push_back(kOpusCodec); parameters.codecs.push_back(kPcmuCodec); parameters.codecs.push_back(kTelephoneEventCodec1); parameters.codecs.push_back(kTelephoneEventCodec2); @@ -953,7 +945,7 @@ TEST_P(WebRtcVoiceEngineTestFake, SetRecvCodecsWithMultipleStreams) { EXPECT_THAT(GetRecvStreamConfig(ssrc).decoder_map, (ContainerEq>( {{0, {"PCMU", 8000, 1}}, - {106, {"ISAC", 16000, 1}}, + {106, {"OPUS", 48000, 2}}, {126, {"telephone-event", 8000, 1}}, {107, {"telephone-event", 32000, 1}}}))); } @@ -962,20 +954,20 @@ TEST_P(WebRtcVoiceEngineTestFake, SetRecvCodecsWithMultipleStreams) { TEST_P(WebRtcVoiceEngineTestFake, SetRecvCodecsAfterAddingStreams) { EXPECT_TRUE(SetupRecvStream()); cricket::AudioRecvParameters parameters; - parameters.codecs.push_back(kIsacCodec); + parameters.codecs.push_back(kOpusCodec); parameters.codecs[0].id = 106; // collide with existing CN 32k EXPECT_TRUE(channel_->SetRecvParameters(parameters)); const auto& dm = GetRecvStreamConfig(kSsrcX).decoder_map; ASSERT_EQ(1u, dm.count(106)); - EXPECT_EQ(webrtc::SdpAudioFormat("isac", 16000, 1), dm.at(106)); + EXPECT_EQ(webrtc::SdpAudioFormat("opus", 48000, 2), dm.at(106)); } // Test that we can apply the same set of codecs again while playing. TEST_P(WebRtcVoiceEngineTestFake, SetRecvCodecsWhilePlaying) { EXPECT_TRUE(SetupRecvStream()); cricket::AudioRecvParameters parameters; - parameters.codecs.push_back(kIsacCodec); + parameters.codecs.push_back(kPcmuCodec); parameters.codecs.push_back(kCn16000Codec); EXPECT_TRUE(channel_->SetRecvParameters(parameters)); channel_->SetPlayout(true); @@ -983,7 +975,7 @@ TEST_P(WebRtcVoiceEngineTestFake, SetRecvCodecsWhilePlaying) { // Remapping a payload type to a different codec should fail. parameters.codecs[0] = kOpusCodec; - parameters.codecs[0].id = kIsacCodec.id; + parameters.codecs[0].id = kPcmuCodec.id; EXPECT_FALSE(channel_->SetRecvParameters(parameters)); EXPECT_TRUE(GetRecvStream(kSsrcX).started()); } @@ -992,7 +984,7 @@ TEST_P(WebRtcVoiceEngineTestFake, SetRecvCodecsWhilePlaying) { TEST_P(WebRtcVoiceEngineTestFake, AddRecvCodecsWhilePlaying) { EXPECT_TRUE(SetupRecvStream()); cricket::AudioRecvParameters parameters; - parameters.codecs.push_back(kIsacCodec); + parameters.codecs.push_back(kPcmuCodec); parameters.codecs.push_back(kCn16000Codec); EXPECT_TRUE(channel_->SetRecvParameters(parameters)); channel_->SetPlayout(true); @@ -1007,7 +999,7 @@ TEST_P(WebRtcVoiceEngineTestFake, AddRecvCodecsWhilePlaying) { TEST_P(WebRtcVoiceEngineTestFake, ChangeRecvCodecPayloadType) { EXPECT_TRUE(SetupRecvStream()); cricket::AudioRecvParameters parameters; - parameters.codecs.push_back(kIsacCodec); + parameters.codecs.push_back(kOpusCodec); EXPECT_TRUE(channel_->SetRecvParameters(parameters)); ++parameters.codecs[0].id; @@ -1035,9 +1027,6 @@ TEST_P(WebRtcVoiceEngineTestFake, SetSendBandwidthAuto) { // value. autobw is enabled for the following tests because the target // bitrate is <= 0. - // ISAC, default bitrate == 32000. - TestMaxSendBandwidth(kIsacCodec, 0, true, 32000); - // PCMU, default bitrate == 64000. TestMaxSendBandwidth(kPcmuCodec, -1, true, 64000); @@ -1048,11 +1037,6 @@ TEST_P(WebRtcVoiceEngineTestFake, SetSendBandwidthAuto) { TEST_P(WebRtcVoiceEngineTestFake, SetMaxSendBandwidthMultiRateAsCaller) { EXPECT_TRUE(SetupSendStream()); - // ISAC, default bitrate == 32000. - TestMaxSendBandwidth(kIsacCodec, 16000, true, 16000); - // Rates above the max (56000) should be capped. - TestMaxSendBandwidth(kIsacCodec, 100000, true, 32000); - // opus, default bitrate == 64000. TestMaxSendBandwidth(kOpusCodec, 96000, true, 96000); TestMaxSendBandwidth(kOpusCodec, 48000, true, 48000); @@ -1264,13 +1248,13 @@ TEST_P(WebRtcVoiceEngineTestFake, RtpParametersArePerStream) { TEST_P(WebRtcVoiceEngineTestFake, GetRtpSendParametersCodecs) { EXPECT_TRUE(SetupSendStream()); cricket::AudioSendParameters parameters; - parameters.codecs.push_back(kIsacCodec); + parameters.codecs.push_back(kOpusCodec); parameters.codecs.push_back(kPcmuCodec); SetSendParameters(parameters); webrtc::RtpParameters rtp_parameters = channel_->GetRtpSendParameters(kSsrcX); ASSERT_EQ(2u, rtp_parameters.codecs.size()); - EXPECT_EQ(kIsacCodec.ToCodecParameters(), rtp_parameters.codecs[0]); + EXPECT_EQ(kOpusCodec.ToCodecParameters(), rtp_parameters.codecs[0]); EXPECT_EQ(kPcmuCodec.ToCodecParameters(), rtp_parameters.codecs[1]); } @@ -1310,7 +1294,7 @@ TEST_P(WebRtcVoiceEngineTestFake, GetRtpSendParametersSsrc) { TEST_P(WebRtcVoiceEngineTestFake, SetAndGetRtpSendParameters) { EXPECT_TRUE(SetupSendStream()); cricket::AudioSendParameters parameters; - parameters.codecs.push_back(kIsacCodec); + parameters.codecs.push_back(kOpusCodec); parameters.codecs.push_back(kPcmuCodec); SetSendParameters(parameters); @@ -1384,14 +1368,14 @@ TEST_P(WebRtcVoiceEngineTestFake, SetRtpSendParameterUpdatesBitratePriority) { TEST_P(WebRtcVoiceEngineTestFake, GetRtpReceiveParametersCodecs) { EXPECT_TRUE(SetupRecvStream()); cricket::AudioRecvParameters parameters; - parameters.codecs.push_back(kIsacCodec); + parameters.codecs.push_back(kOpusCodec); parameters.codecs.push_back(kPcmuCodec); EXPECT_TRUE(channel_->SetRecvParameters(parameters)); webrtc::RtpParameters rtp_parameters = channel_->GetRtpReceiveParameters(kSsrcX); ASSERT_EQ(2u, rtp_parameters.codecs.size()); - EXPECT_EQ(kIsacCodec.ToCodecParameters(), rtp_parameters.codecs[0]); + EXPECT_EQ(kOpusCodec.ToCodecParameters(), rtp_parameters.codecs[0]); EXPECT_EQ(kPcmuCodec.ToCodecParameters(), rtp_parameters.codecs[1]); } @@ -1408,7 +1392,7 @@ TEST_P(WebRtcVoiceEngineTestFake, GetRtpReceiveParametersSsrc) { TEST_P(WebRtcVoiceEngineTestFake, SetAndGetRtpReceiveParameters) { EXPECT_TRUE(SetupRecvStream()); cricket::AudioRecvParameters parameters; - parameters.codecs.push_back(kIsacCodec); + parameters.codecs.push_back(kOpusCodec); parameters.codecs.push_back(kPcmuCodec); EXPECT_TRUE(channel_->SetRecvParameters(parameters)); @@ -1430,7 +1414,7 @@ TEST_P(WebRtcVoiceEngineTestFake, GetRtpReceiveParametersWithUnsignaledSsrc) { // Call necessary methods to configure receiving a default stream as // soon as it arrives. cricket::AudioRecvParameters parameters; - parameters.codecs.push_back(kIsacCodec); + parameters.codecs.push_back(kOpusCodec); parameters.codecs.push_back(kPcmuCodec); EXPECT_TRUE(channel_->SetRecvParameters(parameters)); @@ -1462,7 +1446,7 @@ TEST_P(WebRtcVoiceEngineTestFake, GetRtpReceiveParametersWithUnsignaledSsrc) { TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecs) { EXPECT_TRUE(SetupSendStream()); cricket::AudioSendParameters parameters; - parameters.codecs.push_back(kIsacCodec); + parameters.codecs.push_back(kOpusCodec); parameters.codecs.push_back(kPcmuCodec); parameters.codecs.push_back(kCn8000Codec); parameters.codecs[0].id = 96; @@ -1471,7 +1455,7 @@ TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecs) { const auto& send_codec_spec = *GetSendStreamConfig(kSsrcX).send_codec_spec; EXPECT_EQ(96, send_codec_spec.payload_type); EXPECT_EQ(22000, send_codec_spec.target_bitrate_bps); - EXPECT_STRCASEEQ("ISAC", send_codec_spec.format.name.c_str()); + EXPECT_STRCASEEQ("OPUS", send_codec_spec.format.name.c_str()); EXPECT_NE(send_codec_spec.format.clockrate_hz, 8000); EXPECT_EQ(absl::nullopt, send_codec_spec.cng_payload_type); EXPECT_FALSE(channel_->CanInsertDtmf()); @@ -1567,7 +1551,7 @@ TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsRedFmtpAmountOfRedundancy) { TEST_P(WebRtcVoiceEngineTestFake, DontRecreateSendStream) { EXPECT_TRUE(SetupSendStream()); cricket::AudioSendParameters parameters; - parameters.codecs.push_back(kIsacCodec); + parameters.codecs.push_back(kOpusCodec); parameters.codecs.push_back(kPcmuCodec); parameters.codecs.push_back(kCn8000Codec); parameters.codecs[0].id = 96; @@ -1840,7 +1824,7 @@ TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecDisableNackRecvStreams) { TEST_P(WebRtcVoiceEngineTestFake, AddRecvStreamEnableNack) { EXPECT_TRUE(SetupSendStream()); cricket::AudioSendParameters parameters; - parameters.codecs.push_back(kIsacCodec); + parameters.codecs.push_back(kOpusCodec); parameters.codecs.push_back(kCn16000Codec); parameters.codecs[0].AddFeedbackParam(cricket::FeedbackParam( cricket::kRtcpFbParamNack, cricket::kParamValueEmpty)); @@ -1860,7 +1844,7 @@ TEST_P(WebRtcVoiceEngineTestFake, TransportCcCanBeEnabledAndDisabled) { SetSendParameters(send_parameters); cricket::AudioRecvParameters recv_parameters; - recv_parameters.codecs.push_back(kIsacCodec); + recv_parameters.codecs.push_back(kOpusCodec); EXPECT_TRUE(channel_->SetRecvParameters(recv_parameters)); EXPECT_TRUE(AddRecvStream(kSsrcX)); ASSERT_TRUE(call_.GetAudioReceiveStream(kSsrcX) != nullptr); @@ -1872,8 +1856,8 @@ TEST_P(WebRtcVoiceEngineTestFake, TransportCcCanBeEnabledAndDisabled) { EXPECT_TRUE(call_.GetAudioReceiveStream(kSsrcX)->transport_cc()); } -// Test that we can switch back and forth between Opus and ISAC with CN. -TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsIsacOpusSwitching) { +// Test that we can switch back and forth between Opus and PCMU with CN. +TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsOpusPcmuSwitching) { EXPECT_TRUE(SetupSendStream()); cricket::AudioSendParameters opus_parameters; @@ -1885,15 +1869,15 @@ TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsIsacOpusSwitching) { EXPECT_STRCASEEQ("opus", spec.format.name.c_str()); } - cricket::AudioSendParameters isac_parameters; - isac_parameters.codecs.push_back(kIsacCodec); - isac_parameters.codecs.push_back(kCn16000Codec); - isac_parameters.codecs.push_back(kOpusCodec); - SetSendParameters(isac_parameters); + cricket::AudioSendParameters pcmu_parameters; + pcmu_parameters.codecs.push_back(kPcmuCodec); + pcmu_parameters.codecs.push_back(kCn16000Codec); + pcmu_parameters.codecs.push_back(kOpusCodec); + SetSendParameters(pcmu_parameters); { const auto& spec = *GetSendStreamConfig(kSsrcX).send_codec_spec; - EXPECT_EQ(103, spec.payload_type); - EXPECT_STRCASEEQ("ISAC", spec.format.name.c_str()); + EXPECT_EQ(0, spec.payload_type); + EXPECT_STRCASEEQ("PCMU", spec.format.name.c_str()); } SetSendParameters(opus_parameters); @@ -1908,33 +1892,7 @@ TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsIsacOpusSwitching) { TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsBitrate) { EXPECT_TRUE(SetupSendStream()); cricket::AudioSendParameters parameters; - parameters.codecs.push_back(kIsacCodec); // bitrate == 32000 - SetSendParameters(parameters); - { - const auto& spec = *GetSendStreamConfig(kSsrcX).send_codec_spec; - EXPECT_EQ(103, spec.payload_type); - EXPECT_STRCASEEQ("ISAC", spec.format.name.c_str()); - EXPECT_EQ(32000, spec.target_bitrate_bps); - } - - parameters.codecs[0].bitrate = 0; // bitrate == default - SetSendParameters(parameters); - { - const auto& spec = *GetSendStreamConfig(kSsrcX).send_codec_spec; - EXPECT_EQ(103, spec.payload_type); - EXPECT_STRCASEEQ("ISAC", spec.format.name.c_str()); - EXPECT_EQ(32000, spec.target_bitrate_bps); - } - parameters.codecs[0].bitrate = 28000; // bitrate == 28000 - SetSendParameters(parameters); - { - const auto& spec = *GetSendStreamConfig(kSsrcX).send_codec_spec; - EXPECT_EQ(103, spec.payload_type); - EXPECT_STRCASEEQ("ISAC", spec.format.name.c_str()); - EXPECT_EQ(28000, spec.target_bitrate_bps); - } - - parameters.codecs[0] = kPcmuCodec; // bitrate == 64000 + parameters.codecs.push_back(kPcmuCodec); SetSendParameters(parameters); { const auto& spec = *GetSendStreamConfig(kSsrcX).send_codec_spec; @@ -1976,14 +1934,14 @@ TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsDTMFOnTop) { EXPECT_TRUE(SetupSendStream()); cricket::AudioSendParameters parameters; parameters.codecs.push_back(kTelephoneEventCodec1); - parameters.codecs.push_back(kIsacCodec); + parameters.codecs.push_back(kOpusCodec); parameters.codecs.push_back(kPcmuCodec); parameters.codecs[0].id = 98; // DTMF parameters.codecs[1].id = 96; SetSendParameters(parameters); const auto& spec = *GetSendStreamConfig(kSsrcX).send_codec_spec; EXPECT_EQ(96, spec.payload_type); - EXPECT_STRCASEEQ("ISAC", spec.format.name.c_str()); + EXPECT_STRCASEEQ("OPUS", spec.format.name.c_str()); SetSend(true); EXPECT_TRUE(channel_->CanInsertDtmf()); } @@ -2009,7 +1967,7 @@ TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsDTMFPayloadTypeOutOfRange) { EXPECT_TRUE(SetupSendStream()); cricket::AudioSendParameters parameters; parameters.codecs.push_back(kTelephoneEventCodec2); - parameters.codecs.push_back(kIsacCodec); + parameters.codecs.push_back(kOpusCodec); parameters.codecs[0].id = 0; // DTMF parameters.codecs[1].id = 96; SetSendParameters(parameters); @@ -2031,15 +1989,13 @@ TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsDTMFPayloadTypeOutOfRange) { TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsCNOnTop) { EXPECT_TRUE(SetupSendStream()); cricket::AudioSendParameters parameters; - parameters.codecs.push_back(kCn16000Codec); - parameters.codecs.push_back(kIsacCodec); + parameters.codecs.push_back(kCn8000Codec); parameters.codecs.push_back(kPcmuCodec); - parameters.codecs[0].id = 98; // wideband CN - parameters.codecs[1].id = 96; + parameters.codecs[0].id = 98; // narrowband CN SetSendParameters(parameters); const auto& send_codec_spec = *GetSendStreamConfig(kSsrcX).send_codec_spec; - EXPECT_EQ(96, send_codec_spec.payload_type); - EXPECT_STRCASEEQ("ISAC", send_codec_spec.format.name.c_str()); + EXPECT_EQ(0, send_codec_spec.payload_type); + EXPECT_STRCASEEQ("PCMU", send_codec_spec.format.name.c_str()); EXPECT_EQ(98, send_codec_spec.cng_payload_type); } @@ -2047,19 +2003,17 @@ TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsCNOnTop) { TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsCNandDTMFAsCaller) { EXPECT_TRUE(SetupSendStream()); cricket::AudioSendParameters parameters; - parameters.codecs.push_back(kIsacCodec); parameters.codecs.push_back(kPcmuCodec); - // TODO(juberti): cn 32000 parameters.codecs.push_back(kCn16000Codec); parameters.codecs.push_back(kCn8000Codec); parameters.codecs.push_back(kTelephoneEventCodec1); parameters.codecs[0].id = 96; - parameters.codecs[2].id = 97; // wideband CN - parameters.codecs[4].id = 98; // DTMF + parameters.codecs[2].id = 97; // narrowband CN + parameters.codecs[3].id = 98; // DTMF SetSendParameters(parameters); const auto& send_codec_spec = *GetSendStreamConfig(kSsrcX).send_codec_spec; EXPECT_EQ(96, send_codec_spec.payload_type); - EXPECT_STRCASEEQ("ISAC", send_codec_spec.format.name.c_str()); + EXPECT_STRCASEEQ("PCMU", send_codec_spec.format.name.c_str()); EXPECT_EQ(1u, send_codec_spec.format.num_channels); EXPECT_EQ(97, send_codec_spec.cng_payload_type); SetSend(true); @@ -2070,22 +2024,20 @@ TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsCNandDTMFAsCaller) { TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsCNandDTMFAsCallee) { EXPECT_TRUE(SetupChannel()); cricket::AudioSendParameters parameters; - parameters.codecs.push_back(kIsacCodec); parameters.codecs.push_back(kPcmuCodec); - // TODO(juberti): cn 32000 parameters.codecs.push_back(kCn16000Codec); parameters.codecs.push_back(kCn8000Codec); parameters.codecs.push_back(kTelephoneEventCodec2); parameters.codecs[0].id = 96; - parameters.codecs[2].id = 97; // wideband CN - parameters.codecs[4].id = 98; // DTMF + parameters.codecs[2].id = 97; // narrowband CN + parameters.codecs[3].id = 98; // DTMF SetSendParameters(parameters); EXPECT_TRUE( channel_->AddSendStream(cricket::StreamParams::CreateLegacy(kSsrcX))); const auto& send_codec_spec = *GetSendStreamConfig(kSsrcX).send_codec_spec; EXPECT_EQ(96, send_codec_spec.payload_type); - EXPECT_STRCASEEQ("ISAC", send_codec_spec.format.name.c_str()); + EXPECT_STRCASEEQ("PCMU", send_codec_spec.format.name.c_str()); EXPECT_EQ(1u, send_codec_spec.format.num_channels); EXPECT_EQ(97, send_codec_spec.cng_payload_type); SetSend(true); @@ -2097,20 +2049,11 @@ TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsCNandDTMFAsCallee) { TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsCNNoMatch) { EXPECT_TRUE(SetupSendStream()); cricket::AudioSendParameters parameters; - // Set ISAC(16K) and CN(16K). VAD should be activated. - parameters.codecs.push_back(kIsacCodec); + // Set PCMU(8K) and CN(16K). VAD should not be activated. + parameters.codecs.push_back(kPcmuCodec); parameters.codecs.push_back(kCn16000Codec); parameters.codecs[1].id = 97; SetSendParameters(parameters); - { - const auto& send_codec_spec = *GetSendStreamConfig(kSsrcX).send_codec_spec; - EXPECT_STRCASEEQ("ISAC", send_codec_spec.format.name.c_str()); - EXPECT_EQ(1u, send_codec_spec.format.num_channels); - EXPECT_EQ(97, send_codec_spec.cng_payload_type); - } - // Set PCMU(8K) and CN(16K). VAD should not be activated. - parameters.codecs[0] = kPcmuCodec; - SetSendParameters(parameters); { const auto& send_codec_spec = *GetSendStreamConfig(kSsrcX).send_codec_spec; EXPECT_STRCASEEQ("PCMU", send_codec_spec.format.name.c_str()); @@ -2125,12 +2068,12 @@ TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsCNNoMatch) { EXPECT_EQ(1u, send_codec_spec.format.num_channels); EXPECT_EQ(13, send_codec_spec.cng_payload_type); } - // Set ISAC(16K) and CN(8K). VAD should not be activated. - parameters.codecs[0] = kIsacCodec; + // Set OPUS(48K) and CN(8K). VAD should not be activated. + parameters.codecs[0] = kOpusCodec; SetSendParameters(parameters); { const auto& send_codec_spec = *GetSendStreamConfig(kSsrcX).send_codec_spec; - EXPECT_STRCASEEQ("ISAC", send_codec_spec.format.name.c_str()); + EXPECT_STRCASEEQ("OPUS", send_codec_spec.format.name.c_str()); EXPECT_EQ(absl::nullopt, send_codec_spec.cng_payload_type); } } @@ -2139,19 +2082,18 @@ TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsCNNoMatch) { TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsCaseInsensitive) { EXPECT_TRUE(SetupSendStream()); cricket::AudioSendParameters parameters; - parameters.codecs.push_back(kIsacCodec); parameters.codecs.push_back(kPcmuCodec); parameters.codecs.push_back(kCn16000Codec); parameters.codecs.push_back(kCn8000Codec); parameters.codecs.push_back(kTelephoneEventCodec1); - parameters.codecs[0].name = "iSaC"; + parameters.codecs[0].name = "PcMu"; parameters.codecs[0].id = 96; - parameters.codecs[2].id = 97; // wideband CN - parameters.codecs[4].id = 98; // DTMF + parameters.codecs[2].id = 97; // narrowband CN + parameters.codecs[3].id = 98; // DTMF SetSendParameters(parameters); const auto& send_codec_spec = *GetSendStreamConfig(kSsrcX).send_codec_spec; EXPECT_EQ(96, send_codec_spec.payload_type); - EXPECT_STRCASEEQ("ISAC", send_codec_spec.format.name.c_str()); + EXPECT_STRCASEEQ("PCMU", send_codec_spec.format.name.c_str()); EXPECT_EQ(1u, send_codec_spec.format.num_channels); EXPECT_EQ(97, send_codec_spec.cng_payload_type); SetSend(true); @@ -2292,24 +2234,25 @@ TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsWithMultipleSendStreams) { } cricket::AudioSendParameters parameters; - // Set ISAC(16K) and CN(16K). VAD should be activated. - parameters.codecs.push_back(kIsacCodec); - parameters.codecs.push_back(kCn16000Codec); + // Set PCMU and CN(8K). VAD should be activated. + parameters.codecs.push_back(kPcmuCodec); + parameters.codecs.push_back(kCn8000Codec); parameters.codecs[1].id = 97; SetSendParameters(parameters); - // Verify ISAC and VAD are corrected configured on all send channels. + // Verify PCMU and VAD are corrected configured on all send channels. for (uint32_t ssrc : kSsrcs4) { ASSERT_TRUE(call_.GetAudioSendStream(ssrc) != nullptr); const auto& send_codec_spec = *call_.GetAudioSendStream(ssrc)->GetConfig().send_codec_spec; - EXPECT_STRCASEEQ("ISAC", send_codec_spec.format.name.c_str()); + EXPECT_STRCASEEQ("PCMU", send_codec_spec.format.name.c_str()); EXPECT_EQ(1u, send_codec_spec.format.num_channels); EXPECT_EQ(97, send_codec_spec.cng_payload_type); } // Change to PCMU(8K) and CN(16K). parameters.codecs[0] = kPcmuCodec; + parameters.codecs[1] = kCn16000Codec; SetSendParameters(parameters); for (uint32_t ssrc : kSsrcs4) { ASSERT_TRUE(call_.GetAudioSendStream(ssrc) != nullptr); @@ -2859,13 +2802,13 @@ TEST_P(WebRtcVoiceEngineTestFake, AddRecvStream) { TEST_P(WebRtcVoiceEngineTestFake, AddRecvStreamUnsupportedCodec) { EXPECT_TRUE(SetupSendStream()); cricket::AudioRecvParameters parameters; - parameters.codecs.push_back(kIsacCodec); + parameters.codecs.push_back(kOpusCodec); parameters.codecs.push_back(kPcmuCodec); EXPECT_TRUE(channel_->SetRecvParameters(parameters)); EXPECT_TRUE(AddRecvStream(kSsrcX)); EXPECT_THAT(GetRecvStreamConfig(kSsrcX).decoder_map, (ContainerEq>( - {{0, {"PCMU", 8000, 1}}, {103, {"ISAC", 16000, 1}}}))); + {{0, {"PCMU", 8000, 1}}, {111, {"OPUS", 48000, 2}}}))); } // Test that we properly clean up any streams that were added, even if @@ -3666,7 +3609,7 @@ TEST(WebRtcVoiceEngineTest, StartupShutdownWithExternalADM) { } } -// Verify the payload id of common audio codecs, including CN, ISAC, and G722. +// Verify the payload id of common audio codecs, including CN and G722. TEST(WebRtcVoiceEngineTest, HasCorrectPayloadTypeMapping) { for (bool use_null_apm : {false, true}) { std::unique_ptr task_queue_factory = @@ -3693,10 +3636,6 @@ TEST(WebRtcVoiceEngineTest, HasCorrectPayloadTypeMapping) { EXPECT_EQ(105, codec.id); } else if (is_codec("CN", 32000)) { EXPECT_EQ(106, codec.id); - } else if (is_codec("ISAC", 16000)) { - EXPECT_EQ(103, codec.id); - } else if (is_codec("ISAC", 32000)) { - EXPECT_EQ(104, codec.id); } else if (is_codec("G722", 8000)) { EXPECT_EQ(9, codec.id); } else if (is_codec("telephone-event", 8000)) { diff --git a/modules/audio_coding/BUILD.gn b/modules/audio_coding/BUILD.gn index 0d67cfa3a6..65c849d7eb 100644 --- a/modules/audio_coding/BUILD.gn +++ b/modules/audio_coding/BUILD.gn @@ -381,50 +381,8 @@ rtc_library("ilbc_c") { absl_deps = [ "//third_party/abseil-cpp/absl/base:core_headers" ] } -rtc_source_set("isac_common") { - poisonous = [ "audio_codecs" ] - sources = [ - "codecs/isac/audio_decoder_isac_t.h", - "codecs/isac/audio_decoder_isac_t_impl.h", - "codecs/isac/audio_encoder_isac_t.h", - "codecs/isac/audio_encoder_isac_t_impl.h", - ] - deps = [ - ":isac_bwinfo", - "../../api:scoped_refptr", - "../../api/audio_codecs:audio_codecs_api", - "../../api/units:time_delta", - "../../rtc_base:checks", - "../../rtc_base:safe_minmax", - "../../system_wrappers:field_trial", - ] - absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ] -} - -rtc_library("isac") { - visibility += [ "*" ] - poisonous = [ "audio_codecs" ] - sources = [ - "codecs/isac/main/include/audio_decoder_isac.h", - "codecs/isac/main/include/audio_encoder_isac.h", - "codecs/isac/main/source/audio_decoder_isac.cc", - "codecs/isac/main/source/audio_encoder_isac.cc", - ] - - deps = [ - ":isac_common", - "../../api/audio_codecs:audio_codecs_api", - ] - public_deps = [ ":isac_c" ] # no-presubmit-check TODO(webrtc:8603) -} - -rtc_source_set("isac_bwinfo") { - sources = [ "codecs/isac/bandwidth_info.h" ] - deps = [] -} - rtc_library("isac_vad") { - visibility += webrtc_default_visibility + visibility += [ "../audio_processing/vad:*" ] sources = [ "codecs/isac/main/source/filter_functions.c", "codecs/isac/main/source/filter_functions.h", @@ -447,247 +405,9 @@ rtc_library("isac_vad") { ] } -rtc_library("isac_c") { - poisonous = [ "audio_codecs" ] - sources = [ - "codecs/isac/main/include/isac.h", - "codecs/isac/main/source/arith_routines.c", - "codecs/isac/main/source/arith_routines.h", - "codecs/isac/main/source/arith_routines_hist.c", - "codecs/isac/main/source/arith_routines_logist.c", - "codecs/isac/main/source/bandwidth_estimator.c", - "codecs/isac/main/source/bandwidth_estimator.h", - "codecs/isac/main/source/codec.h", - "codecs/isac/main/source/crc.c", - "codecs/isac/main/source/crc.h", - "codecs/isac/main/source/decode.c", - "codecs/isac/main/source/decode_bwe.c", - "codecs/isac/main/source/encode.c", - "codecs/isac/main/source/encode_lpc_swb.c", - "codecs/isac/main/source/encode_lpc_swb.h", - "codecs/isac/main/source/entropy_coding.c", - "codecs/isac/main/source/entropy_coding.h", - "codecs/isac/main/source/filterbanks.c", - "codecs/isac/main/source/intialize.c", - "codecs/isac/main/source/isac.c", - "codecs/isac/main/source/isac_float_type.h", - "codecs/isac/main/source/lattice.c", - "codecs/isac/main/source/lpc_analysis.c", - "codecs/isac/main/source/lpc_analysis.h", - "codecs/isac/main/source/lpc_gain_swb_tables.c", - "codecs/isac/main/source/lpc_gain_swb_tables.h", - "codecs/isac/main/source/lpc_shape_swb12_tables.c", - "codecs/isac/main/source/lpc_shape_swb12_tables.h", - "codecs/isac/main/source/lpc_shape_swb16_tables.c", - "codecs/isac/main/source/lpc_shape_swb16_tables.h", - "codecs/isac/main/source/lpc_tables.c", - "codecs/isac/main/source/lpc_tables.h", - "codecs/isac/main/source/pitch_gain_tables.c", - "codecs/isac/main/source/pitch_gain_tables.h", - "codecs/isac/main/source/pitch_lag_tables.c", - "codecs/isac/main/source/pitch_lag_tables.h", - "codecs/isac/main/source/spectrum_ar_model_tables.c", - "codecs/isac/main/source/spectrum_ar_model_tables.h", - "codecs/isac/main/source/transform.c", - ] - - if (is_linux || is_chromeos) { - libs = [ "m" ] - } - - deps = [ - ":isac_bwinfo", - ":isac_vad", - "../../common_audio", - "../../common_audio:common_audio_c", - "../../rtc_base:checks", - "../../rtc_base:compile_assert_c", - "../../rtc_base/system:arch", - "../third_party/fft", - ] -} - -rtc_library("isac_fix") { - visibility += [ "*" ] - poisonous = [ "audio_codecs" ] - sources = [ - "codecs/isac/fix/source/audio_decoder_isacfix.cc", - "codecs/isac/fix/source/audio_encoder_isacfix.cc", - ] - - deps = [ - ":isac_common", - "../../api/audio_codecs:audio_codecs_api", - "../../common_audio", - "../../system_wrappers", - ] - public_deps = [ ":isac_fix_c" ] # no-presubmit-check TODO(webrtc:8603) - - if (rtc_build_with_neon) { - deps += [ ":isac_neon" ] - } -} - -rtc_library("isac_fix_common") { - poisonous = [ "audio_codecs" ] - sources = [ - "codecs/isac/fix/source/codec.h", - "codecs/isac/fix/source/entropy_coding.h", - "codecs/isac/fix/source/fft.c", - "codecs/isac/fix/source/fft.h", - "codecs/isac/fix/source/filterbank_internal.h", - "codecs/isac/fix/source/settings.h", - "codecs/isac/fix/source/structs.h", - "codecs/isac/fix/source/transform_tables.c", - ] - deps = [ - ":isac_bwinfo", - "../../common_audio", - "../../common_audio:common_audio_c", - ] -} - -rtc_source_set("isac_fix_c_arm_asm") { - poisonous = [ "audio_codecs" ] - sources = [] - if (current_cpu == "arm" && arm_version >= 7) { - sources += [ - "codecs/isac/fix/source/lattice_armv7.S", - "codecs/isac/fix/source/pitch_filter_armv6.S", - ] - deps = [ - ":isac_fix_common", - "../../rtc_base/system:asm_defines", - ] - } -} - -rtc_library("isac_fix_c") { - poisonous = [ "audio_codecs" ] - sources = [ - "codecs/isac/fix/include/audio_decoder_isacfix.h", - "codecs/isac/fix/include/audio_encoder_isacfix.h", - "codecs/isac/fix/include/isacfix.h", - "codecs/isac/fix/source/arith_routines.c", - "codecs/isac/fix/source/arith_routines_hist.c", - "codecs/isac/fix/source/arith_routines_logist.c", - "codecs/isac/fix/source/arith_routins.h", - "codecs/isac/fix/source/bandwidth_estimator.c", - "codecs/isac/fix/source/bandwidth_estimator.h", - "codecs/isac/fix/source/decode.c", - "codecs/isac/fix/source/decode_bwe.c", - "codecs/isac/fix/source/decode_plc.c", - "codecs/isac/fix/source/encode.c", - "codecs/isac/fix/source/entropy_coding.c", - "codecs/isac/fix/source/filterbank_tables.c", - "codecs/isac/fix/source/filterbank_tables.h", - "codecs/isac/fix/source/filterbanks.c", - "codecs/isac/fix/source/filters.c", - "codecs/isac/fix/source/initialize.c", - "codecs/isac/fix/source/isac_fix_type.h", - "codecs/isac/fix/source/isacfix.c", - "codecs/isac/fix/source/lattice.c", - "codecs/isac/fix/source/lattice_c.c", - "codecs/isac/fix/source/lpc_masking_model.c", - "codecs/isac/fix/source/lpc_masking_model.h", - "codecs/isac/fix/source/lpc_tables.c", - "codecs/isac/fix/source/lpc_tables.h", - "codecs/isac/fix/source/pitch_estimator.c", - "codecs/isac/fix/source/pitch_estimator.h", - "codecs/isac/fix/source/pitch_estimator_c.c", - "codecs/isac/fix/source/pitch_filter.c", - "codecs/isac/fix/source/pitch_filter_c.c", - "codecs/isac/fix/source/pitch_gain_tables.c", - "codecs/isac/fix/source/pitch_gain_tables.h", - "codecs/isac/fix/source/pitch_lag_tables.c", - "codecs/isac/fix/source/pitch_lag_tables.h", - "codecs/isac/fix/source/spectrum_ar_model_tables.c", - "codecs/isac/fix/source/spectrum_ar_model_tables.h", - "codecs/isac/fix/source/transform.c", - ] - - deps = [ - ":isac_bwinfo", - ":isac_common", - ":isac_fix_common", - "../../api/audio_codecs:audio_codecs_api", - "../../common_audio", - "../../common_audio:common_audio_c", - "../../rtc_base:checks", - "../../rtc_base:compile_assert_c", - "../../rtc_base:sanitizer", - "../../system_wrappers", - "../third_party/fft", - ] - - if (rtc_build_with_neon) { - deps += [ ":isac_neon" ] - - # TODO(bugs.webrtc.org/9579): Consider moving the usage of NEON from - # pitch_estimator_c.c into the "isac_neon" target and delete this flag: - if (current_cpu != "arm64") { - suppressed_configs += [ "//build/config/compiler:compiler_arm_fpu" ] - cflags = [ "-mfpu=neon" ] - } - } - - if (current_cpu == "arm" && arm_version >= 7) { - sources -= [ - "codecs/isac/fix/source/lattice_c.c", - "codecs/isac/fix/source/pitch_filter_c.c", - ] - deps += [ ":isac_fix_c_arm_asm" ] - } - - if (current_cpu == "mipsel") { - sources += [ - "codecs/isac/fix/source/entropy_coding_mips.c", - "codecs/isac/fix/source/filters_mips.c", - "codecs/isac/fix/source/lattice_mips.c", - "codecs/isac/fix/source/pitch_estimator_mips.c", - "codecs/isac/fix/source/transform_mips.c", - ] - sources -= [ - "codecs/isac/fix/source/lattice_c.c", - "codecs/isac/fix/source/pitch_estimator_c.c", - ] - if (mips_dsp_rev > 0) { - sources += [ "codecs/isac/fix/source/filterbanks_mips.c" ] - } - if (mips_dsp_rev > 1) { - sources += [ - "codecs/isac/fix/source/lpc_masking_model_mips.c", - "codecs/isac/fix/source/pitch_filter_mips.c", - ] - sources -= [ "codecs/isac/fix/source/pitch_filter_c.c" ] - } - } -} - -if (rtc_build_with_neon) { - rtc_library("isac_neon") { - poisonous = [ "audio_codecs" ] - sources = [ - "codecs/isac/fix/source/entropy_coding_neon.c", - "codecs/isac/fix/source/filterbanks_neon.c", - "codecs/isac/fix/source/filters_neon.c", - "codecs/isac/fix/source/lattice_neon.c", - "codecs/isac/fix/source/transform_neon.c", - ] - - if (current_cpu != "arm64") { - # Enable compilation for the NEON instruction set. - suppressed_configs += [ "//build/config/compiler:compiler_arm_fpu" ] - cflags = [ "-mfpu=neon" ] - } - - deps = [ - ":isac_fix_common", - "../../common_audio", - "../../common_audio:common_audio_c", - "../../rtc_base:checks", - ] - } +rtc_source_set("isac_bwinfo") { + sources = [ "codecs/isac/bandwidth_info.h" ] + deps = [] } rtc_library("pcm16b") { @@ -1291,11 +1011,6 @@ if (rtc_include_tests) { if (rtc_include_opus) { audio_coding_deps += [ ":webrtc_opus" ] } - if (current_cpu == "arm") { - audio_coding_deps += [ ":isac_fix" ] - } else { - audio_coding_deps += [ ":isac" ] - } if (!build_with_mozilla && !build_with_chromium) { audio_coding_deps += [ ":red" ] } @@ -1327,11 +1042,7 @@ if (rtc_include_tests) { ":g711_test", ":g722_test", ":ilbc_test", - ":isac_api_test", - ":isac_switch_samprate_test", - ":isac_test", ":neteq_ilbc_quality_test", - ":neteq_isac_quality_test", ":neteq_opus_quality_test", ":neteq_pcm16b_quality_test", ":neteq_pcmu_quality_test", @@ -1371,8 +1082,6 @@ if (rtc_include_tests) { "test/Tester.cc", "test/TwoWayCommunication.cc", "test/TwoWayCommunication.h", - "test/iSACTest.cc", - "test/iSACTest.h", "test/target_delay_unittest.cc", ] deps = [ @@ -1397,8 +1106,6 @@ if (rtc_include_tests) { "../../api/audio_codecs/g722:audio_encoder_g722", "../../api/audio_codecs/ilbc:audio_decoder_ilbc", "../../api/audio_codecs/ilbc:audio_encoder_ilbc", - "../../api/audio_codecs/isac:audio_decoder_isac_float", - "../../api/audio_codecs/isac:audio_encoder_isac_float", "../../api/audio_codecs/opus:audio_decoder_opus", "../../api/audio_codecs/opus:audio_encoder_opus", "../../common_audio", @@ -1513,8 +1220,6 @@ if (rtc_include_tests) { deps = [ ":ilbc", - ":isac", - ":isac_fix", ":neteq", ":neteq_input_audio_tools", ":neteq_tools", @@ -1619,12 +1324,10 @@ if (rtc_include_tests) { testonly = true defines = [] deps = [ - ":isac_fix_common", "../../rtc_base:macromagic", "../../test:fileutils", ] sources = [ - "codecs/isac/fix/test/isac_speed_test.cc", "codecs/opus/opus_speed_test.cc", "codecs/tools/audio_codec_speed_test.cc", "codecs/tools/audio_codec_speed_test.h", @@ -1647,7 +1350,6 @@ if (rtc_include_tests) { } deps += [ - ":isac_fix", ":webrtc_opus", "../../rtc_base:checks", "../../test:test_main", @@ -1723,7 +1425,6 @@ if (rtc_include_tests) { "../../api/audio_codecs/g711:audio_encoder_g711", "../../api/audio_codecs/g722:audio_encoder_g722", "../../api/audio_codecs/ilbc:audio_encoder_ilbc", - "../../api/audio_codecs/isac:audio_encoder_isac", "../../api/audio_codecs/opus:audio_encoder_opus", "../../rtc_base:safe_conversions", "//third_party/abseil-cpp/absl/flags:flag", @@ -1832,21 +1533,6 @@ if (rtc_include_tests) { ] } - rtc_executable("neteq_isac_quality_test") { - testonly = true - - sources = [ "neteq/test/neteq_isac_quality_test.cc" ] - - deps = [ - ":isac_fix", - ":neteq", - ":neteq_quality_test_support", - "../../test:test_main", - "//testing/gtest", - "//third_party/abseil-cpp/absl/flags:flag", - ] - } - rtc_executable("neteq_pcmu_quality_test") { testonly = true @@ -1884,28 +1570,6 @@ if (rtc_include_tests) { } } - rtc_library("isac_test_util") { - testonly = true - sources = [ - "codecs/isac/main/util/utility.c", - "codecs/isac/main/util/utility.h", - ] - } - - if (!build_with_chromium) { - rtc_executable("isac_test") { - testonly = true - - sources = [ "codecs/isac/main/test/simpleKenny.c" ] - - deps = [ - ":isac", - ":isac_test_util", - "../../rtc_base:macromagic", - ] - } - } - rtc_executable("g711_test") { testonly = true @@ -1923,32 +1587,6 @@ if (rtc_include_tests) { } if (!build_with_chromium) { - rtc_executable("isac_api_test") { - testonly = true - - sources = [ "codecs/isac/main/test/ReleaseTest-API/ReleaseTest-API.cc" ] - - deps = [ - ":isac", - ":isac_test_util", - "../../rtc_base:macromagic", - ] - } - - rtc_executable("isac_switch_samprate_test") { - testonly = true - - sources = - [ "codecs/isac/main/test/SwitchingSampRate/SwitchingSampRate.cc" ] - - deps = [ - ":isac", - ":isac_test_util", - "../../common_audio", - "../../common_audio:common_audio_c", - ] - } - rtc_executable("ilbc_test") { testonly = true @@ -1997,13 +1635,6 @@ if (rtc_include_tests) { "codecs/cng/audio_encoder_cng_unittest.cc", "codecs/cng/cng_unittest.cc", "codecs/ilbc/ilbc_unittest.cc", - "codecs/isac/fix/source/filterbanks_unittest.cc", - "codecs/isac/fix/source/filters_unittest.cc", - "codecs/isac/fix/source/lpc_masking_model_unittest.cc", - "codecs/isac/fix/source/transform_unittest.cc", - "codecs/isac/isac_webrtc_api_test.cc", - "codecs/isac/main/source/audio_encoder_isac_unittest.cc", - "codecs/isac/main/source/isac_unittest.cc", "codecs/legacy_encoded_audio_frame_unittest.cc", "codecs/opus/audio_decoder_multi_channel_opus_unittest.cc", "codecs/opus/audio_encoder_multi_channel_opus_unittest.cc", @@ -2070,11 +1701,6 @@ if (rtc_include_tests) { ":default_neteq_factory", ":g711", ":ilbc", - ":isac", - ":isac_c", - ":isac_common", - ":isac_fix", - ":isac_fix_common", ":legacy_encoded_audio_frame", ":mocks", ":neteq", @@ -2094,10 +1720,6 @@ if (rtc_include_tests) { "../../api/audio_codecs:audio_codecs_api", "../../api/audio_codecs:builtin_audio_decoder_factory", "../../api/audio_codecs:builtin_audio_encoder_factory", - "../../api/audio_codecs/isac:audio_decoder_isac_fix", - "../../api/audio_codecs/isac:audio_decoder_isac_float", - "../../api/audio_codecs/isac:audio_encoder_isac_fix", - "../../api/audio_codecs/isac:audio_encoder_isac_float", "../../api/audio_codecs/opus:audio_decoder_multiopus", "../../api/audio_codecs/opus:audio_decoder_opus", "../../api/audio_codecs/opus:audio_encoder_multiopus", diff --git a/modules/audio_coding/acm2/acm_receiver_unittest.cc b/modules/audio_coding/acm2/acm_receiver_unittest.cc index e73acc2338..6dd44b696e 100644 --- a/modules/audio_coding/acm2/acm_receiver_unittest.cc +++ b/modules/audio_coding/acm2/acm_receiver_unittest.cc @@ -13,6 +13,7 @@ #include // std::min #include +#include "absl/types/optional.h" #include "api/audio_codecs/builtin_audio_decoder_factory.h" #include "api/audio_codecs/builtin_audio_encoder_factory.h" #include "modules/audio_coding/codecs/cng/audio_encoder_cng.h" @@ -64,12 +65,14 @@ class AcmReceiverTestOldApi : public AudioPacketizationCallback, const SdpAudioFormat& format, const std::map cng_payload_types = {}) { // Create the speech encoder. - AudioCodecInfo info = encoder_factory_->QueryAudioEncoder(format).value(); + absl::optional info = + encoder_factory_->QueryAudioEncoder(format); + RTC_CHECK(info.has_value()); std::unique_ptr enc = encoder_factory_->MakeAudioEncoder(payload_type, format, absl::nullopt); // If we have a compatible CN specification, stack a CNG on top. - auto it = cng_payload_types.find(info.sample_rate_hz); + auto it = cng_payload_types.find(info->sample_rate_hz); if (it != cng_payload_types.end()) { AudioEncoderCngConfig config; config.speech_encoder = std::move(enc); @@ -81,7 +84,7 @@ class AcmReceiverTestOldApi : public AudioPacketizationCallback, // Actually start using the new encoder. acm_->SetEncoder(std::move(enc)); - return info; + return *info; } int InsertOnePacketOfSilence(const AudioCodecInfo& info) { @@ -148,8 +151,7 @@ class AcmReceiverTestOldApi : public AudioPacketizationCallback, #define MAYBE_SampleRate SampleRate #endif TEST_F(AcmReceiverTestOldApi, MAYBE_SampleRate) { - const std::map codecs = {{0, {"ISAC", 16000, 1}}, - {1, {"ISAC", 32000, 1}}}; + const std::map codecs = {{0, {"OPUS", 48000, 2}}}; receiver_->SetCodecs(codecs); constexpr int kOutSampleRateHz = 8000; // Different than codec sample rate. @@ -232,15 +234,6 @@ TEST_F(AcmReceiverTestFaxModeOldApi, MAYBE_VerifyAudioFramePCMU) { RunVerifyAudioFrame({"PCMU", 8000, 1}); } -#if defined(WEBRTC_ANDROID) -#define MAYBE_VerifyAudioFrameISAC DISABLED_VerifyAudioFrameISAC -#else -#define MAYBE_VerifyAudioFrameISAC VerifyAudioFrameISAC -#endif -TEST_F(AcmReceiverTestFaxModeOldApi, MAYBE_VerifyAudioFrameISAC) { - RunVerifyAudioFrame({"ISAC", 16000, 1}); -} - #if defined(WEBRTC_ANDROID) #define MAYBE_VerifyAudioFrameOpus DISABLED_VerifyAudioFrameOpus #else @@ -310,12 +303,10 @@ TEST_F(AcmReceiverTestPostDecodeVadPassiveOldApi, MAYBE_PostdecodingVad) { #else #define MAYBE_LastAudioCodec LastAudioCodec #endif -#if defined(WEBRTC_CODEC_ISAC) +#if defined(WEBRTC_CODEC_OPUS) TEST_F(AcmReceiverTestOldApi, MAYBE_LastAudioCodec) { - const std::map codecs = {{0, {"ISAC", 16000, 1}}, - {1, {"PCMA", 8000, 1}}, - {2, {"ISAC", 32000, 1}}, - {3, {"L16", 32000, 1}}}; + const std::map codecs = { + {0, {"PCMU", 8000, 1}}, {1, {"PCMA", 8000, 1}}, {2, {"L16", 32000, 1}}}; const std::map cng_payload_types = { {8000, 100}, {16000, 101}, {32000, 102}}; { diff --git a/modules/audio_coding/acm2/audio_coding_module_unittest.cc b/modules/audio_coding/acm2/audio_coding_module_unittest.cc index 7e4b764aed..f1eb81c015 100644 --- a/modules/audio_coding/acm2/audio_coding_module_unittest.cc +++ b/modules/audio_coding/acm2/audio_coding_module_unittest.cc @@ -30,7 +30,6 @@ #include "modules/audio_coding/codecs/cng/audio_encoder_cng.h" #include "modules/audio_coding/codecs/g711/audio_decoder_pcm.h" #include "modules/audio_coding/codecs/g711/audio_encoder_pcm.h" -#include "modules/audio_coding/codecs/isac/main/include/audio_encoder_isac.h" #include "modules/audio_coding/include/audio_coding_module_typedefs.h" #include "modules/audio_coding/neteq/tools/audio_checksum.h" #include "modules/audio_coding/neteq/tools/audio_loop.h" @@ -302,44 +301,6 @@ TEST_F(AudioCodingModuleTestOldApi, TransportCallbackIsInvokedForEachPacket) { EXPECT_EQ(AudioFrameType::kAudioFrameSpeech, packet_cb_.last_frame_type()); } -#if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX) -// Verifies that the RTP timestamp series is not reset when the codec is -// changed. -TEST_F(AudioCodingModuleTestOldApi, TimestampSeriesContinuesWhenCodecChanges) { - RegisterCodec(); // This registers the default codec. - uint32_t expected_ts = input_frame_.timestamp_; - int blocks_per_packet = pac_size_ / (kSampleRateHz / 100); - // Encode 5 packets of the first codec type. - const int kNumPackets1 = 5; - for (int j = 0; j < kNumPackets1; ++j) { - for (int i = 0; i < blocks_per_packet; ++i) { - EXPECT_EQ(j, packet_cb_.num_calls()); - InsertAudio(); - } - EXPECT_EQ(j + 1, packet_cb_.num_calls()); - EXPECT_EQ(expected_ts, packet_cb_.last_timestamp()); - expected_ts += pac_size_; - } - - // Change codec. - audio_format_ = SdpAudioFormat("ISAC", kSampleRateHz, 1); - pac_size_ = 480; - RegisterCodec(); - blocks_per_packet = pac_size_ / (kSampleRateHz / 100); - // Encode another 5 packets. - const int kNumPackets2 = 5; - for (int j = 0; j < kNumPackets2; ++j) { - for (int i = 0; i < blocks_per_packet; ++i) { - EXPECT_EQ(kNumPackets1 + j, packet_cb_.num_calls()); - InsertAudio(); - } - EXPECT_EQ(kNumPackets1 + j + 1, packet_cb_.num_calls()); - EXPECT_EQ(expected_ts, packet_cb_.last_timestamp()); - expected_ts += pac_size_; - } -} -#endif - // Introduce this class to set different expectations on the number of encoded // bytes. This class expects all encoded packets to be 9 bytes (matching one // CNG SID frame) or 0 bytes. This test depends on `input_frame_` containing @@ -420,8 +381,7 @@ TEST_F(AudioCodingModuleTestWithComfortNoiseOldApi, DoTest(k10MsBlocksPerPacket, kCngPayloadType); } -// A multi-threaded test for ACM. This base class is using the PCM16b 16 kHz -// codec, while the derive class AcmIsacMtTest is using iSAC. +// A multi-threaded test for ACM that uses the PCM16b 16 kHz codec. class AudioCodingModuleMtTestOldApi : public AudioCodingModuleTestOldApi { protected: static const int kNumPackets = 500; @@ -560,272 +520,6 @@ TEST_F(AudioCodingModuleMtTestOldApi, MAYBE_DoTest) { EXPECT_TRUE(RunTest()); } -// This is a multi-threaded ACM test using iSAC. The test encodes audio -// from a PCM file. The most recent encoded frame is used as input to the -// receiving part. Depending on timing, it may happen that the same RTP packet -// is inserted into the receiver multiple times, but this is a valid use-case, -// and simplifies the test code a lot. -class AcmIsacMtTestOldApi : public AudioCodingModuleMtTestOldApi { - protected: - static const int kNumPackets = 500; - static const int kNumPullCalls = 500; - - AcmIsacMtTestOldApi() - : AudioCodingModuleMtTestOldApi(), last_packet_number_(0) {} - - ~AcmIsacMtTestOldApi() {} - - void SetUp() override { - AudioCodingModuleTestOldApi::SetUp(); - RegisterCodec(); // Must be called before the threads start below. - - // Set up input audio source to read from specified file, loop after 5 - // seconds, and deliver blocks of 10 ms. - const std::string input_file_name = - webrtc::test::ResourcePath("audio_coding/speech_mono_16kHz", "pcm"); - audio_loop_.Init(input_file_name, 5 * kSampleRateHz, kNumSamples10ms); - - // Generate one packet to have something to insert. - int loop_counter = 0; - while (packet_cb_.last_payload_len_bytes() == 0) { - InsertAudio(); - ASSERT_LT(loop_counter++, 10); - } - // Set `last_packet_number_` to one less that `num_calls` so that the packet - // will be fetched in the next InsertPacket() call. - last_packet_number_ = packet_cb_.num_calls() - 1; - - StartThreads(); - } - - void RegisterCodec() override { - static_assert(kSampleRateHz == 16000, "test designed for iSAC 16 kHz"); - audio_format_ = SdpAudioFormat("isac", kSampleRateHz, 1); - pac_size_ = 480; - - // Register iSAC codec in ACM, effectively unregistering the PCM16B codec - // registered in AudioCodingModuleTestOldApi::SetUp(); - acm_->SetReceiveCodecs({{kPayloadType, *audio_format_}}); - acm_->SetEncoder(CreateBuiltinAudioEncoderFactory()->MakeAudioEncoder( - kPayloadType, *audio_format_, absl::nullopt)); - } - - void InsertPacket() override { - int num_calls = packet_cb_.num_calls(); // Store locally for thread safety. - if (num_calls > last_packet_number_) { - // Get the new payload out from the callback handler. - // Note that since we swap buffers here instead of directly inserting - // a pointer to the data in `packet_cb_`, we avoid locking the callback - // for the duration of the IncomingPacket() call. - packet_cb_.SwapBuffers(&last_payload_vec_); - ASSERT_GT(last_payload_vec_.size(), 0u); - rtp_utility_->Forward(&rtp_header_); - last_packet_number_ = num_calls; - } - ASSERT_GT(last_payload_vec_.size(), 0u); - ASSERT_EQ(0, acm_->IncomingPacket(&last_payload_vec_[0], - last_payload_vec_.size(), rtp_header_)); - } - - void InsertAudio() override { - // TODO(kwiberg): Use std::copy here. Might be complications because AFAICS - // this call confuses the number of samples with the number of bytes, and - // ends up copying only half of what it should. - memcpy(input_frame_.mutable_data(), audio_loop_.GetNextBlock().data(), - kNumSamples10ms); - AudioCodingModuleTestOldApi::InsertAudio(); - } - - // Override the verification function with no-op, since iSAC produces variable - // payload sizes. - void VerifyEncoding() override {} - - // This method is the same as AudioCodingModuleMtTestOldApi::TestDone(), but - // here it is using the constants defined in this class (i.e., shorter test - // run). - bool TestDone() override { - if (packet_cb_.num_calls() > kNumPackets) { - MutexLock lock(&mutex_); - if (pull_audio_count_ > kNumPullCalls) { - // Both conditions for completion are met. End the test. - return true; - } - } - return false; - } - - int last_packet_number_; - std::vector last_payload_vec_; - test::AudioLoop audio_loop_; -}; - -#if defined(WEBRTC_IOS) -#define MAYBE_DoTest DISABLED_DoTest -#else -#define MAYBE_DoTest DoTest -#endif -#if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX) -TEST_F(AcmIsacMtTestOldApi, MAYBE_DoTest) { - EXPECT_TRUE(RunTest()); -} -#endif - -class AcmReRegisterIsacMtTestOldApi : public AudioCodingModuleTestOldApi { - protected: - static const int kRegisterAfterNumPackets = 5; - static const int kNumPackets = 10; - static const int kPacketSizeMs = 30; - static const int kPacketSizeSamples = kPacketSizeMs * 16; - - AcmReRegisterIsacMtTestOldApi() - : AudioCodingModuleTestOldApi(), - codec_registered_(false), - receive_packet_count_(0), - next_insert_packet_time_ms_(0), - fake_clock_(new SimulatedClock(0)) { - AudioEncoderIsacFloatImpl::Config config; - config.payload_type = kPayloadType; - isac_encoder_.reset(new AudioEncoderIsacFloatImpl(config)); - clock_ = fake_clock_.get(); - } - - void SetUp() override { - AudioCodingModuleTestOldApi::SetUp(); - // Set up input audio source to read from specified file, loop after 5 - // seconds, and deliver blocks of 10 ms. - const std::string input_file_name = - webrtc::test::ResourcePath("audio_coding/speech_mono_16kHz", "pcm"); - audio_loop_.Init(input_file_name, 5 * kSampleRateHz, kNumSamples10ms); - RegisterCodec(); // Must be called before the threads start below. - StartThreads(); - } - - void RegisterCodec() override { - // Register iSAC codec in ACM, effectively unregistering the PCM16B codec - // registered in AudioCodingModuleTestOldApi::SetUp(); - // Only register the decoder for now. The encoder is registered later. - static_assert(kSampleRateHz == 16000, "test designed for iSAC 16 kHz"); - acm_->SetReceiveCodecs({{kPayloadType, {"ISAC", kSampleRateHz, 1}}}); - } - - void StartThreads() { - quit_.store(false); - const auto attributes = - rtc::ThreadAttributes().SetPriority(rtc::ThreadPriority::kRealtime); - receive_thread_ = rtc::PlatformThread::SpawnJoinable( - [this] { - while (!quit_.load() && CbReceiveImpl()) { - } - }, - "receive", attributes); - codec_registration_thread_ = rtc::PlatformThread::SpawnJoinable( - [this] { - while (!quit_.load()) { - CbCodecRegistrationImpl(); - } - }, - "codec_registration", attributes); - } - - void TearDown() override { - AudioCodingModuleTestOldApi::TearDown(); - quit_.store(true); - receive_thread_.Finalize(); - codec_registration_thread_.Finalize(); - } - - bool RunTest() { return test_complete_.Wait(TimeDelta::Minutes(10)); } - - bool CbReceiveImpl() { - SleepMs(1); - rtc::Buffer encoded; - AudioEncoder::EncodedInfo info; - { - MutexLock lock(&mutex_); - if (clock_->TimeInMilliseconds() < next_insert_packet_time_ms_) { - return true; - } - next_insert_packet_time_ms_ += kPacketSizeMs; - ++receive_packet_count_; - - // Encode new frame. - uint32_t input_timestamp = rtp_header_.timestamp; - while (info.encoded_bytes == 0) { - info = isac_encoder_->Encode(input_timestamp, - audio_loop_.GetNextBlock(), &encoded); - input_timestamp += 160; // 10 ms at 16 kHz. - } - EXPECT_EQ(rtp_header_.timestamp + kPacketSizeSamples, input_timestamp); - EXPECT_EQ(rtp_header_.timestamp, info.encoded_timestamp); - EXPECT_EQ(rtp_header_.payloadType, info.payload_type); - } - // Now we're not holding the crit sect when calling ACM. - - // Insert into ACM. - EXPECT_EQ(0, acm_->IncomingPacket(encoded.data(), info.encoded_bytes, - rtp_header_)); - - // Pull audio. - for (int i = 0; i < rtc::CheckedDivExact(kPacketSizeMs, 10); ++i) { - AudioFrame audio_frame; - bool muted; - EXPECT_EQ(0, acm_->PlayoutData10Ms(-1 /* default output frequency */, - &audio_frame, &muted)); - if (muted) { - ADD_FAILURE(); - return false; - } - fake_clock_->AdvanceTimeMilliseconds(10); - } - rtp_utility_->Forward(&rtp_header_); - return true; - } - - void CbCodecRegistrationImpl() { - SleepMs(1); - if (HasFatalFailure()) { - // End the test early if a fatal failure (ASSERT_*) has occurred. - test_complete_.Set(); - } - MutexLock lock(&mutex_); - if (!codec_registered_ && - receive_packet_count_ > kRegisterAfterNumPackets) { - // Register the iSAC encoder. - acm_->SetEncoder(CreateBuiltinAudioEncoderFactory()->MakeAudioEncoder( - kPayloadType, *audio_format_, absl::nullopt)); - codec_registered_ = true; - } - if (codec_registered_ && receive_packet_count_ > kNumPackets) { - test_complete_.Set(); - } - } - - rtc::PlatformThread receive_thread_; - rtc::PlatformThread codec_registration_thread_; - // Used to force worker threads to stop looping. - std::atomic quit_; - - rtc::Event test_complete_; - Mutex mutex_; - bool codec_registered_ RTC_GUARDED_BY(mutex_); - int receive_packet_count_ RTC_GUARDED_BY(mutex_); - int64_t next_insert_packet_time_ms_ RTC_GUARDED_BY(mutex_); - std::unique_ptr isac_encoder_; - std::unique_ptr fake_clock_; - test::AudioLoop audio_loop_; -}; - -#if defined(WEBRTC_IOS) -#define MAYBE_DoTest DISABLED_DoTest -#else -#define MAYBE_DoTest DoTest -#endif -#if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX) -TEST_F(AcmReRegisterIsacMtTestOldApi, MAYBE_DoTest) { - EXPECT_TRUE(RunTest()); -} -#endif - // Disabling all of these tests on iOS until file support has been added. // See https://code.google.com/p/webrtc/issues/detail?id=4752 for details. #if !defined(WEBRTC_IOS) @@ -1025,38 +719,6 @@ class AcmSenderBitExactnessOldApi : public ::testing::Test, class AcmSenderBitExactnessNewApi : public AcmSenderBitExactnessOldApi {}; -// Run bit exactness tests only for release builds. -#if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)) && \ - defined(NDEBUG) && defined(WEBRTC_LINUX) && defined(WEBRTC_ARCH_X86_64) -TEST_F(AcmSenderBitExactnessOldApi, IsacWb30ms) { - ASSERT_NO_FATAL_FAILURE(SetUpTest("ISAC", 16000, 1, 103, 480, 480)); - Run(/*audio_checksum_ref=*/"37ecdabad1698a857cf811e6d1fa91df", - /*payload_checksum_ref=*/"3c79f16f34218271f3dca4e2b1dfe1bb", - /*expected_packets=*/33, - /*expected_channels=*/test::AcmReceiveTestOldApi::kMonoOutput); -} - -TEST_F(AcmSenderBitExactnessOldApi, IsacWb60ms) { - ASSERT_NO_FATAL_FAILURE(SetUpTest("ISAC", 16000, 1, 103, 960, 960)); - Run(/*audio_checksum_ref=*/"0e9078d23454901496a88362ba0740c3", - /*payload_checksum_ref=*/"9e0a0ab743ad987b55b8e14802769c56", - /*expected_packets=*/16, - /*expected_channels=*/test::AcmReceiveTestOldApi::kMonoOutput); -} -#endif - -// Run bit exactness test only for release build. -#if defined(WEBRTC_CODEC_ISAC) && defined(NDEBUG) && defined(WEBRTC_LINUX) && \ - defined(WEBRTC_ARCH_X86_64) -TEST_F(AcmSenderBitExactnessOldApi, IsacSwb30ms) { - ASSERT_NO_FATAL_FAILURE(SetUpTest("ISAC", 32000, 1, 104, 960, 960)); - Run(/*audio_checksum_ref=*/"f4cf577f28a0dcbac33358b757518e0c", - /*payload_checksum_ref=*/"ce86106a93419aefb063097108ec94ab", - /*expected_packets=*/33, - /*expected_channels=*/test::AcmReceiveTestOldApi::kMonoOutput); -} -#endif - TEST_F(AcmSenderBitExactnessOldApi, Pcm16_8000khz_10ms) { ASSERT_NO_FATAL_FAILURE(SetUpTest("L16", 8000, 1, 107, 80, 80)); Run(/*audio_checksum_ref=*/"69118ed438ac76252d023e0463819471", diff --git a/modules/audio_coding/audio_coding.gni b/modules/audio_coding/audio_coding.gni index bf67d9cb8d..3b147091de 100644 --- a/modules/audio_coding/audio_coding.gni +++ b/modules/audio_coding/audio_coding.gni @@ -20,11 +20,6 @@ if (rtc_opus_support_120ms_ptime) { } else { audio_codec_defines += [ "WEBRTC_OPUS_SUPPORT_120MS_PTIME=0" ] } -if (current_cpu == "arm") { - audio_codec_defines += [ "WEBRTC_CODEC_ISACFX" ] -} else { - audio_codec_defines += [ "WEBRTC_CODEC_ISAC" ] -} audio_coding_defines = audio_codec_defines neteq_defines = audio_codec_defines diff --git a/modules/audio_coding/codecs/builtin_audio_decoder_factory_unittest.cc b/modules/audio_coding/codecs/builtin_audio_decoder_factory_unittest.cc index 109da78eea..4a2b261a59 100644 --- a/modules/audio_coding/codecs/builtin_audio_decoder_factory_unittest.cc +++ b/modules/audio_coding/codecs/builtin_audio_decoder_factory_unittest.cc @@ -75,31 +75,6 @@ TEST(AudioDecoderFactoryTest, CreateIlbc) { adf->MakeAudioDecoder(SdpAudioFormat("ilbc", 16000, 1), absl::nullopt)); } -TEST(AudioDecoderFactoryTest, CreateIsac) { - rtc::scoped_refptr adf = - CreateBuiltinAudioDecoderFactory(); - ASSERT_TRUE(adf); - // iSAC supports 16 kHz, 1 channel. The float implementation additionally - // supports 32 kHz, 1 channel. - EXPECT_FALSE( - adf->MakeAudioDecoder(SdpAudioFormat("isac", 16000, 0), absl::nullopt)); - EXPECT_TRUE( - adf->MakeAudioDecoder(SdpAudioFormat("isac", 16000, 1), absl::nullopt)); - EXPECT_FALSE( - adf->MakeAudioDecoder(SdpAudioFormat("isac", 16000, 2), absl::nullopt)); - EXPECT_FALSE( - adf->MakeAudioDecoder(SdpAudioFormat("isac", 8000, 1), absl::nullopt)); - EXPECT_FALSE( - adf->MakeAudioDecoder(SdpAudioFormat("isac", 48000, 1), absl::nullopt)); -#ifdef WEBRTC_ARCH_ARM - EXPECT_FALSE( - adf->MakeAudioDecoder(SdpAudioFormat("isac", 32000, 1), absl::nullopt)); -#else - EXPECT_TRUE( - adf->MakeAudioDecoder(SdpAudioFormat("isac", 32000, 1), absl::nullopt)); -#endif -} - TEST(AudioDecoderFactoryTest, CreateL16) { rtc::scoped_refptr adf = CreateBuiltinAudioDecoderFactory(); @@ -125,9 +100,6 @@ TEST(AudioDecoderFactoryTest, MaxNrOfChannels) { #ifdef WEBRTC_CODEC_OPUS "opus", #endif -#if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX) - "isac", -#endif #ifdef WEBRTC_CODEC_ILBC "ilbc", #endif diff --git a/modules/audio_coding/codecs/isac/audio_decoder_isac_t.h b/modules/audio_coding/codecs/isac/audio_decoder_isac_t.h deleted file mode 100644 index aae708f295..0000000000 --- a/modules/audio_coding/codecs/isac/audio_decoder_isac_t.h +++ /dev/null @@ -1,54 +0,0 @@ -/* - * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_DECODER_ISAC_T_H_ -#define MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_DECODER_ISAC_T_H_ - -#include - -#include "absl/types/optional.h" -#include "api/audio_codecs/audio_decoder.h" -#include "api/scoped_refptr.h" - -namespace webrtc { - -template -class AudioDecoderIsacT final : public AudioDecoder { - public: - struct Config { - bool IsOk() const; - int sample_rate_hz = 16000; - }; - explicit AudioDecoderIsacT(const Config& config); - virtual ~AudioDecoderIsacT() override; - - AudioDecoderIsacT(const AudioDecoderIsacT&) = delete; - AudioDecoderIsacT& operator=(const AudioDecoderIsacT&) = delete; - - bool HasDecodePlc() const override; - size_t DecodePlc(size_t num_frames, int16_t* decoded) override; - void Reset() override; - int ErrorCode() override; - int SampleRateHz() const override; - size_t Channels() const override; - int DecodeInternal(const uint8_t* encoded, - size_t encoded_len, - int sample_rate_hz, - int16_t* decoded, - SpeechType* speech_type) override; - - private: - typename T::instance_type* isac_state_; - int sample_rate_hz_; -}; - -} // namespace webrtc - -#endif // MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_DECODER_ISAC_T_H_ diff --git a/modules/audio_coding/codecs/isac/audio_decoder_isac_t_impl.h b/modules/audio_coding/codecs/isac/audio_decoder_isac_t_impl.h deleted file mode 100644 index 9aa498866b..0000000000 --- a/modules/audio_coding/codecs/isac/audio_decoder_isac_t_impl.h +++ /dev/null @@ -1,85 +0,0 @@ -/* - * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_DECODER_ISAC_T_IMPL_H_ -#define MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_DECODER_ISAC_T_IMPL_H_ - -#include "modules/audio_coding/codecs/isac/audio_decoder_isac_t.h" -#include "rtc_base/checks.h" - -namespace webrtc { - -template -bool AudioDecoderIsacT::Config::IsOk() const { - return (sample_rate_hz == 16000 || sample_rate_hz == 32000); -} - -template -AudioDecoderIsacT::AudioDecoderIsacT(const Config& config) - : sample_rate_hz_(config.sample_rate_hz) { - RTC_CHECK(config.IsOk()) << "Unsupported sample rate " - << config.sample_rate_hz; - RTC_CHECK_EQ(0, T::Create(&isac_state_)); - T::DecoderInit(isac_state_); - RTC_CHECK_EQ(0, T::SetDecSampRate(isac_state_, sample_rate_hz_)); -} - -template -AudioDecoderIsacT::~AudioDecoderIsacT() { - RTC_CHECK_EQ(0, T::Free(isac_state_)); -} - -template -int AudioDecoderIsacT::DecodeInternal(const uint8_t* encoded, - size_t encoded_len, - int sample_rate_hz, - int16_t* decoded, - SpeechType* speech_type) { - RTC_CHECK_EQ(sample_rate_hz_, sample_rate_hz); - int16_t temp_type = 1; // Default is speech. - int ret = - T::DecodeInternal(isac_state_, encoded, encoded_len, decoded, &temp_type); - *speech_type = ConvertSpeechType(temp_type); - return ret; -} - -template -bool AudioDecoderIsacT::HasDecodePlc() const { - return false; -} - -template -size_t AudioDecoderIsacT::DecodePlc(size_t num_frames, int16_t* decoded) { - return T::DecodePlc(isac_state_, decoded, num_frames); -} - -template -void AudioDecoderIsacT::Reset() { - T::DecoderInit(isac_state_); -} - -template -int AudioDecoderIsacT::ErrorCode() { - return T::GetErrorCode(isac_state_); -} - -template -int AudioDecoderIsacT::SampleRateHz() const { - return sample_rate_hz_; -} - -template -size_t AudioDecoderIsacT::Channels() const { - return 1; -} - -} // namespace webrtc - -#endif // MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_DECODER_ISAC_T_IMPL_H_ diff --git a/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h b/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h deleted file mode 100644 index c382ea076e..0000000000 --- a/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h +++ /dev/null @@ -1,108 +0,0 @@ -/* - * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_ENCODER_ISAC_T_H_ -#define MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_ENCODER_ISAC_T_H_ - -#include -#include - -#include "absl/types/optional.h" -#include "api/audio_codecs/audio_encoder.h" -#include "api/scoped_refptr.h" -#include "api/units/time_delta.h" -#include "system_wrappers/include/field_trial.h" - -namespace webrtc { - -template -class AudioEncoderIsacT final : public AudioEncoder { - public: - // Allowed combinations of sample rate, frame size, and bit rate are - // - 16000 Hz, 30 ms, 10000-32000 bps - // - 16000 Hz, 60 ms, 10000-32000 bps - // - 32000 Hz, 30 ms, 10000-56000 bps (if T has super-wideband support) - struct Config { - bool IsOk() const; - int payload_type = 103; - int sample_rate_hz = 16000; - int frame_size_ms = 30; - int bit_rate = kDefaultBitRate; // Limit on the short-term average bit - // rate, in bits/s. - int max_payload_size_bytes = -1; - int max_bit_rate = -1; - }; - - explicit AudioEncoderIsacT(const Config& config); - ~AudioEncoderIsacT() override; - - AudioEncoderIsacT(const AudioEncoderIsacT&) = delete; - AudioEncoderIsacT& operator=(const AudioEncoderIsacT&) = delete; - - int SampleRateHz() const override; - size_t NumChannels() const override; - size_t Num10MsFramesInNextPacket() const override; - size_t Max10MsFramesInAPacket() const override; - int GetTargetBitrate() const override; - void SetTargetBitrate(int target_bps) override; - void OnReceivedTargetAudioBitrate(int target_bps) override; - void OnReceivedUplinkBandwidth( - int target_audio_bitrate_bps, - absl::optional bwe_period_ms) override; - void OnReceivedUplinkAllocation(BitrateAllocationUpdate update) override; - void OnReceivedOverhead(size_t overhead_bytes_per_packet) override; - EncodedInfo EncodeImpl(uint32_t rtp_timestamp, - rtc::ArrayView audio, - rtc::Buffer* encoded) override; - void Reset() override; - absl::optional> GetFrameLengthRange() - const override; - - private: - // This value is taken from STREAM_SIZE_MAX_60 for iSAC float (60 ms) and - // STREAM_MAXW16_60MS for iSAC fix (60 ms). - static const size_t kSufficientEncodeBufferSizeBytes = 400; - - static constexpr int kDefaultBitRate = 32000; - static constexpr int kMinBitrateBps = 10000; - static constexpr int MaxBitrateBps(int sample_rate_hz) { - return sample_rate_hz == 32000 ? 56000 : 32000; - } - - void SetTargetBitrate(int target_bps, bool subtract_per_packet_overhead); - - // Recreate the iSAC encoder instance with the given settings, and save them. - void RecreateEncoderInstance(const Config& config); - - Config config_; - typename T::instance_type* isac_state_ = nullptr; - - // Have we accepted input but not yet emitted it in a packet? - bool packet_in_progress_ = false; - - // Timestamp of the first input of the currently in-progress packet. - uint32_t packet_timestamp_; - - // Timestamp of the previously encoded packet. - uint32_t last_encoded_timestamp_; - - // Cache the value of the "WebRTC-SendSideBwe-WithOverhead" field trial. - const bool send_side_bwe_with_overhead_ = - !field_trial::IsDisabled("WebRTC-SendSideBwe-WithOverhead"); - - // When we send a packet, expect this many bytes of headers to be added to it. - // Start out with a reasonable default that we can use until we receive a real - // value. - DataSize overhead_per_packet_ = DataSize::Bytes(28); -}; - -} // namespace webrtc - -#endif // MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_ENCODER_ISAC_T_H_ diff --git a/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h b/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h deleted file mode 100644 index 1bd27cf80d..0000000000 --- a/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h +++ /dev/null @@ -1,225 +0,0 @@ -/* - * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_ENCODER_ISAC_T_IMPL_H_ -#define MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_ENCODER_ISAC_T_IMPL_H_ - -#include "modules/audio_coding/codecs/isac/audio_encoder_isac_t.h" -#include "rtc_base/checks.h" -#include "rtc_base/numerics/safe_minmax.h" - -namespace webrtc { - -template -bool AudioEncoderIsacT::Config::IsOk() const { - if (max_bit_rate < 32000 && max_bit_rate != -1) - return false; - if (max_payload_size_bytes < 120 && max_payload_size_bytes != -1) - return false; - - switch (sample_rate_hz) { - case 16000: - if (max_bit_rate > 53400) - return false; - if (max_payload_size_bytes > 400) - return false; - return (frame_size_ms == 30 || frame_size_ms == 60) && - (bit_rate == 0 || (bit_rate >= 10000 && bit_rate <= 32000)); - case 32000: - if (max_bit_rate > 160000) - return false; - if (max_payload_size_bytes > 600) - return false; - return T::has_swb && - (frame_size_ms == 30 && - (bit_rate == 0 || (bit_rate >= 10000 && bit_rate <= 56000))); - default: - return false; - } -} - -template -AudioEncoderIsacT::AudioEncoderIsacT(const Config& config) { - RecreateEncoderInstance(config); -} - -template -AudioEncoderIsacT::~AudioEncoderIsacT() { - RTC_CHECK_EQ(0, T::Free(isac_state_)); -} - -template -int AudioEncoderIsacT::SampleRateHz() const { - return T::EncSampRate(isac_state_); -} - -template -size_t AudioEncoderIsacT::NumChannels() const { - return 1; -} - -template -size_t AudioEncoderIsacT::Num10MsFramesInNextPacket() const { - const int samples_in_next_packet = T::GetNewFrameLen(isac_state_); - return static_cast(rtc::CheckedDivExact( - samples_in_next_packet, rtc::CheckedDivExact(SampleRateHz(), 100))); -} - -template -size_t AudioEncoderIsacT::Max10MsFramesInAPacket() const { - return 6; // iSAC puts at most 60 ms in a packet. -} - -template -int AudioEncoderIsacT::GetTargetBitrate() const { - return config_.bit_rate == 0 ? kDefaultBitRate : config_.bit_rate; -} - -template -void AudioEncoderIsacT::SetTargetBitrate(int target_bps) { - // Set target bitrate directly without subtracting per-packet overhead, - // because that's what AudioEncoderOpus does. - SetTargetBitrate(target_bps, - /*subtract_per_packet_overhead=*/false); -} - -template -void AudioEncoderIsacT::OnReceivedTargetAudioBitrate(int target_bps) { - // Set target bitrate directly without subtracting per-packet overhead, - // because that's what AudioEncoderOpus does. - SetTargetBitrate(target_bps, - /*subtract_per_packet_overhead=*/false); -} - -template -void AudioEncoderIsacT::OnReceivedUplinkBandwidth( - int target_audio_bitrate_bps, - absl::optional /*bwe_period_ms*/) { - // Set target bitrate, subtracting the per-packet overhead if - // WebRTC-SendSideBwe-WithOverhead is enabled, because that's what - // AudioEncoderOpus does. - SetTargetBitrate( - target_audio_bitrate_bps, - /*subtract_per_packet_overhead=*/send_side_bwe_with_overhead_); -} - -template -void AudioEncoderIsacT::OnReceivedUplinkAllocation( - BitrateAllocationUpdate update) { - // Set target bitrate, subtracting the per-packet overhead if - // WebRTC-SendSideBwe-WithOverhead is enabled, because that's what - // AudioEncoderOpus does. - SetTargetBitrate( - update.target_bitrate.bps(), - /*subtract_per_packet_overhead=*/send_side_bwe_with_overhead_); -} - -template -void AudioEncoderIsacT::OnReceivedOverhead( - size_t overhead_bytes_per_packet) { - overhead_per_packet_ = DataSize::Bytes(overhead_bytes_per_packet); -} - -template -AudioEncoder::EncodedInfo AudioEncoderIsacT::EncodeImpl( - uint32_t rtp_timestamp, - rtc::ArrayView audio, - rtc::Buffer* encoded) { - if (!packet_in_progress_) { - // Starting a new packet; remember the timestamp for later. - packet_in_progress_ = true; - packet_timestamp_ = rtp_timestamp; - } - size_t encoded_bytes = encoded->AppendData( - kSufficientEncodeBufferSizeBytes, [&](rtc::ArrayView encoded) { - int r = T::Encode(isac_state_, audio.data(), encoded.data()); - - if (T::GetErrorCode(isac_state_) == 6450) { - // Isac is not able to effectively compress all types of signals. This - // is a limitation of the codec that cannot be easily fixed. - r = 0; - } - RTC_CHECK_GE(r, 0) << "Encode failed (error code " - << T::GetErrorCode(isac_state_) << ")"; - - return static_cast(r); - }); - - if (encoded_bytes == 0) - return EncodedInfo(); - - // Got enough input to produce a packet. Return the saved timestamp from - // the first chunk of input that went into the packet. - packet_in_progress_ = false; - EncodedInfo info; - info.encoded_bytes = encoded_bytes; - info.encoded_timestamp = packet_timestamp_; - info.payload_type = config_.payload_type; - info.encoder_type = CodecType::kIsac; - return info; -} - -template -void AudioEncoderIsacT::Reset() { - RecreateEncoderInstance(config_); -} - -template -absl::optional> -AudioEncoderIsacT::GetFrameLengthRange() const { - return {{TimeDelta::Millis(config_.frame_size_ms), - TimeDelta::Millis(config_.frame_size_ms)}}; -} - -template -void AudioEncoderIsacT::SetTargetBitrate(int target_bps, - bool subtract_per_packet_overhead) { - if (subtract_per_packet_overhead) { - const DataRate overhead_rate = - overhead_per_packet_ / TimeDelta::Millis(config_.frame_size_ms); - target_bps -= overhead_rate.bps(); - } - target_bps = rtc::SafeClamp(target_bps, kMinBitrateBps, - MaxBitrateBps(config_.sample_rate_hz)); - int result = T::Control(isac_state_, target_bps, config_.frame_size_ms); - RTC_DCHECK_EQ(result, 0); - config_.bit_rate = target_bps; -} - -template -void AudioEncoderIsacT::RecreateEncoderInstance(const Config& config) { - RTC_CHECK(config.IsOk()); - packet_in_progress_ = false; - if (isac_state_) - RTC_CHECK_EQ(0, T::Free(isac_state_)); - RTC_CHECK_EQ(0, T::Create(&isac_state_)); - RTC_CHECK_EQ(0, T::EncoderInit(isac_state_, /*coding_mode=*/1)); - RTC_CHECK_EQ(0, T::SetEncSampRate(isac_state_, config.sample_rate_hz)); - const int bit_rate = config.bit_rate == 0 ? kDefaultBitRate : config.bit_rate; - RTC_CHECK_EQ(0, T::Control(isac_state_, bit_rate, config.frame_size_ms)); - - if (config.max_payload_size_bytes != -1) - RTC_CHECK_EQ( - 0, T::SetMaxPayloadSize(isac_state_, config.max_payload_size_bytes)); - if (config.max_bit_rate != -1) - RTC_CHECK_EQ(0, T::SetMaxRate(isac_state_, config.max_bit_rate)); - - // Set the decoder sample rate even though we just use the encoder. This - // doesn't appear to be necessary to produce a valid encoding, but without it - // we get an encoding that isn't bit-for-bit identical with what a combined - // encoder+decoder object produces. - RTC_CHECK_EQ(0, T::SetDecSampRate(isac_state_, config.sample_rate_hz)); - - config_ = config; -} - -} // namespace webrtc - -#endif // MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_ENCODER_ISAC_T_IMPL_H_ diff --git a/modules/audio_coding/codecs/isac/empty.cc b/modules/audio_coding/codecs/isac/empty.cc deleted file mode 100644 index e69de29bb2..0000000000 diff --git a/modules/audio_coding/codecs/isac/fix/include/audio_decoder_isacfix.h b/modules/audio_coding/codecs/isac/fix/include/audio_decoder_isacfix.h deleted file mode 100644 index 0b4eadd448..0000000000 --- a/modules/audio_coding/codecs/isac/fix/include/audio_decoder_isacfix.h +++ /dev/null @@ -1,22 +0,0 @@ -/* - * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_FIX_INCLUDE_AUDIO_DECODER_ISACFIX_H_ -#define MODULES_AUDIO_CODING_CODECS_ISAC_FIX_INCLUDE_AUDIO_DECODER_ISACFIX_H_ - -#include "modules/audio_coding/codecs/isac/audio_decoder_isac_t.h" -#include "modules/audio_coding/codecs/isac/fix/source/isac_fix_type.h" - -namespace webrtc { - -using AudioDecoderIsacFixImpl = AudioDecoderIsacT; - -} // namespace webrtc -#endif // MODULES_AUDIO_CODING_CODECS_ISAC_FIX_INCLUDE_AUDIO_DECODER_ISACFIX_H_ diff --git a/modules/audio_coding/codecs/isac/fix/include/audio_encoder_isacfix.h b/modules/audio_coding/codecs/isac/fix/include/audio_encoder_isacfix.h deleted file mode 100644 index f0cc038328..0000000000 --- a/modules/audio_coding/codecs/isac/fix/include/audio_encoder_isacfix.h +++ /dev/null @@ -1,22 +0,0 @@ -/* - * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_FIX_INCLUDE_AUDIO_ENCODER_ISACFIX_H_ -#define MODULES_AUDIO_CODING_CODECS_ISAC_FIX_INCLUDE_AUDIO_ENCODER_ISACFIX_H_ - -#include "modules/audio_coding/codecs/isac/audio_encoder_isac_t.h" -#include "modules/audio_coding/codecs/isac/fix/source/isac_fix_type.h" - -namespace webrtc { - -using AudioEncoderIsacFixImpl = AudioEncoderIsacT; - -} // namespace webrtc -#endif // MODULES_AUDIO_CODING_CODECS_ISAC_FIX_INCLUDE_AUDIO_ENCODER_ISACFIX_H_ diff --git a/modules/audio_coding/codecs/isac/fix/include/isacfix.h b/modules/audio_coding/codecs/isac/fix/include/isacfix.h deleted file mode 100644 index dcc7b0991d..0000000000 --- a/modules/audio_coding/codecs/isac/fix/include/isacfix.h +++ /dev/null @@ -1,486 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_FIX_INCLUDE_ISACFIX_H_ -#define MODULES_AUDIO_CODING_CODECS_ISAC_FIX_INCLUDE_ISACFIX_H_ - -#include - -#include "modules/audio_coding/codecs/isac/bandwidth_info.h" - -typedef struct { - void* dummy; -} ISACFIX_MainStruct; - -#if defined(__cplusplus) -extern "C" { -#endif - -/**************************************************************************** - * WebRtcIsacfix_Create(...) - * - * This function creates an ISAC instance, which will contain the state - * information for one coding/decoding channel. - * - * Input: - * - *ISAC_main_inst : a pointer to the coder instance. - * - * Return value : 0 - Ok - * -1 - Error - */ - -int16_t WebRtcIsacfix_Create(ISACFIX_MainStruct** ISAC_main_inst); - -/**************************************************************************** - * WebRtcIsacfix_Free(...) - * - * This function frees the ISAC instance created at the beginning. - * - * Input: - * - ISAC_main_inst : a ISAC instance. - * - * Return value : 0 - Ok - * -1 - Error - */ - -int16_t WebRtcIsacfix_Free(ISACFIX_MainStruct* ISAC_main_inst); - -/**************************************************************************** - * WebRtcIsacfix_EncoderInit(...) - * - * This function initializes an ISAC instance prior to the encoder calls. - * - * Input: - * - ISAC_main_inst : ISAC instance. - * - CodingMode : 0 - Bit rate and frame length are automatically - * adjusted to available bandwidth on - * transmission channel. - * 1 - User sets a frame length and a target bit - * rate which is taken as the maximum short-term - * average bit rate. - * - * Return value : 0 - Ok - * -1 - Error - */ - -int16_t WebRtcIsacfix_EncoderInit(ISACFIX_MainStruct* ISAC_main_inst, - int16_t CodingMode); - -/**************************************************************************** - * WebRtcIsacfix_Encode(...) - * - * This function encodes 10ms frame(s) and inserts it into a package. - * Input speech length has to be 160 samples (10ms). The encoder buffers those - * 10ms frames until it reaches the chosen Framesize (480 or 960 samples - * corresponding to 30 or 60 ms frames), and then proceeds to the encoding. - * - * Input: - * - ISAC_main_inst : ISAC instance. - * - speechIn : input speech vector. - * - * Output: - * - encoded : the encoded data vector - * - * Return value : >0 - Length (in bytes) of coded data - * 0 - The buffer didn't reach the chosen framesize - * so it keeps buffering speech samples. - * -1 - Error - */ - -int WebRtcIsacfix_Encode(ISACFIX_MainStruct* ISAC_main_inst, - const int16_t* speechIn, - uint8_t* encoded); - -/**************************************************************************** - * WebRtcIsacfix_DecoderInit(...) - * - * This function initializes an ISAC instance prior to the decoder calls. - * - * Input: - * - ISAC_main_inst : ISAC instance. - */ - -void WebRtcIsacfix_DecoderInit(ISACFIX_MainStruct* ISAC_main_inst); - -/**************************************************************************** - * WebRtcIsacfix_UpdateBwEstimate1(...) - * - * This function updates the estimate of the bandwidth. - * - * Input: - * - ISAC_main_inst : ISAC instance. - * - encoded : encoded ISAC frame(s). - * - packet_size : size of the packet in bytes. - * - rtp_seq_number : the RTP number of the packet. - * - arr_ts : the arrival time of the packet (from NetEq) - * in samples. - * - * Return value : 0 - Ok - * -1 - Error - */ - -int16_t WebRtcIsacfix_UpdateBwEstimate1(ISACFIX_MainStruct* ISAC_main_inst, - const uint8_t* encoded, - size_t packet_size, - uint16_t rtp_seq_number, - uint32_t arr_ts); - -/**************************************************************************** - * WebRtcIsacfix_UpdateBwEstimate(...) - * - * This function updates the estimate of the bandwidth. - * - * Input: - * - ISAC_main_inst : ISAC instance. - * - encoded : encoded ISAC frame(s). - * - packet_size : size of the packet in bytes. - * - rtp_seq_number : the RTP number of the packet. - * - send_ts : the send time of the packet from RTP header, - * in samples. - * - arr_ts : the arrival time of the packet (from NetEq) - * in samples. - * - * Return value : 0 - Ok - * -1 - Error - */ - -int16_t WebRtcIsacfix_UpdateBwEstimate(ISACFIX_MainStruct* ISAC_main_inst, - const uint8_t* encoded, - size_t packet_size, - uint16_t rtp_seq_number, - uint32_t send_ts, - uint32_t arr_ts); - -/**************************************************************************** - * WebRtcIsacfix_Decode(...) - * - * This function decodes an ISAC frame. Output speech length - * will be a multiple of 480 samples: 480 or 960 samples, - * depending on the framesize (30 or 60 ms). - * - * Input: - * - ISAC_main_inst : ISAC instance. - * - encoded : encoded ISAC frame(s) - * - len : bytes in encoded vector - * - * Output: - * - decoded : The decoded vector - * - * Return value : >0 - number of samples in decoded vector - * -1 - Error - */ - -int WebRtcIsacfix_Decode(ISACFIX_MainStruct* ISAC_main_inst, - const uint8_t* encoded, - size_t len, - int16_t* decoded, - int16_t* speechType); - -/**************************************************************************** - * WebRtcIsacfix_DecodePlc(...) - * - * This function conducts PLC for ISAC frame(s) in wide-band (16kHz sampling). - * Output speech length will be "480*noOfLostFrames" samples - * that is equevalent of "30*noOfLostFrames" millisecond. - * - * Input: - * - ISAC_main_inst : ISAC instance. - * - noOfLostFrames : Number of PLC frames (480sample = 30ms) - * to produce - * NOTE! Maximum number is 2 (960 samples = 60ms) - * - * Output: - * - decoded : The decoded vector - * - * Return value : Number of samples in decoded PLC vector - */ - -size_t WebRtcIsacfix_DecodePlc(ISACFIX_MainStruct* ISAC_main_inst, - int16_t* decoded, - size_t noOfLostFrames); - -/**************************************************************************** - * WebRtcIsacfix_ReadFrameLen(...) - * - * This function returns the length of the frame represented in the packet. - * - * Input: - * - encoded : Encoded bitstream - * - encoded_len_bytes : Length of the bitstream in bytes. - * - * Output: - * - frameLength : Length of frame in packet (in samples) - * - */ - -int16_t WebRtcIsacfix_ReadFrameLen(const uint8_t* encoded, - size_t encoded_len_bytes, - size_t* frameLength); - -/**************************************************************************** - * WebRtcIsacfix_Control(...) - * - * This function sets the limit on the short-term average bit rate and the - * frame length. Should be used only in Instantaneous mode. - * - * Input: - * - ISAC_main_inst : ISAC instance. - * - rate : limit on the short-term average bit rate, - * in bits/second (between 10000 and 32000) - * - framesize : number of milliseconds per frame (30 or 60) - * - * Return value : 0 - ok - * -1 - Error - */ - -int16_t WebRtcIsacfix_Control(ISACFIX_MainStruct* ISAC_main_inst, - int16_t rate, - int framesize); - -void WebRtcIsacfix_SetInitialBweBottleneck(ISACFIX_MainStruct* ISAC_main_inst, - int bottleneck_bits_per_second); - -/**************************************************************************** - * WebRtcIsacfix_ControlBwe(...) - * - * This function sets the initial values of bottleneck and frame-size if - * iSAC is used in channel-adaptive mode. Through this API, users can - * enforce a frame-size for all values of bottleneck. Then iSAC will not - * automatically change the frame-size. - * - * - * Input: - * - ISAC_main_inst : ISAC instance. - * - rateBPS : initial value of bottleneck in bits/second - * 10000 <= rateBPS <= 32000 is accepted - * - frameSizeMs : number of milliseconds per frame (30 or 60) - * - enforceFrameSize : 1 to enforce the given frame-size through out - * the adaptation process, 0 to let iSAC change - * the frame-size if required. - * - * Return value : 0 - ok - * -1 - Error - */ - -int16_t WebRtcIsacfix_ControlBwe(ISACFIX_MainStruct* ISAC_main_inst, - int16_t rateBPS, - int frameSizeMs, - int16_t enforceFrameSize); - -/**************************************************************************** - * WebRtcIsacfix_version(...) - * - * This function returns the version number. - * - * Output: - * - version : Pointer to character string - * - */ - -void WebRtcIsacfix_version(char* version); - -/**************************************************************************** - * WebRtcIsacfix_GetErrorCode(...) - * - * This function can be used to check the error code of an iSAC instance. When - * a function returns -1 a error code will be set for that instance. The - * function below extract the code of the last error that occured in the - * specified instance. - * - * Input: - * - ISAC_main_inst : ISAC instance - * - * Return value : Error code - */ - -int16_t WebRtcIsacfix_GetErrorCode(ISACFIX_MainStruct* ISAC_main_inst); - -/**************************************************************************** - * WebRtcIsacfix_GetUplinkBw(...) - * - * This function return iSAC send bitrate - * - * Input: - * - ISAC_main_inst : iSAC instance - * - * Return value : <0 Error code - * else bitrate - */ - -int32_t WebRtcIsacfix_GetUplinkBw(ISACFIX_MainStruct* ISAC_main_inst); - -/**************************************************************************** - * WebRtcIsacfix_SetMaxPayloadSize(...) - * - * This function sets a limit for the maximum payload size of iSAC. The same - * value is used both for 30 and 60 msec packets. - * The absolute max will be valid until next time the function is called. - * NOTE! This function may override the function WebRtcIsacfix_SetMaxRate() - * - * Input: - * - ISAC_main_inst : iSAC instance - * - maxPayloadBytes : maximum size of the payload in bytes - * valid values are between 100 and 400 bytes - * - * - * Return value : 0 if sucessful - * -1 if error happens - */ - -int16_t WebRtcIsacfix_SetMaxPayloadSize(ISACFIX_MainStruct* ISAC_main_inst, - int16_t maxPayloadBytes); - -/**************************************************************************** - * WebRtcIsacfix_SetMaxRate(...) - * - * This function sets the maximum rate which the codec may not exceed for a - * singel packet. The maximum rate is set in bits per second. - * The codec has an absolute maximum rate of 53400 bits per second (200 bytes - * per 30 msec). - * It is possible to set a maximum rate between 32000 and 53400 bits per second. - * - * The rate limit is valid until next time the function is called. - * - * NOTE! Packet size will never go above the value set if calling - * WebRtcIsacfix_SetMaxPayloadSize() (default max packet size is 400 bytes). - * - * Input: - * - ISAC_main_inst : iSAC instance - * - maxRateInBytes : maximum rate in bits per second, - * valid values are 32000 to 53400 bits - * - * Return value : 0 if sucessful - * -1 if error happens - */ - -int16_t WebRtcIsacfix_SetMaxRate(ISACFIX_MainStruct* ISAC_main_inst, - int32_t maxRate); - -/**************************************************************************** - * WebRtcIsacfix_CreateInternal(...) - * - * This function creates the memory that is used to store data in the encoder - * - * Input: - * - *ISAC_main_inst : a pointer to the coder instance. - * - * Return value : 0 - Ok - * -1 - Error - */ - -int16_t WebRtcIsacfix_CreateInternal(ISACFIX_MainStruct* ISAC_main_inst); - -/**************************************************************************** - * WebRtcIsacfix_FreeInternal(...) - * - * This function frees the internal memory for storing encoder data. - * - * Input: - * - ISAC_main_inst : an ISAC instance. - * - * Return value : 0 - Ok - * -1 - Error - */ - -int16_t WebRtcIsacfix_FreeInternal(ISACFIX_MainStruct* ISAC_main_inst); - -/**************************************************************************** - * WebRtcIsacfix_GetNewBitStream(...) - * - * This function returns encoded data, with the received bwe-index in the - * stream. It should always return a complete packet, i.e. only called once - * even for 60 msec frames - * - * Input: - * - ISAC_main_inst : ISAC instance. - * - bweIndex : index of bandwidth estimate to put in new - * bitstream - scale : factor for rate change (0.4 ~=> half the - * rate, 1 no change). - * - * Output: - * - encoded : the encoded data vector - * - * Return value : >0 - Length (in bytes) of coded data - * -1 - Error - */ - -int16_t WebRtcIsacfix_GetNewBitStream(ISACFIX_MainStruct* ISAC_main_inst, - int16_t bweIndex, - float scale, - uint8_t* encoded); - -/**************************************************************************** - * WebRtcIsacfix_GetDownLinkBwIndex(...) - * - * This function returns index representing the Bandwidth estimate from - * other side to this side. - * - * Input: - * - ISAC_main_inst : iSAC struct - * - * Output: - * - rateIndex : Bandwidth estimate to transmit to other side. - * - */ - -int16_t WebRtcIsacfix_GetDownLinkBwIndex(ISACFIX_MainStruct* ISAC_main_inst, - int16_t* rateIndex); - -/**************************************************************************** - * WebRtcIsacfix_UpdateUplinkBw(...) - * - * This function takes an index representing the Bandwidth estimate from - * this side to other side and updates BWE. - * - * Input: - * - ISAC_main_inst : iSAC struct - * - rateIndex : Bandwidth estimate from other side. - * - */ - -int16_t WebRtcIsacfix_UpdateUplinkBw(ISACFIX_MainStruct* ISAC_main_inst, - int16_t rateIndex); - -/**************************************************************************** - * WebRtcIsacfix_ReadBwIndex(...) - * - * This function returns the index of the Bandwidth estimate from the bitstream. - * - * Input: - * - encoded : Encoded bitstream - * - encoded_len_bytes : Length of the bitstream in bytes. - * - * Output: - * - rateIndex : Bandwidth estimate in bitstream - * - */ - -int16_t WebRtcIsacfix_ReadBwIndex(const uint8_t* encoded, - size_t encoded_len_bytes, - int16_t* rateIndex); - -/**************************************************************************** - * WebRtcIsacfix_GetNewFrameLen(...) - * - * This function return the next frame length (in samples) of iSAC. - * - * Input: - * -ISAC_main_inst : iSAC instance - * - * Return value : frame lenght in samples - */ - -int16_t WebRtcIsacfix_GetNewFrameLen(ISACFIX_MainStruct* ISAC_main_inst); - -#if defined(__cplusplus) -} -#endif - -#endif /* MODULES_AUDIO_CODING_CODECS_ISAC_FIX_INCLUDE_ISACFIX_H_ */ diff --git a/modules/audio_coding/codecs/isac/fix/source/arith_routines.c b/modules/audio_coding/codecs/isac/fix/source/arith_routines.c deleted file mode 100644 index eaeef50f04..0000000000 --- a/modules/audio_coding/codecs/isac/fix/source/arith_routines.c +++ /dev/null @@ -1,122 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/* - * arith_routins.c - * - * This C file contains a function for finalizing the bitstream - * after arithmetic coding. - * - */ - -#include "modules/audio_coding/codecs/isac/fix/source/arith_routins.h" - - -/**************************************************************************** - * WebRtcIsacfix_EncTerminate(...) - * - * Final call to the arithmetic coder for an encoder call. This function - * terminates and return byte stream. - * - * Input: - * - streamData : in-/output struct containing bitstream - * - * Return value : number of bytes in the stream - */ -int16_t WebRtcIsacfix_EncTerminate(Bitstr_enc *streamData) -{ - uint16_t *streamPtr; - uint16_t negCarry; - - /* point to the right place in the stream buffer */ - streamPtr = streamData->stream + streamData->stream_index; - - /* find minimum length (determined by current interval width) */ - if ( streamData->W_upper > 0x01FFFFFF ) - { - streamData->streamval += 0x01000000; - - /* if result is less than the added value we must take care of the carry */ - if (streamData->streamval < 0x01000000) - { - /* propagate carry */ - if (streamData->full == 0) { - /* Add value to current value */ - negCarry = *streamPtr; - negCarry += 0x0100; - *streamPtr = negCarry; - - /* if value is too big, propagate carry to next byte, and so on */ - while (!(negCarry)) - { - negCarry = *--streamPtr; - negCarry++; - *streamPtr = negCarry; - } - } else { - /* propagate carry by adding one to the previous byte in the - * stream if that byte is 0xFFFF we need to propagate the carry - * furhter back in the stream */ - while ( !(++(*--streamPtr)) ); - } - - /* put pointer back to the old value */ - streamPtr = streamData->stream + streamData->stream_index; - } - /* write remaining data to bitstream, if "full == 0" first byte has data */ - if (streamData->full == 0) { - *streamPtr++ += (uint16_t)(streamData->streamval >> 24); - streamData->full = 1; - } else { - *streamPtr = (uint16_t)((streamData->streamval >> 24) << 8); - streamData->full = 0; - } - } - else - { - streamData->streamval += 0x00010000; - - /* if result is less than the added value we must take care of the carry */ - if (streamData->streamval < 0x00010000) - { - /* propagate carry */ - if (streamData->full == 0) { - /* Add value to current value */ - negCarry = *streamPtr; - negCarry += 0x0100; - *streamPtr = negCarry; - - /* if value to big, propagate carry to next byte, and so on */ - while (!(negCarry)) - { - negCarry = *--streamPtr; - negCarry++; - *streamPtr = negCarry; - } - } else { - /* Add carry to previous byte */ - while ( !(++(*--streamPtr)) ); - } - - /* put pointer back to the old value */ - streamPtr = streamData->stream + streamData->stream_index; - } - /* write remaining data (2 bytes) to bitstream */ - if (streamData->full) { - *streamPtr++ = (uint16_t)(streamData->streamval >> 16); - } else { - *streamPtr++ |= (uint16_t)(streamData->streamval >> 24); - *streamPtr = (uint16_t)(streamData->streamval >> 8) & 0xFF00; - } - } - - /* calculate stream length in bytes */ - return (((streamPtr - streamData->stream)<<1) + !(streamData->full)); -} diff --git a/modules/audio_coding/codecs/isac/fix/source/arith_routines_hist.c b/modules/audio_coding/codecs/isac/fix/source/arith_routines_hist.c deleted file mode 100644 index cad3056b37..0000000000 --- a/modules/audio_coding/codecs/isac/fix/source/arith_routines_hist.c +++ /dev/null @@ -1,401 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/* - * arith_routinshist.c - * - * This C file contains arithmetic encoding and decoding. - * - */ - -#include "modules/audio_coding/codecs/isac/fix/source/arith_routins.h" - - -/**************************************************************************** - * WebRtcIsacfix_EncHistMulti(...) - * - * Encode the histogram interval - * - * Input: - * - streamData : in-/output struct containing bitstream - * - data : data vector - * - cdf : array of cdf arrays - * - lenData : data vector length - * - * Return value : 0 if ok - * <0 if error detected - */ -int WebRtcIsacfix_EncHistMulti(Bitstr_enc *streamData, - const int16_t *data, - const uint16_t *const *cdf, - const int16_t lenData) -{ - uint32_t W_lower; - uint32_t W_upper; - uint32_t W_upper_LSB; - uint32_t W_upper_MSB; - uint16_t *streamPtr; - uint16_t negCarry; - uint16_t *maxStreamPtr; - uint16_t *streamPtrCarry; - uint32_t cdfLo; - uint32_t cdfHi; - int k; - - - /* point to beginning of stream buffer - * and set maximum streamPtr value */ - streamPtr = streamData->stream + streamData->stream_index; - maxStreamPtr = streamData->stream + STREAM_MAXW16_60MS - 1; - - W_upper = streamData->W_upper; - - for (k = lenData; k > 0; k--) - { - /* fetch cdf_lower and cdf_upper from cdf tables */ - cdfLo = (uint32_t) *(*cdf + (uint32_t)*data); - cdfHi = (uint32_t) *(*cdf++ + (uint32_t)*data++ + 1); - - /* update interval */ - W_upper_LSB = W_upper & 0x0000FFFF; - W_upper_MSB = W_upper >> 16; - W_lower = WEBRTC_SPL_UMUL(W_upper_MSB, cdfLo); - W_lower += ((W_upper_LSB * cdfLo) >> 16); - W_upper = WEBRTC_SPL_UMUL(W_upper_MSB, cdfHi); - W_upper += ((W_upper_LSB * cdfHi) >> 16); - - /* shift interval such that it begins at zero */ - W_upper -= ++W_lower; - - /* add integer to bitstream */ - streamData->streamval += W_lower; - - /* handle carry */ - if (streamData->streamval < W_lower) - { - /* propagate carry */ - streamPtrCarry = streamPtr; - if (streamData->full == 0) { - negCarry = *streamPtrCarry; - negCarry += 0x0100; - *streamPtrCarry = negCarry; - while (!(negCarry)) - { - negCarry = *--streamPtrCarry; - negCarry++; - *streamPtrCarry = negCarry; - } - } else { - while ( !(++(*--streamPtrCarry)) ); - } - } - - /* renormalize interval, store most significant byte of streamval and update streamval - * W_upper < 2^24 */ - while ( !(W_upper & 0xFF000000) ) - { - W_upper <<= 8; - if (streamData->full == 0) { - *streamPtr++ += (uint16_t)(streamData->streamval >> 24); - streamData->full = 1; - } else { - *streamPtr = (uint16_t)((streamData->streamval >> 24) << 8); - streamData->full = 0; - } - - if( streamPtr > maxStreamPtr ) { - return -ISAC_DISALLOWED_BITSTREAM_LENGTH; - } - streamData->streamval <<= 8; - } - } - - /* calculate new stream_index */ - streamData->stream_index = streamPtr - streamData->stream; - streamData->W_upper = W_upper; - - return 0; -} - - -/**************************************************************************** - * WebRtcIsacfix_DecHistBisectMulti(...) - * - * Function to decode more symbols from the arithmetic bytestream, using - * method of bisection cdf tables should be of size 2^k-1 (which corresponds - * to an alphabet size of 2^k-2) - * - * Input: - * - streamData : in-/output struct containing bitstream - * - cdf : array of cdf arrays - * - cdfSize : array of cdf table sizes+1 (power of two: 2^k) - * - lenData : data vector length - * - * Output: - * - data : data vector - * - * Return value : number of bytes in the stream - * <0 if error detected - */ -int16_t WebRtcIsacfix_DecHistBisectMulti(int16_t *data, - Bitstr_dec *streamData, - const uint16_t *const *cdf, - const uint16_t *cdfSize, - const int16_t lenData) -{ - uint32_t W_lower = 0; - uint32_t W_upper; - uint32_t W_tmp; - uint32_t W_upper_LSB; - uint32_t W_upper_MSB; - uint32_t streamval; - const uint16_t *streamPtr; - const uint16_t *cdfPtr; - int16_t sizeTmp; - int k; - - - streamPtr = streamData->stream + streamData->stream_index; - W_upper = streamData->W_upper; - - /* Error check: should not be possible in normal operation */ - if (W_upper == 0) { - return -2; - } - - /* first time decoder is called for this stream */ - if (streamData->stream_index == 0) - { - /* read first word from bytestream */ - streamval = (uint32_t)*streamPtr++ << 16; - streamval |= *streamPtr++; - } else { - streamval = streamData->streamval; - } - - for (k = lenData; k > 0; k--) - { - /* find the integer *data for which streamval lies in [W_lower+1, W_upper] */ - W_upper_LSB = W_upper & 0x0000FFFF; - W_upper_MSB = W_upper >> 16; - - /* start halfway the cdf range */ - sizeTmp = *cdfSize++ / 2; - cdfPtr = *cdf + (sizeTmp - 1); - - /* method of bisection */ - for ( ;; ) - { - W_tmp = WEBRTC_SPL_UMUL_32_16(W_upper_MSB, *cdfPtr); - W_tmp += (W_upper_LSB * (*cdfPtr)) >> 16; - sizeTmp /= 2; - if (sizeTmp == 0) { - break; - } - - if (streamval > W_tmp) - { - W_lower = W_tmp; - cdfPtr += sizeTmp; - } else { - W_upper = W_tmp; - cdfPtr -= sizeTmp; - } - } - if (streamval > W_tmp) - { - W_lower = W_tmp; - *data++ = cdfPtr - *cdf++; - } else { - W_upper = W_tmp; - *data++ = cdfPtr - *cdf++ - 1; - } - - /* shift interval to start at zero */ - W_upper -= ++W_lower; - - /* add integer to bitstream */ - streamval -= W_lower; - - /* renormalize interval and update streamval */ - /* W_upper < 2^24 */ - while ( !(W_upper & 0xFF000000) ) - { - /* read next byte from stream */ - if (streamData->full == 0) { - streamval = (streamval << 8) | (*streamPtr++ & 0x00FF); - streamData->full = 1; - } else { - streamval = (streamval << 8) | (*streamPtr >> 8); - streamData->full = 0; - } - W_upper <<= 8; - } - - - /* Error check: should not be possible in normal operation */ - if (W_upper == 0) { - return -2; - } - - } - - streamData->stream_index = streamPtr - streamData->stream; - streamData->W_upper = W_upper; - streamData->streamval = streamval; - - if ( W_upper > 0x01FFFFFF ) { - return (streamData->stream_index*2 - 3 + !streamData->full); - } else { - return (streamData->stream_index*2 - 2 + !streamData->full); - } -} - - -/**************************************************************************** - * WebRtcIsacfix_DecHistOneStepMulti(...) - * - * Function to decode more symbols from the arithmetic bytestream, taking - * single step up or down at a time. - * cdf tables can be of arbitrary size, but large tables may take a lot of - * iterations. - * - * Input: - * - streamData : in-/output struct containing bitstream - * - cdf : array of cdf arrays - * - initIndex : vector of initial cdf table search entries - * - lenData : data vector length - * - * Output: - * - data : data vector - * - * Return value : number of bytes in original stream - * <0 if error detected - */ -int16_t WebRtcIsacfix_DecHistOneStepMulti(int16_t *data, - Bitstr_dec *streamData, - const uint16_t *const *cdf, - const uint16_t *initIndex, - const int16_t lenData) -{ - uint32_t W_lower; - uint32_t W_upper; - uint32_t W_tmp; - uint32_t W_upper_LSB; - uint32_t W_upper_MSB; - uint32_t streamval; - const uint16_t *streamPtr; - const uint16_t *cdfPtr; - int k; - - - streamPtr = streamData->stream + streamData->stream_index; - W_upper = streamData->W_upper; - /* Error check: Should not be possible in normal operation */ - if (W_upper == 0) { - return -2; - } - - /* Check if it is the first time decoder is called for this stream */ - if (streamData->stream_index == 0) - { - /* read first word from bytestream */ - streamval = (uint32_t)(*streamPtr++) << 16; - streamval |= *streamPtr++; - } else { - streamval = streamData->streamval; - } - - for (k = lenData; k > 0; k--) - { - /* find the integer *data for which streamval lies in [W_lower+1, W_upper] */ - W_upper_LSB = W_upper & 0x0000FFFF; - W_upper_MSB = WEBRTC_SPL_RSHIFT_U32(W_upper, 16); - - /* start at the specified table entry */ - cdfPtr = *cdf + (*initIndex++); - W_tmp = WEBRTC_SPL_UMUL_32_16(W_upper_MSB, *cdfPtr); - W_tmp += (W_upper_LSB * (*cdfPtr)) >> 16; - - if (streamval > W_tmp) - { - for ( ;; ) - { - W_lower = W_tmp; - - /* range check */ - if (cdfPtr[0] == 65535) { - return -3; - } - - W_tmp = WEBRTC_SPL_UMUL_32_16(W_upper_MSB, *++cdfPtr); - W_tmp += (W_upper_LSB * (*cdfPtr)) >> 16; - - if (streamval <= W_tmp) { - break; - } - } - W_upper = W_tmp; - *data++ = cdfPtr - *cdf++ - 1; - } else { - for ( ;; ) - { - W_upper = W_tmp; - --cdfPtr; - - /* range check */ - if (cdfPtr < *cdf) { - return -3; - } - - W_tmp = WEBRTC_SPL_UMUL_32_16(W_upper_MSB, *cdfPtr); - W_tmp += (W_upper_LSB * (*cdfPtr)) >> 16; - - if (streamval > W_tmp) { - break; - } - } - W_lower = W_tmp; - *data++ = cdfPtr - *cdf++; - } - - /* shift interval to start at zero */ - W_upper -= ++W_lower; - - /* add integer to bitstream */ - streamval -= W_lower; - - /* renormalize interval and update streamval */ - /* W_upper < 2^24 */ - while ( !(W_upper & 0xFF000000) ) - { - /* read next byte from stream */ - if (streamData->full == 0) { - streamval = (streamval << 8) | (*streamPtr++ & 0x00FF); - streamData->full = 1; - } else { - streamval = (streamval << 8) | (*streamPtr >> 8); - streamData->full = 0; - } - W_upper <<= 8; - } - } - - streamData->stream_index = streamPtr - streamData->stream; - streamData->W_upper = W_upper; - streamData->streamval = streamval; - - /* find number of bytes in original stream (determined by current interval width) */ - if ( W_upper > 0x01FFFFFF ) { - return (streamData->stream_index*2 - 3 + !streamData->full); - } else { - return (streamData->stream_index*2 - 2 + !streamData->full); - } -} diff --git a/modules/audio_coding/codecs/isac/fix/source/arith_routines_logist.c b/modules/audio_coding/codecs/isac/fix/source/arith_routines_logist.c deleted file mode 100644 index 8e97960461..0000000000 --- a/modules/audio_coding/codecs/isac/fix/source/arith_routines_logist.c +++ /dev/null @@ -1,413 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/* - * arith_routinslogist.c - * - * This C file contains arithmetic encode and decode logistic - * - */ - -#include "modules/audio_coding/codecs/isac/fix/source/arith_routins.h" - -/* Tables for piecewise linear cdf functions: y = k*x */ - -/* x Points for function piecewise() in Q15 */ -static const int32_t kHistEdges[51] = { - -327680, -314573, -301466, -288359, -275252, -262144, -249037, -235930, -222823, -209716, - -196608, -183501, -170394, -157287, -144180, -131072, -117965, -104858, -91751, -78644, - -65536, -52429, -39322, -26215, -13108, 0, 13107, 26214, 39321, 52428, - 65536, 78643, 91750, 104857, 117964, 131072, 144179, 157286, 170393, 183500, - 196608, 209715, 222822, 235929, 249036, 262144, 275251, 288358, 301465, 314572, - 327680 -}; - - -/* k Points for function piecewise() in Q0 */ -static const uint16_t kCdfSlope[51] = { - 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, - 5, 5, 13, 23, 47, 87, 154, 315, 700, 1088, - 2471, 6064, 14221, 21463, 36634, 36924, 19750, 13270, 5806, 2312, - 1095, 660, 316, 145, 86, 41, 32, 5, 5, 5, - 5, 5, 5, 5, 5, 5, 5, 5, 5, 2, - 0 -}; - -/* y Points for function piecewise() in Q0 */ -static const uint16_t kCdfLogistic[51] = { - 0, 2, 4, 6, 8, 10, 12, 14, 16, 18, - 20, 22, 24, 29, 38, 57, 92, 153, 279, 559, - 994, 1983, 4408, 10097, 18682, 33336, 48105, 56005, 61313, 63636, - 64560, 64998, 65262, 65389, 65447, 65481, 65497, 65510, 65512, 65514, - 65516, 65518, 65520, 65522, 65524, 65526, 65528, 65530, 65532, 65534, - 65535 -}; - - -/**************************************************************************** - * WebRtcIsacfix_Piecewise(...) - * - * Piecewise linear function - * - * Input: - * - xinQ15 : input value x in Q15 - * - * Return value : korresponding y-value in Q0 - */ - - -static __inline uint16_t WebRtcIsacfix_Piecewise(int32_t xinQ15) { - int32_t ind; - int32_t qtmp1; - uint16_t qtmp2; - - /* Find index for x-value */ - qtmp1 = WEBRTC_SPL_SAT(kHistEdges[50],xinQ15,kHistEdges[0]); - ind = WEBRTC_SPL_MUL(5, qtmp1 - kHistEdges[0]); - ind >>= 16; - - /* Calculate corresponding y-value ans return*/ - qtmp1 = qtmp1 - kHistEdges[ind]; - qtmp2 = (uint16_t)WEBRTC_SPL_RSHIFT_U32( - WEBRTC_SPL_UMUL_32_16(qtmp1,kCdfSlope[ind]), 15); - return (kCdfLogistic[ind] + qtmp2); -} - -/**************************************************************************** - * WebRtcIsacfix_EncLogisticMulti2(...) - * - * Arithmetic coding of spectrum. - * - * Input: - * - streamData : in-/output struct containing bitstream - * - dataQ7 : data vector in Q7 - * - envQ8 : side info vector defining the width of the pdf - * in Q8 - * - lenData : data vector length - * - * Return value : 0 if ok, - * <0 otherwise. - */ -int WebRtcIsacfix_EncLogisticMulti2(Bitstr_enc *streamData, - int16_t *dataQ7, - const uint16_t *envQ8, - const int16_t lenData) -{ - uint32_t W_lower; - uint32_t W_upper; - uint16_t W_upper_LSB; - uint16_t W_upper_MSB; - uint16_t *streamPtr; - uint16_t *maxStreamPtr; - uint16_t *streamPtrCarry; - uint16_t negcarry; - uint32_t cdfLo; - uint32_t cdfHi; - int k; - - /* point to beginning of stream buffer - * and set maximum streamPtr value */ - streamPtr = streamData->stream + streamData->stream_index; - maxStreamPtr = streamData->stream + STREAM_MAXW16_60MS - 1; - W_upper = streamData->W_upper; - - for (k = 0; k < lenData; k++) - { - /* compute cdf_lower and cdf_upper by evaluating the - * WebRtcIsacfix_Piecewise linear cdf */ - cdfLo = WebRtcIsacfix_Piecewise(WEBRTC_SPL_MUL_16_U16(*dataQ7 - 64, *envQ8)); - cdfHi = WebRtcIsacfix_Piecewise(WEBRTC_SPL_MUL_16_U16(*dataQ7 + 64, *envQ8)); - - /* test and clip if probability gets too small */ - while ((cdfLo + 1) >= cdfHi) { - /* clip */ - if (*dataQ7 > 0) { - *dataQ7 -= 128; - cdfHi = cdfLo; - cdfLo = WebRtcIsacfix_Piecewise( - WEBRTC_SPL_MUL_16_U16(*dataQ7 - 64, *envQ8)); - } else { - *dataQ7 += 128; - cdfLo = cdfHi; - cdfHi = WebRtcIsacfix_Piecewise( - WEBRTC_SPL_MUL_16_U16(*dataQ7 + 64, *envQ8)); - } - } - - dataQ7++; - /* increment only once per 4 iterations */ - envQ8 += (k & 1) & (k >> 1); - - - /* update interval */ - W_upper_LSB = (uint16_t)W_upper; - W_upper_MSB = (uint16_t)WEBRTC_SPL_RSHIFT_U32(W_upper, 16); - W_lower = WEBRTC_SPL_UMUL_32_16(cdfLo, W_upper_MSB); - W_lower += (cdfLo * W_upper_LSB) >> 16; - W_upper = WEBRTC_SPL_UMUL_32_16(cdfHi, W_upper_MSB); - W_upper += (cdfHi * W_upper_LSB) >> 16; - - /* shift interval such that it begins at zero */ - W_upper -= ++W_lower; - - /* add integer to bitstream */ - streamData->streamval += W_lower; - - /* handle carry */ - if (streamData->streamval < W_lower) - { - /* propagate carry */ - streamPtrCarry = streamPtr; - if (streamData->full == 0) { - negcarry = *streamPtrCarry; - negcarry += 0x0100; - *streamPtrCarry = negcarry; - while (!(negcarry)) - { - negcarry = *--streamPtrCarry; - negcarry++; - *streamPtrCarry = negcarry; - } - } else { - while (!(++(*--streamPtrCarry))); - } - } - - /* renormalize interval, store most significant byte of streamval and update streamval - * W_upper < 2^24 */ - while ( !(W_upper & 0xFF000000) ) - { - W_upper <<= 8; - if (streamData->full == 0) { - *streamPtr++ += (uint16_t) WEBRTC_SPL_RSHIFT_U32( - streamData->streamval, 24); - streamData->full = 1; - } else { - *streamPtr = (uint16_t)((streamData->streamval >> 24) << 8); - streamData->full = 0; - } - - if( streamPtr > maxStreamPtr ) - return -ISAC_DISALLOWED_BITSTREAM_LENGTH; - - streamData->streamval <<= 8; - } - } - - /* calculate new stream_index */ - streamData->stream_index = streamPtr - streamData->stream; - streamData->W_upper = W_upper; - - return 0; -} - - -/**************************************************************************** - * WebRtcIsacfix_DecLogisticMulti2(...) - * - * Arithmetic decoding of spectrum. - * - * Input: - * - streamData : in-/output struct containing bitstream - * - envQ8 : side info vector defining the width of the pdf - * in Q8 - * - lenData : data vector length - * - * Input/Output: - * - dataQ7 : input: dither vector, output: data vector - * - * Return value : number of bytes in the stream so far - * -1 if error detected - */ -int WebRtcIsacfix_DecLogisticMulti2(int16_t *dataQ7, - Bitstr_dec *streamData, - const int32_t *envQ8, - const int16_t lenData) -{ - uint32_t W_lower; - uint32_t W_upper; - uint32_t W_tmp; - uint16_t W_upper_LSB; - uint16_t W_upper_MSB; - uint32_t streamVal; - uint16_t cdfTmp; - int32_t res; - int32_t inSqrt; - int32_t newRes; - const uint16_t *streamPtr; - int16_t candQ7; - int16_t envCount; - uint16_t tmpARSpecQ8 = 0; - int k, i; - int offset = 0; - - /* point to beginning of stream buffer */ - streamPtr = streamData->stream + streamData->stream_index; - W_upper = streamData->W_upper; - - /* Check if it is first time decoder is called for this stream */ - if (streamData->stream_index == 0) - { - /* read first word from bytestream */ - streamVal = (uint32_t)(*streamPtr++) << 16; - streamVal |= *streamPtr++; - - } else { - streamVal = streamData->streamval; - } - - - res = 1 << (WebRtcSpl_GetSizeInBits(envQ8[0]) >> 1); - envCount = 0; - - /* code assumes lenData%4 == 0 */ - for (k = 0; k < lenData; k += 4) - { - int k4; - - /* convert to magnitude spectrum, by doing square-roots (modified from SPLIB) */ - inSqrt = envQ8[envCount]; - i = 10; - - /* For safty reasons */ - if (inSqrt < 0) - inSqrt=-inSqrt; - - newRes = (inSqrt / res + res) >> 1; - do - { - res = newRes; - newRes = (inSqrt / res + res) >> 1; - } while (newRes != res && i-- > 0); - - tmpARSpecQ8 = (uint16_t)newRes; - - for(k4 = 0; k4 < 4; k4++) - { - /* find the integer *data for which streamVal lies in [W_lower+1, W_upper] */ - W_upper_LSB = (uint16_t) (W_upper & 0x0000FFFF); - W_upper_MSB = (uint16_t) WEBRTC_SPL_RSHIFT_U32(W_upper, 16); - - /* find first candidate by inverting the logistic cdf - * Input dither value collected from io-stream */ - candQ7 = - *dataQ7 + 64; - cdfTmp = WebRtcIsacfix_Piecewise(WEBRTC_SPL_MUL_16_U16(candQ7, tmpARSpecQ8)); - - W_tmp = (uint32_t)cdfTmp * W_upper_MSB; - W_tmp += ((uint32_t)cdfTmp * (uint32_t)W_upper_LSB) >> 16; - - if (streamVal > W_tmp) - { - W_lower = W_tmp; - candQ7 += 128; - cdfTmp = WebRtcIsacfix_Piecewise(WEBRTC_SPL_MUL_16_U16(candQ7, tmpARSpecQ8)); - - W_tmp = (uint32_t)cdfTmp * W_upper_MSB; - W_tmp += ((uint32_t)cdfTmp * (uint32_t)W_upper_LSB) >> 16; - - while (streamVal > W_tmp) - { - W_lower = W_tmp; - candQ7 += 128; - cdfTmp = WebRtcIsacfix_Piecewise( - WEBRTC_SPL_MUL_16_U16(candQ7, tmpARSpecQ8)); - - W_tmp = (uint32_t)cdfTmp * W_upper_MSB; - W_tmp += ((uint32_t)cdfTmp * (uint32_t)W_upper_LSB) >> 16; - - /* error check */ - if (W_lower == W_tmp) { - return -1; - } - } - W_upper = W_tmp; - - /* Output value put in dataQ7: another sample decoded */ - *dataQ7 = candQ7 - 64; - } - else - { - W_upper = W_tmp; - candQ7 -= 128; - cdfTmp = WebRtcIsacfix_Piecewise(WEBRTC_SPL_MUL_16_U16(candQ7, tmpARSpecQ8)); - - W_tmp = (uint32_t)cdfTmp * W_upper_MSB; - W_tmp += ((uint32_t)cdfTmp * (uint32_t)W_upper_LSB) >> 16; - - while ( !(streamVal > W_tmp) ) - { - W_upper = W_tmp; - candQ7 -= 128; - cdfTmp = WebRtcIsacfix_Piecewise( - WEBRTC_SPL_MUL_16_U16(candQ7, tmpARSpecQ8)); - - W_tmp = (uint32_t)cdfTmp * W_upper_MSB; - W_tmp += ((uint32_t)cdfTmp * (uint32_t)W_upper_LSB) >> 16; - - /* error check */ - if (W_upper == W_tmp){ - return -1; - } - } - W_lower = W_tmp; - - /* Output value put in dataQ7: another sample decoded */ - *dataQ7 = candQ7 + 64; - } - - dataQ7++; - - /* shift interval to start at zero */ - W_upper -= ++W_lower; - - /* add integer to bitstream */ - streamVal -= W_lower; - - /* renormalize interval and update streamVal - * W_upper < 2^24 */ - while ( !(W_upper & 0xFF000000) ) - { - if (streamPtr < streamData->stream + streamData->stream_size) { - /* read next byte from stream */ - if (streamData->full == 0) { - streamVal = (streamVal << 8) | (*streamPtr++ & 0x00FF); - streamData->full = 1; - } else { - streamVal = (streamVal << 8) | (*streamPtr >> 8); - streamData->full = 0; - } - } else { - /* Intending to read outside the stream. This can happen for the last - * two or three bytes. It is how the algorithm is implemented. Do - * not read from the bit stream and insert zeros instead. */ - streamVal <<= 8; - if (streamData->full == 0) { - offset++; // We would have incremented the pointer in this case. - streamData->full = 1; - } else { - streamData->full = 0; - } - } - W_upper <<= 8; - } - } - envCount++; - } - - streamData->stream_index = streamPtr + offset - streamData->stream; - streamData->W_upper = W_upper; - streamData->streamval = streamVal; - - /* find number of bytes in original stream (determined by current interval width) */ - if ( W_upper > 0x01FFFFFF ) - return (streamData->stream_index*2 - 3 + !streamData->full); - else - return (streamData->stream_index*2 - 2 + !streamData->full); -} diff --git a/modules/audio_coding/codecs/isac/fix/source/arith_routins.h b/modules/audio_coding/codecs/isac/fix/source/arith_routins.h deleted file mode 100644 index d112bfe7f2..0000000000 --- a/modules/audio_coding/codecs/isac/fix/source/arith_routins.h +++ /dev/null @@ -1,149 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/* - * arith_routins.h - * - * Functions for arithmetic coding. - * - */ - -#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_ARITH_ROUTINS_H_ -#define MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_ARITH_ROUTINS_H_ - -#include "modules/audio_coding/codecs/isac/fix/source/structs.h" - -/**************************************************************************** - * WebRtcIsacfix_EncLogisticMulti2(...) - * - * Arithmetic coding of spectrum. - * - * Input: - * - streamData : in-/output struct containing bitstream - * - dataQ7 : data vector in Q7 - * - envQ8 : side info vector defining the width of the pdf - * in Q8 - * - lenData : data vector length - * - * Return value : 0 if ok, - * <0 otherwise. - */ -int WebRtcIsacfix_EncLogisticMulti2(Bitstr_enc* streamData, - int16_t* dataQ7, - const uint16_t* env, - int16_t lenData); - -/**************************************************************************** - * WebRtcIsacfix_EncTerminate(...) - * - * Final call to the arithmetic coder for an encoder call. This function - * terminates and return byte stream. - * - * Input: - * - streamData : in-/output struct containing bitstream - * - * Return value : number of bytes in the stream - */ -int16_t WebRtcIsacfix_EncTerminate(Bitstr_enc* streamData); - -/**************************************************************************** - * WebRtcIsacfix_DecLogisticMulti2(...) - * - * Arithmetic decoding of spectrum. - * - * Input: - * - streamData : in-/output struct containing bitstream - * - envQ8 : side info vector defining the width of the pdf - * in Q8 - * - lenData : data vector length - * - * Input/Output: - * - dataQ7 : input: dither vector, output: data vector, in Q7 - * - * Return value : number of bytes in the stream so far - * <0 if error detected - */ -int WebRtcIsacfix_DecLogisticMulti2(int16_t* data, - Bitstr_dec* streamData, - const int32_t* env, - int16_t lenData); - -/**************************************************************************** - * WebRtcIsacfix_EncHistMulti(...) - * - * Encode the histogram interval - * - * Input: - * - streamData : in-/output struct containing bitstream - * - data : data vector - * - cdf : array of cdf arrays - * - lenData : data vector length - * - * Return value : 0 if ok - * <0 if error detected - */ -int WebRtcIsacfix_EncHistMulti(Bitstr_enc* streamData, - const int16_t* data, - const uint16_t* const* cdf, - int16_t lenData); - -/**************************************************************************** - * WebRtcIsacfix_DecHistBisectMulti(...) - * - * Function to decode more symbols from the arithmetic bytestream, using - * method of bisection. - * C df tables should be of size 2^k-1 (which corresponds to an - * alphabet size of 2^k-2) - * - * Input: - * - streamData : in-/output struct containing bitstream - * - cdf : array of cdf arrays - * - cdfSize : array of cdf table sizes+1 (power of two: 2^k) - * - lenData : data vector length - * - * Output: - * - data : data vector - * - * Return value : number of bytes in the stream - * <0 if error detected - */ -int16_t WebRtcIsacfix_DecHistBisectMulti(int16_t* data, - Bitstr_dec* streamData, - const uint16_t* const* cdf, - const uint16_t* cdfSize, - int16_t lenData); - -/**************************************************************************** - * WebRtcIsacfix_DecHistOneStepMulti(...) - * - * Function to decode more symbols from the arithmetic bytestream, taking - * single step up or down at a time. - * cdf tables can be of arbitrary size, but large tables may take a lot of - * iterations. - * - * Input: - * - streamData : in-/output struct containing bitstream - * - cdf : array of cdf arrays - * - initIndex : vector of initial cdf table search entries - * - lenData : data vector length - * - * Output: - * - data : data vector - * - * Return value : number of bytes in original stream - * <0 if error detected - */ -int16_t WebRtcIsacfix_DecHistOneStepMulti(int16_t* data, - Bitstr_dec* streamData, - const uint16_t* const* cdf, - const uint16_t* initIndex, - int16_t lenData); - -#endif /* MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_ARITH_ROUTINS_H_ */ diff --git a/modules/audio_coding/codecs/isac/fix/source/audio_decoder_isacfix.cc b/modules/audio_coding/codecs/isac/fix/source/audio_decoder_isacfix.cc deleted file mode 100644 index 21259ee2e2..0000000000 --- a/modules/audio_coding/codecs/isac/fix/source/audio_decoder_isacfix.cc +++ /dev/null @@ -1,20 +0,0 @@ -/* - * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#include "modules/audio_coding/codecs/isac/fix/include/audio_decoder_isacfix.h" - -#include "modules/audio_coding/codecs/isac/audio_decoder_isac_t_impl.h" - -namespace webrtc { - -// Explicit instantiation: -template class AudioDecoderIsacT; - -} // namespace webrtc diff --git a/modules/audio_coding/codecs/isac/fix/source/audio_encoder_isacfix.cc b/modules/audio_coding/codecs/isac/fix/source/audio_encoder_isacfix.cc deleted file mode 100644 index 0190ab91b6..0000000000 --- a/modules/audio_coding/codecs/isac/fix/source/audio_encoder_isacfix.cc +++ /dev/null @@ -1,20 +0,0 @@ -/* - * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#include "modules/audio_coding/codecs/isac/fix/include/audio_encoder_isacfix.h" - -#include "modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h" - -namespace webrtc { - -// Explicit instantiation: -template class AudioEncoderIsacT; - -} // namespace webrtc diff --git a/modules/audio_coding/codecs/isac/fix/source/bandwidth_estimator.c b/modules/audio_coding/codecs/isac/fix/source/bandwidth_estimator.c deleted file mode 100644 index 8845357d59..0000000000 --- a/modules/audio_coding/codecs/isac/fix/source/bandwidth_estimator.c +++ /dev/null @@ -1,1021 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/* - * bandwidth_estimator.c - * - * This file contains the code for the Bandwidth Estimator designed - * for iSAC. - * - * NOTE! Castings needed for C55, do not remove! - * - */ - -#include "modules/audio_coding/codecs/isac/fix/source/bandwidth_estimator.h" - -#include "modules/audio_coding/codecs/isac/fix/source/settings.h" -#include "rtc_base/checks.h" - -/* array of quantization levels for bottle neck info; Matlab code: */ -/* sprintf('%4.1ff, ', logspace(log10(5000), log10(40000), 12)) */ -static const int16_t kQRateTable[12] = { - 10000, 11115, 12355, 13733, 15265, 16967, - 18860, 20963, 23301, 25900, 28789, 32000 -}; - -/* 0.1 times the values in the table kQRateTable */ -/* values are in Q16 */ -static const int32_t KQRate01[12] = { - 65536000, 72843264, 80969728, 90000589, 100040704, 111194931, - 123600896, 137383117, 152705434, 169738240, 188671590, 209715200 -}; - -/* Bits per Bytes Seconds - * 8 bits/byte * 1000 msec/sec * 1/framelength (in msec)->bits/byte*sec - * frame length will either be 30 or 60 msec. 8738 is 1/60 in Q19 and 1/30 in Q18 - * The following number is either in Q15 or Q14 depending on the current frame length */ -static const int32_t kBitsByteSec = 4369000; - -/* Received header rate. First value is for 30 ms packets and second for 60 ms */ -static const int16_t kRecHeaderRate[2] = { - 9333, 4666 -}; - -/* Inverted minimum and maximum bandwidth in Q30. - minBwInv 30 ms, maxBwInv 30 ms, - minBwInv 60 ms, maxBwInv 69 ms -*/ -static const int32_t kInvBandwidth[4] = { - 55539, 25978, - 73213, 29284 -}; - -/* Number of samples in 25 msec */ -static const int32_t kSamplesIn25msec = 400; - - -/**************************************************************************** - * WebRtcIsacfix_InitBandwidthEstimator(...) - * - * This function initializes the struct for the bandwidth estimator - * - * Input/Output: - * - bweStr : Struct containing bandwidth information. - * - * Return value : 0 - */ -int32_t WebRtcIsacfix_InitBandwidthEstimator(BwEstimatorstr *bweStr) -{ - bweStr->prevFrameSizeMs = INIT_FRAME_LEN; - bweStr->prevRtpNumber = 0; - bweStr->prevSendTime = 0; - bweStr->prevArrivalTime = 0; - bweStr->prevRtpRate = 1; - bweStr->lastUpdate = 0; - bweStr->lastReduction = 0; - bweStr->countUpdates = -9; - - /* INIT_BN_EST = 20000 - * INIT_BN_EST_Q7 = 2560000 - * INIT_HDR_RATE = 4666 - * INIT_REC_BN_EST_Q5 = 789312 - * - * recBwInv = 1/(INIT_BN_EST + INIT_HDR_RATE) in Q30 - * recBwAvg = INIT_BN_EST + INIT_HDR_RATE in Q5 - */ - bweStr->recBwInv = 43531; - bweStr->recBw = INIT_BN_EST; - bweStr->recBwAvgQ = INIT_BN_EST_Q7; - bweStr->recBwAvg = INIT_REC_BN_EST_Q5; - bweStr->recJitter = (int32_t) 327680; /* 10 in Q15 */ - bweStr->recJitterShortTerm = 0; - bweStr->recJitterShortTermAbs = (int32_t) 40960; /* 5 in Q13 */ - bweStr->recMaxDelay = (int32_t) 10; - bweStr->recMaxDelayAvgQ = (int32_t) 5120; /* 10 in Q9 */ - bweStr->recHeaderRate = INIT_HDR_RATE; - bweStr->countRecPkts = 0; - bweStr->sendBwAvg = INIT_BN_EST_Q7; - bweStr->sendMaxDelayAvg = (int32_t) 5120; /* 10 in Q9 */ - - bweStr->countHighSpeedRec = 0; - bweStr->highSpeedRec = 0; - bweStr->countHighSpeedSent = 0; - bweStr->highSpeedSend = 0; - bweStr->inWaitPeriod = 0; - - /* Find the inverse of the max bw and min bw in Q30 - * (1 / (MAX_ISAC_BW + INIT_HDR_RATE) in Q30 - * (1 / (MIN_ISAC_BW + INIT_HDR_RATE) in Q30 - */ - bweStr->maxBwInv = kInvBandwidth[3]; - bweStr->minBwInv = kInvBandwidth[2]; - - bweStr->external_bw_info.in_use = 0; - - return 0; -} - -/**************************************************************************** - * WebRtcIsacfix_UpdateUplinkBwImpl(...) - * - * This function updates bottle neck rate received from other side in payload - * and calculates a new bottle neck to send to the other side. - * - * Input/Output: - * - bweStr : struct containing bandwidth information. - * - rtpNumber : value from RTP packet, from NetEq - * - frameSize : length of signal frame in ms, from iSAC decoder - * - sendTime : value in RTP header giving send time in samples - * - arrivalTime : value given by timeGetTime() time of arrival in - * samples of packet from NetEq - * - pksize : size of packet in bytes, from NetEq - * - Index : integer (range 0...23) indicating bottle neck & - * jitter as estimated by other side - * - * Return value : 0 if everything went fine, - * -1 otherwise - */ -int32_t WebRtcIsacfix_UpdateUplinkBwImpl(BwEstimatorstr *bweStr, - const uint16_t rtpNumber, - const int16_t frameSize, - const uint32_t sendTime, - const uint32_t arrivalTime, - const size_t pksize, - const uint16_t Index) -{ - uint16_t weight = 0; - uint32_t currBwInv = 0; - uint16_t recRtpRate; - uint32_t arrTimeProj; - int32_t arrTimeDiff; - int32_t arrTimeNoise; - int32_t arrTimeNoiseAbs; - int32_t sendTimeDiff; - - int32_t delayCorrFactor = DELAY_CORRECTION_MED; - int32_t lateDiff = 0; - int16_t immediateSet = 0; - int32_t frameSizeSampl; - - int32_t temp; - int32_t msec; - uint32_t exponent; - uint32_t reductionFactor; - uint32_t numBytesInv; - int32_t sign; - - uint32_t byteSecondsPerBit; - uint32_t tempLower; - uint32_t tempUpper; - int32_t recBwAvgInv; - int32_t numPktsExpected; - - int16_t errCode; - - RTC_DCHECK(!bweStr->external_bw_info.in_use); - - /* UPDATE ESTIMATES FROM OTHER SIDE */ - - /* The function also checks if Index has a valid value */ - errCode = WebRtcIsacfix_UpdateUplinkBwRec(bweStr, Index); - if (errCode <0) { - return(errCode); - } - - - /* UPDATE ESTIMATES ON THIS SIDE */ - - /* Bits per second per byte * 1/30 or 1/60 */ - if (frameSize == 60) { - /* If frameSize changed since last call, from 30 to 60, recalculate some values */ - if ( (frameSize != bweStr->prevFrameSizeMs) && (bweStr->countUpdates > 0)) { - bweStr->countUpdates = 10; - bweStr->recHeaderRate = kRecHeaderRate[1]; - - bweStr->maxBwInv = kInvBandwidth[3]; - bweStr->minBwInv = kInvBandwidth[2]; - bweStr->recBwInv = 1073741824 / (bweStr->recBw + bweStr->recHeaderRate); - } - - /* kBitsByteSec is in Q15 */ - recRtpRate = (int16_t)((kBitsByteSec * pksize) >> 15) + - bweStr->recHeaderRate; - - } else { - /* If frameSize changed since last call, from 60 to 30, recalculate some values */ - if ( (frameSize != bweStr->prevFrameSizeMs) && (bweStr->countUpdates > 0)) { - bweStr->countUpdates = 10; - bweStr->recHeaderRate = kRecHeaderRate[0]; - - bweStr->maxBwInv = kInvBandwidth[1]; - bweStr->minBwInv = kInvBandwidth[0]; - bweStr->recBwInv = 1073741824 / (bweStr->recBw + bweStr->recHeaderRate); - } - - /* kBitsByteSec is in Q14 */ - recRtpRate = (uint16_t)((kBitsByteSec * pksize) >> 14) + - bweStr->recHeaderRate; - } - - - /* Check for timer wrap-around */ - if (arrivalTime < bweStr->prevArrivalTime) { - bweStr->prevArrivalTime = arrivalTime; - bweStr->lastUpdate = arrivalTime; - bweStr->lastReduction = arrivalTime + FS3; - - bweStr->countRecPkts = 0; - - /* store frame size */ - bweStr->prevFrameSizeMs = frameSize; - - /* store far-side transmission rate */ - bweStr->prevRtpRate = recRtpRate; - - /* store far-side RTP time stamp */ - bweStr->prevRtpNumber = rtpNumber; - - return 0; - } - - bweStr->countRecPkts++; - - /* Calculate framesize in msec */ - frameSizeSampl = SAMPLES_PER_MSEC * frameSize; - - /* Check that it's not one of the first 9 packets */ - if ( bweStr->countUpdates > 0 ) { - - /* Stay in Wait Period for 1.5 seconds (no updates in wait period) */ - if(bweStr->inWaitPeriod) { - if ((arrivalTime - bweStr->startWaitPeriod)> FS_1_HALF) { - bweStr->inWaitPeriod = 0; - } - } - - /* If not been updated for a long time, reduce the BN estimate */ - - /* Check send time difference between this packet and previous received */ - sendTimeDiff = sendTime - bweStr->prevSendTime; - if (sendTimeDiff <= frameSizeSampl * 2) { - - /* Only update if 3 seconds has past since last update */ - if ((arrivalTime - bweStr->lastUpdate) > FS3) { - - /* Calculate expected number of received packets since last update */ - numPktsExpected = (arrivalTime - bweStr->lastUpdate) / frameSizeSampl; - - /* If received number of packets is more than 90% of expected (922 = 0.9 in Q10): */ - /* do the update, else not */ - if ((int32_t)bweStr->countRecPkts << 10 > 922 * numPktsExpected) { - /* Q4 chosen to approx dividing by 16 */ - msec = (arrivalTime - bweStr->lastReduction); - - /* the number below represents 13 seconds, highly unlikely - but to insure no overflow when reduction factor is multiplied by recBw inverse */ - if (msec > 208000) { - msec = 208000; - } - - /* Q20 2^(negative number: - 76/1048576) = .99995 - product is Q24 */ - exponent = WEBRTC_SPL_UMUL(0x0000004C, msec); - - /* do the approx with positive exponent so that value is actually rf^-1 - and multiply by bw inverse */ - reductionFactor = WEBRTC_SPL_RSHIFT_U32(0x01000000 | (exponent & 0x00FFFFFF), - WEBRTC_SPL_RSHIFT_U32(exponent, 24)); - - /* reductionFactor in Q13 */ - reductionFactor = WEBRTC_SPL_RSHIFT_U32(reductionFactor, 11); - - if ( reductionFactor != 0 ) { - bweStr->recBwInv = WEBRTC_SPL_MUL((int32_t)bweStr->recBwInv, (int32_t)reductionFactor); - bweStr->recBwInv = (int32_t)bweStr->recBwInv >> 13; - - } else { - static const uint32_t kInitRate = INIT_BN_EST + INIT_HDR_RATE; - /* recBwInv = 1 / kInitRate in Q26 (Q30??)*/ - bweStr->recBwInv = (1073741824 + kInitRate / 2) / kInitRate; - } - - /* reset time-since-update counter */ - bweStr->lastReduction = arrivalTime; - } else { - /* Delay last reduction with 3 seconds */ - bweStr->lastReduction = arrivalTime + FS3; - bweStr->lastUpdate = arrivalTime; - bweStr->countRecPkts = 0; - } - } - } else { - bweStr->lastReduction = arrivalTime + FS3; - bweStr->lastUpdate = arrivalTime; - bweStr->countRecPkts = 0; - } - - - /* update only if previous packet was not lost */ - if ( rtpNumber == bweStr->prevRtpNumber + 1 ) { - arrTimeDiff = arrivalTime - bweStr->prevArrivalTime; - - if (!(bweStr->highSpeedSend && bweStr->highSpeedRec)) { - if (arrTimeDiff > frameSizeSampl) { - if (sendTimeDiff > 0) { - lateDiff = arrTimeDiff - sendTimeDiff - frameSizeSampl * 2; - } else { - lateDiff = arrTimeDiff - frameSizeSampl; - } - - /* 8000 is 1/2 second (in samples at FS) */ - if (lateDiff > 8000) { - delayCorrFactor = (int32_t) DELAY_CORRECTION_MAX; - bweStr->inWaitPeriod = 1; - bweStr->startWaitPeriod = arrivalTime; - immediateSet = 1; - } else if (lateDiff > 5120) { - delayCorrFactor = (int32_t) DELAY_CORRECTION_MED; - immediateSet = 1; - bweStr->inWaitPeriod = 1; - bweStr->startWaitPeriod = arrivalTime; - } - } - } - - if ((bweStr->prevRtpRate > (int32_t)bweStr->recBwAvg >> 5) && - (recRtpRate > (int32_t)bweStr->recBwAvg >> 5) && - !bweStr->inWaitPeriod) { - - /* test if still in initiation period and increment counter */ - if (bweStr->countUpdates++ > 99) { - /* constant weight after initiation part, 0.01 in Q13 */ - weight = (uint16_t) 82; - } else { - /* weight decreases with number of updates, 1/countUpdates in Q13 */ - weight = (uint16_t) WebRtcSpl_DivW32W16( - 8192 + (bweStr->countUpdates >> 1), - (int16_t)bweStr->countUpdates); - } - - /* Bottle Neck Estimation */ - - /* limit outliers, if more than 25 ms too much */ - if (arrTimeDiff > frameSizeSampl + kSamplesIn25msec) { - arrTimeDiff = frameSizeSampl + kSamplesIn25msec; - } - - /* don't allow it to be less than frame rate - 10 ms */ - if (arrTimeDiff < frameSizeSampl - FRAMESAMPLES_10ms) { - arrTimeDiff = frameSizeSampl - FRAMESAMPLES_10ms; - } - - /* compute inverse receiving rate for last packet, in Q19 */ - numBytesInv = (uint16_t) WebRtcSpl_DivW32W16( - (int32_t)(524288 + ((pksize + HEADER_SIZE) >> 1)), - (int16_t)(pksize + HEADER_SIZE)); - - /* 8389 is ~ 1/128000 in Q30 */ - byteSecondsPerBit = (uint32_t)(arrTimeDiff * 8389); - - /* get upper N bits */ - tempUpper = WEBRTC_SPL_RSHIFT_U32(byteSecondsPerBit, 15); - - /* get lower 15 bits */ - tempLower = byteSecondsPerBit & 0x00007FFF; - - tempUpper = WEBRTC_SPL_MUL(tempUpper, numBytesInv); - tempLower = WEBRTC_SPL_MUL(tempLower, numBytesInv); - tempLower = WEBRTC_SPL_RSHIFT_U32(tempLower, 15); - - currBwInv = tempUpper + tempLower; - currBwInv = WEBRTC_SPL_RSHIFT_U32(currBwInv, 4); - - /* Limit inv rate. Note that minBwInv > maxBwInv! */ - if(currBwInv < bweStr->maxBwInv) { - currBwInv = bweStr->maxBwInv; - } else if(currBwInv > bweStr->minBwInv) { - currBwInv = bweStr->minBwInv; - } - - /* update bottle neck rate estimate */ - bweStr->recBwInv = WEBRTC_SPL_UMUL(weight, currBwInv) + - WEBRTC_SPL_UMUL((uint32_t) 8192 - weight, bweStr->recBwInv); - - /* Shift back to Q30 from Q40 (actual used bits shouldn't be more than 27 based on minBwInv) - up to 30 bits used with Q13 weight */ - bweStr->recBwInv = WEBRTC_SPL_RSHIFT_U32(bweStr->recBwInv, 13); - - /* reset time-since-update counter */ - bweStr->lastUpdate = arrivalTime; - bweStr->lastReduction = arrivalTime + FS3; - bweStr->countRecPkts = 0; - - /* to save resolution compute the inverse of recBwAvg in Q26 by left shifting numerator to 2^31 - and NOT right shifting recBwAvg 5 bits to an integer - At max 13 bits are used - shift to Q5 */ - recBwAvgInv = (0x80000000 + bweStr->recBwAvg / 2) / bweStr->recBwAvg; - - /* Calculate Projected arrival time difference */ - - /* The numerator of the quotient can be 22 bits so right shift inv by 4 to avoid overflow - result in Q22 */ - arrTimeProj = WEBRTC_SPL_MUL((int32_t)8000, recBwAvgInv); - /* shift to Q22 */ - arrTimeProj = WEBRTC_SPL_RSHIFT_U32(arrTimeProj, 4); - /* complete calulation */ - arrTimeProj = WEBRTC_SPL_MUL(((int32_t)pksize + HEADER_SIZE), arrTimeProj); - /* shift to Q10 */ - arrTimeProj = WEBRTC_SPL_RSHIFT_U32(arrTimeProj, 12); - - /* difference between projected and actual arrival time differences */ - /* Q9 (only shift arrTimeDiff by 5 to simulate divide by 16 (need to revisit if change sampling rate) DH */ - if ((arrTimeDiff << 6) > (int32_t)arrTimeProj) { - arrTimeNoise = (arrTimeDiff << 6) - arrTimeProj; - sign = 1; - } else { - arrTimeNoise = arrTimeProj - (arrTimeDiff << 6); - sign = -1; - } - - /* Q9 */ - arrTimeNoiseAbs = arrTimeNoise; - - /* long term averaged absolute jitter, Q15 */ - weight >>= 3; - bweStr->recJitter = weight * (arrTimeNoiseAbs << 5) + - (1024 - weight) * bweStr->recJitter; - - /* remove the fractional portion */ - bweStr->recJitter >>= 10; - - /* Maximum jitter is 10 msec in Q15 */ - if (bweStr->recJitter > (int32_t)327680) { - bweStr->recJitter = (int32_t)327680; - } - - /* short term averaged absolute jitter */ - /* Calculation in Q13 products in Q23 */ - bweStr->recJitterShortTermAbs = 51 * (arrTimeNoiseAbs << 3) + - WEBRTC_SPL_MUL(973, bweStr->recJitterShortTermAbs); - bweStr->recJitterShortTermAbs >>= 10; - - /* short term averaged jitter */ - /* Calculation in Q13 products in Q23 */ - bweStr->recJitterShortTerm = 205 * (arrTimeNoise << 3) * sign + - WEBRTC_SPL_MUL(3891, bweStr->recJitterShortTerm); - - if (bweStr->recJitterShortTerm < 0) { - temp = -bweStr->recJitterShortTerm; - temp >>= 12; - bweStr->recJitterShortTerm = -temp; - } else { - bweStr->recJitterShortTerm >>= 12; - } - } - } - } else { - /* reset time-since-update counter when receiving the first 9 packets */ - bweStr->lastUpdate = arrivalTime; - bweStr->lastReduction = arrivalTime + FS3; - bweStr->countRecPkts = 0; - bweStr->countUpdates++; - } - - /* Limit to minimum or maximum bottle neck rate (in Q30) */ - if (bweStr->recBwInv > bweStr->minBwInv) { - bweStr->recBwInv = bweStr->minBwInv; - } else if (bweStr->recBwInv < bweStr->maxBwInv) { - bweStr->recBwInv = bweStr->maxBwInv; - } - - - /* store frame length */ - bweStr->prevFrameSizeMs = frameSize; - - /* store far-side transmission rate */ - bweStr->prevRtpRate = recRtpRate; - - /* store far-side RTP time stamp */ - bweStr->prevRtpNumber = rtpNumber; - - /* Replace bweStr->recMaxDelay by the new value (atomic operation) */ - if (bweStr->prevArrivalTime != 0xffffffff) { - bweStr->recMaxDelay = WEBRTC_SPL_MUL(3, bweStr->recJitter); - } - - /* store arrival time stamp */ - bweStr->prevArrivalTime = arrivalTime; - bweStr->prevSendTime = sendTime; - - /* Replace bweStr->recBw by the new value */ - bweStr->recBw = 1073741824 / bweStr->recBwInv - bweStr->recHeaderRate; - - if (immediateSet) { - /* delay correction factor is in Q10 */ - bweStr->recBw = WEBRTC_SPL_UMUL(delayCorrFactor, bweStr->recBw); - bweStr->recBw = WEBRTC_SPL_RSHIFT_U32(bweStr->recBw, 10); - - if (bweStr->recBw < (int32_t) MIN_ISAC_BW) { - bweStr->recBw = (int32_t) MIN_ISAC_BW; - } - - bweStr->recBwAvg = (bweStr->recBw + bweStr->recHeaderRate) << 5; - - bweStr->recBwAvgQ = bweStr->recBw << 7; - - bweStr->recJitterShortTerm = 0; - - bweStr->recBwInv = 1073741824 / (bweStr->recBw + bweStr->recHeaderRate); - - immediateSet = 0; - } - - - return 0; -} - -/* This function updates the send bottle neck rate */ -/* Index - integer (range 0...23) indicating bottle neck & jitter as estimated by other side */ -/* returns 0 if everything went fine, -1 otherwise */ -int16_t WebRtcIsacfix_UpdateUplinkBwRec(BwEstimatorstr *bweStr, - const int16_t Index) -{ - uint16_t RateInd; - - RTC_DCHECK(!bweStr->external_bw_info.in_use); - - if ( (Index < 0) || (Index > 23) ) { - return -ISAC_RANGE_ERROR_BW_ESTIMATOR; - } - - /* UPDATE ESTIMATES FROM OTHER SIDE */ - - if ( Index > 11 ) { - RateInd = Index - 12; - /* compute the jitter estimate as decoded on the other side in Q9 */ - /* sendMaxDelayAvg = 0.9 * sendMaxDelayAvg + 0.1 * MAX_ISAC_MD */ - bweStr->sendMaxDelayAvg = WEBRTC_SPL_MUL(461, bweStr->sendMaxDelayAvg) + - 51 * (MAX_ISAC_MD << 9); - bweStr->sendMaxDelayAvg >>= 9; - - } else { - RateInd = Index; - /* compute the jitter estimate as decoded on the other side in Q9 */ - /* sendMaxDelayAvg = 0.9 * sendMaxDelayAvg + 0.1 * MIN_ISAC_MD */ - bweStr->sendMaxDelayAvg = WEBRTC_SPL_MUL(461, bweStr->sendMaxDelayAvg) + - 51 * (MIN_ISAC_MD << 9); - bweStr->sendMaxDelayAvg >>= 9; - - } - - - /* compute the BN estimate as decoded on the other side */ - /* sendBwAvg = 0.9 * sendBwAvg + 0.1 * kQRateTable[RateInd]; */ - bweStr->sendBwAvg = 461 * bweStr->sendBwAvg + - 51 * ((uint32_t)kQRateTable[RateInd] << 7); - bweStr->sendBwAvg = WEBRTC_SPL_RSHIFT_U32(bweStr->sendBwAvg, 9); - - - if (WEBRTC_SPL_RSHIFT_U32(bweStr->sendBwAvg, 7) > 28000 && !bweStr->highSpeedSend) { - bweStr->countHighSpeedSent++; - - /* approx 2 seconds with 30ms frames */ - if (bweStr->countHighSpeedSent >= 66) { - bweStr->highSpeedSend = 1; - } - } else if (!bweStr->highSpeedSend) { - bweStr->countHighSpeedSent = 0; - } - - return 0; -} - -/**************************************************************************** - * WebRtcIsacfix_GetDownlinkBwIndexImpl(...) - * - * This function calculates and returns the bandwidth/jitter estimation code - * (integer 0...23) to put in the sending iSAC payload. - * - * Input: - * - bweStr : BWE struct - * - * Return: - * bandwith and jitter index (0..23) - */ -uint16_t WebRtcIsacfix_GetDownlinkBwIndexImpl(BwEstimatorstr *bweStr) -{ - int32_t rate; - int32_t maxDelay; - uint16_t rateInd; - uint16_t maxDelayBit; - int32_t tempTerm1; - int32_t tempTerm2; - int32_t tempTermX; - int32_t tempTermY; - int32_t tempMin; - int32_t tempMax; - - if (bweStr->external_bw_info.in_use) - return bweStr->external_bw_info.bottleneck_idx; - - /* Get Rate Index */ - - /* Get unquantized rate. Always returns 10000 <= rate <= 32000 */ - rate = WebRtcIsacfix_GetDownlinkBandwidth(bweStr); - - /* Compute the averaged BN estimate on this side */ - - /* recBwAvg = 0.9 * recBwAvg + 0.1 * (rate + bweStr->recHeaderRate), 0.9 and 0.1 in Q9 */ - bweStr->recBwAvg = 922 * bweStr->recBwAvg + - 102 * (((uint32_t)rate + bweStr->recHeaderRate) << 5); - bweStr->recBwAvg = WEBRTC_SPL_RSHIFT_U32(bweStr->recBwAvg, 10); - - /* Find quantization index that gives the closest rate after averaging. - * Note that we don't need to check the last value, rate <= kQRateTable[11], - * because we will use rateInd = 11 even if rate > kQRateTable[11]. */ - for (rateInd = 1; rateInd < 11; rateInd++) { - if (rate <= kQRateTable[rateInd]){ - break; - } - } - - /* find closest quantization index, and update quantized average by taking: */ - /* 0.9*recBwAvgQ + 0.1*kQRateTable[rateInd] */ - - /* 0.9 times recBwAvgQ in Q16 */ - /* 461/512 - 25/65536 =0.900009 */ - tempTerm1 = WEBRTC_SPL_MUL(bweStr->recBwAvgQ, 25); - tempTerm1 >>= 7; - tempTermX = WEBRTC_SPL_UMUL(461, bweStr->recBwAvgQ) - tempTerm1; - - /* rate in Q16 */ - tempTermY = rate << 16; - - /* 0.1 * kQRateTable[rateInd] = KQRate01[rateInd] */ - tempTerm1 = tempTermX + KQRate01[rateInd] - tempTermY; - tempTerm2 = tempTermY - tempTermX - KQRate01[rateInd-1]; - - /* Compare (0.9 * recBwAvgQ + 0.1 * kQRateTable[rateInd] - rate) > - (rate - 0.9 * recBwAvgQ - 0.1 * kQRateTable[rateInd-1]) */ - if (tempTerm1 > tempTerm2) { - rateInd--; - } - - /* Update quantized average by taking: */ - /* 0.9*recBwAvgQ + 0.1*kQRateTable[rateInd] */ - - /* Add 0.1 times kQRateTable[rateInd], in Q16 */ - tempTermX += KQRate01[rateInd]; - - /* Shift back to Q7 */ - bweStr->recBwAvgQ = tempTermX >> 9; - - /* Count consecutive received bandwidth above 28000 kbps (28000 in Q7 = 3584000) */ - /* If 66 high estimates in a row, set highSpeedRec to one */ - /* 66 corresponds to ~2 seconds in 30 msec mode */ - if ((bweStr->recBwAvgQ > 3584000) && !bweStr->highSpeedRec) { - bweStr->countHighSpeedRec++; - if (bweStr->countHighSpeedRec >= 66) { - bweStr->highSpeedRec = 1; - } - } else if (!bweStr->highSpeedRec) { - bweStr->countHighSpeedRec = 0; - } - - /* Get Max Delay Bit */ - - /* get unquantized max delay */ - maxDelay = WebRtcIsacfix_GetDownlinkMaxDelay(bweStr); - - /* Update quantized max delay average */ - tempMax = 652800; /* MAX_ISAC_MD * 0.1 in Q18 */ - tempMin = 130560; /* MIN_ISAC_MD * 0.1 in Q18 */ - tempTermX = WEBRTC_SPL_MUL((int32_t)bweStr->recMaxDelayAvgQ, (int32_t)461); - tempTermY = maxDelay << 18; - - tempTerm1 = tempTermX + tempMax - tempTermY; - tempTerm2 = tempTermY - tempTermX - tempMin; - - if ( tempTerm1 > tempTerm2) { - maxDelayBit = 0; - tempTerm1 = tempTermX + tempMin; - - /* update quantized average, shift back to Q9 */ - bweStr->recMaxDelayAvgQ = tempTerm1 >> 9; - } else { - maxDelayBit = 12; - tempTerm1 = tempTermX + tempMax; - - /* update quantized average, shift back to Q9 */ - bweStr->recMaxDelayAvgQ = tempTerm1 >> 9; - } - - /* Return bandwitdh and jitter index (0..23) */ - return (uint16_t)(rateInd + maxDelayBit); -} - -/* get the bottle neck rate from far side to here, as estimated on this side */ -uint16_t WebRtcIsacfix_GetDownlinkBandwidth(const BwEstimatorstr *bweStr) -{ - uint32_t recBw; - int32_t jitter_sign; /* Q8 */ - int32_t bw_adjust; /* Q16 */ - int32_t rec_jitter_short_term_abs_inv; /* Q18 */ - int32_t temp; - - RTC_DCHECK(!bweStr->external_bw_info.in_use); - - /* Q18 rec jitter short term abs is in Q13, multiply it by 2^13 to save precision - 2^18 then needs to be shifted 13 bits to 2^31 */ - rec_jitter_short_term_abs_inv = 0x80000000u / bweStr->recJitterShortTermAbs; - - /* Q27 = 9 + 18 */ - jitter_sign = (bweStr->recJitterShortTerm >> 4) * - rec_jitter_short_term_abs_inv; - - if (jitter_sign < 0) { - temp = -jitter_sign; - temp >>= 19; - jitter_sign = -temp; - } else { - jitter_sign >>= 19; - } - - /* adjust bw proportionally to negative average jitter sign */ - //bw_adjust = 1.0f - jitter_sign * (0.15f + 0.15f * jitter_sign * jitter_sign); - //Q8 -> Q16 .15 +.15 * jitter^2 first term is .15 in Q16 latter term is Q8*Q8*Q8 - //38 in Q8 ~.15 9830 in Q16 ~.15 - temp = 9830 + ((38 * jitter_sign * jitter_sign) >> 8); - - if (jitter_sign < 0) { - temp = WEBRTC_SPL_MUL(jitter_sign, temp); - temp = -temp; - temp >>= 8; - bw_adjust = (uint32_t)65536 + temp; /* (1 << 16) + temp; */ - } else { - /* (1 << 16) - ((jitter_sign * temp) >> 8); */ - bw_adjust = 65536 - ((jitter_sign * temp) >> 8); - } - - //make sure following multiplication won't overflow - //bw adjust now Q14 - bw_adjust >>= 2; // See if good resolution is maintained. - - /* adjust Rate if jitter sign is mostly constant */ - recBw = WEBRTC_SPL_UMUL(bweStr->recBw, bw_adjust); - - recBw >>= 14; - - /* limit range of bottle neck rate */ - if (recBw < MIN_ISAC_BW) { - recBw = MIN_ISAC_BW; - } else if (recBw > MAX_ISAC_BW) { - recBw = MAX_ISAC_BW; - } - - return (uint16_t) recBw; -} - -/* Returns the mmax delay (in ms) */ -int16_t WebRtcIsacfix_GetDownlinkMaxDelay(const BwEstimatorstr *bweStr) -{ - int16_t recMaxDelay = (int16_t)(bweStr->recMaxDelay >> 15); - - RTC_DCHECK(!bweStr->external_bw_info.in_use); - - /* limit range of jitter estimate */ - if (recMaxDelay < MIN_ISAC_MD) { - recMaxDelay = MIN_ISAC_MD; - } else if (recMaxDelay > MAX_ISAC_MD) { - recMaxDelay = MAX_ISAC_MD; - } - - return recMaxDelay; -} - -/* Clamp val to the closed interval [min,max]. */ -static int16_t clamp(int16_t val, int16_t min, int16_t max) { - RTC_DCHECK_LE(min, max); - return val < min ? min : (val > max ? max : val); -} - -int16_t WebRtcIsacfix_GetUplinkBandwidth(const BwEstimatorstr* bweStr) { - return bweStr->external_bw_info.in_use - ? bweStr->external_bw_info.send_bw_avg - : clamp(bweStr->sendBwAvg >> 7, MIN_ISAC_BW, MAX_ISAC_BW); -} - -int16_t WebRtcIsacfix_GetUplinkMaxDelay(const BwEstimatorstr* bweStr) { - return bweStr->external_bw_info.in_use - ? bweStr->external_bw_info.send_max_delay_avg - : clamp(bweStr->sendMaxDelayAvg >> 9, MIN_ISAC_MD, MAX_ISAC_MD); -} - -/* - * update long-term average bitrate and amount of data in buffer - * returns minimum payload size (bytes) - */ -uint16_t WebRtcIsacfix_GetMinBytes(RateModel *State, - int16_t StreamSize, /* bytes in bitstream */ - const int16_t FrameSamples, /* samples per frame */ - const int16_t BottleNeck, /* bottle neck rate; excl headers (bps) */ - const int16_t DelayBuildUp) /* max delay from bottle neck buffering (ms) */ -{ - int32_t MinRate = 0; - uint16_t MinBytes; - int16_t TransmissionTime; - int32_t inv_Q12; - int32_t den; - - - /* first 10 packets @ low rate, then INIT_BURST_LEN packets @ fixed rate of INIT_RATE bps */ - if (State->InitCounter > 0) { - if (State->InitCounter-- <= INIT_BURST_LEN) { - MinRate = INIT_RATE; - } else { - MinRate = 0; - } - } else { - /* handle burst */ - if (State->BurstCounter) { - if (State->StillBuffered < - (((512 - 512 / BURST_LEN) * DelayBuildUp) >> 9)) { - /* max bps derived from BottleNeck and DelayBuildUp values */ - inv_Q12 = 4096 / (BURST_LEN * FrameSamples); - MinRate = (512 + SAMPLES_PER_MSEC * ((DelayBuildUp * inv_Q12) >> 3)) * - BottleNeck; - } else { - /* max bps derived from StillBuffered and DelayBuildUp values */ - inv_Q12 = 4096 / FrameSamples; - if (DelayBuildUp > State->StillBuffered) { - MinRate = (512 + SAMPLES_PER_MSEC * (((DelayBuildUp - - State->StillBuffered) * inv_Q12) >> 3)) * BottleNeck; - } else if ((den = WEBRTC_SPL_MUL(SAMPLES_PER_MSEC, (State->StillBuffered - DelayBuildUp))) >= FrameSamples) { - /* MinRate will be negative here */ - MinRate = 0; - } else { - MinRate = (512 - ((den * inv_Q12) >> 3)) * BottleNeck; - } - //if (MinRate < 1.04 * BottleNeck) - // MinRate = 1.04 * BottleNeck; - //Q9 - if (MinRate < WEBRTC_SPL_MUL(532, BottleNeck)) { - MinRate += WEBRTC_SPL_MUL(22, BottleNeck); - } - } - - State->BurstCounter--; - } - } - - - /* convert rate from bits/second to bytes/packet */ - //round and shift before conversion - MinRate += 256; - MinRate >>= 9; - MinBytes = MinRate * FrameSamples / FS8; - - /* StreamSize will be adjusted if less than MinBytes */ - if (StreamSize < MinBytes) { - StreamSize = MinBytes; - } - - /* keep track of when bottle neck was last exceeded by at least 1% */ - //517/512 ~ 1.01 - if ((StreamSize * (int32_t)FS8) / FrameSamples > (517 * BottleNeck) >> 9) { - if (State->PrevExceed) { - /* bottle_neck exceded twice in a row, decrease ExceedAgo */ - State->ExceedAgo -= BURST_INTERVAL / (BURST_LEN - 1); - if (State->ExceedAgo < 0) { - State->ExceedAgo = 0; - } - } else { - State->ExceedAgo += FrameSamples / SAMPLES_PER_MSEC; /* ms */ - State->PrevExceed = 1; - } - } else { - State->PrevExceed = 0; - State->ExceedAgo += FrameSamples / SAMPLES_PER_MSEC; /* ms */ - } - - /* set burst flag if bottle neck not exceeded for long time */ - if ((State->ExceedAgo > BURST_INTERVAL) && (State->BurstCounter == 0)) { - if (State->PrevExceed) { - State->BurstCounter = BURST_LEN - 1; - } else { - State->BurstCounter = BURST_LEN; - } - } - - - /* Update buffer delay */ - TransmissionTime = (StreamSize * 8000) / BottleNeck; /* ms */ - State->StillBuffered += TransmissionTime; - State->StillBuffered -= FrameSamples / SAMPLES_PER_MSEC; /* ms */ - if (State->StillBuffered < 0) { - State->StillBuffered = 0; - } - - if (State->StillBuffered > 2000) { - State->StillBuffered = 2000; - } - - return MinBytes; -} - - -/* - * update long-term average bitrate and amount of data in buffer - */ -void WebRtcIsacfix_UpdateRateModel(RateModel *State, - int16_t StreamSize, /* bytes in bitstream */ - const int16_t FrameSamples, /* samples per frame */ - const int16_t BottleNeck) /* bottle neck rate; excl headers (bps) */ -{ - const int16_t TransmissionTime = (StreamSize * 8000) / BottleNeck; /* ms */ - - /* avoid the initial "high-rate" burst */ - State->InitCounter = 0; - - /* Update buffer delay */ - State->StillBuffered += TransmissionTime; - State->StillBuffered -= FrameSamples >> 4; /* ms */ - if (State->StillBuffered < 0) { - State->StillBuffered = 0; - } - -} - - -void WebRtcIsacfix_InitRateModel(RateModel *State) -{ - State->PrevExceed = 0; /* boolean */ - State->ExceedAgo = 0; /* ms */ - State->BurstCounter = 0; /* packets */ - State->InitCounter = INIT_BURST_LEN + 10; /* packets */ - State->StillBuffered = 1; /* ms */ -} - - - - - -int16_t WebRtcIsacfix_GetNewFrameLength(int16_t bottle_neck, int16_t current_framesamples) -{ - int16_t new_framesamples; - - new_framesamples = current_framesamples; - - /* find new framelength */ - switch(current_framesamples) { - case 480: - if (bottle_neck < Thld_30_60) { - new_framesamples = 960; - } - break; - case 960: - if (bottle_neck >= Thld_60_30) { - new_framesamples = 480; - } - break; - default: - new_framesamples = -1; /* Error */ - } - - return new_framesamples; -} - -int16_t WebRtcIsacfix_GetSnr(int16_t bottle_neck, int16_t framesamples) -{ - int16_t s2nr = 0; - - /* find new SNR value */ - //consider BottleNeck to be in Q10 ( * 1 in Q10) - switch(framesamples) { - // TODO(bjornv): The comments below confuses me. I don't know if there is a - // difference between frame lengths (in which case the implementation is - // wrong), or if it is frame length independent in which case we should - // correct the comment and simplify the implementation. - case 480: - /*s2nr = -1*(a_30 << 10) + ((b_30 * bottle_neck) >> 10);*/ - s2nr = -22500 + (int16_t)(500 * bottle_neck >> 10); - break; - case 960: - /*s2nr = -1*(a_60 << 10) + ((b_60 * bottle_neck) >> 10);*/ - s2nr = -22500 + (int16_t)(500 * bottle_neck >> 10); - break; - default: - s2nr = -1; /* Error */ - } - - return s2nr; //return in Q10 - -} diff --git a/modules/audio_coding/codecs/isac/fix/source/bandwidth_estimator.h b/modules/audio_coding/codecs/isac/fix/source/bandwidth_estimator.h deleted file mode 100644 index f106746f14..0000000000 --- a/modules/audio_coding/codecs/isac/fix/source/bandwidth_estimator.h +++ /dev/null @@ -1,128 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/* - * bandwidth_estimator.h - * - * This header file contains the API for the Bandwidth Estimator - * designed for iSAC. - * - */ - -#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_BANDWIDTH_ESTIMATOR_H_ -#define MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_BANDWIDTH_ESTIMATOR_H_ - -#include "modules/audio_coding/codecs/isac/fix/source/structs.h" - -/**************************************************************************** - * WebRtcIsacfix_InitBandwidthEstimator(...) - * - * This function initializes the struct for the bandwidth estimator - * - * Input/Output: - * - bwest_str : Struct containing bandwidth information. - * - * Return value : 0 - */ - -int32_t WebRtcIsacfix_InitBandwidthEstimator(BwEstimatorstr* bwest_str); - -/**************************************************************************** - * WebRtcIsacfix_UpdateUplinkBwImpl(...) - * - * This function updates bottle neck rate received from other side in payload - * and calculates a new bottle neck to send to the other side. - * - * Input/Output: - * - bweStr : struct containing bandwidth information. - * - rtpNumber : value from RTP packet, from NetEq - * - frameSize : length of signal frame in ms, from iSAC decoder - * - sendTime : value in RTP header giving send time in samples - * - arrivalTime : value given by timeGetTime() time of arrival in - * samples of packet from NetEq - * - pksize : size of packet in bytes, from NetEq - * - Index : integer (range 0...23) indicating bottle neck & - * jitter as estimated by other side - * - * Return value : 0 if everything went fine, - * -1 otherwise - */ - -int32_t WebRtcIsacfix_UpdateUplinkBwImpl(BwEstimatorstr* bwest_str, - uint16_t rtp_number, - int16_t frameSize, - uint32_t send_ts, - uint32_t arr_ts, - size_t pksize, - uint16_t Index); - -/* Update receiving estimates. Used when we only receive BWE index, no iSAC data - * packet. */ -int16_t WebRtcIsacfix_UpdateUplinkBwRec(BwEstimatorstr* bwest_str, - int16_t Index); - -/**************************************************************************** - * WebRtcIsacfix_GetDownlinkBwIndexImpl(...) - * - * This function calculates and returns the bandwidth/jitter estimation code - * (integer 0...23) to put in the sending iSAC payload. - * - * Input: - * - bweStr : BWE struct - * - * Return: - * bandwith and jitter index (0..23) - */ -uint16_t WebRtcIsacfix_GetDownlinkBwIndexImpl(BwEstimatorstr* bwest_str); - -/* Returns the bandwidth estimation (in bps) */ -uint16_t WebRtcIsacfix_GetDownlinkBandwidth(const BwEstimatorstr* bwest_str); - -/* Returns the bandwidth that iSAC should send with in bps */ -int16_t WebRtcIsacfix_GetUplinkBandwidth(const BwEstimatorstr* bwest_str); - -/* Returns the max delay (in ms) */ -int16_t WebRtcIsacfix_GetDownlinkMaxDelay(const BwEstimatorstr* bwest_str); - -/* Returns the max delay value from the other side in ms */ -int16_t WebRtcIsacfix_GetUplinkMaxDelay(const BwEstimatorstr* bwest_str); - -/* - * update amount of data in bottle neck buffer and burst handling - * returns minimum payload size (bytes) - */ -uint16_t WebRtcIsacfix_GetMinBytes( - RateModel* State, - int16_t StreamSize, /* bytes in bitstream */ - int16_t FrameLen, /* ms per frame */ - int16_t BottleNeck, /* bottle neck rate; excl headers (bps) */ - int16_t DelayBuildUp); /* max delay from bottle neck buffering (ms) */ - -/* - * update long-term average bitrate and amount of data in buffer - */ -void WebRtcIsacfix_UpdateRateModel( - RateModel* State, - int16_t StreamSize, /* bytes in bitstream */ - int16_t FrameSamples, /* samples per frame */ - int16_t BottleNeck); /* bottle neck rate; excl headers (bps) */ - -void WebRtcIsacfix_InitRateModel(RateModel* State); - -/* Returns the new framelength value (input argument: bottle_neck) */ -int16_t WebRtcIsacfix_GetNewFrameLength(int16_t bottle_neck, - int16_t current_framelength); - -/* Returns the new SNR value (input argument: bottle_neck) */ -// returns snr in Q10 -int16_t WebRtcIsacfix_GetSnr(int16_t bottle_neck, int16_t framesamples); - -#endif /* MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_BANDWIDTH_ESTIMATOR_H_ \ - */ diff --git a/modules/audio_coding/codecs/isac/fix/source/codec.h b/modules/audio_coding/codecs/isac/fix/source/codec.h deleted file mode 100644 index 01d6fb907e..0000000000 --- a/modules/audio_coding/codecs/isac/fix/source/codec.h +++ /dev/null @@ -1,212 +0,0 @@ -/* - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/* - * codec.h - * - * This header file contains the calls to the internal encoder - * and decoder functions. - * - */ - -#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_CODEC_H_ -#define MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_CODEC_H_ - -#include "modules/audio_coding/codecs/isac/fix/source/structs.h" - -#ifdef __cplusplus -extern "C" { -#endif - -int WebRtcIsacfix_EstimateBandwidth(BwEstimatorstr* bwest_str, - Bitstr_dec* streamdata, - size_t packet_size, - uint16_t rtp_seq_number, - uint32_t send_ts, - uint32_t arr_ts); - -int WebRtcIsacfix_DecodeImpl(int16_t* signal_out16, - IsacFixDecoderInstance* ISACdec_obj, - size_t* current_framesamples); - -void WebRtcIsacfix_DecodePlcImpl(int16_t* decoded, - IsacFixDecoderInstance* ISACdec_obj, - size_t* current_framesample); - -int WebRtcIsacfix_EncodeImpl(int16_t* in, - IsacFixEncoderInstance* ISACenc_obj, - BwEstimatorstr* bw_estimatordata, - int16_t CodingMode); - -int WebRtcIsacfix_EncodeStoredData(IsacFixEncoderInstance* ISACenc_obj, - int BWnumber, - float scale); - -/* initialization functions */ - -void WebRtcIsacfix_InitMaskingEnc(MaskFiltstr_enc* maskdata); -void WebRtcIsacfix_InitMaskingDec(MaskFiltstr_dec* maskdata); - -void WebRtcIsacfix_InitPreFilterbank(PreFiltBankstr* prefiltdata); - -void WebRtcIsacfix_InitPostFilterbank(PostFiltBankstr* postfiltdata); - -void WebRtcIsacfix_InitPitchFilter(PitchFiltstr* pitchfiltdata); - -void WebRtcIsacfix_InitPitchAnalysis(PitchAnalysisStruct* State); - -void WebRtcIsacfix_InitPlc(PLCstr* State); - -/* transform functions */ - -void WebRtcIsacfix_InitTransform(void); - -typedef void (*Time2Spec)(int16_t* inre1Q9, - int16_t* inre2Q9, - int16_t* outre, - int16_t* outim); -typedef void (*Spec2Time)(int16_t* inreQ7, - int16_t* inimQ7, - int32_t* outre1Q16, - int32_t* outre2Q16); - -extern Time2Spec WebRtcIsacfix_Time2Spec; -extern Spec2Time WebRtcIsacfix_Spec2Time; - -void WebRtcIsacfix_Time2SpecC(int16_t* inre1Q9, - int16_t* inre2Q9, - int16_t* outre, - int16_t* outim); -void WebRtcIsacfix_Spec2TimeC(int16_t* inreQ7, - int16_t* inimQ7, - int32_t* outre1Q16, - int32_t* outre2Q16); - -#if defined(WEBRTC_HAS_NEON) -void WebRtcIsacfix_Time2SpecNeon(int16_t* inre1Q9, - int16_t* inre2Q9, - int16_t* outre, - int16_t* outim); -void WebRtcIsacfix_Spec2TimeNeon(int16_t* inreQ7, - int16_t* inimQ7, - int32_t* outre1Q16, - int32_t* outre2Q16); -#endif - -#if defined(MIPS32_LE) -void WebRtcIsacfix_Time2SpecMIPS(int16_t* inre1Q9, - int16_t* inre2Q9, - int16_t* outre, - int16_t* outim); -void WebRtcIsacfix_Spec2TimeMIPS(int16_t* inreQ7, - int16_t* inimQ7, - int32_t* outre1Q16, - int32_t* outre2Q16); -#endif - -/* filterbank functions */ - -void WebRtcIsacfix_SplitAndFilter1(int16_t* in, - int16_t* LP16, - int16_t* HP16, - PreFiltBankstr* prefiltdata); - -void WebRtcIsacfix_FilterAndCombine1(int16_t* tempin_ch1, - int16_t* tempin_ch2, - int16_t* out16, - PostFiltBankstr* postfiltdata); - -/* normalized lattice filters */ - -void WebRtcIsacfix_NormLatticeFilterMa(size_t orderCoef, - int32_t* stateGQ15, - int16_t* lat_inQ0, - int16_t* filt_coefQ15, - int32_t* gain_lo_hiQ17, - int16_t lo_hi, - int16_t* lat_outQ9); - -void WebRtcIsacfix_NormLatticeFilterAr(size_t orderCoef, - int16_t* stateGQ0, - int32_t* lat_inQ25, - int16_t* filt_coefQ15, - int32_t* gain_lo_hiQ17, - int16_t lo_hi, - int16_t* lat_outQ0); - -/* TODO(kma): Remove the following functions into individual header files. */ - -/* Internal functions in both C and ARM Neon versions */ - -int WebRtcIsacfix_AutocorrC(int32_t* __restrict r, - const int16_t* __restrict x, - int16_t N, - int16_t order, - int16_t* __restrict scale); - -void WebRtcIsacfix_FilterMaLoopC(int16_t input0, - int16_t input1, - int32_t input2, - int32_t* ptr0, - int32_t* ptr1, - int32_t* ptr2); - -#if defined(WEBRTC_HAS_NEON) -int WebRtcIsacfix_AutocorrNeon(int32_t* __restrict r, - const int16_t* __restrict x, - int16_t N, - int16_t order, - int16_t* __restrict scale); - -void WebRtcIsacfix_FilterMaLoopNeon(int16_t input0, - int16_t input1, - int32_t input2, - int32_t* ptr0, - int32_t* ptr1, - int32_t* ptr2); -#endif - -#if defined(MIPS32_LE) -int WebRtcIsacfix_AutocorrMIPS(int32_t* __restrict r, - const int16_t* __restrict x, - int16_t N, - int16_t order, - int16_t* __restrict scale); - -void WebRtcIsacfix_FilterMaLoopMIPS(int16_t input0, - int16_t input1, - int32_t input2, - int32_t* ptr0, - int32_t* ptr1, - int32_t* ptr2); -#endif - -/* Function pointers associated with the above functions. */ - -typedef int (*AutocorrFix)(int32_t* __restrict r, - const int16_t* __restrict x, - int16_t N, - int16_t order, - int16_t* __restrict scale); -extern AutocorrFix WebRtcIsacfix_AutocorrFix; - -typedef void (*FilterMaLoopFix)(int16_t input0, - int16_t input1, - int32_t input2, - int32_t* ptr0, - int32_t* ptr1, - int32_t* ptr2); -extern FilterMaLoopFix WebRtcIsacfix_FilterMaLoopFix; - -#ifdef __cplusplus -} // extern "C" -#endif - -#endif /* MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_CODEC_H_ */ diff --git a/modules/audio_coding/codecs/isac/fix/source/decode.c b/modules/audio_coding/codecs/isac/fix/source/decode.c deleted file mode 100644 index 144208818a..0000000000 --- a/modules/audio_coding/codecs/isac/fix/source/decode.c +++ /dev/null @@ -1,221 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/* - * decode.c - * - * This C file contains the internal decoding function. - * - */ - -#include - -#include "modules/audio_coding/codecs/isac/fix/source/bandwidth_estimator.h" -#include "modules/audio_coding/codecs/isac/fix/source/codec.h" -#include "modules/audio_coding/codecs/isac/fix/source/entropy_coding.h" -#include "modules/audio_coding/codecs/isac/fix/source/pitch_estimator.h" -#include "modules/audio_coding/codecs/isac/fix/source/settings.h" -#include "modules/audio_coding/codecs/isac/fix/source/structs.h" - - - - -int WebRtcIsacfix_DecodeImpl(int16_t* signal_out16, - IsacFixDecoderInstance* ISACdec_obj, - size_t* current_framesamples) -{ - int k; - int err; - int16_t BWno; - int len = 0; - - int16_t model; - - - int16_t Vector_Word16_1[FRAMESAMPLES/2]; - int16_t Vector_Word16_2[FRAMESAMPLES/2]; - - int32_t Vector_Word32_1[FRAMESAMPLES/2]; - int32_t Vector_Word32_2[FRAMESAMPLES/2]; - - int16_t lofilt_coefQ15[ORDERLO*SUBFRAMES]; //refl. coeffs - int16_t hifilt_coefQ15[ORDERHI*SUBFRAMES]; //refl. coeffs - int32_t gain_lo_hiQ17[2*SUBFRAMES]; - - int16_t PitchLags_Q7[PITCH_SUBFRAMES]; - int16_t PitchGains_Q12[PITCH_SUBFRAMES]; - int16_t AvgPitchGain_Q12; - - int16_t tmp_1, tmp_2; - int32_t tmp32a; - int16_t gainQ13; - - - size_t frame_nb; /* counter */ - size_t frame_mode; /* 0 for 30ms, 1 for 60ms */ - static const size_t kProcessedSamples = 480; /* 480 (for both 30, 60 ms) */ - - /* PLC */ - int16_t overlapWin[ 240 ]; - - (ISACdec_obj->bitstr_obj).W_upper = 0xFFFFFFFF; - (ISACdec_obj->bitstr_obj).streamval = 0; - (ISACdec_obj->bitstr_obj).stream_index = 0; - (ISACdec_obj->bitstr_obj).full = 1; - - - /* decode framelength and BW estimation - not used, only for stream pointer*/ - err = WebRtcIsacfix_DecodeFrameLen(&ISACdec_obj->bitstr_obj, current_framesamples); - if (err<0) // error check - return err; - - frame_mode = *current_framesamples / MAX_FRAMESAMPLES; /* 0, or 1 */ - - err = WebRtcIsacfix_DecodeSendBandwidth(&ISACdec_obj->bitstr_obj, &BWno); - if (err<0) // error check - return err; - - /* one loop if it's one frame (30ms), two loops if two frames bundled together - * (60ms) */ - for (frame_nb = 0; frame_nb <= frame_mode; frame_nb++) { - - /* decode & dequantize pitch parameters */ - err = WebRtcIsacfix_DecodePitchGain(&(ISACdec_obj->bitstr_obj), PitchGains_Q12); - if (err<0) // error check - return err; - - err = WebRtcIsacfix_DecodePitchLag(&ISACdec_obj->bitstr_obj, PitchGains_Q12, PitchLags_Q7); - if (err<0) // error check - return err; - - AvgPitchGain_Q12 = (int16_t)(((int32_t)PitchGains_Q12[0] + PitchGains_Q12[1] + PitchGains_Q12[2] + PitchGains_Q12[3])>>2); - - /* decode & dequantize FiltCoef */ - err = WebRtcIsacfix_DecodeLpc(gain_lo_hiQ17, lofilt_coefQ15, hifilt_coefQ15, - &ISACdec_obj->bitstr_obj, &model); - - if (err<0) // error check - return err; - - /* decode & dequantize spectrum */ - len = WebRtcIsacfix_DecodeSpec(&ISACdec_obj->bitstr_obj, Vector_Word16_1, Vector_Word16_2, AvgPitchGain_Q12); - if (len < 0) // error check - return len; - - // Why does this need Q16 in and out? /JS - WebRtcIsacfix_Spec2Time(Vector_Word16_1, Vector_Word16_2, Vector_Word32_1, Vector_Word32_2); - - for (k=0; k Q9. - Vector_Word16_1[k] = (int16_t)((Vector_Word32_1[k] + 64) >> 7); - } - - /* ---- If this is recovery frame ---- */ - if( (ISACdec_obj->plcstr_obj).used == PLC_WAS_USED ) - { - (ISACdec_obj->plcstr_obj).used = PLC_NOT_USED; - if( (ISACdec_obj->plcstr_obj).B < 1000 ) - { - (ISACdec_obj->plcstr_obj).decayCoeffPriodic = 4000; - } - - ISACdec_obj->plcstr_obj.decayCoeffPriodic = WEBRTC_SPL_WORD16_MAX; /* DECAY_RATE is in Q15 */ - ISACdec_obj->plcstr_obj.decayCoeffNoise = WEBRTC_SPL_WORD16_MAX; /* DECAY_RATE is in Q15 */ - ISACdec_obj->plcstr_obj.pitchCycles = 0; - - PitchGains_Q12[0] = (int16_t)(PitchGains_Q12[0] * 700 >> 10); - - /* ---- Add-overlap ---- */ - WebRtcSpl_GetHanningWindow( overlapWin, RECOVERY_OVERLAP ); - for( k = 0; k < RECOVERY_OVERLAP; k++ ) - Vector_Word16_1[k] = WebRtcSpl_AddSatW16( - (int16_t)(ISACdec_obj->plcstr_obj.overlapLP[k] * - overlapWin[RECOVERY_OVERLAP - k - 1] >> 14), - (int16_t)(Vector_Word16_1[k] * overlapWin[k] >> 14)); - - - - } - - /* --- Store side info --- */ - if( frame_nb == frame_mode ) - { - /* --- LPC info */ - WEBRTC_SPL_MEMCPY_W16( (ISACdec_obj->plcstr_obj).lofilt_coefQ15, &lofilt_coefQ15[(SUBFRAMES-1)*ORDERLO], ORDERLO ); - WEBRTC_SPL_MEMCPY_W16( (ISACdec_obj->plcstr_obj).hifilt_coefQ15, &hifilt_coefQ15[(SUBFRAMES-1)*ORDERHI], ORDERHI ); - (ISACdec_obj->plcstr_obj).gain_lo_hiQ17[0] = gain_lo_hiQ17[(SUBFRAMES-1) * 2]; - (ISACdec_obj->plcstr_obj).gain_lo_hiQ17[1] = gain_lo_hiQ17[(SUBFRAMES-1) * 2 + 1]; - - /* --- LTP info */ - (ISACdec_obj->plcstr_obj).AvgPitchGain_Q12 = PitchGains_Q12[3]; - (ISACdec_obj->plcstr_obj).lastPitchGain_Q12 = PitchGains_Q12[3]; - (ISACdec_obj->plcstr_obj).lastPitchLag_Q7 = PitchLags_Q7[3]; - - if( PitchLags_Q7[3] < 3000 ) - (ISACdec_obj->plcstr_obj).lastPitchLag_Q7 += PitchLags_Q7[3]; - - WEBRTC_SPL_MEMCPY_W16( (ISACdec_obj->plcstr_obj).prevPitchInvIn, Vector_Word16_1, FRAMESAMPLES/2 ); - - } - /* ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ */ - - /* inverse pitch filter */ - WebRtcIsacfix_PitchFilter(Vector_Word16_1, Vector_Word16_2, &ISACdec_obj->pitchfiltstr_obj, PitchLags_Q7, PitchGains_Q12, 4); - - if( frame_nb == frame_mode ) - { - WEBRTC_SPL_MEMCPY_W16( (ISACdec_obj->plcstr_obj).prevPitchInvOut, &(Vector_Word16_2[FRAMESAMPLES/2 - (PITCH_MAX_LAG + 10)]), PITCH_MAX_LAG ); - } - - - /* reduce gain to compensate for pitch enhancer */ - /* gain = 1.0f - 0.45f * AvgPitchGain; */ - tmp32a = AvgPitchGain_Q12 * 29; // Q18 - gainQ13 = (int16_t)((262144 - tmp32a) >> 5); // Q18 -> Q13. - - for (k = 0; k < FRAMESAMPLES/2; k++) - { - Vector_Word32_1[k] = (Vector_Word16_2[k] * gainQ13) * (1 << 3); // Q25 - } - - - /* perceptual post-filtering (using normalized lattice filter) */ - WebRtcIsacfix_NormLatticeFilterAr(ORDERLO, (ISACdec_obj->maskfiltstr_obj).PostStateLoGQ0, - Vector_Word32_1, lofilt_coefQ15, gain_lo_hiQ17, 0, Vector_Word16_1); - - /* --- Store Highpass Residual --- */ - for (k = 0; k < FRAMESAMPLES/2; k++) - Vector_Word32_1[k] = Vector_Word32_2[k] * (1 << 9); // Q16 -> Q25 - - for( k = 0; k < PITCH_MAX_LAG + 10; k++ ) - (ISACdec_obj->plcstr_obj).prevHP[k] = Vector_Word32_1[FRAMESAMPLES/2 - (PITCH_MAX_LAG + 10) + k]; - - - WebRtcIsacfix_NormLatticeFilterAr(ORDERHI, (ISACdec_obj->maskfiltstr_obj).PostStateHiGQ0, - Vector_Word32_1, hifilt_coefQ15, gain_lo_hiQ17, 1, Vector_Word16_2); - - /* recombine the 2 bands */ - - /* Form the polyphase signals, and compensate for DC offset */ - for (k=0;kpostfiltbankstr_obj); - - } - return len; -} diff --git a/modules/audio_coding/codecs/isac/fix/source/decode_bwe.c b/modules/audio_coding/codecs/isac/fix/source/decode_bwe.c deleted file mode 100644 index 99676504cd..0000000000 --- a/modules/audio_coding/codecs/isac/fix/source/decode_bwe.c +++ /dev/null @@ -1,69 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/* - * decode_bwe.c - * - * This C file contains the internal decode bandwidth estimate function. - * - */ - - -#include "modules/audio_coding/codecs/isac/fix/source/bandwidth_estimator.h" -#include "modules/audio_coding/codecs/isac/fix/source/codec.h" -#include "modules/audio_coding/codecs/isac/fix/source/entropy_coding.h" -#include "modules/audio_coding/codecs/isac/fix/source/structs.h" - - - - -int WebRtcIsacfix_EstimateBandwidth(BwEstimatorstr *bwest_str, - Bitstr_dec *streamdata, - size_t packet_size, - uint16_t rtp_seq_number, - uint32_t send_ts, - uint32_t arr_ts) -{ - int16_t index; - size_t frame_samples; - int err; - - /* decode framelength */ - err = WebRtcIsacfix_DecodeFrameLen(streamdata, &frame_samples); - /* error check */ - if (err<0) { - return err; - } - - /* decode BW estimation */ - err = WebRtcIsacfix_DecodeSendBandwidth(streamdata, &index); - /* error check */ - if (err<0) { - return err; - } - - /* Update BWE with received data */ - err = WebRtcIsacfix_UpdateUplinkBwImpl( - bwest_str, - rtp_seq_number, - (int16_t)(frame_samples * 1000 / FS), - send_ts, - arr_ts, - packet_size, /* in bytes */ - index); - - /* error check */ - if (err<0) { - return err; - } - - /* Succesful */ - return 0; -} diff --git a/modules/audio_coding/codecs/isac/fix/source/decode_plc.c b/modules/audio_coding/codecs/isac/fix/source/decode_plc.c deleted file mode 100644 index 873cf951ba..0000000000 --- a/modules/audio_coding/codecs/isac/fix/source/decode_plc.c +++ /dev/null @@ -1,805 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/* - * decode_plc.c - * - * Packet Loss Concealment. - * - */ - -#include - -#include "modules/audio_coding/codecs/isac/fix/source/settings.h" -#include "modules/audio_coding/codecs/isac/fix/source/entropy_coding.h" -#include "modules/audio_coding/codecs/isac/fix/source/pitch_estimator.h" -#include "modules/audio_coding/codecs/isac/fix/source/bandwidth_estimator.h" -#include "modules/audio_coding/codecs/isac/fix/source/structs.h" -#include "modules/audio_coding/codecs/isac/fix/source/codec.h" - - -#define NO_OF_PRIMES 8 -#define NOISE_FILTER_LEN 30 - -/* - * function to decode the bitstream - * returns the total number of bytes in the stream - */ - -static int16_t plc_filterma_Fast( - int16_t *In, /* (i) Vector to be filtered. InOut[-orderCoef+1] - to InOut[-1] contains state */ - int16_t *Out, /* (o) Filtered vector */ - int16_t *B, /* (i) The filter coefficients (in Q0) */ - int16_t Blen, /* (i) Number of B coefficients */ - int16_t len, /* (i) Number of samples to be filtered */ - int16_t reduceDecay, - int16_t decay, - int16_t rshift ) -{ - int i, j; - int32_t o; - int32_t lim = (1 << (15 + rshift)) - 1; - - for (i = 0; i < len; i++) - { - const int16_t *b_ptr = &B[0]; - const int16_t *x_ptr = &In[i]; - - o = (int32_t)0; - - for (j = 0;j < Blen; j++) - { - o = WebRtcSpl_AddSatW32(o, *b_ptr * *x_ptr); - b_ptr++; - x_ptr--; - } - - /* to round off correctly */ - o = WebRtcSpl_AddSatW32(o, 1 << (rshift - 1)); - - /* saturate according to the domain of the filter coefficients */ - o = WEBRTC_SPL_SAT((int32_t)lim, o, (int32_t)-lim); - - /* o should be in the range of int16_t */ - o >>= rshift; - - /* decay the output signal; this is specific to plc */ - *Out++ = (int16_t)((int16_t)o * decay >> 15); - - /* change the decay */ - decay -= reduceDecay; - if( decay < 0 ) - decay = 0; - } - return( decay ); -} - - - - - - - - -static __inline int32_t log2_Q8_T( uint32_t x ) { - - int32_t zeros; - int16_t frac; - - zeros=WebRtcSpl_NormU32(x); - frac = (int16_t)(((x << zeros) & 0x7FFFFFFF) >> 23); - - /* log2(magn(i)) */ - return ((31 - zeros) << 8) + frac; -} - -static __inline int16_t exp2_Q10_T(int16_t x) { // Both in and out in Q10 - - int16_t tmp16_1, tmp16_2; - - tmp16_2=(int16_t)(0x0400|(x&0x03FF)); - tmp16_1 = -(x >> 10); - if(tmp16_1>0) - return tmp16_2 >> tmp16_1; - else - return tmp16_2 << -tmp16_1; - -} - - -/* - This is a fixed-point version of the above code with limLow = 700 and limHigh = 5000, - hard-coded. The values 700 and 5000 were experimentally obtained. - - The function implements membership values for two sets. The mebership functions are - of second orders corresponding to half-bell-shapped pulses. -*/ -static void MemshipValQ15( int16_t in, int16_t *A, int16_t *B ) -{ - int16_t x; - - in -= 700; /* translate the lowLim to 0, limHigh = 5000 - 700, M = 2150 */ - - if( in <= 2150 ) - { - if( in > 0 ) - { - /* b = in^2 / (2 * M^2), a = 1 - b in Q0. - We have to compute in Q15 */ - - /* x = in / 2150 {in Q15} = x * 15.2409 {in Q15} = - x*15 + (x*983)/(2^12); note that 983/2^12 = 0.23999 */ - - /* we are sure that x is in the range of int16_t */ - x = (int16_t)(in * 15 + (in * 983 >> 12)); - /* b = x^2 / 2 {in Q15} so a shift of 16 is required to - be in correct domain and one more for the division by 2 */ - *B = (int16_t)((x * x + 0x00010000) >> 17); - *A = WEBRTC_SPL_WORD16_MAX - *B; - } - else - { - *B = 0; - *A = WEBRTC_SPL_WORD16_MAX; - } - } - else - { - if( in < 4300 ) - { - /* This is a mirror case of the above */ - in = 4300 - in; - x = (int16_t)(in * 15 + (in * 983 >> 12)); - /* b = x^2 / 2 {in Q15} so a shift of 16 is required to - be in correct domain and one more for the division by 2 */ - *A = (int16_t)((x * x + 0x00010000) >> 17); - *B = WEBRTC_SPL_WORD16_MAX - *A; - - } - else - { - *A = 0; - *B = WEBRTC_SPL_WORD16_MAX; - } - } -} - - - - -static void LinearResampler(int16_t* in, - int16_t* out, - size_t lenIn, - size_t lenOut) -{ - size_t n = (lenIn - 1) * RESAMP_RES; - int16_t resOut, relativePos, diff; /* */ - size_t i, j; - uint16_t udiff; - - if( lenIn == lenOut ) - { - WEBRTC_SPL_MEMCPY_W16( out, in, lenIn ); - return; - } - - resOut = WebRtcSpl_DivW32W16ResW16( (int32_t)n, (int16_t)(lenOut-1) ); - - out[0] = in[0]; - for( i = 1, j = 0, relativePos = 0; i < lenOut; i++ ) - { - - relativePos += resOut; - while( relativePos > RESAMP_RES ) - { - j++; - relativePos -= RESAMP_RES; - } - - - /* an overflow may happen and the differce in sample values may - * require more than 16 bits. We like to avoid 32 bit arithmatic - * as much as possible */ - - if( (in[ j ] > 0) && (in[j + 1] < 0) ) - { - udiff = (uint16_t)(in[ j ] - in[j + 1]); - out[ i ] = in[ j ] - (uint16_t)( ((int32_t)( udiff * relativePos )) >> RESAMP_RES_BIT); - } - else - { - if( (in[j] < 0) && (in[j+1] > 0) ) - { - udiff = (uint16_t)( in[j + 1] - in[ j ] ); - out[ i ] = in[ j ] + (uint16_t)( ((int32_t)( udiff * relativePos )) >> RESAMP_RES_BIT); - } - else - { - diff = in[ j + 1 ] - in[ j ]; - out[i] = in[j] + (int16_t)(diff * relativePos >> RESAMP_RES_BIT); - } - } - } -} - - - - - -void WebRtcIsacfix_DecodePlcImpl(int16_t *signal_out16, - IsacFixDecoderInstance *ISACdec_obj, - size_t *current_framesamples ) -{ - int subframecnt; - - int16_t* Vector_Word16_1; - int16_t Vector_Word16_Extended_1[FRAMESAMPLES_HALF + NOISE_FILTER_LEN]; - int16_t* Vector_Word16_2; - int16_t Vector_Word16_Extended_2[FRAMESAMPLES_HALF + NOISE_FILTER_LEN]; - - int32_t Vector_Word32_1[FRAMESAMPLES_HALF]; - int32_t Vector_Word32_2[FRAMESAMPLES_HALF]; - - int16_t lofilt_coefQ15[ORDERLO*SUBFRAMES]; //refl. coeffs - int16_t hifilt_coefQ15[ORDERHI*SUBFRAMES]; //refl. coeffs - - int16_t pitchLags_Q7[PITCH_SUBFRAMES]; - int16_t pitchGains_Q12[PITCH_SUBFRAMES]; - - int16_t tmp_1, tmp_2; - int32_t tmp32a, tmp32b; - int16_t gainQ13; - - int16_t myDecayRate; - - /* ---------- PLC variables ------------ */ - size_t lag0, i, k; - int16_t noiseIndex; - int16_t stretchPitchLP[PITCH_MAX_LAG + 10], stretchPitchLP1[PITCH_MAX_LAG + 10]; - - int32_t gain_lo_hiQ17[2*SUBFRAMES]; - - int16_t nLP, pLP, wNoisyLP, wPriodicLP, tmp16; - size_t minIdx; - int32_t nHP, pHP, wNoisyHP, wPriodicHP, corr, minCorr, maxCoeff; - int16_t noise1, rshift; - - - int16_t ltpGain, pitchGain, myVoiceIndicator, myAbs, maxAbs; - int32_t varIn, varOut, logVarIn, logVarOut, Q, logMaxAbs; - int rightShiftIn, rightShiftOut; - - - /* ------------------------------------- */ - - - myDecayRate = (DECAY_RATE); - Vector_Word16_1 = &Vector_Word16_Extended_1[NOISE_FILTER_LEN]; - Vector_Word16_2 = &Vector_Word16_Extended_2[NOISE_FILTER_LEN]; - - - /* ----- Simply Copy Previous LPC parameters ------ */ - for( subframecnt = 0; subframecnt < SUBFRAMES; subframecnt++ ) - { - /* lower Band */ - WEBRTC_SPL_MEMCPY_W16(&lofilt_coefQ15[ subframecnt * ORDERLO ], - (ISACdec_obj->plcstr_obj).lofilt_coefQ15, ORDERLO); - gain_lo_hiQ17[2*subframecnt] = (ISACdec_obj->plcstr_obj).gain_lo_hiQ17[0]; - - /* Upper Band */ - WEBRTC_SPL_MEMCPY_W16(&hifilt_coefQ15[ subframecnt * ORDERHI ], - (ISACdec_obj->plcstr_obj).hifilt_coefQ15, ORDERHI); - gain_lo_hiQ17[2*subframecnt + 1] = (ISACdec_obj->plcstr_obj).gain_lo_hiQ17[1]; - } - - - - - lag0 = (size_t)(((ISACdec_obj->plcstr_obj.lastPitchLag_Q7 + 64) >> 7) + 1); - - - if( (ISACdec_obj->plcstr_obj).used != PLC_WAS_USED ) - { - (ISACdec_obj->plcstr_obj).pitchCycles = 0; - - (ISACdec_obj->plcstr_obj).lastPitchLP = - &((ISACdec_obj->plcstr_obj).prevPitchInvIn[FRAMESAMPLES_HALF - lag0]); - minCorr = WEBRTC_SPL_WORD32_MAX; - - if ((FRAMESAMPLES_HALF - 10) > 2 * lag0) - { - minIdx = 11; - for( i = 0; i < 21; i++ ) - { - corr = 0; - for( k = 0; k < lag0; k++ ) - { - corr = WebRtcSpl_AddSatW32(corr, WEBRTC_SPL_ABS_W32( - WebRtcSpl_SubSatW16( - (ISACdec_obj->plcstr_obj).lastPitchLP[k], - (ISACdec_obj->plcstr_obj).prevPitchInvIn[ - FRAMESAMPLES_HALF - 2*lag0 - 10 + i + k ] ) ) ); - } - if( corr < minCorr ) - { - minCorr = corr; - minIdx = i; - } - } - (ISACdec_obj->plcstr_obj).prevPitchLP = - &( (ISACdec_obj->plcstr_obj).prevPitchInvIn[ - FRAMESAMPLES_HALF - lag0*2 - 10 + minIdx] ); - } - else - { - (ISACdec_obj->plcstr_obj).prevPitchLP = - (ISACdec_obj->plcstr_obj).lastPitchLP; - } - pitchGain = (ISACdec_obj->plcstr_obj).lastPitchGain_Q12; - - WebRtcSpl_AutoCorrelation( - &(ISACdec_obj->plcstr_obj).prevPitchInvIn[FRAMESAMPLES_HALF - lag0], - lag0, 0, &varIn, &rightShiftIn); - WebRtcSpl_AutoCorrelation( - &(ISACdec_obj->plcstr_obj).prevPitchInvOut[PITCH_MAX_LAG + 10 - lag0], - lag0, 0, &varOut, &rightShiftOut); - - maxAbs = 0; - for( i = 0; i< lag0; i++) - { - myAbs = WEBRTC_SPL_ABS_W16( - (ISACdec_obj->plcstr_obj).prevPitchInvOut[ - PITCH_MAX_LAG + 10 - lag0 + i] ); - maxAbs = (myAbs > maxAbs)? myAbs:maxAbs; - } - logVarIn = log2_Q8_T( (uint32_t)( varIn ) ) + - (int32_t)(rightShiftIn << 8); - logVarOut = log2_Q8_T( (uint32_t)( varOut ) ) + - (int32_t)(rightShiftOut << 8); - logMaxAbs = log2_Q8_T( (uint32_t)( maxAbs ) ); - - ltpGain = (int16_t)(logVarOut - logVarIn); - Q = 2 * logMaxAbs - ( logVarOut - 1512 ); - - /* - * --- - * We are computing sqrt( (VarIn/lag0) / var( noise ) ) - * var( noise ) is almost 256. we have already computed log2( VarIn ) in Q8 - * so we actually compute 2^( 0.5*(log2( VarIn ) - log2( lag0 ) - log2( var(noise ) ) ). - * Note that put log function is in Q8 but the exponential function is in Q10. - * -- - */ - - logVarIn -= log2_Q8_T( (uint32_t)( lag0 ) ); - tmp16 = (int16_t)((logVarIn<<1) - (4<<10) ); - rightShiftIn = 0; - if( tmp16 > 4096 ) - { - tmp16 -= 4096; - tmp16 = exp2_Q10_T( tmp16 ); - tmp16 >>= 6; - } - else - tmp16 = exp2_Q10_T( tmp16 )>>10; - - (ISACdec_obj->plcstr_obj).std = tmp16 - 4; - - if( (ltpGain < 110) || (ltpGain > 230) ) - { - if( ltpGain < 100 && (pitchGain < 1800) ) - { - (ISACdec_obj->plcstr_obj).A = WEBRTC_SPL_WORD16_MAX; - } - else - { - (ISACdec_obj->plcstr_obj).A = ((ltpGain < 110) && (Q < 800) - )? WEBRTC_SPL_WORD16_MAX:0; - } - (ISACdec_obj->plcstr_obj).B = WEBRTC_SPL_WORD16_MAX - - (ISACdec_obj->plcstr_obj).A; - } - else - { - if( (pitchGain < 450) || (pitchGain > 1600) ) - { - (ISACdec_obj->plcstr_obj).A = ((pitchGain < 450) - )? WEBRTC_SPL_WORD16_MAX:0; - (ISACdec_obj->plcstr_obj).B = WEBRTC_SPL_WORD16_MAX - - (ISACdec_obj->plcstr_obj).A; - } - else - { - myVoiceIndicator = ltpGain * 2 + pitchGain; - MemshipValQ15( myVoiceIndicator, - &(ISACdec_obj->plcstr_obj).A, &(ISACdec_obj->plcstr_obj).B ); - } - } - - - - myVoiceIndicator = ltpGain * 16 + pitchGain * 2 + (pitchGain >> 8); - MemshipValQ15( myVoiceIndicator, - &(ISACdec_obj->plcstr_obj).A, &(ISACdec_obj->plcstr_obj).B ); - - - - (ISACdec_obj->plcstr_obj).stretchLag = lag0; - (ISACdec_obj->plcstr_obj).pitchIndex = 0; - - } - else - { - myDecayRate = (DECAY_RATE<<2); - } - - if( (ISACdec_obj->plcstr_obj).B < 1000 ) - { - myDecayRate += (DECAY_RATE<<3); - } - - /* ------------ reconstructing the residual signal ------------------ */ - - LinearResampler( (ISACdec_obj->plcstr_obj).lastPitchLP, - stretchPitchLP, lag0, (ISACdec_obj->plcstr_obj).stretchLag ); - /* inverse pitch filter */ - - pitchLags_Q7[0] = pitchLags_Q7[1] = pitchLags_Q7[2] = pitchLags_Q7[3] = - (int16_t)((ISACdec_obj->plcstr_obj).stretchLag<<7); - pitchGains_Q12[3] = ( (ISACdec_obj->plcstr_obj).lastPitchGain_Q12); - pitchGains_Q12[2] = (int16_t)(pitchGains_Q12[3] * 1010 >> 10); - pitchGains_Q12[1] = (int16_t)(pitchGains_Q12[2] * 1010 >> 10); - pitchGains_Q12[0] = (int16_t)(pitchGains_Q12[1] * 1010 >> 10); - - - /* most of the time either B or A are zero so seperating */ - if( (ISACdec_obj->plcstr_obj).B == 0 ) - { - for( i = 0; i < FRAMESAMPLES_HALF; i++ ) - { - /* --- Low Pass */ - (ISACdec_obj->plcstr_obj).seed = WEBRTC_SPL_RAND( - (ISACdec_obj->plcstr_obj).seed ); - Vector_Word16_1[i] = (ISACdec_obj->plcstr_obj.seed >> 10) - 16; - - /* --- Highpass */ - (ISACdec_obj->plcstr_obj).seed = WEBRTC_SPL_RAND( - (ISACdec_obj->plcstr_obj).seed ); - Vector_Word16_2[i] = (ISACdec_obj->plcstr_obj.seed >> 10) - 16; - - } - for( i = 1; i < NOISE_FILTER_LEN; i++ ) - { - (ISACdec_obj->plcstr_obj).seed = WEBRTC_SPL_RAND( - (ISACdec_obj->plcstr_obj).seed ); - Vector_Word16_Extended_1[i] = (ISACdec_obj->plcstr_obj.seed >> 10) - 16; - - (ISACdec_obj->plcstr_obj).seed = WEBRTC_SPL_RAND( - (ISACdec_obj->plcstr_obj).seed ); - Vector_Word16_Extended_2[i] = (ISACdec_obj->plcstr_obj.seed >> 10) - 16; - } - plc_filterma_Fast(Vector_Word16_1, Vector_Word16_Extended_1, - &(ISACdec_obj->plcstr_obj).prevPitchInvIn[FRAMESAMPLES_HALF - - NOISE_FILTER_LEN], (int16_t) NOISE_FILTER_LEN, - (int16_t) FRAMESAMPLES_HALF, (int16_t)(5), - (ISACdec_obj->plcstr_obj).decayCoeffNoise, (int16_t)(6)); - - maxCoeff = WebRtcSpl_MaxAbsValueW32( - &(ISACdec_obj->plcstr_obj).prevHP[ - PITCH_MAX_LAG + 10 - NOISE_FILTER_LEN], NOISE_FILTER_LEN ); - - rshift = 0; - while( maxCoeff > WEBRTC_SPL_WORD16_MAX ) - { - maxCoeff >>= 1; - rshift++; - } - for( i = 0; i < NOISE_FILTER_LEN; i++ ) { - Vector_Word16_1[FRAMESAMPLES_HALF - NOISE_FILTER_LEN + i] =(int16_t)( - ISACdec_obj->plcstr_obj.prevHP[PITCH_MAX_LAG + 10 - NOISE_FILTER_LEN + - i] >> rshift); - } - (ISACdec_obj->plcstr_obj).decayCoeffNoise = plc_filterma_Fast( - Vector_Word16_2, - Vector_Word16_Extended_2, - &Vector_Word16_1[FRAMESAMPLES_HALF - NOISE_FILTER_LEN], - (int16_t) NOISE_FILTER_LEN, - (int16_t) FRAMESAMPLES_HALF, - (int16_t) (5), - (ISACdec_obj->plcstr_obj).decayCoeffNoise, - (int16_t) (7) ); - - for( i = 0; i < FRAMESAMPLES_HALF; i++ ) - Vector_Word32_2[i] = Vector_Word16_Extended_2[i] << rshift; - - Vector_Word16_1 = Vector_Word16_Extended_1; - } - else - { - if( (ISACdec_obj->plcstr_obj).A == 0 ) - { - /* ------ Periodic Vector --- */ - for( i = 0, noiseIndex = 0; i < FRAMESAMPLES_HALF; i++, noiseIndex++ ) - { - /* --- Lowpass */ - pLP = (int16_t)(stretchPitchLP[ISACdec_obj->plcstr_obj.pitchIndex] * - ISACdec_obj->plcstr_obj.decayCoeffPriodic >> 15); - - /* --- Highpass */ - pHP = (int32_t)WEBRTC_SPL_MUL_16_32_RSFT15( - (ISACdec_obj->plcstr_obj).decayCoeffPriodic, - (ISACdec_obj->plcstr_obj).prevHP[PITCH_MAX_LAG + 10 - - (ISACdec_obj->plcstr_obj).stretchLag + - (ISACdec_obj->plcstr_obj).pitchIndex] ); - - /* --- lower the muliplier (more decay at next sample) --- */ - (ISACdec_obj->plcstr_obj).decayCoeffPriodic -= (myDecayRate); - if( (ISACdec_obj->plcstr_obj).decayCoeffPriodic < 0 ) - (ISACdec_obj->plcstr_obj).decayCoeffPriodic = 0; - - (ISACdec_obj->plcstr_obj).pitchIndex++; - - if( (ISACdec_obj->plcstr_obj).pitchIndex == - (ISACdec_obj->plcstr_obj).stretchLag ) - { - (ISACdec_obj->plcstr_obj).pitchIndex = 0; - (ISACdec_obj->plcstr_obj).pitchCycles++; - - if( (ISACdec_obj->plcstr_obj).stretchLag != (lag0 + 1) ) - { - (ISACdec_obj->plcstr_obj).stretchLag = lag0 + 1; - } - else - { - (ISACdec_obj->plcstr_obj).stretchLag = lag0; - } - - (ISACdec_obj->plcstr_obj).stretchLag = ( - (ISACdec_obj->plcstr_obj).stretchLag > PITCH_MAX_LAG - )? (PITCH_MAX_LAG):(ISACdec_obj->plcstr_obj).stretchLag; - - LinearResampler( (ISACdec_obj->plcstr_obj).lastPitchLP, - stretchPitchLP, lag0, (ISACdec_obj->plcstr_obj).stretchLag ); - - LinearResampler( (ISACdec_obj->plcstr_obj).prevPitchLP, - stretchPitchLP1, lag0, (ISACdec_obj->plcstr_obj).stretchLag ); - - switch( (ISACdec_obj->plcstr_obj).pitchCycles ) - { - case 1: - { - for( k=0; k<(ISACdec_obj->plcstr_obj).stretchLag; k++ ) - { - stretchPitchLP[k] = (int16_t)(( - (int32_t)stretchPitchLP[k]* 3 + - (int32_t)stretchPitchLP1[k])>>2); - } - break; - } - case 2: - { - for( k=0; k<(ISACdec_obj->plcstr_obj).stretchLag; k++ ) - { - stretchPitchLP[k] = (int16_t)(( - (int32_t)stretchPitchLP[k] + - (int32_t)stretchPitchLP1[k] )>>1); - } - break; - } - case 3: - { - for( k=0; k<(ISACdec_obj->plcstr_obj).stretchLag; k++ ) - { - stretchPitchLP[k] = (int16_t)((stretchPitchLP[k] + - (int32_t)stretchPitchLP1[k]*3 )>>2); - } - break; - } - } - - if( (ISACdec_obj->plcstr_obj).pitchCycles == 3 ) - { - myDecayRate += 35; //(myDecayRate>>1); - (ISACdec_obj->plcstr_obj).pitchCycles = 0; - } - - } - - /* ------ Sum the noisy and periodic signals ------ */ - Vector_Word16_1[i] = pLP; - Vector_Word32_2[i] = pHP; - } - } - else - { - for( i = 0, noiseIndex = 0; i < FRAMESAMPLES_HALF; i++, noiseIndex++ ) - { - - (ISACdec_obj->plcstr_obj).seed = WEBRTC_SPL_RAND( - (ISACdec_obj->plcstr_obj).seed ); - - noise1 = (ISACdec_obj->plcstr_obj.seed >> 10) - 16; - - nLP = (int16_t)((int16_t)(noise1 * ISACdec_obj->plcstr_obj.std) * - ISACdec_obj->plcstr_obj.decayCoeffNoise >> 15); - - /* --- Highpass */ - (ISACdec_obj->plcstr_obj).seed = WEBRTC_SPL_RAND( - (ISACdec_obj->plcstr_obj).seed ); - noise1 = (ISACdec_obj->plcstr_obj.seed >> 11) - 8; - - nHP = (int32_t)WEBRTC_SPL_MUL_16_32_RSFT15( - (ISACdec_obj->plcstr_obj).decayCoeffNoise, - (int32_t)(noise1*(ISACdec_obj->plcstr_obj).std) ); - - /* --- lower the muliplier (more decay at next sample) --- */ - (ISACdec_obj->plcstr_obj).decayCoeffNoise -= (myDecayRate); - if( (ISACdec_obj->plcstr_obj).decayCoeffNoise < 0 ) - (ISACdec_obj->plcstr_obj).decayCoeffNoise = 0; - - /* ------ Periodic Vector --- */ - /* --- Lowpass */ - pLP = (int16_t)(stretchPitchLP[ISACdec_obj->plcstr_obj.pitchIndex] * - ISACdec_obj->plcstr_obj.decayCoeffPriodic >> 15); - - /* --- Highpass */ - pHP = (int32_t)WEBRTC_SPL_MUL_16_32_RSFT15( - (ISACdec_obj->plcstr_obj).decayCoeffPriodic, - (ISACdec_obj->plcstr_obj).prevHP[PITCH_MAX_LAG + 10 - - (ISACdec_obj->plcstr_obj).stretchLag + - (ISACdec_obj->plcstr_obj).pitchIndex] ); - - /* --- lower the muliplier (more decay at next sample) --- */ - (ISACdec_obj->plcstr_obj).decayCoeffPriodic -= (myDecayRate); - if( (ISACdec_obj->plcstr_obj).decayCoeffPriodic < 0 ) - { - (ISACdec_obj->plcstr_obj).decayCoeffPriodic = 0; - } - - /* ------ Weighting the noisy and periodic vectors ------- */ - wNoisyLP = (int16_t)(ISACdec_obj->plcstr_obj.A * nLP >> 15); - wNoisyHP = (int32_t)(WEBRTC_SPL_MUL_16_32_RSFT15( - (ISACdec_obj->plcstr_obj).A, (nHP) ) ); - - wPriodicLP = (int16_t)(ISACdec_obj->plcstr_obj.B * pLP >> 15); - wPriodicHP = (int32_t)(WEBRTC_SPL_MUL_16_32_RSFT15( - (ISACdec_obj->plcstr_obj).B, pHP)); - - (ISACdec_obj->plcstr_obj).pitchIndex++; - - if((ISACdec_obj->plcstr_obj).pitchIndex == - (ISACdec_obj->plcstr_obj).stretchLag) - { - (ISACdec_obj->plcstr_obj).pitchIndex = 0; - (ISACdec_obj->plcstr_obj).pitchCycles++; - - if( (ISACdec_obj->plcstr_obj).stretchLag != (lag0 + 1) ) - (ISACdec_obj->plcstr_obj).stretchLag = lag0 + 1; - else - (ISACdec_obj->plcstr_obj).stretchLag = lag0; - - (ISACdec_obj->plcstr_obj).stretchLag = ( - (ISACdec_obj->plcstr_obj).stretchLag > PITCH_MAX_LAG - )? (PITCH_MAX_LAG):(ISACdec_obj->plcstr_obj).stretchLag; - LinearResampler( - (ISACdec_obj->plcstr_obj).lastPitchLP, - stretchPitchLP, lag0, (ISACdec_obj->plcstr_obj).stretchLag ); - - LinearResampler((ISACdec_obj->plcstr_obj).prevPitchLP, - stretchPitchLP1, lag0, (ISACdec_obj->plcstr_obj).stretchLag ); - - switch((ISACdec_obj->plcstr_obj).pitchCycles) - { - case 1: - { - for( k=0; k<(ISACdec_obj->plcstr_obj).stretchLag; k++ ) - { - stretchPitchLP[k] = (int16_t)(( - (int32_t)stretchPitchLP[k]* 3 + - (int32_t)stretchPitchLP1[k] )>>2); - } - break; - } - case 2: - { - for( k=0; k<(ISACdec_obj->plcstr_obj).stretchLag; k++ ) - { - stretchPitchLP[k] = (int16_t)(( - (int32_t)stretchPitchLP[k] + - (int32_t)stretchPitchLP1[k])>>1); - } - break; - } - case 3: - { - for( k=0; k<(ISACdec_obj->plcstr_obj).stretchLag; k++ ) - { - stretchPitchLP[k] = (int16_t)( - (stretchPitchLP[k] + - (int32_t)stretchPitchLP1[k]*3 )>>2); - } - break; - } - } - - if( (ISACdec_obj->plcstr_obj).pitchCycles == 3 ) - { - myDecayRate += 55; //(myDecayRate>>1); - (ISACdec_obj->plcstr_obj).pitchCycles = 0; - } - } - - /* ------ Sum the noisy and periodic signals ------ */ - Vector_Word16_1[i] = WebRtcSpl_AddSatW16(wNoisyLP, wPriodicLP); - Vector_Word32_2[i] = WebRtcSpl_AddSatW32(wNoisyHP, wPriodicHP); - } - } - } - /* ----------------- residual signal is reconstructed ------------------ */ - - k = (ISACdec_obj->plcstr_obj).pitchIndex; - /* --- Write one pitch cycle for recovery block --- */ - - for( i = 0; i < RECOVERY_OVERLAP; i++ ) - { - ISACdec_obj->plcstr_obj.overlapLP[i] = (int16_t)( - stretchPitchLP[k] * ISACdec_obj->plcstr_obj.decayCoeffPriodic >> 15); - k = ( k < ((ISACdec_obj->plcstr_obj).stretchLag - 1) )? (k+1):0; - } - - (ISACdec_obj->plcstr_obj).lastPitchLag_Q7 = - (int16_t)((ISACdec_obj->plcstr_obj).stretchLag << 7); - - - /* --- Inverse Pitch Filter --- */ - WebRtcIsacfix_PitchFilter(Vector_Word16_1, Vector_Word16_2, - &ISACdec_obj->pitchfiltstr_obj, pitchLags_Q7, pitchGains_Q12, 4); - - /* reduce gain to compensate for pitch enhancer */ - /* gain = 1.0f - 0.45f * AvgPitchGain; */ - tmp32a = ISACdec_obj->plcstr_obj.AvgPitchGain_Q12 * 29; // Q18 - tmp32b = 262144 - tmp32a; // Q18 - gainQ13 = (int16_t) (tmp32b >> 5); // Q13 - - /* perceptual post-filtering (using normalized lattice filter) */ - for (k = 0; k < FRAMESAMPLES_HALF; k++) - Vector_Word32_1[k] = (Vector_Word16_2[k] * gainQ13) << 3; // Q25 - - - WebRtcIsacfix_NormLatticeFilterAr(ORDERLO, - (ISACdec_obj->maskfiltstr_obj).PostStateLoGQ0, - Vector_Word32_1, lofilt_coefQ15, gain_lo_hiQ17, 0, Vector_Word16_1); - - WebRtcIsacfix_NormLatticeFilterAr(ORDERHI, - (ISACdec_obj->maskfiltstr_obj).PostStateHiGQ0, - Vector_Word32_2, hifilt_coefQ15, gain_lo_hiQ17, 1, Vector_Word16_2); - - /* recombine the 2 bands */ - - /* Form the polyphase signals, and compensate for DC offset */ - for (k=0;kpostfiltbankstr_obj); - - (ISACdec_obj->plcstr_obj).used = PLC_WAS_USED; - *current_framesamples = 480; -} diff --git a/modules/audio_coding/codecs/isac/fix/source/encode.c b/modules/audio_coding/codecs/isac/fix/source/encode.c deleted file mode 100644 index ef3e320e2c..0000000000 --- a/modules/audio_coding/codecs/isac/fix/source/encode.c +++ /dev/null @@ -1,635 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/* - * encode.c - * - * Encoding function for the iSAC coder. - * - */ - -#include "rtc_base/checks.h" -#include "modules/audio_coding/codecs/isac/fix/source/codec.h" - -#include - -#include "modules/audio_coding/codecs/isac/fix/source/arith_routins.h" -#include "modules/audio_coding/codecs/isac/fix/source/bandwidth_estimator.h" -#include "modules/audio_coding/codecs/isac/fix/source/entropy_coding.h" -#include "modules/audio_coding/codecs/isac/fix/source/lpc_masking_model.h" -#include "modules/audio_coding/codecs/isac/fix/source/lpc_tables.h" -#include "modules/audio_coding/codecs/isac/fix/source/pitch_estimator.h" -#include "modules/audio_coding/codecs/isac/fix/source/pitch_gain_tables.h" -#include "modules/audio_coding/codecs/isac/fix/source/pitch_lag_tables.h" -#include "modules/audio_coding/codecs/isac/fix/source/structs.h" - - -int WebRtcIsacfix_EncodeImpl(int16_t *in, - IsacFixEncoderInstance *ISACenc_obj, - BwEstimatorstr *bw_estimatordata, - int16_t CodingMode) -{ - int16_t stream_length = 0; - int16_t usefulstr_len = 0; - int k; - int16_t BWno; - - int16_t lofilt_coefQ15[(ORDERLO)*SUBFRAMES]; - int16_t hifilt_coefQ15[(ORDERHI)*SUBFRAMES]; - int32_t gain_lo_hiQ17[2*SUBFRAMES]; - - int16_t LPandHP[FRAMESAMPLES/2 + QLOOKAHEAD]; - int16_t LP16a[FRAMESAMPLES/2 + QLOOKAHEAD]; - int16_t HP16a[FRAMESAMPLES/2 + QLOOKAHEAD]; - - int16_t PitchLags_Q7[PITCH_SUBFRAMES]; - int16_t PitchGains_Q12[PITCH_SUBFRAMES]; - int16_t AvgPitchGain_Q12; - - int16_t frame_mode; /* 0 for 30ms, 1 for 60ms */ - int16_t processed_samples; - int status; - - int32_t bits_gainsQ11; - int16_t MinBytes; - int16_t bmodel; - - transcode_obj transcodingParam; - int16_t payloadLimitBytes; - int16_t arithLenBeforeEncodingDFT; - int16_t iterCntr; - - /* copy new frame length and bottle neck rate only for the first 10 ms data */ - if (ISACenc_obj->buffer_index == 0) { - /* set the framelength for the next packet */ - ISACenc_obj->current_framesamples = ISACenc_obj->new_framelength; - } - - frame_mode = ISACenc_obj->current_framesamples/MAX_FRAMESAMPLES; /* 0 (30 ms) or 1 (60 ms) */ - processed_samples = ISACenc_obj->current_framesamples/(frame_mode+1); /* 480 (30, 60 ms) */ - - /* buffer speech samples (by 10ms packet) until the framelength is reached (30 or 60 ms) */ - /**************************************************************************************/ - /* fill the buffer with 10ms input data */ - for(k=0; kdata_buffer_fix[k + ISACenc_obj->buffer_index] = in[k]; - } - /* if buffersize is not equal to current framesize, and end of file is not reached yet, */ - /* increase index and go back to main to get more speech samples */ - if (ISACenc_obj->buffer_index + FRAMESAMPLES_10ms != processed_samples) { - ISACenc_obj->buffer_index = ISACenc_obj->buffer_index + FRAMESAMPLES_10ms; - return 0; - } - /* if buffer reached the right size, reset index and continue with encoding the frame */ - ISACenc_obj->buffer_index = 0; - - /* end of buffer function */ - /**************************/ - - /* encoding */ - /************/ - - if (frame_mode == 0 || ISACenc_obj->frame_nb == 0 ) - { - /* reset bitstream */ - ISACenc_obj->bitstr_obj.W_upper = 0xFFFFFFFF; - ISACenc_obj->bitstr_obj.streamval = 0; - ISACenc_obj->bitstr_obj.stream_index = 0; - ISACenc_obj->bitstr_obj.full = 1; - - if (CodingMode == 0) { - ISACenc_obj->BottleNeck = WebRtcIsacfix_GetUplinkBandwidth(bw_estimatordata); - ISACenc_obj->MaxDelay = WebRtcIsacfix_GetUplinkMaxDelay(bw_estimatordata); - } - if (CodingMode == 0 && frame_mode == 0 && (ISACenc_obj->enforceFrameSize == 0)) { - ISACenc_obj->new_framelength = WebRtcIsacfix_GetNewFrameLength(ISACenc_obj->BottleNeck, - ISACenc_obj->current_framesamples); - } - - // multiply the bottleneck by 0.88 before computing SNR, 0.88 is tuned by experimenting on TIMIT - // 901/1024 is 0.87988281250000 - ISACenc_obj->s2nr = WebRtcIsacfix_GetSnr( - (int16_t)(ISACenc_obj->BottleNeck * 901 >> 10), - ISACenc_obj->current_framesamples); - - /* encode frame length */ - status = WebRtcIsacfix_EncodeFrameLen(ISACenc_obj->current_framesamples, &ISACenc_obj->bitstr_obj); - if (status < 0) - { - /* Wrong frame size */ - if (frame_mode == 1 && ISACenc_obj->frame_nb == 1) - { - // If this is the second 30ms of a 60ms frame reset this such that in the next call - // encoder starts fresh. - ISACenc_obj->frame_nb = 0; - } - return status; - } - - /* Save framelength for multiple packets memory */ - if (ISACenc_obj->SaveEnc_ptr != NULL) { - (ISACenc_obj->SaveEnc_ptr)->framelength=ISACenc_obj->current_framesamples; - } - - /* bandwidth estimation and coding */ - BWno = WebRtcIsacfix_GetDownlinkBwIndexImpl(bw_estimatordata); - status = WebRtcIsacfix_EncodeReceiveBandwidth(&BWno, &ISACenc_obj->bitstr_obj); - if (status < 0) - { - if (frame_mode == 1 && ISACenc_obj->frame_nb == 1) - { - // If this is the second 30ms of a 60ms frame reset this such that in the next call - // encoder starts fresh. - ISACenc_obj->frame_nb = 0; - } - return status; - } - } - - /* split signal in two bands */ - WebRtcIsacfix_SplitAndFilter1(ISACenc_obj->data_buffer_fix, LP16a, HP16a, &ISACenc_obj->prefiltbankstr_obj ); - - /* estimate pitch parameters and pitch-filter lookahead signal */ - WebRtcIsacfix_PitchAnalysis(LP16a+QLOOKAHEAD, LPandHP, - &ISACenc_obj->pitchanalysisstr_obj, PitchLags_Q7, PitchGains_Q12); /* LPandHP = LP_lookahead_pfQ0, */ - - /* Set where to store data in multiple packets memory */ - if (ISACenc_obj->SaveEnc_ptr != NULL) { - if (frame_mode == 0 || ISACenc_obj->frame_nb == 0) - { - (ISACenc_obj->SaveEnc_ptr)->startIdx = 0; - } - else - { - (ISACenc_obj->SaveEnc_ptr)->startIdx = 1; - } - } - - /* quantize & encode pitch parameters */ - status = WebRtcIsacfix_EncodePitchGain(PitchGains_Q12, &ISACenc_obj->bitstr_obj, ISACenc_obj->SaveEnc_ptr); - if (status < 0) - { - if (frame_mode == 1 && ISACenc_obj->frame_nb == 1) - { - // If this is the second 30ms of a 60ms frame reset this such that in the next call - // encoder starts fresh. - ISACenc_obj->frame_nb = 0; - } - return status; - } - status = WebRtcIsacfix_EncodePitchLag(PitchLags_Q7 , PitchGains_Q12, &ISACenc_obj->bitstr_obj, ISACenc_obj->SaveEnc_ptr); - if (status < 0) - { - if (frame_mode == 1 && ISACenc_obj->frame_nb == 1) - { - // If this is the second 30ms of a 60ms frame reset this such that in the next call - // encoder starts fresh. - ISACenc_obj->frame_nb = 0; - } - return status; - } - AvgPitchGain_Q12 = (PitchGains_Q12[0] + PitchGains_Q12[1] + - PitchGains_Q12[2] + PitchGains_Q12[3]) >> 2; - - /* find coefficients for perceptual pre-filters */ - WebRtcIsacfix_GetLpcCoef(LPandHP, HP16a+QLOOKAHEAD, &ISACenc_obj->maskfiltstr_obj, - ISACenc_obj->s2nr, PitchGains_Q12, - gain_lo_hiQ17, lofilt_coefQ15, hifilt_coefQ15); /*LPandHP = LP_lookahead_pfQ0*/ - - // record LPC Gains for possible bit-rate reduction - for(k = 0; k < KLT_ORDER_GAIN; k++) - { - transcodingParam.lpcGains[k] = gain_lo_hiQ17[k]; - } - - /* code LPC model and shape - gains not quantized yet */ - status = WebRtcIsacfix_EncodeLpc(gain_lo_hiQ17, lofilt_coefQ15, hifilt_coefQ15, - &bmodel, &bits_gainsQ11, &ISACenc_obj->bitstr_obj, ISACenc_obj->SaveEnc_ptr, &transcodingParam); - if (status < 0) - { - if (frame_mode == 1 && ISACenc_obj->frame_nb == 1) - { - // If this is the second 30ms of a 60ms frame reset this such that in the next call - // encoder starts fresh. - ISACenc_obj->frame_nb = 0; - } - return status; - } - arithLenBeforeEncodingDFT = (ISACenc_obj->bitstr_obj.stream_index << 1) + (1-ISACenc_obj->bitstr_obj.full); - - /* low-band filtering */ - WebRtcIsacfix_NormLatticeFilterMa(ORDERLO, ISACenc_obj->maskfiltstr_obj.PreStateLoGQ15, - LP16a, lofilt_coefQ15, gain_lo_hiQ17, 0, LPandHP);/* LPandHP = LP16b */ - - /* pitch filter */ - WebRtcIsacfix_PitchFilter(LPandHP, LP16a, &ISACenc_obj->pitchfiltstr_obj, PitchLags_Q7, PitchGains_Q12, 1);/* LPandHP = LP16b */ - - /* high-band filtering */ - WebRtcIsacfix_NormLatticeFilterMa(ORDERHI, ISACenc_obj->maskfiltstr_obj.PreStateHiGQ15, - HP16a, hifilt_coefQ15, gain_lo_hiQ17, 1, LPandHP);/*LPandHP = HP16b*/ - - /* transform */ - WebRtcIsacfix_Time2Spec(LP16a, LPandHP, LP16a, LPandHP); /*LPandHP = HP16b*/ - - /* Save data for multiple packets memory */ - if (ISACenc_obj->SaveEnc_ptr != NULL) { - for (k = 0; k < FRAMESAMPLES_HALF; k++) { - (ISACenc_obj->SaveEnc_ptr)->fre[k + (ISACenc_obj->SaveEnc_ptr)->startIdx*FRAMESAMPLES_HALF] = LP16a[k]; - (ISACenc_obj->SaveEnc_ptr)->fim[k + (ISACenc_obj->SaveEnc_ptr)->startIdx*FRAMESAMPLES_HALF] = LPandHP[k]; - } - (ISACenc_obj->SaveEnc_ptr)->AvgPitchGain[(ISACenc_obj->SaveEnc_ptr)->startIdx] = AvgPitchGain_Q12; - } - - /* quantization and lossless coding */ - status = WebRtcIsacfix_EncodeSpec(LP16a, LPandHP, &ISACenc_obj->bitstr_obj, AvgPitchGain_Q12); - if((status <= -1) && (status != -ISAC_DISALLOWED_BITSTREAM_LENGTH)) /*LPandHP = HP16b*/ - { - if (frame_mode == 1 && ISACenc_obj->frame_nb == 1) - { - // If this is the second 30ms of a 60ms frame reset this such that in the next call - // encoder starts fresh. - ISACenc_obj->frame_nb = 0; - } - return status; - } - - if((frame_mode == 1) && (ISACenc_obj->frame_nb == 0)) - { - // it is a 60ms and we are in the first 30ms - // then the limit at this point should be half of the assigned value - payloadLimitBytes = ISACenc_obj->payloadLimitBytes60 >> 1; - } - else if (frame_mode == 0) - { - // it is a 30ms frame - payloadLimitBytes = (ISACenc_obj->payloadLimitBytes30) - 3; - } - else - { - // this is the second half of a 60ms frame. - payloadLimitBytes = ISACenc_obj->payloadLimitBytes60 - 3; // subract 3 because termination process may add 3 bytes - } - - iterCntr = 0; - while((((ISACenc_obj->bitstr_obj.stream_index) << 1) > payloadLimitBytes) || - (status == -ISAC_DISALLOWED_BITSTREAM_LENGTH)) - { - int16_t arithLenDFTByte; - int16_t bytesLeftQ5; - int16_t ratioQ5[8] = {0, 6, 9, 12, 16, 19, 22, 25}; - - // According to experiments on TIMIT the following is proper for audio, but it is not agressive enough for tonal inputs - // such as DTMF, sweep-sine, ... - // - // (0.55 - (0.8 - ratio[i]/32) * 5 / 6) * 2^14 - // int16_t scaleQ14[8] = {0, 648, 1928, 3208, 4915, 6195, 7475, 8755}; - - - // This is a supper-agressive scaling passed the tests (tonal inputs) tone with one iteration for payload limit - // of 120 (32kbps bottleneck), number of frames needed a rate-reduction was 58403 - // - int16_t scaleQ14[8] = {0, 348, 828, 1408, 2015, 3195, 3500, 3500}; - int16_t idx; - - if(iterCntr >= MAX_PAYLOAD_LIMIT_ITERATION) - { - // We were not able to limit the payload size - - if((frame_mode == 1) && (ISACenc_obj->frame_nb == 0)) - { - // This was the first 30ms of a 60ms frame. Although the payload is larger than it - // should be but we let the second 30ms be encoded. Maybe togetehr we won't exceed - // the limit. - ISACenc_obj->frame_nb = 1; - return 0; - } - else if((frame_mode == 1) && (ISACenc_obj->frame_nb == 1)) - { - ISACenc_obj->frame_nb = 0; - } - - if(status != -ISAC_DISALLOWED_BITSTREAM_LENGTH) - { - return -ISAC_PAYLOAD_LARGER_THAN_LIMIT; - } - else - { - return status; - } - } - if(status != -ISAC_DISALLOWED_BITSTREAM_LENGTH) - { - arithLenDFTByte = (ISACenc_obj->bitstr_obj.stream_index << 1) + (1-ISACenc_obj->bitstr_obj.full) - arithLenBeforeEncodingDFT; - bytesLeftQ5 = (payloadLimitBytes - arithLenBeforeEncodingDFT) << 5; - - // bytesLeft / arithLenDFTBytes indicates how much scaling is required a rough estimate (agressive) - // scale = 0.55 - (0.8 - bytesLeft / arithLenDFTBytes) * 5 / 6 - // bytesLeft / arithLenDFTBytes below 0.2 will have a scale of zero and above 0.8 are treated as 0.8 - // to avoid division we do more simplification. - // - // values of (bytesLeft / arithLenDFTBytes)*32 between ratioQ5[i] and ratioQ5[i+1] are rounded to ratioQ5[i] - // and the corresponding scale is chosen - - // we compare bytesLeftQ5 with ratioQ5[]*arithLenDFTByte; - idx = 4; - idx += (bytesLeftQ5 >= ratioQ5[idx] * arithLenDFTByte) ? 2 : -2; - idx += (bytesLeftQ5 >= ratioQ5[idx] * arithLenDFTByte) ? 1 : -1; - idx += (bytesLeftQ5 >= ratioQ5[idx] * arithLenDFTByte) ? 0 : -1; - } - else - { - // we are here because the bit-stream did not fit into the buffer, in this case, the stream_index is not - // trustable, especially if the is the first 30ms of a packet. Thereforem, we will go for the most agressive - // case. - idx = 0; - } - // scale FFT coefficients to reduce the bit-rate - for(k = 0; k < FRAMESAMPLES_HALF; k++) - { - LP16a[k] = (int16_t)(LP16a[k] * scaleQ14[idx] >> 14); - LPandHP[k] = (int16_t)(LPandHP[k] * scaleQ14[idx] >> 14); - } - - // Save data for multiple packets memory - if (ISACenc_obj->SaveEnc_ptr != NULL) - { - for(k = 0; k < FRAMESAMPLES_HALF; k++) - { - (ISACenc_obj->SaveEnc_ptr)->fre[k + (ISACenc_obj->SaveEnc_ptr)->startIdx*FRAMESAMPLES_HALF] = LP16a[k]; - (ISACenc_obj->SaveEnc_ptr)->fim[k + (ISACenc_obj->SaveEnc_ptr)->startIdx*FRAMESAMPLES_HALF] = LPandHP[k]; - } - } - - // scale the unquantized LPC gains and save the scaled version for the future use - for(k = 0; k < KLT_ORDER_GAIN; k++) - { - gain_lo_hiQ17[k] = WEBRTC_SPL_MUL_16_32_RSFT14(scaleQ14[idx], transcodingParam.lpcGains[k]);//transcodingParam.lpcGains[k]; // - transcodingParam.lpcGains[k] = gain_lo_hiQ17[k]; - } - - // reset the bit-stream object to the state which it had before encoding LPC Gains - ISACenc_obj->bitstr_obj.full = transcodingParam.full; - ISACenc_obj->bitstr_obj.stream_index = transcodingParam.stream_index; - ISACenc_obj->bitstr_obj.streamval = transcodingParam.streamval; - ISACenc_obj->bitstr_obj.W_upper = transcodingParam.W_upper; - ISACenc_obj->bitstr_obj.stream[transcodingParam.stream_index-1] = transcodingParam.beforeLastWord; - ISACenc_obj->bitstr_obj.stream[transcodingParam.stream_index] = transcodingParam.lastWord; - - - // quantize and encode LPC gain - WebRtcIsacfix_EstCodeLpcGain(gain_lo_hiQ17, &ISACenc_obj->bitstr_obj, ISACenc_obj->SaveEnc_ptr); - arithLenBeforeEncodingDFT = (ISACenc_obj->bitstr_obj.stream_index << 1) + (1-ISACenc_obj->bitstr_obj.full); - status = WebRtcIsacfix_EncodeSpec(LP16a, LPandHP, &ISACenc_obj->bitstr_obj, AvgPitchGain_Q12); - if((status <= -1) && (status != -ISAC_DISALLOWED_BITSTREAM_LENGTH)) /*LPandHP = HP16b*/ - { - if (frame_mode == 1 && ISACenc_obj->frame_nb == 1) - { - // If this is the second 30ms of a 60ms frame reset this such that in the next call - // encoder starts fresh. - ISACenc_obj->frame_nb = 0; - } - return status; - } - iterCntr++; - } - - if (frame_mode == 1 && ISACenc_obj->frame_nb == 0) - /* i.e. 60 ms framesize and just processed the first 30ms, */ - /* go back to main function to buffer the other 30ms speech frame */ - { - ISACenc_obj->frame_nb = 1; - return 0; - } - else if (frame_mode == 1 && ISACenc_obj->frame_nb == 1) - { - ISACenc_obj->frame_nb = 0; - /* also update the framelength for next packet, in Adaptive mode only */ - if (CodingMode == 0 && (ISACenc_obj->enforceFrameSize == 0)) { - ISACenc_obj->new_framelength = WebRtcIsacfix_GetNewFrameLength(ISACenc_obj->BottleNeck, - ISACenc_obj->current_framesamples); - } - } - - - /* complete arithmetic coding */ - stream_length = WebRtcIsacfix_EncTerminate(&ISACenc_obj->bitstr_obj); - /* can this be negative? */ - - if(CodingMode == 0) - { - - /* update rate model and get minimum number of bytes in this packet */ - MinBytes = WebRtcIsacfix_GetMinBytes(&ISACenc_obj->rate_data_obj, (int16_t) stream_length, - ISACenc_obj->current_framesamples, ISACenc_obj->BottleNeck, ISACenc_obj->MaxDelay); - - /* if bitstream is too short, add garbage at the end */ - - /* Store length of coded data */ - usefulstr_len = stream_length; - - /* Make sure MinBytes does not exceed packet size limit */ - if ((ISACenc_obj->frame_nb == 0) && (MinBytes > ISACenc_obj->payloadLimitBytes30)) { - MinBytes = ISACenc_obj->payloadLimitBytes30; - } else if ((ISACenc_obj->frame_nb == 1) && (MinBytes > ISACenc_obj->payloadLimitBytes60)) { - MinBytes = ISACenc_obj->payloadLimitBytes60; - } - - /* Make sure we don't allow more than 255 bytes of garbage data. - We store the length of the garbage data in 8 bits in the bitstream, - 255 is the max garbage lenght we can signal using 8 bits. */ - if( MinBytes > usefulstr_len + 255 ) { - MinBytes = usefulstr_len + 255; - } - - /* Save data for creation of multiple bitstreams */ - if (ISACenc_obj->SaveEnc_ptr != NULL) { - (ISACenc_obj->SaveEnc_ptr)->minBytes = MinBytes; - } - - while (stream_length < MinBytes) - { - RTC_DCHECK_GE(stream_length, 0); - if (stream_length & 0x0001){ - ISACenc_obj->bitstr_seed = WEBRTC_SPL_RAND( ISACenc_obj->bitstr_seed ); - ISACenc_obj->bitstr_obj.stream[stream_length / 2] |= - (uint16_t)(ISACenc_obj->bitstr_seed & 0xFF); - } else { - ISACenc_obj->bitstr_seed = WEBRTC_SPL_RAND( ISACenc_obj->bitstr_seed ); - ISACenc_obj->bitstr_obj.stream[stream_length / 2] = - ((uint16_t)ISACenc_obj->bitstr_seed << 8); - } - stream_length++; - } - - /* to get the real stream_length, without garbage */ - if (usefulstr_len & 0x0001) { - ISACenc_obj->bitstr_obj.stream[usefulstr_len>>1] &= 0xFF00; - ISACenc_obj->bitstr_obj.stream[usefulstr_len>>1] += (MinBytes - usefulstr_len) & 0x00FF; - } - else { - ISACenc_obj->bitstr_obj.stream[usefulstr_len>>1] &= 0x00FF; - ISACenc_obj->bitstr_obj.stream[usefulstr_len >> 1] += - ((uint16_t)((MinBytes - usefulstr_len) & 0x00FF) << 8); - } - } - else - { - /* update rate model */ - WebRtcIsacfix_UpdateRateModel(&ISACenc_obj->rate_data_obj, (int16_t) stream_length, - ISACenc_obj->current_framesamples, ISACenc_obj->BottleNeck); - } - return stream_length; -} - -/* This function is used to create a new bitstream with new BWE. - The same data as previously encoded with the fucntion WebRtcIsacfix_EncodeImpl() - is used. The data needed is taken from the struct, where it was stored - when calling the encoder. */ -int WebRtcIsacfix_EncodeStoredData(IsacFixEncoderInstance *ISACenc_obj, - int BWnumber, - float scale) -{ - int ii; - int status; - int16_t BWno = (int16_t)BWnumber; - int stream_length = 0; - - int16_t model; - const uint16_t *Q_PitchGain_cdf_ptr[1]; - const uint16_t **cdf; - const IsacSaveEncoderData *SaveEnc_str; - int32_t tmpLPCcoeffs_g[KLT_ORDER_GAIN<<1]; - int16_t tmpLPCindex_g[KLT_ORDER_GAIN<<1]; - int16_t tmp_fre[FRAMESAMPLES]; - int16_t tmp_fim[FRAMESAMPLES]; - - SaveEnc_str = ISACenc_obj->SaveEnc_ptr; - - /* Check if SaveEnc memory exists */ - if (SaveEnc_str == NULL) { - return (-1); - } - - /* Sanity Check - possible values for BWnumber is 0 - 23 */ - if ((BWnumber < 0) || (BWnumber > 23)) { - return -ISAC_RANGE_ERROR_BW_ESTIMATOR; - } - - /* reset bitstream */ - ISACenc_obj->bitstr_obj.W_upper = 0xFFFFFFFF; - ISACenc_obj->bitstr_obj.streamval = 0; - ISACenc_obj->bitstr_obj.stream_index = 0; - ISACenc_obj->bitstr_obj.full = 1; - - /* encode frame length */ - status = WebRtcIsacfix_EncodeFrameLen(SaveEnc_str->framelength, &ISACenc_obj->bitstr_obj); - if (status < 0) { - /* Wrong frame size */ - return status; - } - - /* encode bandwidth estimate */ - status = WebRtcIsacfix_EncodeReceiveBandwidth(&BWno, &ISACenc_obj->bitstr_obj); - if (status < 0) { - return status; - } - - /* Transcoding */ - /* If scale < 1, rescale data to produce lower bitrate signal */ - if ((0.0 < scale) && (scale < 1.0)) { - /* Compensate LPC gain */ - for (ii = 0; ii < (KLT_ORDER_GAIN*(1+SaveEnc_str->startIdx)); ii++) { - tmpLPCcoeffs_g[ii] = (int32_t) ((scale) * (float) SaveEnc_str->LPCcoeffs_g[ii]); - } - - /* Scale DFT */ - for (ii = 0; ii < (FRAMESAMPLES_HALF*(1+SaveEnc_str->startIdx)); ii++) { - tmp_fre[ii] = (int16_t) ((scale) * (float) SaveEnc_str->fre[ii]) ; - tmp_fim[ii] = (int16_t) ((scale) * (float) SaveEnc_str->fim[ii]) ; - } - } else { - for (ii = 0; ii < (KLT_ORDER_GAIN*(1+SaveEnc_str->startIdx)); ii++) { - tmpLPCindex_g[ii] = SaveEnc_str->LPCindex_g[ii]; - } - - for (ii = 0; ii < (FRAMESAMPLES_HALF*(1+SaveEnc_str->startIdx)); ii++) { - tmp_fre[ii] = SaveEnc_str->fre[ii]; - tmp_fim[ii] = SaveEnc_str->fim[ii]; - } - } - - /* Loop over number of 30 msec */ - for (ii = 0; ii <= SaveEnc_str->startIdx; ii++) - { - - /* encode pitch gains */ - *Q_PitchGain_cdf_ptr = WebRtcIsacfix_kPitchGainCdf; - status = WebRtcIsacfix_EncHistMulti(&ISACenc_obj->bitstr_obj, &SaveEnc_str->pitchGain_index[ii], - Q_PitchGain_cdf_ptr, 1); - if (status < 0) { - return status; - } - - /* entropy coding of quantization pitch lags */ - /* voicing classificiation */ - if (SaveEnc_str->meanGain[ii] <= 819) { - cdf = WebRtcIsacfix_kPitchLagPtrLo; - } else if (SaveEnc_str->meanGain[ii] <= 1638) { - cdf = WebRtcIsacfix_kPitchLagPtrMid; - } else { - cdf = WebRtcIsacfix_kPitchLagPtrHi; - } - status = WebRtcIsacfix_EncHistMulti(&ISACenc_obj->bitstr_obj, - &SaveEnc_str->pitchIndex[PITCH_SUBFRAMES*ii], cdf, PITCH_SUBFRAMES); - if (status < 0) { - return status; - } - - /* LPC */ - /* entropy coding of model number */ - model = 0; - status = WebRtcIsacfix_EncHistMulti(&ISACenc_obj->bitstr_obj, &model, - WebRtcIsacfix_kModelCdfPtr, 1); - if (status < 0) { - return status; - } - - /* entropy coding of quantization indices - LPC shape only */ - status = WebRtcIsacfix_EncHistMulti(&ISACenc_obj->bitstr_obj, &SaveEnc_str->LPCindex_s[KLT_ORDER_SHAPE*ii], - WebRtcIsacfix_kCdfShapePtr[0], KLT_ORDER_SHAPE); - if (status < 0) { - return status; - } - - /* If transcoding, get new LPC gain indices */ - if (scale < 1.0) { - WebRtcIsacfix_TranscodeLpcCoef(&tmpLPCcoeffs_g[KLT_ORDER_GAIN*ii], &tmpLPCindex_g[KLT_ORDER_GAIN*ii]); - } - - /* entropy coding of quantization indices - LPC gain */ - status = WebRtcIsacfix_EncHistMulti(&ISACenc_obj->bitstr_obj, &tmpLPCindex_g[KLT_ORDER_GAIN*ii], - WebRtcIsacfix_kCdfGainPtr[0], KLT_ORDER_GAIN); - if (status < 0) { - return status; - } - - /* quantization and lossless coding */ - status = WebRtcIsacfix_EncodeSpec(&tmp_fre[ii*FRAMESAMPLES_HALF], &tmp_fim[ii*FRAMESAMPLES_HALF], - &ISACenc_obj->bitstr_obj, SaveEnc_str->AvgPitchGain[ii]); - if (status < 0) { - return status; - } - } - - /* complete arithmetic coding */ - stream_length = WebRtcIsacfix_EncTerminate(&ISACenc_obj->bitstr_obj); - - return stream_length; -} diff --git a/modules/audio_coding/codecs/isac/fix/source/entropy_coding.c b/modules/audio_coding/codecs/isac/fix/source/entropy_coding.c deleted file mode 100644 index 842e77f47e..0000000000 --- a/modules/audio_coding/codecs/isac/fix/source/entropy_coding.c +++ /dev/null @@ -1,2056 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/* - * entropy_coding.c - * - * This file contains all functions used to arithmetically - * encode the iSAC bistream. - * - */ - -#include - -#include "common_audio/signal_processing/include/signal_processing_library.h" -#include "modules/audio_coding/codecs/isac/fix/source/arith_routins.h" -#include "modules/audio_coding/codecs/isac/fix/source/entropy_coding.h" -#include "modules/audio_coding/codecs/isac/fix/source/lpc_tables.h" -#include "modules/audio_coding/codecs/isac/fix/source/pitch_gain_tables.h" -#include "modules/audio_coding/codecs/isac/fix/source/pitch_lag_tables.h" -#include "modules/audio_coding/codecs/isac/fix/source/settings.h" -#include "modules/audio_coding/codecs/isac/fix/source/spectrum_ar_model_tables.h" -#include "rtc_base/sanitizer.h" - -/* - * Eenumerations for arguments to functions WebRtcIsacfix_MatrixProduct1() - * and WebRtcIsacfix_MatrixProduct2(). -*/ - -enum matrix_index_factor { - kTIndexFactor1 = 1, - kTIndexFactor2 = 2, - kTIndexFactor3 = SUBFRAMES, - kTIndexFactor4 = LPC_SHAPE_ORDER -}; - -enum matrix_index_step { - kTIndexStep1 = 1, - kTIndexStep2 = SUBFRAMES, - kTIndexStep3 = LPC_SHAPE_ORDER -}; - -enum matrixprod_loop_count { - kTLoopCount1 = SUBFRAMES, - kTLoopCount2 = 2, - kTLoopCount3 = LPC_SHAPE_ORDER -}; - -enum matrix1_shift_value { - kTMatrix1_shift0 = 0, - kTMatrix1_shift1 = 1, - kTMatrix1_shift5 = 5 -}; - -enum matrixprod_init_case { - kTInitCase0 = 0, - kTInitCase1 = 1 -}; - -/* - This function implements the fix-point correspondant function to lrint. - - FLP: (int32_t)floor(flt+.499999999999) - FIP: (fixVal+roundVal)>>qDomain - - where roundVal = 2^(qDomain-1) = 1<<(qDomain-1) - -*/ -static __inline int32_t CalcLrIntQ(int32_t fixVal, int16_t qDomain) { - return (fixVal + (1 << (qDomain - 1))) >> qDomain; -} - -/* - __inline uint32_t stepwise(int32_t dinQ10) { - - int32_t ind, diQ10, dtQ10; - - diQ10 = dinQ10; - if (diQ10 < DPMIN_Q10) - diQ10 = DPMIN_Q10; - if (diQ10 >= DPMAX_Q10) - diQ10 = DPMAX_Q10 - 1; - - dtQ10 = diQ10 - DPMIN_Q10;*/ /* Q10 + Q10 = Q10 */ -/* ind = (dtQ10 * 5) >> 10; */ /* 2^10 / 5 = 0.2 in Q10 */ -/* Q10 -> Q0 */ - -/* return rpointsFIX_Q10[ind]; - - } -*/ - -/* logN(x) = logN(2)*log2(x) = 0.6931*log2(x). Output in Q8. */ -/* The input argument X to logN(X) is 2^17 times higher than the - input floating point argument Y to log(Y), since the X value - is a Q17 value. This can be compensated for after the call, by - subraction a value Z for each Q-step. One Q-step means that - X gets 2 thimes higher, i.e. Z = logN(2)*256 = 0.693147180559*256 = - 177.445678 should be subtracted (since logN() returns a Q8 value). - For a X value in Q17, the value 177.445678*17 = 3017 should be - subtracted */ -static int16_t CalcLogN(int32_t arg) { - int16_t zeros, log2, frac, logN; - - zeros=WebRtcSpl_NormU32(arg); - frac = (int16_t)((uint32_t)((arg << zeros) & 0x7FFFFFFF) >> 23); - log2 = (int16_t)(((31 - zeros) << 8) + frac); // log2(x) in Q8 - logN = (int16_t)(log2 * 22713 >> 15); // log(2) = 0.693147 = 22713 in Q15 - logN=logN+11; //Scalar compensation which minimizes the (log(x)-logN(x))^2 error over all x. - - return logN; -} - - -/* - expN(x) = 2^(a*x), where a = log2(e) ~= 1.442695 - - Input: Q8 (int16_t) - Output: Q17 (int32_t) - - a = log2(e) = log2(exp(1)) ~= 1.442695 ==> a = 23637 in Q14 (1.442688) - To this value, 700 is added or subtracted in order to get an average error - nearer zero, instead of always same-sign. -*/ - -static int32_t CalcExpN(int16_t x) { - int16_t axINT, axFRAC; - int16_t exp16; - int32_t exp; - int16_t ax = (int16_t)(x * 23637 >> 14); // Q8 - - if (x>=0) { - axINT = ax >> 8; //Q0 - axFRAC = ax&0x00FF; - exp16 = 1 << axINT; // Q0 - axFRAC = axFRAC+256; //Q8 - exp = exp16 * axFRAC; // Q0*Q8 = Q8 - exp <<= 9; // Q17 - } else { - ax = -ax; - axINT = 1 + (ax >> 8); //Q0 - axFRAC = 0x00FF - (ax&0x00FF); - exp16 = (int16_t)(32768 >> axINT); // Q15 - axFRAC = axFRAC+256; //Q8 - exp = exp16 * axFRAC; // Q15*Q8 = Q23 - exp >>= 6; // Q17 - } - - return exp; -} - - -/* compute correlation from power spectrum */ -static void CalcCorrelation(int32_t *PSpecQ12, int32_t *CorrQ7) -{ - int32_t summ[FRAMESAMPLES/8]; - int32_t diff[FRAMESAMPLES/8]; - int32_t sum; - int k, n; - - for (k = 0; k < FRAMESAMPLES/8; k++) { - summ[k] = (PSpecQ12[k] + PSpecQ12[FRAMESAMPLES / 4 - 1 - k] + 16) >> 5; - diff[k] = (PSpecQ12[k] - PSpecQ12[FRAMESAMPLES / 4 - 1 - k] + 16) >> 5; - } - - sum = 2; - for (n = 0; n < FRAMESAMPLES/8; n++) - sum += summ[n]; - CorrQ7[0] = sum; - - for (k = 0; k < AR_ORDER; k += 2) { - sum = 0; - for (n = 0; n < FRAMESAMPLES/8; n++) - sum += (WebRtcIsacfix_kCos[k][n] * diff[n] + 256) >> 9; - CorrQ7[k+1] = sum; - } - - for (k=1; k> 9; - CorrQ7[k+1] = sum; - } -} - -// Some arithmetic operations that are allowed to overflow. (It's still -// undefined behavior, so not a good idea; this just makes UBSan ignore the -// violations, so that our old code can continue to do what it's always been -// doing.) -static inline int32_t RTC_NO_SANITIZE("signed-integer-overflow") - OverflowingMulS16S32ToS32(int16_t a, int32_t b) { - return a * b; -} -static inline int32_t RTC_NO_SANITIZE("signed-integer-overflow") - OverflowingAddS32S32ToS32(int32_t a, int32_t b) { - return a + b; -} -static inline int32_t RTC_NO_SANITIZE("signed-integer-overflow") - OverflowingSubS32S32ToS32(int32_t a, int32_t b) { - return a - b; -} - -/* compute inverse AR power spectrum */ -static void CalcInvArSpec(const int16_t *ARCoefQ12, - const int32_t gainQ10, - int32_t *CurveQ16) -{ - int32_t CorrQ11[AR_ORDER+1]; - int32_t sum, tmpGain; - int32_t diffQ16[FRAMESAMPLES/8]; - const int16_t *CS_ptrQ9; - int k, n; - int16_t round, shftVal = 0, sh; - - sum = 0; - for (n = 0; n < AR_ORDER+1; n++) - sum += WEBRTC_SPL_MUL(ARCoefQ12[n], ARCoefQ12[n]); /* Q24 */ - sum = ((sum >> 6) * 65 + 32768) >> 16; /* Result in Q8. */ - CorrQ11[0] = (sum * gainQ10 + 256) >> 9; - - /* To avoid overflow, we shift down gainQ10 if it is large. We will not lose any precision */ - if(gainQ10>400000){ - tmpGain = gainQ10 >> 3; - round = 32; - shftVal = 6; - } else { - tmpGain = gainQ10; - round = 256; - shftVal = 9; - } - - for (k = 1; k < AR_ORDER+1; k++) { - sum = 16384; - for (n = k; n < AR_ORDER+1; n++) - sum += WEBRTC_SPL_MUL(ARCoefQ12[n-k], ARCoefQ12[n]); /* Q24 */ - sum >>= 15; - CorrQ11[k] = (sum * tmpGain + round) >> shftVal; - } - sum = CorrQ11[0] << 7; - for (n = 0; n < FRAMESAMPLES/8; n++) - CurveQ16[n] = sum; - - for (k = 1; k < AR_ORDER; k += 2) { - for (n = 0; n < FRAMESAMPLES/8; n++) - CurveQ16[n] += - (OverflowingMulS16S32ToS32(WebRtcIsacfix_kCos[k][n], CorrQ11[k + 1]) + - 2) >> - 2; - } - - CS_ptrQ9 = WebRtcIsacfix_kCos[0]; - - /* If CorrQ11[1] too large we avoid getting overflow in the calculation by shifting */ - sh=WebRtcSpl_NormW32(CorrQ11[1]); - if (CorrQ11[1]==0) /* Use next correlation */ - sh=WebRtcSpl_NormW32(CorrQ11[2]); - - if (sh<9) - shftVal = 9 - sh; - else - shftVal = 0; - - for (n = 0; n < FRAMESAMPLES/8; n++) - diffQ16[n] = (CS_ptrQ9[n] * (CorrQ11[1] >> shftVal) + 2) >> 2; - for (k = 2; k < AR_ORDER; k += 2) { - CS_ptrQ9 = WebRtcIsacfix_kCos[k]; - for (n = 0; n < FRAMESAMPLES/8; n++) - diffQ16[n] += (CS_ptrQ9[n] * (CorrQ11[k + 1] >> shftVal) + 2) >> 2; - } - - for (k=0; k> 6) * 65 + 32768) >> 16; /* Result in Q8. */ - CorrQ11[0] = (sum * gainQ10 + 256) >> 9; - - /* To avoid overflow, we shift down gainQ10 if it is large. We will not lose any precision */ - if(gainQ10>400000){ - tmpGain = gainQ10 >> 3; - round = 32; - shftVal = 6; - } else { - tmpGain = gainQ10; - round = 256; - shftVal = 9; - } - - for (k = 1; k < AR_ORDER+1; k++) { - sum = 16384; - for (n = k; n < AR_ORDER+1; n++) - sum += WEBRTC_SPL_MUL(ARCoefQ12[n-k], ARCoefQ12[n]); /* Q24 */ - sum >>= 15; - CorrQ11[k] = (sum * tmpGain + round) >> shftVal; - } - sum = CorrQ11[0] << 7; - for (n = 0; n < FRAMESAMPLES/8; n++) - summQ16[n] = sum; - - for (k = 1; k < (AR_ORDER); k += 2) { - for (n = 0; n < FRAMESAMPLES/8; n++) - summQ16[n] += ((CorrQ11[k + 1] * WebRtcIsacfix_kCos[k][n]) + 2) >> 2; - } - - CS_ptrQ9 = WebRtcIsacfix_kCos[0]; - - /* If CorrQ11[1] too large we avoid getting overflow in the calculation by shifting */ - sh=WebRtcSpl_NormW32(CorrQ11[1]); - if (CorrQ11[1]==0) /* Use next correlation */ - sh=WebRtcSpl_NormW32(CorrQ11[2]); - - if (sh<9) - shftVal = 9 - sh; - else - shftVal = 0; - - for (n = 0; n < FRAMESAMPLES/8; n++) - diffQ16[n] = (CS_ptrQ9[n] * (CorrQ11[1] >> shftVal) + 2) >> 2; - for (k = 2; k < AR_ORDER; k += 2) { - CS_ptrQ9 = WebRtcIsacfix_kCos[k]; - for (n = 0; n < FRAMESAMPLES/8; n++) - diffQ16[n] += (CS_ptrQ9[n] * (CorrQ11[k + 1] >> shftVal) + 2) >> 2; - } - - in_sqrt = summQ16[0] + (diffQ16[0] << shftVal); - - /* convert to magnitude spectrum, by doing square-roots (modified from SPLIB) */ - res = 1 << (WebRtcSpl_GetSizeInBits(in_sqrt) >> 1); - - for (k = 0; k < FRAMESAMPLES/8; k++) - { - in_sqrt = summQ16[k] + (diffQ16[k] << shftVal); - i = 10; - - /* make in_sqrt positive to prohibit sqrt of negative values */ - if(in_sqrt<0) - in_sqrt=-in_sqrt; - - newRes = (in_sqrt / res + res) >> 1; - do - { - res = newRes; - newRes = (in_sqrt / res + res) >> 1; - } while (newRes != res && i-- > 0); - - CurveQ8[k] = (int16_t)newRes; - } - for (k = FRAMESAMPLES/8; k < FRAMESAMPLES/4; k++) { - - in_sqrt = summQ16[FRAMESAMPLES / 4 - 1 - k] - - (diffQ16[FRAMESAMPLES / 4 - 1 - k] << shftVal); - i = 10; - - /* make in_sqrt positive to prohibit sqrt of negative values */ - if(in_sqrt<0) - in_sqrt=-in_sqrt; - - newRes = (in_sqrt / res + res) >> 1; - do - { - res = newRes; - newRes = (in_sqrt / res + res) >> 1; - } while (newRes != res && i-- > 0); - - CurveQ8[k] = (int16_t)newRes; - } - -} - - - -/* generate array of dither samples in Q7 */ -static void GenerateDitherQ7(int16_t *bufQ7, - uint32_t seed, - int16_t length, - int16_t AvgPitchGain_Q12) -{ - int k; - int16_t dither1_Q7, dither2_Q7, dither_gain_Q14, shft; - - if (AvgPitchGain_Q12 < 614) /* this threshold should be equal to that in decode_spec() */ - { - for (k = 0; k < length-2; k += 3) - { - /* new random unsigned int32_t */ - seed = WEBRTC_SPL_UMUL(seed, 196314165) + 907633515; - - /* fixed-point dither sample between -64 and 64 (Q7) */ - dither1_Q7 = (int16_t)(((int32_t)(seed + 16777216)) >> 25); - - /* new random unsigned int32_t */ - seed = WEBRTC_SPL_UMUL(seed, 196314165) + 907633515; - - /* fixed-point dither sample between -64 and 64 */ - dither2_Q7 = (int16_t)(((int32_t)(seed + 16777216)) >> 25); - - shft = (int16_t)(WEBRTC_SPL_RSHIFT_U32(seed, 25) & 15); - if (shft < 5) - { - bufQ7[k] = dither1_Q7; - bufQ7[k+1] = dither2_Q7; - bufQ7[k+2] = 0; - } - else if (shft < 10) - { - bufQ7[k] = dither1_Q7; - bufQ7[k+1] = 0; - bufQ7[k+2] = dither2_Q7; - } - else - { - bufQ7[k] = 0; - bufQ7[k+1] = dither1_Q7; - bufQ7[k+2] = dither2_Q7; - } - } - } - else - { - dither_gain_Q14 = (int16_t)(22528 - WEBRTC_SPL_MUL(10, AvgPitchGain_Q12)); - - /* dither on half of the coefficients */ - for (k = 0; k < length-1; k += 2) - { - /* new random unsigned int32_t */ - seed = WEBRTC_SPL_UMUL(seed, 196314165) + 907633515; - - /* fixed-point dither sample between -64 and 64 */ - dither1_Q7 = (int16_t)(((int32_t)(seed + 16777216)) >> 25); - - /* dither sample is placed in either even or odd index */ - shft = (int16_t)(WEBRTC_SPL_RSHIFT_U32(seed, 25) & 1); /* either 0 or 1 */ - - bufQ7[k + shft] = (int16_t)((dither_gain_Q14 * dither1_Q7 + 8192) >> 14); - bufQ7[k + 1 - shft] = 0; - } - } -} - - - - -/* - * function to decode the complex spectrum from the bitstream - * returns the total number of bytes in the stream - */ -int WebRtcIsacfix_DecodeSpec(Bitstr_dec *streamdata, - int16_t *frQ7, - int16_t *fiQ7, - int16_t AvgPitchGain_Q12) -{ - int16_t data[FRAMESAMPLES]; - int32_t invARSpec2_Q16[FRAMESAMPLES/4]; - int16_t ARCoefQ12[AR_ORDER+1]; - int16_t RCQ15[AR_ORDER]; - int16_t gainQ10; - int32_t gain2_Q10; - int len; - int k; - - /* create dither signal */ - GenerateDitherQ7(data, streamdata->W_upper, FRAMESAMPLES, AvgPitchGain_Q12); /* Dither is output in vector 'Data' */ - - /* decode model parameters */ - if (WebRtcIsacfix_DecodeRcCoef(streamdata, RCQ15) < 0) - return -ISAC_RANGE_ERROR_DECODE_SPECTRUM; - - - WebRtcSpl_ReflCoefToLpc(RCQ15, AR_ORDER, ARCoefQ12); - - if (WebRtcIsacfix_DecodeGain2(streamdata, &gain2_Q10) < 0) - return -ISAC_RANGE_ERROR_DECODE_SPECTRUM; - - /* compute inverse AR power spectrum */ - CalcInvArSpec(ARCoefQ12, gain2_Q10, invARSpec2_Q16); - - /* arithmetic decoding of spectrum */ - /* 'data' input and output. Input = Dither */ - len = WebRtcIsacfix_DecLogisticMulti2(data, streamdata, invARSpec2_Q16, (int16_t)FRAMESAMPLES); - - if (len<1) - return -ISAC_RANGE_ERROR_DECODE_SPECTRUM; - - /* subtract dither and scale down spectral samples with low SNR */ - if (AvgPitchGain_Q12 <= 614) - { - for (k = 0; k < FRAMESAMPLES; k += 4) - { - gainQ10 = WebRtcSpl_DivW32W16ResW16(30 << 10, - (int16_t)((uint32_t)(invARSpec2_Q16[k >> 2] + 2195456) >> 16)); - *frQ7++ = (int16_t)((data[k] * gainQ10 + 512) >> 10); - *fiQ7++ = (int16_t)((data[k + 1] * gainQ10 + 512) >> 10); - *frQ7++ = (int16_t)((data[k + 2] * gainQ10 + 512) >> 10); - *fiQ7++ = (int16_t)((data[k + 3] * gainQ10 + 512) >> 10); - } - } - else - { - for (k = 0; k < FRAMESAMPLES; k += 4) - { - gainQ10 = WebRtcSpl_DivW32W16ResW16(36 << 10, - (int16_t)((uint32_t)(invARSpec2_Q16[k >> 2] + 2654208) >> 16)); - *frQ7++ = (int16_t)((data[k] * gainQ10 + 512) >> 10); - *fiQ7++ = (int16_t)((data[k + 1] * gainQ10 + 512) >> 10); - *frQ7++ = (int16_t)((data[k + 2] * gainQ10 + 512) >> 10); - *fiQ7++ = (int16_t)((data[k + 3] * gainQ10 + 512) >> 10); - } - } - - return len; -} - - -int WebRtcIsacfix_EncodeSpec(const int16_t *fr, - const int16_t *fi, - Bitstr_enc *streamdata, - int16_t AvgPitchGain_Q12) -{ - int16_t dataQ7[FRAMESAMPLES]; - int32_t PSpec[FRAMESAMPLES/4]; - uint16_t invARSpecQ8[FRAMESAMPLES/4]; - int32_t CorrQ7[AR_ORDER+1]; - int32_t CorrQ7_norm[AR_ORDER+1]; - int16_t RCQ15[AR_ORDER]; - int16_t ARCoefQ12[AR_ORDER+1]; - int32_t gain2_Q10; - int16_t val; - int32_t nrg; - uint32_t sum; - int16_t lft_shft; - int16_t status; - int k, n, j; - - - /* create dither_float signal */ - GenerateDitherQ7(dataQ7, streamdata->W_upper, FRAMESAMPLES, AvgPitchGain_Q12); - - /* add dither and quantize, and compute power spectrum */ - /* Vector dataQ7 contains Dither in Q7 */ - for (k = 0; k < FRAMESAMPLES; k += 4) - { - val = ((*fr++ + dataQ7[k] + 64) & 0xFF80) - dataQ7[k]; /* Data = Dither */ - dataQ7[k] = val; /* New value in Data */ - sum = WEBRTC_SPL_UMUL(val, val); - - val = ((*fi++ + dataQ7[k+1] + 64) & 0xFF80) - dataQ7[k+1]; /* Data = Dither */ - dataQ7[k+1] = val; /* New value in Data */ - sum += WEBRTC_SPL_UMUL(val, val); - - val = ((*fr++ + dataQ7[k+2] + 64) & 0xFF80) - dataQ7[k+2]; /* Data = Dither */ - dataQ7[k+2] = val; /* New value in Data */ - sum += WEBRTC_SPL_UMUL(val, val); - - val = ((*fi++ + dataQ7[k+3] + 64) & 0xFF80) - dataQ7[k+3]; /* Data = Dither */ - dataQ7[k+3] = val; /* New value in Data */ - sum += WEBRTC_SPL_UMUL(val, val); - - PSpec[k>>2] = WEBRTC_SPL_RSHIFT_U32(sum, 2); - } - - /* compute correlation from power spectrum */ - CalcCorrelation(PSpec, CorrQ7); - - - /* find AR coefficients */ - /* number of bit shifts to 14-bit normalize CorrQ7[0] (leaving room for sign) */ - lft_shft = WebRtcSpl_NormW32(CorrQ7[0]) - 18; - - if (lft_shft > 0) { - for (k=0; k> -lft_shft; - } - - /* find RC coefficients */ - WebRtcSpl_AutoCorrToReflCoef(CorrQ7_norm, AR_ORDER, RCQ15); - - /* quantize & code RC Coef */ - status = WebRtcIsacfix_EncodeRcCoef(RCQ15, streamdata); - if (status < 0) { - return status; - } - - /* RC -> AR coefficients */ - WebRtcSpl_ReflCoefToLpc(RCQ15, AR_ORDER, ARCoefQ12); - - /* compute ARCoef' * Corr * ARCoef in Q19 */ - nrg = 0; - for (j = 0; j <= AR_ORDER; j++) { - for (n = 0; n <= j; n++) - nrg += (ARCoefQ12[j] * ((CorrQ7_norm[j - n] * ARCoefQ12[n] + 256) >> 9) + - 4) >> 3; - for (n = j+1; n <= AR_ORDER; n++) - nrg += (ARCoefQ12[j] * ((CorrQ7_norm[n - j] * ARCoefQ12[n] + 256) >> 9) + - 4) >> 3; - } - - if (lft_shft > 0) - nrg >>= lft_shft; - else - nrg <<= -lft_shft; - - if(nrg>131072) - gain2_Q10 = WebRtcSpl_DivResultInQ31(FRAMESAMPLES >> 2, nrg); /* also shifts 31 bits to the left! */ - else - gain2_Q10 = FRAMESAMPLES >> 2; - - /* quantize & code gain2_Q10 */ - if (WebRtcIsacfix_EncodeGain2(&gain2_Q10, streamdata)) - return -1; - - /* compute inverse AR magnitude spectrum */ - CalcRootInvArSpec(ARCoefQ12, gain2_Q10, invARSpecQ8); - - - /* arithmetic coding of spectrum */ - status = WebRtcIsacfix_EncLogisticMulti2(streamdata, dataQ7, invARSpecQ8, (int16_t)FRAMESAMPLES); - if ( status ) - return( status ); - - return 0; -} - - -/* Matlab's LAR definition */ -static void Rc2LarFix(const int16_t *rcQ15, int32_t *larQ17, int16_t order) { - - /* - - This is a piece-wise implemenetation of a rc2lar-function (all values in the comment - are Q15 values and are based on [0 24956/32768 30000/32768 32500/32768], i.e. - [0.76159667968750 0.91552734375000 0.99182128906250] - - x0 x1 a k x0(again) b - ================================================================================== - 0.00 0.76: 0 2.625997508581 0 0 - 0.76 0.91: 2.000012018559 7.284502668663 0.761596679688 -3.547841027073 - 0.91 0.99: 3.121320351712 31.115835041229 0.915527343750 -25.366077452148 - 0.99 1.00: 5.495270168700 686.663805654056 0.991821289063 -675.552510708011 - - The implementation is y(x)= a + (x-x0)*k, but this can be simplified to - - y(x) = a-x0*k + x*k = b + x*k, where b = a-x0*k - - akx=[0 2.625997508581 0 - 2.000012018559 7.284502668663 0.761596679688 - 3.121320351712 31.115835041229 0.915527343750 - 5.495270168700 686.663805654056 0.991821289063]; - - b = akx(:,1) - akx(:,3).*akx(:,2) - - [ 0.0 - -3.547841027073 - -25.366077452148 - -675.552510708011] - - */ - - int k; - int16_t rc; - int32_t larAbsQ17; - - for (k = 0; k < order; k++) { - - rc = WEBRTC_SPL_ABS_W16(rcQ15[k]); //Q15 - - /* Calculate larAbsQ17 in Q17 from rc in Q15 */ - - if (rc<24956) { //0.7615966 in Q15 - // (Q15*Q13)>>11 = Q17 - larAbsQ17 = rc * 21512 >> 11; - } else if (rc<30000) { //0.91552734375 in Q15 - // Q17 + (Q15*Q12)>>10 = Q17 - larAbsQ17 = -465024 + (rc * 29837 >> 10); - } else if (rc<32500) { //0.99182128906250 in Q15 - // Q17 + (Q15*Q10)>>8 = Q17 - larAbsQ17 = -3324784 + (rc * 31863 >> 8); - } else { - // Q17 + (Q15*Q5)>>3 = Q17 - larAbsQ17 = -88546020 + (rc * 21973 >> 3); - } - - if (rcQ15[k]>0) { - larQ17[k] = larAbsQ17; - } else { - larQ17[k] = -larAbsQ17; - } - } -} - - -static void Lar2RcFix(const int32_t *larQ17, int16_t *rcQ15, int16_t order) { - - /* - This is a piece-wise implemenetation of a lar2rc-function - See comment in Rc2LarFix() about details. - */ - - int k; - int16_t larAbsQ11; - int32_t rc; - - for (k = 0; k < order; k++) { - - larAbsQ11 = (int16_t)WEBRTC_SPL_ABS_W32((larQ17[k] + 32) >> 6); // Q11 - - if (larAbsQ11<4097) { //2.000012018559 in Q11 - // Q11*Q16>>12 = Q15 - rc = larAbsQ11 * 24957 >> 12; - } else if (larAbsQ11<6393) { //3.121320351712 in Q11 - // (Q11*Q17 + Q13)>>13 = Q15 - rc = (larAbsQ11 * 17993 + 130738688) >> 13; - } else if (larAbsQ11<11255) { //5.495270168700 in Q11 - // (Q11*Q19 + Q30)>>15 = Q15 - rc = (larAbsQ11 * 16850 + 875329820) >> 15; - } else { - // (Q11*Q24>>16 + Q19)>>4 = Q15 - rc = (((larAbsQ11 * 24433) >> 16) + 515804) >> 4; - } - - if (larQ17[k]<=0) { - rc = -rc; - } - - rcQ15[k] = (int16_t) rc; // Q15 - } -} - -static void Poly2LarFix(int16_t *lowbandQ15, - int16_t orderLo, - int16_t *hibandQ15, - int16_t orderHi, - int16_t Nsub, - int32_t *larsQ17) { - - int k, n; - int32_t *outpQ17; - int16_t orderTot; - int32_t larQ17[MAX_ORDER]; // Size 7+6 is enough - - orderTot = (orderLo + orderHi); - outpQ17 = larsQ17; - for (k = 0; k < Nsub; k++) { - - Rc2LarFix(lowbandQ15, larQ17, orderLo); - - for (n = 0; n < orderLo; n++) - outpQ17[n] = larQ17[n]; //Q17 - - Rc2LarFix(hibandQ15, larQ17, orderHi); - - for (n = 0; n < orderHi; n++) - outpQ17[n + orderLo] = larQ17[n]; //Q17; - - outpQ17 += orderTot; - lowbandQ15 += orderLo; - hibandQ15 += orderHi; - } -} - - -static void Lar2polyFix(int32_t *larsQ17, - int16_t *lowbandQ15, - int16_t orderLo, - int16_t *hibandQ15, - int16_t orderHi, - int16_t Nsub) { - - int k, n; - int16_t orderTot; - int16_t *outplQ15, *outphQ15; - int32_t *inpQ17; - int16_t rcQ15[7+6]; - - orderTot = (orderLo + orderHi); - outplQ15 = lowbandQ15; - outphQ15 = hibandQ15; - inpQ17 = larsQ17; - for (k = 0; k < Nsub; k++) { - - /* gains not handled here as in the FLP version */ - - /* Low band */ - Lar2RcFix(&inpQ17[0], rcQ15, orderLo); - for (n = 0; n < orderLo; n++) - outplQ15[n] = rcQ15[n]; // Refl. coeffs - - /* High band */ - Lar2RcFix(&inpQ17[orderLo], rcQ15, orderHi); - for (n = 0; n < orderHi; n++) - outphQ15[n] = rcQ15[n]; // Refl. coeffs - - inpQ17 += orderTot; - outplQ15 += orderLo; - outphQ15 += orderHi; - } -} - -/* -Function WebRtcIsacfix_MatrixProduct1C() does one form of matrix multiplication. -It first shifts input data of one matrix, determines the right indexes for the -two matrixes, multiply them, and write the results into an output buffer. - -Note that two factors (or, multipliers) determine the initialization values of -the variable `matrix1_index` in the code. The relationship is -`matrix1_index` = `matrix1_index_factor1` * `matrix1_index_factor2`, where -`matrix1_index_factor1` is given by the argument while `matrix1_index_factor2` -is determined by the value of argument `matrix1_index_init_case`; -`matrix1_index_factor2` is the value of the outmost loop counter j (when -`matrix1_index_init_case` is 0), or the value of the middle loop counter k (when -`matrix1_index_init_case` is non-zero). - -`matrix0_index` is determined the same way. - -Arguments: - matrix0[]: matrix0 data in Q15 domain. - matrix1[]: matrix1 data. - matrix_product[]: output data (matrix product). - matrix1_index_factor1: The first of two factors determining the - initialization value of matrix1_index. - matrix0_index_factor1: The first of two factors determining the - initialization value of matrix0_index. - matrix1_index_init_case: Case number for selecting the second of two - factors determining the initialization value - of matrix1_index and matrix0_index. - matrix1_index_step: Incremental step for matrix1_index. - matrix0_index_step: Incremental step for matrix0_index. - inner_loop_count: Maximum count of the inner loop. - mid_loop_count: Maximum count of the intermediate loop. - shift: Left shift value for matrix1. -*/ -void WebRtcIsacfix_MatrixProduct1C(const int16_t matrix0[], - const int32_t matrix1[], - int32_t matrix_product[], - const int matrix1_index_factor1, - const int matrix0_index_factor1, - const int matrix1_index_init_case, - const int matrix1_index_step, - const int matrix0_index_step, - const int inner_loop_count, - const int mid_loop_count, - const int shift) { - int j = 0, k = 0, n = 0; - int matrix0_index = 0, matrix1_index = 0, matrix_prod_index = 0; - int* matrix0_index_factor2 = &k; - int* matrix1_index_factor2 = &j; - if (matrix1_index_init_case != 0) { - matrix0_index_factor2 = &j; - matrix1_index_factor2 = &k; - } - - for (j = 0; j < SUBFRAMES; j++) { - matrix_prod_index = mid_loop_count * j; - for (k = 0; k < mid_loop_count; k++) { - int32_t sum32 = 0; - matrix0_index = matrix0_index_factor1 * (*matrix0_index_factor2); - matrix1_index = matrix1_index_factor1 * (*matrix1_index_factor2); - for (n = 0; n < inner_loop_count; n++) { - sum32 += WEBRTC_SPL_MUL_16_32_RSFT16( - matrix0[matrix0_index], matrix1[matrix1_index] * (1 << shift)); - matrix0_index += matrix0_index_step; - matrix1_index += matrix1_index_step; - } - matrix_product[matrix_prod_index] = sum32; - matrix_prod_index++; - } - } -} - -/* -Function WebRtcIsacfix_MatrixProduct2C() returns the product of two matrixes, -one of which has two columns. It first has to determine the correct index of -the first matrix before doing the actual element multiplication. - -Arguments: - matrix0[]: A matrix in Q15 domain. - matrix1[]: A matrix in Q21 domain. - matrix_product[]: Output data in Q17 domain. - matrix0_index_factor: A factor determining the initialization value - of matrix0_index. - matrix0_index_step: Incremental step for matrix0_index. -*/ -void WebRtcIsacfix_MatrixProduct2C(const int16_t matrix0[], - const int32_t matrix1[], - int32_t matrix_product[], - const int matrix0_index_factor, - const int matrix0_index_step) { - int j = 0, n = 0; - int matrix1_index = 0, matrix0_index = 0, matrix_prod_index = 0; - for (j = 0; j < SUBFRAMES; j++) { - int32_t sum32 = 0, sum32_2 = 0; - matrix1_index = 0; - matrix0_index = matrix0_index_factor * j; - for (n = SUBFRAMES; n > 0; n--) { - sum32 += (WEBRTC_SPL_MUL_16_32_RSFT16(matrix0[matrix0_index], - matrix1[matrix1_index])); - sum32_2 += (WEBRTC_SPL_MUL_16_32_RSFT16(matrix0[matrix0_index], - matrix1[matrix1_index + 1])); - matrix1_index += 2; - matrix0_index += matrix0_index_step; - } - matrix_product[matrix_prod_index] = sum32 >> 3; - matrix_product[matrix_prod_index + 1] = sum32_2 >> 3; - matrix_prod_index += 2; - } -} - -int WebRtcIsacfix_DecodeLpc(int32_t *gain_lo_hiQ17, - int16_t *LPCCoef_loQ15, - int16_t *LPCCoef_hiQ15, - Bitstr_dec *streamdata, - int16_t *outmodel) { - - int32_t larsQ17[KLT_ORDER_SHAPE]; // KLT_ORDER_GAIN+KLT_ORDER_SHAPE == (ORDERLO+ORDERHI)*SUBFRAMES - int err; - - err = WebRtcIsacfix_DecodeLpcCoef(streamdata, larsQ17, gain_lo_hiQ17, outmodel); - if (err<0) // error check - return -ISAC_RANGE_ERROR_DECODE_LPC; - - Lar2polyFix(larsQ17, LPCCoef_loQ15, ORDERLO, LPCCoef_hiQ15, ORDERHI, SUBFRAMES); - - return 0; -} - -/* decode & dequantize LPC Coef */ -int WebRtcIsacfix_DecodeLpcCoef(Bitstr_dec *streamdata, - int32_t *LPCCoefQ17, - int32_t *gain_lo_hiQ17, - int16_t *outmodel) -{ - int j, k, n; - int err; - int16_t pos, pos2, posg, poss; - int16_t gainpos; - int16_t model; - int16_t index_QQ[KLT_ORDER_SHAPE]; - int32_t tmpcoeffs_gQ17[KLT_ORDER_GAIN]; - int32_t tmpcoeffs2_gQ21[KLT_ORDER_GAIN]; - int16_t tmpcoeffs_sQ10[KLT_ORDER_SHAPE]; - int32_t tmpcoeffs_sQ17[KLT_ORDER_SHAPE]; - int32_t tmpcoeffs2_sQ18[KLT_ORDER_SHAPE]; - int32_t sumQQ; - int16_t sumQQ16; - int32_t tmp32; - - - - /* entropy decoding of model number */ - err = WebRtcIsacfix_DecHistOneStepMulti(&model, streamdata, WebRtcIsacfix_kModelCdfPtr, WebRtcIsacfix_kModelInitIndex, 1); - if (err<0) // error check - return err; - - /* entropy decoding of quantization indices */ - err = WebRtcIsacfix_DecHistOneStepMulti(index_QQ, streamdata, WebRtcIsacfix_kCdfShapePtr[model], WebRtcIsacfix_kInitIndexShape[model], KLT_ORDER_SHAPE); - if (err<0) // error check - return err; - /* find quantization levels for coefficients */ - for (k=0; k> 7; // (Q10*Q15)>>7 = Q18 - pos++; - pos2++; - } - tmpcoeffs2_sQ18[poss] = sumQQ; //Q18 - poss++; - } - } - - /* right transform */ // Transpose matrix - WebRtcIsacfix_MatrixProduct2(WebRtcIsacfix_kT2GainQ15[0], tmpcoeffs2_gQ21, - tmpcoeffs_gQ17, kTIndexFactor1, kTIndexStep2); - WebRtcIsacfix_MatrixProduct1(WebRtcIsacfix_kT2ShapeQ15[model], - tmpcoeffs2_sQ18, tmpcoeffs_sQ17, kTIndexFactor1, kTIndexFactor1, - kTInitCase1, kTIndexStep3, kTIndexStep2, kTLoopCount1, kTLoopCount3, - kTMatrix1_shift0); - - /* scaling, mean addition, and gain restoration */ - gainpos = 0; - posg = 0;poss = 0;pos=0; - for (k=0; k> 11); - sumQQ16 += WebRtcIsacfix_kMeansGainQ8[model][posg]; - sumQQ = CalcExpN(sumQQ16); // Q8 in and Q17 out - gain_lo_hiQ17[gainpos] = sumQQ; //Q17 - gainpos++; - posg++; - - // Divide by 4 and get Q17 to Q8, i.e. shift 2+9. - sumQQ16 = (int16_t)(tmpcoeffs_gQ17[posg] >> 11); - sumQQ16 += WebRtcIsacfix_kMeansGainQ8[model][posg]; - sumQQ = CalcExpN(sumQQ16); // Q8 in and Q17 out - gain_lo_hiQ17[gainpos] = sumQQ; //Q17 - gainpos++; - posg++; - - /* lo band LAR coeffs */ - for (n=0; n>16 = Q17, with 1/2.1 = 0.47619047619 ~= 31208 in Q16 - tmp32 = tmp32 + WebRtcIsacfix_kMeansShapeQ17[model][poss]; // Q17+Q17 = Q17 - LPCCoefQ17[pos] = tmp32; - } - - /* hi band LAR coeffs */ - for (n=0; n>16)<<3 = Q17, with 1/0.45 = 2.222222222222 ~= 18204 in Q13 - tmp32 = - WEBRTC_SPL_MUL_16_32_RSFT16(18204, tmpcoeffs_sQ17[poss]) * (1 << 3); - tmp32 = tmp32 + WebRtcIsacfix_kMeansShapeQ17[model][poss]; // Q17+Q17 = Q17 - LPCCoefQ17[pos] = tmp32; - } - } - - - *outmodel=model; - - return 0; -} - -/* estimate codel length of LPC Coef */ -static int EstCodeLpcCoef(int32_t *LPCCoefQ17, - int32_t *gain_lo_hiQ17, - int16_t *model, - int32_t *sizeQ11, - Bitstr_enc *streamdata, - IsacSaveEncoderData* encData, - transcode_obj *transcodingParam) { - int j, k, n; - int16_t posQQ, pos2QQ, gainpos; - int16_t pos, poss, posg, offsg; - int16_t index_gQQ[KLT_ORDER_GAIN], index_sQQ[KLT_ORDER_SHAPE]; - int16_t index_ovr_gQQ[KLT_ORDER_GAIN], index_ovr_sQQ[KLT_ORDER_SHAPE]; - int32_t BitsQQ; - - int16_t tmpcoeffs_gQ6[KLT_ORDER_GAIN]; - int32_t tmpcoeffs_gQ17[KLT_ORDER_GAIN]; - int32_t tmpcoeffs_sQ17[KLT_ORDER_SHAPE]; - int32_t tmpcoeffs2_gQ21[KLT_ORDER_GAIN]; - int32_t tmpcoeffs2_sQ17[KLT_ORDER_SHAPE]; - int32_t sumQQ; - int32_t tmp32; - int16_t sumQQ16; - int status = 0; - - /* write LAR coefficients to statistics file */ - /* Save data for creation of multiple bitstreams (and transcoding) */ - if (encData != NULL) { - for (k=0; kLPCcoeffs_g[KLT_ORDER_GAIN*encData->startIdx + k] = gain_lo_hiQ17[k]; - } - } - - /* log gains, mean removal and scaling */ - posg = 0;poss = 0;pos=0; gainpos=0; - - for (k=0; k WebRtcIsacfix_kMaxIndGain[k]) { - index_gQQ[k] = WebRtcIsacfix_kMaxIndGain[k]; - } - index_ovr_gQQ[k] = WebRtcIsacfix_kOffsetGain[0][k]+index_gQQ[k]; - posQQ = WebRtcIsacfix_kOfLevelsGain[0] + index_ovr_gQQ[k]; - - /* Save data for creation of multiple bitstreams */ - if (encData != NULL) { - encData->LPCindex_g[KLT_ORDER_GAIN*encData->startIdx + k] = index_gQQ[k]; - } - - /* determine number of bits */ - sumQQ = WebRtcIsacfix_kCodeLenGainQ11[posQQ]; //Q11 - BitsQQ += sumQQ; - } - - for (k=0; k WebRtcIsacfix_kMaxIndShape[k]) - index_sQQ[k] = WebRtcIsacfix_kMaxIndShape[k]; - index_ovr_sQQ[k] = WebRtcIsacfix_kOffsetShape[0][k]+index_sQQ[k]; - - posQQ = WebRtcIsacfix_kOfLevelsShape[0] + index_ovr_sQQ[k]; - sumQQ = WebRtcIsacfix_kCodeLenShapeQ11[posQQ]; //Q11 - BitsQQ += sumQQ; - } - - - - *model = 0; - *sizeQ11=BitsQQ; - - /* entropy coding of model number */ - status = WebRtcIsacfix_EncHistMulti(streamdata, model, WebRtcIsacfix_kModelCdfPtr, 1); - if (status < 0) { - return status; - } - - /* entropy coding of quantization indices - shape only */ - status = WebRtcIsacfix_EncHistMulti(streamdata, index_sQQ, WebRtcIsacfix_kCdfShapePtr[0], KLT_ORDER_SHAPE); - if (status < 0) { - return status; - } - - /* Save data for creation of multiple bitstreams */ - if (encData != NULL) { - for (k=0; kLPCindex_s[KLT_ORDER_SHAPE*encData->startIdx + k] = index_sQQ[k]; - } - } - /* save the state of the bitstream object 'streamdata' for the possible bit-rate reduction */ - transcodingParam->full = streamdata->full; - transcodingParam->stream_index = streamdata->stream_index; - transcodingParam->streamval = streamdata->streamval; - transcodingParam->W_upper = streamdata->W_upper; - transcodingParam->beforeLastWord = streamdata->stream[streamdata->stream_index-1]; - transcodingParam->lastWord = streamdata->stream[streamdata->stream_index]; - - /* entropy coding of index */ - status = WebRtcIsacfix_EncHistMulti(streamdata, index_gQQ, WebRtcIsacfix_kCdfGainPtr[0], KLT_ORDER_GAIN); - if (status < 0) { - return status; - } - - /* find quantization levels for shape coefficients */ - for (k=0; k>16 = Q17, with 1/2.1 = 0.47619047619 ~= 31208 in Q16 - tmp32 = tmp32 + WebRtcIsacfix_kMeansShapeQ17[0][poss]; // Q17+Q17 = Q17 - LPCCoefQ17[pos] = tmp32; - } - - /* hi band LAR coeffs */ - for (n=0; n>16)<<3 = Q17, with 1/0.45 = 2.222222222222 ~= 18204 in Q13 - tmp32 = WEBRTC_SPL_MUL_16_32_RSFT16(18204, tmpcoeffs_sQ17[poss]) << 3; - tmp32 = tmp32 + WebRtcIsacfix_kMeansShapeQ17[0][poss]; // Q17+Q17 = Q17 - LPCCoefQ17[pos] = tmp32; - } - - } - - //to update tmpcoeffs_gQ17 to the proper state - for (k=0; k> (16 - 1) = Q17; Q17 << 4 = Q21. - sumQQ = (WEBRTC_SPL_MUL_16_32_RSFT16(WebRtcIsacfix_kT1GainQ15[0][0], - tmpcoeffs_gQ17[offsg]) << 1); - sumQQ += (WEBRTC_SPL_MUL_16_32_RSFT16(WebRtcIsacfix_kT1GainQ15[0][1], - tmpcoeffs_gQ17[offsg + 1]) << 1); - tmpcoeffs2_gQ21[posg] = sumQQ << 4; - posg++; - - sumQQ = (WEBRTC_SPL_MUL_16_32_RSFT16(WebRtcIsacfix_kT1GainQ15[0][2], - tmpcoeffs_gQ17[offsg]) << 1); - sumQQ += (WEBRTC_SPL_MUL_16_32_RSFT16(WebRtcIsacfix_kT1GainQ15[0][3], - tmpcoeffs_gQ17[offsg + 1]) << 1); - tmpcoeffs2_gQ21[posg] = sumQQ << 4; - posg++; - offsg += 2; - } - - /* right transform */ // Transpose matrix - WebRtcIsacfix_MatrixProduct2(WebRtcIsacfix_kT2GainQ15[0], tmpcoeffs2_gQ21, - tmpcoeffs_gQ17, kTIndexFactor1, kTIndexStep2); - - /* scaling, mean addition, and gain restoration */ - posg = 0; - gainpos = 0; - for (k=0; k<2*SUBFRAMES; k++) { - - // Divide by 4 and get Q17 to Q8, i.e. shift 2+9. - sumQQ16 = (int16_t)(tmpcoeffs_gQ17[posg] >> 11); - sumQQ16 += WebRtcIsacfix_kMeansGainQ8[0][posg]; - sumQQ = CalcExpN(sumQQ16); // Q8 in and Q17 out - gain_lo_hiQ17[gainpos] = sumQQ; //Q17 - - gainpos++; - pos++;posg++; - } - - return 0; -} - -int WebRtcIsacfix_EstCodeLpcGain(int32_t *gain_lo_hiQ17, - Bitstr_enc *streamdata, - IsacSaveEncoderData* encData) { - int j, k; - int16_t posQQ, pos2QQ, gainpos; - int16_t posg; - int16_t index_gQQ[KLT_ORDER_GAIN]; - - int16_t tmpcoeffs_gQ6[KLT_ORDER_GAIN]; - int32_t tmpcoeffs_gQ17[KLT_ORDER_GAIN]; - int32_t tmpcoeffs2_gQ21[KLT_ORDER_GAIN]; - int32_t sumQQ; - int status = 0; - - /* write LAR coefficients to statistics file */ - /* Save data for creation of multiple bitstreams (and transcoding) */ - if (encData != NULL) { - for (k=0; kLPCcoeffs_g[KLT_ORDER_GAIN*encData->startIdx + k] = gain_lo_hiQ17[k]; - } - } - - /* log gains, mean removal and scaling */ - posg = 0; gainpos = 0; - - for (k=0; k WebRtcIsacfix_kMaxIndGain[k]) { - index_gQQ[k] = WebRtcIsacfix_kMaxIndGain[k]; - } - - /* Save data for creation of multiple bitstreams */ - if (encData != NULL) { - encData->LPCindex_g[KLT_ORDER_GAIN*encData->startIdx + k] = index_gQQ[k]; - } - } - - /* entropy coding of index */ - status = WebRtcIsacfix_EncHistMulti(streamdata, index_gQQ, WebRtcIsacfix_kCdfGainPtr[0], KLT_ORDER_GAIN); - if (status < 0) { - return status; - } - - return 0; -} - - -int WebRtcIsacfix_EncodeLpc(int32_t *gain_lo_hiQ17, - int16_t *LPCCoef_loQ15, - int16_t *LPCCoef_hiQ15, - int16_t *model, - int32_t *sizeQ11, - Bitstr_enc *streamdata, - IsacSaveEncoderData* encData, - transcode_obj *transcodeParam) -{ - int status = 0; - int32_t larsQ17[KLT_ORDER_SHAPE]; // KLT_ORDER_SHAPE == (ORDERLO+ORDERHI)*SUBFRAMES - // = (6+12)*6 == 108 - - Poly2LarFix(LPCCoef_loQ15, ORDERLO, LPCCoef_hiQ15, ORDERHI, SUBFRAMES, larsQ17); - - status = EstCodeLpcCoef(larsQ17, gain_lo_hiQ17, model, sizeQ11, - streamdata, encData, transcodeParam); - if (status < 0) { - return (status); - } - - Lar2polyFix(larsQ17, LPCCoef_loQ15, ORDERLO, LPCCoef_hiQ15, ORDERHI, SUBFRAMES); - - return 0; -} - - -/* decode & dequantize RC */ -int WebRtcIsacfix_DecodeRcCoef(Bitstr_dec *streamdata, int16_t *RCQ15) -{ - int k, err; - int16_t index[AR_ORDER]; - - /* entropy decoding of quantization indices */ - err = WebRtcIsacfix_DecHistOneStepMulti(index, streamdata, WebRtcIsacfix_kRcCdfPtr, WebRtcIsacfix_kRcInitInd, AR_ORDER); - if (err<0) // error check - return err; - - /* find quantization levels for reflection coefficients */ - for (k=0; k WebRtcIsacfix_kRcBound[index[k]]) - { - while (RCQ15[k] > WebRtcIsacfix_kRcBound[index[k] + 1]) - index[k]++; - } - else - { - while (RCQ15[k] < WebRtcIsacfix_kRcBound[--index[k]]) ; - } - - RCQ15[k] = *(WebRtcIsacfix_kRcLevPtr[k] + index[k]); - } - - - /* entropy coding of quantization indices */ - status = WebRtcIsacfix_EncHistMulti(streamdata, index, WebRtcIsacfix_kRcCdfPtr, AR_ORDER); - - /* If error in WebRtcIsacfix_EncHistMulti(), status will be negative, otherwise 0 */ - return status; -} - - -/* decode & dequantize squared Gain */ -int WebRtcIsacfix_DecodeGain2(Bitstr_dec *streamdata, int32_t *gainQ10) -{ - int err; - int16_t index; - - /* entropy decoding of quantization index */ - err = WebRtcIsacfix_DecHistOneStepMulti( - &index, - streamdata, - WebRtcIsacfix_kGainPtr, - WebRtcIsacfix_kGainInitInd, - 1); - /* error check */ - if (err<0) { - return err; - } - - /* find quantization level */ - *gainQ10 = WebRtcIsacfix_kGain2Lev[index]; - - return 0; -} - - - -/* quantize & code squared Gain */ -int WebRtcIsacfix_EncodeGain2(int32_t *gainQ10, Bitstr_enc *streamdata) -{ - int16_t index; - int status = 0; - - /* find quantization index */ - index = WebRtcIsacfix_kGainInitInd[0]; - if (*gainQ10 > WebRtcIsacfix_kGain2Bound[index]) - { - while (*gainQ10 > WebRtcIsacfix_kGain2Bound[index + 1]) - index++; - } - else - { - while (*gainQ10 < WebRtcIsacfix_kGain2Bound[--index]) ; - } - - /* dequantize */ - *gainQ10 = WebRtcIsacfix_kGain2Lev[index]; - - /* entropy coding of quantization index */ - status = WebRtcIsacfix_EncHistMulti(streamdata, &index, WebRtcIsacfix_kGainPtr, 1); - - /* If error in WebRtcIsacfix_EncHistMulti(), status will be negative, otherwise 0 */ - return status; -} - - -/* code and decode Pitch Gains and Lags functions */ - -/* decode & dequantize Pitch Gains */ -int WebRtcIsacfix_DecodePitchGain(Bitstr_dec *streamdata, int16_t *PitchGains_Q12) -{ - int err; - int16_t index_comb; - const uint16_t *pitch_gain_cdf_ptr[1]; - - /* entropy decoding of quantization indices */ - *pitch_gain_cdf_ptr = WebRtcIsacfix_kPitchGainCdf; - err = WebRtcIsacfix_DecHistBisectMulti(&index_comb, streamdata, pitch_gain_cdf_ptr, WebRtcIsacfix_kCdfTableSizeGain, 1); - /* error check, Q_mean_Gain.. tables are of size 144 */ - if ((err < 0) || (index_comb < 0) || (index_comb >= 144)) - return -ISAC_RANGE_ERROR_DECODE_PITCH_GAIN; - - /* unquantize back to pitch gains by table look-up */ - PitchGains_Q12[0] = WebRtcIsacfix_kPitchGain1[index_comb]; - PitchGains_Q12[1] = WebRtcIsacfix_kPitchGain2[index_comb]; - PitchGains_Q12[2] = WebRtcIsacfix_kPitchGain3[index_comb]; - PitchGains_Q12[3] = WebRtcIsacfix_kPitchGain4[index_comb]; - - return 0; -} - - -/* quantize & code Pitch Gains */ -int WebRtcIsacfix_EncodePitchGain(int16_t* PitchGains_Q12, - Bitstr_enc* streamdata, - IsacSaveEncoderData* encData) { - int k,j; - int16_t SQ15[PITCH_SUBFRAMES]; - int16_t index[3]; - int16_t index_comb; - const uint16_t *pitch_gain_cdf_ptr[1]; - int32_t CQ17; - int status = 0; - - - /* get the approximate arcsine (almost linear)*/ - for (k=0; k> 2); // Q15 - - - /* find quantization index; only for the first three transform coefficients */ - for (k=0; k<3; k++) - { - /* transform */ - CQ17=0; - for (j=0; j> 10; // Q17 - } - - index[k] = (int16_t)((CQ17 + 8192)>>14); // Rounding and scaling with stepsize (=1/0.125=8) - - /* check that the index is not outside the boundaries of the table */ - if (index[k] < WebRtcIsacfix_kLowerlimiGain[k]) index[k] = WebRtcIsacfix_kLowerlimiGain[k]; - else if (index[k] > WebRtcIsacfix_kUpperlimitGain[k]) index[k] = WebRtcIsacfix_kUpperlimitGain[k]; - index[k] -= WebRtcIsacfix_kLowerlimiGain[k]; - } - - /* calculate unique overall index */ - index_comb = (int16_t)(WEBRTC_SPL_MUL(WebRtcIsacfix_kMultsGain[0], index[0]) + - WEBRTC_SPL_MUL(WebRtcIsacfix_kMultsGain[1], index[1]) + index[2]); - - /* unquantize back to pitch gains by table look-up */ - // (Y) - PitchGains_Q12[0] = WebRtcIsacfix_kPitchGain1[index_comb]; - PitchGains_Q12[1] = WebRtcIsacfix_kPitchGain2[index_comb]; - PitchGains_Q12[2] = WebRtcIsacfix_kPitchGain3[index_comb]; - PitchGains_Q12[3] = WebRtcIsacfix_kPitchGain4[index_comb]; - - - /* entropy coding of quantization pitch gains */ - *pitch_gain_cdf_ptr = WebRtcIsacfix_kPitchGainCdf; - status = WebRtcIsacfix_EncHistMulti(streamdata, &index_comb, pitch_gain_cdf_ptr, 1); - if (status < 0) { - return status; - } - - /* Save data for creation of multiple bitstreams */ - if (encData != NULL) { - encData->pitchGain_index[encData->startIdx] = index_comb; - } - - return 0; -} - - - -/* Pitch LAG */ - - -/* decode & dequantize Pitch Lags */ -int WebRtcIsacfix_DecodePitchLag(Bitstr_dec *streamdata, - int16_t *PitchGain_Q12, - int16_t *PitchLags_Q7) -{ - int k, err; - int16_t index[PITCH_SUBFRAMES]; - const int16_t *mean_val2Q10, *mean_val4Q10; - - const int16_t *lower_limit; - const uint16_t *init_index; - const uint16_t *cdf_size; - const uint16_t **cdf; - - int32_t meangainQ12; - int32_t CQ11, CQ10,tmp32a,tmp32b; - int16_t shft; - - meangainQ12=0; - for (k = 0; k < 4; k++) - meangainQ12 += PitchGain_Q12[k]; - - meangainQ12 >>= 2; // Get average. - - /* voicing classificiation */ - if (meangainQ12 <= 819) { // mean_gain < 0.2 - shft = -1; // StepSize=2.0; - cdf = WebRtcIsacfix_kPitchLagPtrLo; - cdf_size = WebRtcIsacfix_kPitchLagSizeLo; - mean_val2Q10 = WebRtcIsacfix_kMeanLag2Lo; - mean_val4Q10 = WebRtcIsacfix_kMeanLag4Lo; - lower_limit = WebRtcIsacfix_kLowerLimitLo; - init_index = WebRtcIsacfix_kInitIndLo; - } else if (meangainQ12 <= 1638) { // mean_gain < 0.4 - shft = 0; // StepSize=1.0; - cdf = WebRtcIsacfix_kPitchLagPtrMid; - cdf_size = WebRtcIsacfix_kPitchLagSizeMid; - mean_val2Q10 = WebRtcIsacfix_kMeanLag2Mid; - mean_val4Q10 = WebRtcIsacfix_kMeanLag4Mid; - lower_limit = WebRtcIsacfix_kLowerLimitMid; - init_index = WebRtcIsacfix_kInitIndMid; - } else { - shft = 1; // StepSize=0.5; - cdf = WebRtcIsacfix_kPitchLagPtrHi; - cdf_size = WebRtcIsacfix_kPitchLagSizeHi; - mean_val2Q10 = WebRtcIsacfix_kMeanLag2Hi; - mean_val4Q10 = WebRtcIsacfix_kMeanLag4Hi; - lower_limit = WebRtcIsacfix_kLowerLimitHi; - init_index = WebRtcIsacfix_kInitIndHi; - } - - /* entropy decoding of quantization indices */ - err = WebRtcIsacfix_DecHistBisectMulti(index, streamdata, cdf, cdf_size, 1); - if ((err<0) || (index[0]<0)) // error check - return -ISAC_RANGE_ERROR_DECODE_PITCH_LAG; - - err = WebRtcIsacfix_DecHistOneStepMulti(index+1, streamdata, cdf+1, init_index, 3); - if (err<0) // error check - return -ISAC_RANGE_ERROR_DECODE_PITCH_LAG; - - - /* unquantize back to transform coefficients and do the inverse transform: S = T'*C */ - CQ11 = ((int32_t)index[0] + lower_limit[0]); // Q0 - CQ11 = WEBRTC_SPL_SHIFT_W32(CQ11,11-shft); // Scale with StepSize, Q11 - for (k=0; k> 5); - } - - CQ10 = mean_val2Q10[index[1]]; - for (k=0; k> 10; - PitchLags_Q7[k] += (int16_t)(tmp32b >> 5); - } - - CQ10 = mean_val4Q10[index[3]]; - for (k=0; k> 10; - PitchLags_Q7[k] += (int16_t)(tmp32b >> 5); - } - - return 0; -} - - - -/* quantize & code Pitch Lags */ -int WebRtcIsacfix_EncodePitchLag(int16_t* PitchLagsQ7, - int16_t* PitchGain_Q12, - Bitstr_enc* streamdata, - IsacSaveEncoderData* encData) { - int k, j; - int16_t index[PITCH_SUBFRAMES]; - int32_t meangainQ12, CQ17; - int32_t CQ11, CQ10,tmp32a; - - const int16_t *mean_val2Q10,*mean_val4Q10; - const int16_t *lower_limit, *upper_limit; - const uint16_t **cdf; - int16_t shft, tmp16b; - int32_t tmp32b; - int status = 0; - - /* compute mean pitch gain */ - meangainQ12=0; - for (k = 0; k < 4; k++) - meangainQ12 += PitchGain_Q12[k]; - - meangainQ12 >>= 2; - - /* Save data for creation of multiple bitstreams */ - if (encData != NULL) { - encData->meanGain[encData->startIdx] = meangainQ12; - } - - /* voicing classificiation */ - if (meangainQ12 <= 819) { // mean_gain < 0.2 - shft = -1; // StepSize=2.0; - cdf = WebRtcIsacfix_kPitchLagPtrLo; - mean_val2Q10 = WebRtcIsacfix_kMeanLag2Lo; - mean_val4Q10 = WebRtcIsacfix_kMeanLag4Lo; - lower_limit = WebRtcIsacfix_kLowerLimitLo; - upper_limit = WebRtcIsacfix_kUpperLimitLo; - } else if (meangainQ12 <= 1638) { // mean_gain < 0.4 - shft = 0; // StepSize=1.0; - cdf = WebRtcIsacfix_kPitchLagPtrMid; - mean_val2Q10 = WebRtcIsacfix_kMeanLag2Mid; - mean_val4Q10 = WebRtcIsacfix_kMeanLag4Mid; - lower_limit = WebRtcIsacfix_kLowerLimitMid; - upper_limit = WebRtcIsacfix_kUpperLimitMid; - } else { - shft = 1; // StepSize=0.5; - cdf = WebRtcIsacfix_kPitchLagPtrHi; - mean_val2Q10 = WebRtcIsacfix_kMeanLag2Hi; - mean_val4Q10 = WebRtcIsacfix_kMeanLag4Hi; - lower_limit = WebRtcIsacfix_kLowerLimitHi; - upper_limit = WebRtcIsacfix_kUpperLimitHi; - } - - /* find quantization index */ - for (k=0; k<4; k++) - { - /* transform */ - CQ17=0; - for (j=0; j> 2; // Q17 - - CQ17 = WEBRTC_SPL_SHIFT_W32(CQ17,shft); // Scale with StepSize - - /* quantize */ - tmp16b = (int16_t)((CQ17 + 65536) >> 17); - index[k] = tmp16b; - - /* check that the index is not outside the boundaries of the table */ - if (index[k] < lower_limit[k]) index[k] = lower_limit[k]; - else if (index[k] > upper_limit[k]) index[k] = upper_limit[k]; - index[k] -= lower_limit[k]; - - /* Save data for creation of multiple bitstreams */ - if(encData != NULL) { - encData->pitchIndex[PITCH_SUBFRAMES*encData->startIdx + k] = index[k]; - } - } - - /* unquantize back to transform coefficients and do the inverse transform: S = T'*C */ - CQ11 = (index[0] + lower_limit[0]); // Q0 - CQ11 = WEBRTC_SPL_SHIFT_W32(CQ11,11-shft); // Scale with StepSize, Q11 - - for (k=0; k> 5); // Q7. - } - - CQ10 = mean_val2Q10[index[1]]; - for (k=0; k> 10; - PitchLagsQ7[k] += (int16_t)(tmp32b >> 5); // Q7. - } - - CQ10 = mean_val4Q10[index[3]]; - for (k=0; k> 10; - PitchLagsQ7[k] += (int16_t)(tmp32b >> 5); // Q7. - } - - /* entropy coding of quantization pitch lags */ - status = WebRtcIsacfix_EncHistMulti(streamdata, index, cdf, PITCH_SUBFRAMES); - - /* If error in WebRtcIsacfix_EncHistMulti(), status will be negative, otherwise 0 */ - return status; -} - - - -/* Routines for inband signaling of bandwitdh estimation */ -/* Histograms based on uniform distribution of indices */ -/* Move global variables later! */ - - -/* cdf array for frame length indicator */ -const uint16_t kFrameLenCdf[4] = { - 0, 21845, 43690, 65535}; - -/* pointer to cdf array for frame length indicator */ -const uint16_t * const kFrameLenCdfPtr[1] = {kFrameLenCdf}; - -/* initial cdf index for decoder of frame length indicator */ -const uint16_t kFrameLenInitIndex[1] = {1}; - - -int WebRtcIsacfix_DecodeFrameLen(Bitstr_dec *streamdata, - size_t *framesamples) -{ - - int err; - int16_t frame_mode; - - err = 0; - /* entropy decoding of frame length [1:30ms,2:60ms] */ - err = WebRtcIsacfix_DecHistOneStepMulti(&frame_mode, streamdata, kFrameLenCdfPtr, kFrameLenInitIndex, 1); - if (err<0) // error check - return -ISAC_RANGE_ERROR_DECODE_FRAME_LENGTH; - - switch(frame_mode) { - case 1: - *framesamples = 480; /* 30ms */ - break; - case 2: - *framesamples = 960; /* 60ms */ - break; - default: - err = -ISAC_DISALLOWED_FRAME_MODE_DECODER; - } - - return err; -} - - -int WebRtcIsacfix_EncodeFrameLen(int16_t framesamples, Bitstr_enc *streamdata) { - - int status; - int16_t frame_mode; - - status = 0; - frame_mode = 0; - /* entropy coding of frame length [1:480 samples,2:960 samples] */ - switch(framesamples) { - case 480: - frame_mode = 1; - break; - case 960: - frame_mode = 2; - break; - default: - status = - ISAC_DISALLOWED_FRAME_MODE_ENCODER; - } - - if (status < 0) - return status; - - status = WebRtcIsacfix_EncHistMulti(streamdata, &frame_mode, kFrameLenCdfPtr, 1); - - return status; -} - -/* cdf array for estimated bandwidth */ -const uint16_t kBwCdf[25] = { - 0, 2731, 5461, 8192, 10923, 13653, 16384, 19114, 21845, 24576, 27306, 30037, - 32768, 35498, 38229, 40959, 43690, 46421, 49151, 51882, 54613, 57343, 60074, - 62804, 65535}; - -/* pointer to cdf array for estimated bandwidth */ -const uint16_t * const kBwCdfPtr[1] = {kBwCdf}; - -/* initial cdf index for decoder of estimated bandwidth*/ -const uint16_t kBwInitIndex[1] = {7}; - - -int WebRtcIsacfix_DecodeSendBandwidth(Bitstr_dec *streamdata, int16_t *BWno) { - - int err; - int16_t BWno32; - - /* entropy decoding of sender's BW estimation [0..23] */ - err = WebRtcIsacfix_DecHistOneStepMulti(&BWno32, streamdata, kBwCdfPtr, kBwInitIndex, 1); - if (err<0) // error check - return -ISAC_RANGE_ERROR_DECODE_BANDWIDTH; - *BWno = (int16_t)BWno32; - return err; - -} - - -int WebRtcIsacfix_EncodeReceiveBandwidth(int16_t *BWno, Bitstr_enc *streamdata) -{ - int status = 0; - /* entropy encoding of receiver's BW estimation [0..23] */ - status = WebRtcIsacfix_EncHistMulti(streamdata, BWno, kBwCdfPtr, 1); - - return status; -} - -/* estimate codel length of LPC Coef */ -void WebRtcIsacfix_TranscodeLpcCoef(int32_t *gain_lo_hiQ17, - int16_t *index_gQQ) { - int j, k; - int16_t posQQ, pos2QQ; - int16_t posg, offsg, gainpos; - int32_t tmpcoeffs_gQ6[KLT_ORDER_GAIN]; - int32_t tmpcoeffs_gQ17[KLT_ORDER_GAIN]; - int32_t tmpcoeffs2_gQ21[KLT_ORDER_GAIN]; - int32_t sumQQ; - - - /* log gains, mean removal and scaling */ - posg = 0; gainpos=0; - - for (k=0; k WebRtcIsacfix_kMaxIndGain[k]) { - index_gQQ[k] = WebRtcIsacfix_kMaxIndGain[k]; - } - } -} diff --git a/modules/audio_coding/codecs/isac/fix/source/entropy_coding.h b/modules/audio_coding/codecs/isac/fix/source/entropy_coding.h deleted file mode 100644 index ae11394f7c..0000000000 --- a/modules/audio_coding/codecs/isac/fix/source/entropy_coding.h +++ /dev/null @@ -1,177 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/* - * entropy_coding.h - * - * This header file contains all of the functions used to arithmetically - * encode the iSAC bistream - * - */ - -#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_ENTROPY_CODING_H_ -#define MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_ENTROPY_CODING_H_ - -#include "modules/audio_coding/codecs/isac/fix/source/structs.h" - -/* decode complex spectrum (return number of bytes in stream) */ -int WebRtcIsacfix_DecodeSpec(Bitstr_dec* streamdata, - int16_t* frQ7, - int16_t* fiQ7, - int16_t AvgPitchGain_Q12); - -/* encode complex spectrum */ -int WebRtcIsacfix_EncodeSpec(const int16_t* fr, - const int16_t* fi, - Bitstr_enc* streamdata, - int16_t AvgPitchGain_Q12); - -/* decode & dequantize LPC Coef */ -int WebRtcIsacfix_DecodeLpcCoef(Bitstr_dec* streamdata, - int32_t* LPCCoefQ17, - int32_t* gain_lo_hiQ17, - int16_t* outmodel); - -int WebRtcIsacfix_DecodeLpc(int32_t* gain_lo_hiQ17, - int16_t* LPCCoef_loQ15, - int16_t* LPCCoef_hiQ15, - Bitstr_dec* streamdata, - int16_t* outmodel); - -/* quantize & code LPC Coef */ -int WebRtcIsacfix_EncodeLpc(int32_t* gain_lo_hiQ17, - int16_t* LPCCoef_loQ15, - int16_t* LPCCoef_hiQ15, - int16_t* model, - int32_t* sizeQ11, - Bitstr_enc* streamdata, - IsacSaveEncoderData* encData, - transcode_obj* transcodeParam); - -int WebRtcIsacfix_EstCodeLpcGain(int32_t* gain_lo_hiQ17, - Bitstr_enc* streamdata, - IsacSaveEncoderData* encData); -/* decode & dequantize RC */ -int WebRtcIsacfix_DecodeRcCoef(Bitstr_dec* streamdata, int16_t* RCQ15); - -/* quantize & code RC */ -int WebRtcIsacfix_EncodeRcCoef(int16_t* RCQ15, Bitstr_enc* streamdata); - -/* decode & dequantize squared Gain */ -int WebRtcIsacfix_DecodeGain2(Bitstr_dec* streamdata, int32_t* Gain2); - -/* quantize & code squared Gain (input is squared gain) */ -int WebRtcIsacfix_EncodeGain2(int32_t* gain2, Bitstr_enc* streamdata); - -int WebRtcIsacfix_EncodePitchGain(int16_t* PitchGains_Q12, - Bitstr_enc* streamdata, - IsacSaveEncoderData* encData); - -int WebRtcIsacfix_EncodePitchLag(int16_t* PitchLagQ7, - int16_t* PitchGain_Q12, - Bitstr_enc* streamdata, - IsacSaveEncoderData* encData); - -int WebRtcIsacfix_DecodePitchGain(Bitstr_dec* streamdata, - int16_t* PitchGain_Q12); - -int WebRtcIsacfix_DecodePitchLag(Bitstr_dec* streamdata, - int16_t* PitchGain_Q12, - int16_t* PitchLagQ7); - -int WebRtcIsacfix_DecodeFrameLen(Bitstr_dec* streamdata, size_t* framelength); - -int WebRtcIsacfix_EncodeFrameLen(int16_t framelength, Bitstr_enc* streamdata); - -int WebRtcIsacfix_DecodeSendBandwidth(Bitstr_dec* streamdata, int16_t* BWno); - -int WebRtcIsacfix_EncodeReceiveBandwidth(int16_t* BWno, Bitstr_enc* streamdata); - -void WebRtcIsacfix_TranscodeLpcCoef(int32_t* tmpcoeffs_gQ6, int16_t* index_gQQ); - -// Pointer functions for LPC transforms. - -typedef void (*MatrixProduct1)(const int16_t matrix0[], - const int32_t matrix1[], - int32_t matrix_product[], - int matrix1_index_factor1, - int matrix0_index_factor1, - int matrix1_index_init_case, - int matrix1_index_step, - int matrix0_index_step, - int inner_loop_count, - int mid_loop_count, - int shift); -typedef void (*MatrixProduct2)(const int16_t matrix0[], - const int32_t matrix1[], - int32_t matrix_product[], - int matrix0_index_factor, - int matrix0_index_step); - -extern MatrixProduct1 WebRtcIsacfix_MatrixProduct1; -extern MatrixProduct2 WebRtcIsacfix_MatrixProduct2; - -void WebRtcIsacfix_MatrixProduct1C(const int16_t matrix0[], - const int32_t matrix1[], - int32_t matrix_product[], - int matrix1_index_factor1, - int matrix0_index_factor1, - int matrix1_index_init_case, - int matrix1_index_step, - int matrix0_index_step, - int inner_loop_count, - int mid_loop_count, - int shift); -void WebRtcIsacfix_MatrixProduct2C(const int16_t matrix0[], - const int32_t matrix1[], - int32_t matrix_product[], - int matrix0_index_factor, - int matrix0_index_step); - -#if defined(WEBRTC_HAS_NEON) -void WebRtcIsacfix_MatrixProduct1Neon(const int16_t matrix0[], - const int32_t matrix1[], - int32_t matrix_product[], - int matrix1_index_factor1, - int matrix0_index_factor1, - int matrix1_index_init_case, - int matrix1_index_step, - int matrix0_index_step, - int inner_loop_count, - int mid_loop_count, - int shift); -void WebRtcIsacfix_MatrixProduct2Neon(const int16_t matrix0[], - const int32_t matrix1[], - int32_t matrix_product[], - int matrix0_index_factor, - int matrix0_index_step); -#endif - -#if defined(MIPS32_LE) -void WebRtcIsacfix_MatrixProduct1MIPS(const int16_t matrix0[], - const int32_t matrix1[], - int32_t matrix_product[], - int matrix1_index_factor1, - int matrix0_index_factor1, - int matrix1_index_init_case, - int matrix1_index_step, - int matrix0_index_step, - int inner_loop_count, - int mid_loop_count, - int shift); - -void WebRtcIsacfix_MatrixProduct2MIPS(const int16_t matrix0[], - const int32_t matrix1[], - int32_t matrix_product[], - int matrix0_index_factor, - int matrix0_index_step); -#endif - -#endif // MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_ENTROPY_CODING_H_ diff --git a/modules/audio_coding/codecs/isac/fix/source/entropy_coding_mips.c b/modules/audio_coding/codecs/isac/fix/source/entropy_coding_mips.c deleted file mode 100644 index a66a43ef99..0000000000 --- a/modules/audio_coding/codecs/isac/fix/source/entropy_coding_mips.c +++ /dev/null @@ -1,249 +0,0 @@ -/* - * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#include "modules/audio_coding/codecs/isac/fix/source/entropy_coding.h" -#include "modules/audio_coding/codecs/isac/fix/source/settings.h" - -// MIPS optimization of the function WebRtcIsacfix_MatrixProduct1. -// Bit-exact with the function WebRtcIsacfix_MatrixProduct1C from -// entropy_coding.c file. -void WebRtcIsacfix_MatrixProduct1MIPS(const int16_t matrix0[], - const int32_t matrix1[], - int32_t matrix_product[], - const int matrix1_index_factor1, - const int matrix0_index_factor1, - const int matrix1_index_init_case, - const int matrix1_index_step, - const int matrix0_index_step, - const int inner_loop_count, - const int mid_loop_count, - const int shift) { - if (matrix1_index_init_case != 0) { - int j = SUBFRAMES, k = 0, n = 0; - int32_t r0, r1, r2, sum32; - int32_t* product_start = matrix_product; - int32_t* product_ptr; - const uint32_t product_step = 4 * mid_loop_count; - const uint32_t matrix0_step = 2 * matrix0_index_step; - const uint32_t matrix1_step = 4 * matrix1_index_step; - const uint32_t matrix0_step2 = 2 * matrix0_index_factor1; - const uint32_t matrix1_step2 = 4 * matrix1_index_factor1; - const int16_t* matrix0_start = matrix0; - const int32_t* matrix1_start = matrix1; - int16_t* matrix0_ptr; - int32_t* matrix1_ptr; - - __asm __volatile ( - ".set push \n\t" - ".set noreorder \n\t" - "1: \n\t" - "addu %[product_ptr], %[product_start], $0 \n\t" - "addu %[k], %[product_step], $0 \n\t" - "addiu %[j], %[j], -1 \n\t" - "addu %[matrix1_start], %[matrix1], $0 \n\t" - "2: \n\t" - "addu %[matrix1_ptr], %[matrix1_start], $0 \n\t" - "addu %[matrix0_ptr], %[matrix0_start], $0 \n\t" - "addu %[n], %[inner_loop_count], $0 \n\t" - "mul %[sum32], $0, $0 \n\t" - "3: \n\t" - "lw %[r0], 0(%[matrix1_ptr]) \n\t" - "lh %[r1], 0(%[matrix0_ptr]) \n\t" - "addu %[matrix1_ptr], %[matrix1_ptr], %[matrix1_step] \n\t" - "sllv %[r0], %[r0], %[shift] \n\t" - "andi %[r2], %[r0], 0xffff \n\t" - "sra %[r2], %[r2], 1 \n\t" - "mul %[r2], %[r2], %[r1] \n\t" - "sra %[r0], %[r0], 16 \n\t" - "mul %[r0], %[r0], %[r1] \n\t" - "addu %[matrix0_ptr], %[matrix0_ptr], %[matrix0_step] \n\t" - "addiu %[n], %[n], -1 \n\t" -#if defined(MIPS_DSP_R1_LE) - "shra_r.w %[r2], %[r2], 15 \n\t" -#else - "addiu %[r2], %[r2], 0x4000 \n\t" - "sra %[r2], %[r2], 15 \n\t" -#endif - "addu %[sum32], %[sum32], %[r2] \n\t" - "bgtz %[n], 3b \n\t" - " addu %[sum32], %[sum32], %[r0] \n\t" - "addiu %[k], %[k], -4 \n\t" - "addu %[matrix1_start], %[matrix1_start], %[matrix1_step2] \n\t" - "sw %[sum32], 0(%[product_ptr]) \n\t" - "bgtz %[k], 2b \n\t" - " addiu %[product_ptr], %[product_ptr], 4 \n\t" - "addu %[matrix0_start], %[matrix0_start], %[matrix0_step2] \n\t" - "bgtz %[j], 1b \n\t" - " addu %[product_start], %[product_start], %[product_step] \n\t" - ".set pop \n\t" - : [product_ptr] "=&r" (product_ptr), [product_start] "+r" (product_start), - [k] "=&r" (k), [j] "+r" (j), [matrix1_start] "=&r"(matrix1_start), - [matrix1_ptr] "=&r" (matrix1_ptr), [matrix0_ptr] "=&r" (matrix0_ptr), - [matrix0_start] "+r" (matrix0_start), [n] "=&r" (n), [r0] "=&r" (r0), - [sum32] "=&r" (sum32), [r1] "=&r" (r1),[r2] "=&r" (r2) - : [product_step] "r" (product_step), [matrix1] "r" (matrix1), - [inner_loop_count] "r" (inner_loop_count), - [matrix1_step] "r" (matrix1_step), [shift] "r" (shift), - [matrix0_step] "r" (matrix0_step), [matrix1_step2] "r" (matrix1_step2), - [matrix0_step2] "r" (matrix0_step2) - : "hi", "lo", "memory" - ); - } else { - int j = SUBFRAMES, k = 0, n = 0; - int32_t r0, r1, r2, sum32; - int32_t* product_start = matrix_product; - int32_t* product_ptr; - const uint32_t product_step = 4 * mid_loop_count; - const uint32_t matrix0_step = 2 * matrix0_index_step; - const uint32_t matrix1_step = 4 * matrix1_index_step; - const uint32_t matrix0_step2 = 2 * matrix0_index_factor1; - const uint32_t matrix1_step2 = 4 * matrix1_index_factor1; - const int16_t* matrix0_start = matrix0; - const int32_t* matrix1_start = matrix1; - int16_t* matrix0_ptr; - int32_t* matrix1_ptr; - - __asm __volatile ( - ".set push \n\t" - ".set noreorder \n\t" - "1: \n\t" - "addu %[product_ptr], %[product_start], $0 \n\t" - "addu %[k], %[product_step], $0 \n\t" - "addiu %[j], %[j], -1 \n\t" - "addu %[matrix0_start], %[matrix0], $0 \n\t" - "2: \n\t" - "addu %[matrix1_ptr], %[matrix1_start], $0 \n\t" - "addu %[matrix0_ptr], %[matrix0_start], $0 \n\t" - "addu %[n], %[inner_loop_count], $0 \n\t" - "mul %[sum32], $0, $0 \n\t" - "3: \n\t" - "lw %[r0], 0(%[matrix1_ptr]) \n\t" - "lh %[r1], 0(%[matrix0_ptr]) \n\t" - "addu %[matrix1_ptr], %[matrix1_ptr], %[matrix1_step] \n\t" - "sllv %[r0], %[r0], %[shift] \n\t" - "andi %[r2], %[r0], 0xffff \n\t" - "sra %[r2], %[r2], 1 \n\t" - "mul %[r2], %[r2], %[r1] \n\t" - "sra %[r0], %[r0], 16 \n\t" - "mul %[r0], %[r0], %[r1] \n\t" - "addu %[matrix0_ptr], %[matrix0_ptr], %[matrix0_step] \n\t" - "addiu %[n], %[n], -1 \n\t" -#if defined(MIPS_DSP_R1_LE) - "shra_r.w %[r2], %[r2], 15 \n\t" -#else - "addiu %[r2], %[r2], 0x4000 \n\t" - "sra %[r2], %[r2], 15 \n\t" -#endif - "addu %[sum32], %[sum32], %[r2] \n\t" - "bgtz %[n], 3b \n\t" - " addu %[sum32], %[sum32], %[r0] \n\t" - "addiu %[k], %[k], -4 \n\t" - "addu %[matrix0_start], %[matrix0_start], %[matrix0_step2] \n\t" - "sw %[sum32], 0(%[product_ptr]) \n\t" - "bgtz %[k], 2b \n\t" - " addiu %[product_ptr], %[product_ptr], 4 \n\t" - "addu %[matrix1_start], %[matrix1_start], %[matrix1_step2] \n\t" - "bgtz %[j], 1b \n\t" - " addu %[product_start], %[product_start], %[product_step] \n\t" - ".set pop \n\t" - : [product_ptr] "=&r" (product_ptr), [product_start] "+r" (product_start), - [k] "=&r" (k), [j] "+r" (j), [matrix1_start] "+r"(matrix1_start), - [matrix1_ptr] "=&r" (matrix1_ptr), [matrix0_ptr] "=&r" (matrix0_ptr), - [matrix0_start] "=&r" (matrix0_start), [n] "=&r" (n), [r0] "=&r" (r0), - [sum32] "=&r" (sum32), [r1] "=&r" (r1),[r2] "=&r" (r2) - : [product_step] "r" (product_step), [matrix0] "r" (matrix0), - [inner_loop_count] "r" (inner_loop_count), - [matrix1_step] "r" (matrix1_step), [shift] "r" (shift), - [matrix0_step] "r" (matrix0_step), [matrix1_step2] "r" (matrix1_step2), - [matrix0_step2] "r" (matrix0_step2) - : "hi", "lo", "memory" - ); - } -} - -// MIPS optimization of the function WebRtcIsacfix_MatrixProduct2. -// Bit-exact with the function WebRtcIsacfix_MatrixProduct2C from -// entropy_coding.c file. -void WebRtcIsacfix_MatrixProduct2MIPS(const int16_t matrix0[], - const int32_t matrix1[], - int32_t matrix_product[], - const int matrix0_index_factor, - const int matrix0_index_step) { - int j = 0, n = 0; - int loop_count = SUBFRAMES; - const int16_t* matrix0_ptr; - const int32_t* matrix1_ptr; - const int16_t* matrix0_start = matrix0; - const int matrix0_step = 2 * matrix0_index_step; - const int matrix0_step2 = 2 * matrix0_index_factor; - int32_t r0, r1, r2, r3, r4, sum32, sum32_2; - - __asm __volatile ( - ".set push \n\t" - ".set noreorder \n\t" - "addu %[j], %[loop_count], $0 \n\t" - "addu %[matrix0_start], %[matrix0], $0 \n\t" - "1: \n\t" - "addu %[matrix1_ptr], %[matrix1], $0 \n\t" - "addu %[matrix0_ptr], %[matrix0_start], $0 \n\t" - "addu %[n], %[loop_count], $0 \n\t" - "mul %[sum32], $0, $0 \n\t" - "mul %[sum32_2], $0, $0 \n\t" - "2: \n\t" - "lw %[r0], 0(%[matrix1_ptr]) \n\t" - "lw %[r1], 4(%[matrix1_ptr]) \n\t" - "lh %[r2], 0(%[matrix0_ptr]) \n\t" - "andi %[r3], %[r0], 0xffff \n\t" - "sra %[r3], %[r3], 1 \n\t" - "mul %[r3], %[r3], %[r2] \n\t" - "andi %[r4], %[r1], 0xffff \n\t" - "sra %[r4], %[r4], 1 \n\t" - "mul %[r4], %[r4], %[r2] \n\t" - "sra %[r0], %[r0], 16 \n\t" - "mul %[r0], %[r0], %[r2] \n\t" - "sra %[r1], %[r1], 16 \n\t" - "mul %[r1], %[r1], %[r2] \n\t" -#if defined(MIPS_DSP_R1_LE) - "shra_r.w %[r3], %[r3], 15 \n\t" - "shra_r.w %[r4], %[r4], 15 \n\t" -#else - "addiu %[r3], %[r3], 0x4000 \n\t" - "sra %[r3], %[r3], 15 \n\t" - "addiu %[r4], %[r4], 0x4000 \n\t" - "sra %[r4], %[r4], 15 \n\t" -#endif - "addiu %[matrix1_ptr], %[matrix1_ptr], 8 \n\t" - "addu %[matrix0_ptr], %[matrix0_ptr], %[matrix0_step] \n\t" - "addiu %[n], %[n], -1 \n\t" - "addu %[sum32], %[sum32], %[r3] \n\t" - "addu %[sum32_2], %[sum32_2], %[r4] \n\t" - "addu %[sum32], %[sum32], %[r0] \n\t" - "bgtz %[n], 2b \n\t" - " addu %[sum32_2], %[sum32_2], %[r1] \n\t" - "sra %[sum32], %[sum32], 3 \n\t" - "sra %[sum32_2], %[sum32_2], 3 \n\t" - "addiu %[j], %[j], -1 \n\t" - "addu %[matrix0_start], %[matrix0_start], %[matrix0_step2] \n\t" - "sw %[sum32], 0(%[matrix_product]) \n\t" - "sw %[sum32_2], 4(%[matrix_product]) \n\t" - "bgtz %[j], 1b \n\t" - " addiu %[matrix_product], %[matrix_product], 8 \n\t" - ".set pop \n\t" - : [j] "=&r" (j), [matrix0_start] "=&r" (matrix0_start), - [matrix1_ptr] "=&r" (matrix1_ptr), [matrix0_ptr] "=&r" (matrix0_ptr), - [n] "=&r" (n), [sum32] "=&r" (sum32), [sum32_2] "=&r" (sum32_2), - [r0] "=&r" (r0), [r1] "=&r" (r1), [r2] "=&r" (r2), [r3] "=&r" (r3), - [r4] "=&r" (r4), [matrix_product] "+r" (matrix_product) - : [loop_count] "r" (loop_count), [matrix0] "r" (matrix0), - [matrix1] "r" (matrix1), [matrix0_step] "r" (matrix0_step), - [matrix0_step2] "r" (matrix0_step2) - : "hi", "lo", "memory" - ); -} diff --git a/modules/audio_coding/codecs/isac/fix/source/entropy_coding_neon.c b/modules/audio_coding/codecs/isac/fix/source/entropy_coding_neon.c deleted file mode 100644 index 0200567880..0000000000 --- a/modules/audio_coding/codecs/isac/fix/source/entropy_coding_neon.c +++ /dev/null @@ -1,217 +0,0 @@ -/* - * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/* This file contains WebRtcIsacfix_MatrixProduct1Neon() and - * WebRtcIsacfix_MatrixProduct2Neon() for ARM Neon platform. API's are in - * entropy_coding.c. Results are bit exact with the c code for - * generic platforms. - */ - -#include -#include - -#include "modules/audio_coding/codecs/isac/fix/source/entropy_coding.h" -#include "common_audio/signal_processing/include/signal_processing_library.h" -#include "rtc_base/checks.h" - -void WebRtcIsacfix_MatrixProduct1Neon(const int16_t matrix0[], - const int32_t matrix1[], - int32_t matrix_product[], - const int matrix1_index_factor1, - const int matrix0_index_factor1, - const int matrix1_index_init_case, - const int matrix1_index_step, - const int matrix0_index_step, - const int inner_loop_count, - const int mid_loop_count, - const int shift) { - int j = 0, k = 0, n = 0; - int matrix1_index = 0, matrix0_index = 0, matrix_prod_index = 0; - int* matrix1_index_factor2 = &j; - int* matrix0_index_factor2 = &k; - if (matrix1_index_init_case != 0) { - matrix1_index_factor2 = &k; - matrix0_index_factor2 = &j; - } - int32x4_t shift32x4 = vdupq_n_s32(shift); - int32x2_t shift32x2 = vdup_n_s32(shift); - int32x4_t sum_32x4 = vdupq_n_s32(0); - int32x2_t sum_32x2 = vdup_n_s32(0); - - RTC_DCHECK_EQ(0, inner_loop_count % 2); - RTC_DCHECK_EQ(0, mid_loop_count % 2); - - if (matrix1_index_init_case != 0 && matrix1_index_factor1 == 1) { - for (j = 0; j < SUBFRAMES; j++) { - matrix_prod_index = mid_loop_count * j; - for (k = 0; k < (mid_loop_count >> 2) << 2; k += 4) { - sum_32x4 = veorq_s32(sum_32x4, sum_32x4); // Initialize to zeros. - matrix1_index = k; - matrix0_index = matrix0_index_factor1 * j; - for (n = 0; n < inner_loop_count; n++) { - int32x4_t matrix0_32x4 = - vdupq_n_s32((int32_t)(matrix0[matrix0_index]) << 15); - int32x4_t matrix1_32x4 = - vshlq_s32(vld1q_s32(&matrix1[matrix1_index]), shift32x4); - int32x4_t multi_32x4 = vqdmulhq_s32(matrix0_32x4, matrix1_32x4); - sum_32x4 = vqaddq_s32(sum_32x4, multi_32x4); - matrix1_index += matrix1_index_step; - matrix0_index += matrix0_index_step; - } - vst1q_s32(&matrix_product[matrix_prod_index], sum_32x4); - matrix_prod_index += 4; - } - if (mid_loop_count % 4 > 1) { - sum_32x2 = veor_s32(sum_32x2, sum_32x2); // Initialize to zeros. - matrix1_index = k; - k += 2; - matrix0_index = matrix0_index_factor1 * j; - for (n = 0; n < inner_loop_count; n++) { - int32x2_t matrix0_32x2 = - vdup_n_s32((int32_t)(matrix0[matrix0_index]) << 15); - int32x2_t matrix1_32x2 = - vshl_s32(vld1_s32(&matrix1[matrix1_index]), shift32x2); - int32x2_t multi_32x2 = vqdmulh_s32(matrix0_32x2, matrix1_32x2); - sum_32x2 = vqadd_s32(sum_32x2, multi_32x2); - matrix1_index += matrix1_index_step; - matrix0_index += matrix0_index_step; - } - vst1_s32(&matrix_product[matrix_prod_index], sum_32x2); - matrix_prod_index += 2; - } - } - } - else if (matrix1_index_init_case == 0 && matrix0_index_factor1 == 1) { - int32x2_t multi_32x2 = vdup_n_s32(0); - int32x2_t matrix0_32x2 = vdup_n_s32(0); - for (j = 0; j < SUBFRAMES; j++) { - matrix_prod_index = mid_loop_count * j; - for (k = 0; k < (mid_loop_count >> 2) << 2; k += 4) { - sum_32x4 = veorq_s32(sum_32x4, sum_32x4); // Initialize to zeros. - matrix1_index = matrix1_index_factor1 * j; - matrix0_index = k; - for (n = 0; n < inner_loop_count; n++) { - int32x4_t matrix1_32x4 = vdupq_n_s32(matrix1[matrix1_index] << shift); - int32x4_t matrix0_32x4 = - vshll_n_s16(vld1_s16(&matrix0[matrix0_index]), 15); - int32x4_t multi_32x4 = vqdmulhq_s32(matrix0_32x4, matrix1_32x4); - sum_32x4 = vqaddq_s32(sum_32x4, multi_32x4); - matrix1_index += matrix1_index_step; - matrix0_index += matrix0_index_step; - } - vst1q_s32(&matrix_product[matrix_prod_index], sum_32x4); - matrix_prod_index += 4; - } - if (mid_loop_count % 4 > 1) { - sum_32x2 = veor_s32(sum_32x2, sum_32x2); // Initialize to zeros. - matrix1_index = matrix1_index_factor1 * j; - matrix0_index = k; - for (n = 0; n < inner_loop_count; n++) { - int32x2_t matrix1_32x2 = vdup_n_s32(matrix1[matrix1_index] << shift); - matrix0_32x2 = - vset_lane_s32((int32_t)matrix0[matrix0_index], matrix0_32x2, 0); - matrix0_32x2 = vset_lane_s32((int32_t)matrix0[matrix0_index + 1], - matrix0_32x2, 1); - matrix0_32x2 = vshl_n_s32(matrix0_32x2, 15); - multi_32x2 = vqdmulh_s32(matrix1_32x2, matrix0_32x2); - sum_32x2 = vqadd_s32(sum_32x2, multi_32x2); - matrix1_index += matrix1_index_step; - matrix0_index += matrix0_index_step; - } - vst1_s32(&matrix_product[matrix_prod_index], sum_32x2); - matrix_prod_index += 2; - } - } - } - else if (matrix1_index_init_case == 0 && - matrix1_index_step == 1 && - matrix0_index_step == 1) { - int32x2_t multi_32x2 = vdup_n_s32(0); - int32x2_t matrix0_32x2 = vdup_n_s32(0); - for (j = 0; j < SUBFRAMES; j++) { - matrix_prod_index = mid_loop_count * j; - for (k = 0; k < mid_loop_count; k++) { - sum_32x4 = veorq_s32(sum_32x4, sum_32x4); // Initialize to zeros. - matrix1_index = matrix1_index_factor1 * j; - matrix0_index = matrix0_index_factor1 * k; - for (n = 0; n < (inner_loop_count >> 2) << 2; n += 4) { - int32x4_t matrix1_32x4 = - vshlq_s32(vld1q_s32(&matrix1[matrix1_index]), shift32x4); - int32x4_t matrix0_32x4 = - vshll_n_s16(vld1_s16(&matrix0[matrix0_index]), 15); - int32x4_t multi_32x4 = vqdmulhq_s32(matrix0_32x4, matrix1_32x4); - sum_32x4 = vqaddq_s32(sum_32x4, multi_32x4); - matrix1_index += 4; - matrix0_index += 4; - } - sum_32x2 = vqadd_s32(vget_low_s32(sum_32x4), vget_high_s32(sum_32x4)); - if (inner_loop_count % 4 > 1) { - int32x2_t matrix1_32x2 = - vshl_s32(vld1_s32(&matrix1[matrix1_index]), shift32x2); - matrix0_32x2 = - vset_lane_s32((int32_t)matrix0[matrix0_index], matrix0_32x2, 0); - matrix0_32x2 = vset_lane_s32((int32_t)matrix0[matrix0_index + 1], - matrix0_32x2, 1); - matrix0_32x2 = vshl_n_s32(matrix0_32x2, 15); - multi_32x2 = vqdmulh_s32(matrix1_32x2, matrix0_32x2); - sum_32x2 = vqadd_s32(sum_32x2, multi_32x2); - } - sum_32x2 = vpadd_s32(sum_32x2, sum_32x2); - vst1_lane_s32(&matrix_product[matrix_prod_index], sum_32x2, 0); - matrix_prod_index++; - } - } - } - else { - for (j = 0; j < SUBFRAMES; j++) { - matrix_prod_index = mid_loop_count * j; - for (k=0; k < mid_loop_count; k++) { - int32_t sum32 = 0; - matrix1_index = matrix1_index_factor1 * (*matrix1_index_factor2); - matrix0_index = matrix0_index_factor1 * (*matrix0_index_factor2); - for (n = 0; n < inner_loop_count; n++) { - sum32 += (WEBRTC_SPL_MUL_16_32_RSFT16(matrix0[matrix0_index], - matrix1[matrix1_index] << shift)); - matrix1_index += matrix1_index_step; - matrix0_index += matrix0_index_step; - } - matrix_product[matrix_prod_index] = sum32; - matrix_prod_index++; - } - } - } -} - -void WebRtcIsacfix_MatrixProduct2Neon(const int16_t matrix0[], - const int32_t matrix1[], - int32_t matrix_product[], - const int matrix0_index_factor, - const int matrix0_index_step) { - int j = 0, n = 0; - int matrix1_index = 0, matrix0_index = 0, matrix_prod_index = 0; - int32x2_t sum_32x2 = vdup_n_s32(0); - for (j = 0; j < SUBFRAMES; j++) { - sum_32x2 = veor_s32(sum_32x2, sum_32x2); // Initialize to zeros. - matrix1_index = 0; - matrix0_index = matrix0_index_factor * j; - for (n = SUBFRAMES; n > 0; n--) { - int32x2_t matrix0_32x2 = - vdup_n_s32((int32_t)(matrix0[matrix0_index]) << 15); - int32x2_t matrix1_32x2 = vld1_s32(&matrix1[matrix1_index]); - int32x2_t multi_32x2 = vqdmulh_s32(matrix0_32x2, matrix1_32x2); - sum_32x2 = vqadd_s32(sum_32x2, multi_32x2); - matrix1_index += 2; - matrix0_index += matrix0_index_step; - } - sum_32x2 = vshr_n_s32(sum_32x2, 3); - vst1_s32(&matrix_product[matrix_prod_index], sum_32x2); - matrix_prod_index += 2; - } -} diff --git a/modules/audio_coding/codecs/isac/fix/source/fft.c b/modules/audio_coding/codecs/isac/fix/source/fft.c deleted file mode 100644 index a0ed3f83ce..0000000000 --- a/modules/audio_coding/codecs/isac/fix/source/fft.c +++ /dev/null @@ -1,415 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/* - * fft.c - * - * Fast Fourier Transform - * - */ - - -#include "modules/audio_coding/codecs/isac/fix/source/fft.h" - -static const int16_t kSortTabFft[240] = { - 0, 60, 120, 180, 20, 80, 140, 200, 40, 100, 160, 220, - 4, 64, 124, 184, 24, 84, 144, 204, 44, 104, 164, 224, - 8, 68, 128, 188, 28, 88, 148, 208, 48, 108, 168, 228, - 12, 72, 132, 192, 32, 92, 152, 212, 52, 112, 172, 232, - 16, 76, 136, 196, 36, 96, 156, 216, 56, 116, 176, 236, - 1, 61, 121, 181, 21, 81, 141, 201, 41, 101, 161, 221, - 5, 65, 125, 185, 25, 85, 145, 205, 45, 105, 165, 225, - 9, 69, 129, 189, 29, 89, 149, 209, 49, 109, 169, 229, - 13, 73, 133, 193, 33, 93, 153, 213, 53, 113, 173, 233, - 17, 77, 137, 197, 37, 97, 157, 217, 57, 117, 177, 237, - 2, 62, 122, 182, 22, 82, 142, 202, 42, 102, 162, 222, - 6, 66, 126, 186, 26, 86, 146, 206, 46, 106, 166, 226, - 10, 70, 130, 190, 30, 90, 150, 210, 50, 110, 170, 230, - 14, 74, 134, 194, 34, 94, 154, 214, 54, 114, 174, 234, - 18, 78, 138, 198, 38, 98, 158, 218, 58, 118, 178, 238, - 3, 63, 123, 183, 23, 83, 143, 203, 43, 103, 163, 223, - 7, 67, 127, 187, 27, 87, 147, 207, 47, 107, 167, 227, - 11, 71, 131, 191, 31, 91, 151, 211, 51, 111, 171, 231, - 15, 75, 135, 195, 35, 95, 155, 215, 55, 115, 175, 235, - 19, 79, 139, 199, 39, 99, 159, 219, 59, 119, 179, 239 -}; - -/* Cosine table in Q14 */ -static const int16_t kCosTabFfftQ14[240] = { - 16384, 16378, 16362, 16333, 16294, 16244, 16182, 16110, 16026, 15931, 15826, 15709, - 15582, 15444, 15296, 15137, 14968, 14788, 14598, 14399, 14189, 13970, 13741, 13502, - 13255, 12998, 12733, 12458, 12176, 11885, 11585, 11278, 10963, 10641, 10311, 9974, - 9630, 9280, 8923, 8561, 8192, 7818, 7438, 7053, 6664, 6270, 5872, 5469, - 5063, 4653, 4240, 3825, 3406, 2986, 2563, 2139, 1713, 1285, 857, 429, - 0, -429, -857, -1285, -1713, -2139, -2563, -2986, -3406, -3825, -4240, -4653, - -5063, -5469, -5872, -6270, -6664, -7053, -7438, -7818, -8192, -8561, -8923, -9280, - -9630, -9974, -10311, -10641, -10963, -11278, -11585, -11885, -12176, -12458, -12733, -12998, - -13255, -13502, -13741, -13970, -14189, -14399, -14598, -14788, -14968, -15137, -15296, -15444, - -15582, -15709, -15826, -15931, -16026, -16110, -16182, -16244, -16294, -16333, -16362, -16378, - -16384, -16378, -16362, -16333, -16294, -16244, -16182, -16110, -16026, -15931, -15826, -15709, - -15582, -15444, -15296, -15137, -14968, -14788, -14598, -14399, -14189, -13970, -13741, -13502, - -13255, -12998, -12733, -12458, -12176, -11885, -11585, -11278, -10963, -10641, -10311, -9974, - -9630, -9280, -8923, -8561, -8192, -7818, -7438, -7053, -6664, -6270, -5872, -5469, - -5063, -4653, -4240, -3825, -3406, -2986, -2563, -2139, -1713, -1285, -857, -429, - 0, 429, 857, 1285, 1713, 2139, 2563, 2986, 3406, 3825, 4240, 4653, - 5063, 5469, 5872, 6270, 6664, 7053, 7438, 7818, 8192, 8561, 8923, 9280, - 9630, 9974, 10311, 10641, 10963, 11278, 11585, 11885, 12176, 12458, 12733, 12998, - 13255, 13502, 13741, 13970, 14189, 14399, 14598, 14788, 14968, 15137, 15296, 15444, - 15582, 15709, 15826, 15931, 16026, 16110, 16182, 16244, 16294, 16333, 16362, 16378 -}; - - - -/* Uses 16x16 mul, without rounding, which is faster. Uses WEBRTC_SPL_MUL_16_16_RSFT */ -int16_t WebRtcIsacfix_FftRadix16Fastest(int16_t RexQx[], int16_t ImxQx[], int16_t iSign) { - - int16_t dd, ee, ff, gg, hh, ii; - int16_t k0, k1, k2, k3, k4, kk; - int16_t tmp116, tmp216; - - int16_t ccc1Q14, ccc2Q14, ccc3Q14, sss1Q14, sss2Q14, sss3Q14; - int16_t sss60Q14, ccc72Q14, sss72Q14; - int16_t aaQx, ajQx, akQx, ajmQx, ajpQx, akmQx, akpQx; - int16_t bbQx, bjQx, bkQx, bjmQx, bjpQx, bkmQx, bkpQx; - - int16_t ReDATAQx[240], ImDATAQx[240]; - - sss60Q14 = kCosTabFfftQ14[20]; - ccc72Q14 = kCosTabFfftQ14[48]; - sss72Q14 = kCosTabFfftQ14[12]; - - if (iSign < 0) { - sss72Q14 = -sss72Q14; - sss60Q14 = -sss60Q14; - } - /* Complexity is: 10 cycles */ - - /* compute fourier transform */ - - // transform for factor of 4 - for (kk=0; kk<60; kk++) { - k0 = kk; - k1 = k0 + 60; - k2 = k1 + 60; - k3 = k2 + 60; - - akpQx = RexQx[k0] + RexQx[k2]; - akmQx = RexQx[k0] - RexQx[k2]; - ajpQx = RexQx[k1] + RexQx[k3]; - ajmQx = RexQx[k1] - RexQx[k3]; - bkpQx = ImxQx[k0] + ImxQx[k2]; - bkmQx = ImxQx[k0] - ImxQx[k2]; - bjpQx = ImxQx[k1] + ImxQx[k3]; - bjmQx = ImxQx[k1] - ImxQx[k3]; - - RexQx[k0] = akpQx + ajpQx; - ImxQx[k0] = bkpQx + bjpQx; - ajpQx = akpQx - ajpQx; - bjpQx = bkpQx - bjpQx; - if (iSign < 0) { - akpQx = akmQx + bjmQx; - bkpQx = bkmQx - ajmQx; - akmQx -= bjmQx; - bkmQx += ajmQx; - } else { - akpQx = akmQx - bjmQx; - bkpQx = bkmQx + ajmQx; - akmQx += bjmQx; - bkmQx -= ajmQx; - } - - ccc1Q14 = kCosTabFfftQ14[kk]; - ccc2Q14 = kCosTabFfftQ14[2 * kk]; - ccc3Q14 = kCosTabFfftQ14[3 * kk]; - sss1Q14 = kCosTabFfftQ14[kk + 60]; - sss2Q14 = kCosTabFfftQ14[2 * kk + 60]; - sss3Q14 = kCosTabFfftQ14[3 * kk + 60]; - if (iSign==1) { - sss1Q14 = -sss1Q14; - sss2Q14 = -sss2Q14; - sss3Q14 = -sss3Q14; - } - - //Do several multiplications like Q14*Q16>>14 = Q16 - // RexQ16[k1] = akpQ16 * ccc1Q14 - bkpQ16 * sss1Q14; - // RexQ16[k2] = ajpQ16 * ccc2Q14 - bjpQ16 * sss2Q14; - // RexQ16[k3] = akmQ16 * ccc3Q14 - bkmQ16 * sss3Q14; - // ImxQ16[k1] = akpQ16 * sss1Q14 + bkpQ16 * ccc1Q14; - // ImxQ16[k2] = ajpQ16 * sss2Q14 + bjpQ16 * ccc2Q14; - // ImxQ16[k3] = akmQ16 * sss3Q14 + bkmQ16 * ccc3Q14; - - RexQx[k1] = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(ccc1Q14, akpQx, 14) - - (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(sss1Q14, bkpQx, 14); // 6 non-mul + 2 mul cycles, i.e. 8 cycles (6+2*7=20 cycles if 16x32mul) - RexQx[k2] = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(ccc2Q14, ajpQx, 14) - - (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(sss2Q14, bjpQx, 14); - RexQx[k3] = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(ccc3Q14, akmQx, 14) - - (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(sss3Q14, bkmQx, 14); - ImxQx[k1] = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(sss1Q14, akpQx, 14) + - (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(ccc1Q14, bkpQx, 14); - ImxQx[k2] = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(sss2Q14, ajpQx, 14) + - (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(ccc2Q14, bjpQx, 14); - ImxQx[k3] = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(sss3Q14, akmQx, 14) + - (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(ccc3Q14, bkmQx, 14); - //This mul segment needs 6*8 = 48 cycles for 16x16 muls, but 6*20 = 120 cycles for 16x32 muls - - - } - /* Complexity is: 51+48 = 99 cycles for 16x16 muls, but 51+120 = 171 cycles for 16x32 muls*/ - - // transform for factor of 3 - kk=0; - k1=20; - k2=40; - - for (hh=0; hh<4; hh++) { - for (ii=0; ii<20; ii++) { - akQx = RexQx[kk]; - bkQx = ImxQx[kk]; - ajQx = RexQx[k1] + RexQx[k2]; - bjQx = ImxQx[k1] + ImxQx[k2]; - RexQx[kk] = akQx + ajQx; - ImxQx[kk] = bkQx + bjQx; - tmp116 = ajQx >> 1; - tmp216 = bjQx >> 1; - akQx = akQx - tmp116; - bkQx = bkQx - tmp216; - tmp116 = RexQx[k1] - RexQx[k2]; - tmp216 = ImxQx[k1] - ImxQx[k2]; - - ajQx = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(sss60Q14, tmp116, 14); // Q14*Qx>>14 = Qx - bjQx = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(sss60Q14, tmp216, 14); // Q14*Qx>>14 = Qx - RexQx[k1] = akQx - bjQx; - RexQx[k2] = akQx + bjQx; - ImxQx[k1] = bkQx + ajQx; - ImxQx[k2] = bkQx - ajQx; - - kk++; - k1++; - k2++; - } - /* Complexity : (31+6)*20 = 740 cycles for 16x16 muls, but (31+18)*20 = 980 cycles for 16x32 muls*/ - kk=kk+40; - k1=k1+40; - k2=k2+40; - } - /* Complexity : 4*(740+3) = 2972 cycles for 16x16 muls, but 4*(980+3) = 3932 cycles for 16x32 muls*/ - - /* multiply by rotation factor for odd factor 3 or 5 (not for 4) - Same code (duplicated) for both ii=2 and ii=3 */ - kk = 1; - ee = 0; - ff = 0; - - for (gg=0; gg<19; gg++) { - kk += 20; - ff = ff+4; - for (hh=0; hh<2; hh++) { - ee = ff + hh * ff; - dd = ee + 60; - ccc2Q14 = kCosTabFfftQ14[ee]; - sss2Q14 = kCosTabFfftQ14[dd]; - if (iSign==1) { - sss2Q14 = -sss2Q14; - } - for (ii=0; ii<4; ii++) { - akQx = RexQx[kk]; - bkQx = ImxQx[kk]; - RexQx[kk] = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(ccc2Q14, akQx, 14) - // Q14*Qx>>14 = Qx - (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(sss2Q14, bkQx, 14); - ImxQx[kk] = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(sss2Q14, akQx, 14) + // Q14*Qx>>14 = Qx - (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(ccc2Q14, bkQx, 14); - - - kk += 60; - } - kk = kk - 220; - } - // Complexity: 2*(13+5+4*13+2) = 144 for 16x16 muls, but 2*(13+5+4*33+2) = 304 cycles for 16x32 muls - kk = kk - 59; - } - // Complexity: 19*144 = 2736 for 16x16 muls, but 19*304 = 5776 cycles for 16x32 muls - - // transform for factor of 5 - kk = 0; - ccc2Q14 = kCosTabFfftQ14[96]; - sss2Q14 = kCosTabFfftQ14[84]; - if (iSign==1) { - sss2Q14 = -sss2Q14; - } - - for (hh=0; hh<4; hh++) { - for (ii=0; ii<12; ii++) { - k1 = kk + 4; - k2 = k1 + 4; - k3 = k2 + 4; - k4 = k3 + 4; - - akpQx = RexQx[k1] + RexQx[k4]; - akmQx = RexQx[k1] - RexQx[k4]; - bkpQx = ImxQx[k1] + ImxQx[k4]; - bkmQx = ImxQx[k1] - ImxQx[k4]; - ajpQx = RexQx[k2] + RexQx[k3]; - ajmQx = RexQx[k2] - RexQx[k3]; - bjpQx = ImxQx[k2] + ImxQx[k3]; - bjmQx = ImxQx[k2] - ImxQx[k3]; - aaQx = RexQx[kk]; - bbQx = ImxQx[kk]; - RexQx[kk] = aaQx + akpQx + ajpQx; - ImxQx[kk] = bbQx + bkpQx + bjpQx; - - akQx = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(ccc72Q14, akpQx, 14) + - (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(ccc2Q14, ajpQx, 14) + aaQx; - bkQx = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(ccc72Q14, bkpQx, 14) + - (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(ccc2Q14, bjpQx, 14) + bbQx; - ajQx = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(sss72Q14, akmQx, 14) + - (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(sss2Q14, ajmQx, 14); - bjQx = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(sss72Q14, bkmQx, 14) + - (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(sss2Q14, bjmQx, 14); - // 32+4*8=64 or 32+4*20=112 - - RexQx[k1] = akQx - bjQx; - RexQx[k4] = akQx + bjQx; - ImxQx[k1] = bkQx + ajQx; - ImxQx[k4] = bkQx - ajQx; - - akQx = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(ccc2Q14, akpQx, 14) + - (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(ccc72Q14, ajpQx, 14) + aaQx; - bkQx = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(ccc2Q14, bkpQx, 14) + - (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(ccc72Q14, bjpQx, 14) + bbQx; - ajQx = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(sss2Q14, akmQx, 14) - - (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(sss72Q14, ajmQx, 14); - bjQx = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(sss2Q14, bkmQx, 14) - - (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(sss72Q14, bjmQx, 14); - // 8+4*8=40 or 8+4*20=88 - - RexQx[k2] = akQx - bjQx; - RexQx[k3] = akQx + bjQx; - ImxQx[k2] = bkQx + ajQx; - ImxQx[k3] = bkQx - ajQx; - - kk = k4 + 4; - } - // Complexity: 12*(64+40+10) = 1368 for 16x16 muls, but 12*(112+88+10) = 2520 cycles for 16x32 muls - kk -= 239; - } - // Complexity: 4*1368 = 5472 for 16x16 muls, but 4*2520 = 10080 cycles for 16x32 muls - - /* multiply by rotation factor for odd factor 3 or 5 (not for 4) - Same code (duplicated) for both ii=2 and ii=3 */ - kk = 1; - ee=0; - - for (gg=0; gg<3; gg++) { - kk += 4; - dd = 12 + 12 * gg; - ff = 0; - for (hh=0; hh<4; hh++) { - ff = ff+dd; - ee = ff+60; - for (ii=0; ii<12; ii++) { - akQx = RexQx[kk]; - bkQx = ImxQx[kk]; - - ccc2Q14 = kCosTabFfftQ14[ff]; - sss2Q14 = kCosTabFfftQ14[ee]; - - if (iSign==1) { - sss2Q14 = -sss2Q14; - } - - RexQx[kk] = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(ccc2Q14, akQx, 14) - - (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(sss2Q14, bkQx, 14); - ImxQx[kk] = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(sss2Q14, akQx, 14) + - (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(ccc2Q14, bkQx, 14); - - kk += 20; - } - kk = kk - 236; - // Complexity: 12*(12+12) = 288 for 16x16 muls, but 12*(12+32) = 528 cycles for 16x32 muls - } - kk = kk - 19; - // Complexity: 4*288+6 for 16x16 muls, but 4*528+6 cycles for 16x32 muls - } - // Complexity: 3*4*288+6 = 3462 for 16x16 muls, but 3*4*528+6 = 6342 cycles for 16x32 muls - - - // last transform for factor of 4 */ - for (kk=0; kk<240; kk=kk+4) { - k1 = kk + 1; - k2 = k1 + 1; - k3 = k2 + 1; - - akpQx = RexQx[kk] + RexQx[k2]; - akmQx = RexQx[kk] - RexQx[k2]; - ajpQx = RexQx[k1] + RexQx[k3]; - ajmQx = RexQx[k1] - RexQx[k3]; - bkpQx = ImxQx[kk] + ImxQx[k2]; - bkmQx = ImxQx[kk] - ImxQx[k2]; - bjpQx = ImxQx[k1] + ImxQx[k3]; - bjmQx = ImxQx[k1] - ImxQx[k3]; - RexQx[kk] = akpQx + ajpQx; - ImxQx[kk] = bkpQx + bjpQx; - ajpQx = akpQx - ajpQx; - bjpQx = bkpQx - bjpQx; - if (iSign < 0) { - akpQx = akmQx + bjmQx; - bkpQx = bkmQx - ajmQx; - akmQx -= bjmQx; - bkmQx += ajmQx; - } else { - akpQx = akmQx - bjmQx; - bkpQx = bkmQx + ajmQx; - akmQx += bjmQx; - bkmQx -= ajmQx; - } - RexQx[k1] = akpQx; - RexQx[k2] = ajpQx; - RexQx[k3] = akmQx; - ImxQx[k1] = bkpQx; - ImxQx[k2] = bjpQx; - ImxQx[k3] = bkmQx; - } - // Complexity: 60*45 = 2700 for 16x16 muls, but 60*45 = 2700 cycles for 16x32 muls - - /* permute the results to normal order */ - for (ii=0; ii<240; ii++) { - ReDATAQx[ii]=RexQx[ii]; - ImDATAQx[ii]=ImxQx[ii]; - } - // Complexity: 240*2=480 cycles - - for (ii=0; ii<240; ii++) { - RexQx[ii]=ReDATAQx[kSortTabFft[ii]]; - ImxQx[ii]=ImDATAQx[kSortTabFft[ii]]; - } - // Complexity: 240*2*2=960 cycles - - // Total complexity: - // 16x16 16x32 - // Complexity: 10 10 - // Complexity: 99 171 - // Complexity: 2972 3932 - // Complexity: 2736 5776 - // Complexity: 5472 10080 - // Complexity: 3462 6342 - // Complexity: 2700 2700 - // Complexity: 480 480 - // Complexity: 960 960 - // ======================= - // 18891 30451 - // - // If this FFT is called 2 time each frame, i.e. 67 times per second, it will correspond to - // a C54 complexity of 67*18891/1000000 = 1.27 MIPS with 16x16-muls, and 67*30451/1000000 = - // = 2.04 MIPS with 16x32-muls. Note that this routine somtimes is called 6 times during the - // encoding of a frame, i.e. the max complexity would be 7/2*1.27 = 4.4 MIPS for the 16x16 mul case. - - - return 0; -} diff --git a/modules/audio_coding/codecs/isac/fix/source/fft.h b/modules/audio_coding/codecs/isac/fix/source/fft.h deleted file mode 100644 index 4fe9b96be4..0000000000 --- a/modules/audio_coding/codecs/isac/fix/source/fft.h +++ /dev/null @@ -1,39 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/*--------------------------------*-C-*---------------------------------* - * File: - * fft.h - * ---------------------------------------------------------------------* - * Re[]: real value array - * Im[]: imaginary value array - * nTotal: total number of complex values - * nPass: number of elements involved in this pass of transform - * nSpan: nspan/nPass = number of bytes to increment pointer - * in Re[] and Im[] - * isign: exponent: +1 = forward -1 = reverse - * scaling: normalizing constant by which the final result is *divided* - * scaling == -1, normalize by total dimension of the transform - * scaling < -1, normalize by the square-root of the total dimension - * - * ---------------------------------------------------------------------- - * See the comments in the code for correct usage! - */ - -#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_FFT_H_ -#define MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_FFT_H_ - -#include "modules/audio_coding/codecs/isac/fix/source/structs.h" - -int16_t WebRtcIsacfix_FftRadix16Fastest(int16_t RexQx[], - int16_t ImxQx[], - int16_t iSign); - -#endif /* MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_FFT_H_ */ diff --git a/modules/audio_coding/codecs/isac/fix/source/filterbank_internal.h b/modules/audio_coding/codecs/isac/fix/source/filterbank_internal.h deleted file mode 100644 index f741e6f677..0000000000 --- a/modules/audio_coding/codecs/isac/fix/source/filterbank_internal.h +++ /dev/null @@ -1,87 +0,0 @@ -/* - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_FILTERBANK_INTERNAL_H_ -#define MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_FILTERBANK_INTERNAL_H_ - -#include - -#if defined(__cplusplus) || defined(c_plusplus) -extern "C" { -#endif - -/* Arguments: - * io: Input/output, in Q0. - * len: Input, sample length. - * coefficient: Input. - * state: Input/output, filter state, in Q4. - */ -typedef void (*HighpassFilterFixDec32)(int16_t* io, - int16_t len, - const int16_t* coefficient, - int32_t* state); -extern HighpassFilterFixDec32 WebRtcIsacfix_HighpassFilterFixDec32; - -void WebRtcIsacfix_HighpassFilterFixDec32C(int16_t* io, - int16_t len, - const int16_t* coefficient, - int32_t* state); - -#if defined(MIPS_DSP_R1_LE) -void WebRtcIsacfix_HighpassFilterFixDec32MIPS(int16_t* io, - int16_t len, - const int16_t* coefficient, - int32_t* state); -#endif - -typedef void (*AllpassFilter2FixDec16)( - int16_t* data_ch1, // Input and output in channel 1, in Q0 - int16_t* data_ch2, // Input and output in channel 2, in Q0 - const int16_t* factor_ch1, // Scaling factor for channel 1, in Q15 - const int16_t* factor_ch2, // Scaling factor for channel 2, in Q15 - int length, // Length of the data buffers - int32_t* filter_state_ch1, // Filter state for channel 1, in Q16 - int32_t* filter_state_ch2); // Filter state for channel 2, in Q16 -extern AllpassFilter2FixDec16 WebRtcIsacfix_AllpassFilter2FixDec16; - -void WebRtcIsacfix_AllpassFilter2FixDec16C(int16_t* data_ch1, - int16_t* data_ch2, - const int16_t* factor_ch1, - const int16_t* factor_ch2, - int length, - int32_t* filter_state_ch1, - int32_t* filter_state_ch2); - -#if defined(WEBRTC_HAS_NEON) -void WebRtcIsacfix_AllpassFilter2FixDec16Neon(int16_t* data_ch1, - int16_t* data_ch2, - const int16_t* factor_ch1, - const int16_t* factor_ch2, - int length, - int32_t* filter_state_ch1, - int32_t* filter_state_ch2); -#endif - -#if defined(MIPS_DSP_R1_LE) -void WebRtcIsacfix_AllpassFilter2FixDec16MIPS(int16_t* data_ch1, - int16_t* data_ch2, - const int16_t* factor_ch1, - const int16_t* factor_ch2, - int length, - int32_t* filter_state_ch1, - int32_t* filter_state_ch2); -#endif - -#if defined(__cplusplus) || defined(c_plusplus) -} -#endif - -#endif -/* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_FILTERBANK_INTERNAL_H_ */ diff --git a/modules/audio_coding/codecs/isac/fix/source/filterbank_tables.c b/modules/audio_coding/codecs/isac/fix/source/filterbank_tables.c deleted file mode 100644 index f2dec79c2d..0000000000 --- a/modules/audio_coding/codecs/isac/fix/source/filterbank_tables.c +++ /dev/null @@ -1,62 +0,0 @@ -/* - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/* - * filterbank_tables.c - * - * This file contains variables that are used in - * filterbanks.c - * - */ - -#include "modules/audio_coding/codecs/isac/fix/source/filterbank_tables.h" - -/* HPstcoeff_in_Q14 = {a1, a2, b1 - b0 * a1, b2 - b0 * a2}; - * In float, they are: {-1.94895953203325f, 0.94984516000000f, - * -0.05101826139794f, 0.05015484000000f}; - */ -const int16_t WebRtcIsacfix_kHpStCoeffInQ30[8] = { - 16189, -31932, /* Q30 lo/hi pair */ - 17243, 15562, /* Q30 lo/hi pair */ - -17186, -26748, /* Q35 lo/hi pair */ - -27476, 26296 /* Q35 lo/hi pair */ -}; - -/* HPstcoeff_out_1_Q14 = {a1, a2, b1 - b0 * a1, b2 - b0 * a2}; - * In float, they are: {-1.99701049409000f, 0.99714204490000f, - * 0.01701049409000f, -0.01704204490000f}; - */ -const int16_t WebRtcIsacfix_kHPStCoeffOut1Q30[8] = { - -1306, -32719, /* Q30 lo/hi pair */ - 11486, 16337, /* Q30 lo/hi pair */ - 26078, 8918, /* Q35 lo/hi pair */ - 3956, -8935 /* Q35 lo/hi pair */ -}; - -/* HPstcoeff_out_2_Q14 = {a1, a2, b1 - b0 * a1, b2 - b0 * a2}; - * In float, they are: {-1.98645294509837f, 0.98672435560000f, - * 0.00645294509837f, -0.00662435560000f}; - */ -const int16_t WebRtcIsacfix_kHPStCoeffOut2Q30[8] = { - -2953, -32546, /* Q30 lo/hi pair */ - 32233, 16166, /* Q30 lo/hi pair */ - 13217, 3383, /* Q35 lo/hi pair */ - -4597, -3473 /* Q35 lo/hi pair */ -}; - -/* The upper channel all-pass filter factors */ -const int16_t WebRtcIsacfix_kUpperApFactorsQ15[2] = { - 1137, 12537 -}; - -/* The lower channel all-pass filter factors */ -const int16_t WebRtcIsacfix_kLowerApFactorsQ15[2] = { - 5059, 24379 -}; diff --git a/modules/audio_coding/codecs/isac/fix/source/filterbank_tables.h b/modules/audio_coding/codecs/isac/fix/source/filterbank_tables.h deleted file mode 100644 index 01e5a7ba85..0000000000 --- a/modules/audio_coding/codecs/isac/fix/source/filterbank_tables.h +++ /dev/null @@ -1,52 +0,0 @@ -/* - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/* - * filterbank_tables.h - * - * Header file for variables that are defined in - * filterbank_tables.c. - * - */ - -#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_FILTERBANK_TABLES_H_ -#define MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_FILTERBANK_TABLES_H_ - -#include - -#if defined(__cplusplus) || defined(c_plusplus) -extern "C" { -#endif - -/********************* Coefficient Tables ************************/ - -/* HPstcoeff_in_Q14 = {a1, a2, b1 - b0 * a1, b2 - b0 * a2}; */ -/* [Q30lo Q30hi Q30lo Q30hi Q35lo Q35hi Q35lo Q35hi] */ -extern const int16_t WebRtcIsacfix_kHpStCoeffInQ30[8]; - -/* HPstcoeff_out_1_Q14 = {a1, a2, b1 - b0 * a1, b2 - b0 * a2}; */ -/* [Q30lo Q30hi Q30lo Q30hi Q35lo Q35hi Q35lo Q35hi] */ -extern const int16_t WebRtcIsacfix_kHPStCoeffOut1Q30[8]; - -/* HPstcoeff_out_2_Q14 = {a1, a2, b1 - b0 * a1, b2 - b0 * a2}; */ -/* [Q30lo Q30hi Q30lo Q30hi Q35lo Q35hi Q35lo Q35hi] */ -extern const int16_t WebRtcIsacfix_kHPStCoeffOut2Q30[8]; - -/* The upper channel all-pass filter factors */ -extern const int16_t WebRtcIsacfix_kUpperApFactorsQ15[2]; - -/* The lower channel all-pass filter factors */ -extern const int16_t WebRtcIsacfix_kLowerApFactorsQ15[2]; - -#if defined(__cplusplus) || defined(c_plusplus) -} -#endif - -#endif /* MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_FILTERBANK_TABLES_H_ */ diff --git a/modules/audio_coding/codecs/isac/fix/source/filterbanks.c b/modules/audio_coding/codecs/isac/fix/source/filterbanks.c deleted file mode 100644 index 57b3e70b89..0000000000 --- a/modules/audio_coding/codecs/isac/fix/source/filterbanks.c +++ /dev/null @@ -1,297 +0,0 @@ -/* - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/* - * filterbanks.c - * - * This file contains function - * WebRtcIsacfix_SplitAndFilter, and WebRtcIsacfix_FilterAndCombine - * which implement filterbanks that produce decimated lowpass and - * highpass versions of a signal, and performs reconstruction. - * - */ - -#include "modules/audio_coding/codecs/isac/fix/source/filterbank_internal.h" - -#include "modules/audio_coding/codecs/isac/fix/source/codec.h" -#include "modules/audio_coding/codecs/isac/fix/source/filterbank_tables.h" -#include "modules/audio_coding/codecs/isac/fix/source/settings.h" -#include "rtc_base/checks.h" - -// Declare a function pointer. -AllpassFilter2FixDec16 WebRtcIsacfix_AllpassFilter2FixDec16; - -void WebRtcIsacfix_AllpassFilter2FixDec16C( - int16_t *data_ch1, // Input and output in channel 1, in Q0 - int16_t *data_ch2, // Input and output in channel 2, in Q0 - const int16_t *factor_ch1, // Scaling factor for channel 1, in Q15 - const int16_t *factor_ch2, // Scaling factor for channel 2, in Q15 - const int length, // Length of the data buffers - int32_t *filter_state_ch1, // Filter state for channel 1, in Q16 - int32_t *filter_state_ch2) { // Filter state for channel 2, in Q16 - int n = 0; - int32_t state0_ch1 = filter_state_ch1[0], state1_ch1 = filter_state_ch1[1]; - int32_t state0_ch2 = filter_state_ch2[0], state1_ch2 = filter_state_ch2[1]; - int16_t in_out = 0; - int32_t a = 0, b = 0; - - // Assembly file assumption. - RTC_DCHECK_EQ(0, length % 2); - - for (n = 0; n < length; n++) { - // Process channel 1: - in_out = data_ch1[n]; - a = factor_ch1[0] * in_out; // Q15 * Q0 = Q15 - a *= 1 << 1; // Q15 -> Q16 - b = WebRtcSpl_AddSatW32(a, state0_ch1); - a = -factor_ch1[0] * (int16_t)(b >> 16); // Q15 - state0_ch1 = - WebRtcSpl_AddSatW32(a * (1 << 1), (int32_t)in_out * (1 << 16)); // Q16 - in_out = (int16_t) (b >> 16); // Save as Q0 - - a = factor_ch1[1] * in_out; // Q15 * Q0 = Q15 - a *= 1 << 1; // Q15 -> Q16 - b = WebRtcSpl_AddSatW32(a, state1_ch1); // Q16 - a = -factor_ch1[1] * (int16_t)(b >> 16); // Q15 - state1_ch1 = - WebRtcSpl_AddSatW32(a * (1 << 1), (int32_t)in_out * (1 << 16)); // Q16 - data_ch1[n] = (int16_t) (b >> 16); // Save as Q0 - - // Process channel 2: - in_out = data_ch2[n]; - a = factor_ch2[0] * in_out; // Q15 * Q0 = Q15 - a *= 1 << 1; // Q15 -> Q16 - b = WebRtcSpl_AddSatW32(a, state0_ch2); // Q16 - a = -factor_ch2[0] * (int16_t)(b >> 16); // Q15 - state0_ch2 = - WebRtcSpl_AddSatW32(a * (1 << 1), (int32_t)in_out * (1 << 16)); // Q16 - in_out = (int16_t) (b >> 16); // Save as Q0 - - a = factor_ch2[1] * in_out; // Q15 * Q0 = Q15 - a *= (1 << 1); // Q15 -> Q16 - b = WebRtcSpl_AddSatW32(a, state1_ch2); // Q16 - a = -factor_ch2[1] * (int16_t)(b >> 16); // Q15 - state1_ch2 = - WebRtcSpl_AddSatW32(a * (1 << 1), (int32_t)in_out * (1 << 16)); // Q16 - data_ch2[n] = (int16_t) (b >> 16); // Save as Q0 - } - - filter_state_ch1[0] = state0_ch1; - filter_state_ch1[1] = state1_ch1; - filter_state_ch2[0] = state0_ch2; - filter_state_ch2[1] = state1_ch2; -} - -// Declare a function pointer. -HighpassFilterFixDec32 WebRtcIsacfix_HighpassFilterFixDec32; - -void WebRtcIsacfix_HighpassFilterFixDec32C(int16_t *io, - int16_t len, - const int16_t *coefficient, - int32_t *state) -{ - int k; - int32_t a1 = 0, b1 = 0, c = 0, in = 0; - int32_t a2 = 0, b2 = 0; - int32_t state0 = state[0]; - int32_t state1 = state[1]; - - for (k=0; k Q7 */ - a1 = WEBRTC_SPL_MUL_16_32_RSFT16(coefficient[5], state0) + - (WEBRTC_SPL_MUL_16_32_RSFT16(coefficient[4], state0) >> 16); - b1 = WEBRTC_SPL_MUL_16_32_RSFT16(coefficient[7], state1) + - (WEBRTC_SPL_MUL_16_32_RSFT16(coefficient[6], state1) >> 16); - - /* Q30 * Q4 = Q34 ; shift 32 bit => Q2 */ - a2 = WEBRTC_SPL_MUL_16_32_RSFT16(coefficient[1], state0) + - (WEBRTC_SPL_MUL_16_32_RSFT16(coefficient[0], state0) >> 16); - b2 = WEBRTC_SPL_MUL_16_32_RSFT16(coefficient[3], state1) + - (WEBRTC_SPL_MUL_16_32_RSFT16(coefficient[2], state1) >> 16); -#endif - - c = in + ((a1 + b1) >> 7); // Q0. - io[k] = (int16_t)WebRtcSpl_SatW32ToW16(c); // Write output as Q0. - - c = in * (1 << 2) - a2 - b2; // In Q2. - c = (int32_t)WEBRTC_SPL_SAT(536870911, c, -536870912); - - state1 = state0; - state0 = c * (1 << 2); // Write state as Q4 - } - state[0] = state0; - state[1] = state1; -} - - -void WebRtcIsacfix_SplitAndFilter1(int16_t *pin, - int16_t *LP16, - int16_t *HP16, - PreFiltBankstr *prefiltdata) -{ - /* Function WebRtcIsacfix_SplitAndFilter */ - /* This function creates low-pass and high-pass decimated versions of part of - the input signal, and part of the signal in the input 'lookahead buffer'. */ - - int k; - - int16_t tempin_ch1[FRAMESAMPLES/2 + QLOOKAHEAD]; - int16_t tempin_ch2[FRAMESAMPLES/2 + QLOOKAHEAD]; - int32_t tmpState_ch1[2 * (QORDER-1)]; /* 4 */ - int32_t tmpState_ch2[2 * (QORDER-1)]; /* 4 */ - - /* High pass filter */ - WebRtcIsacfix_HighpassFilterFixDec32(pin, FRAMESAMPLES, WebRtcIsacfix_kHpStCoeffInQ30, prefiltdata->HPstates_fix); - - - /* First Channel */ - for (k=0;kINLABUF1_fix[k]; - prefiltdata->INLABUF1_fix[k] = pin[FRAMESAMPLES + 1 - 2 * (QLOOKAHEAD - k)]; - } - - /* Second Channel. This is exactly like the first channel, except that the - even samples are now filtered instead (lower channel). */ - for (k=0;kINLABUF2_fix[k]; - prefiltdata->INLABUF2_fix[k] = pin[FRAMESAMPLES - 2 * (QLOOKAHEAD - k)]; - } - - - /*obtain polyphase components by forward all-pass filtering through each channel */ - /* The all pass filtering automatically updates the filter states which are exported in the - prefiltdata structure */ - WebRtcIsacfix_AllpassFilter2FixDec16(tempin_ch1, - tempin_ch2, - WebRtcIsacfix_kUpperApFactorsQ15, - WebRtcIsacfix_kLowerApFactorsQ15, - FRAMESAMPLES/2, - prefiltdata->INSTAT1_fix, - prefiltdata->INSTAT2_fix); - - for (k = 0; k < 2 * (QORDER - 1); k++) { - tmpState_ch1[k] = prefiltdata->INSTAT1_fix[k]; - tmpState_ch2[k] = prefiltdata->INSTAT2_fix[k]; - } - WebRtcIsacfix_AllpassFilter2FixDec16(tempin_ch1 + FRAMESAMPLES/2, - tempin_ch2 + FRAMESAMPLES/2, - WebRtcIsacfix_kUpperApFactorsQ15, - WebRtcIsacfix_kLowerApFactorsQ15, - QLOOKAHEAD, - tmpState_ch1, - tmpState_ch2); - - /* Now Construct low-pass and high-pass signals as combinations of polyphase components */ - for (k=0; k Q0 - tmp2 = (int32_t)tempin_ch2[k]; // Q0 -> Q0 - tmp3 = (tmp1 + tmp2) >> 1; /* Low pass signal. */ - LP16[k] = (int16_t)WebRtcSpl_SatW32ToW16(tmp3); /*low pass */ - tmp3 = (tmp1 - tmp2) >> 1; /* High pass signal. */ - HP16[k] = (int16_t)WebRtcSpl_SatW32ToW16(tmp3); /*high pass */ - } - -}/*end of WebRtcIsacfix_SplitAndFilter */ - - - -////////////////////////////////////////////////////////// -////////// Combining -/* Function WebRtcIsacfix_FilterAndCombine */ -/* This is a decoder function that takes the decimated - length FRAMESAMPLES/2 input low-pass and - high-pass signals and creates a reconstructed fullband - output signal of length FRAMESAMPLES. WebRtcIsacfix_FilterAndCombine - is the sibling function of WebRtcIsacfix_SplitAndFilter */ -/* INPUTS: - inLP: a length FRAMESAMPLES/2 array of input low-pass - samples. - inHP: a length FRAMESAMPLES/2 array of input high-pass - samples. - postfiltdata: input data structure containing the filterbank - states from the previous decoding iteration. - OUTPUTS: - Out: a length FRAMESAMPLES array of output reconstructed - samples (fullband) based on the input low-pass and - high-pass signals. - postfiltdata: the input data structure containing the filterbank - states is updated for the next decoding iteration */ -void WebRtcIsacfix_FilterAndCombine1(int16_t *tempin_ch1, - int16_t *tempin_ch2, - int16_t *out16, - PostFiltBankstr *postfiltdata) -{ - int k; - int16_t in[FRAMESAMPLES]; - - /* all-pass filter the new upper and lower channel signal. - For upper channel, use the all-pass filter factors that were used as a - lower channel at the encoding side. So at the decoder, the corresponding - all-pass filter factors for each channel are swapped. - For lower channel signal, since all-pass filter factors at the decoder are - swapped from the ones at the encoder, the 'upper' channel all-pass filter - factors (kUpperApFactors) are used to filter this new lower channel signal. - */ - WebRtcIsacfix_AllpassFilter2FixDec16(tempin_ch1, - tempin_ch2, - WebRtcIsacfix_kLowerApFactorsQ15, - WebRtcIsacfix_kUpperApFactorsQ15, - FRAMESAMPLES/2, - postfiltdata->STATE_0_UPPER_fix, - postfiltdata->STATE_0_LOWER_fix); - - /* Merge outputs to form the full length output signal.*/ - for (k=0;kHPstates1_fix); - WebRtcIsacfix_HighpassFilterFixDec32(in, FRAMESAMPLES, WebRtcIsacfix_kHPStCoeffOut2Q30, postfiltdata->HPstates2_fix); - - for (k=0;k - -#include "modules/audio_coding/codecs/isac/fix/source/filterbank_internal.h" -#include "rtc_base/checks.h" - -void WebRtcIsacfix_AllpassFilter2FixDec16Neon( - int16_t* data_ch1, // Input and output in channel 1, in Q0 - int16_t* data_ch2, // Input and output in channel 2, in Q0 - const int16_t* factor_ch1, // Scaling factor for channel 1, in Q15 - const int16_t* factor_ch2, // Scaling factor for channel 2, in Q15 - const int length, // Length of the data buffers - int32_t* filter_state_ch1, // Filter state for channel 1, in Q16 - int32_t* filter_state_ch2) { // Filter state for channel 2, in Q16 - RTC_DCHECK_EQ(0, length % 2); - int n = 0; - int16x4_t factorv; - int16x4_t datav; - int32x4_t statev; - - // Load factor_ch1 and factor_ch2. - factorv = vld1_dup_s16(factor_ch1); - factorv = vld1_lane_s16(factor_ch1 + 1, factorv, 1); - factorv = vld1_lane_s16(factor_ch2, factorv, 2); - factorv = vld1_lane_s16(factor_ch2 + 1, factorv, 3); - - // Load filter_state_ch1[0] and filter_state_ch2[0]. - statev = vld1q_dup_s32(filter_state_ch1); - statev = vld1q_lane_s32(filter_state_ch2, statev, 2); - - // Loop unrolling preprocessing. - int32x4_t a; - int16x4_t tmp1, tmp2; - - // Load data_ch1[0] and data_ch2[0]. - datav = vld1_dup_s16(data_ch1); - datav = vld1_lane_s16(data_ch2, datav, 2); - - a = vqdmlal_s16(statev, datav, factorv); - tmp1 = vshrn_n_s32(a, 16); - - // Update filter_state_ch1[0] and filter_state_ch2[0]. - statev = vqdmlsl_s16(vshll_n_s16(datav, 16), tmp1, factorv); - - // Load filter_state_ch1[1] and filter_state_ch2[1]. - statev = vld1q_lane_s32(filter_state_ch1 + 1, statev, 1); - statev = vld1q_lane_s32(filter_state_ch2 + 1, statev, 3); - - // Load data_ch1[1] and data_ch2[1]. - tmp1 = vld1_lane_s16(data_ch1 + 1, tmp1, 1); - tmp1 = vld1_lane_s16(data_ch2 + 1, tmp1, 3); - datav = vrev32_s16(tmp1); - - // Loop unrolling processing. - for (n = 0; n < length - 2; n += 2) { - a = vqdmlal_s16(statev, datav, factorv); - tmp1 = vshrn_n_s32(a, 16); - // Store data_ch1[n] and data_ch2[n]. - vst1_lane_s16(data_ch1 + n, tmp1, 1); - vst1_lane_s16(data_ch2 + n, tmp1, 3); - - // Update filter_state_ch1[0], filter_state_ch1[1] - // and filter_state_ch2[0], filter_state_ch2[1]. - statev = vqdmlsl_s16(vshll_n_s16(datav, 16), tmp1, factorv); - - // Load data_ch1[n + 2] and data_ch2[n + 2]. - tmp1 = vld1_lane_s16(data_ch1 + n + 2, tmp1, 1); - tmp1 = vld1_lane_s16(data_ch2 + n + 2, tmp1, 3); - datav = vrev32_s16(tmp1); - - a = vqdmlal_s16(statev, datav, factorv); - tmp2 = vshrn_n_s32(a, 16); - // Store data_ch1[n + 1] and data_ch2[n + 1]. - vst1_lane_s16(data_ch1 + n + 1, tmp2, 1); - vst1_lane_s16(data_ch2 + n + 1, tmp2, 3); - - // Update filter_state_ch1[0], filter_state_ch1[1] - // and filter_state_ch2[0], filter_state_ch2[1]. - statev = vqdmlsl_s16(vshll_n_s16(datav, 16), tmp2, factorv); - - // Load data_ch1[n + 3] and data_ch2[n + 3]. - tmp2 = vld1_lane_s16(data_ch1 + n + 3, tmp2, 1); - tmp2 = vld1_lane_s16(data_ch2 + n + 3, tmp2, 3); - datav = vrev32_s16(tmp2); - } - - // Loop unrolling post-processing. - a = vqdmlal_s16(statev, datav, factorv); - tmp1 = vshrn_n_s32(a, 16); - // Store data_ch1[n] and data_ch2[n]. - vst1_lane_s16(data_ch1 + n, tmp1, 1); - vst1_lane_s16(data_ch2 + n, tmp1, 3); - - // Update filter_state_ch1[0], filter_state_ch1[1] - // and filter_state_ch2[0], filter_state_ch2[1]. - statev = vqdmlsl_s16(vshll_n_s16(datav, 16), tmp1, factorv); - // Store filter_state_ch1[0] and filter_state_ch2[0]. - vst1q_lane_s32(filter_state_ch1, statev, 0); - vst1q_lane_s32(filter_state_ch2, statev, 2); - - datav = vrev32_s16(tmp1); - a = vqdmlal_s16(statev, datav, factorv); - tmp2 = vshrn_n_s32(a, 16); - // Store data_ch1[n + 1] and data_ch2[n + 1]. - vst1_lane_s16(data_ch1 + n + 1, tmp2, 1); - vst1_lane_s16(data_ch2 + n + 1, tmp2, 3); - - // Update filter_state_ch1[1] and filter_state_ch2[1]. - statev = vqdmlsl_s16(vshll_n_s16(datav, 16), tmp2, factorv); - // Store filter_state_ch1[1] and filter_state_ch2[1]. - vst1q_lane_s32(filter_state_ch1 + 1, statev, 1); - vst1q_lane_s32(filter_state_ch2 + 1, statev, 3); -} - -// This function is the prototype for above neon optimized function. -//void AllpassFilter2FixDec16BothChannels( -// int16_t *data_ch1, // Input and output in channel 1, in Q0 -// int16_t *data_ch2, // Input and output in channel 2, in Q0 -// const int16_t *factor_ch1, // Scaling factor for channel 1, in Q15 -// const int16_t *factor_ch2, // Scaling factor for channel 2, in Q15 -// const int length, // Length of the data buffers -// int32_t *filter_state_ch1, // Filter state for channel 1, in Q16 -// int32_t *filter_state_ch2) { // Filter state for channel 2, in Q16 -// int n = 0; -// int32_t state0_ch1 = filter_state_ch1[0], state1_ch1 = filter_state_ch1[1]; -// int32_t state0_ch2 = filter_state_ch2[0], state1_ch2 = filter_state_ch2[1]; -// int16_t sample0_ch1 = 0, sample0_ch2 = 0; -// int16_t sample1_ch1 = 0, sample1_ch2 = 0; -// int32_t a0_ch1 = 0, a0_ch2 = 0; -// int32_t b0_ch1 = 0, b0_ch2 = 0; -// -// int32_t a1_ch1 = 0, a1_ch2 = 0; -// int32_t b1_ch1 = 0, b1_ch2 = 0; -// int32_t b2_ch1 = 0, b2_ch2 = 0; -// -// // Loop unrolling preprocessing. -// -// sample0_ch1 = data_ch1[n]; -// sample0_ch2 = data_ch2[n]; -// -// a0_ch1 = (factor_ch1[0] * sample0_ch1) << 1; -// a0_ch2 = (factor_ch2[0] * sample0_ch2) << 1; -// -// b0_ch1 = WebRtcSpl_AddSatW32(a0_ch1, state0_ch1); -// b0_ch2 = WebRtcSpl_AddSatW32(a0_ch2, state0_ch2); //Q16+Q16=Q16 -// -// a0_ch1 = -factor_ch1[0] * (int16_t)(b0_ch1 >> 16); -// a0_ch2 = -factor_ch2[0] * (int16_t)(b0_ch2 >> 16); -// -// state0_ch1 = WebRtcSpl_AddSatW32(a0_ch1 <<1, (uint32_t)sample0_ch1 << 16); -// state0_ch2 = WebRtcSpl_AddSatW32(a0_ch2 <<1, (uint32_t)sample0_ch2 << 16); -// -// sample1_ch1 = data_ch1[n + 1]; -// sample0_ch1 = (int16_t) (b0_ch1 >> 16); //Save as Q0 -// sample1_ch2 = data_ch2[n + 1]; -// sample0_ch2 = (int16_t) (b0_ch2 >> 16); //Save as Q0 -// -// -// for (n = 0; n < length - 2; n += 2) { -// a1_ch1 = (factor_ch1[0] * sample1_ch1) << 1; -// a0_ch1 = (factor_ch1[1] * sample0_ch1) << 1; -// a1_ch2 = (factor_ch2[0] * sample1_ch2) << 1; -// a0_ch2 = (factor_ch2[1] * sample0_ch2) << 1; -// -// b1_ch1 = WebRtcSpl_AddSatW32(a1_ch1, state0_ch1); -// b0_ch1 = WebRtcSpl_AddSatW32(a0_ch1, state1_ch1); //Q16+Q16=Q16 -// b1_ch2 = WebRtcSpl_AddSatW32(a1_ch2, state0_ch2); //Q16+Q16=Q16 -// b0_ch2 = WebRtcSpl_AddSatW32(a0_ch2, state1_ch2); //Q16+Q16=Q16 -// -// a1_ch1 = -factor_ch1[0] * (int16_t)(b1_ch1 >> 16); -// a0_ch1 = -factor_ch1[1] * (int16_t)(b0_ch1 >> 16); -// a1_ch2 = -factor_ch2[0] * (int16_t)(b1_ch2 >> 16); -// a0_ch2 = -factor_ch2[1] * (int16_t)(b0_ch2 >> 16); -// -// state0_ch1 = WebRtcSpl_AddSatW32(a1_ch1<<1, (uint32_t)sample1_ch1 <<16); -// state1_ch1 = WebRtcSpl_AddSatW32(a0_ch1<<1, (uint32_t)sample0_ch1 <<16); -// state0_ch2 = WebRtcSpl_AddSatW32(a1_ch2<<1, (uint32_t)sample1_ch2 <<16); -// state1_ch2 = WebRtcSpl_AddSatW32(a0_ch2<<1, (uint32_t)sample0_ch2 <<16); -// -// sample0_ch1 = data_ch1[n + 2]; -// sample1_ch1 = (int16_t) (b1_ch1 >> 16); //Save as Q0 -// sample0_ch2 = data_ch2[n + 2]; -// sample1_ch2 = (int16_t) (b1_ch2 >> 16); //Save as Q0 -// -// a0_ch1 = (factor_ch1[0] * sample0_ch1) << 1; -// a1_ch1 = (factor_ch1[1] * sample1_ch1) << 1; -// a0_ch2 = (factor_ch2[0] * sample0_ch2) << 1; -// a1_ch2 = (factor_ch2[1] * sample1_ch2) << 1; -// -// b2_ch1 = WebRtcSpl_AddSatW32(a0_ch1, state0_ch1); -// b1_ch1 = WebRtcSpl_AddSatW32(a1_ch1, state1_ch1); //Q16+Q16=Q16 -// b2_ch2 = WebRtcSpl_AddSatW32(a0_ch2, state0_ch2); //Q16+Q16=Q16 -// b1_ch2 = WebRtcSpl_AddSatW32(a1_ch2, state1_ch2); //Q16+Q16=Q16 -// -// a0_ch1 = -factor_ch1[0] * (int16_t)(b2_ch1 >> 16); -// a1_ch1 = -factor_ch1[1] * (int16_t)(b1_ch1 >> 16); -// a0_ch2 = -factor_ch2[0] * (int16_t)(b2_ch2 >> 16); -// a1_ch2 = -factor_ch2[1] * (int16_t)(b1_ch2 >> 16); -// -// state0_ch1 = WebRtcSpl_AddSatW32(a0_ch1<<1, (uint32_t)sample0_ch1<<16); -// state1_ch1 = WebRtcSpl_AddSatW32(a1_ch1<<1, (uint32_t)sample1_ch1<<16); -// state0_ch2 = WebRtcSpl_AddSatW32(a0_ch2<<1, (uint32_t)sample0_ch2<<16); -// state1_ch2 = WebRtcSpl_AddSatW32(a1_ch2<<1, (uint32_t)sample1_ch2<<16); -// -// -// sample1_ch1 = data_ch1[n + 3]; -// sample0_ch1 = (int16_t) (b2_ch1 >> 16); //Save as Q0 -// sample1_ch2 = data_ch2[n + 3]; -// sample0_ch2 = (int16_t) (b2_ch2 >> 16); //Save as Q0 -// -// data_ch1[n] = (int16_t) (b0_ch1 >> 16); //Save as Q0 -// data_ch1[n + 1] = (int16_t) (b1_ch1 >> 16); //Save as Q0 -// data_ch2[n] = (int16_t) (b0_ch2 >> 16); -// data_ch2[n + 1] = (int16_t) (b1_ch2 >> 16); -// } -// -// // Loop unrolling post-processing. -// -// a1_ch1 = (factor_ch1[0] * sample1_ch1) << 1; -// a0_ch1 = (factor_ch1[1] * sample0_ch1) << 1; -// a1_ch2 = (factor_ch2[0] * sample1_ch2) << 1; -// a0_ch2 = (factor_ch2[1] * sample0_ch2) << 1; -// -// b1_ch1 = WebRtcSpl_AddSatW32(a1_ch1, state0_ch1); -// b0_ch1 = WebRtcSpl_AddSatW32(a0_ch1, state1_ch1); -// b1_ch2 = WebRtcSpl_AddSatW32(a1_ch2, state0_ch2); -// b0_ch2 = WebRtcSpl_AddSatW32(a0_ch2, state1_ch2); -// -// a1_ch1 = -factor_ch1[0] * (int16_t)(b1_ch1 >> 16); -// a0_ch1 = -factor_ch1[1] * (int16_t)(b0_ch1 >> 16); -// a1_ch2 = -factor_ch2[0] * (int16_t)(b1_ch2 >> 16); -// a0_ch2 = -factor_ch2[1] * (int16_t)(b0_ch2 >> 16); -// -// state0_ch1 = WebRtcSpl_AddSatW32(a1_ch1<<1, (uint32_t)sample1_ch1 << 16); -// state1_ch1 = WebRtcSpl_AddSatW32(a0_ch1<<1, (uint32_t)sample0_ch1 << 16); -// state0_ch2 = WebRtcSpl_AddSatW32(a1_ch2<<1, (uint32_t)sample1_ch2 << 16); -// state1_ch2 = WebRtcSpl_AddSatW32(a0_ch2<<1, (uint32_t)sample0_ch2 << 16); -// -// data_ch1[n] = (int16_t) (b0_ch1 >> 16); //Save as Q0 -// data_ch2[n] = (int16_t) (b0_ch2 >> 16); -// -// sample1_ch1 = (int16_t) (b1_ch1 >> 16); //Save as Q0 -// sample1_ch2 = (int16_t) (b1_ch2 >> 16); //Save as Q0 -// -// a1_ch1 = (factor_ch1[1] * sample1_ch1) << 1; -// a1_ch2 = (factor_ch2[1] * sample1_ch2) << 1; -// -// b1_ch1 = WebRtcSpl_AddSatW32(a1_ch1, state1_ch1); //Q16+Q16=Q16 -// b1_ch2 = WebRtcSpl_AddSatW32(a1_ch2, state1_ch2); //Q16+Q16=Q16 -// -// a1_ch1 = -factor_ch1[1] * (int16_t)(b1_ch1 >> 16); -// a1_ch2 = -factor_ch2[1] * (int16_t)(b1_ch2 >> 16); -// -// state1_ch1 = WebRtcSpl_AddSatW32(a1_ch1<<1, (uint32_t)sample1_ch1<<16); -// state1_ch2 = WebRtcSpl_AddSatW32(a1_ch2<<1, (uint32_t)sample1_ch2<<16); -// -// data_ch1[n + 1] = (int16_t) (b1_ch1 >> 16); //Save as Q0 -// data_ch2[n + 1] = (int16_t) (b1_ch2 >> 16); -// -// filter_state_ch1[0] = state0_ch1; -// filter_state_ch1[1] = state1_ch1; -// filter_state_ch2[0] = state0_ch2; -// filter_state_ch2[1] = state1_ch2; -//} diff --git a/modules/audio_coding/codecs/isac/fix/source/filterbanks_unittest.cc b/modules/audio_coding/codecs/isac/fix/source/filterbanks_unittest.cc deleted file mode 100644 index 4a3db2324a..0000000000 --- a/modules/audio_coding/codecs/isac/fix/source/filterbanks_unittest.cc +++ /dev/null @@ -1,103 +0,0 @@ -/* - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#include "common_audio/signal_processing/include/signal_processing_library.h" -#include "modules/audio_coding/codecs/isac/fix/source/filterbank_internal.h" -#include "modules/audio_coding/codecs/isac/fix/source/filterbank_tables.h" -#include "modules/audio_coding/codecs/isac/fix/source/settings.h" -#include "rtc_base/sanitizer.h" -#include "system_wrappers/include/cpu_features_wrapper.h" -#include "test/gtest.h" - -class FilterBanksTest : public ::testing::Test { - protected: - // Pass a function pointer to the Tester function. - void RTC_NO_SANITIZE("signed-integer-overflow") // bugs.webrtc.org/5513 - CalculateResidualEnergyTester( - AllpassFilter2FixDec16 AllpassFilter2FixDec16Function) { - const int kSamples = QLOOKAHEAD; - const int kState = 2; - int16_t data_ch1[kSamples] = {0}; - int16_t data_ch2[kSamples] = {0}; - int32_t state_ch1[kState] = {0}; - int32_t state_ch2[kState] = {0}; - const int32_t out_state_ch1[kState] = {-809122714, 1645972152}; - const int32_t out_state_ch2[kState] = {428019288, 1057309936}; - const int32_t out_data_ch1[kSamples] = { - 0, 0, 347, 10618, 16718, -7089, 32767, 16913, - 27042, 8377, -22973, -28372, -27603, -14804, 398, -25332, - -11200, 18044, 25223, -6839, 1116, -23984, 32717, 7364}; - const int32_t out_data_ch2[kSamples] = { - 0, 0, 3010, 22351, 21106, 16969, -2095, -664, - 3513, -30980, 32767, -23839, 13335, 20289, -6831, 339, - -17207, 32767, 4959, 6177, 32767, 16599, -4747, 20504}; - int sign = 1; - - for (int i = 0; i < kSamples; i++) { - sign *= -1; - data_ch1[i] = sign * WEBRTC_SPL_WORD32_MAX / (i * i + 1); - data_ch2[i] = sign * WEBRTC_SPL_WORD32_MIN / (i * i + 1); - // UBSan: -1 * -2147483648 cannot be represented in type 'int' - }; - - AllpassFilter2FixDec16Function( - data_ch1, data_ch2, WebRtcIsacfix_kUpperApFactorsQ15, - WebRtcIsacfix_kLowerApFactorsQ15, kSamples, state_ch1, state_ch2); - - for (int i = 0; i < kSamples; i++) { - EXPECT_EQ(out_data_ch1[i], data_ch1[i]); - EXPECT_EQ(out_data_ch2[i], data_ch2[i]); - } - for (int i = 0; i < kState; i++) { - EXPECT_EQ(out_state_ch1[i], state_ch1[i]); - EXPECT_EQ(out_state_ch2[i], state_ch2[i]); - } - } -}; - -TEST_F(FilterBanksTest, AllpassFilter2FixDec16Test) { - CalculateResidualEnergyTester(WebRtcIsacfix_AllpassFilter2FixDec16C); -#if defined(WEBRTC_HAS_NEON) - CalculateResidualEnergyTester(WebRtcIsacfix_AllpassFilter2FixDec16Neon); -#endif -} - -TEST_F(FilterBanksTest, HighpassFilterFixDec32Test) { - const int kSamples = 20; - int16_t in[kSamples]; - int32_t state[2] = {12345, 987654}; -#ifdef WEBRTC_ARCH_ARM_V7 - int32_t out[kSamples] = {-1040, -1035, -22875, -1397, -27604, 20018, 7917, - -1279, -8552, -14494, -7558, -23537, -27258, -30554, - -32768, -3432, -32768, 25215, -27536, 22436}; -#else - int32_t out[kSamples] = {-1040, -1035, -22875, -1397, -27604, 20017, 7915, - -1280, -8554, -14496, -7561, -23541, -27263, -30560, - -32768, -3441, -32768, 25203, -27550, 22419}; -#endif - HighpassFilterFixDec32 WebRtcIsacfix_HighpassFilterFixDec32; -#if defined(MIPS_DSP_R1_LE) - WebRtcIsacfix_HighpassFilterFixDec32 = - WebRtcIsacfix_HighpassFilterFixDec32MIPS; -#else - WebRtcIsacfix_HighpassFilterFixDec32 = WebRtcIsacfix_HighpassFilterFixDec32C; -#endif - - for (int i = 0; i < kSamples; i++) { - in[i] = WEBRTC_SPL_WORD32_MAX / (i + 1); - } - - WebRtcIsacfix_HighpassFilterFixDec32(in, kSamples, - WebRtcIsacfix_kHPStCoeffOut1Q30, state); - - for (int i = 0; i < kSamples; i++) { - EXPECT_EQ(out[i], in[i]); - } -} diff --git a/modules/audio_coding/codecs/isac/fix/source/filters.c b/modules/audio_coding/codecs/isac/fix/source/filters.c deleted file mode 100644 index 838ba4b3e8..0000000000 --- a/modules/audio_coding/codecs/isac/fix/source/filters.c +++ /dev/null @@ -1,112 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#include "rtc_base/checks.h" -#include "modules/audio_coding/codecs/isac/fix/source/codec.h" - -// Autocorrelation function in fixed point. -// NOTE! Different from SPLIB-version in how it scales the signal. -int WebRtcIsacfix_AutocorrC(int32_t* __restrict r, - const int16_t* __restrict x, - int16_t N, - int16_t order, - int16_t* __restrict scale) { - int i = 0; - int j = 0; - int16_t scaling = 0; - int32_t sum = 0; - uint32_t temp = 0; - int64_t prod = 0; - - // The ARM assembly code assumptoins. - RTC_DCHECK_EQ(0, N % 4); - RTC_DCHECK_GE(N, 8); - - // Calculate r[0]. - for (i = 0; i < N; i++) { - prod += x[i] * x[i]; - } - - // Calculate scaling (the value of shifting). - temp = (uint32_t)(prod >> 31); - if(temp == 0) { - scaling = 0; - } else { - scaling = 32 - WebRtcSpl_NormU32(temp); - } - r[0] = (int32_t)(prod >> scaling); - - // Perform the actual correlation calculation. - for (i = 1; i < order + 1; i++) { - prod = 0; - for (j = 0; j < N - i; j++) { - prod += x[j] * x[i + j]; - } - sum = (int32_t)(prod >> scaling); - r[i] = sum; - } - - *scale = scaling; - - return(order + 1); -} - -static const int32_t kApUpperQ15[ALLPASSSECTIONS] = { 1137, 12537 }; -static const int32_t kApLowerQ15[ALLPASSSECTIONS] = { 5059, 24379 }; - - -static void AllpassFilterForDec32(int16_t *InOut16, //Q0 - const int32_t *APSectionFactors, //Q15 - int16_t lengthInOut, - int32_t *FilterState) //Q16 -{ - int n, j; - int32_t a, b; - - for (j=0; j Q16 - b = WebRtcSpl_AddSatW32(a, FilterState[j]); //Q16+Q16=Q16 - // `a` in Q15 (Q0*Q31=Q31 shifted 16 gives Q15). - a = WEBRTC_SPL_MUL_16_32_RSFT16(b >> 16, -APSectionFactors[j]); - // FilterState[j]: Q15<<1 + Q0<<16 = Q16 + Q16 = Q16 - FilterState[j] = WebRtcSpl_AddSatW32(a << 1, (uint32_t)InOut16[n] << 16); - InOut16[n] = (int16_t)(b >> 16); // Save as Q0. - } - } -} - - - - -void WebRtcIsacfix_DecimateAllpass32(const int16_t *in, - int32_t *state_in, /* array of size: 2*ALLPASSSECTIONS+1 */ - int16_t N, /* number of input samples */ - int16_t *out) /* array of size N/2 */ -{ - int n; - int16_t data_vec[PITCH_FRAME_LEN]; - - /* copy input */ - memcpy(data_vec + 1, in, sizeof(int16_t) * (N - 1)); - - data_vec[0] = (int16_t)(state_in[2 * ALLPASSSECTIONS] >> 16); // z^-1 state. - state_in[2 * ALLPASSSECTIONS] = (uint32_t)in[N - 1] << 16; - - - - AllpassFilterForDec32(data_vec+1, kApUpperQ15, N, state_in); - AllpassFilterForDec32(data_vec, kApLowerQ15, N, state_in+ALLPASSSECTIONS); - - for (n=0;n> 3); - int count = (int)(N & 7); - // Declare temporary variables used as registry values. - int32_t r0, r1, r2, r3; -#if !defined(MIPS_DSP_R2_LE) - // For non-DSPR2 optimizations 4 more registers are used. - int32_t r4, r5, r6, r7; -#endif - - // Calculate r[0] and scaling needed. - __asm __volatile ( - ".set push \n\t" - ".set noreorder \n\t" - "mult $0, $0 \n\t" - // Loop is unrolled 8 times, set accumulator to zero in branch delay slot. - "beqz %[loop_size], 2f \n\t" - " mult $0, $0 \n\t" - "1: \n\t" - // Load 8 samples per loop iteration. -#if defined(MIPS_DSP_R2_LE) - "ulw %[r0], 0(%[in]) \n\t" - "ulw %[r1], 4(%[in]) \n\t" - "ulw %[r2], 8(%[in]) \n\t" - "ulw %[r3], 12(%[in]) \n\t" -#else - "lh %[r0], 0(%[in]) \n\t" - "lh %[r1], 2(%[in]) \n\t" - "lh %[r2], 4(%[in]) \n\t" - "lh %[r3], 6(%[in]) \n\t" - "lh %[r4], 8(%[in]) \n\t" - "lh %[r5], 10(%[in]) \n\t" - "lh %[r6], 12(%[in]) \n\t" - "lh %[r7], 14(%[in]) \n\t" -#endif - "addiu %[loop_size], %[loop_size], -1 \n\t" - // Multiply and accumulate. -#if defined(MIPS_DSP_R2_LE) - "dpa.w.ph $ac0, %[r0], %[r0] \n\t" - "dpa.w.ph $ac0, %[r1], %[r1] \n\t" - "dpa.w.ph $ac0, %[r2], %[r2] \n\t" - "dpa.w.ph $ac0, %[r3], %[r3] \n\t" -#else - "madd %[r0], %[r0] \n\t" - "madd %[r1], %[r1] \n\t" - "madd %[r2], %[r2] \n\t" - "madd %[r3], %[r3] \n\t" - "madd %[r4], %[r4] \n\t" - "madd %[r5], %[r5] \n\t" - "madd %[r6], %[r6] \n\t" - "madd %[r7], %[r7] \n\t" -#endif - "bnez %[loop_size], 1b \n\t" - " addiu %[in], %[in], 16 \n\t" - "2: \n\t" - "beqz %[count], 4f \n\t" -#if defined(MIPS_DSP_R1_LE) - " extr.w %[r0], $ac0, 31 \n\t" -#else - " mfhi %[r2] \n\t" -#endif - // Process remaining samples (if any). - "3: \n\t" - "lh %[r0], 0(%[in]) \n\t" - "addiu %[count], %[count], -1 \n\t" - "madd %[r0], %[r0] \n\t" - "bnez %[count], 3b \n\t" - " addiu %[in], %[in], 2 \n\t" -#if defined(MIPS_DSP_R1_LE) - "extr.w %[r0], $ac0, 31 \n\t" -#else - "mfhi %[r2] \n\t" -#endif - "4: \n\t" -#if !defined(MIPS_DSP_R1_LE) - "mflo %[r3] \n\t" - "sll %[r0], %[r2], 1 \n\t" - "srl %[r1], %[r3], 31 \n\t" - "addu %[r0], %[r0], %[r1] \n\t" -#endif - // Calculate scaling (the value of shifting). - "clz %[r1], %[r0] \n\t" - "addiu %[r1], %[r1], -32 \n\t" - "subu %[scaling], $0, %[r1] \n\t" - "slti %[r1], %[r0], 0x1 \n\t" - "movn %[scaling], $0, %[r1] \n\t" -#if defined(MIPS_DSP_R1_LE) - "extrv.w %[r0], $ac0, %[scaling] \n\t" - "mfhi %[r2], $ac0 \n\t" -#else - "addiu %[r1], %[scaling], -32 \n\t" - "subu %[r1], $0, %[r1] \n\t" - "sllv %[r1], %[r2], %[r1] \n\t" - "srlv %[r0], %[r3], %[scaling] \n\t" - "addu %[r0], %[r0], %[r1] \n\t" -#endif - "slti %[r1], %[scaling], 32 \n\t" - "movz %[r0], %[r2], %[r1] \n\t" - ".set pop \n\t" - : [loop_size] "+r" (loop_size), [in] "+r" (in), [r0] "=&r" (r0), - [r1] "=&r" (r1), [r2] "=&r" (r2), [r3] "=&r" (r3), -#if !defined(MIPS_DSP_R2_LE) - [r4] "=&r" (r4), [r5] "=&r" (r5), [r6] "=&r" (r6), [r7] "=&r" (r7), -#endif - [count] "+r" (count), [scaling] "=r" (scaling) - : [N] "r" (N) - : "memory", "hi", "lo" - ); - r[0] = r0; - - // Correlation calculation is divided in 3 cases depending on the scaling - // value (different accumulator manipulation needed). Three slightly different - // loops are written in order to avoid branches inside the loop. - if (scaling == 0) { - // In this case, the result will be in low part of the accumulator. - for (i = 1; i < order + 1; i++) { - in = (int16_t*)x; - int16_t* in1 = (int16_t*)x + i; - count = N - i; - loop_size = (count) >> 2; - __asm __volatile ( - ".set push \n\t" - ".set noreorder \n\t" - "mult $0, $0 \n\t" - "beqz %[loop_size], 2f \n\t" - " andi %[count], %[count], 0x3 \n\t" - // Loop processing 4 pairs of samples per iteration. - "1: \n\t" -#if defined(MIPS_DSP_R2_LE) - "ulw %[r0], 0(%[in]) \n\t" - "ulw %[r1], 0(%[in1]) \n\t" - "ulw %[r2], 4(%[in]) \n\t" - "ulw %[r3], 4(%[in1]) \n\t" -#else - "lh %[r0], 0(%[in]) \n\t" - "lh %[r1], 0(%[in1]) \n\t" - "lh %[r2], 2(%[in]) \n\t" - "lh %[r3], 2(%[in1]) \n\t" - "lh %[r4], 4(%[in]) \n\t" - "lh %[r5], 4(%[in1]) \n\t" - "lh %[r6], 6(%[in]) \n\t" - "lh %[r7], 6(%[in1]) \n\t" -#endif - "addiu %[loop_size], %[loop_size], -1 \n\t" -#if defined(MIPS_DSP_R2_LE) - "dpa.w.ph $ac0, %[r0], %[r1] \n\t" - "dpa.w.ph $ac0, %[r2], %[r3] \n\t" -#else - "madd %[r0], %[r1] \n\t" - "madd %[r2], %[r3] \n\t" - "madd %[r4], %[r5] \n\t" - "madd %[r6], %[r7] \n\t" -#endif - "addiu %[in], %[in], 8 \n\t" - "bnez %[loop_size], 1b \n\t" - " addiu %[in1], %[in1], 8 \n\t" - "2: \n\t" - "beqz %[count], 4f \n\t" - " mflo %[r0] \n\t" - // Process remaining samples (if any). - "3: \n\t" - "lh %[r0], 0(%[in]) \n\t" - "lh %[r1], 0(%[in1]) \n\t" - "addiu %[count], %[count], -1 \n\t" - "addiu %[in], %[in], 2 \n\t" - "madd %[r0], %[r1] \n\t" - "bnez %[count], 3b \n\t" - " addiu %[in1], %[in1], 2 \n\t" - "mflo %[r0] \n\t" - "4: \n\t" - ".set pop \n\t" - : [loop_size] "+r" (loop_size), [in] "+r" (in), [in1] "+r" (in1), -#if !defined(MIPS_DSP_R2_LE) - [r4] "=&r" (r4), [r5] "=&r" (r5), [r6] "=&r" (r6), [r7] "=&r" (r7), -#endif - [r0] "=&r" (r0), [r1] "=&r" (r1), [r2] "=&r" (r2), [r3] "=&r" (r3), - [count] "+r" (count) - : - : "memory", "hi", "lo" - ); - r[i] = r0; - } - } else if (scaling == 32) { - // In this case, the result will be high part of the accumulator. - for (i = 1; i < order + 1; i++) { - in = (int16_t*)x; - int16_t* in1 = (int16_t*)x + i; - count = N - i; - loop_size = (count) >> 2; - __asm __volatile ( - ".set push \n\t" - ".set noreorder \n\t" - "mult $0, $0 \n\t" - "beqz %[loop_size], 2f \n\t" - " andi %[count], %[count], 0x3 \n\t" - // Loop processing 4 pairs of samples per iteration. - "1: \n\t" -#if defined(MIPS_DSP_R2_LE) - "ulw %[r0], 0(%[in]) \n\t" - "ulw %[r1], 0(%[in1]) \n\t" - "ulw %[r2], 4(%[in]) \n\t" - "ulw %[r3], 4(%[in1]) \n\t" -#else - "lh %[r0], 0(%[in]) \n\t" - "lh %[r1], 0(%[in1]) \n\t" - "lh %[r2], 2(%[in]) \n\t" - "lh %[r3], 2(%[in1]) \n\t" - "lh %[r4], 4(%[in]) \n\t" - "lh %[r5], 4(%[in1]) \n\t" - "lh %[r6], 6(%[in]) \n\t" - "lh %[r7], 6(%[in1]) \n\t" -#endif - "addiu %[loop_size], %[loop_size], -1 \n\t" -#if defined(MIPS_DSP_R2_LE) - "dpa.w.ph $ac0, %[r0], %[r1] \n\t" - "dpa.w.ph $ac0, %[r2], %[r3] \n\t" -#else - "madd %[r0], %[r1] \n\t" - "madd %[r2], %[r3] \n\t" - "madd %[r4], %[r5] \n\t" - "madd %[r6], %[r7] \n\t" -#endif - "addiu %[in], %[in], 8 \n\t" - "bnez %[loop_size], 1b \n\t" - " addiu %[in1], %[in1], 8 \n\t" - "2: \n\t" - "beqz %[count], 4f \n\t" - " mfhi %[r0] \n\t" - // Process remaining samples (if any). - "3: \n\t" - "lh %[r0], 0(%[in]) \n\t" - "lh %[r1], 0(%[in1]) \n\t" - "addiu %[count], %[count], -1 \n\t" - "addiu %[in], %[in], 2 \n\t" - "madd %[r0], %[r1] \n\t" - "bnez %[count], 3b \n\t" - " addiu %[in1], %[in1], 2 \n\t" - "mfhi %[r0] \n\t" - "4: \n\t" - ".set pop \n\t" - : [loop_size] "+r" (loop_size), [in] "+r" (in), [in1] "+r" (in1), -#if !defined(MIPS_DSP_R2_LE) - [r4] "=&r" (r4), [r5] "=&r" (r5), [r6] "=&r" (r6), [r7] "=&r" (r7), -#endif - [r0] "=&r" (r0), [r1] "=&r" (r1), [r2] "=&r" (r2), [r3] "=&r" (r3), - [count] "+r" (count) - : - : "memory", "hi", "lo" - ); - r[i] = r0; - } - } else { - // In this case, the result is obtained by combining low and high parts - // of the accumulator. -#if !defined(MIPS_DSP_R1_LE) - int32_t tmp_shift = 32 - scaling; -#endif - for (i = 1; i < order + 1; i++) { - in = (int16_t*)x; - int16_t* in1 = (int16_t*)x + i; - count = N - i; - loop_size = (count) >> 2; - __asm __volatile ( - ".set push \n\t" - ".set noreorder \n\t" - "mult $0, $0 \n\t" - "beqz %[loop_size], 2f \n\t" - " andi %[count], %[count], 0x3 \n\t" - "1: \n\t" -#if defined(MIPS_DSP_R2_LE) - "ulw %[r0], 0(%[in]) \n\t" - "ulw %[r1], 0(%[in1]) \n\t" - "ulw %[r2], 4(%[in]) \n\t" - "ulw %[r3], 4(%[in1]) \n\t" -#else - "lh %[r0], 0(%[in]) \n\t" - "lh %[r1], 0(%[in1]) \n\t" - "lh %[r2], 2(%[in]) \n\t" - "lh %[r3], 2(%[in1]) \n\t" - "lh %[r4], 4(%[in]) \n\t" - "lh %[r5], 4(%[in1]) \n\t" - "lh %[r6], 6(%[in]) \n\t" - "lh %[r7], 6(%[in1]) \n\t" -#endif - "addiu %[loop_size], %[loop_size], -1 \n\t" -#if defined(MIPS_DSP_R2_LE) - "dpa.w.ph $ac0, %[r0], %[r1] \n\t" - "dpa.w.ph $ac0, %[r2], %[r3] \n\t" -#else - "madd %[r0], %[r1] \n\t" - "madd %[r2], %[r3] \n\t" - "madd %[r4], %[r5] \n\t" - "madd %[r6], %[r7] \n\t" -#endif - "addiu %[in], %[in], 8 \n\t" - "bnez %[loop_size], 1b \n\t" - " addiu %[in1], %[in1], 8 \n\t" - "2: \n\t" - "beqz %[count], 4f \n\t" -#if defined(MIPS_DSP_R1_LE) - " extrv.w %[r0], $ac0, %[scaling] \n\t" -#else - " mfhi %[r0] \n\t" -#endif - "3: \n\t" - "lh %[r0], 0(%[in]) \n\t" - "lh %[r1], 0(%[in1]) \n\t" - "addiu %[count], %[count], -1 \n\t" - "addiu %[in], %[in], 2 \n\t" - "madd %[r0], %[r1] \n\t" - "bnez %[count], 3b \n\t" - " addiu %[in1], %[in1], 2 \n\t" -#if defined(MIPS_DSP_R1_LE) - "extrv.w %[r0], $ac0, %[scaling] \n\t" -#else - "mfhi %[r0] \n\t" -#endif - "4: \n\t" -#if !defined(MIPS_DSP_R1_LE) - "mflo %[r1] \n\t" - "sllv %[r0], %[r0], %[tmp_shift] \n\t" - "srlv %[r1], %[r1], %[scaling] \n\t" - "addu %[r0], %[r0], %[r1] \n\t" -#endif - ".set pop \n\t" - : [loop_size] "+r" (loop_size), [in] "+r" (in), [in1] "+r" (in1), -#if !defined(MIPS_DSP_R2_LE) - [r4] "=&r" (r4), [r5] "=&r" (r5), [r6] "=&r" (r6), [r7] "=&r" (r7), -#endif - [r0] "=&r" (r0), [r1] "=&r" (r1), [r2] "=&r" (r2), [r3] "=&r" (r3), - [count] "+r" (count) - : [scaling] "r" (scaling) -#if !defined(MIPS_DSP_R1_LE) - , [tmp_shift] "r" (tmp_shift) -#endif - : "memory", "hi", "lo" - ); - r[i] = r0; - } - } - *scale = scaling; - - return (order + 1); -} diff --git a/modules/audio_coding/codecs/isac/fix/source/filters_neon.c b/modules/audio_coding/codecs/isac/fix/source/filters_neon.c deleted file mode 100644 index 1734a969cb..0000000000 --- a/modules/audio_coding/codecs/isac/fix/source/filters_neon.c +++ /dev/null @@ -1,114 +0,0 @@ -/* - * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#include - -#include "rtc_base/checks.h" -#include "modules/audio_coding/codecs/isac/fix/source/codec.h" - -// Autocorrelation function in fixed point. -// NOTE! Different from SPLIB-version in how it scales the signal. -int WebRtcIsacfix_AutocorrNeon(int32_t* __restrict r, - const int16_t* x, - int16_t n, - int16_t order, - int16_t* __restrict scale) { - int i = 0; - int16_t scaling = 0; - uint32_t temp = 0; - int64_t prod = 0; - int64_t prod_tail = 0; - - RTC_DCHECK_EQ(0, n % 4); - RTC_DCHECK_GE(n, 8); - - // Calculate r[0]. - int16x4_t x0_v; - int32x4_t tmpa0_v; - int64x2_t tmpb_v; - - tmpb_v = vdupq_n_s64(0); - const int16_t* x_start = x; - const int16_t* x_end0 = x_start + n; - while (x_start < x_end0) { - x0_v = vld1_s16(x_start); - tmpa0_v = vmull_s16(x0_v, x0_v); - tmpb_v = vpadalq_s32(tmpb_v, tmpa0_v); - x_start += 4; - } - -#ifdef WEBRTC_ARCH_ARM64 - prod = vaddvq_s64(tmpb_v); -#else - prod = vget_lane_s64(vadd_s64(vget_low_s64(tmpb_v), vget_high_s64(tmpb_v)), - 0); -#endif - // Calculate scaling (the value of shifting). - temp = (uint32_t)(prod >> 31); - - scaling = temp ? 32 - WebRtcSpl_NormU32(temp) : 0; - r[0] = (int32_t)(prod >> scaling); - - int16x8_t x1_v; - int16x8_t y_v; - int32x4_t tmpa1_v; - // Perform the actual correlation calculation. - for (i = 1; i < order + 1; i++) { - tmpb_v = vdupq_n_s64(0); - int rest = (n - i) % 8; - x_start = x; - x_end0 = x_start + n - i - rest; - const int16_t* y_start = x_start + i; - while (x_start < x_end0) { - x1_v = vld1q_s16(x_start); - y_v = vld1q_s16(y_start); - tmpa0_v = vmull_s16(vget_low_s16(x1_v), vget_low_s16(y_v)); -#ifdef WEBRTC_ARCH_ARM64 - tmpa1_v = vmull_high_s16(x1_v, y_v); -#else - tmpa1_v = vmull_s16(vget_high_s16(x1_v), vget_high_s16(y_v)); -#endif - tmpb_v = vpadalq_s32(tmpb_v, tmpa0_v); - tmpb_v = vpadalq_s32(tmpb_v, tmpa1_v); - x_start += 8; - y_start += 8; - } - // The remaining calculation. - const int16_t* x_end1 = x + n - i; - if (rest >= 4) { - int16x4_t x2_v = vld1_s16(x_start); - int16x4_t y2_v = vld1_s16(y_start); - tmpa0_v = vmull_s16(x2_v, y2_v); - tmpb_v = vpadalq_s32(tmpb_v, tmpa0_v); - x_start += 4; - y_start += 4; - } -#ifdef WEBRTC_ARCH_ARM64 - prod = vaddvq_s64(tmpb_v); -#else - prod = vget_lane_s64(vadd_s64(vget_low_s64(tmpb_v), vget_high_s64(tmpb_v)), - 0); -#endif - - prod_tail = 0; - while (x_start < x_end1) { - prod_tail += *x_start * *y_start; - ++x_start; - ++y_start; - } - - r[i] = (int32_t)((prod + prod_tail) >> scaling); - } - - *scale = scaling; - - return order + 1; -} - diff --git a/modules/audio_coding/codecs/isac/fix/source/filters_unittest.cc b/modules/audio_coding/codecs/isac/fix/source/filters_unittest.cc deleted file mode 100644 index 192ef89f9f..0000000000 --- a/modules/audio_coding/codecs/isac/fix/source/filters_unittest.cc +++ /dev/null @@ -1,68 +0,0 @@ -/* - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#include "modules/audio_coding/codecs/isac/fix/source/codec.h" -#include "system_wrappers/include/cpu_features_wrapper.h" -#include "test/gtest.h" - -class FiltersTest : public ::testing::Test { - protected: - // Pass a function pointer to the Tester function. - void FiltersTester(AutocorrFix WebRtcIsacfix_AutocorrFixFunction) { - const int kOrder = 12; - const int kBuffer = 40; - int16_t scale = 0; - int32_t r_buffer[kOrder + 2] = {0}; - - // Test an overflow case. - const int16_t x_buffer_0[kBuffer] = { - 0, 0, 3010, 22351, 21106, 16969, -2095, -664, - 3513, -30980, 32767, -23839, 13335, 20289, -6831, 339, - -17207, 32767, 4959, 6177, 32767, 16599, -4747, 20504, - 3513, -30980, 32767, -23839, 13335, 20289, 0, -16969, - -2095, -664, 3513, 31981, 32767, -13839, 23336, 30281}; - const int32_t r_expected_0[kOrder + 2] = { - 1872498461, -224288754, 203789985, 483400487, -208272635, - 2436500, 137785322, 266600814, -208486262, 329510080, - 137949184, -161738972, -26894267, 237630192}; - - WebRtcIsacfix_AutocorrFixFunction(r_buffer, x_buffer_0, kBuffer, kOrder + 1, - &scale); - for (int i = 0; i < kOrder + 2; i++) { - EXPECT_EQ(r_expected_0[i], r_buffer[i]); - } - EXPECT_EQ(3, scale); - - // Test a no-overflow case. - const int16_t x_buffer_1[kBuffer] = { - 0, 0, 300, 21, 206, 169, -295, -664, 3513, -300, - 327, -29, 15, 289, -6831, 339, -107, 37, 59, 6177, - 327, 169, -4747, 204, 313, -980, 767, -9, 135, 289, - 0, -6969, -2095, -664, 0, 1, 7, -39, 236, 281}; - const int32_t r_expected_1[kOrder + 2] = { - 176253864, 8126617, 1983287, -26196788, -3487363, - -42839676, -24644043, 3469813, 30559879, 31905045, - 5101567, 29328896, -55787438, -13163978}; - - WebRtcIsacfix_AutocorrFixFunction(r_buffer, x_buffer_1, kBuffer, kOrder + 1, - &scale); - for (int i = 0; i < kOrder + 2; i++) { - EXPECT_EQ(r_expected_1[i], r_buffer[i]); - } - EXPECT_EQ(0, scale); - } -}; - -TEST_F(FiltersTest, AutocorrFixTest) { - FiltersTester(WebRtcIsacfix_AutocorrC); -#if defined(WEBRTC_HAS_NEON) - FiltersTester(WebRtcIsacfix_AutocorrNeon); -#endif -} diff --git a/modules/audio_coding/codecs/isac/fix/source/initialize.c b/modules/audio_coding/codecs/isac/fix/source/initialize.c deleted file mode 100644 index 1b82958883..0000000000 --- a/modules/audio_coding/codecs/isac/fix/source/initialize.c +++ /dev/null @@ -1,173 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/* - * initialize.c - * - * Internal initfunctions - * - */ - -#include "modules/audio_coding/codecs/isac/fix/source/codec.h" -#include "modules/audio_coding/codecs/isac/fix/source/structs.h" -#include "modules/audio_coding/codecs/isac/fix/source/pitch_estimator.h" - - -void WebRtcIsacfix_InitMaskingEnc(MaskFiltstr_enc *maskdata) { - - int k; - - for (k = 0; k < WINLEN; k++) { - maskdata->DataBufferLoQ0[k] = (int16_t) 0; - maskdata->DataBufferHiQ0[k] = (int16_t) 0; - } - for (k = 0; k < ORDERLO+1; k++) { - maskdata->CorrBufLoQQ[k] = (int32_t) 0; - maskdata->CorrBufLoQdom[k] = 0; - - maskdata->PreStateLoGQ15[k] = 0; - - } - for (k = 0; k < ORDERHI+1; k++) { - maskdata->CorrBufHiQQ[k] = (int32_t) 0; - maskdata->CorrBufHiQdom[k] = 0; - maskdata->PreStateHiGQ15[k] = 0; - } - - maskdata->OldEnergy = 10; - - return; -} - -void WebRtcIsacfix_InitMaskingDec(MaskFiltstr_dec *maskdata) { - - int k; - - for (k = 0; k < ORDERLO+1; k++) - { - maskdata->PostStateLoGQ0[k] = 0; - } - for (k = 0; k < ORDERHI+1; k++) - { - maskdata->PostStateHiGQ0[k] = 0; - } - - maskdata->OldEnergy = 10; - - return; -} - - - - - - - -void WebRtcIsacfix_InitPreFilterbank(PreFiltBankstr *prefiltdata) -{ - int k; - - for (k = 0; k < QLOOKAHEAD; k++) { - prefiltdata->INLABUF1_fix[k] = 0; - prefiltdata->INLABUF2_fix[k] = 0; - } - for (k = 0; k < 2 * (QORDER - 1); k++) { - prefiltdata->INSTAT1_fix[k] = 0; - prefiltdata->INSTAT2_fix[k] = 0; - } - - /* High pass filter states */ - prefiltdata->HPstates_fix[0] = 0; - prefiltdata->HPstates_fix[1] = 0; - - return; -} - -void WebRtcIsacfix_InitPostFilterbank(PostFiltBankstr *postfiltdata) -{ - int k; - - for (k = 0; k < 2 * POSTQORDER; k++) { - postfiltdata->STATE_0_LOWER_fix[k] = 0; - postfiltdata->STATE_0_UPPER_fix[k] = 0; - } - - /* High pass filter states */ - - postfiltdata->HPstates1_fix[0] = 0; - postfiltdata->HPstates1_fix[1] = 0; - - postfiltdata->HPstates2_fix[0] = 0; - postfiltdata->HPstates2_fix[1] = 0; - - return; -} - - -void WebRtcIsacfix_InitPitchFilter(PitchFiltstr *pitchfiltdata) -{ - int k; - - for (k = 0; k < PITCH_BUFFSIZE; k++) - pitchfiltdata->ubufQQ[k] = 0; - for (k = 0; k < (PITCH_DAMPORDER); k++) - pitchfiltdata->ystateQQ[k] = 0; - - pitchfiltdata->oldlagQ7 = 6400; /* 50.0 in Q7 */ - pitchfiltdata->oldgainQ12 = 0; -} - -void WebRtcIsacfix_InitPitchAnalysis(PitchAnalysisStruct *State) -{ - int k; - - for (k = 0; k < PITCH_CORR_LEN2+PITCH_CORR_STEP2+PITCH_MAX_LAG/2-PITCH_FRAME_LEN/2+2; k++) { - State->dec_buffer16[k] = 0; - } - for (k = 0; k < 2 * ALLPASSSECTIONS + 1; k++) { - State->decimator_state32[k] = 0; - } - - for (k = 0; k < QLOOKAHEAD; k++) - State->inbuf[k] = 0; - - WebRtcIsacfix_InitPitchFilter(&(State->PFstr_wght)); - - WebRtcIsacfix_InitPitchFilter(&(State->PFstr)); -} - - -void WebRtcIsacfix_InitPlc( PLCstr *State ) -{ - State->decayCoeffPriodic = WEBRTC_SPL_WORD16_MAX; - State->decayCoeffNoise = WEBRTC_SPL_WORD16_MAX; - - State->used = PLC_WAS_USED; - - WebRtcSpl_ZerosArrayW16(State->overlapLP, RECOVERY_OVERLAP); - WebRtcSpl_ZerosArrayW16(State->lofilt_coefQ15, ORDERLO); - WebRtcSpl_ZerosArrayW16(State->hifilt_coefQ15, ORDERHI ); - - State->AvgPitchGain_Q12 = 0; - State->lastPitchGain_Q12 = 0; - State->lastPitchLag_Q7 = 0; - State->gain_lo_hiQ17[0]=State->gain_lo_hiQ17[1] = 0; - WebRtcSpl_ZerosArrayW16(State->prevPitchInvIn, FRAMESAMPLES/2); - WebRtcSpl_ZerosArrayW16(State->prevPitchInvOut, PITCH_MAX_LAG + 10 ); - WebRtcSpl_ZerosArrayW32(State->prevHP, PITCH_MAX_LAG + 10 ); - State->pitchCycles = 0; - State->A = 0; - State->B = 0; - State->pitchIndex = 0; - State->stretchLag = 240; - State->seed = 4447; - - -} diff --git a/modules/audio_coding/codecs/isac/fix/source/isac_fix_type.h b/modules/audio_coding/codecs/isac/fix/source/isac_fix_type.h deleted file mode 100644 index 512911a8bb..0000000000 --- a/modules/audio_coding/codecs/isac/fix/source/isac_fix_type.h +++ /dev/null @@ -1,106 +0,0 @@ -/* - * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_ISAC_FIX_TYPE_H_ -#define MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_ISAC_FIX_TYPE_H_ - -#include "modules/audio_coding/codecs/isac/fix/include/isacfix.h" -#include "rtc_base/checks.h" - -namespace webrtc { - -class IsacFix { - public: - using instance_type = ISACFIX_MainStruct; - static const bool has_swb = false; - static inline int16_t Control(instance_type* inst, - int32_t rate, - int framesize) { - return WebRtcIsacfix_Control(inst, rate, framesize); - } - static inline int16_t ControlBwe(instance_type* inst, - int32_t rate_bps, - int frame_size_ms, - int16_t enforce_frame_size) { - return WebRtcIsacfix_ControlBwe(inst, rate_bps, frame_size_ms, - enforce_frame_size); - } - static inline int16_t Create(instance_type** inst) { - return WebRtcIsacfix_Create(inst); - } - static inline int DecodeInternal(instance_type* inst, - const uint8_t* encoded, - size_t len, - int16_t* decoded, - int16_t* speech_type) { - return WebRtcIsacfix_Decode(inst, encoded, len, decoded, speech_type); - } - static inline size_t DecodePlc(instance_type* inst, - int16_t* decoded, - size_t num_lost_frames) { - return WebRtcIsacfix_DecodePlc(inst, decoded, num_lost_frames); - } - static inline void DecoderInit(instance_type* inst) { - WebRtcIsacfix_DecoderInit(inst); - } - static inline int Encode(instance_type* inst, - const int16_t* speech_in, - uint8_t* encoded) { - return WebRtcIsacfix_Encode(inst, speech_in, encoded); - } - static inline int16_t EncoderInit(instance_type* inst, int16_t coding_mode) { - return WebRtcIsacfix_EncoderInit(inst, coding_mode); - } - static inline uint16_t EncSampRate(instance_type* inst) { - return kFixSampleRate; - } - - static inline int16_t Free(instance_type* inst) { - return WebRtcIsacfix_Free(inst); - } - static inline int16_t GetErrorCode(instance_type* inst) { - return WebRtcIsacfix_GetErrorCode(inst); - } - - static inline int16_t GetNewFrameLen(instance_type* inst) { - return WebRtcIsacfix_GetNewFrameLen(inst); - } - static inline int16_t SetDecSampRate(instance_type* inst, - uint16_t sample_rate_hz) { - RTC_DCHECK_EQ(sample_rate_hz, kFixSampleRate); - return 0; - } - static inline int16_t SetEncSampRate(instance_type* inst, - uint16_t sample_rate_hz) { - RTC_DCHECK_EQ(sample_rate_hz, kFixSampleRate); - return 0; - } - static inline void SetEncSampRateInDecoder(instance_type* inst, - uint16_t sample_rate_hz) { - RTC_DCHECK_EQ(sample_rate_hz, kFixSampleRate); - } - static inline void SetInitialBweBottleneck(instance_type* inst, - int bottleneck_bits_per_second) { - WebRtcIsacfix_SetInitialBweBottleneck(inst, bottleneck_bits_per_second); - } - static inline int16_t SetMaxPayloadSize(instance_type* inst, - int16_t max_payload_size_bytes) { - return WebRtcIsacfix_SetMaxPayloadSize(inst, max_payload_size_bytes); - } - static inline int16_t SetMaxRate(instance_type* inst, int32_t max_bit_rate) { - return WebRtcIsacfix_SetMaxRate(inst, max_bit_rate); - } - - private: - enum { kFixSampleRate = 16000 }; -}; - -} // namespace webrtc -#endif // MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_ISAC_FIX_TYPE_H_ diff --git a/modules/audio_coding/codecs/isac/fix/source/isacfix.c b/modules/audio_coding/codecs/isac/fix/source/isacfix.c deleted file mode 100644 index a7d44e883d..0000000000 --- a/modules/audio_coding/codecs/isac/fix/source/isacfix.c +++ /dev/null @@ -1,1230 +0,0 @@ -/* - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/* - * isacfix.c - * - * This C file contains the functions for the ISAC API - * - */ - -#include "modules/audio_coding/codecs/isac/fix/include/isacfix.h" - -#include - -#include "rtc_base/checks.h" -#include "modules/audio_coding/codecs/isac/fix/source/bandwidth_estimator.h" -#include "modules/audio_coding/codecs/isac/fix/source/codec.h" -#include "modules/audio_coding/codecs/isac/fix/source/entropy_coding.h" -#include "modules/audio_coding/codecs/isac/fix/source/filterbank_internal.h" -#include "modules/audio_coding/codecs/isac/fix/source/lpc_masking_model.h" -#include "modules/audio_coding/codecs/isac/fix/source/structs.h" - -// Declare function pointers. -FilterMaLoopFix WebRtcIsacfix_FilterMaLoopFix; -Spec2Time WebRtcIsacfix_Spec2Time; -Time2Spec WebRtcIsacfix_Time2Spec; -MatrixProduct1 WebRtcIsacfix_MatrixProduct1; -MatrixProduct2 WebRtcIsacfix_MatrixProduct2; - -/* This method assumes that `stream_size_bytes` is in valid range, - * i.e. >= 0 && <= STREAM_MAXW16_60MS - */ -static void InitializeDecoderBitstream(size_t stream_size_bytes, - Bitstr_dec* bitstream) { - bitstream->W_upper = 0xFFFFFFFF; - bitstream->streamval = 0; - bitstream->stream_index = 0; - bitstream->full = 1; - bitstream->stream_size = (stream_size_bytes + 1) >> 1; - memset(bitstream->stream, 0, sizeof(bitstream->stream)); -} - -/**************************************************************************** - * WebRtcIsacfix_Create(...) - * - * This function creates a ISAC instance, which will contain the state - * information for one coding/decoding channel. - * - * Input: - * - *ISAC_main_inst : a pointer to the coder instance. - * - * Return value : 0 - Ok - * -1 - Error - */ - -int16_t WebRtcIsacfix_Create(ISACFIX_MainStruct **ISAC_main_inst) -{ - ISACFIX_SubStruct *tempo; - tempo = malloc(1 * sizeof(ISACFIX_SubStruct)); - *ISAC_main_inst = (ISACFIX_MainStruct *)tempo; - if (*ISAC_main_inst!=NULL) { - (*(ISACFIX_SubStruct**)ISAC_main_inst)->errorcode = 0; - (*(ISACFIX_SubStruct**)ISAC_main_inst)->initflag = 0; - (*(ISACFIX_SubStruct**)ISAC_main_inst)->ISACenc_obj.SaveEnc_ptr = NULL; - WebRtcIsacfix_InitBandwidthEstimator(&tempo->bwestimator_obj); - return(0); - } else { - return(-1); - } -} - - -/**************************************************************************** - * WebRtcIsacfix_CreateInternal(...) - * - * This function creates the memory that is used to store data in the encoder - * - * Input: - * - *ISAC_main_inst : a pointer to the coder instance. - * - * Return value : 0 - Ok - * -1 - Error - */ - -int16_t WebRtcIsacfix_CreateInternal(ISACFIX_MainStruct *ISAC_main_inst) -{ - ISACFIX_SubStruct *ISAC_inst; - - /* typecast pointer to real structure */ - ISAC_inst = (ISACFIX_SubStruct *)ISAC_main_inst; - - /* Allocate memory for storing encoder data */ - ISAC_inst->ISACenc_obj.SaveEnc_ptr = malloc(1 * sizeof(IsacSaveEncoderData)); - - if (ISAC_inst->ISACenc_obj.SaveEnc_ptr!=NULL) { - return(0); - } else { - return(-1); - } -} - - -/**************************************************************************** - * WebRtcIsacfix_Free(...) - * - * This function frees the ISAC instance created at the beginning. - * - * Input: - * - ISAC_main_inst : a ISAC instance. - * - * Return value : 0 - Ok - * -1 - Error - */ - -int16_t WebRtcIsacfix_Free(ISACFIX_MainStruct *ISAC_main_inst) -{ - free(ISAC_main_inst); - return(0); -} - -/**************************************************************************** - * WebRtcIsacfix_FreeInternal(...) - * - * This function frees the internal memory for storing encoder data. - * - * Input: - * - ISAC_main_inst : a ISAC instance. - * - * Return value : 0 - Ok - * -1 - Error - */ - -int16_t WebRtcIsacfix_FreeInternal(ISACFIX_MainStruct *ISAC_main_inst) -{ - ISACFIX_SubStruct *ISAC_inst; - - /* typecast pointer to real structure */ - ISAC_inst = (ISACFIX_SubStruct *)ISAC_main_inst; - - /* Release memory */ - free(ISAC_inst->ISACenc_obj.SaveEnc_ptr); - - return(0); -} - -/**************************************************************************** - * WebRtcIsacfix_InitNeon(...) - * - * This function initializes function pointers for ARM Neon platform. - */ - -#if defined(WEBRTC_HAS_NEON) -static void WebRtcIsacfix_InitNeon(void) { - WebRtcIsacfix_AutocorrFix = WebRtcIsacfix_AutocorrNeon; - WebRtcIsacfix_FilterMaLoopFix = WebRtcIsacfix_FilterMaLoopNeon; - WebRtcIsacfix_Spec2Time = WebRtcIsacfix_Spec2TimeNeon; - WebRtcIsacfix_Time2Spec = WebRtcIsacfix_Time2SpecNeon; - WebRtcIsacfix_AllpassFilter2FixDec16 = - WebRtcIsacfix_AllpassFilter2FixDec16Neon; - WebRtcIsacfix_MatrixProduct1 = WebRtcIsacfix_MatrixProduct1Neon; - WebRtcIsacfix_MatrixProduct2 = WebRtcIsacfix_MatrixProduct2Neon; -} -#endif - -/**************************************************************************** - * WebRtcIsacfix_InitMIPS(...) - * - * This function initializes function pointers for MIPS platform. - */ - -#if defined(MIPS32_LE) -static void WebRtcIsacfix_InitMIPS(void) { - WebRtcIsacfix_AutocorrFix = WebRtcIsacfix_AutocorrMIPS; - WebRtcIsacfix_FilterMaLoopFix = WebRtcIsacfix_FilterMaLoopMIPS; - WebRtcIsacfix_Spec2Time = WebRtcIsacfix_Spec2TimeMIPS; - WebRtcIsacfix_Time2Spec = WebRtcIsacfix_Time2SpecMIPS; - WebRtcIsacfix_MatrixProduct1 = WebRtcIsacfix_MatrixProduct1MIPS; - WebRtcIsacfix_MatrixProduct2 = WebRtcIsacfix_MatrixProduct2MIPS; -#if defined(MIPS_DSP_R1_LE) - WebRtcIsacfix_AllpassFilter2FixDec16 = - WebRtcIsacfix_AllpassFilter2FixDec16MIPS; - WebRtcIsacfix_HighpassFilterFixDec32 = - WebRtcIsacfix_HighpassFilterFixDec32MIPS; -#endif -#if defined(MIPS_DSP_R2_LE) - WebRtcIsacfix_CalculateResidualEnergy = - WebRtcIsacfix_CalculateResidualEnergyMIPS; -#endif -} -#endif - -static void InitFunctionPointers(void) { - WebRtcIsacfix_AutocorrFix = WebRtcIsacfix_AutocorrC; - WebRtcIsacfix_FilterMaLoopFix = WebRtcIsacfix_FilterMaLoopC; - WebRtcIsacfix_CalculateResidualEnergy = - WebRtcIsacfix_CalculateResidualEnergyC; - WebRtcIsacfix_AllpassFilter2FixDec16 = WebRtcIsacfix_AllpassFilter2FixDec16C; - WebRtcIsacfix_HighpassFilterFixDec32 = WebRtcIsacfix_HighpassFilterFixDec32C; - WebRtcIsacfix_Time2Spec = WebRtcIsacfix_Time2SpecC; - WebRtcIsacfix_Spec2Time = WebRtcIsacfix_Spec2TimeC; - WebRtcIsacfix_MatrixProduct1 = WebRtcIsacfix_MatrixProduct1C; - WebRtcIsacfix_MatrixProduct2 = WebRtcIsacfix_MatrixProduct2C; - -#if defined(WEBRTC_HAS_NEON) - WebRtcIsacfix_InitNeon(); -#endif - -#if defined(MIPS32_LE) - WebRtcIsacfix_InitMIPS(); -#endif -} - -/**************************************************************************** - * WebRtcIsacfix_EncoderInit(...) - * - * This function initializes a ISAC instance prior to the encoder calls. - * - * Input: - * - ISAC_main_inst : ISAC instance. - * - CodingMode : 0 -> Bit rate and frame length are automatically - * adjusted to available bandwidth on - * transmission channel. - * 1 -> User sets a frame length and a target bit - * rate which is taken as the maximum short-term - * average bit rate. - * - * Return value : 0 - Ok - * -1 - Error - */ - -int16_t WebRtcIsacfix_EncoderInit(ISACFIX_MainStruct *ISAC_main_inst, - int16_t CodingMode) -{ - int k; - int16_t statusInit; - ISACFIX_SubStruct *ISAC_inst; - - statusInit = 0; - /* typecast pointer to rela structure */ - ISAC_inst = (ISACFIX_SubStruct *)ISAC_main_inst; - - /* flag encoder init */ - ISAC_inst->initflag |= 2; - - if (CodingMode == 0) - /* Adaptive mode */ - ISAC_inst->ISACenc_obj.new_framelength = INITIAL_FRAMESAMPLES; - else if (CodingMode == 1) - /* Instantaneous mode */ - ISAC_inst->ISACenc_obj.new_framelength = 480; /* default for I-mode */ - else { - ISAC_inst->errorcode = ISAC_DISALLOWED_CODING_MODE; - statusInit = -1; - } - - ISAC_inst->CodingMode = CodingMode; - - WebRtcIsacfix_InitMaskingEnc(&ISAC_inst->ISACenc_obj.maskfiltstr_obj); - WebRtcIsacfix_InitPreFilterbank(&ISAC_inst->ISACenc_obj.prefiltbankstr_obj); - WebRtcIsacfix_InitPitchFilter(&ISAC_inst->ISACenc_obj.pitchfiltstr_obj); - WebRtcIsacfix_InitPitchAnalysis(&ISAC_inst->ISACenc_obj.pitchanalysisstr_obj); - - WebRtcIsacfix_InitRateModel(&ISAC_inst->ISACenc_obj.rate_data_obj); - - - ISAC_inst->ISACenc_obj.buffer_index = 0; - ISAC_inst->ISACenc_obj.frame_nb = 0; - ISAC_inst->ISACenc_obj.BottleNeck = 32000; /* default for I-mode */ - ISAC_inst->ISACenc_obj.MaxDelay = 10; /* default for I-mode */ - ISAC_inst->ISACenc_obj.current_framesamples = 0; - ISAC_inst->ISACenc_obj.s2nr = 0; - ISAC_inst->ISACenc_obj.MaxBits = 0; - ISAC_inst->ISACenc_obj.bitstr_seed = 4447; - ISAC_inst->ISACenc_obj.payloadLimitBytes30 = STREAM_MAXW16_30MS << 1; - ISAC_inst->ISACenc_obj.payloadLimitBytes60 = STREAM_MAXW16_60MS << 1; - ISAC_inst->ISACenc_obj.maxPayloadBytes = STREAM_MAXW16_60MS << 1; - ISAC_inst->ISACenc_obj.maxRateInBytes = STREAM_MAXW16_30MS << 1; - ISAC_inst->ISACenc_obj.enforceFrameSize = 0; - - /* Init the bistream data area to zero */ - for (k=0; kISACenc_obj.bitstr_obj.stream[k] = 0; - } - - InitFunctionPointers(); - - return statusInit; -} - -/* Read the given number of bytes of big-endian 16-bit integers from `src` and - write them to `dest` in host endian. If `nbytes` is odd, the number of - output elements is rounded up, and the least significant byte of the last - element is set to 0. */ -static void read_be16(const uint8_t* src, size_t nbytes, uint16_t* dest) { - size_t i; - for (i = 0; i < nbytes / 2; ++i) - dest[i] = src[2 * i] << 8 | src[2 * i + 1]; - if (nbytes % 2 == 1) - dest[nbytes / 2] = src[nbytes - 1] << 8; -} - -/* Read the given number of bytes of host-endian 16-bit integers from `src` and - write them to `dest` in big endian. If `nbytes` is odd, the number of source - elements is rounded up (but only the most significant byte of the last - element is used), and the number of output bytes written will be - nbytes + 1. */ -static void write_be16(const uint16_t* src, size_t nbytes, uint8_t* dest) { - size_t i; - for (i = 0; i < nbytes / 2; ++i) { - dest[2 * i] = src[i] >> 8; - dest[2 * i + 1] = src[i]; - } - if (nbytes % 2 == 1) { - dest[nbytes - 1] = src[nbytes / 2] >> 8; - dest[nbytes] = 0; - } -} - -/**************************************************************************** - * WebRtcIsacfix_Encode(...) - * - * This function encodes 10ms frame(s) and inserts it into a package. - * Input speech length has to be 160 samples (10ms). The encoder buffers those - * 10ms frames until it reaches the chosen Framesize (480 or 960 samples - * corresponding to 30 or 60 ms frames), and then proceeds to the encoding. - * - * Input: - * - ISAC_main_inst : ISAC instance. - * - speechIn : input speech vector. - * - * Output: - * - encoded : the encoded data vector - * - * Return value: - * : >0 - Length (in bytes) of coded data - * : 0 - The buffer didn't reach the chosen framesize - * so it keeps buffering speech samples. - * : -1 - Error - */ - -int WebRtcIsacfix_Encode(ISACFIX_MainStruct *ISAC_main_inst, - const int16_t *speechIn, - uint8_t* encoded) -{ - ISACFIX_SubStruct *ISAC_inst; - int stream_len; - - /* typecast pointer to rela structure */ - ISAC_inst = (ISACFIX_SubStruct *)ISAC_main_inst; - - - /* check if encoder initiated */ - if ((ISAC_inst->initflag & 2) != 2) { - ISAC_inst->errorcode = ISAC_ENCODER_NOT_INITIATED; - return (-1); - } - - stream_len = WebRtcIsacfix_EncodeImpl((int16_t*)speechIn, - &ISAC_inst->ISACenc_obj, - &ISAC_inst->bwestimator_obj, - ISAC_inst->CodingMode); - if (stream_len<0) { - ISAC_inst->errorcode = -(int16_t)stream_len; - return -1; - } - - write_be16(ISAC_inst->ISACenc_obj.bitstr_obj.stream, (size_t)stream_len, - encoded); - return stream_len; - -} - - -/**************************************************************************** - * WebRtcIsacfix_GetNewBitStream(...) - * - * This function returns encoded data, with the received bwe-index in the - * stream. It should always return a complete packet, i.e. only called once - * even for 60 msec frames - * - * Input: - * - ISAC_main_inst : ISAC instance. - * - bweIndex : index of bandwidth estimate to put in new bitstream - * - * Output: - * - encoded : the encoded data vector - * - * Return value: - * : >0 - Length (in bytes) of coded data - * : -1 - Error - */ - -int16_t WebRtcIsacfix_GetNewBitStream(ISACFIX_MainStruct *ISAC_main_inst, - int16_t bweIndex, - float scale, - uint8_t* encoded) -{ - ISACFIX_SubStruct *ISAC_inst; - int16_t stream_len; - - /* typecast pointer to rela structure */ - ISAC_inst = (ISACFIX_SubStruct *)ISAC_main_inst; - - - /* check if encoder initiated */ - if ((ISAC_inst->initflag & 2) != 2) { - ISAC_inst->errorcode = ISAC_ENCODER_NOT_INITIATED; - return (-1); - } - - stream_len = WebRtcIsacfix_EncodeStoredData(&ISAC_inst->ISACenc_obj, - bweIndex, - scale); - if (stream_len<0) { - ISAC_inst->errorcode = - stream_len; - return -1; - } - - write_be16(ISAC_inst->ISACenc_obj.bitstr_obj.stream, stream_len, encoded); - return stream_len; -} - - - -/**************************************************************************** - * WebRtcIsacfix_DecoderInit(...) - * - * This function initializes a ISAC instance prior to the decoder calls. - * - * Input: - * - ISAC_main_inst : ISAC instance. - */ - -void WebRtcIsacfix_DecoderInit(ISACFIX_MainStruct *ISAC_main_inst) -{ - ISACFIX_SubStruct *ISAC_inst; - - InitFunctionPointers(); - - /* typecast pointer to real structure */ - ISAC_inst = (ISACFIX_SubStruct *)ISAC_main_inst; - - /* flag decoder init */ - ISAC_inst->initflag |= 1; - - WebRtcIsacfix_InitMaskingDec(&ISAC_inst->ISACdec_obj.maskfiltstr_obj); - WebRtcIsacfix_InitPostFilterbank(&ISAC_inst->ISACdec_obj.postfiltbankstr_obj); - WebRtcIsacfix_InitPitchFilter(&ISAC_inst->ISACdec_obj.pitchfiltstr_obj); - - /* TS */ - WebRtcIsacfix_InitPlc( &ISAC_inst->ISACdec_obj.plcstr_obj ); -} - - -/**************************************************************************** - * WebRtcIsacfix_UpdateBwEstimate1(...) - * - * This function updates the estimate of the bandwidth. - * - * Input: - * - ISAC_main_inst : ISAC instance. - * - encoded : encoded ISAC frame(s). - * - packet_size : size of the packet. - * - rtp_seq_number : the RTP number of the packet. - * - arr_ts : the arrival time of the packet (from NetEq) - * in samples. - * - * Return value : 0 - Ok - * -1 - Error - */ - -int16_t WebRtcIsacfix_UpdateBwEstimate1(ISACFIX_MainStruct *ISAC_main_inst, - const uint8_t* encoded, - size_t packet_size, - uint16_t rtp_seq_number, - uint32_t arr_ts) -{ - ISACFIX_SubStruct *ISAC_inst; - Bitstr_dec streamdata; - int16_t err; - const size_t kRequiredEncodedLenBytes = 10; - - /* typecast pointer to real structure */ - ISAC_inst = (ISACFIX_SubStruct *)ISAC_main_inst; - - /* Sanity check of packet length */ - if (packet_size == 0) { - /* return error code if the packet length is null or less */ - ISAC_inst->errorcode = ISAC_EMPTY_PACKET; - return -1; - } else if (packet_size > (STREAM_MAXW16<<1)) { - /* return error code if length of stream is too long */ - ISAC_inst->errorcode = ISAC_LENGTH_MISMATCH; - return -1; - } - - /* check if decoder initiated */ - if ((ISAC_inst->initflag & 1) != 1) { - ISAC_inst->errorcode = ISAC_DECODER_NOT_INITIATED; - return (-1); - } - - InitializeDecoderBitstream(packet_size, &streamdata); - - read_be16(encoded, kRequiredEncodedLenBytes, streamdata.stream); - - err = WebRtcIsacfix_EstimateBandwidth(&ISAC_inst->bwestimator_obj, - &streamdata, - packet_size, - rtp_seq_number, - 0, - arr_ts); - - - if (err < 0) - { - /* return error code if something went wrong */ - ISAC_inst->errorcode = -err; - return -1; - } - - - return 0; -} - -/**************************************************************************** - * WebRtcIsacfix_UpdateBwEstimate(...) - * - * This function updates the estimate of the bandwidth. - * - * Input: - * - ISAC_main_inst : ISAC instance. - * - encoded : encoded ISAC frame(s). - * - packet_size : size of the packet. - * - rtp_seq_number : the RTP number of the packet. - * - send_ts : Send Time Stamp from RTP header - * - arr_ts : the arrival time of the packet (from NetEq) - * in samples. - * - * Return value : 0 - Ok - * -1 - Error - */ - -int16_t WebRtcIsacfix_UpdateBwEstimate(ISACFIX_MainStruct *ISAC_main_inst, - const uint8_t* encoded, - size_t packet_size, - uint16_t rtp_seq_number, - uint32_t send_ts, - uint32_t arr_ts) -{ - ISACFIX_SubStruct *ISAC_inst; - Bitstr_dec streamdata; - int16_t err; - const size_t kRequiredEncodedLenBytes = 10; - - /* typecast pointer to real structure */ - ISAC_inst = (ISACFIX_SubStruct *)ISAC_main_inst; - - /* Sanity check of packet length */ - if (packet_size == 0) { - /* return error code if the packet length is null or less */ - ISAC_inst->errorcode = ISAC_EMPTY_PACKET; - return -1; - } else if (packet_size < kRequiredEncodedLenBytes) { - ISAC_inst->errorcode = ISAC_PACKET_TOO_SHORT; - return -1; - } else if (packet_size > (STREAM_MAXW16<<1)) { - /* return error code if length of stream is too long */ - ISAC_inst->errorcode = ISAC_LENGTH_MISMATCH; - return -1; - } - - /* check if decoder initiated */ - if ((ISAC_inst->initflag & 1) != 1) { - ISAC_inst->errorcode = ISAC_DECODER_NOT_INITIATED; - return (-1); - } - - InitializeDecoderBitstream(packet_size, &streamdata); - - read_be16(encoded, kRequiredEncodedLenBytes, streamdata.stream); - - err = WebRtcIsacfix_EstimateBandwidth(&ISAC_inst->bwestimator_obj, - &streamdata, - packet_size, - rtp_seq_number, - send_ts, - arr_ts); - - if (err < 0) - { - /* return error code if something went wrong */ - ISAC_inst->errorcode = -err; - return -1; - } - - - return 0; -} - -/**************************************************************************** - * WebRtcIsacfix_Decode(...) - * - * This function decodes a ISAC frame. Output speech length - * will be a multiple of 480 samples: 480 or 960 samples, - * depending on the framesize (30 or 60 ms). - * - * Input: - * - ISAC_main_inst : ISAC instance. - * - encoded : encoded ISAC frame(s) - * - len : bytes in encoded vector - * - * Output: - * - decoded : The decoded vector - * - * Return value : >0 - number of samples in decoded vector - * -1 - Error - */ - - -int WebRtcIsacfix_Decode(ISACFIX_MainStruct* ISAC_main_inst, - const uint8_t* encoded, - size_t len, - int16_t* decoded, - int16_t* speechType) -{ - ISACFIX_SubStruct *ISAC_inst; - /* number of samples (480 or 960), output from decoder */ - /* that were actually used in the encoder/decoder (determined on the fly) */ - size_t number_of_samples; - int declen_int = 0; - size_t declen; - - /* typecast pointer to real structure */ - ISAC_inst = (ISACFIX_SubStruct *)ISAC_main_inst; - - /* check if decoder initiated */ - if ((ISAC_inst->initflag & 1) != 1) { - ISAC_inst->errorcode = ISAC_DECODER_NOT_INITIATED; - return (-1); - } - - /* Sanity check of packet length */ - if (len == 0) { - /* return error code if the packet length is null or less */ - ISAC_inst->errorcode = ISAC_EMPTY_PACKET; - return -1; - } else if (len > (STREAM_MAXW16<<1)) { - /* return error code if length of stream is too long */ - ISAC_inst->errorcode = ISAC_LENGTH_MISMATCH; - return -1; - } - - InitializeDecoderBitstream(len, &ISAC_inst->ISACdec_obj.bitstr_obj); - - read_be16(encoded, len, ISAC_inst->ISACdec_obj.bitstr_obj.stream); - - /* added for NetEq purposes (VAD/DTX related) */ - *speechType=1; - - declen_int = WebRtcIsacfix_DecodeImpl(decoded, &ISAC_inst->ISACdec_obj, - &number_of_samples); - if (declen_int < 0) { - /* Some error inside the decoder */ - ISAC_inst->errorcode = -(int16_t)declen_int; - memset(decoded, 0, sizeof(int16_t) * MAX_FRAMESAMPLES); - return -1; - } - declen = (size_t)declen_int; - - /* error check */ - - if (declen & 1) { - if (len != declen && - len != declen + - ((ISAC_inst->ISACdec_obj.bitstr_obj.stream[declen >> 1]) & 0xFF)) { - ISAC_inst->errorcode = ISAC_LENGTH_MISMATCH; - memset(decoded, 0, sizeof(int16_t) * number_of_samples); - return -1; - } - } else { - if (len != declen && - len != declen + - ((ISAC_inst->ISACdec_obj.bitstr_obj.stream[declen >> 1]) >> 8)) { - ISAC_inst->errorcode = ISAC_LENGTH_MISMATCH; - memset(decoded, 0, sizeof(int16_t) * number_of_samples); - return -1; - } - } - - return (int)number_of_samples; -} - - -/**************************************************************************** - * WebRtcIsacfix_DecodePlc(...) - * - * This function conducts PLC for ISAC frame(s) in wide-band (16kHz sampling). - * Output speech length will be "480*noOfLostFrames" samples - * that is equevalent of "30*noOfLostFrames" millisecond. - * - * Input: - * - ISAC_main_inst : ISAC instance. - * - noOfLostFrames : Number of PLC frames (480sample = 30ms) - * to produce - * - * Output: - * - decoded : The decoded vector - * - * Return value : Number of samples in decoded PLC vector - */ - -size_t WebRtcIsacfix_DecodePlc(ISACFIX_MainStruct* ISAC_main_inst, - int16_t* decoded, - size_t noOfLostFrames) -{ - - size_t no_of_samples, declen, k; - int16_t outframe16[MAX_FRAMESAMPLES]; - - ISACFIX_SubStruct *ISAC_inst; - /* typecast pointer to real structure */ - ISAC_inst = (ISACFIX_SubStruct *)ISAC_main_inst; - - /* Limit number of frames to two = 60 msec. Otherwise we exceed data vectors */ - if (noOfLostFrames > 2) { - noOfLostFrames = 2; - } - k = 0; - declen = 0; - while( noOfLostFrames > 0 ) - { - WebRtcIsacfix_DecodePlcImpl(&(outframe16[k*480]), &ISAC_inst->ISACdec_obj, - &no_of_samples); - declen += no_of_samples; - noOfLostFrames--; - k++; - } - - for (k=0;kCodingMode == 0) - { - /* in adaptive mode */ - ISAC_inst->errorcode = ISAC_MODE_MISMATCH; - return -1; - } - - - if (rate >= 10000 && rate <= 32000) - ISAC_inst->ISACenc_obj.BottleNeck = rate; - else { - ISAC_inst->errorcode = ISAC_DISALLOWED_BOTTLENECK; - return -1; - } - - - - if (framesize == 30 || framesize == 60) - ISAC_inst->ISACenc_obj.new_framelength = (int16_t)((FS/1000) * framesize); - else { - ISAC_inst->errorcode = ISAC_DISALLOWED_FRAME_LENGTH; - return -1; - } - - return 0; -} - -void WebRtcIsacfix_SetInitialBweBottleneck(ISACFIX_MainStruct* ISAC_main_inst, - int bottleneck_bits_per_second) { - ISACFIX_SubStruct* inst = (ISACFIX_SubStruct*)ISAC_main_inst; - RTC_DCHECK_GE(bottleneck_bits_per_second, 10000); - RTC_DCHECK_LE(bottleneck_bits_per_second, 32000); - inst->bwestimator_obj.sendBwAvg = ((uint32_t)bottleneck_bits_per_second) << 7; -} - -/**************************************************************************** - * WebRtcIsacfix_ControlBwe(...) - * - * This function sets the initial values of bottleneck and frame-size if - * iSAC is used in channel-adaptive mode. Through this API, users can - * enforce a frame-size for all values of bottleneck. Then iSAC will not - * automatically change the frame-size. - * - * - * Input: - * - ISAC_main_inst : ISAC instance. - * - rateBPS : initial value of bottleneck in bits/second - * 10000 <= rateBPS <= 32000 is accepted - * For default bottleneck set rateBPS = 0 - * - frameSizeMs : number of milliseconds per frame (30 or 60) - * - enforceFrameSize : 1 to enforce the given frame-size through out - * the adaptation process, 0 to let iSAC change - * the frame-size if required. - * - * Return value : 0 - ok - * -1 - Error - */ - -int16_t WebRtcIsacfix_ControlBwe(ISACFIX_MainStruct *ISAC_main_inst, - int16_t rateBPS, - int frameSizeMs, - int16_t enforceFrameSize) -{ - ISACFIX_SubStruct *ISAC_inst; - /* Typecast pointer to real structure */ - ISAC_inst = (ISACFIX_SubStruct *)ISAC_main_inst; - - /* check if encoder initiated */ - if ((ISAC_inst->initflag & 2) != 2) { - ISAC_inst->errorcode = ISAC_ENCODER_NOT_INITIATED; - return (-1); - } - - /* Check that we are in channel-adaptive mode, otherwise, return -1 */ - if (ISAC_inst->CodingMode != 0) { - ISAC_inst->errorcode = ISAC_MODE_MISMATCH; - return (-1); - } - - /* Set struct variable if enforceFrameSize is set. ISAC will then keep the */ - /* chosen frame size. */ - ISAC_inst->ISACenc_obj.enforceFrameSize = (enforceFrameSize != 0)? 1:0; - - /* Set initial rate, if value between 10000 and 32000, */ - /* if rateBPS is 0, keep the default initial bottleneck value (15000) */ - if ((rateBPS >= 10000) && (rateBPS <= 32000)) { - ISAC_inst->bwestimator_obj.sendBwAvg = (((uint32_t)rateBPS) << 7); - } else if (rateBPS != 0) { - ISAC_inst->errorcode = ISAC_DISALLOWED_BOTTLENECK; - return -1; - } - - /* Set initial framesize. If enforceFrameSize is set the frame size will not change */ - if ((frameSizeMs == 30) || (frameSizeMs == 60)) { - ISAC_inst->ISACenc_obj.new_framelength = (int16_t)((FS/1000) * frameSizeMs); - } else { - ISAC_inst->errorcode = ISAC_DISALLOWED_FRAME_LENGTH; - return -1; - } - - return 0; -} - - - - - -/**************************************************************************** - * WebRtcIsacfix_GetDownLinkBwIndex(...) - * - * This function returns index representing the Bandwidth estimate from - * other side to this side. - * - * Input: - * - ISAC_main_inst: iSAC struct - * - * Output: - * - rateIndex : Bandwidth estimate to transmit to other side. - * - */ - -int16_t WebRtcIsacfix_GetDownLinkBwIndex(ISACFIX_MainStruct* ISAC_main_inst, - int16_t* rateIndex) -{ - ISACFIX_SubStruct *ISAC_inst; - - /* typecast pointer to real structure */ - ISAC_inst = (ISACFIX_SubStruct *)ISAC_main_inst; - - /* Call function to get Bandwidth Estimate */ - *rateIndex = WebRtcIsacfix_GetDownlinkBwIndexImpl(&ISAC_inst->bwestimator_obj); - - return 0; -} - - -/**************************************************************************** - * WebRtcIsacfix_UpdateUplinkBw(...) - * - * This function takes an index representing the Bandwidth estimate from - * this side to other side and updates BWE. - * - * Input: - * - ISAC_main_inst: iSAC struct - * - rateIndex : Bandwidth estimate from other side. - * - */ - -int16_t WebRtcIsacfix_UpdateUplinkBw(ISACFIX_MainStruct* ISAC_main_inst, - int16_t rateIndex) -{ - int16_t err = 0; - ISACFIX_SubStruct *ISAC_inst; - - /* typecast pointer to real structure */ - ISAC_inst = (ISACFIX_SubStruct *)ISAC_main_inst; - - /* Call function to update BWE with received Bandwidth Estimate */ - err = WebRtcIsacfix_UpdateUplinkBwRec(&ISAC_inst->bwestimator_obj, rateIndex); - if (err < 0) { - ISAC_inst->errorcode = -err; - return (-1); - } - - return 0; -} - -/**************************************************************************** - * WebRtcIsacfix_ReadFrameLen(...) - * - * This function returns the length of the frame represented in the packet. - * - * Input: - * - encoded : Encoded bitstream - * - * Output: - * - frameLength : Length of frame in packet (in samples) - * - */ - -int16_t WebRtcIsacfix_ReadFrameLen(const uint8_t* encoded, - size_t encoded_len_bytes, - size_t* frameLength) -{ - Bitstr_dec streamdata; - int16_t err; - const size_t kRequiredEncodedLenBytes = 10; - - if (encoded_len_bytes < kRequiredEncodedLenBytes) { - return -1; - } - - InitializeDecoderBitstream(encoded_len_bytes, &streamdata); - - read_be16(encoded, kRequiredEncodedLenBytes, streamdata.stream); - - /* decode frame length */ - err = WebRtcIsacfix_DecodeFrameLen(&streamdata, frameLength); - if (err<0) // error check - return err; - - return 0; -} - - -/**************************************************************************** - * WebRtcIsacfix_ReadBwIndex(...) - * - * This function returns the index of the Bandwidth estimate from the bitstream. - * - * Input: - * - encoded : Encoded bitstream - * - * Output: - * - frameLength : Length of frame in packet (in samples) - * - rateIndex : Bandwidth estimate in bitstream - * - */ - -int16_t WebRtcIsacfix_ReadBwIndex(const uint8_t* encoded, - size_t encoded_len_bytes, - int16_t* rateIndex) -{ - Bitstr_dec streamdata; - int16_t err; - const size_t kRequiredEncodedLenBytes = 10; - - if (encoded_len_bytes < kRequiredEncodedLenBytes) { - return -1; - } - - InitializeDecoderBitstream(encoded_len_bytes, &streamdata); - - read_be16(encoded, kRequiredEncodedLenBytes, streamdata.stream); - - /* decode frame length, needed to get to the rateIndex in the bitstream */ - size_t frameLength; - err = WebRtcIsacfix_DecodeFrameLen(&streamdata, &frameLength); - if (err<0) // error check - return err; - - /* decode BW estimation */ - err = WebRtcIsacfix_DecodeSendBandwidth(&streamdata, rateIndex); - if (err<0) // error check - return err; - - return 0; -} - - - - -/**************************************************************************** - * WebRtcIsacfix_GetErrorCode(...) - * - * This function can be used to check the error code of an iSAC instance. When - * a function returns -1 a error code will be set for that instance. The - * function below extract the code of the last error that occured in the - * specified instance. - * - * Input: - * - ISAC_main_inst : ISAC instance - * - * Return value : Error code - */ - -int16_t WebRtcIsacfix_GetErrorCode(ISACFIX_MainStruct *ISAC_main_inst) -{ - ISACFIX_SubStruct *ISAC_inst; - /* typecast pointer to real structure */ - ISAC_inst = (ISACFIX_SubStruct *)ISAC_main_inst; - - return ISAC_inst->errorcode; -} - - - -/**************************************************************************** - * WebRtcIsacfix_GetUplinkBw(...) - * - * This function returns the inst quantized iSAC send bitrate - * - * Input: - * - ISAC_main_inst : iSAC instance - * - * Return value : bitrate - */ - -int32_t WebRtcIsacfix_GetUplinkBw(ISACFIX_MainStruct *ISAC_main_inst) -{ - ISACFIX_SubStruct *ISAC_inst = (ISACFIX_SubStruct *)ISAC_main_inst; - BwEstimatorstr * bw = (BwEstimatorstr*)&(ISAC_inst->bwestimator_obj); - - return (int32_t) WebRtcIsacfix_GetUplinkBandwidth(bw); -} - -/**************************************************************************** - * WebRtcIsacfix_GetNewFrameLen(...) - * - * This function return the next frame length (in samples) of iSAC. - * - * Input: - * - ISAC_main_inst : iSAC instance - * - * Return value : frame lenght in samples - */ - -int16_t WebRtcIsacfix_GetNewFrameLen(ISACFIX_MainStruct *ISAC_main_inst) -{ - ISACFIX_SubStruct *ISAC_inst = (ISACFIX_SubStruct *)ISAC_main_inst; - return ISAC_inst->ISACenc_obj.new_framelength; -} - - -/**************************************************************************** - * WebRtcIsacfix_SetMaxPayloadSize(...) - * - * This function sets a limit for the maximum payload size of iSAC. The same - * value is used both for 30 and 60 msec packets. - * The absolute max will be valid until next time the function is called. - * NOTE! This function may override the function WebRtcIsacfix_SetMaxRate() - * - * Input: - * - ISAC_main_inst : iSAC instance - * - maxPayloadBytes : maximum size of the payload in bytes - * valid values are between 100 and 400 bytes - * - * - * Return value : 0 if sucessful - * -1 if error happens - */ - -int16_t WebRtcIsacfix_SetMaxPayloadSize(ISACFIX_MainStruct *ISAC_main_inst, - int16_t maxPayloadBytes) -{ - ISACFIX_SubStruct *ISAC_inst; - - /* typecast pointer to real structure */ - ISAC_inst = (ISACFIX_SubStruct *)ISAC_main_inst; - - if((maxPayloadBytes < 100) || (maxPayloadBytes > 400)) - { - /* maxPayloadBytes is out of valid range */ - return -1; - } - else - { - /* Set new absolute max, which will not change unless this function - is called again with a new value */ - ISAC_inst->ISACenc_obj.maxPayloadBytes = maxPayloadBytes; - - /* Set new maximum values for 30 and 60 msec packets */ - if (maxPayloadBytes < ISAC_inst->ISACenc_obj.maxRateInBytes) { - ISAC_inst->ISACenc_obj.payloadLimitBytes30 = maxPayloadBytes; - } else { - ISAC_inst->ISACenc_obj.payloadLimitBytes30 = ISAC_inst->ISACenc_obj.maxRateInBytes; - } - - if ( maxPayloadBytes < (ISAC_inst->ISACenc_obj.maxRateInBytes << 1)) { - ISAC_inst->ISACenc_obj.payloadLimitBytes60 = maxPayloadBytes; - } else { - ISAC_inst->ISACenc_obj.payloadLimitBytes60 = (ISAC_inst->ISACenc_obj.maxRateInBytes << 1); - } - } - return 0; -} - - -/**************************************************************************** - * WebRtcIsacfix_SetMaxRate(...) - * - * This function sets the maximum rate which the codec may not exceed for a - * singel packet. The maximum rate is set in bits per second. - * The codec has an absolute maximum rate of 53400 bits per second (200 bytes - * per 30 msec). - * It is possible to set a maximum rate between 32000 and 53400 bits per second. - * - * The rate limit is valid until next time the function is called. - * - * NOTE! Packet size will never go above the value set if calling - * WebRtcIsacfix_SetMaxPayloadSize() (default max packet size is 400 bytes). - * - * Input: - * - ISAC_main_inst : iSAC instance - * - maxRateInBytes : maximum rate in bits per second, - * valid values are 32000 to 53400 bits - * - * Return value : 0 if sucessful - * -1 if error happens - */ - -int16_t WebRtcIsacfix_SetMaxRate(ISACFIX_MainStruct *ISAC_main_inst, - int32_t maxRate) -{ - ISACFIX_SubStruct *ISAC_inst; - int16_t maxRateInBytes; - - /* typecast pointer to real structure */ - ISAC_inst = (ISACFIX_SubStruct *)ISAC_main_inst; - - if((maxRate < 32000) || (maxRate > 53400)) - { - /* maxRate is out of valid range */ - return -1; - } - else - { - /* Calculate maximum number of bytes per 30 msec packets for the given - maximum rate. Multiply with 30/1000 to get number of bits per 30 msec, - divide by 8 to get number of bytes per 30 msec: - maxRateInBytes = floor((maxRate * 30/1000) / 8); */ - maxRateInBytes = (int16_t)( WebRtcSpl_DivW32W16ResW16(WEBRTC_SPL_MUL(maxRate, 3), 800) ); - - /* Store the value for usage in the WebRtcIsacfix_SetMaxPayloadSize-function */ - ISAC_inst->ISACenc_obj.maxRateInBytes = maxRateInBytes; - - /* For 30 msec packets: if the new limit is below the maximum - payload size, set a new limit */ - if (maxRateInBytes < ISAC_inst->ISACenc_obj.maxPayloadBytes) { - ISAC_inst->ISACenc_obj.payloadLimitBytes30 = maxRateInBytes; - } else { - ISAC_inst->ISACenc_obj.payloadLimitBytes30 = ISAC_inst->ISACenc_obj.maxPayloadBytes; - } - - /* For 60 msec packets: if the new limit (times 2) is below the - maximum payload size, set a new limit */ - if ( (maxRateInBytes << 1) < ISAC_inst->ISACenc_obj.maxPayloadBytes) { - ISAC_inst->ISACenc_obj.payloadLimitBytes60 = (maxRateInBytes << 1); - } else { - ISAC_inst->ISACenc_obj.payloadLimitBytes60 = ISAC_inst->ISACenc_obj.maxPayloadBytes; - } - } - - return 0; -} - - - -/**************************************************************************** - * WebRtcIsacfix_version(...) - * - * This function returns the version number. - * - * Output: - * - version : Pointer to character string - * - */ - -void WebRtcIsacfix_version(char *version) -{ - strcpy(version, "3.6.0"); -} diff --git a/modules/audio_coding/codecs/isac/fix/source/lattice.c b/modules/audio_coding/codecs/isac/fix/source/lattice.c deleted file mode 100644 index 7bbf4e054a..0000000000 --- a/modules/audio_coding/codecs/isac/fix/source/lattice.c +++ /dev/null @@ -1,321 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/* - * lattice.c - * - * Contains the normalized lattice filter routines (MA and AR) for iSAC codec - * - */ - -#include "modules/audio_coding/codecs/isac/fix/source/codec.h" -#include "modules/audio_coding/codecs/isac/fix/source/settings.h" -#include "rtc_base/sanitizer.h" - -#define LATTICE_MUL_32_32_RSFT16(a32a, a32b, b32) \ - ((int32_t)(WEBRTC_SPL_MUL(a32a, b32) + (WEBRTC_SPL_MUL_16_32_RSFT16(a32b, b32)))) -/* This macro is FORBIDDEN to use elsewhere than in a function in this file and - its corresponding neon version. It might give unpredictable results, since a - general int32_t*int32_t multiplication results in a 64 bit value. - The result is then shifted just 16 steps to the right, giving need for 48 - bits, i.e. in the generel case, it will NOT fit in a int32_t. In the - cases used in here, the int32_t will be enough, since (for a good - reason) the involved multiplicands aren't big enough to overflow a - int32_t after shifting right 16 bits. I have compared the result of a - multiplication between t32 and tmp32, done in two ways: - 1) Using (int32_t) (((float)(tmp32))*((float)(tmp32b))/65536.0); - 2) Using LATTICE_MUL_32_32_RSFT16(t16a, t16b, tmp32b); - By running 25 files, I haven't found any bigger diff than 64 - this was in the - case when method 1) gave 650235648 and 2) gave 650235712. -*/ - -/* Function prototype: filtering ar_g_Q0[] and ar_f_Q0[] through an AR filter - with coefficients cth_Q15[] and sth_Q15[]. - Implemented for both generic and ARMv7 platforms. - */ -void WebRtcIsacfix_FilterArLoop(int16_t* ar_g_Q0, - int16_t* ar_f_Q0, - int16_t* cth_Q15, - int16_t* sth_Q15, - size_t order_coef); - -/* Inner loop used for function WebRtcIsacfix_NormLatticeFilterMa(). It does: - for 0 <= n < HALF_SUBFRAMELEN - 1: - *ptr2 = input2 * (*ptr2) + input0 * (*ptr0)); - *ptr1 = input1 * (*ptr0) + input0 * (*ptr2); - Note, function WebRtcIsacfix_FilterMaLoopNeon and WebRtcIsacfix_FilterMaLoopC - are not bit-exact. The accuracy by the ARM Neon function is same or better. -*/ -void WebRtcIsacfix_FilterMaLoopC(int16_t input0, // Filter coefficient - int16_t input1, // Filter coefficient - int32_t input2, // Inverse coeff. (1/input1) - int32_t* ptr0, // Sample buffer - int32_t* ptr1, // Sample buffer - int32_t* ptr2) { // Sample buffer - int n = 0; - - // Separate the 32-bit variable input2 into two 16-bit integers (high 16 and - // low 16 bits), for using LATTICE_MUL_32_32_RSFT16 in the loop. - int16_t t16a = (int16_t)(input2 >> 16); - int16_t t16b = (int16_t)input2; - if (t16b < 0) t16a++; - - // The loop filtering the samples *ptr0, *ptr1, *ptr2 with filter coefficients - // input0, input1, and input2. - for(n = 0; n < HALF_SUBFRAMELEN - 1; n++, ptr0++, ptr1++, ptr2++) { - int32_t tmp32a = 0; - int32_t tmp32b = 0; - - // Calculate *ptr2 = input2 * (*ptr2 + input0 * (*ptr0)); - tmp32a = WEBRTC_SPL_MUL_16_32_RSFT15(input0, *ptr0); // Q15 * Q15 >> 15 = Q15 - tmp32b = *ptr2 + tmp32a; // Q15 + Q15 = Q15 - *ptr2 = LATTICE_MUL_32_32_RSFT16(t16a, t16b, tmp32b); - - // Calculate *ptr1 = input1 * (*ptr0) + input0 * (*ptr2); - tmp32a = WEBRTC_SPL_MUL_16_32_RSFT15(input1, *ptr0); // Q15*Q15>>15 = Q15 - tmp32b = WEBRTC_SPL_MUL_16_32_RSFT15(input0, *ptr2); // Q15*Q15>>15 = Q15 - *ptr1 = tmp32a + tmp32b; // Q15 + Q15 = Q15 - } -} - -/* filter the signal using normalized lattice filter */ -/* MA filter */ -void WebRtcIsacfix_NormLatticeFilterMa(size_t orderCoef, - int32_t *stateGQ15, - int16_t *lat_inQ0, - int16_t *filt_coefQ15, - int32_t *gain_lo_hiQ17, - int16_t lo_hi, - int16_t *lat_outQ9) -{ - int16_t sthQ15[MAX_AR_MODEL_ORDER]; - int16_t cthQ15[MAX_AR_MODEL_ORDER]; - - int u, n; - size_t i, k; - int16_t temp2,temp3; - size_t ord_1 = orderCoef+1; - int32_t inv_cthQ16[MAX_AR_MODEL_ORDER]; - - int32_t gain32, fQtmp; - int16_t gain16; - int16_t gain_sh; - - int32_t tmp32, tmp32b; - int32_t fQ15vec[HALF_SUBFRAMELEN]; - int32_t gQ15[MAX_AR_MODEL_ORDER+1][HALF_SUBFRAMELEN]; - int16_t sh; - int16_t t16a; - int16_t t16b; - - for (u=0;u>15 = Q(17+gain_sh) - inv_cthQ16[k] = WebRtcSpl_DivW32W16((int32_t)2147483647, cthQ15[k]); // 1/cth[k] in Q31/Q15 = Q16 - } - gain16 = (int16_t)(gain32 >> 16); // Q(1+gain_sh). - - /* normalized lattice filter */ - /*****************************/ - - /* initial conditions */ - for (i=0;i>15 = Q15 - tmp32b= fQtmp + tmp32; //Q15+Q15=Q15 - tmp32 = inv_cthQ16[i-1]; //Q16 - t16a = (int16_t)(tmp32 >> 16); - t16b = (int16_t)(tmp32 - (t16a << 16)); - if (t16b<0) t16a++; - tmp32 = LATTICE_MUL_32_32_RSFT16(t16a, t16b, tmp32b); - fQtmp = tmp32; // Q15 - - // Calculate g[i][0] = cth[i-1]*stateG[i-1] + sth[i-1]* f[i][0]; - tmp32 = WEBRTC_SPL_MUL_16_32_RSFT15(cthQ15[i-1], stateGQ15[i-1]); //Q15*Q15>>15 = Q15 - tmp32b = WEBRTC_SPL_MUL_16_32_RSFT15(sthQ15[i-1], fQtmp); //Q15*Q15>>15 = Q15 - tmp32 = tmp32 + tmp32b;//Q15+Q15 = Q15 - gQ15[i][0] = tmp32; // Q15 - } - - /* filtering */ - /* save the states */ - for(k=0;k>= gain_sh; // Q(17+gain_sh) -> Q17 - tmp32 = WEBRTC_SPL_MUL_16_32_RSFT16(gain16, fQ15vec[n]); //Q(1+gain_sh)*Q15>>16 = Q(gain_sh) - sh = 9-gain_sh; //number of needed shifts to reach Q9 - t16a = (int16_t) WEBRTC_SPL_SHIFT_W32(tmp32, sh); - lat_outQ9[n + temp1] = t16a; - } - - /* save the states */ - for (i=0;i>15 = Q27 - } - - sh = WebRtcSpl_NormW32(tmp32); // tmp32 is the gain - den16 = (int16_t) WEBRTC_SPL_SHIFT_W32(tmp32, sh-16); //Q(27+sh-16) = Q(sh+11) (all 16 bits are value bits) - inv_gain32 = WebRtcSpl_DivW32W16((int32_t)2147483647, den16); // 1/gain in Q31/Q(sh+11) = Q(20-sh) - - //initial conditions - inv_gain16 = (int16_t)(inv_gain32 >> 2); // 1/gain in Q(20-sh-2) = Q(18-sh) - - for (i=0;iQ26 - tmp32 = WEBRTC_SPL_MUL_16_32_RSFT16(inv_gain16, tmp32); //lat_in[]*inv_gain in (Q(18-sh)*Q26)>>16 = Q(28-sh) - tmp32 = WEBRTC_SPL_SHIFT_W32(tmp32, -(28-sh)); // lat_in[]*inv_gain in Q0 - - ARfQ0vec[i] = (int16_t)WebRtcSpl_SatW32ToW16(tmp32); // Q0 - } - - // Get the state of f & g for the first input, for all orders. - for (i = orderCoef; i > 0; i--) - { - tmp32 = (cthQ15[i - 1] * ARfQ0vec[0] - sthQ15[i - 1] * stateGQ0[i - 1] + - 16384) >> 15; - tmpAR = (int16_t)WebRtcSpl_SatW32ToW16(tmp32); // Q0 - - tmp32 = (sthQ15[i - 1] * ARfQ0vec[0] + cthQ15[i - 1] * stateGQ0[i - 1] + - 16384) >> 15; - ARgQ0vec[i] = (int16_t)WebRtcSpl_SatW32ToW16(tmp32); // Q0 - ARfQ0vec[0] = tmpAR; - } - ARgQ0vec[0] = ARfQ0vec[0]; - - // Filter ARgQ0vec[] and ARfQ0vec[] through coefficients cthQ15[] and sthQ15[]. - WebRtcIsacfix_FilterArLoop(ARgQ0vec, ARfQ0vec, cthQ15, sthQ15, orderCoef); - - for(n=0;n 0; k--) - - ldrh r7, [r3, #-2]! @ sth_Q15[k - 1] - ldrh r6, [r2, #-2]! @ cth_Q15[k - 1] - - ldrh r8, [r0, #-2] @ ar_g_Q0[k - 1] - smlabb r11, r7, r5, r12 @ sth_Q15[k - 1] * tmpAR + 16384 - smlabb r10, r6, r5, r12 @ cth_Q15[k - 1] * tmpAR + 16384 - smulbb r7, r7, r8 @ sth_Q15[k - 1] * ar_g_Q0[k - 1] - smlabb r11, r6, r8, r11 @ cth_Q15[k - 1] * ar_g_Q0[k - 1] + - @ (sth_Q15[k - 1] * tmpAR + 16384) - - sub r10, r10, r7 @ cth_Q15[k - 1] * tmpAR + 16384 - - @ (sth_Q15[k - 1] * ar_g_Q0[k - 1]) - ssat r11, #16, r11, asr #15 - ssat r5, #16, r10, asr #15 - strh r11, [r0], #-2 @ Output: ar_g_Q0[k] - - subs r9, #1 - bgt ORDER_COEF_LOOP - - strh r5, [r0] @ Output: ar_g_Q0[0] = tmpAR; - strh r5, [r1], #2 @ Output: ar_f_Q0[n+1] = tmpAR; - - subs r4, #1 - bne HALF_SUBFRAME_LOOP - - pop {r4-r11} - bx lr diff --git a/modules/audio_coding/codecs/isac/fix/source/lattice_c.c b/modules/audio_coding/codecs/isac/fix/source/lattice_c.c deleted file mode 100644 index 43406612e8..0000000000 --- a/modules/audio_coding/codecs/isac/fix/source/lattice_c.c +++ /dev/null @@ -1,48 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/* - * Contains the core loop function for the lattice filter AR routine - * for iSAC codec. - * - */ - -#include "common_audio/signal_processing/include/signal_processing_library.h" -#include "modules/audio_coding/codecs/isac/fix/source/settings.h" - -/* Filter ar_g_Q0[] and ar_f_Q0[] through an AR filter with coefficients - * cth_Q15[] and sth_Q15[]. - */ -void WebRtcIsacfix_FilterArLoop(int16_t* ar_g_Q0, // Input samples - int16_t* ar_f_Q0, // Input samples - int16_t* cth_Q15, // Filter coefficients - int16_t* sth_Q15, // Filter coefficients - size_t order_coef) { // order of the filter - int n = 0; - - for (n = 0; n < HALF_SUBFRAMELEN - 1; n++) { - size_t k = 0; - int16_t tmpAR = 0; - int32_t tmp32 = 0; - int32_t tmp32_2 = 0; - - tmpAR = ar_f_Q0[n + 1]; - for (k = order_coef; k > 0; k--) { - tmp32 = (cth_Q15[k - 1] * tmpAR - sth_Q15[k - 1] * ar_g_Q0[k - 1] + - 16384) >> 15; - tmp32_2 = (sth_Q15[k - 1] * tmpAR + cth_Q15[k - 1] * ar_g_Q0[k - 1] + - 16384) >> 15; - tmpAR = (int16_t)WebRtcSpl_SatW32ToW16(tmp32); - ar_g_Q0[k] = (int16_t)WebRtcSpl_SatW32ToW16(tmp32_2); - } - ar_f_Q0[n + 1] = tmpAR; - ar_g_Q0[0] = tmpAR; - } -} diff --git a/modules/audio_coding/codecs/isac/fix/source/lattice_mips.c b/modules/audio_coding/codecs/isac/fix/source/lattice_mips.c deleted file mode 100644 index 3189726629..0000000000 --- a/modules/audio_coding/codecs/isac/fix/source/lattice_mips.c +++ /dev/null @@ -1,329 +0,0 @@ -/* - * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#include -#include - -#include "modules/audio_coding/codecs/isac/fix/source/settings.h" - -// Filter ar_g_Q0[] and ar_f_Q0[] through an AR filter with coefficients -// cth_Q15[] and sth_Q15[]. -void WebRtcIsacfix_FilterArLoop(int16_t* ar_g_Q0, // Input samples - int16_t* ar_f_Q0, // Input samples - int16_t* cth_Q15, // Filter coefficients - int16_t* sth_Q15, // Filter coefficients - size_t order_coef) { // order of the filter - int n = 0; - - for (n = 0; n < HALF_SUBFRAMELEN - 1; n++) { - int count = (int)(order_coef - 1); - int offset; -#if !defined(MIPS_DSP_R1_LE) - int16_t* tmp_cth; - int16_t* tmp_sth; - int16_t* tmp_arg; - int32_t max_q16 = 0x7fff; - int32_t min_q16 = 0xffff8000; -#endif - // Declare variables used as temporary registers. - int32_t r0, r1, r2, t0, t1, t2, t_ar; - - __asm __volatile ( - ".set push \n\t" - ".set noreorder \n\t" - "bltz %[count], 2f \n\t" - " lh %[t_ar], 0(%[tmp]) \n\t" - // Inner loop - "1: \n\t" - "sll %[offset], %[count], 1 \n\t" -#if defined(MIPS_DSP_R1_LE) - "lhx %[r0], %[offset](%[cth_Q15]) \n\t" - "lhx %[r1], %[offset](%[sth_Q15]) \n\t" - "lhx %[r2], %[offset](%[ar_g_Q0]) \n\t" -#else - "addu %[tmp_cth], %[cth_Q15], %[offset] \n\t" - "addu %[tmp_sth], %[sth_Q15], %[offset] \n\t" - "addu %[tmp_arg], %[ar_g_Q0], %[offset] \n\t" - "lh %[r0], 0(%[tmp_cth]) \n\t" - "lh %[r1], 0(%[tmp_sth]) \n\t" - "lh %[r2], 0(%[tmp_arg]) \n\t" -#endif - "mul %[t0], %[r0], %[t_ar] \n\t" - "mul %[t1], %[r1], %[t_ar] \n\t" - "mul %[t2], %[r1], %[r2] \n\t" - "mul %[r0], %[r0], %[r2] \n\t" - "subu %[t0], %[t0], %[t2] \n\t" - "addu %[t1], %[t1], %[r0] \n\t" -#if defined(MIPS_DSP_R1_LE) - "shra_r.w %[t1], %[t1], 15 \n\t" - "shra_r.w %[t0], %[t0], 15 \n\t" -#else - "addiu %[t1], %[t1], 0x4000 \n\t" - "sra %[t1], %[t1], 15 \n\t" - "addiu %[t0], %[t0], 0x4000 \n\t" - "sra %[t0], %[t0], 15 \n\t" -#endif - "addiu %[offset], %[offset], 2 \n\t" -#if defined(MIPS_DSP_R1_LE) - "shll_s.w %[t1], %[t1], 16 \n\t" - "shll_s.w %[t_ar], %[t0], 16 \n\t" -#else - "slt %[r0], %[t1], %[max_q16] \n\t" - "slt %[r1], %[t0], %[max_q16] \n\t" - "movz %[t1], %[max_q16], %[r0] \n\t" - "movz %[t0], %[max_q16], %[r1] \n\t" -#endif - "addu %[offset], %[offset], %[ar_g_Q0] \n\t" -#if defined(MIPS_DSP_R1_LE) - "sra %[t1], %[t1], 16 \n\t" - "sra %[t_ar], %[t_ar], 16 \n\t" -#else - "slt %[r0], %[t1], %[min_q16] \n\t" - "slt %[r1], %[t0], %[min_q16] \n\t" - "movn %[t1], %[min_q16], %[r0] \n\t" - "movn %[t0], %[min_q16], %[r1] \n\t" - "addu %[t_ar], $zero, %[t0] \n\t" -#endif - "sh %[t1], 0(%[offset]) \n\t" - "bgtz %[count], 1b \n\t" - " addiu %[count], %[count], -1 \n\t" - "2: \n\t" - "sh %[t_ar], 0(%[tmp]) \n\t" - "sh %[t_ar], 0(%[ar_g_Q0]) \n\t" - ".set pop \n\t" - : [t_ar] "=&r" (t_ar), [count] "+r" (count), [offset] "=&r" (offset), - [r0] "=&r" (r0), [r1] "=&r" (r1), [r2] "=&r" (r2), [t0] "=&r" (t0), -#if !defined(MIPS_DSP_R1_LE) - [tmp_cth] "=&r" (tmp_cth), [tmp_sth] "=&r" (tmp_sth), - [tmp_arg] "=&r" (tmp_arg), -#endif - [t1] "=&r" (t1), [t2] "=&r" (t2) - : [tmp] "r" (&ar_f_Q0[n+1]), [cth_Q15] "r" (cth_Q15), -#if !defined(MIPS_DSP_R1_LE) - [max_q16] "r" (max_q16), [min_q16] "r" (min_q16), -#endif - [sth_Q15] "r" (sth_Q15), [ar_g_Q0] "r" (ar_g_Q0) - : "memory", "hi", "lo" - ); - } -} - -// MIPS optimization of the inner loop used for function -// WebRtcIsacfix_NormLatticeFilterMa(). It does: -// -// for 0 <= n < HALF_SUBFRAMELEN - 1: -// *ptr2 = input2 * (*ptr2) + input0 * (*ptr0)); -// *ptr1 = input1 * (*ptr0) + input0 * (*ptr2); -// -// Note, function WebRtcIsacfix_FilterMaLoopMIPS and WebRtcIsacfix_FilterMaLoopC -// are not bit-exact. The accuracy of the MIPS function is same or better. -void WebRtcIsacfix_FilterMaLoopMIPS(int16_t input0, // Filter coefficient - int16_t input1, // Filter coefficient - int32_t input2, // Inverse coeff (1/input1) - int32_t* ptr0, // Sample buffer - int32_t* ptr1, // Sample buffer - int32_t* ptr2) { // Sample buffer -#if defined(MIPS_DSP_R2_LE) - // MIPS DSPR2 version. 4 available accumulators allows loop unrolling 4 times. - // This variant is not bit-exact with WebRtcIsacfix_FilterMaLoopC, since we - // are exploiting 64-bit accumulators. The accuracy of the MIPS DSPR2 function - // is same or better. - int n = (HALF_SUBFRAMELEN - 1) >> 2; - int m = (HALF_SUBFRAMELEN - 1) & 3; - - int r0, r1, r2, r3; - int t0, t1, t2, t3; - int s0, s1, s2, s3; - - __asm __volatile ( - ".set push \n\t" - ".set noreorder \n\t" - "1: \n\t" - "lw %[r0], 0(%[ptr0]) \n\t" - "lw %[r1], 4(%[ptr0]) \n\t" - "lw %[r2], 8(%[ptr0]) \n\t" - "lw %[r3], 12(%[ptr0]) \n\t" - "mult $ac0, %[r0], %[input0] \n\t" - "mult $ac1, %[r1], %[input0] \n\t" - "mult $ac2, %[r2], %[input0] \n\t" - "mult $ac3, %[r3], %[input0] \n\t" - "lw %[t0], 0(%[ptr2]) \n\t" - "extr_rs.w %[s0], $ac0, 15 \n\t" - "extr_rs.w %[s1], $ac1, 15 \n\t" - "extr_rs.w %[s2], $ac2, 15 \n\t" - "extr_rs.w %[s3], $ac3, 15 \n\t" - "lw %[t1], 4(%[ptr2]) \n\t" - "lw %[t2], 8(%[ptr2]) \n\t" - "lw %[t3], 12(%[ptr2]) \n\t" - "addu %[t0], %[t0], %[s0] \n\t" - "addu %[t1], %[t1], %[s1] \n\t" - "addu %[t2], %[t2], %[s2] \n\t" - "addu %[t3], %[t3], %[s3] \n\t" - "mult $ac0, %[t0], %[input2] \n\t" - "mult $ac1, %[t1], %[input2] \n\t" - "mult $ac2, %[t2], %[input2] \n\t" - "mult $ac3, %[t3], %[input2] \n\t" - "addiu %[ptr0], %[ptr0], 16 \n\t" - "extr_rs.w %[t0], $ac0, 16 \n\t" - "extr_rs.w %[t1], $ac1, 16 \n\t" - "extr_rs.w %[t2], $ac2, 16 \n\t" - "extr_rs.w %[t3], $ac3, 16 \n\t" - "addiu %[n], %[n], -1 \n\t" - "mult $ac0, %[r0], %[input1] \n\t" - "mult $ac1, %[r1], %[input1] \n\t" - "mult $ac2, %[r2], %[input1] \n\t" - "mult $ac3, %[r3], %[input1] \n\t" - "sw %[t0], 0(%[ptr2]) \n\t" - "extr_rs.w %[s0], $ac0, 15 \n\t" - "extr_rs.w %[s1], $ac1, 15 \n\t" - "extr_rs.w %[s2], $ac2, 15 \n\t" - "extr_rs.w %[s3], $ac3, 15 \n\t" - "sw %[t1], 4(%[ptr2]) \n\t" - "sw %[t2], 8(%[ptr2]) \n\t" - "sw %[t3], 12(%[ptr2]) \n\t" - "mult $ac0, %[t0], %[input0] \n\t" - "mult $ac1, %[t1], %[input0] \n\t" - "mult $ac2, %[t2], %[input0] \n\t" - "mult $ac3, %[t3], %[input0] \n\t" - "addiu %[ptr2], %[ptr2], 16 \n\t" - "extr_rs.w %[t0], $ac0, 15 \n\t" - "extr_rs.w %[t1], $ac1, 15 \n\t" - "extr_rs.w %[t2], $ac2, 15 \n\t" - "extr_rs.w %[t3], $ac3, 15 \n\t" - "addu %[t0], %[t0], %[s0] \n\t" - "addu %[t1], %[t1], %[s1] \n\t" - "addu %[t2], %[t2], %[s2] \n\t" - "addu %[t3], %[t3], %[s3] \n\t" - "sw %[t0], 0(%[ptr1]) \n\t" - "sw %[t1], 4(%[ptr1]) \n\t" - "sw %[t2], 8(%[ptr1]) \n\t" - "sw %[t3], 12(%[ptr1]) \n\t" - "bgtz %[n], 1b \n\t" - " addiu %[ptr1], %[ptr1], 16 \n\t" - "beq %[m], %0, 3f \n\t" - " nop \n\t" - "2: \n\t" - "lw %[r0], 0(%[ptr0]) \n\t" - "lw %[t0], 0(%[ptr2]) \n\t" - "addiu %[ptr0], %[ptr0], 4 \n\t" - "mult $ac0, %[r0], %[input0] \n\t" - "mult $ac1, %[r0], %[input1] \n\t" - "extr_rs.w %[r1], $ac0, 15 \n\t" - "extr_rs.w %[t1], $ac1, 15 \n\t" - "addu %[t0], %[t0], %[r1] \n\t" - "mult $ac0, %[t0], %[input2] \n\t" - "extr_rs.w %[t0], $ac0, 16 \n\t" - "sw %[t0], 0(%[ptr2]) \n\t" - "mult $ac0, %[t0], %[input0] \n\t" - "addiu %[ptr2], %[ptr2], 4 \n\t" - "addiu %[m], %[m], -1 \n\t" - "extr_rs.w %[t0], $ac0, 15 \n\t" - "addu %[t0], %[t0], %[t1] \n\t" - "sw %[t0], 0(%[ptr1]) \n\t" - "bgtz %[m], 2b \n\t" - " addiu %[ptr1], %[ptr1], 4 \n\t" - "3: \n\t" - ".set pop \n\t" - : [r0] "=&r" (r0), [r1] "=&r" (r1), [r2] "=&r" (r2), - [r3] "=&r" (r3), [t0] "=&r" (t0), [t1] "=&r" (t1), - [t2] "=&r" (t2), [t3] "=&r" (t3), [s0] "=&r" (s0), - [s1] "=&r" (s1), [s2] "=&r" (s2), [s3] "=&r" (s3), - [ptr0] "+r" (ptr0), [ptr1] "+r" (ptr1), [m] "+r" (m), - [ptr2] "+r" (ptr2), [n] "+r" (n) - : [input0] "r" (input0), [input1] "r" (input1), - [input2] "r" (input2) - : "memory", "hi", "lo", "$ac1hi", "$ac1lo", "$ac2hi", - "$ac2lo", "$ac3hi", "$ac3lo" - ); -#else - // Non-DSPR2 version of the function. Avoiding the accumulator usage due to - // large latencies. This variant is bit-exact with C code. - int n = HALF_SUBFRAMELEN - 1; - int32_t t16a, t16b; - int32_t r0, r1, r2, r3, r4; - - __asm __volatile ( - ".set push \n\t" - ".set noreorder \n\t" - "sra %[t16a], %[input2], 16 \n\t" - "andi %[t16b], %[input2], 0xFFFF \n\t" -#if defined(MIPS32R2_LE) - "seh %[t16b], %[t16b] \n\t" - "seh %[input0], %[input0] \n\t" - "seh %[input1], %[input1] \n\t" -#else - "sll %[t16b], %[t16b], 16 \n\t" - "sra %[t16b], %[t16b], 16 \n\t" - "sll %[input0], %[input0], 16 \n\t" - "sra %[input0], %[input0], 16 \n\t" - "sll %[input1], %[input1], 16 \n\t" - "sra %[input1], %[input1], 16 \n\t" -#endif - "addiu %[r0], %[t16a], 1 \n\t" - "slt %[r1], %[t16b], $zero \n\t" - "movn %[t16a], %[r0], %[r1] \n\t" - "1: \n\t" - "lw %[r0], 0(%[ptr0]) \n\t" - "lw %[r1], 0(%[ptr2]) \n\t" - "addiu %[ptr0], %[ptr0], 4 \n\t" - "sra %[r2], %[r0], 16 \n\t" - "andi %[r0], %[r0], 0xFFFF \n\t" - "mul %[r3], %[r2], %[input0] \n\t" - "mul %[r4], %[r0], %[input0] \n\t" - "mul %[r2], %[r2], %[input1] \n\t" - "mul %[r0], %[r0], %[input1] \n\t" - "addiu %[ptr2], %[ptr2], 4 \n\t" - "sll %[r3], %[r3], 1 \n\t" - "sra %[r4], %[r4], 1 \n\t" - "addiu %[r4], %[r4], 0x2000 \n\t" - "sra %[r4], %[r4], 14 \n\t" - "addu %[r3], %[r3], %[r4] \n\t" - "addu %[r1], %[r1], %[r3] \n\t" - "sra %[r3], %[r1], 16 \n\t" - "andi %[r4], %[r1], 0xFFFF \n\t" - "sra %[r4], %[r4], 1 \n\t" - "mul %[r1], %[r1], %[t16a] \n\t" - "mul %[r3], %[r3], %[t16b] \n\t" - "mul %[r4], %[r4], %[t16b] \n\t" - "sll %[r2], %[r2], 1 \n\t" - "sra %[r0], %[r0], 1 \n\t" - "addiu %[r0], %[r0], 0x2000 \n\t" - "sra %[r0], %[r0], 14 \n\t" - "addu %[r0], %[r0], %[r2] \n\t" - "addiu %[n], %[n], -1 \n\t" - "addu %[r1], %[r1], %[r3] \n\t" - "addiu %[r4], %[r4], 0x4000 \n\t" - "sra %[r4], %[r4], 15 \n\t" - "addu %[r1], %[r1], %[r4] \n\t" - "sra %[r2], %[r1], 16 \n\t" - "andi %[r3], %[r1], 0xFFFF \n\t" - "mul %[r3], %[r3], %[input0] \n\t" - "mul %[r2], %[r2], %[input0] \n\t" - "sw %[r1], -4(%[ptr2]) \n\t" - "sra %[r3], %[r3], 1 \n\t" - "addiu %[r3], %[r3], 0x2000 \n\t" - "sra %[r3], %[r3], 14 \n\t" - "addu %[r0], %[r0], %[r3] \n\t" - "sll %[r2], %[r2], 1 \n\t" - "addu %[r0], %[r0], %[r2] \n\t" - "sw %[r0], 0(%[ptr1]) \n\t" - "bgtz %[n], 1b \n\t" - " addiu %[ptr1], %[ptr1], 4 \n\t" - ".set pop \n\t" - : [t16a] "=&r" (t16a), [t16b] "=&r" (t16b), [r0] "=&r" (r0), - [r1] "=&r" (r1), [r2] "=&r" (r2), [r3] "=&r" (r3), - [r4] "=&r" (r4), [ptr0] "+r" (ptr0), [ptr1] "+r" (ptr1), - [ptr2] "+r" (ptr2), [n] "+r" (n) - : [input0] "r" (input0), [input1] "r" (input1), - [input2] "r" (input2) - : "hi", "lo", "memory" - ); -#endif -} diff --git a/modules/audio_coding/codecs/isac/fix/source/lattice_neon.c b/modules/audio_coding/codecs/isac/fix/source/lattice_neon.c deleted file mode 100644 index 8ea9b63578..0000000000 --- a/modules/audio_coding/codecs/isac/fix/source/lattice_neon.c +++ /dev/null @@ -1,195 +0,0 @@ -/* - * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#include - -#include "modules/audio_coding/codecs/isac/fix/source/codec.h" -#include "modules/audio_coding/codecs/isac/fix/source/settings.h" - -// Contains a function for the core loop in the normalized lattice MA -// filter routine for iSAC codec, optimized for ARM Neon platform. -// It does: -// for 0 <= n < HALF_SUBFRAMELEN - 1: -// *ptr2 = input2 * ((*ptr2) + input0 * (*ptr0)); -// *ptr1 = input1 * (*ptr0) + input0 * (*ptr2); -// Output is not bit-exact with the reference C code, due to the replacement -// of WEBRTC_SPL_MUL_16_32_RSFT15 and LATTICE_MUL_32_32_RSFT16 with Neon -// instructions. The difference should not be bigger than 1. -void WebRtcIsacfix_FilterMaLoopNeon(int16_t input0, // Filter coefficient - int16_t input1, // Filter coefficient - int32_t input2, // Inverse coefficient - int32_t* ptr0, // Sample buffer - int32_t* ptr1, // Sample buffer - int32_t* ptr2) // Sample buffer -{ - int n = 0; - int loop = (HALF_SUBFRAMELEN - 1) >> 3; - int loop_tail = (HALF_SUBFRAMELEN - 1) & 0x7; - - int32x4_t input0_v = vdupq_n_s32((int32_t)input0 << 16); - int32x4_t input1_v = vdupq_n_s32((int32_t)input1 << 16); - int32x4_t input2_v = vdupq_n_s32(input2); - int32x4_t tmp0a, tmp1a, tmp2a, tmp3a; - int32x4_t tmp0b, tmp1b, tmp2b, tmp3b; - int32x4_t ptr0va, ptr1va, ptr2va; - int32x4_t ptr0vb, ptr1vb, ptr2vb; - - int64x2_t tmp2al_low, tmp2al_high, tmp2bl_low, tmp2bl_high; - // Unroll to process 8 samples at once. - for (n = 0; n < loop; n++) { - ptr0va = vld1q_s32(ptr0); - ptr0vb = vld1q_s32(ptr0 + 4); - ptr0 += 8; - - ptr2va = vld1q_s32(ptr2); - ptr2vb = vld1q_s32(ptr2 + 4); - - // Calculate tmp0 = (*ptr0) * input0. - tmp0a = vqrdmulhq_s32(ptr0va, input0_v); - tmp0b = vqrdmulhq_s32(ptr0vb, input0_v); - - // Calculate tmp1 = (*ptr0) * input1. - tmp1a = vqrdmulhq_s32(ptr0va, input1_v); - tmp1b = vqrdmulhq_s32(ptr0vb, input1_v); - - // Calculate tmp2 = tmp0 + *(ptr2). - tmp2a = vaddq_s32(tmp0a, ptr2va); - tmp2b = vaddq_s32(tmp0b, ptr2vb); - - // Calculate *ptr2 = input2 * tmp2. - tmp2al_low = vmull_s32(vget_low_s32(tmp2a), vget_low_s32(input2_v)); -#if defined(WEBRTC_ARCH_ARM64) - tmp2al_high = vmull_high_s32(tmp2a, input2_v); -#else - tmp2al_high = vmull_s32(vget_high_s32(tmp2a), vget_high_s32(input2_v)); -#endif - ptr2va = vcombine_s32(vrshrn_n_s64(tmp2al_low, 16), - vrshrn_n_s64(tmp2al_high, 16)); - - tmp2bl_low = vmull_s32(vget_low_s32(tmp2b), vget_low_s32(input2_v)); -#if defined(WEBRTC_ARCH_ARM64) - tmp2bl_high = vmull_high_s32(tmp2b, input2_v); -#else - tmp2bl_high = vmull_s32(vget_high_s32(tmp2b), vget_high_s32(input2_v)); -#endif - ptr2vb = vcombine_s32(vrshrn_n_s64(tmp2bl_low, 16), - vrshrn_n_s64(tmp2bl_high, 16)); - - vst1q_s32(ptr2, ptr2va); - vst1q_s32(ptr2 + 4, ptr2vb); - ptr2 += 8; - - // Calculate tmp3 = ptr2v * input0. - tmp3a = vqrdmulhq_s32(ptr2va, input0_v); - tmp3b = vqrdmulhq_s32(ptr2vb, input0_v); - - // Calculate *ptr1 = tmp1 + tmp3. - ptr1va = vaddq_s32(tmp1a, tmp3a); - ptr1vb = vaddq_s32(tmp1b, tmp3b); - - vst1q_s32(ptr1, ptr1va); - vst1q_s32(ptr1 + 4, ptr1vb); - ptr1 += 8; - } - - // Process four more samples. - if (loop_tail & 0x4) { - ptr0va = vld1q_s32(ptr0); - ptr2va = vld1q_s32(ptr2); - ptr0 += 4; - - // Calculate tmp0 = (*ptr0) * input0. - tmp0a = vqrdmulhq_s32(ptr0va, input0_v); - - // Calculate tmp1 = (*ptr0) * input1. - tmp1a = vqrdmulhq_s32(ptr0va, input1_v); - - // Calculate tmp2 = tmp0 + *(ptr2). - tmp2a = vaddq_s32(tmp0a, ptr2va); - - // Calculate *ptr2 = input2 * tmp2. - tmp2al_low = vmull_s32(vget_low_s32(tmp2a), vget_low_s32(input2_v)); - -#if defined(WEBRTC_ARCH_ARM64) - tmp2al_high = vmull_high_s32(tmp2a, input2_v); -#else - tmp2al_high = vmull_s32(vget_high_s32(tmp2a), vget_high_s32(input2_v)); -#endif - ptr2va = vcombine_s32(vrshrn_n_s64(tmp2al_low, 16), - vrshrn_n_s64(tmp2al_high, 16)); - - vst1q_s32(ptr2, ptr2va); - ptr2 += 4; - - // Calculate tmp3 = *(ptr2) * input0. - tmp3a = vqrdmulhq_s32(ptr2va, input0_v); - - // Calculate *ptr1 = tmp1 + tmp3. - ptr1va = vaddq_s32(tmp1a, tmp3a); - - vst1q_s32(ptr1, ptr1va); - ptr1 += 4; - } - - // Process two more samples. - if (loop_tail & 0x2) { - int32x2_t ptr0v_tail, ptr2v_tail, ptr1v_tail; - int32x2_t tmp0_tail, tmp1_tail, tmp2_tail, tmp3_tail; - int64x2_t tmp2l_tail; - ptr0v_tail = vld1_s32(ptr0); - ptr2v_tail = vld1_s32(ptr2); - ptr0 += 2; - - // Calculate tmp0 = (*ptr0) * input0. - tmp0_tail = vqrdmulh_s32(ptr0v_tail, vget_low_s32(input0_v)); - - // Calculate tmp1 = (*ptr0) * input1. - tmp1_tail = vqrdmulh_s32(ptr0v_tail, vget_low_s32(input1_v)); - - // Calculate tmp2 = tmp0 + *(ptr2). - tmp2_tail = vadd_s32(tmp0_tail, ptr2v_tail); - - // Calculate *ptr2 = input2 * tmp2. - tmp2l_tail = vmull_s32(tmp2_tail, vget_low_s32(input2_v)); - ptr2v_tail = vrshrn_n_s64(tmp2l_tail, 16); - - vst1_s32(ptr2, ptr2v_tail); - ptr2 += 2; - - // Calculate tmp3 = *(ptr2) * input0. - tmp3_tail = vqrdmulh_s32(ptr2v_tail, vget_low_s32(input0_v)); - - // Calculate *ptr1 = tmp1 + tmp3. - ptr1v_tail = vadd_s32(tmp1_tail, tmp3_tail); - - vst1_s32(ptr1, ptr1v_tail); - ptr1 += 2; - } - - // Process one more sample. - if (loop_tail & 0x1) { - int16_t t16a = (int16_t)(input2 >> 16); - int16_t t16b = (int16_t)input2; - if (t16b < 0) t16a++; - int32_t tmp32a; - int32_t tmp32b; - - // Calculate *ptr2 = input2 * (*ptr2 + input0 * (*ptr0)). - tmp32a = WEBRTC_SPL_MUL_16_32_RSFT15(input0, *ptr0); - tmp32b = *ptr2 + tmp32a; - *ptr2 = (int32_t)(WEBRTC_SPL_MUL(t16a, tmp32b) + - (WEBRTC_SPL_MUL_16_32_RSFT16(t16b, tmp32b))); - - // Calculate *ptr1 = input1 * (*ptr0) + input0 * (*ptr2). - tmp32a = WEBRTC_SPL_MUL_16_32_RSFT15(input1, *ptr0); - tmp32b = WEBRTC_SPL_MUL_16_32_RSFT15(input0, *ptr2); - *ptr1 = tmp32a + tmp32b; - } -} diff --git a/modules/audio_coding/codecs/isac/fix/source/lpc_masking_model.c b/modules/audio_coding/codecs/isac/fix/source/lpc_masking_model.c deleted file mode 100644 index f151cd1c88..0000000000 --- a/modules/audio_coding/codecs/isac/fix/source/lpc_masking_model.c +++ /dev/null @@ -1,949 +0,0 @@ -/* - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/* - * lpc_masking_model.c - * - * LPC analysis and filtering functions - * - */ - -#include "lpc_masking_model.h" - -#include /* For LLONG_MAX and LLONG_MIN. */ - -#include "modules/audio_coding/codecs/isac/fix/source/codec.h" -#include "modules/audio_coding/codecs/isac/fix/source/entropy_coding.h" -#include "modules/audio_coding/codecs/isac/fix/source/settings.h" - -/* The conversion is implemented by the step-down algorithm */ -void WebRtcSpl_AToK_JSK( - int16_t *a16, /* Q11 */ - int16_t useOrder, - int16_t *k16 /* Q15 */ - ) -{ - int m, k; - int32_t tmp32[MAX_AR_MODEL_ORDER]; - int32_t tmp32b; - int32_t tmp_inv_denum32; - int16_t tmp_inv_denum16; - - k16[useOrder-1] = a16[useOrder] << 4; // Q11<<4 => Q15 - - for (m=useOrder-1; m>0; m--) { - // (1 - k^2) in Q30 - tmp_inv_denum32 = 1073741823 - k16[m] * k16[m]; - tmp_inv_denum16 = (int16_t)(tmp_inv_denum32 >> 15); // (1 - k^2) in Q15. - - for (k=1; k<=m; k++) { - tmp32b = (a16[k] << 16) - ((k16[m] * a16[m - k + 1]) << 1); - - tmp32[k] = WebRtcSpl_DivW32W16(tmp32b, tmp_inv_denum16); //Q27/Q15 = Q12 - } - - for (k=1; k> 1); // Q12>>1 => Q11 - } - - tmp32[m] = WEBRTC_SPL_SAT(4092, tmp32[m], -4092); - k16[m - 1] = (int16_t)(tmp32[m] << 3); // Q12<<3 => Q15 - } - - return; -} - - - - - -int16_t WebRtcSpl_LevinsonW32_JSK( - int32_t *R, /* (i) Autocorrelation of length >= order+1 */ - int16_t *A, /* (o) A[0..order] LPC coefficients (Q11) */ - int16_t *K, /* (o) K[0...order-1] Reflection coefficients (Q15) */ - int16_t order /* (i) filter order */ - ) { - int16_t i, j; - int16_t R_hi[LEVINSON_MAX_ORDER+1], R_low[LEVINSON_MAX_ORDER+1]; - /* Aurocorr coefficients in high precision */ - int16_t A_hi[LEVINSON_MAX_ORDER+1], A_low[LEVINSON_MAX_ORDER+1]; - /* LPC coefficients in high precicion */ - int16_t A_upd_hi[LEVINSON_MAX_ORDER+1], A_upd_low[LEVINSON_MAX_ORDER+1]; - /* LPC coefficients for next iteration */ - int16_t K_hi, K_low; /* reflection coefficient in high precision */ - int16_t Alpha_hi, Alpha_low, Alpha_exp; /* Prediction gain Alpha in high precision - and with scale factor */ - int16_t tmp_hi, tmp_low; - int32_t temp1W32, temp2W32, temp3W32; - int16_t norm; - - /* Normalize the autocorrelation R[0]...R[order+1] */ - - norm = WebRtcSpl_NormW32(R[0]); - - for (i=order;i>=0;i--) { - temp1W32 = R[i] << norm; - /* Put R in hi and low format */ - R_hi[i] = (int16_t)(temp1W32 >> 16); - R_low[i] = (int16_t)((temp1W32 - ((int32_t)R_hi[i] << 16)) >> 1); - } - - /* K = A[1] = -R[1] / R[0] */ - - temp2W32 = (R_hi[1] << 16) + (R_low[1] << 1); /* R[1] in Q31 */ - temp3W32 = WEBRTC_SPL_ABS_W32(temp2W32); /* abs R[1] */ - temp1W32 = WebRtcSpl_DivW32HiLow(temp3W32, R_hi[0], R_low[0]); /* abs(R[1])/R[0] in Q31 */ - /* Put back the sign on R[1] */ - if (temp2W32 > 0) { - temp1W32 = -temp1W32; - } - - /* Put K in hi and low format */ - K_hi = (int16_t)(temp1W32 >> 16); - K_low = (int16_t)((temp1W32 - ((int32_t)K_hi << 16)) >> 1); - - /* Store first reflection coefficient */ - K[0] = K_hi; - - temp1W32 >>= 4; /* A[1] in Q27. */ - - /* Put A[1] in hi and low format */ - A_hi[1] = (int16_t)(temp1W32 >> 16); - A_low[1] = (int16_t)((temp1W32 - ((int32_t)A_hi[1] << 16)) >> 1); - - /* Alpha = R[0] * (1-K^2) */ - - temp1W32 = (((K_hi * K_low) >> 14) + K_hi * K_hi) << 1; /* = k^2 in Q31 */ - - temp1W32 = WEBRTC_SPL_ABS_W32(temp1W32); /* Guard against <0 */ - temp1W32 = (int32_t)0x7fffffffL - temp1W32; /* temp1W32 = (1 - K[0]*K[0]) in Q31 */ - - /* Store temp1W32 = 1 - K[0]*K[0] on hi and low format */ - tmp_hi = (int16_t)(temp1W32 >> 16); - tmp_low = (int16_t)((temp1W32 - ((int32_t)tmp_hi << 16)) >> 1); - - /* Calculate Alpha in Q31 */ - temp1W32 = (R_hi[0] * tmp_hi + ((R_hi[0] * tmp_low) >> 15) + - ((R_low[0] * tmp_hi) >> 15)) << 1; - - /* Normalize Alpha and put it in hi and low format */ - - Alpha_exp = WebRtcSpl_NormW32(temp1W32); - temp1W32 <<= Alpha_exp; - Alpha_hi = (int16_t)(temp1W32 >> 16); - Alpha_low = (int16_t)((temp1W32 - ((int32_t)Alpha_hi<< 16)) >> 1); - - /* Perform the iterative calculations in the - Levinson Durbin algorithm */ - - for (i=2; i<=order; i++) - { - - /* ---- - \ - temp1W32 = R[i] + > R[j]*A[i-j] - / - ---- - j=1..i-1 - */ - - temp1W32 = 0; - - for(j=1; j> 15) + - ((R_low[j] * A_hi[i - j]) >> 15)) << 1); - } - - temp1W32 <<= 4; - temp1W32 += (R_hi[i] << 16) + (R_low[i] << 1); - - /* K = -temp1W32 / Alpha */ - temp2W32 = WEBRTC_SPL_ABS_W32(temp1W32); /* abs(temp1W32) */ - temp3W32 = WebRtcSpl_DivW32HiLow(temp2W32, Alpha_hi, Alpha_low); /* abs(temp1W32)/Alpha */ - - /* Put the sign of temp1W32 back again */ - if (temp1W32 > 0) { - temp3W32 = -temp3W32; - } - - /* Use the Alpha shifts from earlier to denormalize */ - norm = WebRtcSpl_NormW32(temp3W32); - if ((Alpha_exp <= norm)||(temp3W32==0)) { - temp3W32 <<= Alpha_exp; - } else { - if (temp3W32 > 0) - { - temp3W32 = (int32_t)0x7fffffffL; - } else - { - temp3W32 = (int32_t)0x80000000L; - } - } - - /* Put K on hi and low format */ - K_hi = (int16_t)(temp3W32 >> 16); - K_low = (int16_t)((temp3W32 - ((int32_t)K_hi << 16)) >> 1); - - /* Store Reflection coefficient in Q15 */ - K[i-1] = K_hi; - - /* Test for unstable filter. If unstable return 0 and let the - user decide what to do in that case - */ - - if ((int32_t)WEBRTC_SPL_ABS_W16(K_hi) > (int32_t)32740) { - return(-i); /* Unstable filter */ - } - - /* - Compute updated LPC coefficient: Anew[i] - Anew[j]= A[j] + K*A[i-j] for j=1..i-1 - Anew[i]= K - */ - - for(j=1; j> 15) + - ((K_low * A_hi[i - j]) >> 15)) << 1; // temp1W32 += K*A[i-j] in Q27. - - /* Put Anew in hi and low format */ - A_upd_hi[j] = (int16_t)(temp1W32 >> 16); - A_upd_low[j] = (int16_t)((temp1W32 - ((int32_t)A_upd_hi[j] << 16)) >> 1); - } - - temp3W32 >>= 4; /* temp3W32 = K in Q27 (Convert from Q31 to Q27) */ - - /* Store Anew in hi and low format */ - A_upd_hi[i] = (int16_t)(temp3W32 >> 16); - A_upd_low[i] = (int16_t)((temp3W32 - ((int32_t)A_upd_hi[i] << 16)) >> 1); - - /* Alpha = Alpha * (1-K^2) */ - - temp1W32 = (((K_hi * K_low) >> 14) + K_hi * K_hi) << 1; /* K*K in Q31 */ - - temp1W32 = WEBRTC_SPL_ABS_W32(temp1W32); /* Guard against <0 */ - temp1W32 = (int32_t)0x7fffffffL - temp1W32; /* 1 - K*K in Q31 */ - - /* Convert 1- K^2 in hi and low format */ - tmp_hi = (int16_t)(temp1W32 >> 16); - tmp_low = (int16_t)((temp1W32 - ((int32_t)tmp_hi << 16)) >> 1); - - /* Calculate Alpha = Alpha * (1-K^2) in Q31 */ - temp1W32 = (Alpha_hi * tmp_hi + ((Alpha_hi * tmp_low) >> 15) + - ((Alpha_low * tmp_hi) >> 15)) << 1; - - /* Normalize Alpha and store it on hi and low format */ - - norm = WebRtcSpl_NormW32(temp1W32); - temp1W32 <<= norm; - - Alpha_hi = (int16_t)(temp1W32 >> 16); - Alpha_low = (int16_t)((temp1W32 - ((int32_t)Alpha_hi << 16)) >> 1); - - /* Update the total nomalization of Alpha */ - Alpha_exp = Alpha_exp + norm; - - /* Update A[] */ - - for(j=1; j<=i; j++) - { - A_hi[j] =A_upd_hi[j]; - A_low[j] =A_upd_low[j]; - } - } - - /* - Set A[0] to 1.0 and store the A[i] i=1...order in Q12 - (Convert from Q27 and use rounding) - */ - - A[0] = 2048; - - for(i=1; i<=order; i++) { - /* temp1W32 in Q27 */ - temp1W32 = (A_hi[i] << 16) + (A_low[i] << 1); - /* Round and store upper word */ - A[i] = (int16_t)((temp1W32 + 32768) >> 16); - } - return(1); /* Stable filters */ -} - - - - - -/* window */ -/* Matlab generation of floating point code: - * t = (1:256)/257; r = 1-(1-t).^.45; w = sin(r*pi).^3; w = w/sum(w); plot((1:256)/8, w); grid; - * for k=1:16, fprintf(1, '%.8f, ', w(k*16 + (-15:0))); fprintf(1, '\n'); end - * All values are multiplyed with 2^21 in fixed point code. - */ -static const int16_t kWindowAutocorr[WINLEN] = { - 0, 0, 0, 0, 0, 1, 1, 2, 2, 3, 5, 6, - 8, 10, 12, 14, 17, 20, 24, 28, 33, 38, 43, 49, - 56, 63, 71, 79, 88, 98, 108, 119, 131, 143, 157, 171, - 186, 202, 219, 237, 256, 275, 296, 318, 341, 365, 390, 416, - 444, 472, 502, 533, 566, 600, 635, 671, 709, 748, 789, 831, - 875, 920, 967, 1015, 1065, 1116, 1170, 1224, 1281, 1339, 1399, 1461, - 1525, 1590, 1657, 1726, 1797, 1870, 1945, 2021, 2100, 2181, 2263, 2348, - 2434, 2523, 2614, 2706, 2801, 2898, 2997, 3099, 3202, 3307, 3415, 3525, - 3637, 3751, 3867, 3986, 4106, 4229, 4354, 4481, 4611, 4742, 4876, 5012, - 5150, 5291, 5433, 5578, 5725, 5874, 6025, 6178, 6333, 6490, 6650, 6811, - 6974, 7140, 7307, 7476, 7647, 7820, 7995, 8171, 8349, 8529, 8711, 8894, - 9079, 9265, 9453, 9642, 9833, 10024, 10217, 10412, 10607, 10803, 11000, 11199, - 11398, 11597, 11797, 11998, 12200, 12401, 12603, 12805, 13008, 13210, 13412, 13614, - 13815, 14016, 14216, 14416, 14615, 14813, 15009, 15205, 15399, 15591, 15782, 15971, - 16157, 16342, 16524, 16704, 16881, 17056, 17227, 17395, 17559, 17720, 17877, 18030, - 18179, 18323, 18462, 18597, 18727, 18851, 18970, 19082, 19189, 19290, 19384, 19471, - 19551, 19623, 19689, 19746, 19795, 19835, 19867, 19890, 19904, 19908, 19902, 19886, - 19860, 19823, 19775, 19715, 19644, 19561, 19465, 19357, 19237, 19102, 18955, 18793, - 18618, 18428, 18223, 18004, 17769, 17518, 17252, 16970, 16672, 16357, 16025, 15677, - 15311, 14929, 14529, 14111, 13677, 13225, 12755, 12268, 11764, 11243, 10706, 10152, - 9583, 8998, 8399, 7787, 7162, 6527, 5883, 5231, 4576, 3919, 3265, 2620, - 1990, 1386, 825, 333 -}; - - -/* By using a hearing threshold level in dB of -28 dB (higher value gives more noise), - the H_T_H (in float) can be calculated as: - H_T_H = pow(10.0, 0.05 * (-28.0)) = 0.039810717055350 - In Q19, H_T_H becomes round(0.039810717055350*2^19) ~= 20872, i.e. - H_T_H = 20872/524288.0, and H_T_HQ19 = 20872; -*/ - - -/* The bandwidth expansion vectors are created from: - kPolyVecLo=[0.900000,0.810000,0.729000,0.656100,0.590490,0.531441,0.478297,0.430467,0.387420,0.348678,0.313811,0.282430]; - kPolyVecHi=[0.800000,0.640000,0.512000,0.409600,0.327680,0.262144]; - round(kPolyVecLo*32768) - round(kPolyVecHi*32768) -*/ -static const int16_t kPolyVecLo[12] = { - 29491, 26542, 23888, 21499, 19349, 17414, 15673, 14106, 12695, 11425, 10283, 9255 -}; -static const int16_t kPolyVecHi[6] = { - 26214, 20972, 16777, 13422, 10737, 8590 -}; - -static __inline int32_t log2_Q8_LPC( uint32_t x ) { - - int32_t zeros; - int16_t frac; - - zeros=WebRtcSpl_NormU32(x); - frac = (int16_t)(((x << zeros) & 0x7FFFFFFF) >> 23); - - /* log2(x) */ - return ((31 - zeros) << 8) + frac; -} - -static const int16_t kMulPitchGain = -25; /* 200/256 in Q5 */ -static const int16_t kChngFactor = 3523; /* log10(2)*10/4*0.4/1.4=log10(2)/1.4= 0.2150 in Q14 */ -static const int16_t kExp2 = 11819; /* 1/log(2) */ -const int kShiftLowerBand = 11; /* Shift value for lower band in Q domain. */ -const int kShiftHigherBand = 12; /* Shift value for higher band in Q domain. */ - -void WebRtcIsacfix_GetVars(const int16_t *input, const int16_t *pitchGains_Q12, - uint32_t *oldEnergy, int16_t *varscale) -{ - int k; - uint32_t nrgQ[4]; - int16_t nrgQlog[4]; - int16_t tmp16, chng1, chng2, chng3, chng4, tmp, chngQ, oldNrgQlog, pgQ, pg3; - int32_t expPg32; - int16_t expPg, divVal; - int16_t tmp16_1, tmp16_2; - - /* Calculate energies of first and second frame halfs */ - nrgQ[0]=0; - for (k = QLOOKAHEAD/2; k < (FRAMESAMPLES/4 + QLOOKAHEAD) / 2; k++) { - nrgQ[0] += (uint32_t)(input[k] * input[k]); - } - nrgQ[1]=0; - for ( ; k < (FRAMESAMPLES/2 + QLOOKAHEAD) / 2; k++) { - nrgQ[1] += (uint32_t)(input[k] * input[k]); - } - nrgQ[2]=0; - for ( ; k < (FRAMESAMPLES * 3 / 4 + QLOOKAHEAD) / 2; k++) { - nrgQ[2] += (uint32_t)(input[k] * input[k]); - } - nrgQ[3]=0; - for ( ; k < (FRAMESAMPLES + QLOOKAHEAD) / 2; k++) { - nrgQ[3] += (uint32_t)(input[k] * input[k]); - } - - for ( k=0; k<4; k++) { - nrgQlog[k] = (int16_t)log2_Q8_LPC(nrgQ[k]); /* log2(nrgQ) */ - } - oldNrgQlog = (int16_t)log2_Q8_LPC(*oldEnergy); - - /* Calculate average level change */ - chng1 = WEBRTC_SPL_ABS_W16(nrgQlog[3]-nrgQlog[2]); - chng2 = WEBRTC_SPL_ABS_W16(nrgQlog[2]-nrgQlog[1]); - chng3 = WEBRTC_SPL_ABS_W16(nrgQlog[1]-nrgQlog[0]); - chng4 = WEBRTC_SPL_ABS_W16(nrgQlog[0]-oldNrgQlog); - tmp = chng1+chng2+chng3+chng4; - chngQ = (int16_t)(tmp * kChngFactor >> 10); /* Q12 */ - chngQ += 2926; /* + 1.0/1.4 in Q12 */ - - /* Find average pitch gain */ - pgQ = 0; - for (k=0; k<4; k++) - { - pgQ += pitchGains_Q12[k]; - } - - pg3 = (int16_t)(pgQ * pgQ >> 11); // pgQ in Q(12+2)=Q14. Q14*Q14>>11 => Q17 - pg3 = (int16_t)(pgQ * pg3 >> 13); /* Q14*Q17>>13 =>Q18 */ - /* kMulPitchGain = -25 = -200 in Q-3. */ - pg3 = (int16_t)(pg3 * kMulPitchGain >> 5); // Q10 - tmp16=(int16_t)WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(kExp2,pg3,13);/* Q13*Q10>>13 => Q10*/ - if (tmp16<0) { - tmp16_2 = (0x0400 | (tmp16 & 0x03FF)); - tmp16_1 = ((uint16_t)(tmp16 ^ 0xFFFF) >> 10) - 3; /* Gives result in Q14 */ - if (tmp16_1<0) - expPg = -(tmp16_2 << -tmp16_1); - else - expPg = -(tmp16_2 >> tmp16_1); - } else - expPg = (int16_t) -16384; /* 1 in Q14, since 2^0=1 */ - - expPg32 = (int32_t)expPg << 8; /* Q22 */ - divVal = WebRtcSpl_DivW32W16ResW16(expPg32, chngQ); /* Q22/Q12=Q10 */ - - tmp16=(int16_t)WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(kExp2,divVal,13);/* Q13*Q10>>13 => Q10*/ - if (tmp16<0) { - tmp16_2 = (0x0400 | (tmp16 & 0x03FF)); - tmp16_1 = ((uint16_t)(tmp16 ^ 0xFFFF) >> 10) - 3; /* Gives result in Q14 */ - if (tmp16_1<0) - expPg = tmp16_2 << -tmp16_1; - else - expPg = tmp16_2 >> tmp16_1; - } else - expPg = (int16_t) 16384; /* 1 in Q14, since 2^0=1 */ - - *varscale = expPg-1; - *oldEnergy = nrgQ[3]; -} - - - -static __inline int16_t exp2_Q10_T(int16_t x) { // Both in and out in Q10 - - int16_t tmp16_1, tmp16_2; - - tmp16_2=(int16_t)(0x0400|(x&0x03FF)); - tmp16_1 = -(x >> 10); - if(tmp16_1>0) - return tmp16_2 >> tmp16_1; - else - return tmp16_2 << -tmp16_1; - -} - - -// Declare function pointers. -AutocorrFix WebRtcIsacfix_AutocorrFix; -CalculateResidualEnergy WebRtcIsacfix_CalculateResidualEnergy; - -/* This routine calculates the residual energy for LPC. - * Formula as shown in comments inside. - */ -int32_t WebRtcIsacfix_CalculateResidualEnergyC(int lpc_order, - int32_t q_val_corr, - int q_val_polynomial, - int16_t* a_polynomial, - int32_t* corr_coeffs, - int* q_val_residual_energy) { - int i = 0, j = 0; - int shift_internal = 0, shift_norm = 0; - int32_t tmp32 = 0, word32_high = 0, word32_low = 0, residual_energy = 0; - int64_t sum64 = 0, sum64_tmp = 0; - - for (i = 0; i <= lpc_order; i++) { - for (j = i; j <= lpc_order; j++) { - /* For the case of i == 0: residual_energy += - * a_polynomial[j] * corr_coeffs[i] * a_polynomial[j - i]; - * For the case of i != 0: residual_energy += - * a_polynomial[j] * corr_coeffs[i] * a_polynomial[j - i] * 2; - */ - - tmp32 = a_polynomial[j] * a_polynomial[j - i]; - /* tmp32 in Q(q_val_polynomial * 2). */ - if (i != 0) { - tmp32 <<= 1; - } - sum64_tmp = (int64_t)tmp32 * (int64_t)corr_coeffs[i]; - sum64_tmp >>= shift_internal; - - /* Test overflow and sum the result. */ - if(((sum64_tmp > 0 && sum64 > 0) && (LLONG_MAX - sum64 < sum64_tmp)) || - ((sum64_tmp < 0 && sum64 < 0) && (LLONG_MIN - sum64 > sum64_tmp))) { - /* Shift right for overflow. */ - shift_internal += 1; - sum64 >>= 1; - sum64 += sum64_tmp >> 1; - } else { - sum64 += sum64_tmp; - } - } - } - - word32_high = (int32_t)(sum64 >> 32); - word32_low = (int32_t)sum64; - - // Calculate the value of shifting (shift_norm) for the 64-bit sum. - if(word32_high != 0) { - shift_norm = 32 - WebRtcSpl_NormW32(word32_high); - residual_energy = (int32_t)(sum64 >> shift_norm); - } else { - if((word32_low & 0x80000000) != 0) { - shift_norm = 1; - residual_energy = (uint32_t)word32_low >> 1; - } else { - shift_norm = WebRtcSpl_NormW32(word32_low); - residual_energy = word32_low << shift_norm; - shift_norm = -shift_norm; - } - } - - /* Q(q_val_polynomial * 2) * Q(q_val_corr) >> shift_internal >> shift_norm - * = Q(q_val_corr - shift_internal - shift_norm + q_val_polynomial * 2) - */ - *q_val_residual_energy = q_val_corr - shift_internal - shift_norm - + q_val_polynomial * 2; - - return residual_energy; -} - -void WebRtcIsacfix_GetLpcCoef(int16_t *inLoQ0, - int16_t *inHiQ0, - MaskFiltstr_enc *maskdata, - int16_t snrQ10, - const int16_t *pitchGains_Q12, - int32_t *gain_lo_hiQ17, - int16_t *lo_coeffQ15, - int16_t *hi_coeffQ15) -{ - int k, n, ii; - int pos1, pos2; - int sh_lo, sh_hi, sh, ssh, shMem; - int16_t varscaleQ14; - - int16_t tmpQQlo, tmpQQhi; - int32_t tmp32; - int16_t tmp16,tmp16b; - - int16_t polyHI[ORDERHI+1]; - int16_t rcQ15_lo[ORDERLO], rcQ15_hi[ORDERHI]; - - - int16_t DataLoQ6[WINLEN], DataHiQ6[WINLEN]; - int32_t corrloQQ[ORDERLO+2]; - int32_t corrhiQQ[ORDERHI+1]; - int32_t corrlo2QQ[ORDERLO+1]; - int16_t scale; - int16_t QdomLO, QdomHI, newQdomHI, newQdomLO; - - int32_t res_nrgQQ; - int32_t sqrt_nrg; - - /* less-noise-at-low-frequencies factor */ - int16_t aaQ14; - - /* Multiplication with 1/sqrt(12) ~= 0.28901734104046 can be done by convertion to - Q15, i.e. round(0.28901734104046*32768) = 9471, and use 9471/32768.0 ~= 0.289032 - */ - int16_t snrq; - int shft; - - int16_t tmp16a; - int32_t tmp32a, tmp32b, tmp32c; - - int16_t a_LOQ11[ORDERLO+1]; - int16_t k_vecloQ15[ORDERLO]; - int16_t a_HIQ12[ORDERHI+1]; - int16_t k_vechiQ15[ORDERHI]; - - int16_t stab; - - snrq=snrQ10; - - /* SNR= C * 2 ^ (D * snrq) ; C=0.289, D=0.05*log2(10)=0.166 (~=172 in Q10)*/ - tmp16 = (int16_t)(snrq * 172 >> 10); // Q10 - tmp16b = exp2_Q10_T(tmp16); // Q10 - snrq = (int16_t)(tmp16b * 285 >> 10); // Q10 - - /* change quallevel depending on pitch gains and level fluctuations */ - WebRtcIsacfix_GetVars(inLoQ0, pitchGains_Q12, &(maskdata->OldEnergy), &varscaleQ14); - - /* less-noise-at-low-frequencies factor */ - /* Calculation of 0.35 * (0.5 + 0.5 * varscale) in fixpoint: - With 0.35 in Q16 (0.35 ~= 22938/65536.0 = 0.3500061) and varscaleQ14 in Q14, - we get Q16*Q14>>16 = Q14 - */ - aaQ14 = (int16_t)((22938 * (8192 + (varscaleQ14 >> 1)) + 32768) >> 16); - - /* Calculate tmp = (1.0 + aa*aa); in Q12 */ - tmp16 = (int16_t)(aaQ14 * aaQ14 >> 15); // Q14*Q14>>15 = Q13 - tmpQQlo = 4096 + (tmp16 >> 1); // Q12 + Q13>>1 = Q12. - - /* Calculate tmp = (1.0+aa) * (1.0+aa); */ - tmp16 = 8192 + (aaQ14 >> 1); // 1+a in Q13. - tmpQQhi = (int16_t)(tmp16 * tmp16 >> 14); // Q13*Q13>>14 = Q12 - - /* replace data in buffer by new look-ahead data */ - for (pos1 = 0; pos1 < QLOOKAHEAD; pos1++) { - maskdata->DataBufferLoQ0[pos1 + WINLEN - QLOOKAHEAD] = inLoQ0[pos1]; - } - - for (k = 0; k < SUBFRAMES; k++) { - - /* Update input buffer and multiply signal with window */ - for (pos1 = 0; pos1 < WINLEN - UPDATE/2; pos1++) { - maskdata->DataBufferLoQ0[pos1] = maskdata->DataBufferLoQ0[pos1 + UPDATE/2]; - maskdata->DataBufferHiQ0[pos1] = maskdata->DataBufferHiQ0[pos1 + UPDATE/2]; - DataLoQ6[pos1] = (int16_t)(maskdata->DataBufferLoQ0[pos1] * - kWindowAutocorr[pos1] >> 15); // Q0*Q21>>15 = Q6 - DataHiQ6[pos1] = (int16_t)(maskdata->DataBufferHiQ0[pos1] * - kWindowAutocorr[pos1] >> 15); // Q0*Q21>>15 = Q6 - } - pos2 = (int16_t)(k * UPDATE / 2); - for (n = 0; n < UPDATE/2; n++, pos1++) { - maskdata->DataBufferLoQ0[pos1] = inLoQ0[QLOOKAHEAD + pos2]; - maskdata->DataBufferHiQ0[pos1] = inHiQ0[pos2++]; - DataLoQ6[pos1] = (int16_t)(maskdata->DataBufferLoQ0[pos1] * - kWindowAutocorr[pos1] >> 15); // Q0*Q21>>15 = Q6 - DataHiQ6[pos1] = (int16_t)(maskdata->DataBufferHiQ0[pos1] * - kWindowAutocorr[pos1] >> 15); // Q0*Q21>>15 = Q6 - } - - /* Get correlation coefficients */ - /* The highest absolute value measured inside DataLo in the test set - For DataHi, corresponding value was 160. - - This means that it should be possible to represent the input values - to WebRtcSpl_AutoCorrelation() as Q6 values (since 307*2^6 = - 19648). Of course, Q0 will also work, but due to the low energy in - DataLo and DataHi, the outputted autocorrelation will be more accurate - and mimic the floating point code better, by being in an high as possible - Q-domain. - */ - - WebRtcIsacfix_AutocorrFix(corrloQQ,DataLoQ6,WINLEN, ORDERLO+1, &scale); - QdomLO = 12-scale; // QdomLO is the Q-domain of corrloQQ - sh_lo = WebRtcSpl_NormW32(corrloQQ[0]); - QdomLO += sh_lo; - for (ii=0; ii> 1) - - (WEBRTC_SPL_MUL_16_32_RSFT16(aaQ14, corrloQQ[1]) >> 2); - - /* Calculate corrlo2[n] = tmpQQlo * corrlo[n] - tmpQQlo * (corrlo[n-1] + corrlo[n+1]);*/ - for (n = 1; n <= ORDERLO; n++) { - - tmp32 = (corrloQQ[n - 1] >> 1) + (corrloQQ[n + 1] >> 1); // Q(QdomLO-1). - corrlo2QQ[n] = (WEBRTC_SPL_MUL_16_32_RSFT16(tmpQQlo, corrloQQ[n]) >> 1) - - (WEBRTC_SPL_MUL_16_32_RSFT16(aaQ14, tmp32) >> 2); - } - QdomLO -= 5; - - /* Calculate corrhi[n] = tmpQQhi * corrhi[n]; */ - for (n = 0; n <= ORDERHI; n++) { - corrhiQQ[n] = WEBRTC_SPL_MUL_16_32_RSFT16(tmpQQhi, corrhiQQ[n]); // Q(12+QdomHI-16) = Q(QdomHI-4) - } - QdomHI -= 4; - - /* add white noise floor */ - /* corrlo2QQ is in Q(QdomLO) and corrhiQQ is in Q(QdomHI) */ - /* Calculate corrlo2[0] += 9.5367431640625e-7; and - corrhi[0] += 9.5367431640625e-7, where the constant is 1/2^20 */ - - tmp32 = WEBRTC_SPL_SHIFT_W32((int32_t) 1, QdomLO-20); - corrlo2QQ[0] += tmp32; - tmp32 = WEBRTC_SPL_SHIFT_W32((int32_t) 1, QdomHI-20); - corrhiQQ[0] += tmp32; - - /* corrlo2QQ is in Q(QdomLO) and corrhiQQ is in Q(QdomHI) before the following - code segment, where we want to make sure we get a 1-bit margin */ - for (n = 0; n <= ORDERLO; n++) { - corrlo2QQ[n] >>= 1; // Make sure we have a 1-bit margin. - } - QdomLO -= 1; // Now, corrlo2QQ is in Q(QdomLO), with a 1-bit margin - - for (n = 0; n <= ORDERHI; n++) { - corrhiQQ[n] >>= 1; // Make sure we have a 1-bit margin. - } - QdomHI -= 1; // Now, corrhiQQ is in Q(QdomHI), with a 1-bit margin - - - newQdomLO = QdomLO; - - for (n = 0; n <= ORDERLO; n++) { - int32_t tmp, tmpB, tmpCorr; - int16_t alpha=328; //0.01 in Q15 - int16_t beta=324; //(1-0.01)*0.01=0.0099 in Q15 - int16_t gamma=32440; //(1-0.01)=0.99 in Q15 - - if (maskdata->CorrBufLoQQ[n] != 0) { - shMem=WebRtcSpl_NormW32(maskdata->CorrBufLoQQ[n]); - sh = QdomLO - maskdata->CorrBufLoQdom[n]; - if (sh<=shMem) { - tmp = WEBRTC_SPL_SHIFT_W32(maskdata->CorrBufLoQQ[n], sh); // Get CorrBufLoQQ to same domain as corrlo2 - tmp = WEBRTC_SPL_MUL_16_32_RSFT15(alpha, tmp); - } else if ((sh-shMem)<7){ - tmp = WEBRTC_SPL_SHIFT_W32(maskdata->CorrBufLoQQ[n], shMem); // Shift up CorrBufLoQQ as much as possible - // Shift `alpha` the number of times required to get `tmp` in QdomLO. - tmp = WEBRTC_SPL_MUL_16_32_RSFT15(alpha << (sh - shMem), tmp); - } else { - tmp = WEBRTC_SPL_SHIFT_W32(maskdata->CorrBufLoQQ[n], shMem); // Shift up CorrBufHiQQ as much as possible - // Shift `alpha` as much as possible without overflow the number of - // times required to get `tmp` in QdomLO. - tmp = WEBRTC_SPL_MUL_16_32_RSFT15(alpha << 6, tmp); - tmpCorr = corrloQQ[n] >> (sh - shMem - 6); - tmp = tmp + tmpCorr; - maskdata->CorrBufLoQQ[n] = tmp; - newQdomLO = QdomLO-(sh-shMem-6); - maskdata->CorrBufLoQdom[n] = newQdomLO; - } - } else - tmp = 0; - - tmp = tmp + corrlo2QQ[n]; - - maskdata->CorrBufLoQQ[n] = tmp; - maskdata->CorrBufLoQdom[n] = QdomLO; - - tmp=WEBRTC_SPL_MUL_16_32_RSFT15(beta, tmp); - tmpB=WEBRTC_SPL_MUL_16_32_RSFT15(gamma, corrlo2QQ[n]); - corrlo2QQ[n] = tmp + tmpB; - } - if( newQdomLO!=QdomLO) { - for (n = 0; n <= ORDERLO; n++) { - if (maskdata->CorrBufLoQdom[n] != newQdomLO) - corrloQQ[n] >>= maskdata->CorrBufLoQdom[n] - newQdomLO; - } - QdomLO = newQdomLO; - } - - - newQdomHI = QdomHI; - - for (n = 0; n <= ORDERHI; n++) { - int32_t tmp, tmpB, tmpCorr; - int16_t alpha=328; //0.01 in Q15 - int16_t beta=324; //(1-0.01)*0.01=0.0099 in Q15 - int16_t gamma=32440; //(1-0.01)=0.99 in Q1 - if (maskdata->CorrBufHiQQ[n] != 0) { - shMem=WebRtcSpl_NormW32(maskdata->CorrBufHiQQ[n]); - sh = QdomHI - maskdata->CorrBufHiQdom[n]; - if (sh<=shMem) { - tmp = WEBRTC_SPL_SHIFT_W32(maskdata->CorrBufHiQQ[n], sh); // Get CorrBufHiQQ to same domain as corrhi - tmp = WEBRTC_SPL_MUL_16_32_RSFT15(alpha, tmp); - tmpCorr = corrhiQQ[n]; - tmp = tmp + tmpCorr; - maskdata->CorrBufHiQQ[n] = tmp; - maskdata->CorrBufHiQdom[n] = QdomHI; - } else if ((sh-shMem)<7) { - tmp = WEBRTC_SPL_SHIFT_W32(maskdata->CorrBufHiQQ[n], shMem); // Shift up CorrBufHiQQ as much as possible - // Shift `alpha` the number of times required to get `tmp` in QdomHI. - tmp = WEBRTC_SPL_MUL_16_32_RSFT15(alpha << (sh - shMem), tmp); - tmpCorr = corrhiQQ[n]; - tmp = tmp + tmpCorr; - maskdata->CorrBufHiQQ[n] = tmp; - maskdata->CorrBufHiQdom[n] = QdomHI; - } else { - tmp = WEBRTC_SPL_SHIFT_W32(maskdata->CorrBufHiQQ[n], shMem); // Shift up CorrBufHiQQ as much as possible - // Shift `alpha` as much as possible without overflow the number of - // times required to get `tmp` in QdomHI. - tmp = WEBRTC_SPL_MUL_16_32_RSFT15(alpha << 6, tmp); - tmpCorr = corrhiQQ[n] >> (sh - shMem - 6); - tmp = tmp + tmpCorr; - maskdata->CorrBufHiQQ[n] = tmp; - newQdomHI = QdomHI-(sh-shMem-6); - maskdata->CorrBufHiQdom[n] = newQdomHI; - } - } else { - tmp = corrhiQQ[n]; - tmpCorr = tmp; - maskdata->CorrBufHiQQ[n] = tmp; - maskdata->CorrBufHiQdom[n] = QdomHI; - } - - tmp=WEBRTC_SPL_MUL_16_32_RSFT15(beta, tmp); - tmpB=WEBRTC_SPL_MUL_16_32_RSFT15(gamma, tmpCorr); - corrhiQQ[n] = tmp + tmpB; - } - - if( newQdomHI!=QdomHI) { - for (n = 0; n <= ORDERHI; n++) { - if (maskdata->CorrBufHiQdom[n] != newQdomHI) - corrhiQQ[n] >>= maskdata->CorrBufHiQdom[n] - newQdomHI; - } - QdomHI = newQdomHI; - } - - stab=WebRtcSpl_LevinsonW32_JSK(corrlo2QQ, a_LOQ11, k_vecloQ15, ORDERLO); - - if (stab<0) { // If unstable use lower order - a_LOQ11[0]=2048; - for (n = 1; n <= ORDERLO; n++) { - a_LOQ11[n]=0; - } - - stab=WebRtcSpl_LevinsonW32_JSK(corrlo2QQ, a_LOQ11, k_vecloQ15, 8); - } - - - WebRtcSpl_LevinsonDurbin(corrhiQQ, a_HIQ12, k_vechiQ15, ORDERHI); - - /* bandwidth expansion */ - for (n = 1; n <= ORDERLO; n++) { - a_LOQ11[n] = (int16_t)((kPolyVecLo[n - 1] * a_LOQ11[n] + (1 << 14)) >> - 15); - } - - - polyHI[0] = a_HIQ12[0]; - for (n = 1; n <= ORDERHI; n++) { - a_HIQ12[n] = (int16_t)(((int32_t)(kPolyVecHi[n - 1] * a_HIQ12[n]) + - (1 << 14)) >> 15); - polyHI[n] = a_HIQ12[n]; - } - - /* Normalize the corrlo2 vector */ - sh = WebRtcSpl_NormW32(corrlo2QQ[0]); - for (n = 0; n <= ORDERLO; n++) { - corrlo2QQ[n] <<= sh; - } - QdomLO += sh; /* Now, corrlo2QQ is still in Q(QdomLO) */ - - - /* residual energy */ - - sh_lo = 31; - res_nrgQQ = WebRtcIsacfix_CalculateResidualEnergy(ORDERLO, QdomLO, - kShiftLowerBand, a_LOQ11, corrlo2QQ, &sh_lo); - - /* Convert to reflection coefficients */ - WebRtcSpl_AToK_JSK(a_LOQ11, ORDERLO, rcQ15_lo); - - if (sh_lo & 0x0001) { - res_nrgQQ >>= 1; - sh_lo-=1; - } - - - if( res_nrgQQ > 0 ) - { - sqrt_nrg=WebRtcSpl_Sqrt(res_nrgQQ); - - /* add hearing threshold and compute the gain */ - /* lo_coeff = varscale * S_N_R / (sqrt_nrg + varscale * H_T_H); */ - - tmp32a = varscaleQ14 >> 1; // H_T_HQ19=65536 (16-17=-1) - ssh = sh_lo >> 1; // sqrt_nrg is in Qssh. - sh = ssh - 14; - tmp32b = WEBRTC_SPL_SHIFT_W32(tmp32a, sh); // Q14->Qssh - tmp32c = sqrt_nrg + tmp32b; // Qssh (denominator) - tmp32a = varscaleQ14 * snrq; // Q24 (numerator) - - sh = WebRtcSpl_NormW32(tmp32c); - shft = 16 - sh; - tmp16a = (int16_t) WEBRTC_SPL_SHIFT_W32(tmp32c, -shft); // Q(ssh-shft) (denominator) - - tmp32b = WebRtcSpl_DivW32W16(tmp32a, tmp16a); // Q(24-ssh+shft) - sh = ssh-shft-7; - *gain_lo_hiQ17 = WEBRTC_SPL_SHIFT_W32(tmp32b, sh); // Gains in Q17 - } - else - { - *gain_lo_hiQ17 = 100; // Gains in Q17 - } - gain_lo_hiQ17++; - - /* copy coefficients to output array */ - for (n = 0; n < ORDERLO; n++) { - *lo_coeffQ15 = (int16_t) (rcQ15_lo[n]); - lo_coeffQ15++; - } - /* residual energy */ - sh_hi = 31; - res_nrgQQ = WebRtcIsacfix_CalculateResidualEnergy(ORDERHI, QdomHI, - kShiftHigherBand, a_HIQ12, corrhiQQ, &sh_hi); - - /* Convert to reflection coefficients */ - WebRtcSpl_LpcToReflCoef(polyHI, ORDERHI, rcQ15_hi); - - if (sh_hi & 0x0001) { - res_nrgQQ >>= 1; - sh_hi-=1; - } - - - if( res_nrgQQ > 0 ) - { - sqrt_nrg=WebRtcSpl_Sqrt(res_nrgQQ); - - - /* add hearing threshold and compute the gain */ - /* hi_coeff = varscale * S_N_R / (sqrt_nrg + varscale * H_T_H); */ - - tmp32a = varscaleQ14 >> 1; // H_T_HQ19=65536 (16-17=-1) - - ssh = sh_hi >> 1; // `sqrt_nrg` is in Qssh. - sh = ssh - 14; - tmp32b = WEBRTC_SPL_SHIFT_W32(tmp32a, sh); // Q14->Qssh - tmp32c = sqrt_nrg + tmp32b; // Qssh (denominator) - tmp32a = varscaleQ14 * snrq; // Q24 (numerator) - - sh = WebRtcSpl_NormW32(tmp32c); - shft = 16 - sh; - tmp16a = (int16_t) WEBRTC_SPL_SHIFT_W32(tmp32c, -shft); // Q(ssh-shft) (denominator) - - tmp32b = WebRtcSpl_DivW32W16(tmp32a, tmp16a); // Q(24-ssh+shft) - sh = ssh-shft-7; - *gain_lo_hiQ17 = WEBRTC_SPL_SHIFT_W32(tmp32b, sh); // Gains in Q17 - } - else - { - *gain_lo_hiQ17 = 100; // Gains in Q17 - } - gain_lo_hiQ17++; - - - /* copy coefficients to output array */ - for (n = 0; n < ORDERHI; n++) { - *hi_coeffQ15 = rcQ15_hi[n]; - hi_coeffQ15++; - } - } -} diff --git a/modules/audio_coding/codecs/isac/fix/source/lpc_masking_model.h b/modules/audio_coding/codecs/isac/fix/source/lpc_masking_model.h deleted file mode 100644 index 40a99e8a77..0000000000 --- a/modules/audio_coding/codecs/isac/fix/source/lpc_masking_model.h +++ /dev/null @@ -1,69 +0,0 @@ -/* - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/* - * lpc_masking_model.h - * - * LPC functions - * - */ - -#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_LPC_MASKING_MODEL_H_ -#define MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_LPC_MASKING_MODEL_H_ - -#ifdef __cplusplus -extern "C" { -#endif - -#include "modules/audio_coding/codecs/isac/fix/source/structs.h" - -void WebRtcIsacfix_GetVars(const int16_t* input, - const int16_t* pitchGains_Q12, - uint32_t* oldEnergy, - int16_t* varscale); - -void WebRtcIsacfix_GetLpcCoef(int16_t* inLoQ0, - int16_t* inHiQ0, - MaskFiltstr_enc* maskdata, - int16_t snrQ10, - const int16_t* pitchGains_Q12, - int32_t* gain_lo_hiQ17, - int16_t* lo_coeffQ15, - int16_t* hi_coeffQ15); - -typedef int32_t (*CalculateResidualEnergy)(int lpc_order, - int32_t q_val_corr, - int q_val_polynomial, - int16_t* a_polynomial, - int32_t* corr_coeffs, - int* q_val_residual_energy); -extern CalculateResidualEnergy WebRtcIsacfix_CalculateResidualEnergy; - -int32_t WebRtcIsacfix_CalculateResidualEnergyC(int lpc_order, - int32_t q_val_corr, - int q_val_polynomial, - int16_t* a_polynomial, - int32_t* corr_coeffs, - int* q_val_residual_energy); - -#if defined(MIPS_DSP_R2_LE) -int32_t WebRtcIsacfix_CalculateResidualEnergyMIPS(int lpc_order, - int32_t q_val_corr, - int q_val_polynomial, - int16_t* a_polynomial, - int32_t* corr_coeffs, - int* q_val_residual_energy); -#endif - -#ifdef __cplusplus -} /* extern "C" */ -#endif - -#endif /* MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_LPC_MASKING_MODEL_H_ */ diff --git a/modules/audio_coding/codecs/isac/fix/source/lpc_masking_model_mips.c b/modules/audio_coding/codecs/isac/fix/source/lpc_masking_model_mips.c deleted file mode 100644 index 727008da32..0000000000 --- a/modules/audio_coding/codecs/isac/fix/source/lpc_masking_model_mips.c +++ /dev/null @@ -1,237 +0,0 @@ -/* - * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#include "modules/audio_coding/codecs/isac/fix/source/lpc_masking_model.h" - -// MIPS DSPR2 optimization for function WebRtcIsacfix_CalculateResidualEnergy -// Bit-exact with WebRtcIsacfix_CalculateResidualEnergyC from file -// lpc_masking_model.c -int32_t WebRtcIsacfix_CalculateResidualEnergyMIPS(int lpc_order, - int32_t q_val_corr, - int q_val_polynomial, - int16_t* a_polynomial, - int32_t* corr_coeffs, - int* q_val_residual_energy) { - - int i = 0, j = 0; - int shift_internal = 0, shift_norm = 0; - int32_t tmp32 = 0, word32_high = 0, word32_low = 0, residual_energy = 0; - int32_t tmp_corr_c = corr_coeffs[0]; - int16_t* tmp_a_poly = &a_polynomial[0]; - int32_t sum64_hi = 0; - int32_t sum64_lo = 0; - - for (j = 0; j <= lpc_order; j++) { - // For the case of i == 0: - // residual_energy += - // a_polynomial[j] * corr_coeffs[i] * a_polynomial[j - i]; - - int32_t tmp2, tmp3; - int16_t sign_1; - int16_t sign_2; - int16_t sign_3; - - __asm __volatile ( - ".set push \n\t" - ".set noreorder \n\t" - "lh %[tmp2], 0(%[tmp_a_poly]) \n\t" - "mul %[tmp32], %[tmp2], %[tmp2] \n\t" - "addiu %[tmp_a_poly], %[tmp_a_poly], 2 \n\t" - "sra %[sign_2], %[sum64_hi], 31 \n\t" - "mult $ac0, %[tmp32], %[tmp_corr_c] \n\t" - "shilov $ac0, %[shift_internal] \n\t" - "mfhi %[tmp2], $ac0 \n\t" - "mflo %[tmp3], $ac0 \n\t" - "sra %[sign_1], %[tmp2], 31 \n\t" - "xor %[sign_3], %[sign_1], %[sign_2] \n\t" - ".set pop \n\t" - : [tmp2] "=&r" (tmp2), [tmp3] "=&r" (tmp3), [tmp32] "=&r" (tmp32), - [tmp_a_poly] "+r" (tmp_a_poly), [sign_1] "=&r" (sign_1), - [sign_3] "=&r" (sign_3), [sign_2] "=&r" (sign_2), - [sum64_hi] "+r" (sum64_hi), [sum64_lo] "+r" (sum64_lo) - : [tmp_corr_c] "r" (tmp_corr_c), [shift_internal] "r" (shift_internal) - : "hi", "lo", "memory" - ); - - if (sign_3 != 0) { - __asm __volatile ( - ".set push \n\t" - ".set noreorder \n\t" - "addsc %[sum64_lo], %[sum64_lo], %[tmp3] \n\t" - "addwc %[sum64_hi], %[sum64_hi], %[tmp2] \n\t" - ".set pop \n\t" - : [sum64_hi] "+r" (sum64_hi), [sum64_lo] "+r" (sum64_lo) - : [tmp2] "r" (tmp2), [tmp3] "r" (tmp3) - : "hi", "lo", "memory" - ); - } else { - if (((!(sign_1 || sign_2)) && (0x7FFFFFFF - sum64_hi < tmp2)) || - ((sign_1 && sign_2) && (sum64_hi + tmp2 > 0))) { - // Shift right for overflow. - __asm __volatile ( - ".set push \n\t" - ".set noreorder \n\t" - "addiu %[shift_internal], %[shift_internal], 1 \n\t" - "prepend %[sum64_lo], %[sum64_hi], 1 \n\t" - "sra %[sum64_hi], %[sum64_hi], 1 \n\t" - "prepend %[tmp3], %[tmp2], 1 \n\t" - "sra %[tmp2], %[tmp2], 1 \n\t" - "addsc %[sum64_lo], %[sum64_lo], %[tmp3] \n\t" - "addwc %[sum64_hi], %[sum64_hi], %[tmp2] \n\t" - ".set pop \n\t" - : [tmp2] "+r" (tmp2), [tmp3] "+r" (tmp3), - [shift_internal] "+r" (shift_internal), - [sum64_hi] "+r" (sum64_hi), [sum64_lo] "+r" (sum64_lo) - : - : "hi", "lo", "memory" - ); - } else { - __asm __volatile ( - ".set push \n\t" - ".set noreorder \n\t" - "addsc %[sum64_lo], %[sum64_lo], %[tmp3] \n\t" - "addwc %[sum64_hi], %[sum64_hi], %[tmp2] \n\t" - ".set pop \n\t" - : [sum64_hi] "+r" (sum64_hi), [sum64_lo] "+r" (sum64_lo) - : [tmp2] "r" (tmp2), [tmp3] "r" (tmp3) - : "hi", "lo", "memory" - ); - } - } - } - - for (i = 1; i <= lpc_order; i++) { - tmp_corr_c = corr_coeffs[i]; - int16_t* tmp_a_poly_j = &a_polynomial[i]; - int16_t* tmp_a_poly_j_i = &a_polynomial[0]; - for (j = i; j <= lpc_order; j++) { - // For the case of i = 1 .. lpc_order: - // residual_energy += - // a_polynomial[j] * corr_coeffs[i] * a_polynomial[j - i] * 2; - - int32_t tmp2, tmp3; - int16_t sign_1; - int16_t sign_2; - int16_t sign_3; - - __asm __volatile ( - ".set push \n\t" - ".set noreorder \n\t" - "lh %[tmp3], 0(%[tmp_a_poly_j]) \n\t" - "lh %[tmp2], 0(%[tmp_a_poly_j_i]) \n\t" - "addiu %[tmp_a_poly_j], %[tmp_a_poly_j], 2 \n\t" - "addiu %[tmp_a_poly_j_i], %[tmp_a_poly_j_i], 2 \n\t" - "mul %[tmp32], %[tmp3], %[tmp2] \n\t" - "sll %[tmp32], %[tmp32], 1 \n\t" - "mult $ac0, %[tmp32], %[tmp_corr_c] \n\t" - "shilov $ac0, %[shift_internal] \n\t" - "mfhi %[tmp2], $ac0 \n\t" - "mflo %[tmp3], $ac0 \n\t" - "sra %[sign_1], %[tmp2], 31 \n\t" - "sra %[sign_2], %[sum64_hi], 31 \n\t" - "xor %[sign_3], %[sign_1], %[sign_2] \n\t" - ".set pop \n\t" - : [tmp2] "=&r" (tmp2), [tmp3] "=&r" (tmp3), [tmp32] "=&r" (tmp32), - [tmp_a_poly_j] "+r" (tmp_a_poly_j), [sign_1] "=&r" (sign_1), - [tmp_a_poly_j_i] "+r" (tmp_a_poly_j_i), [sign_2] "=&r" (sign_2), - [sign_3] "=&r" (sign_3), [sum64_hi] "+r" (sum64_hi), - [sum64_lo] "+r" (sum64_lo) - : [tmp_corr_c] "r" (tmp_corr_c), [shift_internal] "r" (shift_internal) - : "hi", "lo", "memory" - ); - if (sign_3 != 0) { - __asm __volatile ( - ".set push \n\t" - ".set noreorder \n\t" - "addsc %[sum64_lo], %[sum64_lo], %[tmp3] \n\t" - "addwc %[sum64_hi], %[sum64_hi], %[tmp2] \n\t" - ".set pop \n\t" - : [tmp2] "+r" (tmp2), [tmp3] "+r" (tmp3), [sum64_hi] "+r" (sum64_hi), - [sum64_lo] "+r" (sum64_lo) - : - :"memory" - ); - } else { - // Test overflow and sum the result. - if (((!(sign_1 || sign_2)) && (0x7FFFFFFF - sum64_hi < tmp2)) || - ((sign_1 && sign_2) && (sum64_hi + tmp2 > 0))) { - // Shift right for overflow. - __asm __volatile ( - ".set push \n\t" - ".set noreorder \n\t" - "addiu %[shift_internal], %[shift_internal], 1 \n\t" - "prepend %[sum64_lo], %[sum64_hi], 1 \n\t" - "sra %[sum64_hi], %[sum64_hi], 1 \n\t" - "prepend %[tmp3], %[tmp2], 1 \n\t" - "sra %[tmp2], %[tmp2], 1 \n\t" - "addsc %[sum64_lo], %[sum64_lo], %[tmp3] \n\t" - "addwc %[sum64_hi], %[sum64_hi], %[tmp2] \n\t" - ".set pop \n\t" - : [tmp2] "+r" (tmp2), [tmp3] "+r" (tmp3), - [shift_internal] "+r" (shift_internal), - [sum64_hi] "+r" (sum64_hi), [sum64_lo] "+r" (sum64_lo) - : - : "hi", "lo", "memory" - ); - } else { - __asm __volatile ( - ".set push \n\t" - ".set noreorder \n\t" - "addsc %[sum64_lo], %[sum64_lo], %[tmp3] \n\t" - "addwc %[sum64_hi], %[sum64_hi], %[tmp2] \n\t" - ".set pop \n\t" - : [tmp2] "+r" (tmp2), [tmp3] "+r" (tmp3), - [sum64_hi] "+r" (sum64_hi), [sum64_lo] "+r" (sum64_lo) - : - : "hi", "lo", "memory" - ); - } - } - } - } - word32_high = sum64_hi; - word32_low = sum64_lo; - - // Calculate the value of shifting (shift_norm) for the 64-bit sum. - if (word32_high != 0) { - shift_norm = 32 - WebRtcSpl_NormW32(word32_high); - int tmp1; - __asm __volatile ( - ".set push \n\t" - ".set noreorder \n\t" - "srl %[residual_energy], %[sum64_lo], %[shift_norm] \n\t" - "li %[tmp1], 32 \n\t" - "subu %[tmp1], %[tmp1], %[shift_norm] \n\t" - "sll %[tmp1], %[sum64_hi], %[tmp1] \n\t" - "or %[residual_energy], %[residual_energy], %[tmp1] \n\t" - ".set pop \n\t" - : [residual_energy] "=&r" (residual_energy), [tmp1]"=&r"(tmp1), - [sum64_hi] "+r" (sum64_hi), [sum64_lo] "+r" (sum64_lo) - : [shift_norm] "r" (shift_norm) - : "memory" - ); - } else { - if ((word32_low & 0x80000000) != 0) { - shift_norm = 1; - residual_energy = (uint32_t)word32_low >> 1; - } else { - shift_norm = WebRtcSpl_NormW32(word32_low); - residual_energy = word32_low << shift_norm; - shift_norm = -shift_norm; - } - } - - // Q(q_val_polynomial * 2) * Q(q_val_corr) >> shift_internal >> shift_norm - // = Q(q_val_corr - shift_internal - shift_norm + q_val_polynomial * 2) - *q_val_residual_energy = - q_val_corr - shift_internal - shift_norm + q_val_polynomial * 2; - - return residual_energy; -} diff --git a/modules/audio_coding/codecs/isac/fix/source/lpc_masking_model_unittest.cc b/modules/audio_coding/codecs/isac/fix/source/lpc_masking_model_unittest.cc deleted file mode 100644 index 82793f1344..0000000000 --- a/modules/audio_coding/codecs/isac/fix/source/lpc_masking_model_unittest.cc +++ /dev/null @@ -1,61 +0,0 @@ -/* - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#include "modules/audio_coding/codecs/isac/fix/source/lpc_masking_model.h" - -#include "system_wrappers/include/cpu_features_wrapper.h" -#include "test/gtest.h" - -class LpcMaskingModelTest : public ::testing::Test { - protected: - // Pass a function pointer to the Tester function. - void CalculateResidualEnergyTester( - CalculateResidualEnergy CalculateResidualEnergyFunction) { - const int kIntOrder = 10; - const int32_t kInt32QDomain = 5; - const int kIntShift = 11; - int16_t a[kIntOrder + 1] = {32760, 122, 7, 0, -32760, -3958, - -48, 18745, 498, 9, 23456}; - int32_t corr[kIntOrder + 1] = {11443647, -27495, 0, 98745, -11443600, 1, - 1, 498, 9, 888, 23456}; - int q_shift_residual = 0; - int32_t residual_energy = 0; - - // Test the code path where (residual_energy >= 0x10000). - residual_energy = CalculateResidualEnergyFunction( - kIntOrder, kInt32QDomain, kIntShift, a, corr, &q_shift_residual); - EXPECT_EQ(1789023310, residual_energy); - EXPECT_EQ(2, q_shift_residual); - - // Test the code path where (residual_energy < 0x10000) - // and ((energy & 0x8000) != 0). - for (int i = 0; i < kIntOrder + 1; i++) { - a[i] = 24575 >> i; - corr[i] = i; - } - residual_energy = CalculateResidualEnergyFunction( - kIntOrder, kInt32QDomain, kIntShift, a, corr, &q_shift_residual); - EXPECT_EQ(1595279092, residual_energy); - EXPECT_EQ(26, q_shift_residual); - - // Test the code path where (residual_energy <= 0x7fff). - for (int i = 0; i < kIntOrder + 1; i++) { - a[i] = 2457 >> i; - } - residual_energy = CalculateResidualEnergyFunction( - kIntOrder, kInt32QDomain, kIntShift, a, corr, &q_shift_residual); - EXPECT_EQ(2029266944, residual_energy); - EXPECT_EQ(33, q_shift_residual); - } -}; - -TEST_F(LpcMaskingModelTest, CalculateResidualEnergyTest) { - CalculateResidualEnergyTester(WebRtcIsacfix_CalculateResidualEnergyC); -} diff --git a/modules/audio_coding/codecs/isac/fix/source/lpc_tables.c b/modules/audio_coding/codecs/isac/fix/source/lpc_tables.c deleted file mode 100644 index d495d29235..0000000000 --- a/modules/audio_coding/codecs/isac/fix/source/lpc_tables.c +++ /dev/null @@ -1,1281 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/* - * lpc_tables.c - * - * Coding tables for the KLT coefficients - * - */ - - -#include "modules/audio_coding/codecs/isac/fix/source/settings.h" -#include "modules/audio_coding/codecs/isac/fix/source/lpc_tables.h" - -/* indices of KLT coefficients used */ -const uint16_t WebRtcIsacfix_kSelIndGain[12] = { - 0, 1, 2, 3, 4, 5, 6, 7, 8, 9, - 10, 11}; - -const uint16_t WebRtcIsacfix_kSelIndShape[108] = { - 0, 1, 2, 3, 4, 5, 6, 7, 8, 9, - 10, 11, 12, 13, 14, 15, 16, 17, 18, 19, - 20, 21, 22, 23, 24, 25, 26, 27, 28, 29, - 30, 31, 32, 33, 34, 35, 36, 37, 38, 39, - 40, 41, 42, 43, 44, 45, 46, 47, 48, 49, - 50, 51, 52, 53, 54, 55, 56, 57, 58, 59, - 60, 61, 62, 63, 64, 65, 66, 67, 68, 69, - 70, 71, 72, 73, 74, 75, 76, 77, 78, 79, - 80, 81, 82, 83, 84, 85, 86, 87, 88, 89, - 90, 91, 92, 93, 94, 95, 96, 97, 98, 99, - 100, 101, 102, 103, 104, 105, 106, 107 -}; - -/* cdf array for model indicator */ -const uint16_t WebRtcIsacfix_kModelCdf[4] = { - 0, 15434, 37548, 65535 -}; - -/* pointer to cdf array for model indicator */ -const uint16_t *WebRtcIsacfix_kModelCdfPtr[1] = { - WebRtcIsacfix_kModelCdf -}; - -/* initial cdf index for decoder of model indicator */ -const uint16_t WebRtcIsacfix_kModelInitIndex[1] = { - 1 -}; - -/* offset to go from rounded value to quantization index */ -const int16_t WebRtcIsacfix_kQuantMinGain[12] ={ - 3, 6, 4, 6, 6, 9, 5, 16, 11, 34, 32, 47 -}; - -const int16_t WebRtcIsacfix_kQuantMinShape[108] = { - 0, 0, 0, 0, 0, 0, 0, 0, 0, 1, - 1, 1, 1, 1, 2, 2, 2, 3, 0, 0, - 0, 0, 1, 0, 0, 0, 0, 1, 1, 1, - 1, 1, 1, 2, 2, 3, 0, 0, 0, 0, - 1, 0, 1, 1, 1, 1, 1, 1, 1, 2, - 2, 4, 3, 5, 0, 0, 0, 0, 1, 1, - 1, 1, 1, 1, 2, 1, 2, 2, 3, 4, - 4, 7, 0, 0, 1, 1, 1, 1, 1, 1, - 1, 2, 3, 2, 3, 4, 4, 5, 7, 13, - 0, 1, 1, 2, 3, 2, 2, 2, 4, 4, - 5, 6, 7, 11, 9, 13, 12, 26 -}; - -/* maximum quantization index */ -const uint16_t WebRtcIsacfix_kMaxIndGain[12] = { - 6, 12, 8, 14, 10, 19, 12, 31, 22, 56, 52, 138 -}; - -const uint16_t WebRtcIsacfix_kMaxIndShape[108] = { - 0, 0, 0, 0, 0, 0, 0, 0, 0, 1, - 2, 2, 2, 2, 4, 4, 5, 6, 0, 0, - 0, 0, 1, 0, 0, 0, 0, 1, 2, 2, - 2, 2, 3, 4, 5, 7, 0, 0, 0, 0, - 2, 0, 2, 2, 2, 2, 3, 2, 2, 4, - 4, 6, 6, 9, 0, 0, 0, 0, 2, 2, - 2, 2, 2, 2, 3, 2, 4, 4, 7, 7, - 9, 13, 0, 0, 2, 2, 2, 2, 2, 2, - 3, 4, 5, 4, 6, 8, 8, 10, 16, 25, - 0, 2, 2, 4, 5, 4, 4, 4, 7, 8, - 9, 10, 13, 19, 17, 23, 25, 49 -}; - -/* index offset */ -const uint16_t WebRtcIsacfix_kOffsetGain[3][12] = { - { 0, 7, 20, 29, 44, 55, 75, 88, 120, 143, 200, 253}, - { 0, 7, 19, 27, 42, 53, 73, 86, 117, 140, 197, 249}, - { 0, 7, 20, 28, 44, 55, 75, 89, 121, 145, 202, 257} -}; - -const uint16_t WebRtcIsacfix_kOffsetShape[3][108] = { - { - 0, 1, 2, 3, 4, 5, 6, 7, 8, 9, - 11, 14, 17, 20, 23, 28, 33, 39, 46, 47, - 48, 49, 50, 52, 53, 54, 55, 56, 58, 61, - 64, 67, 70, 74, 79, 85, 93, 94, 95, 96, - 97, 100, 101, 104, 107, 110, 113, 117, 120, 123, - 128, 133, 140, 147, 157, 158, 159, 160, 161, 164, - 167, 170, 173, 176, 179, 183, 186, 191, 196, 204, - 212, 222, 236, 237, 238, 241, 244, 247, 250, 253, - 256, 260, 265, 271, 276, 283, 292, 301, 312, 329, - 355, 356, 359, 362, 367, 373, 378, 383, 388, 396, - 405, 415, 426, 440, 460, 478, 502, 528 - }, - { - 0, 1, 2, 3, 4, 6, 7, 8, 9, 11, - 13, 16, 19, 22, 26, 29, 34, 39, 45, 46, - 47, 48, 49, 50, 51, 52, 53, 55, 57, 60, - 63, 66, 70, 73, 78, 84, 91, 92, 93, 94, - 95, 96, 97, 99, 102, 105, 108, 111, 114, 118, - 123, 128, 134, 141, 151, 152, 153, 154, 156, 159, - 162, 165, 168, 171, 174, 177, 181, 186, 194, 200, - 208, 218, 233, 234, 235, 236, 239, 242, 245, 248, - 251, 254, 258, 263, 270, 277, 288, 297, 308, 324, - 349, 351, 354, 357, 361, 366, 372, 378, 383, 390, - 398, 407, 420, 431, 450, 472, 496, 524 - }, - { - 0, 1, 2, 3, 4, 5, 6, 7, 8, 11, - 14, 17, 20, 23, 26, 29, 34, 40, 47, 48, - 49, 50, 51, 52, 53, 54, 55, 58, 61, 64, - 67, 70, 73, 77, 82, 88, 96, 97, 98, 99, - 101, 102, 104, 107, 110, 113, 116, 119, 122, 125, - 129, 134, 141, 150, 160, 161, 162, 163, 166, 168, - 171, 174, 177, 180, 183, 186, 190, 195, 201, 208, - 216, 226, 243, 244, 245, 248, 251, 254, 257, 260, - 263, 268, 273, 278, 284, 291, 299, 310, 323, 340, - 366, 368, 371, 374, 379, 383, 389, 394, 399, 406, - 414, 422, 433, 445, 461, 480, 505, 533 - } -}; - -/* initial cdf index for KLT coefficients */ -const uint16_t WebRtcIsacfix_kInitIndexGain[3][12] = { - { 3, 6, 4, 7, 5, 10, 6, 16, 11, 28, 26, 69}, - { 3, 6, 4, 7, 5, 10, 6, 15, 11, 28, 26, 69}, - { 3, 6, 4, 8, 5, 10, 7, 16, 12, 28, 27, 70} -}; - -const uint16_t WebRtcIsacfix_kInitIndexShape[3][108] = { - { - 0, 0, 0, 0, 0, 0, 0, 0, 0, 1, - 1, 1, 1, 1, 2, 2, 3, 3, 0, 0, - 0, 0, 1, 0, 0, 0, 0, 1, 1, 1, - 1, 1, 2, 2, 3, 4, 0, 0, 0, 0, - 1, 0, 1, 1, 1, 1, 2, 1, 1, 2, - 2, 3, 3, 5, 0, 0, 0, 0, 1, 1, - 1, 1, 1, 1, 2, 1, 2, 2, 4, 4, - 5, 7, 0, 0, 1, 1, 1, 1, 1, 1, - 2, 2, 3, 2, 3, 4, 4, 5, 8, 13, - 0, 1, 1, 2, 3, 2, 2, 2, 4, 4, - 5, 5, 7, 10, 9, 12, 13, 25 - }, - { - 0, 0, 0, 0, 1, 0, 0, 0, 1, 1, - 1, 1, 1, 2, 1, 2, 2, 3, 0, 0, - 0, 0, 0, 0, 0, 0, 1, 1, 1, 1, - 1, 2, 1, 2, 3, 3, 0, 0, 0, 0, - 0, 0, 1, 1, 1, 1, 1, 1, 2, 2, - 2, 3, 3, 5, 0, 0, 0, 1, 1, 1, - 1, 1, 1, 1, 1, 2, 2, 4, 3, 4, - 5, 7, 0, 0, 0, 1, 1, 1, 1, 1, - 1, 2, 2, 3, 3, 5, 4, 5, 8, 12, - 1, 1, 1, 2, 2, 3, 3, 2, 3, 4, - 4, 6, 5, 9, 11, 12, 14, 25 - }, - { - 0, 0, 0, 0, 0, 0, 0, 0, 1, 1, - 1, 1, 1, 1, 1, 2, 3, 3, 0, 0, - 0, 0, 0, 0, 0, 0, 1, 1, 1, 1, - 1, 1, 2, 2, 3, 4, 0, 0, 0, 1, - 0, 1, 1, 1, 1, 1, 1, 1, 1, 2, - 2, 3, 4, 5, 0, 0, 0, 1, 1, 1, - 1, 1, 1, 1, 1, 2, 2, 3, 3, 4, - 5, 8, 0, 0, 1, 1, 1, 1, 1, 1, - 2, 2, 2, 3, 3, 4, 5, 6, 8, 13, - 1, 1, 1, 2, 2, 3, 2, 2, 3, 4, - 4, 5, 6, 8, 9, 12, 14, 25 - } -}; - -/* offsets for quantizer representation levels*/ -const uint16_t WebRtcIsacfix_kOfLevelsGain[3] = { - 0, 392, 779 -}; - -const uint16_t WebRtcIsacfix_kOfLevelsShape[3] = { - 0, 578, 1152 -}; - -/* quantizer representation levels */ - - - -const int32_t WebRtcIsacfix_kLevelsGainQ17[1176] = { - -364547,-231664,-102123,-573,104807,238257,368823,-758583,-640135,-510291 - ,-377662,-252785,-113177,2627,112906,248601,389461,522691,644517,763974 - ,-538963,-368179,-245823,-106095,-890,104299,241111,350730,493190,-800763 - ,-646230,-510239,-382115,-248623,-111829,-2983,113852,251105,388114,519757 - ,644048,774712,896334,1057931,-770009,-635310,-503690,-375087,-248106,-108525 - ,-105,108259,243472,377948,519271,-1160885,-1032044,-914636,-777593,-647891 - ,-518408,-388028,-254321,-115293,-598,117849,251296,385367,515510,652727 - ,777432,920363,1038625,1153637,1316836,-632287,-505430,-379430,-248458,-118013 - ,-888,118762,250266,381650,513327,652169,766471,932113,-2107480,-1971030 - ,-1827020,-1698263,-1558670,-1436314,-1305377,-1172252,-1047355,-914202,-779651,-651001 - ,-520999,-390394,-255761,-123490,-1893,126839,256703,385493,518607,651760 - ,782750,908693,1044131,1163865,1311066,1424177,1582628,1709823,1831740,1955391 - ,-1423044,-1288917,-1181281,-1043222,-911770,-780354,-646799,-522664,-386721,-258266 - ,-128060,-1101,128233,259996,390336,519590,649290,778701,908010,1040796 - ,1161235,1306889,1441882,-4446002,-4301031,-4194304,-4080591,-3947740,-3808975,-3686530 - ,-3567839,-3383251,-3287089,-3136577,-3017405,-2869860,-2751321,-2619984,-2482932,-2354790 - ,-2223147,-2090669,-1964135,-1831208,-1706697,-1570817,-1446008,-1305386,-1175773,-1046066 - ,-915356,-785120,-653614,-524331,-393767,-260442,-130187,-799,128841,261466 - ,393616,520542,652117,784613,914159,1045399,1181072,1308971,1442502,1570346 - ,1693912,1843986,1966014,2090474,2224869,2364593,2475934,2628403,2752512,2856640 - ,-4192441,-4063232,-3917821,-3799195,-3666233,-3519199,-3411021,-3269192,-3135684,-3008826 - ,-2880875,-2747342,-2620981,-2494872,-2354979,-2229718,-2098939,-1964971,-1835399,-1703452 - ,-1572806,-1440482,-1311794,-1179338,-1046521,-919823,-785914,-655335,-523416,-395507 - ,-264833,-132184,-2546,131698,256217,391372,522688,651248,789964,909618 - ,1035305,1179145,1313824,1436934,1552353,1693722,1815508,1972826,2096328,2228224 - ,2359296,2490368,2598848,-6160384,-6029312,-5881382,-5767168,-5636096,-5505024,-5373952 - ,-5228418,-5110384,-4954923,-4880576,-4710990,-4587364,-4471340,-4333905,-4211513,-4051293 - ,-3907927,-3800105,-3675961,-3538640,-3413663,-3271148,-3152105,-3019103,-2869647,-2744015 - ,-2620639,-2479385,-2364211,-2227611,-2095427,-1974497,-1834168,-1703561,-1568961,-1439826 - ,-1309192,-1174050,-1050191,-917836,-786015,-656943,-518934,-394831,-257708,-128041 - ,1610,128991,264442,393977,521383,653849,788164,918641,1049122,1181971 - ,1308934,1439505,1571808,1706305,1836318,1966235,2097269,2228990,2357005,2490292 - ,2617400,2749680,2881234,3014880,3145637,3276467,3409099,3536637,3671493,3802918 - ,3929740,4065036,4194143,4325999,4456126,4586857,4717194,4843923,4978676,5110913 - ,5245281,5371394,5499780,5633779,5762611,5897682,6028688,6167546,6296465,6421682 - ,6548882,6682074,6809432,6941956,7078143,7204509,7334296,7475137,7609896,7732044 - ,7861604,8002039,8131670,8259222,8390299,8522399,8650037,8782348,8908402,9037815 - ,9164594,9300338,9434679,9574500,9699702,9833934,9948152,10083972,10244937,10332822 - ,10485760,10600122,10760754,10892964,11010048,11111004,11272192,11403264,11525091,11624984 - ,11796480,11915146,-393216,-262144,-101702,-740,100568,262144,393216,-786432 - ,-655360,-524288,-383907,-243301,-94956,-156,95547,269629,416691,524288 - ,655360,-393216,-262144,-88448,-37,87318,262144,393216,524288,-917504 - ,-786432,-655360,-495894,-373308,-267503,-93211,4119,91308,250895,393216 - ,526138,655360,786432,917504,-786432,-655360,-524288,-393216,-262144,-83497 - ,222,86893,240922,393216,524288,-1048576,-917504,-790472,-655360,-508639 - ,-383609,-262016,-95550,-3775,96692,256797,364847,534906,655360,786432 - ,889679,1048576,1179648,1310720,1441792,-655360,-524288,-377684,-248408,-93690 - ,1261,95441,227519,393216,524288,655360,786432,917504,-2097152,-1966080 - ,-1809470,-1703936,-1572864,-1441792,-1314289,-1195149,-1056205,-917504,-809951,-657769 - ,-521072,-383788,-248747,-106350,-2944,105550,243408,388548,521064,628732 - ,786432,885456,1064548,1179648,1310720,1441792,1572864,1703936,1835008,-1441792 - ,-1310720,-1179648,-1037570,-888492,-767774,-646634,-519935,-373458,-248029,-111915 - ,760,111232,247735,379432,507672,672699,786432,917504,1048576,1179648 - ,1310720,1441792,-4456448,-4325376,-4194304,-4063232,-3932160,-3801088,-3670016,-3538944 - ,-3407872,-3276800,-3145728,-3014656,-2883584,-2752512,-2647002,-2490368,-2359296,-2228224 - ,-2097152,-1951753,-1835008,-1703936,-1594177,-1462001,-1289150,-1160774,-1025917,-924928 - ,-782509,-641294,-516191,-386630,-251910,-118886,5210,121226,253949,386008 - ,517973,649374,780064,917783,1052462,1183856,1290593,1419389,1556641,1699884 - ,1835008,1988314,2090470,2228224,2359296,2490368,2621440,2752512,2883584,-3801088 - ,-3643514,-3539937,-3409931,-3263294,-3145658,-3012952,-2879230,-2752359,-2622556,-2483471 - ,-2357556,-2226500,-2093112,-1965892,-1833664,-1701035,-1567767,-1440320,-1310556,-1178339 - ,-1049625,-916812,-786477,-655277,-525050,-393773,-264828,-130696,-480,132126 - ,260116,394197,527846,652294,785563,917183,1049511,1175958,1308161,1438759 - ,1572253,1698835,1828535,1967072,2089391,2212798,2348901,2461547,2621440,2752512 - ,2883584,-7309870,-7203780,-7062699,-6939106,-6790819,-6672036,-6553600,-6422317,-6288422 - ,-6164694,-6026456,-5901410,-5754168,-5621459,-5502710,-5369686,-5240454,-5120712,-4976140 - ,-4847970,-4723070,-4589083,-4450923,-4324680,-4189892,-4065551,-3931803,-3800209,-3668539 - ,-3539395,-3404801,-3277470,-3141389,-3016710,-2885724,-2752612,-2618541,-2486762,-2354153 - ,-2225059,-2094984,-1968194,-1830895,-1699508,-1575743,-1444516,-1308683,-1179714,-1053088 - ,-917981,-783707,-653900,-524980,-395409,-260309,-131948,-3452,132113,263241 - ,392185,522597,654134,788288,919810,1045795,1179210,1314201,1444235,1574447 - ,1705193,1834009,1967332,2098102,2229019,2359147,2489859,2619878,2754966,2879671 - ,3014438,3146143,3276733,3405958,3542196,3667493,3798815,3932961,4062458,4187125 - ,4322346,4454875,4587752,4716809,4848274,4975027,5111957,5242215,5373085,5501158 - ,5640140,5762918,5895358,6024008,6157906,6290628,6422713,6546339,6675888,6815606 - ,6955288,7077501,7211630,7337893,7473635,7607175,7728310,7866475,7999658,8127888 - ,8241758,8386483,8522550,8641582,8771915,8922139,9038632,9179385,9313426,9437184 - ,9568256,9699328,9830400,9952933,10120004,10223616,10354688,10474645,10616832,-393216 - ,-262144,-85425,-121,82533,262144,393216,-786432,-655360,-524288,-379928 - ,-222821,-95200,287,95541,227093,393216,493567,655360,786432,-393216 - ,-262144,-86805,510,86722,262144,393216,524288,-1048576,-917504,-786432 - ,-624456,-529951,-395071,-241627,-101168,81,99975,241605,393216,524288 - ,655360,786432,917504,-786432,-655360,-524288,-393216,-230359,-95619,-137 - ,94425,226222,393216,524288,-1179648,-1048576,-917504,-773841,-655360,-492258 - ,-379715,-244707,-103621,-434,104523,242680,381575,523659,650565,786432 - ,917504,1048576,1179648,1310720,-786432,-629344,-524288,-376757,-242858,-101932 - ,-2715,107155,239212,366480,514943,655360,786432,917504,-2228224,-2097152 - ,-1966080,-1835008,-1703936,-1572864,-1441792,-1284584,-1179648,-1048819,-934658,-777181 - ,-626371,-515660,-377493,-248975,-113036,436,113584,248354,379718,512475 - ,653932,796494,917504,1048576,1179648,1310720,1441792,1572864,1703936,1835008 - ,-1572864,-1441792,-1297608,-1161159,-1032316,-917092,-779770,-647384,-515529,-384269 - ,-250003,-119252,1053,118111,249512,380545,512039,648101,770656,907003 - ,1021725,1178082,1310720,1441792,-4587520,-4456448,-4325376,-4194304,-4063232,-3932160 - ,-3801088,-3670016,-3538944,-3407872,-3276800,-3145728,-2999335,-2883584,-2752512,-2621440 - ,-2490368,-2359296,-2228224,-2112691,-1966080,-1848781,-1709830,-1566109,-1438427,-1303530 - ,-1176124,-1040936,-913876,-784585,-652025,-518361,-385267,-256342,-127297,-2733 - ,125422,257792,389363,519911,651106,783805,909407,1044143,1174156,1309267 - ,1436173,1553771,1708958,1814083,1967036,2095386,2255169,2359296,2478303,2621440 - ,2752512,-4456448,-4325376,-4194304,-4063232,-3932160,-3797524,-3670016,-3560250,-3413217 - ,-3257719,-3166416,-2986626,-2878000,-2781144,-2625383,-2495465,-2346792,-2230930,-2077063 - ,-1949225,-1819274,-1697261,-1568664,-1443074,-1304302,-1175289,-1043794,-913423,-785561 - ,-652104,-522835,-392667,-260517,-130088,-2,129509,260990,391931,522470 - ,655770,784902,917093,1046445,1176951,1303121,1441362,1565401,1702022,1822856 - ,1952852,2090384,2214607,2338436,2457483,2621440,-8781824,-8650752,-8519680,-8388608 - ,-8260828,-8126464,-8003337,-7859030,-7750057,-7602176,-7471104,-7340032,-7193045,-7090588 - ,-6946816,-6843344,-6676635,-6557575,-6447804,-6277614,-6159736,-6035729,-5884723,-5739567 - ,-5634818,-5489867,-5372864,-5243300,-5098939,-4988639,-4856258,-4728494,-4591717,-4447428 - ,-4322409,-4192918,-4062638,-3934141,-3797545,-3673373,-3531587,-3407391,-3277404,-3147797 - ,-3013578,-2886548,-2749811,-2616428,-2490949,-2361301,-2228482,-2096883,-1964343,-1831754 - ,-1702201,-1572495,-1442012,-1309242,-1182451,-1048996,-916905,-786510,-657079,-524730 - ,-393672,-261313,-128743,166,130678,261334,393287,524155,655570,786839 - ,917353,1052167,1179013,1309360,1442634,1571153,1703961,1832027,1965014,2097912 - ,2224861,2355341,2490455,2623051,2753484,2877015,3015783,3144157,3273705,3405255 - ,3542006,3669580,3802417,3935413,4065088,4190896,4333521,4456355,4579781,4713832 - ,4845707,4978625,5113278,5243817,5382318,5500592,5638135,5761179,5900822,6029270 - ,6186398,6297816,6436435,6559163,6666389,6806548,6950461,7086078,7195777,7350973 - ,7480132,7614852,7743514,7847288,8014762,8126464,8257536,8388608,8519680,8650752 - ,8781824,8912896,9043968,9175040,9306112,9437184 -}; - - - -const int16_t WebRtcIsacfix_kLevelsShapeQ10[1735] = { - 0, 0, -1, 0, 0, 1, 0, 1, 0, -821 - , 1, -763, -1, 656, -620, 0, 633, -636, 4, 615 - , -630, 1, 649, -1773, -670, 5, 678, 1810, -1876, -676 - , 0, 691, 1843, -1806, -743, -1, 749, 1795, 2920, -2872 - , -1761, -772, -3, 790, 1763, 2942, 0, 0, 0, 0 - , -792, 2, 0, 0, 1, 0, -854, 0, -702, -1 - , 662, -624, -5, 638, -611, -6, 638, -647, 0, 651 - , -685, -4, 679, 2123, -1814, -693, 0, 664, 1791, -1735 - , -737, 0, 771, 1854, 2873, -2867, -1842, -793, -1, 821 - , 1826, 2805, 3922, 0, 0, 0, -1, -779, 1, 786 - , 1, -708, 0, 789, -799, 1, 797, -663, 2, 646 - , -600, 3, 609, -600, 1, 658, 1807, -627, -3, 612 - , -625, 3, 632, -1732, -674, 1, 672, 2048, -1768, -715 - , 0, 724, 1784, -3881, -3072, -1774, -719, -1, 730, 1811 - , -2963, -1829, -806, -1, 816, 1795, 3050, -5389, -3784, -2942 - , -1893, -865, -12, 867, 1885, 2945, 3928, -2, 1, 4 - , 0, -694, 2, 665, -598, 5, 587, -599, -1, 661 - , -656, -7, 611, -607, 5, 603, -618, -4, 620, -1794 - , -645, -2, 654, -655, -1, 658, -1801, -700, 5, 707 - , 1927, -1752, -745, -8, 752, 1843, -2838, -1781, -801, 11 - , 796, 1811, 2942, 3866, -3849, -3026, -1848, -819, 2, 827 - , 1825, 2963, -3873, -2904, -1869, -910, -6, 903, 1902, 2885 - , 3978, 5286, -7168, -6081, -4989, -3968, -2963, -1970, -943, -2 - , 953, 1951, 2968, 3974, 5009, 6032, -2, 3, -1024, 2 - , 1024, -637, 1, 669, -613, -7, 630, -603, 4, 612 - , -612, 0, 590, -645, -11, 627, -657, -2, 671, 1849 - , -1853, -694, 2, 702, 1838, -3304, -1780, -736, -8, 732 - , 1772, -1709, -755, -6, 760, 1780, -2994, -1780, -800, 8 - , 819, 1830, 2816, -4096, -2822, -1881, -851, -4, 855, 1872 - , 2840, 3899, -3908, -2904, -1878, -887, 6, 897, 1872, 2942 - , 4008, -4992, -3881, -2933, -1915, -928, 1, 937, 1919, 2900 - , 4009, 4881, -6848, -6157, -5065, -3981, -2983, -1972, -978, -1 - , 968, 1979, 2988, 4008, 5007, 6108, 7003, 8051, 9027,-13272 - ,-12012,-11228,-10213, -9261, -8084, -7133, -6075, -5052, -4050, -3036 - , -2014, -996, -4, 1007, 2031, 3038, 4049, 5074, 6134, 7069 - , 8094, 9069, 10212, 11049, 12104, 51, -1024, -13, 1024, -609 - , -107, 613, -2048, -687, -95, 667, 2048, -3072, -1724, -785 - , -34, 732, 1819, -2048, -703, -26, 681, 2048, -2048, -686 - , -9, 665, 2048, -2048, -702, 37, 748, 1723, -4096, -2786 - , -1844, -837, 37, 811, 1742, 3072, -4096, -2783, -1848, -881 - , 39, 898, 1843, 2792, 3764, -5120, -4096, -2923, -1833, -852 - , -14, 862, 1824, 2834, 4096, -6144, -5120, -3914, -2842, -1870 - , -886, -27, 888, 1929, 2931, 4051, -7168, -6144, -5120, -3866 - , -2933, -1915, -927, 64, 933, 1902, 2929, 3912, 5063, 6144 - ,-11264,-10240, -9216, -8192, -7086, -6144, -5039, -3972, -2943, -1929 - , -941, 3, 938, 1942, 2959, 3933, 4905, 6088, 6983, 8192 - , -9216, -8192, -7202, -6088, -4983, -4019, -2955, -1975, -966, 17 - , 997, 1981, 2967, 3990, 4948, 6022, 6967, 8192,-13312,-12288 - ,-11264,-10240, -9216, -8049, -6997, -6040, -5026, -4043, -3029, -2034 - , -1015, -23, 984, 1997, 3010, 4038, 5002, 6015, 6946, 8061 - , 9216, 10240,-12381,-11264,-10240, -9060, -8058, -7153, -6085, -5075 - , -4051, -3042, -2037, -1017, -5, 1007, 2028, 3035, 4050, 5088 - , 6111, 7160, 8156, 9215, 10095, 11229, 12202, 13016,-26624,-25600 - ,-24582,-23671,-22674,-21400,-20355,-19508,-18315,-17269,-16361,-15299 - ,-14363,-13294,-12262,-11237,-10203, -9227, -8165, -7156, -6116, -5122 - , -4076, -3056, -2043, -1020, -8, 1027, 2047, 3065, 4110, 5130 - , 6125, 7168, 8195, 9206, 10230, 11227, 12256, 13304, 14281, 15316 - , 16374, 17382, 18428, 19388, 20361, 21468, 22448, 23781, 0, 0 - , -1, 0, -2, 1024, 0, 0, 0, -1, 1024, -1024 - , 1, -1024, 4, 1024, -1024, 2, 1024, -1024, 2, 1024 - , -2048, -1024, -4, 1024, -1024, 2, 1024, -2048, -1024, -3 - , 1024, 2048, -2048, -1024, 4, 1024, 2048, -3072, -2048, -1024 - , -1, 662, 2048, 0, 1, 0, 0, 1, -2, -2 - , 0, 2, 1024, -1, 1024, -1024, 4, 1024, -1024, 1 - , 1024, -1024, 1, 1024, -2048, -781, -4, 844, -807, -5 - , 866, -2048, -726, -13, 777, 2048, -2048, -643, -4, 617 - , 2048, 3072, -3072, -2048, -629, 1, 630, 2048, 3072, 0 - , -1, 1, -2, 2, 1, -1024, 5, -1024, 6, 1024 - , -1024, 4, 1024, -1024, 1, 1024, -1024, -9, 1024, -673 - , -7, 655, -2048, -665, -15, 716, -2048, -647, 4, 640 - , 2048, -2048, -615, -1, 635, 2048, -2048, -613, 10, 637 - , 2048, 3072, -3072, -2048, -647, -3, 641, 2048, 3072, -5120 - , -4096, -3072, -2048, -681, 6, 685, 2048, 3072, 4096, 1 - , 1, 0, -1, 1024, -1024, -3, 1024, -1024, 6, 1024 - , -1024, -1, 769, -733, 0, 1024, -876, -2, 653, -1024 - , -4, 786, -596, -13, 595, -634, -2, 638, 2048, -2048 - , -620, -5, 620, 2048, -4096, -3072, -2048, -639, 11, 655 - , 2048, 3072, -3072, -2048, -659, 5, 663, 2048, -3072, -1823 - , -687, 22, 695, 2048, 3072, 4096, -4096, -3072, -1848, -715 - , -3, 727, 1816, 3072, 4096, 5120, -8192, -7168, -6144, -5120 - , -4096, -2884, -1771, -756, -14, 775, 1844, 3072, 4096, 5120 - , 6144, -1, 1, 0, -1024, 2, 815, -768, 2, 708 - , -1024, -3, 693, -661, -7, 607, -643, -5, 609, -624 - , 3, 631, -682, -3, 691, 2048, -2048, -640, 5, 650 - , 2048, -3072, -2048, -701, 9, 704, 2048, 3072, -3072, -2048 - , -670, 10, 674, 2048, 3072, -5120, -4096, -3072, -1749, -738 - , 0, 733, 1811, 3072, 4096, 5120, -4096, -3072, -1873, -753 - , 0, 756, 1874, 3072, 4096, -5120, -4096, -2900, -1838, -793 - , -6, 793, 1868, 2837, 4096, 5120, -7168, -6144, -5120, -4096 - , -2832, -1891, -828, 1, 828, 1901, 2823, 3912, 5120, 6144 - , 7168, 8192,-13312,-12288,-11264,-10240, -9216, -8192, -7168, -6144 - , -5120, -3976, -3004, -1911, -869, 7, 869, 1932, 3024, 3992 - , 5009, 6144, 7168, 8192, 9216, 10240, 11264, -4, 1024, -629 - , -22, 609, -623, 9, 640, -2048, -768, 1, 682, -2048 - , -741, 49, 722, 2048, -3072, -1706, -808, -20, 768, 1750 - , -1684, -727, -29, 788, 1840, 3033, -1758, -784, 0, 801 - , 1702, -3072, -1813, -814, 38, 820, 1884, 2927, -4096, -3241 - , -1839, -922, 25, 882, 1886, 2812, -4096, -2982, -1923, -894 - , 84, 912, 1869, 2778, 4096, -4928, -3965, -2902, -1920, -883 - , 3, 917, 1953, 2921, 3957, 4922, 6144, 7168, -5120, -3916 - , -2897, -1949, -930, 31, 959, 1934, 2901, 3851, 5120, -9216 - , -8192, -7046, -6029, -5030, -4034, -2980, -1969, -1013, -76, 963 - , 1963, 2901, 3929, 4893, 6270, 7168, 8192, 9216,-12288,-11264 - ,-10240, -9216, -8192, -6846, -6123, -5108, -4008, -3000, -1963, -954 - , -6, 958, 1992, 3009, 4020, 5085, 6097, 7168, 8192, 9216 - ,-11264,-10139, -9194, -8127, -7156, -6102, -5053, -4049, -3036, -2025 - , -1009, -34, 974, 1984, 3034, 4028, 5138, 6000, 7057, 8166 - , 9070, 10033, 11360, 12288,-13312,-12288,-10932,-10190, -9120, -8123 - , -7128, -6103, -5074, -4081, -3053, -2029, -989, -4, 1010, 2028 - , 3051, 4073, 5071, 6099, 7132, 8147, 9295, 10159, 11023, 12263 - , 13312, 14336,-25600,-24576,-23552,-22529,-21504,-20480,-19456,-18637 - ,-17425,-16165,-15316,-14327,-13606,-12135,-11182,-10107, -9153, -8144 - , -7146, -6160, -5129, -4095, -3064, -2038, -1025, 1, 1031, 2072 - , 3074, 4088, 5123, 6149, 7157, 8173, 9198, 10244, 11250, 12268 - , 13263, 14289, 15351, 16370, 17402, 18413, 19474, 20337, 21386, 22521 - , 23367, 24350, 0, 0, 0, 0, 0, 0, 0, 0 - , -1024, 0, 1024, -1024, 0, 1024, -1024, 0, 1024, -1024 - , 0, 1024, -1024, 0, 1024, -773, 0, 1024, -674, 0 - , 645, -2048, -745, 0, 628, 2048, -2048, -712, 0, 681 - , 2048, 3072, -3072, -2048, -673, 0, 682, 1964, 3257, 0 - , 0, 0, 0, 0, 0, 0, 0, -1024, 0, 1024 - , -1024, 0, 1024, -1024, 0, 1024, -705, 0, 623, -771 - , 0, 1024, -786, 0, 688, -631, 0, 652, 2048, -2048 - , -627, -1, 666, 2048, -3072, -1756, -694, 0, 674, 2048 - , -3098, -1879, -720, 5, 694, 1886, 2958, 4096, 0, 0 - , 0, 0, 1024, 0, 0, 1024, -769, 0, 1024, -1024 - , 0, 1024, -1024, 0, 1024, -817, 0, 734, -786, 0 - , 651, -638, 0, 637, -623, 0, 671, -652, 0, 619 - , 2048, -2048, -670, -1, 663, 2048, -1908, -680, 1, 686 - , 2048, 3072, 4096, -4096, -3072, -1833, -711, 0, 727, 1747 - , 3072, 4096, -4096, -2971, -1826, -762, 2, 766, 1832, 2852 - , 3928, 5079, 0, 0, 0, -1024, 0, 1024, -1024, 0 - , -656, 0, 1024, -599, 0, 620, -1024, 0, 1024, -603 - , 0, 622, -643, 0, 660, -599, 0, 611, -641, -1 - , 651, 2048, -2048, -648, -2, 647, 1798, -3072, -2048, -672 - , 2, 670, 2048, -3072, -1780, -694, -1, 706, 1751, 3072 - , -3072, -1862, -757, 7, 739, 1798, 3072, 4096, -5120, -4096 - , -3253, -1811, -787, 3, 782, 1887, 3123, 4096, -7252, -6144 - , -5354, -4060, -2864, -1863, -820, -11, 847, 1903, 2970, 3851 - , 4921, 5957, 7168, 8192, 9306, 0, 0, -1024, 0, 1024 - , -726, 0, 706, -692, 0, 593, -598, 0, 616, -624 - , 0, 616, -605, 0, 613, -2048, -652, 1, 635, 2048 - , -2048, -647, -1, 660, 2048, -1811, -668, -2, 685, 2048 - , -1796, -731, -2, 730, 1702, 3072, -3072, -1766, -747, -4 - , 756, 1770, 3072, -4096, -3024, -1762, -783, 4, 771, 1781 - , 3072, -5120, -4057, -2807, -1832, -822, 0, 816, 1804, 2851 - , 3949, 5120, -6144, -4899, -3927, -2920, -1893, -874, -2, 868 - , 1881, 2905, 3960, 4912, 6144, -9216, -8192, -7168, -6225, -4963 - , -3943, -2956, -1890, -902, 0, 897, 1914, 2916, 3984, 4990 - , 6050, 7168,-11264,-10217, -9114, -8132, -7035, -5988, -4984, -4000 - , -2980, -1962, -927, 7, 931, 1956, 2981, 4031, 4972, 6213 - , 7227, 8192, 9216, 10240, 11170, 12288, 13312, 14336, 0, 1024 - , -557, 1, 571, -606, -4, 612, -1676, -707, 10, 673 - , 2048, -2048, -727, 5, 686, -3072, -1772, -755, 12, 716 - , 1877, -1856, -786, 2, 786, 1712, -1685, -818, -16, 863 - , 1729, -3072, -1762, -857, 3, 866, 1838, 2841, -3862, -2816 - , -1864, -925, -2, 923, 1897, 2779, -2782, -1838, -920, -28 - , 931, 1951, 2835, 3804, -4815, -4001, -2940, -1934, -959, -22 - , 975, 1957, 2904, 3971, 4835, -5148, -3892, -2944, -1953, -986 - , -11, 989, 1968, 2939, 3949, 4947, 5902, -9216, -8192, -6915 - , -6004, -4965, -4013, -3009, -1977, -987, -1, 982, 1972, 3000 - , 3960, 4939, 5814, -8976, -7888, -7084, -5955, -5043, -4009, -2991 - , -2002, -1000, -8, 993, 2011, 3023, 4026, 5028, 6023, 7052 - , 8014, 9216,-11240,-10036, -9125, -8118, -7105, -6062, -5048, -4047 - , -3044, -2025, -1009, -1, 1011, 2023, 3042, 4074, 5085, 6108 - , 7119, 8142, 9152, 10114, 11141, 12250, 13307,-15360,-14099,-13284 - ,-12291,-11223,-10221, -9152, -8147, -7128, -6104, -5077, -4072, -3062 - , -2033, -1020, 7, 1018, 2038, 3059, 4081, 5084, 6109, 7102 - , 8128, 9134, 10125, 11239, 12080,-23552,-22528,-21504,-20480,-19456 - ,-18159,-17240,-16291,-15364,-14285,-13305,-12271,-11233,-10217, -9198 - , -8175, -7157, -6134, -5122, -4089, -3071, -2047, -1018, 3, 1026 - , 2041, 3077, 4090, 5108, 6131, 7150, 8172, 9175, 10196, 11272 - , 12303, 13273, 14328, 15332, 16334, 17381, 18409, 19423, 20423, 21451 - , 22679, 23391, 24568, 25600, 26589 -}; - -/* cdf tables for quantizer indices */ -const uint16_t WebRtcIsacfix_kCdfGain[1212] = { - 0, 13, 301, 3730, 61784, 65167, 65489, 65535, 0, 17, - 142, 314, 929, 2466, 7678, 56450, 63463, 64740, 65204, 65426, - 65527, 65535, 0, 8, 100, 724, 6301, 60105, 65125, 65510, - 65531, 65535, 0, 13, 117, 368, 1068, 3010, 11928, 53603, - 61177, 63404, 64505, 65108, 65422, 65502, 65531, 65535, 0, 4, - 17, 96, 410, 1859, 12125, 54361, 64103, 65305, 65497, 65535, - 0, 4, 88, 230, 469, 950, 1746, 3228, 6092, 16592, - 44756, 56848, 61256, 63308, 64325, 64920, 65309, 65460, 65502, 65522, - 65535, 0, 88, 352, 1675, 6339, 20749, 46686, 59284, 63525, - 64949, 65359, 65502, 65527, 65535, 0, 13, 38, 63, 117, - 234, 381, 641, 929, 1407, 2043, 2809, 4032, 5753, 8792, - 14407, 24308, 38941, 48947, 55403, 59293, 61411, 62688, 63630, 64329, - 64840, 65188, 65376, 65472, 65506, 65527, 65531, 65535, 0, 8, - 29, 75, 222, 615, 1327, 2801, 5623, 9931, 16094, 24966, - 34419, 43458, 50676, 56186, 60055, 62500, 63936, 64765, 65225, 65435, - 65514, 65535, 0, 8, 13, 15, 17, 21, 33, 59, - 71, 92, 151, 243, 360, 456, 674, 934, 1223, 1583, - 1989, 2504, 3031, 3617, 4354, 5154, 6163, 7411, 8780, 10747, - 12874, 15591, 18974, 23027, 27436, 32020, 36948, 41830, 46205, 49797, - 53042, 56094, 58418, 60360, 61763, 62818, 63559, 64103, 64509, 64798, - 65045, 65162, 65288, 65363, 65447, 65506, 65522, 65531, 65533, 65535, - 0, 4, 6, 25, 38, 71, 138, 264, 519, 808, - 1227, 1825, 2516, 3408, 4279, 5560, 7092, 9197, 11420, 14108, - 16947, 20300, 23926, 27459, 31164, 34827, 38575, 42178, 45540, 48747, - 51444, 54090, 56426, 58460, 60080, 61595, 62734, 63668, 64275, 64673, - 64936, 65112, 65217, 65334, 65426, 65464, 65477, 65489, 65518, 65527, - 65529, 65531, 65533, 65535, 0, 2, 4, 8, 10, 12, - 14, 16, 21, 33, 50, 71, 84, 92, 105, 138, - 180, 255, 318, 377, 435, 473, 511, 590, 682, 758, - 913, 1097, 1256, 1449, 1671, 1884, 2169, 2445, 2772, 3157, - 3563, 3944, 4375, 4848, 5334, 5820, 6448, 7101, 7716, 8378, - 9102, 9956, 10752, 11648, 12707, 13670, 14758, 15910, 17187, 18472, - 19627, 20649, 21951, 23169, 24283, 25552, 26862, 28227, 29391, 30764, - 31882, 33213, 34432, 35600, 36910, 38116, 39464, 40729, 41872, 43144, - 44371, 45514, 46762, 47813, 48968, 50069, 51032, 51974, 52908, 53737, - 54603, 55445, 56282, 56990, 57572, 58191, 58840, 59410, 59887, 60264, - 60607, 60946, 61269, 61516, 61771, 61960, 62198, 62408, 62558, 62776, - 62985, 63207, 63408, 63546, 63739, 63906, 64070, 64237, 64371, 64551, - 64677, 64836, 64999, 65095, 65213, 65284, 65338, 65380, 65426, 65447, - 65472, 65485, 65487, 65489, 65502, 65510, 65512, 65514, 65516, 65518, - 65522, 65531, 65533, 65535, 0, 2, 4, 6, 65528, 65531, - 65533, 65535, 0, 2, 4, 6, 8, 10, 222, 65321, - 65513, 65528, 65531, 65533, 65535, 0, 2, 4, 50, 65476, - 65529, 65531, 65533, 65535, 0, 2, 4, 6, 8, 12, - 38, 544, 64936, 65509, 65523, 65525, 65529, 65531, 65533, 65535, - 0, 2, 4, 6, 8, 10, 1055, 64508, 65528, 65531, - 65533, 65535, 0, 2, 4, 6, 8, 10, 12, 123, - 3956, 62999, 65372, 65495, 65515, 65521, 65523, 65525, 65527, 65529, - 65531, 65533, 65535, 0, 2, 4, 12, 53, 4707, 59445, - 65467, 65525, 65527, 65529, 65531, 65533, 65535, 0, 2, 4, - 6, 8, 10, 12, 14, 16, 38, 40, 50, 67, - 96, 234, 929, 14345, 55750, 64866, 65389, 65462, 65514, 65517, - 65519, 65521, 65523, 65525, 65527, 65529, 65531, 65533, 65535, 0, - 2, 4, 6, 8, 10, 15, 35, 91, 377, 1946, - 13618, 52565, 63714, 65184, 65465, 65520, 65523, 65525, 65527, 65529, - 65531, 65533, 65535, 0, 2, 4, 6, 8, 10, 12, - 14, 16, 18, 20, 22, 24, 26, 28, 30, 32, - 34, 36, 38, 40, 42, 44, 46, 48, 50, 52, - 54, 82, 149, 362, 751, 1701, 4239, 12893, 38627, 55072, - 60875, 63071, 64158, 64702, 65096, 65283, 65412, 65473, 65494, 65505, - 65508, 65517, 65519, 65521, 65523, 65525, 65527, 65529, 65531, 65533, - 65535, 0, 2, 15, 23, 53, 143, 260, 418, 698, - 988, 1353, 1812, 2411, 3144, 4015, 5143, 6401, 7611, 8999, - 10653, 12512, 14636, 16865, 19404, 22154, 24798, 27521, 30326, 33102, - 35790, 38603, 41415, 43968, 46771, 49435, 52152, 54715, 57143, 59481, - 61178, 62507, 63603, 64489, 64997, 65257, 65427, 65473, 65503, 65520, - 65529, 65531, 65533, 65535, 0, 3, 6, 9, 26, 32, - 44, 46, 64, 94, 111, 164, 205, 254, 327, 409, - 506, 608, 733, 885, 1093, 1292, 1482, 1742, 1993, 2329, - 2615, 3029, 3374, 3798, 4257, 4870, 5405, 5992, 6618, 7225, - 7816, 8418, 9051, 9761, 10532, 11380, 12113, 13010, 13788, 14594, - 15455, 16361, 17182, 18088, 18997, 20046, 20951, 21968, 22947, 24124, - 25296, 26547, 27712, 28775, 29807, 30835, 31709, 32469, 33201, 34014, - 34876, 35773, 36696, 37620, 38558, 39547, 40406, 41277, 42367, 43290, - 44445, 45443, 46510, 47684, 48973, 50157, 51187, 52242, 53209, 54083, - 55006, 55871, 56618, 57293, 57965, 58556, 59222, 59722, 60180, 60554, - 60902, 61250, 61554, 61837, 62100, 62372, 62631, 62856, 63078, 63324, - 63557, 63768, 63961, 64089, 64235, 64352, 64501, 64633, 64770, 64887, - 65001, 65059, 65121, 65188, 65246, 65302, 65346, 65390, 65428, 65463, - 65477, 65506, 65515, 65517, 65519, 65521, 65523, 65525, 65527, 65529, - 65531, 65533, 65535, 0, 2, 4, 109, 65332, 65531, 65533, - 65535, 0, 2, 4, 6, 8, 25, 1817, 63874, 65511, - 65527, 65529, 65531, 65533, 65535, 0, 2, 4, 907, 65014, - 65529, 65531, 65533, 65535, 0, 2, 4, 6, 8, 10, - 12, 132, 2743, 62708, 65430, 65525, 65527, 65529, 65531, 65533, - 65535, 0, 2, 4, 6, 8, 35, 3743, 61666, 65485, - 65531, 65533, 65535, 0, 2, 4, 6, 8, 10, 23, - 109, 683, 6905, 58417, 64911, 65398, 65497, 65518, 65525, 65527, - 65529, 65531, 65533, 65535, 0, 2, 4, 6, 53, 510, - 10209, 55212, 64573, 65441, 65522, 65529, 65531, 65533, 65535, 0, - 2, 4, 6, 8, 10, 12, 14, 16, 18, 20, - 22, 32, 90, 266, 1037, 3349, 14468, 50488, 62394, 64685, - 65341, 65480, 65514, 65519, 65521, 65523, 65525, 65527, 65529, 65531, - 65533, 65535, 0, 2, 4, 6, 9, 16, 37, 106, - 296, 748, 1868, 5733, 18897, 45553, 60165, 63949, 64926, 65314, - 65441, 65508, 65524, 65529, 65531, 65533, 65535, 0, 2, 4, - 6, 8, 10, 12, 14, 16, 18, 20, 22, 24, - 26, 28, 30, 32, 34, 36, 38, 40, 42, 44, - 46, 48, 50, 83, 175, 344, 667, 1293, 2337, 4357, - 8033, 14988, 28600, 43244, 52011, 57042, 59980, 61779, 63065, 63869, - 64390, 64753, 64988, 65164, 65326, 65422, 65462, 65492, 65506, 65522, - 65524, 65526, 65531, 65533, 65535, 0, 2, 4, 6, 8, - 10, 12, 14, 16, 25, 39, 48, 55, 62, 65, - 85, 106, 139, 169, 194, 252, 323, 485, 688, 1074, - 1600, 2544, 3863, 5733, 8303, 11397, 15529, 20273, 25734, 31455, - 36853, 41891, 46410, 50306, 53702, 56503, 58673, 60479, 61880, 62989, - 63748, 64404, 64852, 65124, 65309, 65424, 65480, 65524, 65528, 65533, - 65535, 0, 2, 4, 6, 8, 10, 12, 14, 21, - 23, 25, 27, 29, 31, 39, 41, 43, 48, 60, - 72, 79, 106, 136, 166, 187, 224, 252, 323, 381, - 427, 478, 568, 660, 783, 912, 1046, 1175, 1365, 1567, - 1768, 2024, 2347, 2659, 3049, 3529, 4033, 4623, 5281, 5925, - 6726, 7526, 8417, 9468, 10783, 12141, 13571, 15222, 16916, 18659, - 20350, 22020, 23725, 25497, 27201, 29026, 30867, 32632, 34323, 36062, - 37829, 39466, 41144, 42654, 43981, 45343, 46579, 47759, 49013, 50171, - 51249, 52283, 53245, 54148, 54938, 55669, 56421, 57109, 57791, 58464, - 59092, 59674, 60105, 60653, 61083, 61407, 61757, 62095, 62388, 62649, - 62873, 63157, 63358, 63540, 63725, 63884, 64046, 64155, 64278, 64426, - 64548, 64654, 64806, 64906, 64994, 65077, 65137, 65215, 65277, 65324, - 65354, 65409, 65437, 65455, 65462, 65490, 65495, 65499, 65508, 65511, - 65513, 65515, 65517, 65519, 65521, 65523, 65525, 65527, 65529, 65531, - 65533, 65535 -}; - -const uint16_t WebRtcIsacfix_kCdfShape[2059] = { - 0, 65535, 0, 65535, 0, 65535, 0, 65535, 0, 65535, - 0, 65535, 0, 65535, 0, 65535, 0, 65535, 0, 4, - 65535, 0, 8, 65514, 65535, 0, 29, 65481, 65535, 0, - 121, 65439, 65535, 0, 239, 65284, 65535, 0, 8, 779, - 64999, 65527, 65535, 0, 8, 888, 64693, 65522, 65535, 0, - 29, 2604, 62843, 65497, 65531, 65535, 0, 25, 176, 4576, - 61164, 65275, 65527, 65535, 0, 65535, 0, 65535, 0, 65535, - 0, 65535, 0, 4, 65535, 0, 65535, 0, 65535, 0, - 65535, 0, 65535, 0, 4, 65535, 0, 33, 65502, 65535, - 0, 54, 65481, 65535, 0, 251, 65309, 65535, 0, 611, - 65074, 65535, 0, 1273, 64292, 65527, 65535, 0, 4, 1809, - 63940, 65518, 65535, 0, 88, 4392, 60603, 65426, 65531, 65535, - 0, 25, 419, 7046, 57756, 64961, 65514, 65531, 65535, 0, - 65535, 0, 65535, 0, 65535, 0, 65535, 0, 4, 65531, - 65535, 0, 65535, 0, 8, 65531, 65535, 0, 4, 65527, - 65535, 0, 17, 65510, 65535, 0, 42, 65481, 65535, 0, - 197, 65342, 65531, 65535, 0, 385, 65154, 65535, 0, 1005, - 64522, 65535, 0, 8, 1985, 63469, 65533, 65535, 0, 38, - 3119, 61884, 65514, 65535, 0, 4, 6, 67, 4961, 60804, - 65472, 65535, 0, 17, 565, 9182, 56538, 65087, 65514, 65535, - 0, 8, 63, 327, 2118, 14490, 52774, 63839, 65376, 65522, - 65535, 0, 65535, 0, 65535, 0, 65535, 0, 65535, 0, - 17, 65522, 65535, 0, 59, 65489, 65535, 0, 50, 65522, - 65535, 0, 54, 65489, 65535, 0, 310, 65179, 65535, 0, - 615, 64836, 65535, 0, 4, 1503, 63965, 65535, 0, 2780, - 63383, 65535, 0, 21, 3919, 61051, 65527, 65535, 0, 84, - 6674, 59929, 65435, 65535, 0, 4, 255, 7976, 55784, 65150, - 65518, 65531, 65535, 0, 4, 8, 582, 10726, 53465, 64949, - 65518, 65535, 0, 29, 339, 3006, 17555, 49517, 62956, 65200, - 65497, 65531, 65535, 0, 2, 33, 138, 565, 2324, 7670, - 22089, 45966, 58949, 63479, 64966, 65380, 65518, 65535, 0, 65535, - 0, 65535, 0, 2, 65533, 65535, 0, 46, 65514, 65535, - 0, 414, 65091, 65535, 0, 540, 64911, 65535, 0, 419, - 65162, 65535, 0, 976, 64790, 65535, 0, 2977, 62495, 65531, - 65535, 0, 4, 3852, 61034, 65527, 65535, 0, 4, 29, - 6021, 60243, 65468, 65535, 0, 84, 6711, 58066, 65418, 65535, - 0, 13, 281, 9550, 54917, 65125, 65506, 65535, 0, 2, - 63, 984, 12108, 52644, 64342, 65435, 65527, 65535, 0, 29, - 251, 2014, 14871, 47553, 62881, 65229, 65518, 65535, 0, 13, - 142, 749, 4220, 18497, 45200, 60913, 64823, 65426, 65527, 65535, - 0, 13, 71, 264, 1176, 3789, 10500, 24480, 43488, 56324, - 62315, 64493, 65242, 65464, 65514, 65522, 65531, 65535, 0, 4, - 13, 38, 109, 205, 448, 850, 1708, 3429, 6276, 11371, - 19221, 29734, 40955, 49391, 55411, 59460, 62102, 63793, 64656, 65150, - 65401, 65485, 65522, 65531, 65535, 0, 65535, 0, 2, 65533, - 65535, 0, 1160, 65476, 65535, 0, 2, 6640, 64763, 65533, - 65535, 0, 2, 38, 9923, 61009, 65527, 65535, 0, 2, - 4949, 63092, 65533, 65535, 0, 2, 3090, 63398, 65533, 65535, - 0, 2, 2520, 58744, 65510, 65535, 0, 2, 13, 544, - 8784, 51403, 65148, 65533, 65535, 0, 2, 25, 1017, 10412, - 43550, 63651, 65489, 65527, 65535, 0, 2, 4, 29, 783, - 13377, 52462, 64524, 65495, 65533, 65535, 0, 2, 4, 6, - 100, 1817, 18451, 52590, 63559, 65376, 65531, 65535, 0, 2, - 4, 6, 46, 385, 2562, 11225, 37416, 60488, 65026, 65487, - 65529, 65533, 65535, 0, 2, 4, 6, 8, 10, 12, - 42, 222, 971, 5221, 19811, 45048, 60312, 64486, 65294, 65474, - 65525, 65529, 65533, 65535, 0, 2, 4, 8, 71, 167, - 666, 2533, 7875, 19622, 38082, 54359, 62108, 64633, 65290, 65495, - 65529, 65533, 65535, 0, 2, 4, 6, 8, 10, 13, - 109, 586, 1930, 4949, 11600, 22641, 36125, 48312, 56899, 61495, - 63927, 64932, 65389, 65489, 65518, 65531, 65533, 65535, 0, 4, - 6, 8, 67, 209, 712, 1838, 4195, 8432, 14432, 22834, - 31723, 40523, 48139, 53929, 57865, 60657, 62403, 63584, 64363, 64907, - 65167, 65372, 65472, 65514, 65535, 0, 2, 4, 13, 25, - 42, 46, 50, 75, 113, 147, 281, 448, 657, 909, - 1185, 1591, 1976, 2600, 3676, 5317, 7398, 9914, 12941, 16169, - 19477, 22885, 26464, 29851, 33360, 37228, 41139, 44802, 48654, 52058, - 55181, 57676, 59581, 61022, 62190, 63107, 63676, 64199, 64547, 64924, - 65158, 65313, 65430, 65481, 65518, 65535, 0, 65535, 0, 65535, - 0, 65535, 0, 65535, 0, 65533, 65535, 0, 65535, 0, - 65535, 0, 65535, 0, 65533, 65535, 0, 2, 65535, 0, - 2, 65533, 65535, 0, 2, 65533, 65535, 0, 2, 65533, - 65535, 0, 2, 4, 65533, 65535, 0, 2, 65533, 65535, - 0, 2, 4, 65531, 65533, 65535, 0, 2, 4, 65531, - 65533, 65535, 0, 2, 4, 6, 65524, 65533, 65535, 0, - 65535, 0, 65535, 0, 65535, 0, 65535, 0, 65535, 0, - 65535, 0, 65535, 0, 65535, 0, 65533, 65535, 0, 65533, - 65535, 0, 2, 65533, 65535, 0, 2, 65533, 65535, 0, - 2, 65533, 65535, 0, 2, 4, 65532, 65535, 0, 6, - 65523, 65535, 0, 2, 15, 65530, 65533, 65535, 0, 2, - 35, 65493, 65531, 65533, 65535, 0, 2, 4, 158, 65382, - 65531, 65533, 65535, 0, 65535, 0, 65535, 0, 65535, 0, - 65535, 0, 65535, 0, 65535, 0, 2, 65535, 0, 2, - 65533, 65535, 0, 2, 65533, 65535, 0, 2, 65533, 65535, - 0, 2, 65533, 65535, 0, 9, 65512, 65535, 0, 2, - 12, 65529, 65535, 0, 2, 73, 65434, 65533, 65535, 0, - 2, 240, 65343, 65533, 65535, 0, 2, 476, 65017, 65531, - 65533, 65535, 0, 2, 4, 1046, 64686, 65531, 65533, 65535, - 0, 2, 4, 6, 8, 1870, 63898, 65529, 65531, 65533, - 65535, 0, 65535, 0, 65535, 0, 65535, 0, 65533, 65535, - 0, 2, 65533, 65535, 0, 2, 65533, 65535, 0, 2, - 65532, 65535, 0, 6, 65533, 65535, 0, 6, 65523, 65535, - 0, 2, 65532, 65535, 0, 137, 65439, 65535, 0, 576, - 64899, 65533, 65535, 0, 2, 289, 65299, 65533, 65535, 0, - 2, 4, 6, 880, 64134, 65531, 65533, 65535, 0, 2, - 4, 1853, 63347, 65533, 65535, 0, 2, 6, 2516, 61762, - 65529, 65531, 65533, 65535, 0, 2, 4, 9, 3980, 61380, - 65503, 65529, 65531, 65533, 65535, 0, 2, 4, 6, 8, - 10, 12, 61, 6393, 59859, 65466, 65527, 65529, 65531, 65533, - 65535, 0, 65535, 0, 65535, 0, 65535, 0, 2, 65532, - 65535, 0, 3, 65529, 65535, 0, 2, 65529, 65535, 0, - 61, 65453, 65535, 0, 234, 65313, 65535, 0, 503, 65138, - 65535, 0, 155, 65402, 65533, 65535, 0, 2, 1058, 64554, - 65533, 65535, 0, 2, 4, 3138, 62109, 65531, 65533, 65535, - 0, 2, 4, 2031, 63339, 65531, 65533, 65535, 0, 2, - 4, 6, 9, 4155, 60778, 65523, 65529, 65531, 65533, 65535, - 0, 2, 4, 41, 6189, 59269, 65490, 65531, 65533, 65535, - 0, 2, 4, 6, 210, 8789, 57043, 65400, 65528, 65531, - 65533, 65535, 0, 2, 4, 6, 8, 26, 453, 10086, - 55499, 64948, 65483, 65524, 65527, 65529, 65531, 65533, 65535, 0, - 2, 4, 6, 8, 10, 12, 14, 16, 18, 20, - 114, 1014, 11202, 52670, 64226, 65356, 65503, 65514, 65523, 65525, - 65527, 65529, 65531, 65533, 65535, 0, 65533, 65535, 0, 15, - 65301, 65535, 0, 152, 64807, 65535, 0, 2, 3328, 63308, - 65535, 0, 2, 4050, 59730, 65533, 65535, 0, 2, 164, - 10564, 61894, 65529, 65535, 0, 15, 6712, 59831, 65076, 65532, - 65535, 0, 32, 7712, 57449, 65459, 65535, 0, 2, 210, - 7849, 53110, 65021, 65523, 65535, 0, 2, 12, 1081, 13883, - 48262, 62870, 65477, 65535, 0, 2, 88, 847, 6145, 37852, - 62012, 65454, 65533, 65535, 0, 9, 47, 207, 1823, 14522, - 45521, 61069, 64891, 65481, 65528, 65531, 65533, 65535, 0, 2, - 9, 488, 2881, 12758, 38703, 58412, 64420, 65410, 65533, 65535, - 0, 2, 4, 6, 61, 333, 1891, 6486, 19720, 43188, - 57547, 62472, 64796, 65421, 65497, 65523, 65529, 65531, 65533, 65535, - 0, 2, 4, 6, 8, 10, 12, 29, 117, 447, - 1528, 6138, 21242, 43133, 56495, 62432, 64746, 65362, 65500, 65529, - 65531, 65533, 65535, 0, 2, 18, 105, 301, 760, 1490, - 3472, 7568, 15002, 26424, 40330, 53029, 60048, 62964, 64274, 64890, - 65337, 65445, 65489, 65513, 65527, 65530, 65533, 65535, 0, 2, - 4, 6, 41, 102, 409, 853, 2031, 4316, 7302, 11328, - 16869, 24825, 34926, 43481, 50877, 56126, 59874, 62103, 63281, 63857, - 64166, 64675, 65382, 65522, 65531, 65533, 65535, 0, 2, 4, - 6, 8, 10, 12, 14, 16, 18, 29, 38, 53, - 58, 96, 181, 503, 1183, 2849, 5590, 8600, 11379, 13942, - 16478, 19453, 22638, 26039, 29411, 32921, 37596, 41433, 44998, 48560, - 51979, 55106, 57666, 59892, 61485, 62616, 63484, 64018, 64375, 64685, - 64924, 65076, 65278, 65395, 65471, 65509, 65529, 65535, 0, 65535, - 0, 65535, 0, 65535, 0, 65535, 0, 65535, 0, 65535, - 0, 65535, 0, 65535, 0, 2, 65533, 65535, 0, 2, - 65533, 65535, 0, 2, 65533, 65535, 0, 2, 65533, 65535, - 0, 2, 65533, 65535, 0, 2, 65533, 65535, 0, 7, - 65519, 65535, 0, 2, 14, 65491, 65533, 65535, 0, 2, - 81, 65427, 65531, 65533, 65535, 0, 2, 4, 312, 65293, - 65528, 65533, 65535, 0, 65535, 0, 65535, 0, 65535, 0, - 65535, 0, 65535, 0, 65535, 0, 65535, 0, 65535, 0, - 2, 65533, 65535, 0, 2, 65533, 65535, 0, 2, 65533, - 65535, 0, 5, 65523, 65535, 0, 2, 65533, 65535, 0, - 7, 65526, 65535, 0, 46, 65464, 65533, 65535, 0, 2, - 120, 65309, 65533, 65535, 0, 2, 5, 362, 65097, 65533, - 65535, 0, 2, 18, 1164, 64785, 65528, 65531, 65533, 65535, - 0, 65535, 0, 65535, 0, 65535, 0, 65533, 65535, 0, - 65535, 0, 65533, 65535, 0, 2, 65533, 65535, 0, 2, - 65533, 65535, 0, 2, 65533, 65535, 0, 2, 65530, 65535, - 0, 2, 65523, 65535, 0, 69, 65477, 65535, 0, 141, - 65459, 65535, 0, 194, 65325, 65533, 65535, 0, 2, 543, - 64912, 65533, 65535, 0, 5, 1270, 64301, 65529, 65531, 65533, - 65535, 0, 2, 4, 12, 2055, 63538, 65508, 65531, 65533, - 65535, 0, 2, 7, 102, 3775, 61970, 65429, 65526, 65528, - 65533, 65535, 0, 65535, 0, 65535, 0, 65535, 0, 2, - 65533, 65535, 0, 2, 65535, 0, 9, 65533, 65535, 0, - 25, 65512, 65535, 0, 2, 65533, 65535, 0, 44, 65480, - 65535, 0, 48, 65475, 65535, 0, 162, 65373, 65535, 0, - 637, 64806, 65533, 65535, 0, 2, 935, 64445, 65533, 65535, - 0, 2, 4, 1662, 64083, 65533, 65535, 0, 2, 12, - 3036, 62469, 65521, 65533, 65535, 0, 2, 120, 5405, 60468, - 65469, 65531, 65533, 65535, 0, 2, 4, 18, 254, 6663, - 58999, 65272, 65528, 65533, 65535, 0, 2, 4, 9, 12, - 67, 591, 8981, 56781, 64564, 65365, 65508, 65524, 65526, 65529, - 65531, 65533, 65535, 0, 65535, 0, 65535, 0, 2, 65533, - 65535, 0, 9, 65526, 65535, 0, 14, 65503, 65535, 0, - 127, 65390, 65535, 0, 517, 64990, 65535, 0, 178, 65330, - 65535, 0, 2, 1055, 64533, 65533, 65535, 0, 2, 1558, - 63942, 65533, 65535, 0, 2, 2205, 63173, 65533, 65535, 0, - 25, 4493, 60862, 65505, 65533, 65535, 0, 2, 48, 5890, - 59442, 65482, 65533, 65535, 0, 2, 4, 127, 7532, 58191, - 65394, 65533, 65535, 0, 2, 5, 32, 550, 10388, 54924, - 65046, 65510, 65531, 65533, 65535, 0, 2, 4, 30, 150, - 1685, 14340, 51375, 63619, 65288, 65503, 65528, 65533, 65535, 0, - 2, 4, 6, 8, 28, 97, 473, 2692, 15407, 50020, - 62880, 65064, 65445, 65508, 65531, 65533, 65535, 0, 2, 4, - 12, 32, 79, 150, 372, 907, 2184, 5868, 18207, 45431, - 59856, 64031, 65096, 65401, 65481, 65507, 65521, 65523, 65525, 65527, - 65529, 65531, 65533, 65535, 0, 65533, 65535, 0, 182, 65491, - 65535, 0, 877, 64286, 65535, 0, 9, 2708, 63612, 65533, - 65535, 0, 2, 6038, 59532, 65535, 0, 2, 92, 5500, - 60539, 65533, 65535, 0, 268, 8908, 56512, 65385, 65535, 0, - 129, 13110, 52742, 65036, 65535, 0, 2, 806, 14003, 51929, - 64732, 65523, 65535, 0, 7, 92, 2667, 18159, 47678, 62610, - 65355, 65535, 0, 32, 1836, 19676, 48237, 61677, 64960, 65526, - 65535, 0, 21, 159, 967, 5668, 22782, 44709, 58317, 64020, - 65406, 65528, 65535, 0, 7, 162, 1838, 8328, 23929, 43014, - 56394, 63374, 65216, 65484, 65521, 65535, 0, 2, 4, 6, - 28, 268, 1120, 3613, 10688, 24185, 40989, 54917, 61684, 64510, - 65403, 65530, 65535, 0, 2, 16, 44, 139, 492, 1739, - 5313, 13558, 26766, 41566, 52446, 58937, 62815, 64480, 65201, 65454, - 65524, 65533, 65535, 0, 7, 25, 76, 263, 612, 1466, - 3325, 6832, 12366, 20152, 29466, 39255, 47360, 53506, 57740, 60726, - 62845, 64131, 64882, 65260, 65459, 65521, 65528, 65530, 65535, 0, - 2, 4, 14, 48, 136, 312, 653, 1240, 2369, 4327, - 7028, 10759, 15449, 21235, 28027, 35386, 42938, 49562, 54990, 59119, - 62086, 63916, 64863, 65249, 65445, 65493, 65523, 65535, 0, 2, - 4, 6, 8, 10, 12, 21, 83, 208, 409, 723, - 1152, 1868, 2951, 4463, 6460, 8979, 11831, 15195, 18863, 22657, - 26762, 30881, 34963, 39098, 43054, 47069, 50620, 53871, 56821, 59386, - 61340, 62670, 63512, 64023, 64429, 64750, 64944, 65126, 65279, 65366, - 65413, 65445, 65473, 65505, 65510, 65521, 65528, 65530, 65535 -}; - -/* pointers to cdf tables for quantizer indices */ -const uint16_t *WebRtcIsacfix_kCdfGainPtr[3][12] = { - { WebRtcIsacfix_kCdfGain +0 +0, WebRtcIsacfix_kCdfGain +0 +8, WebRtcIsacfix_kCdfGain +0 +22, - WebRtcIsacfix_kCdfGain +0 +32, WebRtcIsacfix_kCdfGain +0 +48, WebRtcIsacfix_kCdfGain +0 +60, - WebRtcIsacfix_kCdfGain +0 +81, WebRtcIsacfix_kCdfGain +0 +95, WebRtcIsacfix_kCdfGain +0 +128, - WebRtcIsacfix_kCdfGain +0 +152, WebRtcIsacfix_kCdfGain +0 +210, WebRtcIsacfix_kCdfGain +0 +264 - }, - { WebRtcIsacfix_kCdfGain +404 +0, WebRtcIsacfix_kCdfGain +404 +8, WebRtcIsacfix_kCdfGain +404 +21, - WebRtcIsacfix_kCdfGain +404 +30, WebRtcIsacfix_kCdfGain +404 +46, WebRtcIsacfix_kCdfGain +404 +58, - WebRtcIsacfix_kCdfGain +404 +79, WebRtcIsacfix_kCdfGain +404 +93, WebRtcIsacfix_kCdfGain +404 +125, - WebRtcIsacfix_kCdfGain +404 +149, WebRtcIsacfix_kCdfGain +404 +207, WebRtcIsacfix_kCdfGain +404 +260 - }, - { WebRtcIsacfix_kCdfGain +803 +0, WebRtcIsacfix_kCdfGain +803 +8, WebRtcIsacfix_kCdfGain +803 +22, - WebRtcIsacfix_kCdfGain +803 +31, WebRtcIsacfix_kCdfGain +803 +48, WebRtcIsacfix_kCdfGain +803 +60, - WebRtcIsacfix_kCdfGain +803 +81, WebRtcIsacfix_kCdfGain +803 +96, WebRtcIsacfix_kCdfGain +803 +129, - WebRtcIsacfix_kCdfGain +803 +154, WebRtcIsacfix_kCdfGain +803 +212, WebRtcIsacfix_kCdfGain +803 +268 - } -}; - -const uint16_t *WebRtcIsacfix_kCdfShapePtr[3][108] = { - { WebRtcIsacfix_kCdfShape +0 +0, WebRtcIsacfix_kCdfShape +0 +2, WebRtcIsacfix_kCdfShape +0 +4, - WebRtcIsacfix_kCdfShape +0 +6, WebRtcIsacfix_kCdfShape +0 +8, WebRtcIsacfix_kCdfShape +0 +10, - WebRtcIsacfix_kCdfShape +0 +12, WebRtcIsacfix_kCdfShape +0 +14, WebRtcIsacfix_kCdfShape +0 +16, - WebRtcIsacfix_kCdfShape +0 +18, WebRtcIsacfix_kCdfShape +0 +21, WebRtcIsacfix_kCdfShape +0 +25, - WebRtcIsacfix_kCdfShape +0 +29, WebRtcIsacfix_kCdfShape +0 +33, WebRtcIsacfix_kCdfShape +0 +37, - WebRtcIsacfix_kCdfShape +0 +43, WebRtcIsacfix_kCdfShape +0 +49, WebRtcIsacfix_kCdfShape +0 +56, - WebRtcIsacfix_kCdfShape +0 +64, WebRtcIsacfix_kCdfShape +0 +66, WebRtcIsacfix_kCdfShape +0 +68, - WebRtcIsacfix_kCdfShape +0 +70, WebRtcIsacfix_kCdfShape +0 +72, WebRtcIsacfix_kCdfShape +0 +75, - WebRtcIsacfix_kCdfShape +0 +77, WebRtcIsacfix_kCdfShape +0 +79, WebRtcIsacfix_kCdfShape +0 +81, - WebRtcIsacfix_kCdfShape +0 +83, WebRtcIsacfix_kCdfShape +0 +86, WebRtcIsacfix_kCdfShape +0 +90, - WebRtcIsacfix_kCdfShape +0 +94, WebRtcIsacfix_kCdfShape +0 +98, WebRtcIsacfix_kCdfShape +0 +102, - WebRtcIsacfix_kCdfShape +0 +107, WebRtcIsacfix_kCdfShape +0 +113, WebRtcIsacfix_kCdfShape +0 +120, - WebRtcIsacfix_kCdfShape +0 +129, WebRtcIsacfix_kCdfShape +0 +131, WebRtcIsacfix_kCdfShape +0 +133, - WebRtcIsacfix_kCdfShape +0 +135, WebRtcIsacfix_kCdfShape +0 +137, WebRtcIsacfix_kCdfShape +0 +141, - WebRtcIsacfix_kCdfShape +0 +143, WebRtcIsacfix_kCdfShape +0 +147, WebRtcIsacfix_kCdfShape +0 +151, - WebRtcIsacfix_kCdfShape +0 +155, WebRtcIsacfix_kCdfShape +0 +159, WebRtcIsacfix_kCdfShape +0 +164, - WebRtcIsacfix_kCdfShape +0 +168, WebRtcIsacfix_kCdfShape +0 +172, WebRtcIsacfix_kCdfShape +0 +178, - WebRtcIsacfix_kCdfShape +0 +184, WebRtcIsacfix_kCdfShape +0 +192, WebRtcIsacfix_kCdfShape +0 +200, - WebRtcIsacfix_kCdfShape +0 +211, WebRtcIsacfix_kCdfShape +0 +213, WebRtcIsacfix_kCdfShape +0 +215, - WebRtcIsacfix_kCdfShape +0 +217, WebRtcIsacfix_kCdfShape +0 +219, WebRtcIsacfix_kCdfShape +0 +223, - WebRtcIsacfix_kCdfShape +0 +227, WebRtcIsacfix_kCdfShape +0 +231, WebRtcIsacfix_kCdfShape +0 +235, - WebRtcIsacfix_kCdfShape +0 +239, WebRtcIsacfix_kCdfShape +0 +243, WebRtcIsacfix_kCdfShape +0 +248, - WebRtcIsacfix_kCdfShape +0 +252, WebRtcIsacfix_kCdfShape +0 +258, WebRtcIsacfix_kCdfShape +0 +264, - WebRtcIsacfix_kCdfShape +0 +273, WebRtcIsacfix_kCdfShape +0 +282, WebRtcIsacfix_kCdfShape +0 +293, - WebRtcIsacfix_kCdfShape +0 +308, WebRtcIsacfix_kCdfShape +0 +310, WebRtcIsacfix_kCdfShape +0 +312, - WebRtcIsacfix_kCdfShape +0 +316, WebRtcIsacfix_kCdfShape +0 +320, WebRtcIsacfix_kCdfShape +0 +324, - WebRtcIsacfix_kCdfShape +0 +328, WebRtcIsacfix_kCdfShape +0 +332, WebRtcIsacfix_kCdfShape +0 +336, - WebRtcIsacfix_kCdfShape +0 +341, WebRtcIsacfix_kCdfShape +0 +347, WebRtcIsacfix_kCdfShape +0 +354, - WebRtcIsacfix_kCdfShape +0 +360, WebRtcIsacfix_kCdfShape +0 +368, WebRtcIsacfix_kCdfShape +0 +378, - WebRtcIsacfix_kCdfShape +0 +388, WebRtcIsacfix_kCdfShape +0 +400, WebRtcIsacfix_kCdfShape +0 +418, - WebRtcIsacfix_kCdfShape +0 +445, WebRtcIsacfix_kCdfShape +0 +447, WebRtcIsacfix_kCdfShape +0 +451, - WebRtcIsacfix_kCdfShape +0 +455, WebRtcIsacfix_kCdfShape +0 +461, WebRtcIsacfix_kCdfShape +0 +468, - WebRtcIsacfix_kCdfShape +0 +474, WebRtcIsacfix_kCdfShape +0 +480, WebRtcIsacfix_kCdfShape +0 +486, - WebRtcIsacfix_kCdfShape +0 +495, WebRtcIsacfix_kCdfShape +0 +505, WebRtcIsacfix_kCdfShape +0 +516, - WebRtcIsacfix_kCdfShape +0 +528, WebRtcIsacfix_kCdfShape +0 +543, WebRtcIsacfix_kCdfShape +0 +564, - WebRtcIsacfix_kCdfShape +0 +583, WebRtcIsacfix_kCdfShape +0 +608, WebRtcIsacfix_kCdfShape +0 +635 - }, - { WebRtcIsacfix_kCdfShape +686 +0, WebRtcIsacfix_kCdfShape +686 +2, WebRtcIsacfix_kCdfShape +686 +4, - WebRtcIsacfix_kCdfShape +686 +6, WebRtcIsacfix_kCdfShape +686 +8, WebRtcIsacfix_kCdfShape +686 +11, - WebRtcIsacfix_kCdfShape +686 +13, WebRtcIsacfix_kCdfShape +686 +15, WebRtcIsacfix_kCdfShape +686 +17, - WebRtcIsacfix_kCdfShape +686 +20, WebRtcIsacfix_kCdfShape +686 +23, WebRtcIsacfix_kCdfShape +686 +27, - WebRtcIsacfix_kCdfShape +686 +31, WebRtcIsacfix_kCdfShape +686 +35, WebRtcIsacfix_kCdfShape +686 +40, - WebRtcIsacfix_kCdfShape +686 +44, WebRtcIsacfix_kCdfShape +686 +50, WebRtcIsacfix_kCdfShape +686 +56, - WebRtcIsacfix_kCdfShape +686 +63, WebRtcIsacfix_kCdfShape +686 +65, WebRtcIsacfix_kCdfShape +686 +67, - WebRtcIsacfix_kCdfShape +686 +69, WebRtcIsacfix_kCdfShape +686 +71, WebRtcIsacfix_kCdfShape +686 +73, - WebRtcIsacfix_kCdfShape +686 +75, WebRtcIsacfix_kCdfShape +686 +77, WebRtcIsacfix_kCdfShape +686 +79, - WebRtcIsacfix_kCdfShape +686 +82, WebRtcIsacfix_kCdfShape +686 +85, WebRtcIsacfix_kCdfShape +686 +89, - WebRtcIsacfix_kCdfShape +686 +93, WebRtcIsacfix_kCdfShape +686 +97, WebRtcIsacfix_kCdfShape +686 +102, - WebRtcIsacfix_kCdfShape +686 +106, WebRtcIsacfix_kCdfShape +686 +112, WebRtcIsacfix_kCdfShape +686 +119, - WebRtcIsacfix_kCdfShape +686 +127, WebRtcIsacfix_kCdfShape +686 +129, WebRtcIsacfix_kCdfShape +686 +131, - WebRtcIsacfix_kCdfShape +686 +133, WebRtcIsacfix_kCdfShape +686 +135, WebRtcIsacfix_kCdfShape +686 +137, - WebRtcIsacfix_kCdfShape +686 +139, WebRtcIsacfix_kCdfShape +686 +142, WebRtcIsacfix_kCdfShape +686 +146, - WebRtcIsacfix_kCdfShape +686 +150, WebRtcIsacfix_kCdfShape +686 +154, WebRtcIsacfix_kCdfShape +686 +158, - WebRtcIsacfix_kCdfShape +686 +162, WebRtcIsacfix_kCdfShape +686 +167, WebRtcIsacfix_kCdfShape +686 +173, - WebRtcIsacfix_kCdfShape +686 +179, WebRtcIsacfix_kCdfShape +686 +186, WebRtcIsacfix_kCdfShape +686 +194, - WebRtcIsacfix_kCdfShape +686 +205, WebRtcIsacfix_kCdfShape +686 +207, WebRtcIsacfix_kCdfShape +686 +209, - WebRtcIsacfix_kCdfShape +686 +211, WebRtcIsacfix_kCdfShape +686 +214, WebRtcIsacfix_kCdfShape +686 +218, - WebRtcIsacfix_kCdfShape +686 +222, WebRtcIsacfix_kCdfShape +686 +226, WebRtcIsacfix_kCdfShape +686 +230, - WebRtcIsacfix_kCdfShape +686 +234, WebRtcIsacfix_kCdfShape +686 +238, WebRtcIsacfix_kCdfShape +686 +242, - WebRtcIsacfix_kCdfShape +686 +247, WebRtcIsacfix_kCdfShape +686 +253, WebRtcIsacfix_kCdfShape +686 +262, - WebRtcIsacfix_kCdfShape +686 +269, WebRtcIsacfix_kCdfShape +686 +278, WebRtcIsacfix_kCdfShape +686 +289, - WebRtcIsacfix_kCdfShape +686 +305, WebRtcIsacfix_kCdfShape +686 +307, WebRtcIsacfix_kCdfShape +686 +309, - WebRtcIsacfix_kCdfShape +686 +311, WebRtcIsacfix_kCdfShape +686 +315, WebRtcIsacfix_kCdfShape +686 +319, - WebRtcIsacfix_kCdfShape +686 +323, WebRtcIsacfix_kCdfShape +686 +327, WebRtcIsacfix_kCdfShape +686 +331, - WebRtcIsacfix_kCdfShape +686 +335, WebRtcIsacfix_kCdfShape +686 +340, WebRtcIsacfix_kCdfShape +686 +346, - WebRtcIsacfix_kCdfShape +686 +354, WebRtcIsacfix_kCdfShape +686 +362, WebRtcIsacfix_kCdfShape +686 +374, - WebRtcIsacfix_kCdfShape +686 +384, WebRtcIsacfix_kCdfShape +686 +396, WebRtcIsacfix_kCdfShape +686 +413, - WebRtcIsacfix_kCdfShape +686 +439, WebRtcIsacfix_kCdfShape +686 +442, WebRtcIsacfix_kCdfShape +686 +446, - WebRtcIsacfix_kCdfShape +686 +450, WebRtcIsacfix_kCdfShape +686 +455, WebRtcIsacfix_kCdfShape +686 +461, - WebRtcIsacfix_kCdfShape +686 +468, WebRtcIsacfix_kCdfShape +686 +475, WebRtcIsacfix_kCdfShape +686 +481, - WebRtcIsacfix_kCdfShape +686 +489, WebRtcIsacfix_kCdfShape +686 +498, WebRtcIsacfix_kCdfShape +686 +508, - WebRtcIsacfix_kCdfShape +686 +522, WebRtcIsacfix_kCdfShape +686 +534, WebRtcIsacfix_kCdfShape +686 +554, - WebRtcIsacfix_kCdfShape +686 +577, WebRtcIsacfix_kCdfShape +686 +602, WebRtcIsacfix_kCdfShape +686 +631 - }, - { WebRtcIsacfix_kCdfShape +1368 +0, WebRtcIsacfix_kCdfShape +1368 +2, WebRtcIsacfix_kCdfShape +1368 +4, - WebRtcIsacfix_kCdfShape +1368 +6, WebRtcIsacfix_kCdfShape +1368 +8, WebRtcIsacfix_kCdfShape +1368 +10, - WebRtcIsacfix_kCdfShape +1368 +12, WebRtcIsacfix_kCdfShape +1368 +14, WebRtcIsacfix_kCdfShape +1368 +16, - WebRtcIsacfix_kCdfShape +1368 +20, WebRtcIsacfix_kCdfShape +1368 +24, WebRtcIsacfix_kCdfShape +1368 +28, - WebRtcIsacfix_kCdfShape +1368 +32, WebRtcIsacfix_kCdfShape +1368 +36, WebRtcIsacfix_kCdfShape +1368 +40, - WebRtcIsacfix_kCdfShape +1368 +44, WebRtcIsacfix_kCdfShape +1368 +50, WebRtcIsacfix_kCdfShape +1368 +57, - WebRtcIsacfix_kCdfShape +1368 +65, WebRtcIsacfix_kCdfShape +1368 +67, WebRtcIsacfix_kCdfShape +1368 +69, - WebRtcIsacfix_kCdfShape +1368 +71, WebRtcIsacfix_kCdfShape +1368 +73, WebRtcIsacfix_kCdfShape +1368 +75, - WebRtcIsacfix_kCdfShape +1368 +77, WebRtcIsacfix_kCdfShape +1368 +79, WebRtcIsacfix_kCdfShape +1368 +81, - WebRtcIsacfix_kCdfShape +1368 +85, WebRtcIsacfix_kCdfShape +1368 +89, WebRtcIsacfix_kCdfShape +1368 +93, - WebRtcIsacfix_kCdfShape +1368 +97, WebRtcIsacfix_kCdfShape +1368 +101, WebRtcIsacfix_kCdfShape +1368 +105, - WebRtcIsacfix_kCdfShape +1368 +110, WebRtcIsacfix_kCdfShape +1368 +116, WebRtcIsacfix_kCdfShape +1368 +123, - WebRtcIsacfix_kCdfShape +1368 +132, WebRtcIsacfix_kCdfShape +1368 +134, WebRtcIsacfix_kCdfShape +1368 +136, - WebRtcIsacfix_kCdfShape +1368 +138, WebRtcIsacfix_kCdfShape +1368 +141, WebRtcIsacfix_kCdfShape +1368 +143, - WebRtcIsacfix_kCdfShape +1368 +146, WebRtcIsacfix_kCdfShape +1368 +150, WebRtcIsacfix_kCdfShape +1368 +154, - WebRtcIsacfix_kCdfShape +1368 +158, WebRtcIsacfix_kCdfShape +1368 +162, WebRtcIsacfix_kCdfShape +1368 +166, - WebRtcIsacfix_kCdfShape +1368 +170, WebRtcIsacfix_kCdfShape +1368 +174, WebRtcIsacfix_kCdfShape +1368 +179, - WebRtcIsacfix_kCdfShape +1368 +185, WebRtcIsacfix_kCdfShape +1368 +193, WebRtcIsacfix_kCdfShape +1368 +203, - WebRtcIsacfix_kCdfShape +1368 +214, WebRtcIsacfix_kCdfShape +1368 +216, WebRtcIsacfix_kCdfShape +1368 +218, - WebRtcIsacfix_kCdfShape +1368 +220, WebRtcIsacfix_kCdfShape +1368 +224, WebRtcIsacfix_kCdfShape +1368 +227, - WebRtcIsacfix_kCdfShape +1368 +231, WebRtcIsacfix_kCdfShape +1368 +235, WebRtcIsacfix_kCdfShape +1368 +239, - WebRtcIsacfix_kCdfShape +1368 +243, WebRtcIsacfix_kCdfShape +1368 +247, WebRtcIsacfix_kCdfShape +1368 +251, - WebRtcIsacfix_kCdfShape +1368 +256, WebRtcIsacfix_kCdfShape +1368 +262, WebRtcIsacfix_kCdfShape +1368 +269, - WebRtcIsacfix_kCdfShape +1368 +277, WebRtcIsacfix_kCdfShape +1368 +286, WebRtcIsacfix_kCdfShape +1368 +297, - WebRtcIsacfix_kCdfShape +1368 +315, WebRtcIsacfix_kCdfShape +1368 +317, WebRtcIsacfix_kCdfShape +1368 +319, - WebRtcIsacfix_kCdfShape +1368 +323, WebRtcIsacfix_kCdfShape +1368 +327, WebRtcIsacfix_kCdfShape +1368 +331, - WebRtcIsacfix_kCdfShape +1368 +335, WebRtcIsacfix_kCdfShape +1368 +339, WebRtcIsacfix_kCdfShape +1368 +343, - WebRtcIsacfix_kCdfShape +1368 +349, WebRtcIsacfix_kCdfShape +1368 +355, WebRtcIsacfix_kCdfShape +1368 +361, - WebRtcIsacfix_kCdfShape +1368 +368, WebRtcIsacfix_kCdfShape +1368 +376, WebRtcIsacfix_kCdfShape +1368 +385, - WebRtcIsacfix_kCdfShape +1368 +397, WebRtcIsacfix_kCdfShape +1368 +411, WebRtcIsacfix_kCdfShape +1368 +429, - WebRtcIsacfix_kCdfShape +1368 +456, WebRtcIsacfix_kCdfShape +1368 +459, WebRtcIsacfix_kCdfShape +1368 +463, - WebRtcIsacfix_kCdfShape +1368 +467, WebRtcIsacfix_kCdfShape +1368 +473, WebRtcIsacfix_kCdfShape +1368 +478, - WebRtcIsacfix_kCdfShape +1368 +485, WebRtcIsacfix_kCdfShape +1368 +491, WebRtcIsacfix_kCdfShape +1368 +497, - WebRtcIsacfix_kCdfShape +1368 +505, WebRtcIsacfix_kCdfShape +1368 +514, WebRtcIsacfix_kCdfShape +1368 +523, - WebRtcIsacfix_kCdfShape +1368 +535, WebRtcIsacfix_kCdfShape +1368 +548, WebRtcIsacfix_kCdfShape +1368 +565, - WebRtcIsacfix_kCdfShape +1368 +585, WebRtcIsacfix_kCdfShape +1368 +611, WebRtcIsacfix_kCdfShape +1368 +640 - } -}; - -/* code length for all coefficients using different models */ - -const int16_t WebRtcIsacfix_kCodeLenGainQ11[392] = { - 25189, 16036, 8717, 358, 8757, 15706, 21456, 24397, 18502, 17559 - , 13794, 11088, 7480, 873, 6603, 11636, 14627, 16805, 19132, 26624 - , 26624, 19408, 13751, 7280, 583, 7591, 15178, 23773, 28672, 25189 - , 19045, 16442, 13412, 10397, 5893, 1338, 6376, 9992, 12074, 13853 - , 15781, 19821, 22819, 28672, 28672, 25189, 19858, 15781, 11262, 5477 - , 1298, 5632, 11814, 17234, 22020, 28672, 19677, 18125, 16587, 14521 - , 13032, 11196, 9249, 5411, 2495, 4994, 7975, 10234, 12308, 13892 - , 15148, 17944, 21725, 23917, 25189, 19539, 16293, 11531, 7808, 4475 - , 2739, 4872, 8089, 11314, 14992, 18105, 23257, 26624, 25189, 23257 - , 23257, 20982, 18697, 18023, 16338, 16036, 14539, 13695, 13146, 11763 - , 10754, 9074, 7260, 5584, 4430, 5553, 6848, 8344, 10141, 11636 - , 12535, 13416, 14342, 15477, 17296, 19282, 22349, 23773, 28672, 28672 - , 26624, 23773, 21456, 18023, 15118, 13362, 11212, 9293, 8043, 6985 - , 5908, 5721, 5853, 6518, 7316, 8360, 9716, 11289, 12912, 14652 - , 16969, 19858, 23773, 26624, 28013, 30720, 30720, 28672, 25426, 23141 - , 25426, 23773, 20720, 19408, 18697, 19282, 16859, 16338, 16026, 15377 - , 15021, 14319, 14251, 13937, 13260, 13017, 12332, 11703, 11430, 10359 - , 10128, 9405, 8757, 8223, 7974, 7859, 7646, 7673, 7997, 8580 - , 8880, 9061, 9866, 10397, 11358, 12200, 13244, 14157, 15021, 16026 - , 16490, 18697, 18479, 20011, 19677, 20720, 24576, 26276, 30720, 30720 - , 28672, 30720, 24068, 25189, 22437, 20345, 18479, 16396, 16026, 14928 - , 13877, 13450, 12696, 12766, 11626, 11098, 10159, 9998, 9437, 9275 - , 8783, 8552, 8629, 8488, 8522, 8454, 8571, 8775, 8915, 9427 - , 9483, 9851, 10260, 10933, 11131, 11974, 12560, 13833, 15080, 16304 - , 17491, 19017, 18697, 19408, 22020, 25189, 25426, 22819, 26276, 30720 - , 30720, 30720, 30720, 30720, 30720, 28672, 30720, 30720, 30720, 30720 - , 28013, 25426, 24397, 23773, 25189, 26624, 25189, 22437, 21725, 20011 - , 20527, 20720, 20771, 22020, 22020, 19858, 19408, 19972, 17866, 17360 - , 17791, 17219, 16805, 16927, 16067, 16162, 15661, 15178, 15021, 15209 - , 14845, 14570, 14490, 14490, 13733, 13617, 13794, 13577, 13312, 12824 - , 13032, 12683, 12189, 12469, 12109, 11940, 11636, 11617, 11932, 12294 - , 11578, 11775, 12039, 11654, 11560, 11439, 11909, 11421, 12029, 11513 - , 11773, 11899, 11560, 11805, 11476, 11664, 11963, 11647, 11754, 11963 - , 11703, 12211, 11932, 12074, 12469, 12535, 12560, 12912, 12783, 12866 - , 12884, 13378, 13957, 13775, 13635, 14019, 14545, 15240, 15520, 15554 - , 15697, 16490, 16396, 17281, 16599, 16969, 17963, 16859, 16983, 16805 - , 17099, 18210, 17219, 17646, 17700, 17646, 18297, 17425, 18479, 17791 - , 17718, 19282, 18672, 20173, 20982, 21725, 21456, 23773, 23257, 25189 - , 30720, 30720, 25189, 26624, 30720, 30720, 30720, 30720, 28672, 26276 - , 30720, 30720 -}; - -const int16_t WebRtcIsacfix_kCodeLenShapeQ11[578] = { - 0, 0, 0, 0, 0, 0, 0, 0, 0, 28672 - , 0, 26624, 1, 23773, 22819, 4, 20982, 18598, 10, 19282 - , 16587, 22, 16442, 26624, 13126, 60, 14245, 26624, 26624, 12736 - , 79, 12912, 25189, 22819, 9563, 249, 9474, 22349, 28672, 23257 - , 17944, 7980, 434, 8181, 16431, 26624, 0, 0, 0, 0 - , 28672, 0, 0, 0, 0, 0, 28672, 0, 22437, 3 - , 22437, 20982, 5, 20982, 16442, 22, 16752, 13814, 49, 14646 - , 11645, 116, 11734, 26624, 28672, 10613, 158, 11010, 24397, 19539 - , 8046, 453, 7709, 19017, 28672, 23257, 15110, 6770, 758, 6523 - , 14108, 24397, 28672, 0, 0, 0, 0, 28672, 0, 28672 - , 0, 26624, 1, 28672, 28672, 1, 26624, 24397, 2, 23257 - , 21725, 4, 20982, 17158, 18, 17281, 28672, 15178, 35, 15209 - , 12343, 92, 12320, 26624, 10344, 189, 10217, 30720, 22020, 9033 - , 322, 8549, 23773, 28672, 30720, 20622, 7666, 473, 7806, 20527 - , 24397, 14135, 5995, 960, 6018, 14872, 23773, 26624, 20928, 16293 - , 10636, 4926, 1588, 5256, 11088, 18043, 25189, 0, 0, 0 - , 0, 24397, 1, 25189, 20720, 5, 21456, 21209, 3, 25189 - , 20982, 5, 21456, 15818, 30, 15410, 13794, 60, 13416, 28672 - , 11162, 142, 11025, 9337, 231, 10094, 23773, 8338, 405, 7930 - , 26624, 19677, 6787, 613, 7318, 19161, 28672, 16442, 6319, 932 - , 5748, 15312, 25189, 28672, 28672, 28672, 13998, 5513, 1263, 5146 - , 14024, 24397, 22819, 15818, 9460, 4447, 2122, 4681, 9970, 15945 - , 22349, 28672, 30720, 22622, 19017, 14872, 10689, 7405, 4473, 2983 - , 4783, 7894, 11186, 14964, 18210, 24397, 0, 0, 30720, 0 - , 30720, 21456, 3, 23773, 14964, 39, 14757, 14179, 53, 13751 - , 14928, 36, 15272, 12430, 79, 13228, 9135, 285, 9077, 28672 - , 28672, 8377, 403, 7919, 26624, 28672, 23257, 7068, 560, 7473 - , 20345, 19677, 6770, 720, 6464, 18697, 25189, 16249, 5779, 1087 - , 5494, 15209, 22819, 30720, 20622, 12601, 5240, 1419, 5091, 12095 - , 19408, 26624, 22819, 16805, 10683, 4812, 2056, 4293, 9836, 16026 - , 24397, 25189, 18409, 13833, 8681, 4503, 2653, 4220, 8329, 13853 - , 19132, 26624, 25189, 20771, 17219, 12630, 9520, 6733, 4565, 3657 - , 4817, 7069, 10058, 13212, 16805, 21209, 26624, 26276, 28672, 28672 - , 26276, 23257, 20173, 19282, 16538, 15051, 12811, 10754, 9267, 7547 - , 6270, 5407, 5214, 6057, 7054, 8226, 9488, 10806, 12793, 14442 - , 16442, 19677, 22099, 26276, 28672, 0, 30720, 0, 30720, 11920 - , 56, 20720, 30720, 6766, 355, 13130, 30720, 30720, 22180, 5589 - , 736, 7902, 26624, 30720, 7634, 354, 9721, 30720, 30720, 9027 - , 246, 10117, 30720, 30720, 9630, 453, 6709, 23257, 30720, 25683 - , 14228, 6127, 1271, 4615, 15178, 30720, 30720, 23504, 12382, 5739 - , 2015, 3492, 10560, 22020, 26624, 30720, 30720, 23257, 13192, 4873 - , 1527, 5001, 12445, 22020, 30720, 30720, 30720, 30720, 19344, 10761 - , 4051, 1927, 5281, 10594, 17866, 28672, 30720, 30720, 30720, 21869 - , 15554, 10060, 5979, 2710, 3085, 7889, 14646, 21725, 28672, 30720 - , 30720, 30720, 30720, 30720, 30720, 30720, 22719, 17425, 13212, 8083 - , 4439, 2820, 4305, 8136, 12988, 17425, 21151, 28672, 28672, 30720 - , 30720, 30720, 28672, 20527, 19282, 14412, 10513, 7407, 5079, 3744 - , 4115, 6308, 9621, 13599, 17040, 22349, 28672, 30720, 30720, 30720 - , 30720, 30720, 30720, 29522, 19282, 14545, 11485, 9093, 6760, 5262 - , 4672, 4970, 6005, 7852, 9732, 12343, 14672, 19161, 22819, 25189 - , 30720, 30720, 28672, 30720, 30720, 20720, 18125, 14388, 12007, 9825 - , 8092, 7064, 6069, 5903, 5932, 6359, 7169, 8310, 9324, 10711 - , 11867, 13096, 14157, 16338, 17040, 19161, 21725, 23773, 30720, 30720 - , 26276, 25426, 24397, 28672, 28672, 23257, 22020, 22349, 18297, 17646 - , 16983, 16431, 16162, 15021, 15178, 13751, 12142, 10895, 10193, 9632 - , 9086, 8896, 8823, 8735, 8591, 8754, 8649, 8361, 8329, 8522 - , 8373, 8739, 8993, 9657, 10454, 11279, 11899, 12614, 14024, 14273 - , 15477, 15240, 16649, 17866, 18697, 21151, 22099, 0 - // The final 0 was added due to http://bugs.webrtc.org/10584. -}; - -/* left KLT transforms */ -const int16_t WebRtcIsacfix_kT1GainQ15[3][4] = { - { -26130, 19773, 19773, 26130 }, - { -26664, 19046, 19046, 26664 }, - { -23538, 22797, 22797, 23538 } -}; - - - -const int16_t WebRtcIsacfix_kT1ShapeQ15[3][324] = { - { 52,16,168,7,439,-138,-89,306,671,882, - 157,1301,291,1598,-3571,-1943,-1119,32404,96,-12, - 379,-64,-307,345,-836,539,1045,2541,-2865,-992, - 1683,-4717,5808,7427,30599,2319,183,-73,451,481, - 933,-198,781,-397,1244,-777,3690,-2414,149,-1356, - -2593,-31140,8289,-1737,-202,-14,-214,360,501,450, - -245,-7,797,3638,-2804,3042,-337,22137,-22103,2264, - 6838,-3381,305,172,263,-195,-355,351,179,513, - 2234,3343,5509,7531,19075,-17740,-16836,2244,-629,-1505, - -153,108,124,-324,2694,-124,1492,-850,5347,4285, - 7439,-10229,-22822,-12467,-12891,3645,822,-232,131,13, - 374,565,536,4681,1294,-1935,1926,-5734,-10643,26462, - -12480,-5589,-1038,-2468,964,-704,-247,-106,186,-558, - -4050,3760,2972,2141,-7393,6294,26740,11991,-3251,5461, - 5341,1574,2208,-51,-552,-297,-753,-154,2068,-5371, - 3578,4106,28043,-10533,8041,2353,2389,4609,3410,1906, - 351,-249,18,-15,1117,539,2870,9084,17585,-24528, - -366,-6490,2009,-3170,2942,1116,-232,1672,1065,606, - -399,-388,-518,38,3728,28948,-11936,4543,4104,-4441, - 1545,-4044,1485,622,-68,186,-473,135,-280,125, - -546,-1813,6989,6606,23711,19376,-2636,2870,-4553,-1687, - 878,-375,205,-208,-409,-108,-200,-45,-1670,-337, - 8213,-5524,-2334,5240,-12939,-26205,5937,-1582,-592,-959, - -5374,2449,3400,559,349,-492,668,12379,-27684,3419, - 5117,4415,-297,-8270,-1252,-3490,-1272,-1199,-3159,191, - 630,488,-797,-3071,12912,-27783,-10249,1047,647,619, - 111,-3722,-915,-1055,-502,5,-1384,-306,221,68, - 5219,13173,-26474,-11663,-5626,927,806,-1127,236,-589, - -522,-230,-312,-315,-428,-573,426,192,-11830,-26883, - -14121,-2785,-1429,-109,410,-832,-302,539,-459,104, - 1,-530,-202,-289,153,116,30082,-12944,-671,20, - 649,98,103,215,234,0,280,-51,-169,298, - 31,230,-73,-51 - }, - { -154,-7,-192,61,-739,-389,-947,-162,-60,94, - 511,-716,1520,-1428,4168,-2214,1816,32270,-123,-77, - -199,-99,-42,-588,203,-240,-930,-35,1580,234, - 3206,-5507,-1495,-10946,30000,-2667,-136,-176,-240,-175, - -204,-661,-1796,-1039,-1271,498,3143,734,2663,2699, - -8127,29333,10495,2356,-72,113,-91,118,-2840,-723, - -1733,-1158,-389,-2116,-3054,-3,-5179,8071,29546,6308, - 5657,-3178,-186,-294,-473,-635,1213,-983,-1437,-1715, - -1094,1280,-92,-9573,948,29576,-7060,-5921,2954,1349, - -337,-108,-1099,962,418,-413,-1149,-334,1241,3975, - -6825,26725,-14377,7051,-4772,-1707,2335,2008,-150,570, - 1371,42,-1649,-619,2039,3369,-1225,1583,-2755,-15207, - -27504,-4855,-4304,1495,2733,1324,15,-448,403,353, - 3016,-1242,2338,2673,2064,-7496,-30447,-3686,5833,-1301, - -2455,2122,1519,608,43,-653,773,-3072,912,-1537, - 4505,10284,30237,1549,3200,-691,205,1702,658,1014, - 1499,148,79,-322,-1162,-4639,-813,7536,3204,29109, - -10747,-26,1611,2286,2114,2561,1022,372,348,207, - 1062,-1088,-443,-9849,2381,5671,29097,-7612,-2927,3853, - 194,1155,275,1438,1438,1312,581,888,-784,906, - 112,-11103,25104,14438,-9311,-3068,1210,368,370,-940, - -2434,-1148,1925,392,657,258,-526,1475,-2281,-4265, - -1880,1534,2185,-1472,959,-30934,6306,3114,-4109,1768, - -2612,-703,45,644,2185,2033,5670,7211,19114,-22427, - 6432,5150,-4090,-2694,3860,1245,-596,293,1829,369, - -319,229,-3256,2170,-6374,-26216,-4570,-16053,-5766,-262, - -2006,2873,-1477,147,378,-1544,-344,-544,-985,-481, - 4210,4542,30757,-7291,-4863,1529,-2079,-628,-603,-783, - -408,1646,697,808,-620,-292,181,158,-13313,-29173, - 5984,-1262,859,-1776,-558,-24,-883,-1421,739,210, - -531,-285,131,-160,-246,-56,29345,-13706,-2859,-2966, - -300,-970,-2382,-268,-103,-636,-12,-62,-691,-253, - -147,-127,27,66 - }, - { 55,-212,-198,489,-274,81,682,399,328,-934, - -389,-37,1357,-3632,5276,6581,-9493,-29921,29,-45, - 2,190,172,-15,311,-130,-1085,-25,324,-684, - 3223,-6580,4485,-5280,-29521,9933,82,-320,-530,229, - -705,-533,-414,848,-1842,-4473,1390,-857,6717,-6692, - 4648,29397,576,8339,-68,-85,238,-330,264,-1012, - -381,-203,-3384,-3329,3906,6810,3790,-6250,28312,-8078, - 8089,1565,160,-569,-612,-613,-1063,-1928,-1125,3421, - -7481,-7484,4942,-6984,4330,-25591,-10574,-6982,5682,-1781, - -308,89,178,-1715,-420,-3530,-5776,1219,-8617,-7137, - 7015,4981,24875,12657,-5408,-3356,-785,-1972,326,-858, - -506,-3382,-986,-6258,-2259,4015,-8374,-10482,3127,23826, - -14126,-514,-5417,2178,-2912,-17,-587,80,67,-5881, - -1702,-5351,-4481,398,-10156,-225,20727,-15460,-11603,7752, - 3660,1714,-2001,-359,499,-527,-1225,-7820,-1297,-6326, - -8526,7900,-18328,13311,-17488,-2926,-196,-17,2281,873, - 480,-160,-624,471,780,-8729,1707,-14262,-20647,1721, - 18590,-2206,-1214,-1066,312,-2602,783,-412,-113,49, - -119,1305,-2371,-15132,-1833,-18252,20295,-8316,2227,341, - -2074,-702,3082,-262,-465,-198,430,30,-70,-788, - 2342,-25132,-4863,19783,-484,2137,2811,-1906,799,1586, - 962,-734,-191,-30,-129,-93,-1126,1729,5860,-2030, - 8953,603,-3338,-10869,-1144,22070,12130,10513,3191,-6881, - -3514,2090,711,-666,1843,-5997,-5681,2921,-17641,-2801, - 4969,18590,7169,12214,8587,4405,3008,-1074,-371,-77, - 253,331,-5611,5014,13152,-1985,18483,-1696,8043,20463, - 2381,-393,1688,-1205,618,1220,457,248,-83,176, - 7920,-13676,-22139,-3038,17402,2036,844,3258,994,719, - 2087,-44,426,494,12,-91,46,5,-14204,22912, - -18156,-361,442,2298,-829,2229,386,1433,1335,1323, - 55,-592,-139,49,-12,-57,27783,17134,350,-282, - 552,158,142,2488,465,329,1087,118,143,10, - 56,65,-15,-31 - } -}; - -/* right KLT transforms */ -const int16_t WebRtcIsacfix_kT2GainQ15[3][36] = { - { 4775, -14892, 20313, -17104, 10533, -3613, -6782, 16044, -8889, - -11019, 21330, -10720, 13193, -15678, -11101, 14461, 12250, -13096, - -16951, 2167, 16066, 15569, -702, -16754, -19195, -12823, -4321, - 5128, 13348, 17825, 13232, 13404, 13494, 13490, 13383, 13261 - }, - { -3725, 11408, -18493, 20031, -13097, 3865, 9344, -19294, 10740, - 8856, -18432, 8982, 13975, -14444, -11930, 11774, 14285, -13594, - -16323, -4, 16340, 15609, 359, -17220, -18401, -13471, -4643, - 5225, 13375, 18053, 13124, 13463, 13621, 13583, 13393, 13072 - }, - { -3513, 11402, -17883, 19504, -14399, 4885, 8702, -19513, 12046, - 8533, -18110, 8447, 12778, -14838, -12444, 13177, 14107, -12759, - -17268, 914, 15822, 15661, 838, -16686, -18907, -12936, -4820, - 4175, 12398, 18830, 12913, 13215, 13433, 13572, 13601, 13518 - } -}; - -const int16_t WebRtcIsacfix_kT2ShapeQ15[3][36] = { - { 4400, -11512, 17205, -19470, 14770, -5345, 9784, -19222, 11228, - 6842, -18371, 9909, 14191, -13496, -11563, 14015, 11827, -14839, - -15439, 948, 17802, 14827, -2053, -17132, 18723, 14516, 4135, - -6822, -13869, -16016, 12975, 13341, 13563, 13603, 13478, 13296 - }, - { 5420, -14215, 19060, -18073, 11709, -3911, 9645, -18335, 7717, - 10842, -19283, 9777, 14898, -12555, -13661, 11668, 13520, -13733, - -15936, -1358, 15671, 16728, 328, -17100, 17527, 13973, 5587, - -5194, -14165, -17677, 12970, 13446, 13693, 13660, 13462, 13015 - }, - { 4386, -12426, 18019, -18895, 13894, -5034, 9713, -19270, 10283, - 8692, -18439, 9317, 13992, -13454, -13241, 12850, 13366, -13336, - -16334, -498, 15976, 16213, -114, -16987, 18191, 13659, 4958, - -5116, -13444, -18021, 12911, 13424, 13718, 13674, 13464, 13054 - } -}; - -/* means of log gains and LAR coefficients*/ -const int16_t WebRtcIsacfix_kMeansGainQ8[3][12] = { - { -1758, -1370, -1758, -1373, -1757, -1375, - -1758, -1374, -1758, -1373, -1755, -1370 - }, - { -1569, -1224, -1569, -1225, -1569, -1227, - -1569, -1226, -1567, -1225, -1565, -1224 - }, - { -1452, -957, -1447, -951, -1438, -944, - -1431, -938, -1419, -931, -1406, -926 - } -}; - - -const int32_t WebRtcIsacfix_kMeansShapeQ17[3][108] = { - { -119581, 34418, -44193, 11112, -4428, 18906, 9222, 8068, 1953, 5425, - 1871, 1689, 109933, 33751, 10471, -2566, 1090, 2320, -119219, 33728, - -43759, 11450, -4870, 19117, 9174, 8037, 1972, 5331, 1872, 1843, - 109899, 34301, 10629, -2316, 1272, 2562, -118608, 32318, -44012, 11591, - -4914, 18932, 9456, 8088, 1900, 5419, 1723, 1853, 109963, 35059, - 10745, -2335, 1161, 2520, -119174, 32107, -44462, 11635, -4694, 18611, - 9757, 8108, 1969, 5486, 1673, 1777, 109636, 34907, 10643, -2406, - 1034, 2420, -118597, 32320, -44590, 10854, -4569, 18821, 9701, 7866, - 2003, 5577, 1732, 1626, 109913, 34448, 10714, -2752, 990, 2228, - -118138, 32996, -44352, 10334, -3772, 18488, 9464, 7865, 2208, 5540, - 1745, 1664, 109880, 33381, 10640, -2779, 980, 2054 - }, - { -146328, 46370, 1047, 26431, 10035, 13933, 6415, 14359, -2368, 6661, - 2269, 1764, 96623, 7802, 4163, 10742, 1643, 2954, -146871, 46561, 1127, - 26225, 10113, 14096, 6771, 14323, -2037, 6788, 2297, 1761, 96324, 8382, - 4309, 10450, 1695, 3016, -146502, 46475, 1580, 26118, 10487, 14179, 6622, - 14439, -2034, 6757, 2342, 1761, 95869, 8966, 4347, 10358, 1999, 2855, - -146958, 47717, 826, 25952, 10263, 14061, 5266, 13681, -2417, 6582, 2047, - 1608, 96257, 9107, 4452, 10301, 1792, 2676, -146992, 47123, 446, 25822, - 10405, 14292, 5140, 13804, -2403, 6496, 1834, 1735, 97489, 9253, 4414, - 10684, 1549, 2721, -145811, 46182, 901, 26482, 10241, 14524, 6075, 14514, - -2147, 6691, 2196, 1899, 97011, 8178, 4102, 10758, 1638, 2869 - }, - { -166617, 46969, -43908, 17726, 6330, 25615, 6913, 5450, -2301, 1984, - 507, 2883, 149998, 28709, 19333, 16703, 11093, 8965, -168254, 46604, - -44315, 17862, 6474, 25746, 7018, 5373, -2343, 1930, 513, 2819, 150391, - 28627, 19194, 16678, 10998, 8929, -169093, 46084, -44767, 17427, 6401, - 25674, 7147, 5472, -2336, 1820, 491, 2802, 149860, 28430, 19064, 16524, - 10898, 8875, -170205, 46189, -44877, 17403, 6190, 25209, 7035, 5673, -2173, - 1894, 574, 2756, 148830, 28230, 18819, 16418, 10789, 8811, -171263, 45045, - -44834, 16858, 6103, 24726, 7014, 5713, -2103, 1877, 518, 2729, 147073, - 27744, 18629, 16277, 10690, 8703, -171720, 44153, -45062, 15951, 5872, - 24429, 7044, 5585, -2082, 1807, 519, 2769, 144791, 27402, 18490, 16126, - 10548, 8635 - } -}; diff --git a/modules/audio_coding/codecs/isac/fix/source/lpc_tables.h b/modules/audio_coding/codecs/isac/fix/source/lpc_tables.h deleted file mode 100644 index 50e1b12459..0000000000 --- a/modules/audio_coding/codecs/isac/fix/source/lpc_tables.h +++ /dev/null @@ -1,99 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/* - * lpc_tables.h - * - * header file for coding tables for the LPC coefficients - * - */ - -#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_LPC_TABLES_H_ -#define MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_LPC_TABLES_H_ - -#include - -#include "modules/audio_coding/codecs/isac/fix/source/settings.h" - -/* indices of KLT coefficients used */ -extern const uint16_t WebRtcIsacfix_kSelIndGain[12]; - -extern const uint16_t WebRtcIsacfix_kSelIndShape[108]; - -/* cdf array for model indicator */ -extern const uint16_t WebRtcIsacfix_kModelCdf[KLT_NUM_MODELS + 1]; - -/* pointer to cdf array for model indicator */ -extern const uint16_t* WebRtcIsacfix_kModelCdfPtr[1]; - -/* initial cdf index for decoder of model indicator */ -extern const uint16_t WebRtcIsacfix_kModelInitIndex[1]; - -/* offset to go from rounded value to quantization index */ -extern const int16_t WebRtcIsacfix_kQuantMinGain[12]; - -extern const int16_t WebRtcIsacfix_kQuantMinShape[108]; - -/* maximum quantization index */ -extern const uint16_t WebRtcIsacfix_kMaxIndGain[12]; - -extern const uint16_t WebRtcIsacfix_kMaxIndShape[108]; - -/* index offset */ -extern const uint16_t WebRtcIsacfix_kOffsetGain[KLT_NUM_MODELS][12]; - -extern const uint16_t WebRtcIsacfix_kOffsetShape[KLT_NUM_MODELS][108]; - -/* initial cdf index for KLT coefficients */ -extern const uint16_t WebRtcIsacfix_kInitIndexGain[KLT_NUM_MODELS][12]; - -extern const uint16_t WebRtcIsacfix_kInitIndexShape[KLT_NUM_MODELS][108]; - -/* offsets for quantizer representation levels */ -extern const uint16_t WebRtcIsacfix_kOfLevelsGain[3]; - -extern const uint16_t WebRtcIsacfix_kOfLevelsShape[3]; - -/* quantizer representation levels */ -extern const int32_t WebRtcIsacfix_kLevelsGainQ17[1176]; - -extern const int16_t WebRtcIsacfix_kLevelsShapeQ10[1735]; - -/* cdf tables for quantizer indices */ -extern const uint16_t WebRtcIsacfix_kCdfGain[1212]; - -extern const uint16_t WebRtcIsacfix_kCdfShape[2059]; - -/* pointers to cdf tables for quantizer indices */ -extern const uint16_t* WebRtcIsacfix_kCdfGainPtr[KLT_NUM_MODELS][12]; - -extern const uint16_t* WebRtcIsacfix_kCdfShapePtr[KLT_NUM_MODELS][108]; - -/* code length for all coefficients using different models */ -extern const int16_t WebRtcIsacfix_kCodeLenGainQ11[392]; - -extern const int16_t WebRtcIsacfix_kCodeLenShapeQ11[578]; - -/* left KLT transforms */ -extern const int16_t WebRtcIsacfix_kT1GainQ15[KLT_NUM_MODELS][4]; - -extern const int16_t WebRtcIsacfix_kT1ShapeQ15[KLT_NUM_MODELS][324]; - -/* right KLT transforms */ -extern const int16_t WebRtcIsacfix_kT2GainQ15[KLT_NUM_MODELS][36]; - -extern const int16_t WebRtcIsacfix_kT2ShapeQ15[KLT_NUM_MODELS][36]; - -/* means of log gains and LAR coefficients */ -extern const int16_t WebRtcIsacfix_kMeansGainQ8[KLT_NUM_MODELS][12]; - -extern const int32_t WebRtcIsacfix_kMeansShapeQ17[3][108]; - -#endif /* MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_LPC_TABLES_H_ */ diff --git a/modules/audio_coding/codecs/isac/fix/source/pitch_estimator.c b/modules/audio_coding/codecs/isac/fix/source/pitch_estimator.c deleted file mode 100644 index 78cb93f7ae..0000000000 --- a/modules/audio_coding/codecs/isac/fix/source/pitch_estimator.c +++ /dev/null @@ -1,435 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#include "modules/audio_coding/codecs/isac/fix/source/pitch_estimator.h" -#include "common_audio/signal_processing/include/signal_processing_library.h" -#include "rtc_base/compile_assert_c.h" - -/* log2[0.2, 0.5, 0.98] in Q8 */ -static const int16_t kLogLagWinQ8[3] = { - -594, -256, -7 -}; - -/* [1 -0.75 0.25] in Q12 */ -static const int16_t kACoefQ12[3] = { - 4096, -3072, 1024 -}; - -int32_t WebRtcIsacfix_Log2Q8(uint32_t x) { - int32_t zeros; - int16_t frac; - - zeros=WebRtcSpl_NormU32(x); - frac = (int16_t)(((x << zeros) & 0x7FFFFFFF) >> 23); - /* log2(magn(i)) */ - - return ((31 - zeros) << 8) + frac; -} - -static __inline int16_t Exp2Q10(int16_t x) { // Both in and out in Q10 - - int16_t tmp16_1, tmp16_2; - - tmp16_2=(int16_t)(0x0400|(x&0x03FF)); - tmp16_1 = -(x >> 10); - if(tmp16_1>0) - return tmp16_2 >> tmp16_1; - else - return tmp16_2 << -tmp16_1; - -} - - - -/* 1D parabolic interpolation . All input and output values are in Q8 */ -static __inline void Intrp1DQ8(int32_t *x, int32_t *fx, int32_t *y, int32_t *fy) { - - int16_t sign1=1, sign2=1; - int32_t r32, q32, t32, nom32, den32; - int16_t t16, tmp16, tmp16_1; - - if ((fx[0]>0) && (fx[2]>0)) { - r32=fx[1]-fx[2]; - q32=fx[0]-fx[1]; - nom32=q32+r32; - den32 = (q32 - r32) * 2; - if (nom32<0) - sign1=-1; - if (den32<0) - sign2=-1; - - /* t = (q32+r32)/(2*(q32-r32)) = (fx[0]-fx[1] + fx[1]-fx[2])/(2 * fx[0]-fx[1] - (fx[1]-fx[2]))*/ - /* (Signs are removed because WebRtcSpl_DivResultInQ31 can't handle negative numbers) */ - /* t in Q31, without signs */ - t32 = WebRtcSpl_DivResultInQ31(nom32 * sign1, den32 * sign2); - - t16 = (int16_t)(t32 >> 23); /* Q8 */ - t16=t16*sign1*sign2; /* t in Q8 with signs */ - - *y = x[0]+t16; /* Q8 */ - // *y = x[1]+t16; /* Q8 */ - - /* The following code calculates fy in three steps */ - /* fy = 0.5 * t * (t-1) * fx[0] + (1-t*t) * fx[1] + 0.5 * t * (t+1) * fx[2]; */ - - /* Part I: 0.5 * t * (t-1) * fx[0] */ - tmp16_1 = (int16_t)(t16 * t16); /* Q8*Q8=Q16 */ - tmp16_1 >>= 2; /* Q16>>2 = Q14 */ - t16 <<= 6; /* Q8<<6 = Q14 */ - tmp16 = tmp16_1-t16; - *fy = WEBRTC_SPL_MUL_16_32_RSFT15(tmp16, fx[0]); /* (Q14 * Q8 >>15)/2 = Q8 */ - - /* Part II: (1-t*t) * fx[1] */ - tmp16 = 16384-tmp16_1; /* 1 in Q14 - Q14 */ - *fy += WEBRTC_SPL_MUL_16_32_RSFT14(tmp16, fx[1]);/* Q14 * Q8 >> 14 = Q8 */ - - /* Part III: 0.5 * t * (t+1) * fx[2] */ - tmp16 = tmp16_1+t16; - *fy += WEBRTC_SPL_MUL_16_32_RSFT15(tmp16, fx[2]);/* (Q14 * Q8 >>15)/2 = Q8 */ - } else { - *y = x[0]; - *fy= fx[1]; - } -} - - -static void FindFour32(int32_t *in, int16_t length, int16_t *bestind) -{ - int32_t best[4]= {-100, -100, -100, -100}; - int16_t k; - - for (k=0; k best[3]) { - if (in[k] > best[2]) { - if (in[k] > best[1]) { - if (in[k] > best[0]) { // The Best - best[3] = best[2]; - bestind[3] = bestind[2]; - best[2] = best[1]; - bestind[2] = bestind[1]; - best[1] = best[0]; - bestind[1] = bestind[0]; - best[0] = in[k]; - bestind[0] = k; - } else { // 2nd best - best[3] = best[2]; - bestind[3] = bestind[2]; - best[2] = best[1]; - bestind[2] = bestind[1]; - best[1] = in[k]; - bestind[1] = k; - } - } else { // 3rd best - best[3] = best[2]; - bestind[3] = bestind[2]; - best[2] = in[k]; - bestind[2] = k; - } - } else { // 4th best - best[3] = in[k]; - bestind[3] = k; - } - } - } -} - - - - - -extern void WebRtcIsacfix_PCorr2Q32(const int16_t *in, int32_t *logcorQ8); - - - -void WebRtcIsacfix_InitialPitch(const int16_t *in, /* Q0 */ - PitchAnalysisStruct *State, - int16_t *lagsQ7 /* Q7 */ - ) -{ - int16_t buf_dec16[PITCH_CORR_LEN2+PITCH_CORR_STEP2+PITCH_MAX_LAG/2+2]; - int32_t *crrvecQ8_1,*crrvecQ8_2; - int32_t cv1q[PITCH_LAG_SPAN2+2],cv2q[PITCH_LAG_SPAN2+2], peakvq[PITCH_LAG_SPAN2+2]; - int k; - int16_t peaks_indq; - int16_t peakiq[PITCH_LAG_SPAN2]; - int32_t corr; - int32_t corr32, corr_max32, corr_max_o32; - int16_t npkq; - int16_t best4q[4]={0,0,0,0}; - int32_t xq[3],yq[1],fyq[1]; - int32_t *fxq; - int32_t best_lag1q, best_lag2q; - int32_t tmp32a,tmp32b,lag32,ratq; - int16_t start; - int16_t oldgQ12, tmp16a, tmp16b, gain_bias16,tmp16c, tmp16d, bias16; - int32_t tmp32c,tmp32d, tmp32e; - int16_t old_lagQ; - int32_t old_lagQ8; - int32_t lagsQ8[4]; - - old_lagQ = State->PFstr_wght.oldlagQ7; // Q7 - old_lagQ8 = old_lagQ << 1; // Q8 - - oldgQ12= State->PFstr_wght.oldgainQ12; - - crrvecQ8_1=&cv1q[1]; - crrvecQ8_2=&cv2q[1]; - - - /* copy old values from state buffer */ - memcpy(buf_dec16, State->dec_buffer16, sizeof(State->dec_buffer16)); - - /* decimation; put result after the old values */ - WebRtcIsacfix_DecimateAllpass32(in, State->decimator_state32, PITCH_FRAME_LEN, - &buf_dec16[PITCH_CORR_LEN2+PITCH_CORR_STEP2+PITCH_MAX_LAG/2-PITCH_FRAME_LEN/2+2]); - - /* low-pass filtering */ - start= PITCH_CORR_LEN2+PITCH_CORR_STEP2+PITCH_MAX_LAG/2-PITCH_FRAME_LEN/2+2; - WebRtcSpl_FilterARFastQ12(&buf_dec16[start],&buf_dec16[start],(int16_t*)kACoefQ12,3, PITCH_FRAME_LEN/2); - - /* copy end part back into state buffer */ - for (k = 0; k < (PITCH_CORR_LEN2+PITCH_CORR_STEP2+PITCH_MAX_LAG/2-PITCH_FRAME_LEN/2+2); k++) - State->dec_buffer16[k] = buf_dec16[k+PITCH_FRAME_LEN/2]; - - - /* compute correlation for first and second half of the frame */ - WebRtcIsacfix_PCorr2Q32(buf_dec16, crrvecQ8_1); - WebRtcIsacfix_PCorr2Q32(buf_dec16 + PITCH_CORR_STEP2, crrvecQ8_2); - - - /* bias towards pitch lag of previous frame */ - tmp32a = WebRtcIsacfix_Log2Q8((uint32_t) old_lagQ8) - 2304; - // log2(0.5*oldlag) in Q8 - tmp32b = oldgQ12 * oldgQ12 >> 10; // Q12 & * 4.0; - gain_bias16 = (int16_t) tmp32b; //Q12 - if (gain_bias16 > 3276) gain_bias16 = 3276; // 0.8 in Q12 - - - for (k = 0; k < PITCH_LAG_SPAN2; k++) - { - if (crrvecQ8_1[k]>0) { - tmp32b = WebRtcIsacfix_Log2Q8((uint32_t) (k + (PITCH_MIN_LAG/2-2))); - tmp16a = (int16_t) (tmp32b - tmp32a); // Q8 & fabs(ratio)<4 - tmp32c = tmp16a * tmp16a >> 6; // Q10 - tmp16b = (int16_t) tmp32c; // Q10 & <8 - tmp32d = tmp16b * 177 >> 8; // mult with ln2 in Q8 - tmp16c = (int16_t) tmp32d; // Q10 & <4 - tmp16d = Exp2Q10((int16_t) -tmp16c); //Q10 - tmp32c = gain_bias16 * tmp16d >> 13; // Q10 & * 0.5 - bias16 = (int16_t) (1024 + tmp32c); // Q10 - tmp32b = WebRtcIsacfix_Log2Q8((uint32_t)bias16) - 2560; - // Q10 in -> Q8 out with 10*2^8 offset - crrvecQ8_1[k] += tmp32b ; // -10*2^8 offset - } - } - - /* taper correlation functions */ - for (k = 0; k < 3; k++) { - crrvecQ8_1[k] += kLogLagWinQ8[k]; - crrvecQ8_2[k] += kLogLagWinQ8[k]; - - crrvecQ8_1[PITCH_LAG_SPAN2-1-k] += kLogLagWinQ8[k]; - crrvecQ8_2[PITCH_LAG_SPAN2-1-k] += kLogLagWinQ8[k]; - } - - - /* Make zeropadded corr vectors */ - cv1q[0]=0; - cv2q[0]=0; - cv1q[PITCH_LAG_SPAN2+1]=0; - cv2q[PITCH_LAG_SPAN2+1]=0; - corr_max32 = 0; - - for (k = 1; k <= PITCH_LAG_SPAN2; k++) - { - - - corr32=crrvecQ8_1[k-1]; - if (corr32 > corr_max32) - corr_max32 = corr32; - - corr32=crrvecQ8_2[k-1]; - corr32 += -4; // Compensate for later (log2(0.99)) - - if (corr32 > corr_max32) - corr_max32 = corr32; - - } - - /* threshold value to qualify as a peak */ - // corr_max32 += -726; // log(0.14)/log(2.0) in Q8 - corr_max32 += -1000; // log(0.14)/log(2.0) in Q8 - corr_max_o32 = corr_max32; - - - /* find peaks in corr1 */ - peaks_indq = 0; - for (k = 1; k <= PITCH_LAG_SPAN2; k++) - { - corr32=cv1q[k]; - if (corr32>corr_max32) { // Disregard small peaks - if ((corr32>=cv1q[k-1]) && (corr32>cv1q[k+1])) { // Peak? - peakvq[peaks_indq] = corr32; - peakiq[peaks_indq++] = k; - } - } - } - - - /* find highest interpolated peak */ - corr_max32=0; - best_lag1q =0; - if (peaks_indq > 0) { - FindFour32(peakvq, (int16_t) peaks_indq, best4q); - npkq = WEBRTC_SPL_MIN(peaks_indq, 4); - - for (k=0;k> 8; - tmp32c= tmp32b + 256; - *fyq += tmp32c; - if (*fyq > corr_max32) { - corr_max32 = *fyq; - best_lag1q = *yq; - } - } - tmp32b = (best_lag1q - OFFSET_Q8) * 2; - lagsQ8[0] = tmp32b + PITCH_MIN_LAG_Q8; - lagsQ8[1] = lagsQ8[0]; - } else { - lagsQ8[0] = old_lagQ8; - lagsQ8[1] = lagsQ8[0]; - } - - /* Bias towards constant pitch */ - tmp32a = lagsQ8[0] - PITCH_MIN_LAG_Q8; - ratq = (tmp32a >> 1) + OFFSET_Q8; - - for (k = 1; k <= PITCH_LAG_SPAN2; k++) - { - tmp32a = k << 7; // 0.5*k Q8 - tmp32b = tmp32a * 2 - ratq; // Q8 - tmp32c = (int16_t)tmp32b * (int16_t)tmp32b >> 8; // Q8 - - tmp32b = tmp32c + (ratq >> 1); - // (k-r)^2 + 0.5 * r Q8 - tmp32c = WebRtcIsacfix_Log2Q8((uint32_t)tmp32a) - 2048; - // offset 8*2^8 , log2(0.5*k) Q8 - tmp32d = WebRtcIsacfix_Log2Q8((uint32_t)tmp32b) - 2048; - // offset 8*2^8 , log2(0.5*k) Q8 - tmp32e = tmp32c - tmp32d; - - cv2q[k] += tmp32e >> 1; - - } - - /* find peaks in corr2 */ - corr_max32 = corr_max_o32; - peaks_indq = 0; - - for (k = 1; k <= PITCH_LAG_SPAN2; k++) - { - corr=cv2q[k]; - if (corr>corr_max32) { // Disregard small peaks - if ((corr>=cv2q[k-1]) && (corr>cv2q[k+1])) { // Peak? - peakvq[peaks_indq] = corr; - peakiq[peaks_indq++] = k; - } - } - } - - - - /* find highest interpolated peak */ - corr_max32 = 0; - best_lag2q =0; - if (peaks_indq > 0) { - - FindFour32(peakvq, (int16_t) peaks_indq, best4q); - npkq = WEBRTC_SPL_MIN(peaks_indq, 4); - for (k=0;k> 8; - tmp32c= tmp32b + 256; - *fyq += tmp32c; - if (*fyq > corr_max32) { - corr_max32 = *fyq; - best_lag2q = *yq; - } - } - - tmp32b = (best_lag2q - OFFSET_Q8) * 2; - lagsQ8[2] = tmp32b + PITCH_MIN_LAG_Q8; - lagsQ8[3] = lagsQ8[2]; - } else { - lagsQ8[2] = lagsQ8[0]; - lagsQ8[3] = lagsQ8[0]; - } - - lagsQ7[0] = (int16_t)(lagsQ8[0] >> 1); - lagsQ7[1] = (int16_t)(lagsQ8[1] >> 1); - lagsQ7[2] = (int16_t)(lagsQ8[2] >> 1); - lagsQ7[3] = (int16_t)(lagsQ8[3] >> 1); -} - - - -void WebRtcIsacfix_PitchAnalysis(const int16_t *inn, /* PITCH_FRAME_LEN samples */ - int16_t *outQ0, /* PITCH_FRAME_LEN+QLOOKAHEAD samples */ - PitchAnalysisStruct *State, - int16_t *PitchLags_Q7, - int16_t *PitchGains_Q12) -{ - int16_t inbufQ0[PITCH_FRAME_LEN + QLOOKAHEAD]; - int16_t k; - - /* inital pitch estimate */ - WebRtcIsacfix_InitialPitch(inn, State, PitchLags_Q7); - - - /* Calculate gain */ - WebRtcIsacfix_PitchFilterGains(inn, &(State->PFstr_wght), PitchLags_Q7, PitchGains_Q12); - - /* concatenate previous input's end and current input */ - for (k = 0; k < QLOOKAHEAD; k++) { - inbufQ0[k] = State->inbuf[k]; - } - for (k = 0; k < PITCH_FRAME_LEN; k++) { - inbufQ0[k+QLOOKAHEAD] = (int16_t) inn[k]; - } - - /* lookahead pitch filtering for masking analysis */ - WebRtcIsacfix_PitchFilter(inbufQ0, outQ0, &(State->PFstr), PitchLags_Q7,PitchGains_Q12, 2); - - - /* store last part of input */ - for (k = 0; k < QLOOKAHEAD; k++) { - State->inbuf[k] = inbufQ0[k + PITCH_FRAME_LEN]; - } -} diff --git a/modules/audio_coding/codecs/isac/fix/source/pitch_estimator.h b/modules/audio_coding/codecs/isac/fix/source/pitch_estimator.h deleted file mode 100644 index 4303c82711..0000000000 --- a/modules/audio_coding/codecs/isac/fix/source/pitch_estimator.h +++ /dev/null @@ -1,67 +0,0 @@ -/* - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/* - * pitch_estimator.h - * - * Pitch functions - * - */ - -#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_PITCH_ESTIMATOR_H_ -#define MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_PITCH_ESTIMATOR_H_ - -#include "modules/audio_coding/codecs/isac/fix/source/structs.h" - -void WebRtcIsacfix_PitchAnalysis( - const int16_t* in, /* PITCH_FRAME_LEN samples */ - int16_t* outQ0, /* PITCH_FRAME_LEN+QLOOKAHEAD samples */ - PitchAnalysisStruct* State, - int16_t* lagsQ7, - int16_t* PitchGains_Q12); - -void WebRtcIsacfix_InitialPitch(const int16_t* in, - PitchAnalysisStruct* State, - int16_t* qlags); - -void WebRtcIsacfix_PitchFilter(int16_t* indatFix, - int16_t* outdatQQ, - PitchFiltstr* pfp, - int16_t* lagsQ7, - int16_t* gainsQ12, - int16_t type); - -void WebRtcIsacfix_PitchFilterCore(int loopNumber, - int16_t gain, - size_t index, - int16_t sign, - int16_t* inputState, - int16_t* outputBuff2, - const int16_t* coefficient, - int16_t* inputBuf, - int16_t* outputBuf, - int* index2); - -void WebRtcIsacfix_PitchFilterGains(const int16_t* indatQ0, - PitchFiltstr* pfp, - int16_t* lagsQ7, - int16_t* gainsQ12); - -void WebRtcIsacfix_DecimateAllpass32( - const int16_t* in, - int32_t* state_in, /* array of size: 2*ALLPASSSECTIONS+1 */ - int16_t N, /* number of input samples */ - int16_t* out); /* array of size N/2 */ - -int32_t WebRtcIsacfix_Log2Q8(uint32_t x); - -void WebRtcIsacfix_PCorr2Q32(const int16_t* in, int32_t* logcorQ8); - -#endif /* MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_PITCH_ESTIMATOR_H_ */ diff --git a/modules/audio_coding/codecs/isac/fix/source/pitch_estimator_c.c b/modules/audio_coding/codecs/isac/fix/source/pitch_estimator_c.c deleted file mode 100644 index c4af9ab32a..0000000000 --- a/modules/audio_coding/codecs/isac/fix/source/pitch_estimator_c.c +++ /dev/null @@ -1,114 +0,0 @@ -/* - * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#include "modules/audio_coding/codecs/isac/fix/source/pitch_estimator.h" - -#ifdef WEBRTC_HAS_NEON -#include -#endif - -#include "common_audio/signal_processing/include/signal_processing_library.h" -#include "rtc_base/compile_assert_c.h" - -extern int32_t WebRtcIsacfix_Log2Q8(uint32_t x); - -void WebRtcIsacfix_PCorr2Q32(const int16_t* in, int32_t* logcorQ8) { - int16_t scaling,n,k; - int32_t csum32, lys, lcs; - int64_t ysum64; - const int32_t oneQ8 = 1 << 8; // 1.00 in Q8 - const int16_t* x; - const int16_t* inptr; - - x = in + PITCH_MAX_LAG / 2 + 2; - scaling = WebRtcSpl_GetScalingSquare((int16_t*)in, - PITCH_CORR_LEN2, - PITCH_CORR_LEN2); - ysum64 = 1; - csum32 = 0; - x = in + PITCH_MAX_LAG / 2 + 2; - for (n = 0; n < PITCH_CORR_LEN2; n++) { - ysum64 += in[n] * in[n] >> scaling; // Q0 - csum32 += x[n] * in[n] >> scaling; // Q0 - } - logcorQ8 += PITCH_LAG_SPAN2 - 1; - lys = WebRtcIsacfix_Log2Q8((uint32_t)ysum64) >> 1; // Q8, sqrt(ysum) - if (csum32 > 0) { - lcs = WebRtcIsacfix_Log2Q8((uint32_t)csum32); // 2log(csum) in Q8 - if (lcs > (lys + oneQ8)) { // csum/sqrt(ysum) > 2 in Q8 - *logcorQ8 = lcs - lys; // log2(csum/sqrt(ysum)) - } else { - *logcorQ8 = oneQ8; // 1.00 - } - } else { - *logcorQ8 = 0; - } - - - for (k = 1; k < PITCH_LAG_SPAN2; k++) { - inptr = &in[k]; - ysum64 -= in[k - 1] * in[k - 1] >> scaling; - ysum64 += (int32_t)(in[PITCH_CORR_LEN2 + k - 1]) - * in[PITCH_CORR_LEN2 + k - 1] >> scaling; - -#ifdef WEBRTC_HAS_NEON - { - int32_t vbuff[4]; - int32x4_t int_32x4_sum = vmovq_n_s32(0); - // Can't shift a Neon register to right with a non-constant shift value. - int32x4_t int_32x4_scale = vdupq_n_s32(-scaling); - // Assert a codition used in loop unrolling at compile-time. - RTC_COMPILE_ASSERT(PITCH_CORR_LEN2 %4 == 0); - - for (n = 0; n < PITCH_CORR_LEN2; n += 4) { - int16x4_t int_16x4_x = vld1_s16(&x[n]); - int16x4_t int_16x4_in = vld1_s16(&inptr[n]); - int32x4_t int_32x4 = vmull_s16(int_16x4_x, int_16x4_in); - int_32x4 = vshlq_s32(int_32x4, int_32x4_scale); - int_32x4_sum = vaddq_s32(int_32x4_sum, int_32x4); - } - - // Use vector store to avoid long stall from data trasferring - // from vector to general register. - vst1q_s32(vbuff, int_32x4_sum); - csum32 = vbuff[0] + vbuff[1]; - csum32 += vbuff[2]; - csum32 += vbuff[3]; - } -#else - int64_t csum64_tmp = 0; - if(scaling == 0) { - for (n = 0; n < PITCH_CORR_LEN2; n++) { - csum64_tmp += (int32_t)(x[n]) * inptr[n]; - } - } else { - for (n = 0; n < PITCH_CORR_LEN2; n++) { - csum64_tmp += ((int32_t)(x[n]) * inptr[n]) >> scaling; - } - } - csum32 = csum64_tmp; -#endif - - logcorQ8--; - - lys = WebRtcIsacfix_Log2Q8((uint32_t)ysum64) >> 1; // Q8, sqrt(ysum) - - if (csum32 > 0) { - lcs = WebRtcIsacfix_Log2Q8((uint32_t)csum32); // 2log(csum) in Q8 - if (lcs > (lys + oneQ8)) { // csum/sqrt(ysum) > 2 - *logcorQ8 = lcs - lys; // log2(csum/sqrt(ysum)) - } else { - *logcorQ8 = oneQ8; // 1.00 - } - } else { - *logcorQ8 = 0; - } - } -} diff --git a/modules/audio_coding/codecs/isac/fix/source/pitch_estimator_mips.c b/modules/audio_coding/codecs/isac/fix/source/pitch_estimator_mips.c deleted file mode 100644 index 4ead84c492..0000000000 --- a/modules/audio_coding/codecs/isac/fix/source/pitch_estimator_mips.c +++ /dev/null @@ -1,193 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#include "modules/audio_coding/codecs/isac/fix/source/pitch_estimator.h" -#include "common_audio/signal_processing/include/signal_processing_library.h" -#include "rtc_base/compile_assert_c.h" - -extern int32_t WebRtcIsacfix_Log2Q8(uint32_t x); - -void WebRtcIsacfix_PCorr2Q32(const int16_t* in, int32_t* logcorQ8) { - int16_t scaling,n,k; - int32_t ysum32,csum32, lys, lcs; - const int32_t oneQ8 = 1 << 8; // 1.00 in Q8 - const int16_t* x; - const int16_t* inptr; - - x = in + PITCH_MAX_LAG / 2 + 2; - scaling = WebRtcSpl_GetScalingSquare((int16_t*)in, - PITCH_CORR_LEN2, - PITCH_CORR_LEN2); - ysum32 = 1; - csum32 = 0; - x = in + PITCH_MAX_LAG / 2 + 2; - { - const int16_t* tmp_x = x; - const int16_t* tmp_in = in; - int32_t tmp1, tmp2, tmp3, tmp4, tmp5, tmp6, tmp7, tmp8; - n = PITCH_CORR_LEN2; - RTC_COMPILE_ASSERT(PITCH_CORR_LEN2 % 4 == 0); - __asm __volatile ( - ".set push \n\t" - ".set noreorder \n\t" - "1: \n\t" - "lh %[tmp1], 0(%[tmp_in]) \n\t" - "lh %[tmp2], 2(%[tmp_in]) \n\t" - "lh %[tmp3], 4(%[tmp_in]) \n\t" - "lh %[tmp4], 6(%[tmp_in]) \n\t" - "lh %[tmp5], 0(%[tmp_x]) \n\t" - "lh %[tmp6], 2(%[tmp_x]) \n\t" - "lh %[tmp7], 4(%[tmp_x]) \n\t" - "lh %[tmp8], 6(%[tmp_x]) \n\t" - "mul %[tmp5], %[tmp1], %[tmp5] \n\t" - "mul %[tmp1], %[tmp1], %[tmp1] \n\t" - "mul %[tmp6], %[tmp2], %[tmp6] \n\t" - "mul %[tmp2], %[tmp2], %[tmp2] \n\t" - "mul %[tmp7], %[tmp3], %[tmp7] \n\t" - "mul %[tmp3], %[tmp3], %[tmp3] \n\t" - "mul %[tmp8], %[tmp4], %[tmp8] \n\t" - "mul %[tmp4], %[tmp4], %[tmp4] \n\t" - "addiu %[n], %[n], -4 \n\t" - "srav %[tmp5], %[tmp5], %[scaling] \n\t" - "srav %[tmp1], %[tmp1], %[scaling] \n\t" - "srav %[tmp6], %[tmp6], %[scaling] \n\t" - "srav %[tmp2], %[tmp2], %[scaling] \n\t" - "srav %[tmp7], %[tmp7], %[scaling] \n\t" - "srav %[tmp3], %[tmp3], %[scaling] \n\t" - "srav %[tmp8], %[tmp8], %[scaling] \n\t" - "srav %[tmp4], %[tmp4], %[scaling] \n\t" - "addu %[ysum32], %[ysum32], %[tmp1] \n\t" - "addu %[csum32], %[csum32], %[tmp5] \n\t" - "addu %[ysum32], %[ysum32], %[tmp2] \n\t" - "addu %[csum32], %[csum32], %[tmp6] \n\t" - "addu %[ysum32], %[ysum32], %[tmp3] \n\t" - "addu %[csum32], %[csum32], %[tmp7] \n\t" - "addu %[ysum32], %[ysum32], %[tmp4] \n\t" - "addu %[csum32], %[csum32], %[tmp8] \n\t" - "addiu %[tmp_in], %[tmp_in], 8 \n\t" - "bgtz %[n], 1b \n\t" - " addiu %[tmp_x], %[tmp_x], 8 \n\t" - ".set pop \n\t" - : [tmp1] "=&r" (tmp1), [tmp2] "=&r" (tmp2), [tmp3] "=&r" (tmp3), - [tmp4] "=&r" (tmp4), [tmp5] "=&r" (tmp5), [tmp6] "=&r" (tmp6), - [tmp7] "=&r" (tmp7), [tmp8] "=&r" (tmp8), [tmp_in] "+r" (tmp_in), - [ysum32] "+r" (ysum32), [tmp_x] "+r" (tmp_x), [csum32] "+r" (csum32), - [n] "+r" (n) - : [scaling] "r" (scaling) - : "memory", "hi", "lo" - ); - } - logcorQ8 += PITCH_LAG_SPAN2 - 1; - lys = WebRtcIsacfix_Log2Q8((uint32_t)ysum32) >> 1; // Q8, sqrt(ysum) - if (csum32 > 0) { - lcs = WebRtcIsacfix_Log2Q8((uint32_t)csum32); // 2log(csum) in Q8 - if (lcs > (lys + oneQ8)) { // csum/sqrt(ysum) > 2 in Q8 - *logcorQ8 = lcs - lys; // log2(csum/sqrt(ysum)) - } else { - *logcorQ8 = oneQ8; // 1.00 - } - } else { - *logcorQ8 = 0; - } - - for (k = 1; k < PITCH_LAG_SPAN2; k++) { - inptr = &in[k]; - const int16_t* tmp_in1 = &in[k - 1]; - const int16_t* tmp_in2 = &in[PITCH_CORR_LEN2 + k - 1]; - const int16_t* tmp_x = x; - int32_t tmp1, tmp2, tmp3, tmp4, tmp5, tmp6, tmp7, tmp8; - n = PITCH_CORR_LEN2; - csum32 = 0; - __asm __volatile ( - ".set push \n\t" - ".set noreorder \n\t" - "lh %[tmp1], 0(%[tmp_in1]) \n\t" - "lh %[tmp2], 0(%[tmp_in2]) \n\t" - "mul %[tmp1], %[tmp1], %[tmp1] \n\t" - "mul %[tmp2], %[tmp2], %[tmp2] \n\t" - "srav %[tmp1], %[tmp1], %[scaling] \n\t" - "srav %[tmp2], %[tmp2], %[scaling] \n\t" - "subu %[ysum32], %[ysum32], %[tmp1] \n\t" - "bnez %[scaling], 2f \n\t" - " addu %[ysum32], %[ysum32], %[tmp2] \n\t" - "1: \n\t" - "lh %[tmp1], 0(%[inptr]) \n\t" - "lh %[tmp2], 0(%[tmp_x]) \n\t" - "lh %[tmp3], 2(%[inptr]) \n\t" - "lh %[tmp4], 2(%[tmp_x]) \n\t" - "lh %[tmp5], 4(%[inptr]) \n\t" - "lh %[tmp6], 4(%[tmp_x]) \n\t" - "lh %[tmp7], 6(%[inptr]) \n\t" - "lh %[tmp8], 6(%[tmp_x]) \n\t" - "mul %[tmp1], %[tmp1], %[tmp2] \n\t" - "mul %[tmp2], %[tmp3], %[tmp4] \n\t" - "mul %[tmp3], %[tmp5], %[tmp6] \n\t" - "mul %[tmp4], %[tmp7], %[tmp8] \n\t" - "addiu %[n], %[n], -4 \n\t" - "addiu %[inptr], %[inptr], 8 \n\t" - "addiu %[tmp_x], %[tmp_x], 8 \n\t" - "addu %[csum32], %[csum32], %[tmp1] \n\t" - "addu %[csum32], %[csum32], %[tmp2] \n\t" - "addu %[csum32], %[csum32], %[tmp3] \n\t" - "bgtz %[n], 1b \n\t" - " addu %[csum32], %[csum32], %[tmp4] \n\t" - "b 3f \n\t" - " nop \n\t" - "2: \n\t" - "lh %[tmp1], 0(%[inptr]) \n\t" - "lh %[tmp2], 0(%[tmp_x]) \n\t" - "lh %[tmp3], 2(%[inptr]) \n\t" - "lh %[tmp4], 2(%[tmp_x]) \n\t" - "lh %[tmp5], 4(%[inptr]) \n\t" - "lh %[tmp6], 4(%[tmp_x]) \n\t" - "lh %[tmp7], 6(%[inptr]) \n\t" - "lh %[tmp8], 6(%[tmp_x]) \n\t" - "mul %[tmp1], %[tmp1], %[tmp2] \n\t" - "mul %[tmp2], %[tmp3], %[tmp4] \n\t" - "mul %[tmp3], %[tmp5], %[tmp6] \n\t" - "mul %[tmp4], %[tmp7], %[tmp8] \n\t" - "addiu %[n], %[n], -4 \n\t" - "addiu %[inptr], %[inptr], 8 \n\t" - "addiu %[tmp_x], %[tmp_x], 8 \n\t" - "srav %[tmp1], %[tmp1], %[scaling] \n\t" - "srav %[tmp2], %[tmp2], %[scaling] \n\t" - "srav %[tmp3], %[tmp3], %[scaling] \n\t" - "srav %[tmp4], %[tmp4], %[scaling] \n\t" - "addu %[csum32], %[csum32], %[tmp1] \n\t" - "addu %[csum32], %[csum32], %[tmp2] \n\t" - "addu %[csum32], %[csum32], %[tmp3] \n\t" - "bgtz %[n], 2b \n\t" - " addu %[csum32], %[csum32], %[tmp4] \n\t" - "3: \n\t" - ".set pop \n\t" - : [tmp1] "=&r" (tmp1), [tmp2] "=&r" (tmp2), [tmp3] "=&r" (tmp3), - [tmp4] "=&r" (tmp4), [tmp5] "=&r" (tmp5), [tmp6] "=&r" (tmp6), - [tmp7] "=&r" (tmp7), [tmp8] "=&r" (tmp8), [inptr] "+r" (inptr), - [csum32] "+r" (csum32), [tmp_x] "+r" (tmp_x), [ysum32] "+r" (ysum32), - [n] "+r" (n) - : [tmp_in1] "r" (tmp_in1), [tmp_in2] "r" (tmp_in2), - [scaling] "r" (scaling) - : "memory", "hi", "lo" - ); - - logcorQ8--; - lys = WebRtcIsacfix_Log2Q8((uint32_t)ysum32) >> 1; // Q8, sqrt(ysum) - if (csum32 > 0) { - lcs = WebRtcIsacfix_Log2Q8((uint32_t)csum32); // 2log(csum) in Q8 - if (lcs > (lys + oneQ8)) { // csum/sqrt(ysum) > 2 - *logcorQ8 = lcs - lys; // log2(csum/sqrt(ysum)) - } else { - *logcorQ8 = oneQ8; // 1.00 - } - } else { - *logcorQ8 = 0; - } - } -} diff --git a/modules/audio_coding/codecs/isac/fix/source/pitch_filter.c b/modules/audio_coding/codecs/isac/fix/source/pitch_filter.c deleted file mode 100644 index 735533020e..0000000000 --- a/modules/audio_coding/codecs/isac/fix/source/pitch_filter.c +++ /dev/null @@ -1,248 +0,0 @@ -/* - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#include "modules/audio_coding/codecs/isac/fix/source/pitch_estimator.h" -#include "common_audio/signal_processing/include/signal_processing_library.h" -#include "modules/audio_coding/codecs/isac/fix/source/settings.h" -#include "modules/audio_coding/codecs/isac/fix/source/structs.h" -#include "rtc_base/compile_assert_c.h" - -// Number of segments in a pitch subframe. -static const int kSegments = 5; - -// A division factor of 1/5 in Q15. -static const int16_t kDivFactor = 6553; - -// Interpolation coefficients; generated by design_pitch_filter.m. -// Coefficients are stored in Q14. -static const int16_t kIntrpCoef[PITCH_FRACS][PITCH_FRACORDER] = { - {-367, 1090, -2706, 9945, 10596, -3318, 1626, -781, 287}, - {-325, 953, -2292, 7301, 12963, -3320, 1570, -743, 271}, - {-240, 693, -1622, 4634, 14809, -2782, 1262, -587, 212}, - {-125, 358, -817, 2144, 15982, -1668, 721, -329, 118}, - { 0, 0, -1, 1, 16380, 1, -1, 0, 0}, - { 118, -329, 721, -1668, 15982, 2144, -817, 358, -125}, - { 212, -587, 1262, -2782, 14809, 4634, -1622, 693, -240}, - { 271, -743, 1570, -3320, 12963, 7301, -2292, 953, -325} -}; - -static __inline size_t CalcLrIntQ(int16_t fixVal, - int16_t qDomain) { - int32_t roundVal = 1 << (qDomain - 1); - - return (fixVal + roundVal) >> qDomain; -} - -void WebRtcIsacfix_PitchFilter(int16_t* indatQQ, // Q10 if type is 1 or 4, - // Q0 if type is 2. - int16_t* outdatQQ, - PitchFiltstr* pfp, - int16_t* lagsQ7, - int16_t* gainsQ12, - int16_t type) { - int k, ind, cnt; - int16_t sign = 1; - int16_t inystateQQ[PITCH_DAMPORDER]; - int16_t ubufQQ[PITCH_INTBUFFSIZE + QLOOKAHEAD]; - const int16_t Gain = 21299; // 1.3 in Q14 - int16_t oldLagQ7; - int16_t oldGainQ12, lagdeltaQ7, curLagQ7, gaindeltaQ12, curGainQ12; - size_t frcQQ = 0; - int32_t indW32 = 0; - const int16_t* fracoeffQQ = NULL; - - // Assumptions in ARM assembly for WebRtcIsacfix_PitchFilterCoreARM(). - RTC_COMPILE_ASSERT(PITCH_FRACORDER == 9); - RTC_COMPILE_ASSERT(PITCH_DAMPORDER == 5); - - // Set up buffer and states. - memcpy(ubufQQ, pfp->ubufQQ, sizeof(pfp->ubufQQ)); - memcpy(inystateQQ, pfp->ystateQQ, sizeof(inystateQQ)); - - // Get old lag and gain value from memory. - oldLagQ7 = pfp->oldlagQ7; - oldGainQ12 = pfp->oldgainQ12; - - if (type == 4) { - sign = -1; - - // Make output more periodic. - for (k = 0; k < PITCH_SUBFRAMES; k++) { - gainsQ12[k] = (int16_t)(gainsQ12[k] * Gain >> 14); - } - } - - // No interpolation if pitch lag step is big. - if (((lagsQ7[0] * 3 >> 1) < oldLagQ7) || (lagsQ7[0] > (oldLagQ7 * 3 >> 1))) { - oldLagQ7 = lagsQ7[0]; - oldGainQ12 = gainsQ12[0]; - } - - ind = 0; - - for (k = 0; k < PITCH_SUBFRAMES; k++) { - // Calculate interpolation steps. - lagdeltaQ7 = lagsQ7[k] - oldLagQ7; - lagdeltaQ7 = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND( - lagdeltaQ7, kDivFactor, 15); - curLagQ7 = oldLagQ7; - gaindeltaQ12 = gainsQ12[k] - oldGainQ12; - gaindeltaQ12 = (int16_t)(gaindeltaQ12 * kDivFactor >> 15); - - curGainQ12 = oldGainQ12; - oldLagQ7 = lagsQ7[k]; - oldGainQ12 = gainsQ12[k]; - - // Each frame has 4 60-sample pitch subframes, and each subframe has 5 - // 12-sample segments. Each segment need to be processed with - // newly-updated parameters, so we break the pitch filtering into - // two for-loops (5 x 12) below. It's also why kDivFactor = 0.2 (in Q15). - for (cnt = 0; cnt < kSegments; cnt++) { - // Update parameters for each segment. - curGainQ12 += gaindeltaQ12; - curLagQ7 += lagdeltaQ7; - indW32 = CalcLrIntQ(curLagQ7, 7); - if (indW32 < PITCH_FRACORDER - 2) { - // WebRtcIsacfix_PitchFilterCore requires indW32 >= PITCH_FRACORDER - - // 2; otherwise, it will read from entries of ubufQQ that haven't been - // written yet. (This problem has only been seen in fuzzer tests, not - // in real life.) See Chromium bug 581901. - indW32 = PITCH_FRACORDER - 2; - } - frcQQ = ((indW32 << 7) + 64 - curLagQ7) >> 4; - - if (frcQQ >= PITCH_FRACS) { - frcQQ = 0; - } - fracoeffQQ = kIntrpCoef[frcQQ]; - - // Pitch filtering. - WebRtcIsacfix_PitchFilterCore(PITCH_SUBFRAME_LEN / kSegments, curGainQ12, - indW32, sign, inystateQQ, ubufQQ, fracoeffQQ, indatQQ, outdatQQ, &ind); - } - } - - // Export buffer and states. - memcpy(pfp->ubufQQ, ubufQQ + PITCH_FRAME_LEN, sizeof(pfp->ubufQQ)); - memcpy(pfp->ystateQQ, inystateQQ, sizeof(pfp->ystateQQ)); - - pfp->oldlagQ7 = oldLagQ7; - pfp->oldgainQ12 = oldGainQ12; - - if (type == 2) { - // Filter look-ahead segment. - WebRtcIsacfix_PitchFilterCore(QLOOKAHEAD, curGainQ12, indW32, 1, inystateQQ, - ubufQQ, fracoeffQQ, indatQQ, outdatQQ, &ind); - } -} - - -void WebRtcIsacfix_PitchFilterGains(const int16_t* indatQ0, - PitchFiltstr* pfp, - int16_t* lagsQ7, - int16_t* gainsQ12) { - int k, n, m; - size_t ind, pos, pos3QQ; - - int16_t ubufQQ[PITCH_INTBUFFSIZE]; - int16_t oldLagQ7, lagdeltaQ7, curLagQ7; - const int16_t* fracoeffQQ = NULL; - int16_t scale; - int16_t cnt = 0, tmpW16; - size_t frcQQ, indW16 = 0; - int32_t tmpW32, tmp2W32, csum1QQ, esumxQQ; - - // Set up buffer and states. - memcpy(ubufQQ, pfp->ubufQQ, sizeof(pfp->ubufQQ)); - oldLagQ7 = pfp->oldlagQ7; - - // No interpolation if pitch lag step is big. - if (((lagsQ7[0] * 3 >> 1) < oldLagQ7) || (lagsQ7[0] > (oldLagQ7 * 3 >> 1))) { - oldLagQ7 = lagsQ7[0]; - } - - ind = 0; - pos = ind + PITCH_BUFFSIZE; - scale = 0; - for (k = 0; k < PITCH_SUBFRAMES; k++) { - - // Calculate interpolation steps. - lagdeltaQ7 = lagsQ7[k] - oldLagQ7; - lagdeltaQ7 = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND( - lagdeltaQ7, kDivFactor, 15); - curLagQ7 = oldLagQ7; - oldLagQ7 = lagsQ7[k]; - - csum1QQ = 1; - esumxQQ = 1; - - // Same as function WebRtcIsacfix_PitchFilter(), we break the pitch - // filtering into two for-loops (5 x 12) below. - for (cnt = 0; cnt < kSegments; cnt++) { - // Update parameters for each segment. - curLagQ7 += lagdeltaQ7; - indW16 = CalcLrIntQ(curLagQ7, 7); - frcQQ = ((indW16 << 7) + 64 - curLagQ7) >> 4; - - if (frcQQ >= PITCH_FRACS) { - frcQQ = 0; - } - fracoeffQQ = kIntrpCoef[frcQQ]; - - pos3QQ = pos - (indW16 + 4); - - for (n = 0; n < PITCH_SUBFRAME_LEN / kSegments; n++) { - // Filter to get fractional pitch. - - tmpW32 = 0; - for (m = 0; m < PITCH_FRACORDER; m++) { - tmpW32 += ubufQQ[pos3QQ + m] * fracoeffQQ[m]; - } - - // Subtract from input and update buffer. - ubufQQ[pos] = indatQ0[ind]; - - tmp2W32 = WEBRTC_SPL_MUL_16_32_RSFT14(indatQ0[ind], tmpW32); - tmpW32 += 8192; - tmpW16 = tmpW32 >> 14; - tmpW32 = tmpW16 * tmpW16; - - if ((tmp2W32 > 1073700000) || (csum1QQ > 1073700000) || - (tmpW32 > 1073700000) || (esumxQQ > 1073700000)) { // 2^30 - scale++; - csum1QQ >>= 1; - esumxQQ >>= 1; - } - csum1QQ += tmp2W32 >> scale; - esumxQQ += tmpW32 >> scale; - - ind++; - pos++; - pos3QQ++; - } - } - - if (csum1QQ < esumxQQ) { - tmp2W32 = WebRtcSpl_DivResultInQ31(csum1QQ, esumxQQ); - - // Gain should be half the correlation. - tmpW32 = tmp2W32 >> 20; - } else { - tmpW32 = 4096; - } - gainsQ12[k] = (int16_t)WEBRTC_SPL_SAT(PITCH_MAX_GAIN_Q12, tmpW32, 0); - } - - // Export buffer and states. - memcpy(pfp->ubufQQ, ubufQQ + PITCH_FRAME_LEN, sizeof(pfp->ubufQQ)); - pfp->oldlagQ7 = lagsQ7[PITCH_SUBFRAMES - 1]; - pfp->oldgainQ12 = gainsQ12[PITCH_SUBFRAMES - 1]; - -} diff --git a/modules/audio_coding/codecs/isac/fix/source/pitch_filter_armv6.S b/modules/audio_coding/codecs/isac/fix/source/pitch_filter_armv6.S deleted file mode 100644 index 065946856f..0000000000 --- a/modules/audio_coding/codecs/isac/fix/source/pitch_filter_armv6.S +++ /dev/null @@ -1,143 +0,0 @@ -@ -@ Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. -@ -@ Use of this source code is governed by a BSD-style license -@ that can be found in the LICENSE file in the root of the source -@ tree. An additional intellectual property rights grant can be found -@ in the file PATENTS. All contributing project authors may -@ be found in the AUTHORS file in the root of the source tree. -@ - -@ Contains the core loop routine for the pitch filter function in iSAC, -@ optimized for ARMv7 platforms. -@ -@ Output is bit-exact with the reference C code in pitch_filter.c. - -#include "modules/audio_coding/codecs/isac/fix/source/settings.h" -#include "rtc_base/system/asm_defines.h" - -GLOBAL_FUNCTION WebRtcIsacfix_PitchFilterCore -.align 2 - -@ void WebRtcIsacfix_PitchFilterCore(int loopNumber, -@ int16_t gain, -@ size_t index, -@ int16_t sign, -@ int16_t* inputState, -@ int16_t* outputBuf2, -@ const int16_t* coefficient, -@ int16_t* inputBuf, -@ int16_t* outputBuf, -@ int* index2) { -DEFINE_FUNCTION WebRtcIsacfix_PitchFilterCore - push {r4-r11} - sub sp, #8 - - str r0, [sp] @ loopNumber - str r3, [sp, #4] @ sign - ldr r3, [sp, #44] @ outputBuf2 - ldr r6, [sp, #60] @ index2 - ldr r7, [r6] @ *index2 - ldr r8, [sp, #52] @ inputBuf - ldr r12, [sp, #56] @ outputBuf - - add r4, r7, r0 - str r4, [r6] @ Store return value to index2. - - mov r10, r7, asl #1 - add r12, r10 @ &outputBuf[*index2] - add r8, r10 @ &inputBuf[*index2] - - add r4, r7, #PITCH_BUFFSIZE @ *index2 + PITCH_BUFFSIZE - add r6, r3, r4, lsl #1 @ &outputBuf2[*index2 + PITCH_BUFFSIZE] - sub r4, r2 @ r2: index - sub r4, #2 @ *index2 + PITCH_BUFFSIZE - index - 2 - add r3, r4, lsl #1 @ &ubufQQpos2[*index2] - ldr r9, [sp, #48] @ coefficient - -LOOP: -@ Usage of registers in the loop: -@ r0: loop counter -@ r1: gain -@ r2: tmpW32 -@ r3: &ubufQQpos2[] -@ r6: &outputBuf2[] -@ r8: &inputBuf[] -@ r9: &coefficient[] -@ r12: &outputBuf[] -@ r4, r5, r7, r10, r11: scratch - - @ Filter to get fractional pitch. - @ The pitch filter loop here is unrolled with 9 multipications. - pld [r3] - ldr r10, [r3], #4 @ ubufQQpos2[*index2 + 0, *index2 + 1] - ldr r4, [r9], #4 @ coefficient[0, 1] - ldr r11, [r3], #4 - ldr r5, [r9], #4 - smuad r2, r10, r4 - smlad r2, r11, r5, r2 - - ldr r10, [r3], #4 - ldr r4, [r9], #4 - ldr r11, [r3], #4 - ldr r5, [r9], #4 - smlad r2, r10, r4, r2 - ldrh r10, [r3], #-14 @ r3 back to &ubufQQpos2[*index2]. - ldrh r4, [r9], #-16 @ r9 back to &coefficient[0]. - smlad r2, r11, r5, r2 - smlabb r2, r10, r4, r2 - - @ Saturate to avoid overflow in tmpW16. - asr r2, #1 - add r4, r2, #0x1000 - ssat r7, #16, r4, asr #13 - - @ Shift low pass filter state, and excute the low pass filter. - @ The memmove() and the low pass filter loop are unrolled and mixed. - smulbb r5, r1, r7 - add r7, r5, #0x800 - asr r7, #12 @ Get the value for inputState[0]. - ldr r11, [sp, #40] @ inputState - pld [r11] - adr r10, kDampFilter - ldrsh r4, [r10], #2 @ kDampFilter[0] - mul r2, r7, r4 - ldr r4, [r11] @ inputState[0, 1], before shift. - strh r7, [r11] @ inputState[0], after shift. - ldr r5, [r11, #4] @ inputState[2, 3], before shift. - ldr r7, [r10], #4 @ kDampFilter[1, 2] - ldr r10, [r10] @ kDampFilter[3, 4] - str r4, [r11, #2] @ inputState[1, 2], after shift. - str r5, [r11, #6] @ inputState[3, 4], after shift. - smlad r2, r4, r7, r2 - smlad r2, r5, r10, r2 - - @ Saturate to avoid overflow. - @ First shift the sample to the range of [0xC0000000, 0x3FFFFFFF], - @ to avoid overflow in the next saturation step. - asr r2, #1 - add r10, r2, #0x2000 - ssat r10, #16, r10, asr #14 - - @ Subtract from input and update buffer. - ldr r11, [sp, #4] @ sign - ldrsh r4, [r8] - ldrsh r7, [r8], #2 @ inputBuf[*index2] - smulbb r5, r11, r10 - subs r0, #1 - sub r4, r5 - ssat r2, #16, r4 - strh r2, [r12], #2 @ outputBuf[*index2] - - add r2, r7 - ssat r2, #16, r2 - strh r2, [r6], #2 @ outputBuff2[*index2 + PITCH_BUFFSIZE] - bgt LOOP - - add sp, #8 - pop {r4-r11} - bx lr - -.align 2 -kDampFilter: - .short -2294, 8192, 20972, 8192, -2294 diff --git a/modules/audio_coding/codecs/isac/fix/source/pitch_filter_c.c b/modules/audio_coding/codecs/isac/fix/source/pitch_filter_c.c deleted file mode 100644 index f23d19de9c..0000000000 --- a/modules/audio_coding/codecs/isac/fix/source/pitch_filter_c.c +++ /dev/null @@ -1,73 +0,0 @@ -/* - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#include "common_audio/signal_processing/include/signal_processing_library.h" -#include "modules/audio_coding/codecs/isac/fix/source/pitch_estimator.h" - -/* Filter coefficicients in Q15. */ -static const int16_t kDampFilter[PITCH_DAMPORDER] = { - -2294, 8192, 20972, 8192, -2294 -}; - -void WebRtcIsacfix_PitchFilterCore(int loopNumber, - int16_t gain, - size_t index, - int16_t sign, - int16_t* inputState, - int16_t* outputBuf2, - const int16_t* coefficient, - int16_t* inputBuf, - int16_t* outputBuf, - int* index2) { - int i = 0, j = 0; /* Loop counters. */ - int16_t* ubufQQpos2 = &outputBuf2[PITCH_BUFFSIZE - (index + 2)]; - int16_t tmpW16 = 0; - - for (i = 0; i < loopNumber; i++) { - int32_t tmpW32 = 0; - - /* Filter to get fractional pitch. */ - for (j = 0; j < PITCH_FRACORDER; j++) { - tmpW32 += ubufQQpos2[*index2 + j] * coefficient[j]; - } - - /* Saturate to avoid overflow in tmpW16. */ - tmpW32 = WEBRTC_SPL_SAT(536862719, tmpW32, -536879104); - tmpW32 += 8192; - tmpW16 = (int16_t)(tmpW32 >> 14); - - /* Shift low pass filter state. */ - memmove(&inputState[1], &inputState[0], - (PITCH_DAMPORDER - 1) * sizeof(int16_t)); - inputState[0] = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND( - gain, tmpW16, 12); - - /* Low pass filter. */ - tmpW32 = 0; - /* TODO(kma): Define a static inline function WebRtcSpl_DotProduct() - in spl_inl.h to replace this and other similar loops. */ - for (j = 0; j < PITCH_DAMPORDER; j++) { - tmpW32 += inputState[j] * kDampFilter[j]; - } - - /* Saturate to avoid overflow in tmpW16. */ - tmpW32 = WEBRTC_SPL_SAT(1073725439, tmpW32, -1073758208); - tmpW32 += 16384; - tmpW16 = (int16_t)(tmpW32 >> 15); - - /* Subtract from input and update buffer. */ - tmpW32 = inputBuf[*index2] - sign * tmpW16; - outputBuf[*index2] = WebRtcSpl_SatW32ToW16(tmpW32); - tmpW32 = inputBuf[*index2] + outputBuf[*index2]; - outputBuf2[*index2 + PITCH_BUFFSIZE] = WebRtcSpl_SatW32ToW16(tmpW32); - - (*index2)++; - } -} diff --git a/modules/audio_coding/codecs/isac/fix/source/pitch_filter_mips.c b/modules/audio_coding/codecs/isac/fix/source/pitch_filter_mips.c deleted file mode 100644 index 785fd9464f..0000000000 --- a/modules/audio_coding/codecs/isac/fix/source/pitch_filter_mips.c +++ /dev/null @@ -1,133 +0,0 @@ -/* - * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#include "modules/audio_coding/codecs/isac/fix/source/pitch_estimator.h" - -void WebRtcIsacfix_PitchFilterCore(int loopNumber, - int16_t gain, - size_t index, - int16_t sign, - int16_t* inputState, - int16_t* outputBuf2, - const int16_t* coefficient, - int16_t* inputBuf, - int16_t* outputBuf, - int* index2) { - int ind2t = *index2; - int i = 0; - int16_t* out2_pos2 = &outputBuf2[PITCH_BUFFSIZE - (index + 2)] + ind2t; - int32_t w1, w2, w3, w4, w5, gain32, sign32; - int32_t coef1, coef2, coef3, coef4, coef5 = 0; - // Define damp factors as int32_t (pair of int16_t) - int32_t kDampF0 = 0x0000F70A; - int32_t kDampF1 = 0x51EC2000; - int32_t kDampF2 = 0xF70A2000; - int16_t* input1 = inputBuf + ind2t; - int16_t* output1 = outputBuf + ind2t; - int16_t* output2 = outputBuf2 + ind2t + PITCH_BUFFSIZE; - - // Load coefficients outside the loop and sign-extend gain and sign - __asm __volatile ( - ".set push \n\t" - ".set noreorder \n\t" - "lwl %[coef1], 3(%[coefficient]) \n\t" - "lwl %[coef2], 7(%[coefficient]) \n\t" - "lwl %[coef3], 11(%[coefficient]) \n\t" - "lwl %[coef4], 15(%[coefficient]) \n\t" - "lwr %[coef1], 0(%[coefficient]) \n\t" - "lwr %[coef2], 4(%[coefficient]) \n\t" - "lwr %[coef3], 8(%[coefficient]) \n\t" - "lwr %[coef4], 12(%[coefficient]) \n\t" - "lhu %[coef5], 16(%[coefficient]) \n\t" - "seh %[gain32], %[gain] \n\t" - "seh %[sign32], %[sign] \n\t" - ".set pop \n\t" - : [coef1] "=&r" (coef1), [coef2] "=&r" (coef2), [coef3] "=&r" (coef3), - [coef4] "=&r" (coef4), [coef5] "=&r" (coef5), [gain32] "=&r" (gain32), - [sign32] "=&r" (sign32) - : [coefficient] "r" (coefficient), [gain] "r" (gain), - [sign] "r" (sign) - : "memory" - ); - - for (i = 0; i < loopNumber; i++) { - __asm __volatile ( - ".set push \n\t" - ".set noreorder \n\t" - // Filter to get fractional pitch - "li %[w1], 8192 \n\t" - "mtlo %[w1] \n\t" - "mthi $0 \n\t" - "lwl %[w1], 3(%[out2_pos2]) \n\t" - "lwl %[w2], 7(%[out2_pos2]) \n\t" - "lwl %[w3], 11(%[out2_pos2]) \n\t" - "lwl %[w4], 15(%[out2_pos2]) \n\t" - "lwr %[w1], 0(%[out2_pos2]) \n\t" - "lwr %[w2], 4(%[out2_pos2]) \n\t" - "lwr %[w3], 8(%[out2_pos2]) \n\t" - "lwr %[w4], 12(%[out2_pos2]) \n\t" - "lhu %[w5], 16(%[out2_pos2]) \n\t" - "dpa.w.ph $ac0, %[w1], %[coef1] \n\t" - "dpa.w.ph $ac0, %[w2], %[coef2] \n\t" - "dpa.w.ph $ac0, %[w3], %[coef3] \n\t" - "dpa.w.ph $ac0, %[w4], %[coef4] \n\t" - "dpa.w.ph $ac0, %[w5], %[coef5] \n\t" - "addiu %[out2_pos2], %[out2_pos2], 2 \n\t" - "mthi $0, $ac1 \n\t" - "lwl %[w2], 3(%[inputState]) \n\t" - "lwl %[w3], 7(%[inputState]) \n\t" - // Fractional pitch shift & saturation - "extr_s.h %[w1], $ac0, 14 \n\t" - "li %[w4], 16384 \n\t" - "lwr %[w2], 0(%[inputState]) \n\t" - "lwr %[w3], 4(%[inputState]) \n\t" - "mtlo %[w4], $ac1 \n\t" - // Shift low pass filter state - "swl %[w2], 5(%[inputState]) \n\t" - "swl %[w3], 9(%[inputState]) \n\t" - "mul %[w1], %[gain32], %[w1] \n\t" - "swr %[w2], 2(%[inputState]) \n\t" - "swr %[w3], 6(%[inputState]) \n\t" - // Low pass filter accumulation - "dpa.w.ph $ac1, %[kDampF1], %[w2] \n\t" - "dpa.w.ph $ac1, %[kDampF2], %[w3] \n\t" - "lh %[w4], 0(%[input1]) \n\t" - "addiu %[input1], %[input1], 2 \n\t" - "shra_r.w %[w1], %[w1], 12 \n\t" - "sh %[w1], 0(%[inputState]) \n\t" - "dpa.w.ph $ac1, %[kDampF0], %[w1] \n\t" - // Low pass filter shift & saturation - "extr_s.h %[w2], $ac1, 15 \n\t" - "mul %[w2], %[w2], %[sign32] \n\t" - // Buffer update - "subu %[w2], %[w4], %[w2] \n\t" - "shll_s.w %[w2], %[w2], 16 \n\t" - "sra %[w2], %[w2], 16 \n\t" - "sh %[w2], 0(%[output1]) \n\t" - "addu %[w2], %[w2], %[w4] \n\t" - "shll_s.w %[w2], %[w2], 16 \n\t" - "addiu %[output1], %[output1], 2 \n\t" - "sra %[w2], %[w2], 16 \n\t" - "sh %[w2], 0(%[output2]) \n\t" - "addiu %[output2], %[output2], 2 \n\t" - ".set pop \n\t" - : [w1] "=&r" (w1), [w2] "=&r" (w2), [w3] "=&r" (w3), [w4] "=&r" (w4), - [w5] "=&r" (w5), [input1] "+r" (input1), [out2_pos2] "+r" (out2_pos2), - [output1] "+r" (output1), [output2] "+r" (output2) - : [coefficient] "r" (coefficient), [inputState] "r" (inputState), - [gain32] "r" (gain32), [sign32] "r" (sign32), [kDampF0] "r" (kDampF0), - [kDampF1] "r" (kDampF1), [kDampF2] "r" (kDampF2), - [coef1] "r" (coef1), [coef2] "r" (coef2), [coef3] "r" (coef3), - [coef4] "r" (coef4), [coef5] "r" (coef5) - : "hi", "lo", "$ac1hi", "$ac1lo", "memory" - ); - } - (*index2) += loopNumber; -} diff --git a/modules/audio_coding/codecs/isac/fix/source/pitch_gain_tables.c b/modules/audio_coding/codecs/isac/fix/source/pitch_gain_tables.c deleted file mode 100644 index bfbab1950d..0000000000 --- a/modules/audio_coding/codecs/isac/fix/source/pitch_gain_tables.c +++ /dev/null @@ -1,149 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/* - * pitch_gain_tables.c - * - * This file contains tables for the pitch filter side-info in the entropy coder. - * - */ - -#include "modules/audio_coding/codecs/isac/fix/source/pitch_gain_tables.h" - - -/********************* Pitch Filter Gain Coefficient Tables ************************/ - -/* cdf for quantized pitch filter gains */ -const uint16_t WebRtcIsacfix_kPitchGainCdf[255] = { - 0, 2, 4, 6, 64, 901, 903, 905, 16954, 16956, - 16961, 17360, 17362, 17364, 17366, 17368, 17370, 17372, 17374, 17411, - 17514, 17516, 17583, 18790, 18796, 18802, 20760, 20777, 20782, 21722, - 21724, 21728, 21738, 21740, 21742, 21744, 21746, 21748, 22224, 22227, - 22230, 23214, 23229, 23239, 25086, 25108, 25120, 26088, 26094, 26098, - 26175, 26177, 26179, 26181, 26183, 26185, 26484, 26507, 26522, 27705, - 27731, 27750, 29767, 29799, 29817, 30866, 30883, 30885, 31025, 31029, - 31031, 31033, 31035, 31037, 31114, 31126, 31134, 32687, 32722, 32767, - 35718, 35742, 35757, 36943, 36952, 36954, 37115, 37128, 37130, 37132, - 37134, 37136, 37143, 37145, 37152, 38843, 38863, 38897, 47458, 47467, - 47474, 49040, 49061, 49063, 49145, 49157, 49159, 49161, 49163, 49165, - 49167, 49169, 49171, 49757, 49770, 49782, 61333, 61344, 61346, 62860, - 62883, 62885, 62887, 62889, 62891, 62893, 62895, 62897, 62899, 62901, - 62903, 62905, 62907, 62909, 65496, 65498, 65500, 65521, 65523, 65525, - 65527, 65529, 65531, 65533, 65535, 65535, 65535, 65535, 65535, 65535, - 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, - 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, - 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, - 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, - 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, - 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, - 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, - 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, - 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, - 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, - 65535, 65535, 65535, 65535, 65535 -}; - -/* index limits and ranges */ -const int16_t WebRtcIsacfix_kLowerlimiGain[3] = { - -7, -2, -1 -}; - -const int16_t WebRtcIsacfix_kUpperlimitGain[3] = { - 0, 3, 1 -}; - -const uint16_t WebRtcIsacfix_kMultsGain[2] = { - 18, 3 -}; - -/* size of cdf table */ -const uint16_t WebRtcIsacfix_kCdfTableSizeGain[1] = { - 256 -}; - -/* mean values of pitch filter gains in FIXED point Q12 */ -const int16_t WebRtcIsacfix_kPitchGain1[144] = { - 843, 1092, 1336, 1222, 1405, 1656, 1500, 1815, 1843, 1838, 1839, - 1843, 1843, 1843, 1843, 1843, 1843, 1843, 814, 846, 1092, 1013, - 1174, 1383, 1391, 1511, 1584, 1734, 1753, 1843, 1843, 1843, 1843, - 1843, 1843, 1843, 524, 689, 777, 845, 947, 1069, 1090, 1263, - 1380, 1447, 1559, 1676, 1645, 1749, 1843, 1843, 1843, 1843, 81, - 477, 563, 611, 706, 806, 849, 1012, 1192, 1128, 1330, 1489, - 1425, 1576, 1826, 1741, 1843, 1843, 0, 290, 305, 356, 488, - 575, 602, 741, 890, 835, 1079, 1196, 1182, 1376, 1519, 1506, - 1680, 1843, 0, 47, 97, 69, 289, 381, 385, 474, 617, - 664, 803, 1079, 935, 1160, 1269, 1265, 1506, 1741, 0, 0, - 0, 0, 112, 120, 190, 283, 442, 343, 526, 809, 684, - 935, 1134, 1020, 1265, 1506, 0, 0, 0, 0, 0, 0, - 0, 111, 256, 87, 373, 597, 430, 684, 935, 770, 1020, - 1265 -}; - -const int16_t WebRtcIsacfix_kPitchGain2[144] = { - 1760, 1525, 1285, 1747, 1671, 1393, 1843, 1826, 1555, 1843, 1784, - 1606, 1843, 1843, 1711, 1843, 1843, 1814, 1389, 1275, 1040, 1564, - 1414, 1252, 1610, 1495, 1343, 1753, 1592, 1405, 1804, 1720, 1475, - 1843, 1814, 1581, 1208, 1061, 856, 1349, 1148, 994, 1390, 1253, - 1111, 1495, 1343, 1178, 1770, 1465, 1234, 1814, 1581, 1342, 1040, - 793, 713, 1053, 895, 737, 1128, 1003, 861, 1277, 1094, 981, - 1475, 1192, 1019, 1581, 1342, 1098, 855, 570, 483, 833, 648, - 540, 948, 744, 572, 1009, 844, 636, 1234, 934, 685, 1342, - 1217, 984, 537, 318, 124, 603, 423, 350, 687, 479, 322, - 791, 581, 430, 987, 671, 488, 1098, 849, 597, 283, 27, - 0, 397, 222, 38, 513, 271, 124, 624, 325, 157, 737, - 484, 233, 849, 597, 343, 27, 0, 0, 141, 0, 0, - 256, 69, 0, 370, 87, 0, 484, 229, 0, 597, 343, - 87 -}; - -const int16_t WebRtcIsacfix_kPitchGain3[144] = { - 1843, 1843, 1711, 1843, 1818, 1606, 1843, 1827, 1511, 1814, 1639, - 1393, 1760, 1525, 1285, 1656, 1419, 1176, 1835, 1718, 1475, 1841, - 1650, 1387, 1648, 1498, 1287, 1600, 1411, 1176, 1522, 1299, 1040, - 1419, 1176, 928, 1773, 1461, 1128, 1532, 1355, 1202, 1429, 1260, - 1115, 1398, 1151, 1025, 1172, 1080, 790, 1176, 928, 677, 1475, - 1147, 1019, 1276, 1096, 922, 1214, 1010, 901, 1057, 893, 800, - 1040, 796, 734, 928, 677, 424, 1137, 897, 753, 1120, 830, - 710, 875, 751, 601, 795, 642, 583, 790, 544, 475, 677, - 474, 140, 987, 750, 482, 697, 573, 450, 691, 487, 303, - 661, 394, 332, 537, 303, 220, 424, 168, 0, 737, 484, - 229, 624, 348, 153, 441, 261, 136, 397, 166, 51, 283, - 27, 0, 168, 0, 0, 484, 229, 0, 370, 57, 0, - 256, 43, 0, 141, 0, 0, 27, 0, 0, 0, 0, - 0 -}; - - -const int16_t WebRtcIsacfix_kPitchGain4[144] = { - 1843, 1843, 1843, 1843, 1841, 1843, 1500, 1821, 1843, 1222, 1434, - 1656, 843, 1092, 1336, 504, 757, 1007, 1843, 1843, 1843, 1838, - 1791, 1843, 1265, 1505, 1599, 965, 1219, 1425, 730, 821, 1092, - 249, 504, 757, 1783, 1819, 1843, 1351, 1567, 1727, 1096, 1268, - 1409, 805, 961, 1131, 444, 670, 843, 0, 249, 504, 1425, - 1655, 1743, 1096, 1324, 1448, 822, 1019, 1199, 490, 704, 867, - 81, 450, 555, 0, 0, 249, 1247, 1428, 1530, 881, 1073, - 1283, 610, 759, 939, 278, 464, 645, 0, 200, 270, 0, - 0, 0, 935, 1163, 1410, 528, 790, 1068, 377, 499, 717, - 173, 240, 274, 0, 43, 62, 0, 0, 0, 684, 935, - 1182, 343, 551, 735, 161, 262, 423, 0, 55, 27, 0, - 0, 0, 0, 0, 0, 430, 684, 935, 87, 377, 597, - 0, 46, 256, 0, 0, 0, 0, 0, 0, 0, 0, - 0 -}; - - - -/* transform matrix in Q12*/ -const int16_t WebRtcIsacfix_kTransform[4][4] = { - { -2048, -2048, -2048, -2048 }, - { 2748, 916, -916, -2748 }, - { 2048, -2048, -2048, 2048 }, - { 916, -2748, 2748, -916 } -}; diff --git a/modules/audio_coding/codecs/isac/fix/source/pitch_gain_tables.h b/modules/audio_coding/codecs/isac/fix/source/pitch_gain_tables.h deleted file mode 100644 index 59e1738bce..0000000000 --- a/modules/audio_coding/codecs/isac/fix/source/pitch_gain_tables.h +++ /dev/null @@ -1,46 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/* - * pitch_gain_tables.h - * - * This file contains tables for the pitch filter side-info in the entropy - * coder. - * - */ - -#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_PITCH_GAIN_TABLES_H_ -#define MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_PITCH_GAIN_TABLES_H_ - -#include - -/********************* Pitch Filter Gain Coefficient Tables - * ************************/ -/* cdf for quantized pitch filter gains */ -extern const uint16_t WebRtcIsacfix_kPitchGainCdf[255]; - -/* index limits and ranges */ -extern const int16_t WebRtcIsacfix_kLowerlimiGain[3]; -extern const int16_t WebRtcIsacfix_kUpperlimitGain[3]; -extern const uint16_t WebRtcIsacfix_kMultsGain[2]; - -/* mean values of pitch filter gains in Q12*/ -extern const int16_t WebRtcIsacfix_kPitchGain1[144]; -extern const int16_t WebRtcIsacfix_kPitchGain2[144]; -extern const int16_t WebRtcIsacfix_kPitchGain3[144]; -extern const int16_t WebRtcIsacfix_kPitchGain4[144]; - -/* size of cdf table */ -extern const uint16_t WebRtcIsacfix_kCdfTableSizeGain[1]; - -/* transform matrix */ -extern const int16_t WebRtcIsacfix_kTransform[4][4]; - -#endif /* MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_PITCH_GAIN_TABLES_H_ */ diff --git a/modules/audio_coding/codecs/isac/fix/source/pitch_lag_tables.c b/modules/audio_coding/codecs/isac/fix/source/pitch_lag_tables.c deleted file mode 100644 index 894716e739..0000000000 --- a/modules/audio_coding/codecs/isac/fix/source/pitch_lag_tables.c +++ /dev/null @@ -1,306 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/* - * pitch_lag_tables.c - * - * This file contains tables for the pitch filter side-info in the entropy coder. - * - */ - -#include "modules/audio_coding/codecs/isac/fix/source/settings.h" -#include "modules/audio_coding/codecs/isac/fix/source/pitch_lag_tables.h" - - -/********************* Pitch Filter Gain Coefficient Tables ************************/ - -/* tables for use with small pitch gain */ - -/* cdf for quantized pitch filter lags */ -const uint16_t WebRtcIsacfix_kPitchLagCdf1Lo[127] = { - 0, 134, 336, 549, 778, 998, 1264, 1512, 1777, 2070, - 2423, 2794, 3051, 3361, 3708, 3979, 4315, 4610, 4933, 5269, - 5575, 5896, 6155, 6480, 6816, 7129, 7477, 7764, 8061, 8358, - 8718, 9020, 9390, 9783, 10177, 10543, 10885, 11342, 11795, 12213, - 12680, 13096, 13524, 13919, 14436, 14903, 15349, 15795, 16267, 16734, - 17266, 17697, 18130, 18632, 19080, 19447, 19884, 20315, 20735, 21288, - 21764, 22264, 22723, 23193, 23680, 24111, 24557, 25022, 25537, 26082, - 26543, 27090, 27620, 28139, 28652, 29149, 29634, 30175, 30692, 31273, - 31866, 32506, 33059, 33650, 34296, 34955, 35629, 36295, 36967, 37726, - 38559, 39458, 40364, 41293, 42256, 43215, 44231, 45253, 46274, 47359, - 48482, 49678, 50810, 51853, 53016, 54148, 55235, 56263, 57282, 58363, - 59288, 60179, 61076, 61806, 62474, 63129, 63656, 64160, 64533, 64856, - 65152, 65535, 65535, 65535, 65535, 65535, 65535 -}; - -const uint16_t WebRtcIsacfix_kPitchLagCdf2Lo[20] = { - 0, 429, 3558, 5861, 8558, 11639, 15210, 19502, 24773, 31983, - 42602, 48567, 52601, 55676, 58160, 60172, 61889, 63235, 65383, 65535 -}; - -const uint16_t WebRtcIsacfix_kPitchLagCdf3Lo[2] = { - 0, 65535 -}; - -const uint16_t WebRtcIsacfix_kPitchLagCdf4Lo[10] = { - 0, 2966, 6368, 11182, 19431, 37793, 48532, 55353, 60626, 65535 -}; - -const uint16_t *WebRtcIsacfix_kPitchLagPtrLo[4] = { - WebRtcIsacfix_kPitchLagCdf1Lo, - WebRtcIsacfix_kPitchLagCdf2Lo, - WebRtcIsacfix_kPitchLagCdf3Lo, - WebRtcIsacfix_kPitchLagCdf4Lo -}; - -/* size of first cdf table */ -const uint16_t WebRtcIsacfix_kPitchLagSizeLo[1] = { - 128 -}; - -/* index limits and ranges */ -const int16_t WebRtcIsacfix_kLowerLimitLo[4] = { - -140, -9, 0, -4 -}; - -const int16_t WebRtcIsacfix_kUpperLimitLo[4] = { - -20, 9, 0, 4 -}; - -/* initial index for arithmetic decoder */ -const uint16_t WebRtcIsacfix_kInitIndLo[3] = { - 10, 1, 5 -}; - -/* mean values of pitch filter lags in Q10 */ - -const int16_t WebRtcIsacfix_kMeanLag2Lo[19] = { - -17627, -16207, -14409, -12319, -10253, -8200, -6054, -3986, -1948, -19, - 1937, 3974, 6064, 8155, 10229, 12270, 14296, 16127, 17520 -}; - -const int16_t WebRtcIsacfix_kMeanLag4Lo[9] = { - -7949, -6063, -4036, -1941, 38, 1977, 4060, 6059 -}; - - - -/* tables for use with medium pitch gain */ - -/* cdf for quantized pitch filter lags */ -const uint16_t WebRtcIsacfix_kPitchLagCdf1Mid[255] = { - 0, 28, 61, 88, 121, 149, 233, 331, 475, 559, - 624, 661, 689, 712, 745, 791, 815, 843, 866, 922, - 959, 1024, 1061, 1117, 1178, 1238, 1280, 1350, 1453, 1513, - 1564, 1625, 1671, 1741, 1788, 1904, 2072, 2421, 2626, 2770, - 2840, 2900, 2942, 3012, 3068, 3115, 3147, 3194, 3254, 3319, - 3366, 3520, 3678, 3780, 3850, 3911, 3957, 4032, 4106, 4185, - 4292, 4474, 4683, 4842, 5019, 5191, 5321, 5428, 5540, 5675, - 5763, 5847, 5959, 6127, 6304, 6564, 6839, 7090, 7263, 7421, - 7556, 7728, 7872, 7984, 8142, 8361, 8580, 8743, 8938, 9227, - 9409, 9539, 9674, 9795, 9930, 10060, 10177, 10382, 10614, 10861, - 11038, 11271, 11415, 11629, 11792, 12044, 12193, 12416, 12574, 12821, - 13007, 13235, 13445, 13654, 13901, 14134, 14488, 15000, 15703, 16285, - 16504, 16797, 17086, 17328, 17579, 17807, 17998, 18268, 18538, 18836, - 19087, 19274, 19474, 19716, 19935, 20270, 20833, 21303, 21532, 21741, - 21978, 22207, 22523, 22770, 23054, 23613, 23943, 24204, 24399, 24651, - 24832, 25074, 25270, 25549, 25759, 26015, 26150, 26424, 26713, 27048, - 27342, 27504, 27681, 27854, 28021, 28207, 28412, 28664, 28859, 29064, - 29278, 29548, 29748, 30107, 30377, 30656, 30856, 31164, 31452, 31755, - 32011, 32328, 32626, 32919, 33319, 33789, 34329, 34925, 35396, 35973, - 36443, 36964, 37551, 38156, 38724, 39357, 40023, 40908, 41587, 42602, - 43924, 45037, 45810, 46597, 47421, 48291, 49092, 50051, 51448, 52719, - 53440, 54241, 54944, 55977, 56676, 57299, 57872, 58389, 59059, 59688, - 60237, 60782, 61094, 61573, 61890, 62290, 62658, 63030, 63217, 63454, - 63622, 63882, 64003, 64273, 64427, 64529, 64581, 64697, 64758, 64902, - 65414, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, - 65535, 65535, 65535, 65535, 65535 -}; - -const uint16_t WebRtcIsacfix_kPitchLagCdf2Mid[36] = { - 0, 71, 335, 581, 836, 1039, 1323, 1795, 2258, 2608, - 3005, 3591, 4243, 5344, 7163, 10583, 16848, 28078, 49448, 57007, - 60357, 61850, 62837, 63437, 63872, 64188, 64377, 64614, 64774, 64949, - 65039, 65115, 65223, 65360, 65474, 65535 -}; - -const uint16_t WebRtcIsacfix_kPitchLagCdf3Mid[2] = { - 0, 65535 -}; - -const uint16_t WebRtcIsacfix_kPitchLagCdf4Mid[20] = { - 0, 28, 246, 459, 667, 1045, 1523, 2337, 4337, 11347, - 44231, 56709, 60781, 62243, 63161, 63969, 64608, 65062, 65502, 65535 -}; - -const uint16_t *WebRtcIsacfix_kPitchLagPtrMid[4] = { - WebRtcIsacfix_kPitchLagCdf1Mid, - WebRtcIsacfix_kPitchLagCdf2Mid, - WebRtcIsacfix_kPitchLagCdf3Mid, - WebRtcIsacfix_kPitchLagCdf4Mid -}; - -/* size of first cdf table */ -const uint16_t WebRtcIsacfix_kPitchLagSizeMid[1] = { - 256 -}; - -/* index limits and ranges */ -const int16_t WebRtcIsacfix_kLowerLimitMid[4] = { - -280, -17, 0, -9 -}; - -const int16_t WebRtcIsacfix_kUpperLimitMid[4] = { - -40, 17, 0, 9 -}; - -/* initial index for arithmetic decoder */ -const uint16_t WebRtcIsacfix_kInitIndMid[3] = { - 18, 1, 10 -}; - -/* mean values of pitch filter lags in Q10 */ - -const int16_t WebRtcIsacfix_kMeanLag2Mid[35] = { - -17297, -16250, -15416, -14343, -13341, -12363, -11270, - -10355, -9122, -8217, -7172, -6083, -5102, -4004, -3060, - -1982, -952, -18, 935, 1976, 3040, 4032, - 5082, 6065, 7257, 8202, 9264, 10225, 11242, - 12234, 13337, 14336, 15374, 16187, 17347 -}; - - -const int16_t WebRtcIsacfix_kMeanLag4Mid[19] = { - -8811, -8081, -7203, -6003, -5057, -4025, -2983, -1964, - -891, 29, 921, 1920, 2988, 4064, 5187, 6079, 7173, 8074, 8849 -}; - - -/* tables for use with large pitch gain */ - -/* cdf for quantized pitch filter lags */ -const uint16_t WebRtcIsacfix_kPitchLagCdf1Hi[511] = { - 0, 7, 18, 33, 69, 105, 156, 228, 315, 612, - 680, 691, 709, 724, 735, 738, 742, 746, 749, 753, - 756, 760, 764, 774, 782, 785, 789, 796, 800, 803, - 807, 814, 818, 822, 829, 832, 847, 854, 858, 869, - 876, 883, 898, 908, 934, 977, 1010, 1050, 1060, 1064, - 1075, 1078, 1086, 1089, 1093, 1104, 1111, 1122, 1133, 1136, - 1151, 1162, 1183, 1209, 1252, 1281, 1339, 1364, 1386, 1401, - 1411, 1415, 1426, 1430, 1433, 1440, 1448, 1455, 1462, 1477, - 1487, 1495, 1502, 1506, 1509, 1516, 1524, 1531, 1535, 1542, - 1553, 1556, 1578, 1589, 1611, 1625, 1639, 1643, 1654, 1665, - 1672, 1687, 1694, 1705, 1708, 1719, 1730, 1744, 1752, 1759, - 1791, 1795, 1820, 1867, 1886, 1915, 1936, 1943, 1965, 1987, - 2041, 2099, 2161, 2175, 2200, 2211, 2226, 2233, 2244, 2251, - 2266, 2280, 2287, 2298, 2309, 2316, 2331, 2342, 2356, 2378, - 2403, 2418, 2447, 2497, 2544, 2602, 2863, 2895, 2903, 2935, - 2950, 2971, 3004, 3011, 3018, 3029, 3040, 3062, 3087, 3127, - 3152, 3170, 3199, 3243, 3293, 3322, 3340, 3377, 3402, 3427, - 3474, 3518, 3543, 3579, 3601, 3637, 3659, 3706, 3731, 3760, - 3818, 3847, 3869, 3901, 3920, 3952, 4068, 4169, 4220, 4271, - 4524, 4571, 4604, 4632, 4672, 4730, 4777, 4806, 4857, 4904, - 4951, 5002, 5031, 5060, 5107, 5150, 5212, 5266, 5331, 5382, - 5432, 5490, 5544, 5610, 5700, 5762, 5812, 5874, 5972, 6022, - 6091, 6163, 6232, 6305, 6402, 6540, 6685, 6880, 7090, 7271, - 7379, 7452, 7542, 7625, 7687, 7770, 7843, 7911, 7966, 8024, - 8096, 8190, 8252, 8320, 8411, 8501, 8585, 8639, 8751, 8842, - 8918, 8986, 9066, 9127, 9203, 9269, 9345, 9406, 9464, 9536, - 9612, 9667, 9735, 9844, 9931, 10036, 10119, 10199, 10260, 10358, - 10441, 10514, 10666, 10734, 10872, 10951, 11053, 11125, 11223, 11324, - 11516, 11664, 11737, 11816, 11892, 12008, 12120, 12200, 12280, 12392, - 12490, 12576, 12685, 12812, 12917, 13003, 13108, 13210, 13300, 13384, - 13470, 13579, 13673, 13771, 13879, 13999, 14136, 14201, 14368, 14614, - 14759, 14867, 14958, 15030, 15121, 15189, 15280, 15385, 15461, 15555, - 15653, 15768, 15884, 15971, 16069, 16145, 16210, 16279, 16380, 16463, - 16539, 16615, 16688, 16818, 16919, 17017, 18041, 18338, 18523, 18649, - 18790, 18917, 19047, 19167, 19315, 19460, 19601, 19731, 19858, 20068, - 20173, 20318, 20466, 20625, 20741, 20911, 21045, 21201, 21396, 21588, - 21816, 22022, 22305, 22547, 22786, 23072, 23322, 23600, 23879, 24168, - 24433, 24769, 25120, 25511, 25895, 26289, 26792, 27219, 27683, 28077, - 28566, 29094, 29546, 29977, 30491, 30991, 31573, 32105, 32594, 33173, - 33788, 34497, 35181, 35833, 36488, 37255, 37921, 38645, 39275, 39894, - 40505, 41167, 41790, 42431, 43096, 43723, 44385, 45134, 45858, 46607, - 47349, 48091, 48768, 49405, 49955, 50555, 51167, 51985, 52611, 53078, - 53494, 53965, 54435, 54996, 55601, 56125, 56563, 56838, 57244, 57566, - 57967, 58297, 58771, 59093, 59419, 59647, 59886, 60143, 60461, 60693, - 60917, 61170, 61416, 61634, 61891, 62122, 62310, 62455, 62632, 62839, - 63103, 63436, 63639, 63805, 63906, 64015, 64192, 64355, 64475, 64558, - 64663, 64742, 64811, 64865, 64916, 64956, 64981, 65025, 65068, 65115, - 65195, 65314, 65419, 65535, 65535, 65535, 65535, 65535, 65535, 65535, - 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, - 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, - 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, - 65535 -}; - -const uint16_t WebRtcIsacfix_kPitchLagCdf2Hi[68] = { - 0, 7, 11, 22, 37, 52, 56, 59, 81, 85, - 89, 96, 115, 130, 137, 152, 170, 181, 193, 200, - 207, 233, 237, 259, 289, 318, 363, 433, 592, 992, - 1607, 3062, 6149, 12206, 25522, 48368, 58223, 61918, 63640, 64584, - 64943, 65098, 65206, 65268, 65294, 65335, 65350, 65372, 65387, 65402, - 65413, 65420, 65428, 65435, 65439, 65450, 65454, 65468, 65472, 65476, - 65483, 65491, 65498, 65505, 65516, 65520, 65528, 65535 -}; - -const uint16_t WebRtcIsacfix_kPitchLagCdf3Hi[2] = { - 0, 65535 -}; - -const uint16_t WebRtcIsacfix_kPitchLagCdf4Hi[35] = { - 0, 7, 19, 30, 41, 48, 63, 74, 82, 96, - 122, 152, 215, 330, 701, 2611, 10931, 48106, 61177, 64341, - 65112, 65238, 65309, 65338, 65364, 65379, 65401, 65427, 65453, - 65465, 65476, 65490, 65509, 65528, 65535 -}; - -const uint16_t *WebRtcIsacfix_kPitchLagPtrHi[4] = { - WebRtcIsacfix_kPitchLagCdf1Hi, - WebRtcIsacfix_kPitchLagCdf2Hi, - WebRtcIsacfix_kPitchLagCdf3Hi, - WebRtcIsacfix_kPitchLagCdf4Hi -}; - -/* size of first cdf table */ -const uint16_t WebRtcIsacfix_kPitchLagSizeHi[1] = { - 512 -}; - -/* index limits and ranges */ -const int16_t WebRtcIsacfix_kLowerLimitHi[4] = { - -552, -34, 0, -16 -}; - -const int16_t WebRtcIsacfix_kUpperLimitHi[4] = { - -80, 32, 0, 17 -}; - -/* initial index for arithmetic decoder */ -const uint16_t WebRtcIsacfix_kInitIndHi[3] = { - 34, 1, 18 -}; - -/* mean values of pitch filter lags */ - -const int16_t WebRtcIsacfix_kMeanLag2Hi[67] = { - -17482, -16896, -16220, -15929, -15329, -14848, -14336, -13807, -13312, -12800, -12218, -11720, - -11307, -10649, -10396, -9742, -9148, -8668, -8297, -7718, -7155, -6656, -6231, -5600, -5129, - -4610, -4110, -3521, -3040, -2525, -2016, -1506, -995, -477, -5, 469, 991, 1510, 2025, 2526, 3079, - 3555, 4124, 4601, 5131, 5613, 6194, 6671, 7140, 7645, 8207, 8601, 9132, 9728, 10359, 10752, 11302, - 11776, 12288, 12687, 13204, 13759, 14295, 14810, 15360, 15764, 16350 -}; - - -const int16_t WebRtcIsacfix_kMeanLag4Hi[34] = { - -8175, -7659, -7205, -6684, -6215, -5651, -5180, -4566, -4087, -3536, -3096, - -2532, -1990, -1482, -959, -440, 11, 451, 954, 1492, 2020, 2562, 3059, - 3577, 4113, 4618, 5134, 5724, 6060, 6758, 7015, 7716, 8066, 8741 -}; diff --git a/modules/audio_coding/codecs/isac/fix/source/pitch_lag_tables.h b/modules/audio_coding/codecs/isac/fix/source/pitch_lag_tables.h deleted file mode 100644 index 228da26731..0000000000 --- a/modules/audio_coding/codecs/isac/fix/source/pitch_lag_tables.h +++ /dev/null @@ -1,99 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/* - * pitch_lag_tables.h - * - * This file contains tables for the pitch filter side-info in the entropy - * coder. - * - */ - -#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_PITCH_LAG_TABLES_H_ -#define MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_PITCH_LAG_TABLES_H_ - -#include - -/********************* Pitch Filter Lag Coefficient Tables - * ************************/ - -/* tables for use with small pitch gain */ - -/* cdfs for quantized pitch lags */ -extern const uint16_t WebRtcIsacfix_kPitchLagCdf1Lo[127]; -extern const uint16_t WebRtcIsacfix_kPitchLagCdf2Lo[20]; -extern const uint16_t WebRtcIsacfix_kPitchLagCdf3Lo[2]; -extern const uint16_t WebRtcIsacfix_kPitchLagCdf4Lo[10]; - -extern const uint16_t* WebRtcIsacfix_kPitchLagPtrLo[4]; - -/* size of first cdf table */ -extern const uint16_t WebRtcIsacfix_kPitchLagSizeLo[1]; - -/* index limits and ranges */ -extern const int16_t WebRtcIsacfix_kLowerLimitLo[4]; -extern const int16_t WebRtcIsacfix_kUpperLimitLo[4]; - -/* initial index for arithmetic decoder */ -extern const uint16_t WebRtcIsacfix_kInitIndLo[3]; - -/* mean values of pitch filter lags */ -extern const int16_t WebRtcIsacfix_kMeanLag2Lo[19]; -extern const int16_t WebRtcIsacfix_kMeanLag4Lo[9]; - -/* tables for use with medium pitch gain */ - -/* cdfs for quantized pitch lags */ -extern const uint16_t WebRtcIsacfix_kPitchLagCdf1Mid[255]; -extern const uint16_t WebRtcIsacfix_kPitchLagCdf2Mid[36]; -extern const uint16_t WebRtcIsacfix_kPitchLagCdf3Mid[2]; -extern const uint16_t WebRtcIsacfix_kPitchLagCdf4Mid[20]; - -extern const uint16_t* WebRtcIsacfix_kPitchLagPtrMid[4]; - -/* size of first cdf table */ -extern const uint16_t WebRtcIsacfix_kPitchLagSizeMid[1]; - -/* index limits and ranges */ -extern const int16_t WebRtcIsacfix_kLowerLimitMid[4]; -extern const int16_t WebRtcIsacfix_kUpperLimitMid[4]; - -/* initial index for arithmetic decoder */ -extern const uint16_t WebRtcIsacfix_kInitIndMid[3]; - -/* mean values of pitch filter lags */ -extern const int16_t WebRtcIsacfix_kMeanLag2Mid[35]; -extern const int16_t WebRtcIsacfix_kMeanLag4Mid[19]; - -/* tables for use with large pitch gain */ - -/* cdfs for quantized pitch lags */ -extern const uint16_t WebRtcIsacfix_kPitchLagCdf1Hi[511]; -extern const uint16_t WebRtcIsacfix_kPitchLagCdf2Hi[68]; -extern const uint16_t WebRtcIsacfix_kPitchLagCdf3Hi[2]; -extern const uint16_t WebRtcIsacfix_kPitchLagCdf4Hi[35]; - -extern const uint16_t* WebRtcIsacfix_kPitchLagPtrHi[4]; - -/* size of first cdf table */ -extern const uint16_t WebRtcIsacfix_kPitchLagSizeHi[1]; - -/* index limits and ranges */ -extern const int16_t WebRtcIsacfix_kLowerLimitHi[4]; -extern const int16_t WebRtcIsacfix_kUpperLimitHi[4]; - -/* initial index for arithmetic decoder */ -extern const uint16_t WebRtcIsacfix_kInitIndHi[3]; - -/* mean values of pitch filter lags */ -extern const int16_t WebRtcIsacfix_kMeanLag2Hi[67]; -extern const int16_t WebRtcIsacfix_kMeanLag4Hi[34]; - -#endif /* MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_PITCH_LAG_TABLES_H_ */ diff --git a/modules/audio_coding/codecs/isac/fix/source/settings.h b/modules/audio_coding/codecs/isac/fix/source/settings.h deleted file mode 100644 index 03a2d05457..0000000000 --- a/modules/audio_coding/codecs/isac/fix/source/settings.h +++ /dev/null @@ -1,211 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/* - * settings.h - * - * Declaration of #defines used in the iSAC codec - * - */ - -#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_SETTINGS_H_ -#define MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_SETTINGS_H_ - -/* sampling frequency (Hz) */ -#define FS 16000 -/* 1.5 times Sampling frequency */ -#define FS_1_HALF (uint32_t)24000 -/* Three times Sampling frequency */ -#define FS3 (uint32_t)48000 -/* Eight times Sampling frequency */ -#define FS8 (uint32_t)128000 - -/* number of samples per frame (either 480 (30ms) or 960 (60ms)) */ -#define INITIAL_FRAMESAMPLES 960 - -/* miliseconds */ -#define FRAMESIZE 30 -/* number of samples per frame processed in the encoder (30ms) */ -#define FRAMESAMPLES 480 /* ((FRAMESIZE*FS)/1000) */ -#define FRAMESAMPLES_HALF 240 -/* max number of samples per frame (= 60 ms frame) */ -#define MAX_FRAMESAMPLES 960 -/* number of samples per 10ms frame */ -#define FRAMESAMPLES_10ms 160 /* ((10*FS)/1000) */ -/* Number of samples per 1 ms */ -#define SAMPLES_PER_MSEC 16 -/* number of subframes */ -#define SUBFRAMES 6 -/* length of a subframe */ -#define UPDATE 80 -/* length of half a subframe (low/high band) */ -#define HALF_SUBFRAMELEN 40 /* (UPDATE/2) */ -/* samples of look ahead (in a half-band, so actually half the samples of look - * ahead @ FS) */ -#define QLOOKAHEAD 24 /* 3 ms */ - -/* order of AR model in spectral entropy coder */ -#define AR_ORDER 6 -#define MAX_ORDER 13 -#define LEVINSON_MAX_ORDER 12 - -/* window length (masking analysis) */ -#define WINLEN 256 -/* order of low-band pole filter used to approximate masking curve */ -#define ORDERLO 12 -/* order of hi-band pole filter used to approximate masking curve */ -#define ORDERHI 6 - -#define KLT_NUM_AVG_GAIN 0 -#define KLT_NUM_AVG_SHAPE 0 -#define KLT_NUM_MODELS 3 -#define LPC_SHAPE_ORDER 18 /* (ORDERLO + ORDERHI) */ - -#define KLT_ORDER_GAIN 12 /* (2 * SUBFRAMES) */ -#define KLT_ORDER_SHAPE 108 /* (LPC_SHAPE_ORDER * SUBFRAMES) */ - -/* order for post_filter_bank */ -#define POSTQORDER 3 -/* order for pre-filterbank */ -#define QORDER 3 -/* for decimator */ -#define ALLPASSSECTIONS 2 -/* The number of composite all-pass filter factors */ -#define NUMBEROFCOMPOSITEAPSECTIONS 4 - -/* The number of all-pass filter factors in an upper or lower channel*/ -#define NUMBEROFCHANNELAPSECTIONS 2 - -#define DPMIN_Q10 -10240 /* -10.00 in Q10 */ -#define DPMAX_Q10 10240 /* 10.00 in Q10 */ -#define MINBITS_Q10 10240 /* 10.0 in Q10 */ - -/* array size for byte stream in number of Word16. */ -#define STREAM_MAXW16 \ - 300 /* The old maximum size still needed for the decoding */ -#define STREAM_MAXW16_30MS \ - 100 /* 100 Word16 = 200 bytes = 53.4 kbit/s @ 30 ms.framelength */ -#define STREAM_MAXW16_60MS \ - 200 /* 200 Word16 = 400 bytes = 53.4 kbit/s @ 60 ms.framelength */ -/* This is used only at the decoder bit-stream struct. - * - The encoder and decoder bitstream containers are of different size because - * old iSAC limited the encoded bitstream to 600 bytes. But newer versions - * restrict to shorter bitstream. - * - We add 10 bytes of guards to the internal bitstream container. The reason - * is that entropy decoder might read few bytes (3 according to our - * observations) more than the actual size of the bitstream. To avoid reading - * outside memory, in rare occasion of full-size bitstream we add 10 bytes - * of guard. */ -#define INTERNAL_STREAM_SIZE_W16 (STREAM_MAXW16 + 5) - -/* storage size for bit counts */ -//#define BIT_COUNTER_SIZE 30 -/* maximum order of any AR model or filter */ -#define MAX_AR_MODEL_ORDER 12 - -/* Maximum number of iterations allowed to limit payload size */ -#define MAX_PAYLOAD_LIMIT_ITERATION 1 - -/* Bandwidth estimator */ - -#define MIN_ISAC_BW 10000 /* Minimum bandwidth in bits per sec */ -#define MAX_ISAC_BW 32000 /* Maxmum bandwidth in bits per sec */ -#define MIN_ISAC_MD 5 /* Minimum Max Delay in ?? */ -#define MAX_ISAC_MD 25 /* Maxmum Max Delay in ?? */ -#define DELAY_CORRECTION_MAX 717 -#define DELAY_CORRECTION_MED 819 -#define Thld_30_60 18000 -#define Thld_60_30 27000 - -/* assumed header size; we don't know the exact number (header compression may - * be used) */ -#define HEADER_SIZE 35 /* bytes */ -#define INIT_FRAME_LEN 60 -#define INIT_BN_EST 20000 -#define INIT_BN_EST_Q7 2560000 /* 20 kbps in Q7 */ -#define INIT_REC_BN_EST_Q5 789312 /* INIT_BN_EST + INIT_HDR_RATE in Q5 */ - -/* 8738 in Q18 is ~ 1/30 */ -/* #define INIT_HDR_RATE (((HEADER_SIZE * 8 * 1000) * 8738) >> NUM_BITS_TO_SHIFT - * (INIT_FRAME_LEN)) */ -#define INIT_HDR_RATE 4666 -/* number of packets in a row for a high rate burst */ -#define BURST_LEN 3 -/* ms, max time between two full bursts */ -#define BURST_INTERVAL 800 -/* number of packets in a row for initial high rate burst */ -#define INIT_BURST_LEN 5 -/* bits/s, rate for the first BURST_LEN packets */ -#define INIT_RATE 10240000 /* INIT_BN_EST in Q9 */ - -/* For pitch analysis */ -#define PITCH_FRAME_LEN 240 /* (FRAMESAMPLES/2) 30 ms */ -#define PITCH_MAX_LAG 140 /* 57 Hz */ -#define PITCH_MIN_LAG 20 /* 400 Hz */ -#define PITCH_MIN_LAG_Q8 5120 /* 256 * PITCH_MIN_LAG */ -#define OFFSET_Q8 768 /* 256 * 3 */ - -#define PITCH_MAX_GAIN_Q12 1843 /* 0.45 */ -#define PITCH_LAG_SPAN2 65 /* (PITCH_MAX_LAG/2-PITCH_MIN_LAG/2+5) */ -#define PITCH_CORR_LEN2 60 /* 15 ms */ -#define PITCH_CORR_STEP2 60 /* (PITCH_FRAME_LEN/4) */ -#define PITCH_SUBFRAMES 4 -#define PITCH_SUBFRAME_LEN 60 /* (PITCH_FRAME_LEN/PITCH_SUBFRAMES) */ - -/* For pitch filter */ -#define PITCH_BUFFSIZE \ - 190 /* (PITCH_MAX_LAG + 50) Extra 50 for fraction and LP filters */ -#define PITCH_INTBUFFSIZE 430 /* (PITCH_FRAME_LEN+PITCH_BUFFSIZE) */ -#define PITCH_FRACS 8 -#define PITCH_FRACORDER 9 -#define PITCH_DAMPORDER 5 - -/* Order of high pass filter */ -#define HPORDER 2 - -/* PLC */ -#define DECAY_RATE \ - 10 /* Q15, 20% of decay every lost frame apllied linearly sample by sample*/ -#define PLC_WAS_USED 1 -#define PLC_NOT_USED 3 -#define RECOVERY_OVERLAP 80 -#define RESAMP_RES 256 -#define RESAMP_RES_BIT 8 - -/* Define Error codes */ -/* 6000 General */ -#define ISAC_MEMORY_ALLOCATION_FAILED 6010 -#define ISAC_MODE_MISMATCH 6020 -#define ISAC_DISALLOWED_BOTTLENECK 6030 -#define ISAC_DISALLOWED_FRAME_LENGTH 6040 -/* 6200 Bandwidth estimator */ -#define ISAC_RANGE_ERROR_BW_ESTIMATOR 6240 -/* 6400 Encoder */ -#define ISAC_ENCODER_NOT_INITIATED 6410 -#define ISAC_DISALLOWED_CODING_MODE 6420 -#define ISAC_DISALLOWED_FRAME_MODE_ENCODER 6430 -#define ISAC_DISALLOWED_BITSTREAM_LENGTH 6440 -#define ISAC_PAYLOAD_LARGER_THAN_LIMIT 6450 -/* 6600 Decoder */ -#define ISAC_DECODER_NOT_INITIATED 6610 -#define ISAC_EMPTY_PACKET 6620 -#define ISAC_PACKET_TOO_SHORT 6625 -#define ISAC_DISALLOWED_FRAME_MODE_DECODER 6630 -#define ISAC_RANGE_ERROR_DECODE_FRAME_LENGTH 6640 -#define ISAC_RANGE_ERROR_DECODE_BANDWIDTH 6650 -#define ISAC_RANGE_ERROR_DECODE_PITCH_GAIN 6660 -#define ISAC_RANGE_ERROR_DECODE_PITCH_LAG 6670 -#define ISAC_RANGE_ERROR_DECODE_LPC 6680 -#define ISAC_RANGE_ERROR_DECODE_SPECTRUM 6690 -#define ISAC_LENGTH_MISMATCH 6730 -/* 6800 Call setup formats */ -#define ISAC_INCOMPATIBLE_FORMATS 6810 - -#endif /* MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_SETTINGS_H_ */ diff --git a/modules/audio_coding/codecs/isac/fix/source/spectrum_ar_model_tables.c b/modules/audio_coding/codecs/isac/fix/source/spectrum_ar_model_tables.c deleted file mode 100644 index 4ef9a338cc..0000000000 --- a/modules/audio_coding/codecs/isac/fix/source/spectrum_ar_model_tables.c +++ /dev/null @@ -1,193 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/* - * spectrum_ar_model_tables.c - * - * This file contains tables with AR coefficients, Gain coefficients - * and cosine tables. - * - */ - -#include "modules/audio_coding/codecs/isac/fix/source/spectrum_ar_model_tables.h" -#include "modules/audio_coding/codecs/isac/fix/source/settings.h" - -/********************* AR Coefficient Tables ************************/ - -/* cdf for quantized reflection coefficient 1 */ -const uint16_t WebRtcIsacfix_kRc1Cdf[12] = { - 0, 2, 4, 129, 7707, 57485, 65495, 65527, 65529, 65531, - 65533, 65535 -}; - -/* cdf for quantized reflection coefficient 2 */ -const uint16_t WebRtcIsacfix_kRc2Cdf[12] = { - 0, 2, 4, 7, 531, 25298, 64525, 65526, 65529, 65531, - 65533, 65535 -}; - -/* cdf for quantized reflection coefficient 3 */ -const uint16_t WebRtcIsacfix_kRc3Cdf[12] = { - 0, 2, 4, 6, 620, 22898, 64843, 65527, 65529, 65531, - 65533, 65535 -}; - -/* cdf for quantized reflection coefficient 4 */ -const uint16_t WebRtcIsacfix_kRc4Cdf[12] = { - 0, 2, 4, 6, 35, 10034, 60733, 65506, 65529, 65531, - 65533, 65535 -}; - -/* cdf for quantized reflection coefficient 5 */ -const uint16_t WebRtcIsacfix_kRc5Cdf[12] = { - 0, 2, 4, 6, 36, 7567, 56727, 65385, 65529, 65531, - 65533, 65535 -}; - -/* cdf for quantized reflection coefficient 6 */ -const uint16_t WebRtcIsacfix_kRc6Cdf[12] = { - 0, 2, 4, 6, 14, 6579, 57360, 65409, 65529, 65531, - 65533, 65535 -}; - -/* representation levels for quantized reflection coefficient 1 */ -const int16_t WebRtcIsacfix_kRc1Levels[11] = { - -32104, -29007, -23202, -15496, -9279, -2577, 5934, 17535, 24512, 29503, 32104 -}; - -/* representation levels for quantized reflection coefficient 2 */ -const int16_t WebRtcIsacfix_kRc2Levels[11] = { - -32104, -29503, -23494, -15261, -7309, -1399, 6158, 16381, 24512, 29503, 32104 -}; - -/* representation levels for quantized reflection coefficient 3 */ -const int16_t WebRtcIsacfix_kRc3Levels[11] = { - -32104, -29503, -23157, -15186, -7347, -1359, 5829, 17535, 24512, 29503, 32104 -}; - -/* representation levels for quantized reflection coefficient 4 */ -const int16_t WebRtcIsacfix_kRc4Levels[11] = { - -32104, -29503, -24512, -15362, -6665, -342, 6596, 14585, 24512, 29503, 32104 -}; - -/* representation levels for quantized reflection coefficient 5 */ -const int16_t WebRtcIsacfix_kRc5Levels[11] = { - -32104, -29503, -24512, -15005, -6564, -106, 7123, 14920, 24512, 29503, 32104 -}; - -/* representation levels for quantized reflection coefficient 6 */ -const int16_t WebRtcIsacfix_kRc6Levels[11] = { - -32104, -29503, -24512, -15096, -6656, -37, 7036, 14847, 24512, 29503, 32104 -}; - -/* quantization boundary levels for reflection coefficients */ -const int16_t WebRtcIsacfix_kRcBound[12] = { - -32768, -31441, -27566, -21458, -13612, -4663, - 4663, 13612, 21458, 27566, 31441, 32767 -}; - -/* initial index for AR reflection coefficient quantizer and cdf table search */ -const uint16_t WebRtcIsacfix_kRcInitInd[6] = { - 5, 5, 5, 5, 5, 5 -}; - -/* pointers to AR cdf tables */ -const uint16_t *WebRtcIsacfix_kRcCdfPtr[AR_ORDER] = { - WebRtcIsacfix_kRc1Cdf, - WebRtcIsacfix_kRc2Cdf, - WebRtcIsacfix_kRc3Cdf, - WebRtcIsacfix_kRc4Cdf, - WebRtcIsacfix_kRc5Cdf, - WebRtcIsacfix_kRc6Cdf -}; - -/* pointers to AR representation levels tables */ -const int16_t *WebRtcIsacfix_kRcLevPtr[AR_ORDER] = { - WebRtcIsacfix_kRc1Levels, - WebRtcIsacfix_kRc2Levels, - WebRtcIsacfix_kRc3Levels, - WebRtcIsacfix_kRc4Levels, - WebRtcIsacfix_kRc5Levels, - WebRtcIsacfix_kRc6Levels -}; - - -/******************** GAIN Coefficient Tables ***********************/ - -/* cdf for Gain coefficient */ -const uint16_t WebRtcIsacfix_kGainCdf[19] = { - 0, 2, 4, 6, 8, 10, 12, 14, 16, 1172, - 11119, 29411, 51699, 64445, 65527, 65529, 65531, 65533, 65535 -}; - -/* representation levels for quantized squared Gain coefficient */ -const int32_t WebRtcIsacfix_kGain2Lev[18] = { - 128, 128, 128, 128, 128, 215, 364, 709, 1268, - 1960, 3405, 6078, 11286, 17827, 51918, 134498, 487432, 2048000 -}; - -/* quantization boundary levels for squared Gain coefficient */ -const int32_t WebRtcIsacfix_kGain2Bound[19] = { - 0, 21, 35, 59, 99, 166, 280, 475, 815, 1414, - 2495, 4505, 8397, 16405, 34431, 81359, 240497, 921600, 0x7FFFFFFF -}; - -/* pointers to Gain cdf table */ -const uint16_t *WebRtcIsacfix_kGainPtr[1] = { - WebRtcIsacfix_kGainCdf -}; - -/* gain initial index for gain quantizer and cdf table search */ -const uint16_t WebRtcIsacfix_kGainInitInd[1] = { - 11 -}; - - -/************************* Cosine Tables ****************************/ - -/* cosine table */ -const int16_t WebRtcIsacfix_kCos[6][60] = { - { 512, 512, 511, 510, 508, 507, 505, 502, 499, 496, - 493, 489, 485, 480, 476, 470, 465, 459, 453, 447, - 440, 433, 426, 418, 410, 402, 394, 385, 376, 367, - 357, 348, 338, 327, 317, 306, 295, 284, 273, 262, - 250, 238, 226, 214, 202, 190, 177, 165, 152, 139, - 126, 113, 100, 87, 73, 60, 47, 33, 20, 7 }, - { 512, 510, 508, 503, 498, 491, 483, 473, 462, 450, - 437, 422, 406, 389, 371, 352, 333, 312, 290, 268, - 244, 220, 196, 171, 145, 120, 93, 67, 40, 13, - -13, -40, -67, -93, -120, -145, -171, -196, -220, -244, - -268, -290, -312, -333, -352, -371, -389, -406, -422, -437, - -450, -462, -473, -483, -491, -498, -503, -508, -510, -512 }, - { 512, 508, 502, 493, 480, 465, 447, 426, 402, 376, - 348, 317, 284, 250, 214, 177, 139, 100, 60, 20, - -20, -60, -100, -139, -177, -214, -250, -284, -317, -348, - -376, -402, -426, -447, -465, -480, -493, -502, -508, -512, - -512, -508, -502, -493, -480, -465, -447, -426, -402, -376, - -348, -317, -284, -250, -214, -177, -139, -100, -60, -20 }, - { 511, 506, 495, 478, 456, 429, 398, 362, 322, 279, - 232, 183, 133, 80, 27, -27, -80, -133, -183, -232, - -279, -322, -362, -398, -429, -456, -478, -495, -506, -511, - -511, -506, -495, -478, -456, -429, -398, -362, -322, -279, - -232, -183, -133, -80, -27, 27, 80, 133, 183, 232, - 279, 322, 362, 398, 429, 456, 478, 495, 506, 511 }, - { 511, 502, 485, 459, 426, 385, 338, 284, 226, 165, - 100, 33, -33, -100, -165, -226, -284, -338, -385, -426, - -459, -485, -502, -511, -511, -502, -485, -459, -426, -385, - -338, -284, -226, -165, -100, -33, 33, 100, 165, 226, - 284, 338, 385, 426, 459, 485, 502, 511, 511, 502, - 485, 459, 426, 385, 338, 284, 226, 165, 100, 33 }, - { 510, 498, 473, 437, 389, 333, 268, 196, 120, 40, - -40, -120, -196, -268, -333, -389, -437, -473, -498, -510, - -510, -498, -473, -437, -389, -333, -268, -196, -120, -40, - 40, 120, 196, 268, 333, 389, 437, 473, 498, 510, - 510, 498, 473, 437, 389, 333, 268, 196, 120, 40, - -40, -120, -196, -268, -333, -389, -437, -473, -498, -510 } -}; diff --git a/modules/audio_coding/codecs/isac/fix/source/spectrum_ar_model_tables.h b/modules/audio_coding/codecs/isac/fix/source/spectrum_ar_model_tables.h deleted file mode 100644 index 2282a369cb..0000000000 --- a/modules/audio_coding/codecs/isac/fix/source/spectrum_ar_model_tables.h +++ /dev/null @@ -1,97 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/* - * spectrum_ar_model_tables.h - * - * This file contains definitions of tables with AR coefficients, - * Gain coefficients and cosine tables. - * - */ - -#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_SPECTRUM_AR_MODEL_TABLES_H_ -#define MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_SPECTRUM_AR_MODEL_TABLES_H_ - -#include - -#include "modules/audio_coding/codecs/isac/fix/source/settings.h" - -/********************* AR Coefficient Tables ************************/ -/* cdf for quantized reflection coefficient 1 */ -extern const uint16_t WebRtcIsacfix_kRc1Cdf[12]; - -/* cdf for quantized reflection coefficient 2 */ -extern const uint16_t WebRtcIsacfix_kRc2Cdf[12]; - -/* cdf for quantized reflection coefficient 3 */ -extern const uint16_t WebRtcIsacfix_kRc3Cdf[12]; - -/* cdf for quantized reflection coefficient 4 */ -extern const uint16_t WebRtcIsacfix_kRc4Cdf[12]; - -/* cdf for quantized reflection coefficient 5 */ -extern const uint16_t WebRtcIsacfix_kRc5Cdf[12]; - -/* cdf for quantized reflection coefficient 6 */ -extern const uint16_t WebRtcIsacfix_kRc6Cdf[12]; - -/* representation levels for quantized reflection coefficient 1 */ -extern const int16_t WebRtcIsacfix_kRc1Levels[11]; - -/* representation levels for quantized reflection coefficient 2 */ -extern const int16_t WebRtcIsacfix_kRc2Levels[11]; - -/* representation levels for quantized reflection coefficient 3 */ -extern const int16_t WebRtcIsacfix_kRc3Levels[11]; - -/* representation levels for quantized reflection coefficient 4 */ -extern const int16_t WebRtcIsacfix_kRc4Levels[11]; - -/* representation levels for quantized reflection coefficient 5 */ -extern const int16_t WebRtcIsacfix_kRc5Levels[11]; - -/* representation levels for quantized reflection coefficient 6 */ -extern const int16_t WebRtcIsacfix_kRc6Levels[11]; - -/* quantization boundary levels for reflection coefficients */ -extern const int16_t WebRtcIsacfix_kRcBound[12]; - -/* initial indices for AR reflection coefficient quantizer and cdf table search - */ -extern const uint16_t WebRtcIsacfix_kRcInitInd[AR_ORDER]; - -/* pointers to AR cdf tables */ -extern const uint16_t* WebRtcIsacfix_kRcCdfPtr[AR_ORDER]; - -/* pointers to AR representation levels tables */ -extern const int16_t* WebRtcIsacfix_kRcLevPtr[AR_ORDER]; - -/******************** GAIN Coefficient Tables ***********************/ -/* cdf for Gain coefficient */ -extern const uint16_t WebRtcIsacfix_kGainCdf[19]; - -/* representation levels for quantized Gain coefficient */ -extern const int32_t WebRtcIsacfix_kGain2Lev[18]; - -/* squared quantization boundary levels for Gain coefficient */ -extern const int32_t WebRtcIsacfix_kGain2Bound[19]; - -/* pointer to Gain cdf table */ -extern const uint16_t* WebRtcIsacfix_kGainPtr[1]; - -/* Gain initial index for gain quantizer and cdf table search */ -extern const uint16_t WebRtcIsacfix_kGainInitInd[1]; - -/************************* Cosine Tables ****************************/ -/* Cosine table */ -extern const int16_t WebRtcIsacfix_kCos[6][60]; - -#endif /* MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_SPECTRUM_AR_MODEL_TABLES_H_ \ - */ diff --git a/modules/audio_coding/codecs/isac/fix/source/structs.h b/modules/audio_coding/codecs/isac/fix/source/structs.h deleted file mode 100644 index 3044d5176b..0000000000 --- a/modules/audio_coding/codecs/isac/fix/source/structs.h +++ /dev/null @@ -1,345 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/* - * structs.h - * - * This header file contains all the structs used in the ISAC codec - * - */ - -#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_STRUCTS_H_ -#define MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_STRUCTS_H_ - -#include "common_audio/signal_processing/include/signal_processing_library.h" -#include "modules/audio_coding/codecs/isac/bandwidth_info.h" -#include "modules/audio_coding/codecs/isac/fix/source/settings.h" - -/* Bitstream struct for decoder */ -typedef struct Bitstreamstruct_dec { - uint16_t stream[INTERNAL_STREAM_SIZE_W16]; /* Array bytestream to decode */ - uint32_t W_upper; /* Upper boundary of interval W */ - uint32_t streamval; - uint16_t stream_index; /* Index to the current position in bytestream */ - int16_t full; /* 0 - first byte in memory filled, second empty*/ - /* 1 - both bytes are empty (we just filled the previous memory */ - - size_t stream_size; /* The size of stream in bytes. */ -} Bitstr_dec; - -/* Bitstream struct for encoder */ -typedef struct Bitstreamstruct_enc { - uint16_t - stream[STREAM_MAXW16_60MS]; /* Vector for adding encoded bytestream */ - uint32_t W_upper; /* Upper boundary of interval W */ - uint32_t streamval; - uint16_t stream_index; /* Index to the current position in bytestream */ - int16_t full; /* 0 - first byte in memory filled, second empty*/ - /* 1 - both bytes are empty (we just filled the previous memory */ - -} Bitstr_enc; - -typedef struct { - int16_t DataBufferLoQ0[WINLEN]; - int16_t DataBufferHiQ0[WINLEN]; - - int32_t CorrBufLoQQ[ORDERLO + 1]; - int32_t CorrBufHiQQ[ORDERHI + 1]; - - int16_t CorrBufLoQdom[ORDERLO + 1]; - int16_t CorrBufHiQdom[ORDERHI + 1]; - - int32_t PreStateLoGQ15[ORDERLO + 1]; - int32_t PreStateHiGQ15[ORDERHI + 1]; - - uint32_t OldEnergy; - -} MaskFiltstr_enc; - -typedef struct { - int16_t PostStateLoGQ0[ORDERLO + 1]; - int16_t PostStateHiGQ0[ORDERHI + 1]; - - uint32_t OldEnergy; - -} MaskFiltstr_dec; - -typedef struct { - // state vectors for each of the two analysis filters - - int32_t INSTAT1_fix[2 * (QORDER - 1)]; - int32_t INSTAT2_fix[2 * (QORDER - 1)]; - int16_t INLABUF1_fix[QLOOKAHEAD]; - int16_t INLABUF2_fix[QLOOKAHEAD]; - - /* High pass filter */ - int32_t HPstates_fix[HPORDER]; - -} PreFiltBankstr; - -typedef struct { - // state vectors for each of the two analysis filters - int32_t STATE_0_LOWER_fix[2 * POSTQORDER]; - int32_t STATE_0_UPPER_fix[2 * POSTQORDER]; - - /* High pass filter */ - - int32_t HPstates1_fix[HPORDER]; - int32_t HPstates2_fix[HPORDER]; - -} PostFiltBankstr; - -typedef struct { - /* data buffer for pitch filter */ - int16_t ubufQQ[PITCH_BUFFSIZE]; - - /* low pass state vector */ - int16_t ystateQQ[PITCH_DAMPORDER]; - - /* old lag and gain */ - int16_t oldlagQ7; - int16_t oldgainQ12; - -} PitchFiltstr; - -typedef struct { - // for inital estimator - int16_t dec_buffer16[PITCH_CORR_LEN2 + PITCH_CORR_STEP2 + PITCH_MAX_LAG / 2 - - PITCH_FRAME_LEN / 2 + 2]; - int32_t decimator_state32[2 * ALLPASSSECTIONS + 1]; - int16_t inbuf[QLOOKAHEAD]; - - PitchFiltstr PFstr_wght; - PitchFiltstr PFstr; - -} PitchAnalysisStruct; - -typedef struct { - /* Parameters used in PLC to avoid re-computation */ - - /* --- residual signals --- */ - int16_t prevPitchInvIn[FRAMESAMPLES / 2]; - int16_t prevPitchInvOut[PITCH_MAX_LAG + 10]; // [FRAMESAMPLES/2]; save 90 - int32_t prevHP[PITCH_MAX_LAG + 10]; // [FRAMESAMPLES/2]; save 90 - - int16_t decayCoeffPriodic; /* how much to supress a sample */ - int16_t decayCoeffNoise; - int16_t used; /* if PLC is used */ - - int16_t* lastPitchLP; // [FRAMESAMPLES/2]; saved 240; - - /* --- LPC side info --- */ - int16_t lofilt_coefQ15[ORDERLO]; - int16_t hifilt_coefQ15[ORDERHI]; - int32_t gain_lo_hiQ17[2]; - - /* --- LTP side info --- */ - int16_t AvgPitchGain_Q12; - int16_t lastPitchGain_Q12; - int16_t lastPitchLag_Q7; - - /* --- Add-overlap in recovery packet --- */ - int16_t overlapLP[RECOVERY_OVERLAP]; // [FRAMESAMPLES/2]; saved 160 - - int16_t pitchCycles; - int16_t A; - int16_t B; - size_t pitchIndex; - size_t stretchLag; - int16_t* prevPitchLP; // [ FRAMESAMPLES/2 ]; saved 240 - int16_t seed; - - int16_t std; -} PLCstr; - -/* Have instance of struct together with other iSAC structs */ -typedef struct { - int16_t prevFrameSizeMs; /* Previous frame size (in ms) */ - uint16_t prevRtpNumber; /* Previous RTP timestamp from received packet */ - /* (in samples relative beginning) */ - uint32_t prevSendTime; /* Send time for previous packet, from RTP header */ - uint32_t prevArrivalTime; /* Arrival time for previous packet (in ms using - timeGetTime()) */ - uint16_t prevRtpRate; /* rate of previous packet, derived from RTP timestamps - (in bits/s) */ - uint32_t lastUpdate; /* Time since the last update of the Bottle Neck estimate - (in samples) */ - uint32_t lastReduction; /* Time sinse the last reduction (in samples) */ - int32_t countUpdates; /* How many times the estimate was update in the - beginning */ - - /* The estimated bottle neck rate from there to here (in bits/s) */ - uint32_t recBw; - uint32_t recBwInv; - uint32_t recBwAvg; - uint32_t recBwAvgQ; - - uint32_t minBwInv; - uint32_t maxBwInv; - - /* The estimated mean absolute jitter value, as seen on this side (in ms) */ - int32_t recJitter; - int32_t recJitterShortTerm; - int32_t recJitterShortTermAbs; - int32_t recMaxDelay; - int32_t recMaxDelayAvgQ; - - int16_t recHeaderRate; /* (assumed) bitrate for headers (bps) */ - - uint32_t sendBwAvg; /* The estimated bottle neck rate from here to there (in - bits/s) */ - int32_t sendMaxDelayAvg; /* The estimated mean absolute jitter value, as seen - on the other siee (in ms) */ - - int16_t countRecPkts; /* number of packets received since last update */ - int16_t highSpeedRec; /* flag for marking that a high speed network has been - detected downstream */ - - /* number of consecutive pkts sent during which the bwe estimate has - remained at a value greater than the downstream threshold for determining - highspeed network */ - int16_t countHighSpeedRec; - - /* flag indicating bwe should not adjust down immediately for very late pckts - */ - int16_t inWaitPeriod; - - /* variable holding the time of the start of a window of time when - bwe should not adjust down immediately for very late pckts */ - uint32_t startWaitPeriod; - - /* number of consecutive pkts sent during which the bwe estimate has - remained at a value greater than the upstream threshold for determining - highspeed network */ - int16_t countHighSpeedSent; - - /* flag indicated the desired number of packets over threshold rate have been - sent and bwe will assume the connection is over broadband network */ - int16_t highSpeedSend; - - IsacBandwidthInfo external_bw_info; -} BwEstimatorstr; - -typedef struct { - /* boolean, flags if previous packet exceeded B.N. */ - int16_t PrevExceed; - /* ms */ - int16_t ExceedAgo; - /* packets left to send in current burst */ - int16_t BurstCounter; - /* packets */ - int16_t InitCounter; - /* ms remaining in buffer when next packet will be sent */ - int16_t StillBuffered; - -} RateModel; - -/* The following strutc is used to store data from encoding, to make it - fast and easy to construct a new bitstream with a different Bandwidth - estimate. All values (except framelength and minBytes) is double size to - handle 60 ms of data. -*/ -typedef struct { - /* Used to keep track of if it is first or second part of 60 msec packet */ - int startIdx; - - /* Frame length in samples */ - int16_t framelength; - - /* Pitch Gain */ - int16_t pitchGain_index[2]; - - /* Pitch Lag */ - int32_t meanGain[2]; - int16_t pitchIndex[PITCH_SUBFRAMES * 2]; - - /* LPC */ - int32_t LPCcoeffs_g[12 * 2]; /* KLT_ORDER_GAIN = 12 */ - int16_t LPCindex_s[108 * 2]; /* KLT_ORDER_SHAPE = 108 */ - int16_t LPCindex_g[12 * 2]; /* KLT_ORDER_GAIN = 12 */ - - /* Encode Spec */ - int16_t fre[FRAMESAMPLES]; - int16_t fim[FRAMESAMPLES]; - int16_t AvgPitchGain[2]; - - /* Used in adaptive mode only */ - int minBytes; - -} IsacSaveEncoderData; - -typedef struct { - Bitstr_enc bitstr_obj; - MaskFiltstr_enc maskfiltstr_obj; - PreFiltBankstr prefiltbankstr_obj; - PitchFiltstr pitchfiltstr_obj; - PitchAnalysisStruct pitchanalysisstr_obj; - RateModel rate_data_obj; - - int16_t buffer_index; - int16_t current_framesamples; - - int16_t data_buffer_fix[FRAMESAMPLES]; // the size was MAX_FRAMESAMPLES - - int16_t frame_nb; - int16_t BottleNeck; - int16_t MaxDelay; - int16_t new_framelength; - int16_t s2nr; - uint16_t MaxBits; - - int16_t bitstr_seed; - - IsacSaveEncoderData* SaveEnc_ptr; - int16_t payloadLimitBytes30; /* Maximum allowed number of bits for a 30 msec - packet */ - int16_t payloadLimitBytes60; /* Maximum allowed number of bits for a 30 msec - packet */ - int16_t maxPayloadBytes; /* Maximum allowed number of bits for both 30 and 60 - msec packet */ - int16_t maxRateInBytes; /* Maximum allowed rate in bytes per 30 msec packet */ - int16_t enforceFrameSize; /* If set iSAC will never change packet size */ - -} IsacFixEncoderInstance; - -typedef struct { - Bitstr_dec bitstr_obj; - MaskFiltstr_dec maskfiltstr_obj; - PostFiltBankstr postfiltbankstr_obj; - PitchFiltstr pitchfiltstr_obj; - PLCstr plcstr_obj; /* TS; for packet loss concealment */ -} IsacFixDecoderInstance; - -typedef struct { - IsacFixEncoderInstance ISACenc_obj; - IsacFixDecoderInstance ISACdec_obj; - BwEstimatorstr bwestimator_obj; - int16_t CodingMode; /* 0 = adaptive; 1 = instantaneous */ - int16_t errorcode; - int16_t initflag; /* 0 = nothing initiated; 1 = encoder or decoder */ - /* not initiated; 2 = all initiated */ -} ISACFIX_SubStruct; - -typedef struct { - int32_t lpcGains[12]; /* 6 lower-band & 6 upper-band we may need to double it - for 60*/ - /* */ - uint32_t W_upper; /* Upper boundary of interval W */ - uint32_t streamval; - uint16_t stream_index; /* Index to the current position in bytestream */ - int16_t full; /* 0 - first byte in memory filled, second empty*/ - /* 1 - both bytes are empty (we just filled the previous memory */ - uint16_t beforeLastWord; - uint16_t lastWord; -} transcode_obj; - -// Bitstr_enc myBitStr; - -#endif /* MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_STRUCTS_H_ */ diff --git a/modules/audio_coding/codecs/isac/fix/source/transform.c b/modules/audio_coding/codecs/isac/fix/source/transform.c deleted file mode 100644 index 80b244b5f1..0000000000 --- a/modules/audio_coding/codecs/isac/fix/source/transform.c +++ /dev/null @@ -1,214 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/* - * WebRtcIsacfix_kTransform.c - * - * Transform functions - * - */ - -#include "modules/audio_coding/codecs/isac/fix/source/codec.h" -#include "modules/audio_coding/codecs/isac/fix/source/fft.h" -#include "modules/audio_coding/codecs/isac/fix/source/settings.h" -#include "modules/third_party/fft/fft.h" - -/* Tables are defined in transform_tables.c file or ARM assembly files. */ -/* Cosine table 1 in Q14 */ -extern const int16_t WebRtcIsacfix_kCosTab1[FRAMESAMPLES/2]; -/* Sine table 1 in Q14 */ -extern const int16_t WebRtcIsacfix_kSinTab1[FRAMESAMPLES/2]; -/* Sine table 2 in Q14 */ -extern const int16_t WebRtcIsacfix_kSinTab2[FRAMESAMPLES/4]; - -void WebRtcIsacfix_Time2SpecC(int16_t *inre1Q9, - int16_t *inre2Q9, - int16_t *outreQ7, - int16_t *outimQ7) -{ - - int k; - int32_t tmpreQ16[FRAMESAMPLES/2], tmpimQ16[FRAMESAMPLES/2]; - int16_t tmp1rQ14, tmp1iQ14; - int32_t xrQ16, xiQ16, yrQ16, yiQ16; - int32_t v1Q16, v2Q16; - int16_t factQ19, sh; - - /* Multiply with complex exponentials and combine into one complex vector */ - factQ19 = 16921; // 0.5/sqrt(240) in Q19 is round(.5/sqrt(240)*(2^19)) = 16921 - for (k = 0; k < FRAMESAMPLES/2; k++) { - tmp1rQ14 = WebRtcIsacfix_kCosTab1[k]; - tmp1iQ14 = WebRtcIsacfix_kSinTab1[k]; - xrQ16 = (tmp1rQ14 * inre1Q9[k] + tmp1iQ14 * inre2Q9[k]) >> 7; - xiQ16 = (tmp1rQ14 * inre2Q9[k] - tmp1iQ14 * inre1Q9[k]) >> 7; - // Q-domains below: (Q16*Q19>>16)>>3 = Q16 - tmpreQ16[k] = (WEBRTC_SPL_MUL_16_32_RSFT16(factQ19, xrQ16) + 4) >> 3; - tmpimQ16[k] = (WEBRTC_SPL_MUL_16_32_RSFT16(factQ19, xiQ16) + 4) >> 3; - } - - - xrQ16 = WebRtcSpl_MaxAbsValueW32(tmpreQ16, FRAMESAMPLES/2); - yrQ16 = WebRtcSpl_MaxAbsValueW32(tmpimQ16, FRAMESAMPLES/2); - if (yrQ16>xrQ16) { - xrQ16 = yrQ16; - } - - sh = WebRtcSpl_NormW32(xrQ16); - sh = sh-24; //if sh becomes >=0, then we should shift sh steps to the left, and the domain will become Q(16+sh) - //if sh becomes <0, then we should shift -sh steps to the right, and the domain will become Q(16+sh) - - //"Fastest" vectors - if (sh>=0) { - for (k=0; k> -sh); // Q(16+sh) - inre2Q9[k] = (int16_t)((tmpimQ16[k] + round) >> -sh); // Q(16+sh) - } - } - - /* Get DFT */ - WebRtcIsacfix_FftRadix16Fastest(inre1Q9, inre2Q9, -1); // real call - - //"Fastest" vectors - if (sh>=0) { - for (k=0; k> sh; // Q(16+sh) -> Q16 - tmpimQ16[k] = inre2Q9[k] >> sh; // Q(16+sh) -> Q16 - } - } else { - for (k=0; k Q16 - tmpimQ16[k] = inre2Q9[k] << -sh; // Q(16+sh) -> Q16 - } - } - - - /* Use symmetry to separate into two complex vectors and center frames in time around zero */ - for (k = 0; k < FRAMESAMPLES/4; k++) { - xrQ16 = tmpreQ16[k] + tmpreQ16[FRAMESAMPLES/2 - 1 - k]; - yiQ16 = -tmpreQ16[k] + tmpreQ16[FRAMESAMPLES/2 - 1 - k]; - xiQ16 = tmpimQ16[k] - tmpimQ16[FRAMESAMPLES/2 - 1 - k]; - yrQ16 = tmpimQ16[k] + tmpimQ16[FRAMESAMPLES/2 - 1 - k]; - tmp1rQ14 = -WebRtcIsacfix_kSinTab2[FRAMESAMPLES/4 - 1 - k]; - tmp1iQ14 = WebRtcIsacfix_kSinTab2[k]; - v1Q16 = WEBRTC_SPL_MUL_16_32_RSFT14(tmp1rQ14, xrQ16) - WEBRTC_SPL_MUL_16_32_RSFT14(tmp1iQ14, xiQ16); - v2Q16 = WEBRTC_SPL_MUL_16_32_RSFT14(tmp1iQ14, xrQ16) + WEBRTC_SPL_MUL_16_32_RSFT14(tmp1rQ14, xiQ16); - outreQ7[k] = (int16_t)(v1Q16 >> 9); - outimQ7[k] = (int16_t)(v2Q16 >> 9); - v1Q16 = -WEBRTC_SPL_MUL_16_32_RSFT14(tmp1iQ14, yrQ16) - WEBRTC_SPL_MUL_16_32_RSFT14(tmp1rQ14, yiQ16); - v2Q16 = -WEBRTC_SPL_MUL_16_32_RSFT14(tmp1rQ14, yrQ16) + WEBRTC_SPL_MUL_16_32_RSFT14(tmp1iQ14, yiQ16); - // CalcLrIntQ(v1Q16, 9); - outreQ7[FRAMESAMPLES / 2 - 1 - k] = (int16_t)(v1Q16 >> 9); - // CalcLrIntQ(v2Q16, 9); - outimQ7[FRAMESAMPLES / 2 - 1 - k] = (int16_t)(v2Q16 >> 9); - - } -} - - -void WebRtcIsacfix_Spec2TimeC(int16_t *inreQ7, int16_t *inimQ7, int32_t *outre1Q16, int32_t *outre2Q16) -{ - - int k; - int16_t tmp1rQ14, tmp1iQ14; - int32_t xrQ16, xiQ16, yrQ16, yiQ16; - int32_t tmpInRe, tmpInIm, tmpInRe2, tmpInIm2; - int16_t factQ11; - int16_t sh; - - for (k = 0; k < FRAMESAMPLES/4; k++) { - /* Move zero in time to beginning of frames */ - tmp1rQ14 = -WebRtcIsacfix_kSinTab2[FRAMESAMPLES/4 - 1 - k]; - tmp1iQ14 = WebRtcIsacfix_kSinTab2[k]; - - tmpInRe = inreQ7[k] * (1 << 9); // Q7 -> Q16 - tmpInIm = inimQ7[k] * (1 << 9); // Q7 -> Q16 - tmpInRe2 = inreQ7[FRAMESAMPLES / 2 - 1 - k] * (1 << 9); // Q7 -> Q16 - tmpInIm2 = inimQ7[FRAMESAMPLES / 2 - 1 - k] * (1 << 9); // Q7 -> Q16 - - xrQ16 = WEBRTC_SPL_MUL_16_32_RSFT14(tmp1rQ14, tmpInRe) + WEBRTC_SPL_MUL_16_32_RSFT14(tmp1iQ14, tmpInIm); - xiQ16 = WEBRTC_SPL_MUL_16_32_RSFT14(tmp1rQ14, tmpInIm) - WEBRTC_SPL_MUL_16_32_RSFT14(tmp1iQ14, tmpInRe); - yrQ16 = -WEBRTC_SPL_MUL_16_32_RSFT14(tmp1rQ14, tmpInIm2) - WEBRTC_SPL_MUL_16_32_RSFT14(tmp1iQ14, tmpInRe2); - yiQ16 = -WEBRTC_SPL_MUL_16_32_RSFT14(tmp1rQ14, tmpInRe2) + WEBRTC_SPL_MUL_16_32_RSFT14(tmp1iQ14, tmpInIm2); - - /* Combine into one vector, z = x + j * y */ - outre1Q16[k] = xrQ16 - yiQ16; - outre1Q16[FRAMESAMPLES/2 - 1 - k] = xrQ16 + yiQ16; - outre2Q16[k] = xiQ16 + yrQ16; - outre2Q16[FRAMESAMPLES/2 - 1 - k] = -xiQ16 + yrQ16; - } - - /* Get IDFT */ - tmpInRe = WebRtcSpl_MaxAbsValueW32(outre1Q16, 240); - tmpInIm = WebRtcSpl_MaxAbsValueW32(outre2Q16, 240); - if (tmpInIm>tmpInRe) { - tmpInRe = tmpInIm; - } - - sh = WebRtcSpl_NormW32(tmpInRe); - sh = sh-24; //if sh becomes >=0, then we should shift sh steps to the left, and the domain will become Q(16+sh) - //if sh becomes <0, then we should shift -sh steps to the right, and the domain will become Q(16+sh) - - //"Fastest" vectors - if (sh>=0) { - for (k=0; k<240; k++) { - inreQ7[k] = (int16_t)(outre1Q16[k] << sh); // Q(16+sh) - inimQ7[k] = (int16_t)(outre2Q16[k] << sh); // Q(16+sh) - } - } else { - int32_t round = 1 << (-sh - 1); - for (k=0; k<240; k++) { - inreQ7[k] = (int16_t)((outre1Q16[k] + round) >> -sh); // Q(16+sh) - inimQ7[k] = (int16_t)((outre2Q16[k] + round) >> -sh); // Q(16+sh) - } - } - - WebRtcIsacfix_FftRadix16Fastest(inreQ7, inimQ7, 1); // real call - - //"Fastest" vectors - if (sh>=0) { - for (k=0; k<240; k++) { - outre1Q16[k] = inreQ7[k] >> sh; // Q(16+sh) -> Q16 - outre2Q16[k] = inimQ7[k] >> sh; // Q(16+sh) -> Q16 - } - } else { - for (k=0; k<240; k++) { - outre1Q16[k] = inreQ7[k] * (1 << -sh); // Q(16+sh) -> Q16 - outre2Q16[k] = inimQ7[k] * (1 << -sh); // Q(16+sh) -> Q16 - } - } - - /* Divide through by the normalizing constant: */ - /* scale all values with 1/240, i.e. with 273 in Q16 */ - /* 273/65536 ~= 0.0041656 */ - /* 1/240 ~= 0.0041666 */ - for (k=0; k<240; k++) { - outre1Q16[k] = WEBRTC_SPL_MUL_16_32_RSFT16(273, outre1Q16[k]); - outre2Q16[k] = WEBRTC_SPL_MUL_16_32_RSFT16(273, outre2Q16[k]); - } - - /* Demodulate and separate */ - factQ11 = 31727; // sqrt(240) in Q11 is round(15.49193338482967*2048) = 31727 - for (k = 0; k < FRAMESAMPLES/2; k++) { - tmp1rQ14 = WebRtcIsacfix_kCosTab1[k]; - tmp1iQ14 = WebRtcIsacfix_kSinTab1[k]; - xrQ16 = WEBRTC_SPL_MUL_16_32_RSFT14(tmp1rQ14, outre1Q16[k]) - WEBRTC_SPL_MUL_16_32_RSFT14(tmp1iQ14, outre2Q16[k]); - xiQ16 = WEBRTC_SPL_MUL_16_32_RSFT14(tmp1rQ14, outre2Q16[k]) + WEBRTC_SPL_MUL_16_32_RSFT14(tmp1iQ14, outre1Q16[k]); - xrQ16 = WEBRTC_SPL_MUL_16_32_RSFT11(factQ11, xrQ16); - xiQ16 = WEBRTC_SPL_MUL_16_32_RSFT11(factQ11, xiQ16); - outre2Q16[k] = xiQ16; - outre1Q16[k] = xrQ16; - } -} diff --git a/modules/audio_coding/codecs/isac/fix/source/transform_mips.c b/modules/audio_coding/codecs/isac/fix/source/transform_mips.c deleted file mode 100644 index a87b3b54f2..0000000000 --- a/modules/audio_coding/codecs/isac/fix/source/transform_mips.c +++ /dev/null @@ -1,1294 +0,0 @@ -/* - * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#include "modules/audio_coding/codecs/isac/fix/source/codec.h" -#include "modules/audio_coding/codecs/isac/fix/source/fft.h" -#include "modules/audio_coding/codecs/isac/fix/source/settings.h" - -// The tables are defined in transform_tables.c file. -extern const int16_t WebRtcIsacfix_kCosTab1[FRAMESAMPLES/2]; -extern const int16_t WebRtcIsacfix_kSinTab1[FRAMESAMPLES/2]; -extern const int16_t WebRtcIsacfix_kCosTab2[FRAMESAMPLES/4]; -extern const int16_t WebRtcIsacfix_kSinTab2[FRAMESAMPLES/4]; - -// MIPS DSPr2 version of the WebRtcIsacfix_Time2Spec function -// is not bit-exact with the C version. -// The accuracy of the MIPS DSPr2 version is same or better. -void WebRtcIsacfix_Time2SpecMIPS(int16_t* inre1Q9, - int16_t* inre2Q9, - int16_t* outreQ7, - int16_t* outimQ7) { - int k = FRAMESAMPLES / 2; - int32_t tmpreQ16[FRAMESAMPLES / 2], tmpimQ16[FRAMESAMPLES / 2]; - int32_t r0, r1, r2, r3, r4, r5, r6, r7, r8, r9; - int32_t inre1, inre2, tmpre, tmpim, factor, max, max1; - int16_t* cosptr; - int16_t* sinptr; - - cosptr = (int16_t*)WebRtcIsacfix_kCosTab1; - sinptr = (int16_t*)WebRtcIsacfix_kSinTab1; - - __asm __volatile ( - ".set push \n\t" - ".set noreorder \n\t" - "addiu %[inre1], %[inre1Q9], 0 \n\t" - "addiu %[inre2], %[inre2Q9], 0 \n\t" - "addiu %[tmpre], %[tmpreQ16], 0 \n\t" - "addiu %[tmpim], %[tmpimQ16], 0 \n\t" - "addiu %[factor], $zero, 16921 \n\t" - "mul %[max], $zero, $zero \n\t" - // Multiply with complex exponentials and combine into one complex vector. - // Also, calculate the maximal absolute value in the same loop. - "1: \n\t" -#if defined(MIPS_DSP_R2_LE) - "lwl %[r0], 0(%[inre1]) \n\t" - "lwl %[r2], 0(%[cosptr]) \n\t" - "lwl %[r3], 0(%[sinptr]) \n\t" - "lwl %[r1], 0(%[inre2]) \n\t" - "lwr %[r0], 0(%[inre1]) \n\t" - "lwr %[r2], 0(%[cosptr]) \n\t" - "lwr %[r3], 0(%[sinptr]) \n\t" - "lwr %[r1], 0(%[inre2]) \n\t" - "muleq_s.w.phr %[r4], %[r2], %[r0] \n\t" - "muleq_s.w.phr %[r5], %[r3], %[r0] \n\t" - "muleq_s.w.phr %[r6], %[r3], %[r1] \n\t" - "muleq_s.w.phr %[r7], %[r2], %[r1] \n\t" - "muleq_s.w.phl %[r8], %[r2], %[r0] \n\t" - "muleq_s.w.phl %[r0], %[r3], %[r0] \n\t" - "muleq_s.w.phl %[r3], %[r3], %[r1] \n\t" - "muleq_s.w.phl %[r1], %[r2], %[r1] \n\t" - "addiu %[k], %[k], -2 \n\t" - "addu %[r4], %[r4], %[r6] \n\t" - "subu %[r5], %[r7], %[r5] \n\t" - "sra %[r4], %[r4], 8 \n\t" - "sra %[r5], %[r5], 8 \n\t" - "mult $ac0, %[factor], %[r4] \n\t" - "mult $ac1, %[factor], %[r5] \n\t" - "addu %[r3], %[r8], %[r3] \n\t" - "subu %[r0], %[r1], %[r0] \n\t" - "sra %[r3], %[r3], 8 \n\t" - "sra %[r0], %[r0], 8 \n\t" - "mult $ac2, %[factor], %[r3] \n\t" - "mult $ac3, %[factor], %[r0] \n\t" - "extr_r.w %[r4], $ac0, 16 \n\t" - "extr_r.w %[r5], $ac1, 16 \n\t" - "addiu %[inre1], %[inre1], 4 \n\t" - "addiu %[inre2], %[inre2], 4 \n\t" - "extr_r.w %[r6], $ac2, 16 \n\t" - "extr_r.w %[r7], $ac3, 16 \n\t" - "addiu %[cosptr], %[cosptr], 4 \n\t" - "addiu %[sinptr], %[sinptr], 4 \n\t" - "shra_r.w %[r4], %[r4], 3 \n\t" - "shra_r.w %[r5], %[r5], 3 \n\t" - "sw %[r4], 0(%[tmpre]) \n\t" - "absq_s.w %[r4], %[r4] \n\t" - "sw %[r5], 0(%[tmpim]) \n\t" - "absq_s.w %[r5], %[r5] \n\t" - "shra_r.w %[r6], %[r6], 3 \n\t" - "shra_r.w %[r7], %[r7], 3 \n\t" - "sw %[r6], 4(%[tmpre]) \n\t" - "absq_s.w %[r6], %[r6] \n\t" - "sw %[r7], 4(%[tmpim]) \n\t" - "absq_s.w %[r7], %[r7] \n\t" - "slt %[r0], %[r4], %[r5] \n\t" - "movn %[r4], %[r5], %[r0] \n\t" - "slt %[r1], %[r6], %[r7] \n\t" - "movn %[r6], %[r7], %[r1] \n\t" - "slt %[r0], %[max], %[r4] \n\t" - "movn %[max], %[r4], %[r0] \n\t" - "slt %[r1], %[max], %[r6] \n\t" - "movn %[max], %[r6], %[r1] \n\t" - "addiu %[tmpre], %[tmpre], 8 \n\t" - "bgtz %[k], 1b \n\t" - " addiu %[tmpim], %[tmpim], 8 \n\t" -#else // #if defined(MIPS_DSP_R2_LE) - "lh %[r0], 0(%[inre1]) \n\t" - "lh %[r1], 0(%[inre2]) \n\t" - "lh %[r2], 0(%[cosptr]) \n\t" - "lh %[r3], 0(%[sinptr]) \n\t" - "addiu %[k], %[k], -1 \n\t" - "mul %[r4], %[r0], %[r2] \n\t" - "mul %[r5], %[r1], %[r3] \n\t" - "mul %[r0], %[r0], %[r3] \n\t" - "mul %[r2], %[r1], %[r2] \n\t" - "addiu %[inre1], %[inre1], 2 \n\t" - "addiu %[inre2], %[inre2], 2 \n\t" - "addiu %[cosptr], %[cosptr], 2 \n\t" - "addiu %[sinptr], %[sinptr], 2 \n\t" - "addu %[r1], %[r4], %[r5] \n\t" - "sra %[r1], %[r1], 7 \n\t" - "sra %[r3], %[r1], 16 \n\t" - "andi %[r1], %[r1], 0xFFFF \n\t" - "sra %[r1], %[r1], 1 \n\t" - "mul %[r1], %[factor], %[r1] \n\t" - "mul %[r3], %[factor], %[r3] \n\t" - "subu %[r0], %[r2], %[r0] \n\t" - "sra %[r0], %[r0], 7 \n\t" - "sra %[r2], %[r0], 16 \n\t" - "andi %[r0], %[r0], 0xFFFF \n\t" - "sra %[r0], %[r0], 1 \n\t" - "mul %[r0], %[factor], %[r0] \n\t" - "mul %[r2], %[factor], %[r2] \n\t" -#if defined(MIPS_DSP_R1_LE) - "shra_r.w %[r1], %[r1], 15 \n\t" -#else // #if defined(MIPS_DSP_R1_LE) - "addiu %[r1], %[r1], 0x4000 \n\t" - "sra %[r1], %[r1], 15 \n\t" -#endif // #if defined(MIPS_DSP_R1_LE) - "addu %[r1], %[r3], %[r1] \n\t" -#if defined(MIPS_DSP_R1_LE) - "shra_r.w %[r1], %[r1], 3 \n\t" -#else // #if defined(MIPS_DSP_R1_LE) - "addiu %[r1], %[r1], 4 \n\t" - "sra %[r1], %[r1], 3 \n\t" -#endif // #if defined(MIPS_DSP_R1_LE) - "sw %[r1], 0(%[tmpre]) \n\t" - "addiu %[tmpre], %[tmpre], 4 \n\t" -#if defined(MIPS_DSP_R1_LE) - "absq_s.w %[r1], %[r1] \n\t" - "shra_r.w %[r0], %[r0], 15 \n\t" -#else // #if defined(MIPS_DSP_R1_LE) - "negu %[r4], %[r1] \n\t" - "slt %[r3], %[r1], $zero \n\t" - "movn %[r1], %[r4], %[r3] \n\t" - "addiu %[r0], %[r0], 0x4000 \n\t" - "sra %[r0], %[r0], 15 \n\t" -#endif // #if defined(MIPS_DSP_R1_LE) - "addu %[r0], %[r0], %[r2] \n\t" -#if defined(MIPS_DSP_R1_LE) - "shra_r.w %[r0], %[r0], 3 \n\t" - "sw %[r0], 0(%[tmpim]) \n\t" - "absq_s.w %[r0], %[r0] \n\t" -#else // #if defined(MIPS_DSP_R1_LE) - "addiu %[r0], %[r0], 4 \n\t" - "sra %[r0], %[r0], 3 \n\t" - "sw %[r0], 0(%[tmpim]) \n\t" - "negu %[r2], %[r0] \n\t" - "slt %[r3], %[r0], $zero \n\t" - "movn %[r0], %[r2], %[r3] \n\t" -#endif // #if defined(MIPS_DSP_R1_LE) - "slt %[r2], %[max], %[r1] \n\t" - "movn %[max], %[r1], %[r2] \n\t" - "slt %[r2], %[max], %[r0] \n\t" - "movn %[max], %[r0], %[r2] \n\t" - "bgtz %[k], 1b \n\t" - " addiu %[tmpim], %[tmpim], 4 \n\t" -#endif // #if defined(MIPS_DSP_R2_LE) - // Calculate WebRtcSpl_NormW32(max). - // If max gets value >=0, we should shift max steps to the left, and the - // domain will be Q(16+shift). If max gets value <0, we should shift -max - // steps to the right, and the domain will be Q(16+max) - "clz %[max], %[max] \n\t" - "addiu %[max], %[max], -25 \n\t" - ".set pop \n\t" - : [k] "+r" (k), [inre1] "=&r" (inre1), [inre2] "=&r" (inre2), - [r0] "=&r" (r0), [r1] "=&r" (r1), [r2] "=&r" (r2), - [r3] "=&r" (r3), [r4] "=&r" (r4), [tmpre] "=&r" (tmpre), - [tmpim] "=&r" (tmpim), [max] "=&r" (max), [factor] "=&r" (factor), -#if defined(MIPS_DSP_R2_LE) - [r6] "=&r" (r6), [r7] "=&r" (r7), [r8] "=&r" (r8), -#endif // #if defined(MIPS_DSP_R2_LE) - [r5] "=&r" (r5) - : [inre1Q9] "r" (inre1Q9), [inre2Q9] "r" (inre2Q9), - [tmpreQ16] "r" (tmpreQ16), [tmpimQ16] "r" (tmpimQ16), - [cosptr] "r" (cosptr), [sinptr] "r" (sinptr) - : "hi", "lo", "memory" -#if defined(MIPS_DSP_R2_LE) - , "$ac1hi", "$ac1lo", "$ac2hi", "$ac2lo", "$ac3hi", "$ac3lo" -#endif // #if defined(MIPS_DSP_R2_LE) - ); - - // "Fastest" vectors - k = FRAMESAMPLES / 4; - __asm __volatile ( - ".set push \n\t" - ".set noreorder \n\t" - "addiu %[tmpre], %[tmpreQ16], 0 \n\t" - "addiu %[tmpim], %[tmpimQ16], 0 \n\t" - "addiu %[inre1], %[inre1Q9], 0 \n\t" - "addiu %[inre2], %[inre2Q9], 0 \n\t" - "blez %[max], 2f \n\t" - " subu %[max1], $zero, %[max] \n\t" - "1: \n\t" - "lw %[r0], 0(%[tmpre]) \n\t" - "lw %[r1], 0(%[tmpim]) \n\t" - "lw %[r2], 4(%[tmpre]) \n\t" - "lw %[r3], 4(%[tmpim]) \n\t" - "addiu %[k], %[k], -1 \n\t" - "sllv %[r0], %[r0], %[max] \n\t" - "sllv %[r1], %[r1], %[max] \n\t" - "sllv %[r2], %[r2], %[max] \n\t" - "sllv %[r3], %[r3], %[max] \n\t" - "addiu %[tmpre], %[tmpre], 8 \n\t" - "addiu %[tmpim], %[tmpim], 8 \n\t" - "sh %[r0], 0(%[inre1]) \n\t" - "sh %[r1], 0(%[inre2]) \n\t" - "sh %[r2], 2(%[inre1]) \n\t" - "sh %[r3], 2(%[inre2]) \n\t" - "addiu %[inre1], %[inre1], 4 \n\t" - "bgtz %[k], 1b \n\t" - " addiu %[inre2], %[inre2], 4 \n\t" - "b 4f \n\t" - " nop \n\t" - "2: \n\t" -#if !defined(MIPS_DSP_R1_LE) - "addiu %[r4], %[max1], -1 \n\t" - "addiu %[r5], $zero, 1 \n\t" - "sllv %[r4], %[r5], %[r4] \n\t" -#endif // #if !defined(MIPS_DSP_R1_LE) - "3: \n\t" - "lw %[r0], 0(%[tmpre]) \n\t" - "lw %[r1], 0(%[tmpim]) \n\t" - "lw %[r2], 4(%[tmpre]) \n\t" - "lw %[r3], 4(%[tmpim]) \n\t" - "addiu %[k], %[k], -1 \n\t" -#if defined(MIPS_DSP_R1_LE) - "shrav_r.w %[r0], %[r0], %[max1] \n\t" - "shrav_r.w %[r1], %[r1], %[max1] \n\t" - "shrav_r.w %[r2], %[r2], %[max1] \n\t" - "shrav_r.w %[r3], %[r3], %[max1] \n\t" -#else // #if !defined(MIPS_DSP_R1_LE) - "addu %[r0], %[r0], %[r4] \n\t" - "addu %[r1], %[r1], %[r4] \n\t" - "addu %[r2], %[r2], %[r4] \n\t" - "addu %[r3], %[r3], %[r4] \n\t" - "srav %[r0], %[r0], %[max1] \n\t" - "srav %[r1], %[r1], %[max1] \n\t" - "srav %[r2], %[r2], %[max1] \n\t" - "srav %[r3], %[r3], %[max1] \n\t" -#endif // #if !defined(MIPS_DSP_R1_LE) - "addiu %[tmpre], %[tmpre], 8 \n\t" - "addiu %[tmpim], %[tmpim], 8 \n\t" - "sh %[r0], 0(%[inre1]) \n\t" - "sh %[r1], 0(%[inre2]) \n\t" - "sh %[r2], 2(%[inre1]) \n\t" - "sh %[r3], 2(%[inre2]) \n\t" - "addiu %[inre1], %[inre1], 4 \n\t" - "bgtz %[k], 3b \n\t" - " addiu %[inre2], %[inre2], 4 \n\t" - "4: \n\t" - ".set pop \n\t" - : [tmpre] "=&r" (tmpre), [tmpim] "=&r" (tmpim), [inre1] "=&r" (inre1), - [inre2] "=&r" (inre2), [k] "+r" (k), [max1] "=&r" (max1), -#if !defined(MIPS_DSP_R1_LE) - [r4] "=&r" (r4), [r5] "=&r" (r5), -#endif // #if !defined(MIPS_DSP_R1_LE) - [r0] "=&r" (r0), [r1] "=&r" (r1), [r2] "=&r" (r2), [r3] "=&r" (r3) - : [tmpreQ16] "r" (tmpreQ16), [tmpimQ16] "r" (tmpimQ16), - [inre1Q9] "r" (inre1Q9), [inre2Q9] "r" (inre2Q9), [max] "r" (max) - : "memory" - ); - - // Get DFT - WebRtcIsacfix_FftRadix16Fastest(inre1Q9, inre2Q9, -1); // real call - - // "Fastest" vectors and - // Use symmetry to separate into two complex vectors - // and center frames in time around zero - // merged into one loop - cosptr = (int16_t*)WebRtcIsacfix_kCosTab2; - sinptr = (int16_t*)WebRtcIsacfix_kSinTab2; - k = FRAMESAMPLES / 4; - factor = FRAMESAMPLES - 2; // offset for FRAMESAMPLES / 2 - 1 array member - - __asm __volatile ( - ".set push \n\t" - ".set noreorder \n\t" - "addiu %[inre1], %[inre1Q9], 0 \n\t" - "addiu %[inre2], %[inre2Q9], 0 \n\t" - "addiu %[tmpre], %[outreQ7], 0 \n\t" - "addiu %[tmpim], %[outimQ7], 0 \n\t" - "bltz %[max], 2f \n\t" - " subu %[max1], $zero, %[max] \n\t" - "1: \n\t" -#if !defined(MIPS_DSP_R1_LE) - "addu %[r4], %[inre1], %[offset] \n\t" - "addu %[r5], %[inre2], %[offset] \n\t" -#endif // #if !defined(MIPS_DSP_R1_LE) - "lh %[r0], 0(%[inre1]) \n\t" - "lh %[r1], 0(%[inre2]) \n\t" -#if defined(MIPS_DSP_R1_LE) - "lhx %[r2], %[offset](%[inre1]) \n\t" - "lhx %[r3], %[offset](%[inre2]) \n\t" -#else // #if defined(MIPS_DSP_R1_LE) - "lh %[r2], 0(%[r4]) \n\t" - "lh %[r3], 0(%[r5]) \n\t" -#endif // #if defined(MIPS_DSP_R1_LE) - "srav %[r0], %[r0], %[max] \n\t" - "srav %[r1], %[r1], %[max] \n\t" - "srav %[r2], %[r2], %[max] \n\t" - "srav %[r3], %[r3], %[max] \n\t" - "addu %[r4], %[r0], %[r2] \n\t" - "subu %[r0], %[r2], %[r0] \n\t" - "subu %[r2], %[r1], %[r3] \n\t" - "addu %[r1], %[r1], %[r3] \n\t" - "lh %[r3], 0(%[cosptr]) \n\t" - "lh %[r5], 0(%[sinptr]) \n\t" - "andi %[r6], %[r4], 0xFFFF \n\t" - "sra %[r4], %[r4], 16 \n\t" - "mul %[r7], %[r3], %[r6] \n\t" - "mul %[r8], %[r3], %[r4] \n\t" - "mul %[r6], %[r5], %[r6] \n\t" - "mul %[r4], %[r5], %[r4] \n\t" - "addiu %[k], %[k], -1 \n\t" - "addiu %[inre1], %[inre1], 2 \n\t" - "addiu %[inre2], %[inre2], 2 \n\t" -#if defined(MIPS_DSP_R1_LE) - "shra_r.w %[r7], %[r7], 14 \n\t" -#else // #if defined(MIPS_DSP_R1_LE) - "addiu %[r7], %[r7], 0x2000 \n\t" - "sra %[r7], %[r7], 14 \n\t" -#endif // #if defined(MIPS_DSP_R1_LE) - "sll %[r8], %[r8], 2 \n\t" - "addu %[r8], %[r8], %[r7] \n\t" -#if defined(MIPS_DSP_R1_LE) - "shra_r.w %[r6], %[r6], 14 \n\t" -#else // #if defined(MIPS_DSP_R1_LE) - "addiu %[r6], %[r6], 0x2000 \n\t" - "sra %[r6], %[r6], 14 \n\t" -#endif // #if defined(MIPS_DSP_R1_LE) - "sll %[r4], %[r4], 2 \n\t" - "addu %[r4], %[r4], %[r6] \n\t" - "andi %[r6], %[r2], 0xFFFF \n\t" - "sra %[r2], %[r2], 16 \n\t" - "mul %[r7], %[r5], %[r6] \n\t" - "mul %[r9], %[r5], %[r2] \n\t" - "mul %[r6], %[r3], %[r6] \n\t" - "mul %[r2], %[r3], %[r2] \n\t" - "addiu %[cosptr], %[cosptr], 2 \n\t" - "addiu %[sinptr], %[sinptr], 2 \n\t" -#if defined(MIPS_DSP_R1_LE) - "shra_r.w %[r7], %[r7], 14 \n\t" -#else // #if defined(MIPS_DSP_R1_LE) - "addiu %[r7], %[r7], 0x2000 \n\t" - "sra %[r7], %[r7], 14 \n\t" -#endif // #if defined(MIPS_DSP_R1_LE) - "sll %[r9], %[r9], 2 \n\t" - "addu %[r9], %[r7], %[r9] \n\t" -#if defined(MIPS_DSP_R1_LE) - "shra_r.w %[r6], %[r6], 14 \n\t" -#else // #if defined(MIPS_DSP_R1_LE) - "addiu %[r6], %[r6], 0x2000 \n\t" - "sra %[r6], %[r6], 14 \n\t" -#endif // #if defined(MIPS_DSP_R1_LE) - "sll %[r2], %[r2], 2 \n\t" - "addu %[r2], %[r6], %[r2] \n\t" - "subu %[r8], %[r8], %[r9] \n\t" - "sra %[r8], %[r8], 9 \n\t" - "addu %[r2], %[r4], %[r2] \n\t" - "sra %[r2], %[r2], 9 \n\t" - "sh %[r8], 0(%[tmpre]) \n\t" - "sh %[r2], 0(%[tmpim]) \n\t" - - "andi %[r4], %[r1], 0xFFFF \n\t" - "sra %[r1], %[r1], 16 \n\t" - "andi %[r6], %[r0], 0xFFFF \n\t" - "sra %[r0], %[r0], 16 \n\t" - "mul %[r7], %[r5], %[r4] \n\t" - "mul %[r9], %[r5], %[r1] \n\t" - "mul %[r4], %[r3], %[r4] \n\t" - "mul %[r1], %[r3], %[r1] \n\t" - "mul %[r8], %[r3], %[r0] \n\t" - "mul %[r3], %[r3], %[r6] \n\t" - "mul %[r6], %[r5], %[r6] \n\t" - "mul %[r0], %[r5], %[r0] \n\t" -#if defined(MIPS_DSP_R1_LE) - "shra_r.w %[r7], %[r7], 14 \n\t" -#else // #if defined(MIPS_DSP_R1_LE) - "addiu %[r7], %[r7], 0x2000 \n\t" - "sra %[r7], %[r7], 14 \n\t" -#endif // #if defined(MIPS_DSP_R1_LE) - "sll %[r9], %[r9], 2 \n\t" - "addu %[r9], %[r9], %[r7] \n\t" -#if defined(MIPS_DSP_R1_LE) - "shra_r.w %[r4], %[r4], 14 \n\t" -#else // #if defined(MIPS_DSP_R1_LE) - "addiu %[r4], %[r4], 0x2000 \n\t" - "sra %[r4], %[r4], 14 \n\t" -#endif // #if defined(MIPS_DSP_R1_LE) - "sll %[r1], %[r1], 2 \n\t" - "addu %[r1], %[r1], %[r4] \n\t" -#if defined(MIPS_DSP_R1_LE) - "shra_r.w %[r3], %[r3], 14 \n\t" -#else // #if defined(MIPS_DSP_R1_LE) - "addiu %[r3], %[r3], 0x2000 \n\t" - "sra %[r3], %[r3], 14 \n\t" -#endif // #if defined(MIPS_DSP_R1_LE) - "sll %[r8], %[r8], 2 \n\t" - "addu %[r8], %[r8], %[r3] \n\t" -#if defined(MIPS_DSP_R1_LE) - "shra_r.w %[r6], %[r6], 14 \n\t" -#else // #if defined(MIPS_DSP_R1_LE) - "addiu %[r6], %[r6], 0x2000 \n\t" - "sra %[r6], %[r6], 14 \n\t" -#endif // #if defined(MIPS_DSP_R1_LE) - "sll %[r0], %[r0], 2 \n\t" - "addu %[r0], %[r0], %[r6] \n\t" - "addu %[r3], %[tmpre], %[offset] \n\t" - "addu %[r2], %[tmpim], %[offset] \n\t" - "addu %[r9], %[r9], %[r8] \n\t" - "negu %[r9], %[r9] \n\t" - "sra %[r9], %[r9], 9 \n\t" - "subu %[r0], %[r0], %[r1] \n\t" - "addiu %[offset], %[offset], -4 \n\t" - "sh %[r9], 0(%[r3]) \n\t" - "sh %[r0], 0(%[r2]) \n\t" - "addiu %[tmpre], %[tmpre], 2 \n\t" - "bgtz %[k], 1b \n\t" - " addiu %[tmpim], %[tmpim], 2 \n\t" - "b 3f \n\t" - " nop \n\t" - "2: \n\t" -#if !defined(MIPS_DSP_R1_LE) - "addu %[r4], %[inre1], %[offset] \n\t" - "addu %[r5], %[inre2], %[offset] \n\t" -#endif // #if !defined(MIPS_DSP_R1_LE) - "lh %[r0], 0(%[inre1]) \n\t" - "lh %[r1], 0(%[inre2]) \n\t" -#if defined(MIPS_DSP_R1_LE) - "lhx %[r2], %[offset](%[inre1]) \n\t" - "lhx %[r3], %[offset](%[inre2]) \n\t" -#else // #if defined(MIPS_DSP_R1_LE) - "lh %[r2], 0(%[r4]) \n\t" - "lh %[r3], 0(%[r5]) \n\t" -#endif // #if defined(MIPS_DSP_R1_LE) - "sllv %[r0], %[r0], %[max1] \n\t" - "sllv %[r1], %[r1], %[max1] \n\t" - "sllv %[r2], %[r2], %[max1] \n\t" - "sllv %[r3], %[r3], %[max1] \n\t" - "addu %[r4], %[r0], %[r2] \n\t" - "subu %[r0], %[r2], %[r0] \n\t" - "subu %[r2], %[r1], %[r3] \n\t" - "addu %[r1], %[r1], %[r3] \n\t" - "lh %[r3], 0(%[cosptr]) \n\t" - "lh %[r5], 0(%[sinptr]) \n\t" - "andi %[r6], %[r4], 0xFFFF \n\t" - "sra %[r4], %[r4], 16 \n\t" - "mul %[r7], %[r3], %[r6] \n\t" - "mul %[r8], %[r3], %[r4] \n\t" - "mul %[r6], %[r5], %[r6] \n\t" - "mul %[r4], %[r5], %[r4] \n\t" - "addiu %[k], %[k], -1 \n\t" - "addiu %[inre1], %[inre1], 2 \n\t" - "addiu %[inre2], %[inre2], 2 \n\t" -#if defined(MIPS_DSP_R1_LE) - "shra_r.w %[r7], %[r7], 14 \n\t" -#else // #if defined(MIPS_DSP_R1_LE) - "addiu %[r7], %[r7], 0x2000 \n\t" - "sra %[r7], %[r7], 14 \n\t" -#endif // #if defined(MIPS_DSP_R1_LE) - "sll %[r8], %[r8], 2 \n\t" - "addu %[r8], %[r8], %[r7] \n\t" -#if defined(MIPS_DSP_R1_LE) - "shra_r.w %[r6], %[r6], 14 \n\t" -#else // #if defined(MIPS_DSP_R1_LE) - "addiu %[r6], %[r6], 0x2000 \n\t" - "sra %[r6], %[r6], 14 \n\t" -#endif // #if defined(MIPS_DSP_R1_LE) - "sll %[r4], %[r4], 2 \n\t" - "addu %[r4], %[r4], %[r6] \n\t" - "andi %[r6], %[r2], 0xFFFF \n\t" - "sra %[r2], %[r2], 16 \n\t" - "mul %[r7], %[r5], %[r6] \n\t" - "mul %[r9], %[r5], %[r2] \n\t" - "mul %[r6], %[r3], %[r6] \n\t" - "mul %[r2], %[r3], %[r2] \n\t" - "addiu %[cosptr], %[cosptr], 2 \n\t" - "addiu %[sinptr], %[sinptr], 2 \n\t" -#if defined(MIPS_DSP_R1_LE) - "shra_r.w %[r7], %[r7], 14 \n\t" -#else // #if defined(MIPS_DSP_R1_LE) - "addiu %[r7], %[r7], 0x2000 \n\t" - "sra %[r7], %[r7], 14 \n\t" -#endif // #if defined(MIPS_DSP_R1_LE) - "sll %[r9], %[r9], 2 \n\t" - "addu %[r9], %[r7], %[r9] \n\t" -#if defined(MIPS_DSP_R1_LE) - "shra_r.w %[r6], %[r6], 14 \n\t" -#else // #if defined(MIPS_DSP_R1_LE) - "addiu %[r6], %[r6], 0x2000 \n\t" - "sra %[r6], %[r6], 14 \n\t" -#endif // #if defined(MIPS_DSP_R1_LE) - "sll %[r2], %[r2], 2 \n\t" - "addu %[r2], %[r6], %[r2] \n\t" - "subu %[r8], %[r8], %[r9] \n\t" - "sra %[r8], %[r8], 9 \n\t" - "addu %[r2], %[r4], %[r2] \n\t" - "sra %[r2], %[r2], 9 \n\t" - "sh %[r8], 0(%[tmpre]) \n\t" - "sh %[r2], 0(%[tmpim]) \n\t" - "andi %[r4], %[r1], 0xFFFF \n\t" - "sra %[r1], %[r1], 16 \n\t" - "andi %[r6], %[r0], 0xFFFF \n\t" - "sra %[r0], %[r0], 16 \n\t" - "mul %[r7], %[r5], %[r4] \n\t" - "mul %[r9], %[r5], %[r1] \n\t" - "mul %[r4], %[r3], %[r4] \n\t" - "mul %[r1], %[r3], %[r1] \n\t" - "mul %[r8], %[r3], %[r0] \n\t" - "mul %[r3], %[r3], %[r6] \n\t" - "mul %[r6], %[r5], %[r6] \n\t" - "mul %[r0], %[r5], %[r0] \n\t" -#if defined(MIPS_DSP_R1_LE) - "shra_r.w %[r7], %[r7], 14 \n\t" -#else // #if defined(MIPS_DSP_R1_LE) - "addiu %[r7], %[r7], 0x2000 \n\t" - "sra %[r7], %[r7], 14 \n\t" -#endif // #if defined(MIPS_DSP_R1_LE) - "sll %[r9], %[r9], 2 \n\t" - "addu %[r9], %[r9], %[r7] \n\t" -#if defined(MIPS_DSP_R1_LE) - "shra_r.w %[r4], %[r4], 14 \n\t" -#else // #if defined(MIPS_DSP_R1_LE) - "addiu %[r4], %[r4], 0x2000 \n\t" - "sra %[r4], %[r4], 14 \n\t" -#endif - "sll %[r1], %[r1], 2 \n\t" - "addu %[r1], %[r1], %[r4] \n\t" -#if defined(MIPS_DSP_R1_LE) - "shra_r.w %[r3], %[r3], 14 \n\t" -#else // #if defined(MIPS_DSP_R1_LE) - "addiu %[r3], %[r3], 0x2000 \n\t" - "sra %[r3], %[r3], 14 \n\t" -#endif // #if defined(MIPS_DSP_R1_LE) - "sll %[r8], %[r8], 2 \n\t" - "addu %[r8], %[r8], %[r3] \n\t" -#if defined(MIPS_DSP_R1_LE) - "shra_r.w %[r6], %[r6], 14 \n\t" -#else // #if defined(MIPS_DSP_R1_LE) - "addiu %[r6], %[r6], 0x2000 \n\t" - "sra %[r6], %[r6], 14 \n\t" -#endif // #if defined(MIPS_DSP_R1_LE) - "sll %[r0], %[r0], 2 \n\t" - "addu %[r0], %[r0], %[r6] \n\t" - "addu %[r3], %[tmpre], %[offset] \n\t" - "addu %[r2], %[tmpim], %[offset] \n\t" - "addu %[r9], %[r9], %[r8] \n\t" - "negu %[r9], %[r9] \n\t" - "sra %[r9], %[r9], 9 \n\t" - "subu %[r0], %[r0], %[r1] \n\t" - "sra %[r0], %[r0], 9 \n\t" - "addiu %[offset], %[offset], -4 \n\t" - "sh %[r9], 0(%[r3]) \n\t" - "sh %[r0], 0(%[r2]) \n\t" - "addiu %[tmpre], %[tmpre], 2 \n\t" - "bgtz %[k], 2b \n\t" - " addiu %[tmpim], %[tmpim], 2 \n\t" - "3: \n\t" - ".set pop \n\t" - : [inre1] "=&r" (inre1), [inre2] "=&r" (inre2), [tmpre] "=&r" (tmpre), - [tmpim] "=&r" (tmpim), [offset] "+r" (factor), [k] "+r" (k), - [r0] "=&r" (r0), [r1] "=&r" (r1), [r2] "=&r" (r2), [r3] "=&r" (r3), - [r4] "=&r" (r4), [r5] "=&r" (r5), [r6] "=&r" (r6), [r7] "=&r" (r7), - [r8] "=&r" (r8), [r9] "=&r" (r9), [max1] "=&r" (max1) - : [inre1Q9] "r" (inre1Q9), [inre2Q9] "r" (inre2Q9), - [outreQ7] "r" (outreQ7), [outimQ7] "r" (outimQ7), - [max] "r" (max), [cosptr] "r" (cosptr), [sinptr] "r" (sinptr) - : "hi", "lo", "memory" - ); -} - -void WebRtcIsacfix_Spec2TimeMIPS(int16_t *inreQ7, - int16_t *inimQ7, - int32_t *outre1Q16, - int32_t *outre2Q16) { - int k = FRAMESAMPLES / 4; - int16_t* inre; - int16_t* inim; - int32_t* outre1; - int32_t* outre2; - int16_t* cosptr = (int16_t*)WebRtcIsacfix_kCosTab2; - int16_t* sinptr = (int16_t*)WebRtcIsacfix_kSinTab2; - int32_t r0, r1, r2, r3, r4, r5, r6, r7, r8, r9, max, max1; -#if defined(MIPS_DSP_R1_LE) - int32_t offset = FRAMESAMPLES - 4; -#else // #if defined(MIPS_DSP_R1_LE) - int32_t offset = FRAMESAMPLES - 2; -#endif // #if defined(MIPS_DSP_R1_LE) - - __asm __volatile ( - ".set push \n\t" - ".set noreorder \n\t" - "addiu %[inre], %[inreQ7], 0 \n\t" - "addiu %[inim] , %[inimQ7], 0 \n\t" - "addiu %[outre1], %[outre1Q16], 0 \n\t" - "addiu %[outre2], %[outre2Q16], 0 \n\t" - "mul %[max], $zero, $zero \n\t" - "1: \n\t" -#if defined(MIPS_DSP_R1_LE) - // Process two samples in one iteration avoiding left shift before - // multiplication. MaxAbsValueW32 function inlined into the loop. - "addu %[r8], %[inre], %[offset] \n\t" - "addu %[r9], %[inim], %[offset] \n\t" - "lwl %[r4], 0(%[r8]) \n\t" - "lwl %[r5], 0(%[r9]) \n\t" - "lwl %[r0], 0(%[inre]) \n\t" - "lwl %[r1], 0(%[inim]) \n\t" - "lwl %[r2], 0(%[cosptr]) \n\t" - "lwl %[r3], 0(%[sinptr]) \n\t" - "lwr %[r4], 0(%[r8]) \n\t" - "lwr %[r5], 0(%[r9]) \n\t" - "lwr %[r0], 0(%[inre]) \n\t" - "lwr %[r1], 0(%[inim]) \n\t" - "lwr %[r2], 0(%[cosptr]) \n\t" - "lwr %[r3], 0(%[sinptr]) \n\t" - "packrl.ph %[r4], %[r4], %[r4] \n\t" - "packrl.ph %[r5], %[r5], %[r5] \n\t" - "muleq_s.w.phr %[r6], %[r0], %[r2] \n\t" - "muleq_s.w.phr %[r7], %[r1], %[r3] \n\t" - "muleq_s.w.phr %[r8], %[r4], %[r2] \n\t" - "muleq_s.w.phr %[r9], %[r5], %[r3] \n\t" - "addiu %[k], %[k], -2 \n\t" - "addiu %[cosptr], %[cosptr], 4 \n\t" - "addiu %[sinptr], %[sinptr], 4 \n\t" - "addiu %[inre], %[inre], 4 \n\t" - "addiu %[inim], %[inim], 4 \n\t" - "shra_r.w %[r6], %[r6], 6 \n\t" - "shra_r.w %[r7], %[r7], 6 \n\t" - "shra_r.w %[r8], %[r8], 6 \n\t" - "shra_r.w %[r9], %[r9], 6 \n\t" - "addu %[r6], %[r6], %[r7] \n\t" - "subu %[r9], %[r9], %[r8] \n\t" - "subu %[r7], %[r6], %[r9] \n\t" - "addu %[r6], %[r6], %[r9] \n\t" - "sw %[r7], 0(%[outre1]) \n\t" - "absq_s.w %[r7], %[r7] \n\t" - "slt %[r8], %[max], %[r7] \n\t" - "movn %[max], %[r7], %[r8] \n\t" - "sll %[r7], %[offset], 1 \n\t" - "addu %[r7], %[outre1], %[r7] \n\t" - "sw %[r6], 4(%[r7]) \n\t" - "absq_s.w %[r6], %[r6] \n\t" - "slt %[r8], %[max], %[r6] \n\t" - "movn %[max], %[r6], %[r8] \n\t" - "muleq_s.w.phl %[r6], %[r0], %[r2] \n\t" - "muleq_s.w.phl %[r7], %[r1], %[r3] \n\t" - "muleq_s.w.phl %[r8], %[r4], %[r2] \n\t" - "muleq_s.w.phl %[r9], %[r5], %[r3] \n\t" - "shra_r.w %[r6], %[r6], 6 \n\t" - "shra_r.w %[r7], %[r7], 6 \n\t" - "shra_r.w %[r8], %[r8], 6 \n\t" - "shra_r.w %[r9], %[r9], 6 \n\t" - "addu %[r6], %[r6], %[r7] \n\t" - "subu %[r9], %[r9], %[r8] \n\t" - "subu %[r7], %[r6], %[r9] \n\t" - "addu %[r6], %[r6], %[r9] \n\t" - "sw %[r7], 4(%[outre1]) \n\t" - "absq_s.w %[r7], %[r7] \n\t" - "slt %[r8], %[max], %[r7] \n\t" - "movn %[max], %[r7], %[r8] \n\t" - "sll %[r7], %[offset], 1 \n\t" - "addu %[r7], %[outre1], %[r7] \n\t" - "sw %[r6], 0(%[r7]) \n\t" - "absq_s.w %[r6], %[r6] \n\t" - "slt %[r8], %[max], %[r6] \n\t" - "movn %[max], %[r6], %[r8] \n\t" - "muleq_s.w.phr %[r6], %[r1], %[r2] \n\t" - "muleq_s.w.phr %[r7], %[r0], %[r3] \n\t" - "muleq_s.w.phr %[r8], %[r5], %[r2] \n\t" - "muleq_s.w.phr %[r9], %[r4], %[r3] \n\t" - "addiu %[outre1], %[outre1], 8 \n\t" - "shra_r.w %[r6], %[r6], 6 \n\t" - "shra_r.w %[r7], %[r7], 6 \n\t" - "shra_r.w %[r8], %[r8], 6 \n\t" - "shra_r.w %[r9], %[r9], 6 \n\t" - "subu %[r6], %[r6], %[r7] \n\t" - "addu %[r9], %[r9], %[r8] \n\t" - "subu %[r7], %[r6], %[r9] \n\t" - "addu %[r6], %[r9], %[r6] \n\t" - "negu %[r6], %[r6] \n\t" - "sw %[r7], 0(%[outre2]) \n\t" - "absq_s.w %[r7], %[r7] \n\t" - "slt %[r8], %[max], %[r7] \n\t" - "movn %[max], %[r7], %[r8] \n\t" - "sll %[r7], %[offset], 1 \n\t" - "addu %[r7], %[outre2], %[r7] \n\t" - "sw %[r6], 4(%[r7]) \n\t" - "absq_s.w %[r6], %[r6] \n\t" - "slt %[r8], %[max], %[r6] \n\t" - "movn %[max], %[r6], %[r8] \n\t" - "muleq_s.w.phl %[r6], %[r1], %[r2] \n\t" - "muleq_s.w.phl %[r7], %[r0], %[r3] \n\t" - "muleq_s.w.phl %[r8], %[r5], %[r2] \n\t" - "muleq_s.w.phl %[r9], %[r4], %[r3] \n\t" - "addiu %[offset], %[offset], -8 \n\t" - "shra_r.w %[r6], %[r6], 6 \n\t" - "shra_r.w %[r7], %[r7], 6 \n\t" - "shra_r.w %[r8], %[r8], 6 \n\t" - "shra_r.w %[r9], %[r9], 6 \n\t" - "subu %[r6], %[r6], %[r7] \n\t" - "addu %[r9], %[r9], %[r8] \n\t" - "subu %[r7], %[r6], %[r9] \n\t" - "addu %[r6], %[r9], %[r6] \n\t" - "negu %[r6], %[r6] \n\t" - "sw %[r7], 4(%[outre2]) \n\t" - "absq_s.w %[r7], %[r7] \n\t" - "slt %[r8], %[max], %[r7] \n\t" - "movn %[max], %[r7], %[r8] \n\t" - "sll %[r7], %[offset], 1 \n\t" - "addu %[r7], %[outre2], %[r7] \n\t" - "sw %[r6], 0(%[r7]) \n\t" - "absq_s.w %[r6], %[r6] \n\t" - "slt %[r8], %[max], %[r6] \n\t" - "movn %[max], %[r6], %[r8] \n\t" - "bgtz %[k], 1b \n\t" - " addiu %[outre2], %[outre2], 8 \n\t" -#else // #if defined(MIPS_DSP_R1_LE) - "lh %[r0], 0(%[inre]) \n\t" - "lh %[r1], 0(%[inim]) \n\t" - "lh %[r4], 0(%[cosptr]) \n\t" - "lh %[r5], 0(%[sinptr]) \n\t" - "addiu %[k], %[k], -1 \n\t" - "mul %[r2], %[r0], %[r4] \n\t" - "mul %[r0], %[r0], %[r5] \n\t" - "mul %[r3], %[r1], %[r5] \n\t" - "mul %[r1], %[r1], %[r4] \n\t" - "addiu %[cosptr], %[cosptr], 2 \n\t" - "addiu %[sinptr], %[sinptr], 2 \n\t" - "addu %[r8], %[inre], %[offset] \n\t" - "addu %[r9], %[inim], %[offset] \n\t" - "addiu %[r2], %[r2], 16 \n\t" - "sra %[r2], %[r2], 5 \n\t" - "addiu %[r0], %[r0], 16 \n\t" - "sra %[r0], %[r0], 5 \n\t" - "addiu %[r3], %[r3], 16 \n\t" - "sra %[r3], %[r3], 5 \n\t" - "lh %[r6], 0(%[r8]) \n\t" - "lh %[r7], 0(%[r9]) \n\t" - "addiu %[r1], %[r1], 16 \n\t" - "sra %[r1], %[r1], 5 \n\t" - "mul %[r8], %[r7], %[r4] \n\t" - "mul %[r7], %[r7], %[r5] \n\t" - "mul %[r9], %[r6], %[r4] \n\t" - "mul %[r6], %[r6], %[r5] \n\t" - "addu %[r2], %[r2], %[r3] \n\t" - "subu %[r1], %[r1], %[r0] \n\t" - "sll %[r0], %[offset], 1 \n\t" - "addu %[r4], %[outre1], %[r0] \n\t" - "addu %[r5], %[outre2], %[r0] \n\t" - "addiu %[r8], %[r8], 16 \n\t" - "sra %[r8], %[r8], 5 \n\t" - "addiu %[r7], %[r7], 16 \n\t" - "sra %[r7], %[r7], 5 \n\t" - "addiu %[r6], %[r6], 16 \n\t" - "sra %[r6], %[r6], 5 \n\t" - "addiu %[r9], %[r9], 16 \n\t" - "sra %[r9], %[r9], 5 \n\t" - "addu %[r8], %[r8], %[r6] \n\t" - "negu %[r8], %[r8] \n\t" - "subu %[r7], %[r7], %[r9] \n\t" - "subu %[r6], %[r2], %[r7] \n\t" - "addu %[r0], %[r2], %[r7] \n\t" - "addu %[r3], %[r1], %[r8] \n\t" - "subu %[r1], %[r8], %[r1] \n\t" - "sw %[r6], 0(%[outre1]) \n\t" - "sw %[r0], 0(%[r4]) \n\t" - "sw %[r3], 0(%[outre2]) \n\t" - "sw %[r1], 0(%[r5]) \n\t" - "addiu %[outre1], %[outre1], 4 \n\t" - "addiu %[offset], %[offset], -4 \n\t" - "addiu %[inre], %[inre], 2 \n\t" - "addiu %[inim], %[inim], 2 \n\t" - // Inlined WebRtcSpl_MaxAbsValueW32 - "negu %[r5], %[r6] \n\t" - "slt %[r2], %[r6], $zero \n\t" - "movn %[r6], %[r5], %[r2] \n\t" - "negu %[r5], %[r0] \n\t" - "slt %[r2], %[r0], $zero \n\t" - "movn %[r0], %[r5], %[r2] \n\t" - "negu %[r5], %[r3] \n\t" - "slt %[r2], %[r3], $zero \n\t" - "movn %[r3], %[r5], %[r2] \n\t" - "negu %[r5], %[r1] \n\t" - "slt %[r2], %[r1], $zero \n\t" - "movn %[r1], %[r5], %[r2] \n\t" - "slt %[r2], %[r6], %[r0] \n\t" - "slt %[r5], %[r3], %[r1] \n\t" - "movn %[r6], %[r0], %[r2] \n\t" - "movn %[r3], %[r1], %[r5] \n\t" - "slt %[r2], %[r6], %[r3] \n\t" - "movn %[r6], %[r3], %[r2] \n\t" - "slt %[r2], %[max], %[r6] \n\t" - "movn %[max], %[r6], %[r2] \n\t" - "bgtz %[k], 1b \n\t" - " addiu %[outre2], %[outre2], 4 \n\t" -#endif // #if defined(MIPS_DSP_R1_LE) - "clz %[max], %[max] \n\t" - "addiu %[max], %[max], -25 \n\t" - ".set pop \n\t" - : [inre] "=&r" (inre), [inim] "=&r" (inim), - [outre1] "=&r" (outre1), [outre2] "=&r" (outre2), - [offset] "+r" (offset), [k] "+r" (k), [r0] "=&r" (r0), - [r1] "=&r" (r1), [r2] "=&r" (r2), [r3] "=&r" (r3), - [r4] "=&r" (r4), [r5] "=&r" (r5), [r6] "=&r" (r6), - [r7] "=&r" (r7), [r8] "=&r" (r8), [r9] "=&r" (r9), - [max] "=&r" (max) - : [inreQ7] "r" (inreQ7), [inimQ7] "r" (inimQ7), - [cosptr] "r" (cosptr), [sinptr] "r" (sinptr), - [outre1Q16] "r" (outre1Q16), [outre2Q16] "r" (outre2Q16) - : "hi", "lo", "memory" - ); - - // "Fastest" vectors - k = FRAMESAMPLES / 4; - __asm __volatile ( - ".set push \n\t" - ".set noreorder \n\t" - "addiu %[inre], %[inreQ7], 0 \n\t" - "addiu %[inim], %[inimQ7], 0 \n\t" - "addiu %[outre1], %[outre1Q16], 0 \n\t" - "addiu %[outre2], %[outre2Q16], 0 \n\t" - "bltz %[max], 2f \n\t" - " subu %[max1], $zero, %[max] \n\t" - "1: \n\t" - "lw %[r0], 0(%[outre1]) \n\t" - "lw %[r1], 0(%[outre2]) \n\t" - "lw %[r2], 4(%[outre1]) \n\t" - "lw %[r3], 4(%[outre2]) \n\t" - "sllv %[r0], %[r0], %[max] \n\t" - "sllv %[r1], %[r1], %[max] \n\t" - "sllv %[r2], %[r2], %[max] \n\t" - "sllv %[r3], %[r3], %[max] \n\t" - "addiu %[k], %[k], -1 \n\t" - "addiu %[outre1], %[outre1], 8 \n\t" - "addiu %[outre2], %[outre2], 8 \n\t" - "sh %[r0], 0(%[inre]) \n\t" - "sh %[r1], 0(%[inim]) \n\t" - "sh %[r2], 2(%[inre]) \n\t" - "sh %[r3], 2(%[inim]) \n\t" - "addiu %[inre], %[inre], 4 \n\t" - "bgtz %[k], 1b \n\t" - " addiu %[inim], %[inim], 4 \n\t" - "b 4f \n\t" - " nop \n\t" - "2: \n\t" -#if !defined(MIPS_DSP_R1_LE) - "addiu %[r4], $zero, 1 \n\t" - "addiu %[r5], %[max1], -1 \n\t" - "sllv %[r4], %[r4], %[r5] \n\t" -#endif // #if !defined(MIPS_DSP_R1_LE) - "3: \n\t" - "lw %[r0], 0(%[outre1]) \n\t" - "lw %[r1], 0(%[outre2]) \n\t" - "lw %[r2], 4(%[outre1]) \n\t" - "lw %[r3], 4(%[outre2]) \n\t" -#if defined(MIPS_DSP_R1_LE) - "shrav_r.w %[r0], %[r0], %[max1] \n\t" - "shrav_r.w %[r1], %[r1], %[max1] \n\t" - "shrav_r.w %[r2], %[r2], %[max1] \n\t" - "shrav_r.w %[r3], %[r3], %[max1] \n\t" -#else // #if defined(MIPS_DSP_R1_LE) - "addu %[r0], %[r0], %[r4] \n\t" - "addu %[r1], %[r1], %[r4] \n\t" - "addu %[r2], %[r2], %[r4] \n\t" - "addu %[r3], %[r3], %[r4] \n\t" - "srav %[r0], %[r0], %[max1] \n\t" - "srav %[r1], %[r1], %[max1] \n\t" - "srav %[r2], %[r2], %[max1] \n\t" - "srav %[r3], %[r3], %[max1] \n\t" -#endif // #if defined(MIPS_DSP_R1_LE) - "addiu %[outre1], %[outre1], 8 \n\t" - "addiu %[outre2], %[outre2], 8 \n\t" - "sh %[r0], 0(%[inre]) \n\t" - "sh %[r1], 0(%[inim]) \n\t" - "sh %[r2], 2(%[inre]) \n\t" - "sh %[r3], 2(%[inim]) \n\t" - "addiu %[k], %[k], -1 \n\t" - "addiu %[inre], %[inre], 4 \n\t" - "bgtz %[k], 3b \n\t" - " addiu %[inim], %[inim], 4 \n\t" - "4: \n\t" - ".set pop \n\t" - : [k] "+r" (k), [max1] "=&r" (max1), [r0] "=&r" (r0), - [inre] "=&r" (inre), [inim] "=&r" (inim), - [outre1] "=&r" (outre1), [outre2] "=&r" (outre2), -#if !defined(MIPS_DSP_R1_LE) - [r4] "=&r" (r4), [r5] "=&r" (r5), -#endif // #if !defined(MIPS_DSP_R1_LE) - [r1] "=&r" (r1), [r2] "=&r" (r2), [r3] "=&r" (r3) - : [max] "r" (max), [inreQ7] "r" (inreQ7), - [inimQ7] "r" (inimQ7), [outre1Q16] "r" (outre1Q16), - [outre2Q16] "r" (outre2Q16) - : "memory" - ); - - WebRtcIsacfix_FftRadix16Fastest(inreQ7, inimQ7, 1); // real call - - // All the remaining processing is done inside a single loop to avoid - // unnecessary memory accesses. MIPS DSPr2 version processes two samples - // at a time. - cosptr = (int16_t*)WebRtcIsacfix_kCosTab1; - sinptr = (int16_t*)WebRtcIsacfix_kSinTab1; - k = FRAMESAMPLES / 2; - __asm __volatile ( - ".set push \n\t" - ".set noreorder \n\t" - "addiu %[inre], %[inreQ7], 0 \n\t" - "addiu %[inim], %[inimQ7], 0 \n\t" - "addiu %[outre1], %[outre1Q16], 0 \n\t" - "addiu %[outre2], %[outre2Q16], 0 \n\t" - "addiu %[r4], $zero, 273 \n\t" - "addiu %[r5], $zero, 31727 \n\t" -#if defined(MIPS_DSP_R2_LE) - "addiu %[max], %[max], 16 \n\t" - "replv.ph %[r4], %[r4] \n\t" -#endif // #if defined(MIPS_DSP_R2_LE) - "bltz %[max], 2f \n\t" - " subu %[max1], $zero, %[max] \n\t" -#if defined(MIPS_DSP_R2_LE) - "addiu %[max], %[max], 1 \n\t" -#endif // #if defined(MIPS_DSP_R2_LE) - "1: \n\t" -#if defined(MIPS_DSP_R2_LE) - "lwl %[r0], 0(%[inre]) \n\t" - "lwl %[r1], 0(%[inim]) \n\t" - "lh %[r2], 0(%[cosptr]) \n\t" - "lwr %[r0], 0(%[inre]) \n\t" - "lwr %[r1], 0(%[inim]) \n\t" - "lh %[r3], 0(%[sinptr]) \n\t" - "muleq_s.w.phr %[r6], %[r0], %[r4] \n\t" - "muleq_s.w.phr %[r7], %[r1], %[r4] \n\t" - "muleq_s.w.phl %[r0], %[r0], %[r4] \n\t" - "muleq_s.w.phl %[r1], %[r1], %[r4] \n\t" - "addiu %[k], %[k], -2 \n\t" - "addiu %[inre], %[inre], 4 \n\t" - "addiu %[inim], %[inim], 4 \n\t" - "shrav_r.w %[r6], %[r6], %[max] \n\t" - "shrav_r.w %[r7], %[r7], %[max] \n\t" - "mult $ac0, %[r2], %[r6] \n\t" - "mult $ac1, %[r3], %[r7] \n\t" - "mult $ac2, %[r2], %[r7] \n\t" - "mult $ac3, %[r3], %[r6] \n\t" - "lh %[r2], 2(%[cosptr]) \n\t" - "lh %[r3], 2(%[sinptr]) \n\t" - "extr_r.w %[r6], $ac0, 14 \n\t" - "extr_r.w %[r7], $ac1, 14 \n\t" - "extr_r.w %[r8], $ac2, 14 \n\t" - "extr_r.w %[r9], $ac3, 14 \n\t" - "shrav_r.w %[r0], %[r0], %[max] \n\t" - "shrav_r.w %[r1], %[r1], %[max] \n\t" - "mult $ac0, %[r2], %[r0] \n\t" - "mult $ac1, %[r3], %[r1] \n\t" - "mult $ac2, %[r2], %[r1] \n\t" - "mult $ac3, %[r3], %[r0] \n\t" - "addiu %[cosptr], %[cosptr], 4 \n\t" - "extr_r.w %[r0], $ac0, 14 \n\t" - "extr_r.w %[r1], $ac1, 14 \n\t" - "extr_r.w %[r2], $ac2, 14 \n\t" - "extr_r.w %[r3], $ac3, 14 \n\t" - "subu %[r6], %[r6], %[r7] \n\t" - "addu %[r8], %[r8], %[r9] \n\t" - "mult $ac0, %[r5], %[r6] \n\t" - "mult $ac1, %[r5], %[r8] \n\t" - "addiu %[sinptr], %[sinptr], 4 \n\t" - "subu %[r0], %[r0], %[r1] \n\t" - "addu %[r2], %[r2], %[r3] \n\t" - "extr_r.w %[r1], $ac0, 11 \n\t" - "extr_r.w %[r3], $ac1, 11 \n\t" - "mult $ac2, %[r5], %[r0] \n\t" - "mult $ac3, %[r5], %[r2] \n\t" - "sw %[r1], 0(%[outre1]) \n\t" - "sw %[r3], 0(%[outre2]) \n\t" - "addiu %[outre1], %[outre1], 8 \n\t" - "extr_r.w %[r0], $ac2, 11 \n\t" - "extr_r.w %[r2], $ac3, 11 \n\t" - "sw %[r0], -4(%[outre1]) \n\t" - "sw %[r2], 4(%[outre2]) \n\t" - "bgtz %[k], 1b \n\t" - " addiu %[outre2], %[outre2], 8 \n\t" - "b 3f \n\t" -#else // #if defined(MIPS_DSP_R2_LE) - "lh %[r0], 0(%[inre]) \n\t" - "lh %[r1], 0(%[inim]) \n\t" - "addiu %[k], %[k], -1 \n\t" - "srav %[r0], %[r0], %[max] \n\t" - "srav %[r1], %[r1], %[max] \n\t" - "sra %[r2], %[r0], 16 \n\t" - "andi %[r0], %[r0], 0xFFFF \n\t" - "sra %[r0], %[r0], 1 \n\t" - "sra %[r3], %[r1], 16 \n\t" - "andi %[r1], %[r1], 0xFFFF \n\t" - "sra %[r1], %[r1], 1 \n\t" - "mul %[r2], %[r2], %[r4] \n\t" - "mul %[r0], %[r0], %[r4] \n\t" - "mul %[r3], %[r3], %[r4] \n\t" - "mul %[r1], %[r1], %[r4] \n\t" - "addiu %[inre], %[inre], 2 \n\t" - "addiu %[inim], %[inim], 2 \n\t" - "lh %[r6], 0(%[cosptr]) \n\t" - "lh %[r7], 0(%[sinptr]) \n\t" -#if defined(MIPS_DSP_R1_LE) - "shra_r.w %[r0], %[r0], 15 \n\t" - "shra_r.w %[r1], %[r1], 15 \n\t" -#else // #if defined(MIPS_DSP_R1_LE) - "addiu %[r0], %[r0], 0x4000 \n\t" - "addiu %[r1], %[r1], 0x4000 \n\t" - "sra %[r0], %[r0], 15 \n\t" - "sra %[r1], %[r1], 15 \n\t" -#endif // #if defined(MIPS_DSP_R1_LE) - "addu %[r0], %[r2], %[r0] \n\t" - "addu %[r1], %[r3], %[r1] \n\t" - "sra %[r2], %[r0], 16 \n\t" - "andi %[r0], %[r0], 0xFFFF \n\t" - "mul %[r9], %[r2], %[r6] \n\t" - "mul %[r2], %[r2], %[r7] \n\t" - "mul %[r8], %[r0], %[r6] \n\t" - "mul %[r0], %[r0], %[r7] \n\t" - "sra %[r3], %[r3], 16 \n\t" - "andi %[r1], %[r1], 0xFFFF \n\t" - "sll %[r9], %[r9], 2 \n\t" - "sll %[r2], %[r2], 2 \n\t" -#if defined(MIPS_DSP_R1_LE) - "shra_r.w %[r8], %[r8], 14 \n\t" - "shra_r.w %[r0], %[r0], 14 \n\t" -#else // #if defined(MIPS_DSP_R1_LE) - "addiu %[r8], %[r8], 0x2000 \n\t" - "addiu %[r0], %[r0], 0x2000 \n\t" - "sra %[r8], %[r8], 14 \n\t" - "sra %[r0], %[r0], 14 \n\t" -#endif // #if defined(MIPS_DSP_R1_LE) - "addu %[r9], %[r9], %[r8] \n\t" - "addu %[r2], %[r2], %[r0] \n\t" - "mul %[r0], %[r3], %[r6] \n\t" - "mul %[r3], %[r3], %[r7] \n\t" - "mul %[r8], %[r1], %[r6] \n\t" - "mul %[r1], %[r1], %[r8] \n\t" - "addiu %[cosptr], %[cosptr], 2 \n\t" - "addiu %[sinptr], %[sinptr], 2 \n\t" - "sll %[r0], %[r0], 2 \n\t" - "sll %[r3], %[r3], 2 \n\t" -#if defined(MIPS_DSP_R1_LE) - "shra_r.w %[r8], %[r8], 14 \n\t" - "shra_r.w %[r1], %[r1], 14 \n\t" -#else // #if defined(MIPS_DSP_R1_LE) - "addiu %[r8], %[r8], 0x2000 \n\t" - "addiu %[r1], %[r1], 0x2000 \n\t" - "sra %[r8], %[r8], 14 \n\t" - "sra %[r1], %[r1], 14 \n\t" -#endif // #if defined(MIPS_DSP_R1_LE) - "addu %[r0], %[r0], %[r8] \n\t" - "addu %[r3], %[r3], %[r1] \n\t" - "subu %[r9], %[r9], %[r3] \n\t" - "addu %[r0], %[r0], %[r2] \n\t" - "sra %[r1], %[r9], 16 \n\t" - "andi %[r9], %[r9], 0xFFFF \n\t" - "mul %[r1], %[r1], %[r5] \n\t" - "mul %[r9], %[r9], %[r5] \n\t" - "sra %[r2], %[r0], 16 \n\t" - "andi %[r0], %[r0], 0xFFFF \n\t" - "mul %[r2], %[r2], %[r5] \n\t" - "mul %[r0], %[r0], %[r5] \n\t" - "sll %[r1], %[r1], 5 \n\t" -#if defined(MIPS_DSP_R1_LE) - "shra_r.w %[r9], %[r9], 11 \n\t" -#else // #if defined(MIPS_DSP_R1_LE) - "addiu %[r9], %[r9], 0x400 \n\t" - "sra %[r9], %[r9], 11 \n\t" -#endif // #if defined(MIPS_DSP_R1_LE) - "addu %[r1], %[r1], %[r9] \n\t" - "sll %[r2], %[r2], 5 \n\t" -#if defined(MIPS_DSP_R1_LE) - "shra_r.w %[r0], %[r0], 11 \n\t" -#else // #if defined(MIPS_DSP_R1_LE) - "addiu %[r0], %[r0], 0x400 \n\t" - "sra %[r0], %[r0], 11 \n\t" -#endif // #if defined(MIPS_DSP_R1_LE) - "addu %[r0], %[r0], %[r2] \n\t" - "sw %[r1], 0(%[outre1]) \n\t" - "addiu %[outre1], %[outre1], 4 \n\t" - "sw %[r0], 0(%[outre2]) \n\t" - "bgtz %[k], 1b \n\t" - " addiu %[outre2], %[outre2], 4 \n\t" - "b 3f \n\t" - " nop \n\t" -#endif // #if defined(MIPS_DSP_R2_LE) - "2: \n\t" -#if defined(MIPS_DSP_R2_LE) - "addiu %[max1], %[max1], -1 \n\t" - "21: \n\t" - "lwl %[r0], 0(%[inre]) \n\t" - "lwl %[r1], 0(%[inim]) \n\t" - "lh %[r2], 0(%[cosptr]) \n\t" - "lwr %[r0], 0(%[inre]) \n\t" - "lwr %[r1], 0(%[inim]) \n\t" - "lh %[r3], 0(%[sinptr]) \n\t" - "muleq_s.w.phr %[r6], %[r0], %[r4] \n\t" - "muleq_s.w.phr %[r7], %[r1], %[r4] \n\t" - "muleq_s.w.phl %[r0], %[r0], %[r4] \n\t" - "muleq_s.w.phl %[r1], %[r1], %[r4] \n\t" - "addiu %[k], %[k], -2 \n\t" - "addiu %[inre], %[inre], 4 \n\t" - "addiu %[inim], %[inim], 4 \n\t" - "sllv %[r6], %[r6], %[max1] \n\t" - "sllv %[r7], %[r7], %[max1] \n\t" - "mult $ac0, %[r2], %[r6] \n\t" - "mult $ac1, %[r3], %[r7] \n\t" - "mult $ac2, %[r2], %[r7] \n\t" - "mult $ac3, %[r3], %[r6] \n\t" - "lh %[r2], 2(%[cosptr]) \n\t" - "lh %[r3], 2(%[sinptr]) \n\t" - "extr_r.w %[r6], $ac0, 14 \n\t" - "extr_r.w %[r7], $ac1, 14 \n\t" - "extr_r.w %[r8], $ac2, 14 \n\t" - "extr_r.w %[r9], $ac3, 14 \n\t" - "sllv %[r0], %[r0], %[max1] \n\t" - "sllv %[r1], %[r1], %[max1] \n\t" - "mult $ac0, %[r2], %[r0] \n\t" - "mult $ac1, %[r3], %[r1] \n\t" - "mult $ac2, %[r2], %[r1] \n\t" - "mult $ac3, %[r3], %[r0] \n\t" - "addiu %[cosptr], %[cosptr], 4 \n\t" - "extr_r.w %[r0], $ac0, 14 \n\t" - "extr_r.w %[r1], $ac1, 14 \n\t" - "extr_r.w %[r2], $ac2, 14 \n\t" - "extr_r.w %[r3], $ac3, 14 \n\t" - "subu %[r6], %[r6], %[r7] \n\t" - "addu %[r8], %[r8], %[r9] \n\t" - "mult $ac0, %[r5], %[r6] \n\t" - "mult $ac1, %[r5], %[r8] \n\t" - "addiu %[sinptr], %[sinptr], 4 \n\t" - "subu %[r0], %[r0], %[r1] \n\t" - "addu %[r2], %[r2], %[r3] \n\t" - "extr_r.w %[r1], $ac0, 11 \n\t" - "extr_r.w %[r3], $ac1, 11 \n\t" - "mult $ac2, %[r5], %[r0] \n\t" - "mult $ac3, %[r5], %[r2] \n\t" - "sw %[r1], 0(%[outre1]) \n\t" - "sw %[r3], 0(%[outre2]) \n\t" - "addiu %[outre1], %[outre1], 8 \n\t" - "extr_r.w %[r0], $ac2, 11 \n\t" - "extr_r.w %[r2], $ac3, 11 \n\t" - "sw %[r0], -4(%[outre1]) \n\t" - "sw %[r2], 4(%[outre2]) \n\t" - "bgtz %[k], 21b \n\t" - " addiu %[outre2], %[outre2], 8 \n\t" - "b 3f \n\t" - " nop \n\t" -#else // #if defined(MIPS_DSP_R2_LE) - "lh %[r0], 0(%[inre]) \n\t" - "lh %[r1], 0(%[inim]) \n\t" - "addiu %[k], %[k], -1 \n\t" - "sllv %[r0], %[r0], %[max1] \n\t" - "sllv %[r1], %[r1], %[max1] \n\t" - "sra %[r2], %[r0], 16 \n\t" - "andi %[r0], %[r0], 0xFFFF \n\t" - "sra %[r0], %[r0], 1 \n\t" - "sra %[r3], %[r1], 16 \n\t" - "andi %[r1], %[r1], 0xFFFF \n\t" - "sra %[r1], %[r1], 1 \n\t" - "mul %[r2], %[r2], %[r4] \n\t" - "mul %[r0], %[r0], %[r4] \n\t" - "mul %[r3], %[r3], %[r4] \n\t" - "mul %[r1], %[r1], %[r4] \n\t" - "addiu %[inre], %[inre], 2 \n\t" - "addiu %[inim], %[inim], 2 \n\t" - "lh %[r6], 0(%[cosptr]) \n\t" - "lh %[r7], 0(%[sinptr]) \n\t" -#if defined(MIPS_DSP_R1_LE) - "shra_r.w %[r0], %[r0], 15 \n\t" - "shra_r.w %[r1], %[r1], 15 \n\t" -#else // #if defined(MIPS_DSP_R1_LE) - "addiu %[r0], %[r0], 0x4000 \n\t" - "addiu %[r1], %[r1], 0x4000 \n\t" - "sra %[r0], %[r0], 15 \n\t" - "sra %[r1], %[r1], 15 \n\t" -#endif // #if defined(MIPS_DSP_R1_LE) - "addu %[r0], %[r2], %[r0] \n\t" - "addu %[r1], %[r3], %[r1] \n\t" - "sra %[r2], %[r0], 16 \n\t" - "andi %[r0], %[r0], 0xFFFF \n\t" - "mul %[r9], %[r2], %[r6] \n\t" - "mul %[r2], %[r2], %[r7] \n\t" - "mul %[r8], %[r0], %[r6] \n\t" - "mul %[r0], %[r0], %[r7] \n\t" - "sra %[r3], %[r1], 16 \n\t" - "andi %[r1], %[r1], 0xFFFF \n\t" - "sll %[r9], %[r9], 2 \n\t" - "sll %[r2], %[r2], 2 \n\t" -#if defined(MIPS_DSP_R1_LE) - "shra_r.w %[r8], %[r8], 14 \n\t" - "shra_r.w %[r0], %[r0], 14 \n\t" -#else // #if defined(MIPS_DSP_R1_LE) - "addiu %[r8], %[r8], 0x2000 \n\t" - "addiu %[r0], %[r0], 0x2000 \n\t" - "sra %[r8], %[r8], 14 \n\t" - "sra %[r0], %[r0], 14 \n\t" -#endif // #if defined(MIPS_DSP_R1_LE) - "addu %[r9], %[r9], %[r8] \n\t" - "addu %[r2], %[r2], %[r0] \n\t" - "mul %[r0], %[r3], %[r6] \n\t" - "mul %[r3], %[r3], %[r7] \n\t" - "mul %[r8], %[r1], %[r6] \n\t" - "mul %[r1], %[r1], %[r7] \n\t" - "addiu %[cosptr], %[cosptr], 2 \n\t" - "addiu %[sinptr], %[sinptr], 2 \n\t" - "sll %[r0], %[r0], 2 \n\t" - "sll %[r3], %[r3], 2 \n\t" -#if defined(MIPS_DSP_R1_LE) - "shra_r.w %[r8], %[r8], 14 \n\t" - "shra_r.w %[r1], %[r1], 14 \n\t" -#else // #if defined(MIPS_DSP_R1_LE) - "addiu %[r8], %[r8], 0x2000 \n\t" - "addiu %[r1], %[r1], 0x2000 \n\t" - "sra %[r8], %[r8], 14 \n\t" - "sra %[r1], %[r1], 14 \n\t" -#endif // #if defined(MIPS_DSP_R1_LE) - "addu %[r0], %[r0], %[r8] \n\t" - "addu %[r3], %[r3], %[r1] \n\t" - "subu %[r9], %[r9], %[r3] \n\t" - "addu %[r0], %[r0], %[r2] \n\t" - "sra %[r1], %[r9], 16 \n\t" - "andi %[r9], %[r9], 0xFFFF \n\t" - "mul %[r1], %[r1], %[r5] \n\t" - "mul %[r9], %[r9], %[r5] \n\t" - "sra %[r2], %[r0], 16 \n\t" - "andi %[r0], %[r0], 0xFFFF \n\t" - "mul %[r2], %[r2], %[r5] \n\t" - "mul %[r0], %[r0], %[r5] \n\t" - "sll %[r1], %[r1], 5 \n\t" -#if defined(MIPS_DSP_R1_LE) - "shra_r.w %[r9], %[r9], 11 \n\t" -#else // #if defined(MIPS_DSP_R1_LE) - "addiu %[r9], %[r9], 0x400 \n\t" - "sra %[r9], %[r9], 11 \n\t" -#endif // #if defined(MIPS_DSP_R1_LE) - "addu %[r1], %[r1], %[r9] \n\t" - "sll %[r2], %[r2], 5 \n\t" -#if defined(MIPS_DSP_R1_LE) - "shra_r.w %[r0], %[r0], 11 \n\t" -#else // #if defined(MIPS_DSP_R1_LE) - "addiu %[r0], %[r0], 0x400 \n\t" - "sra %[r0], %[r0], 11 \n\t" -#endif // #if defined(MIPS_DSP_R1_LE) - "addu %[r0], %[r0], %[r2] \n\t" - "sw %[r1], 0(%[outre1]) \n\t" - "addiu %[outre1], %[outre1], 4 \n\t" - "sw %[r0], 0(%[outre2]) \n\t" - "bgtz %[k], 2b \n\t" - " addiu %[outre2], %[outre2], 4 \n\t" -#endif // #if defined(MIPS_DSP_R2_LE) - "3: \n\t" - ".set pop \n\t" - : [k] "+r" (k), [r0] "=&r" (r0), [r1] "=&r" (r1), - [r2] "=&r" (r2), [r3] "=&r" (r3), [r4] "=&r" (r4), - [r5] "=&r" (r5), [r6] "=&r" (r6), [r7] "=&r" (r7), - [r8] "=&r" (r8), [r9] "=&r" (r9), [max1] "=&r" (max1), - [inre] "=&r" (inre), [inim] "=&r" (inim), - [outre1] "=&r" (outre1), [outre2] "=&r" (outre2) - : [max] "r" (max), [inreQ7] "r" (inreQ7), - [inimQ7] "r" (inimQ7), [cosptr] "r" (cosptr), - [sinptr] "r" (sinptr), [outre1Q16] "r" (outre1Q16), - [outre2Q16] "r" (outre2Q16) - : "hi", "lo", "memory" -#if defined(MIPS_DSP_R2_LE) - , "$ac1hi", "$ac1lo", "$ac2hi", "$ac2lo", "$ac3hi", "$ac3lo" -#endif // #if defined(MIPS_DSP_R2_LE) - ); -} diff --git a/modules/audio_coding/codecs/isac/fix/source/transform_neon.c b/modules/audio_coding/codecs/isac/fix/source/transform_neon.c deleted file mode 100644 index 79dadc4600..0000000000 --- a/modules/audio_coding/codecs/isac/fix/source/transform_neon.c +++ /dev/null @@ -1,479 +0,0 @@ -/* - * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#include - -#include "modules/audio_coding/codecs/isac/fix/source/codec.h" -#include "modules/audio_coding/codecs/isac/fix/source/fft.h" -#include "modules/audio_coding/codecs/isac/fix/source/settings.h" - -// Tables are defined in transform_tables.c file. -// Cosine table 1 in Q14. -extern const int16_t WebRtcIsacfix_kCosTab1[FRAMESAMPLES/2]; -// Sine table 1 in Q14. -extern const int16_t WebRtcIsacfix_kSinTab1[FRAMESAMPLES/2]; -// Sine table 2 in Q14. -extern const int16_t WebRtcIsacfix_kSinTab2[FRAMESAMPLES/4]; - -static inline int32_t ComplexMulAndFindMaxNeon(int16_t* inre1Q9, - int16_t* inre2Q9, - int32_t* outreQ16, - int32_t* outimQ16) { - int k; - const int16_t* kCosTab = &WebRtcIsacfix_kCosTab1[0]; - const int16_t* kSinTab = &WebRtcIsacfix_kSinTab1[0]; - // 0.5 / sqrt(240) in Q19 is round((.5 / sqrt(240)) * (2^19)) = 16921. - // Use "16921 << 5" and vqdmulh, instead of ">> 26" as in the C code. - int32_t fact = 16921 << 5; - int32x4_t factq = vdupq_n_s32(fact); - uint32x4_t max_r = vdupq_n_u32(0); - uint32x4_t max_i = vdupq_n_u32(0); - - for (k = 0; k < FRAMESAMPLES/2; k += 8) { - int16x8_t tmpr = vld1q_s16(kCosTab); - int16x8_t tmpi = vld1q_s16(kSinTab); - int16x8_t inre1 = vld1q_s16(inre1Q9); - int16x8_t inre2 = vld1q_s16(inre2Q9); - kCosTab += 8; - kSinTab += 8; - inre1Q9 += 8; - inre2Q9 += 8; - - // Use ">> 26", instead of ">> 7", ">> 16" and then ">> 3" as in the C code. - int32x4_t tmp0 = vmull_s16(vget_low_s16(tmpr), vget_low_s16(inre1)); - int32x4_t tmp1 = vmull_s16(vget_low_s16(tmpr), vget_low_s16(inre2)); - tmp0 = vmlal_s16(tmp0, vget_low_s16(tmpi), vget_low_s16(inre2)); - tmp1 = vmlsl_s16(tmp1, vget_low_s16(tmpi), vget_low_s16(inre1)); -#if defined(WEBRTC_ARCH_ARM64) - int32x4_t tmp2 = vmull_high_s16(tmpr, inre1); - int32x4_t tmp3 = vmull_high_s16(tmpr, inre2); - tmp2 = vmlal_high_s16(tmp2, tmpi, inre2); - tmp3 = vmlsl_high_s16(tmp3, tmpi, inre1); -#else - int32x4_t tmp2 = vmull_s16(vget_high_s16(tmpr), vget_high_s16(inre1)); - int32x4_t tmp3 = vmull_s16(vget_high_s16(tmpr), vget_high_s16(inre2)); - tmp2 = vmlal_s16(tmp2, vget_high_s16(tmpi), vget_high_s16(inre2)); - tmp3 = vmlsl_s16(tmp3, vget_high_s16(tmpi), vget_high_s16(inre1)); -#endif - - int32x4_t outr_0 = vqdmulhq_s32(tmp0, factq); - int32x4_t outr_1 = vqdmulhq_s32(tmp2, factq); - int32x4_t outi_0 = vqdmulhq_s32(tmp1, factq); - int32x4_t outi_1 = vqdmulhq_s32(tmp3, factq); - vst1q_s32(outreQ16, outr_0); - outreQ16 += 4; - vst1q_s32(outreQ16, outr_1); - outreQ16 += 4; - vst1q_s32(outimQ16, outi_0); - outimQ16 += 4; - vst1q_s32(outimQ16, outi_1); - outimQ16 += 4; - - // Find the absolute maximum in the vectors. - tmp0 = vabsq_s32(outr_0); - tmp1 = vabsq_s32(outr_1); - tmp2 = vabsq_s32(outi_0); - tmp3 = vabsq_s32(outi_1); - // vabs doesn't change the value of 0x80000000. - // Use u32 so we don't lose the value 0x80000000. - max_r = vmaxq_u32(max_r, vreinterpretq_u32_s32(tmp0)); - max_i = vmaxq_u32(max_i, vreinterpretq_u32_s32(tmp2)); - max_r = vmaxq_u32(max_r, vreinterpretq_u32_s32(tmp1)); - max_i = vmaxq_u32(max_i, vreinterpretq_u32_s32(tmp3)); - } - - max_r = vmaxq_u32(max_r, max_i); -#if defined(WEBRTC_ARCH_ARM64) - uint32_t maximum = vmaxvq_u32(max_r); -#else - uint32x2_t max32x2_r = vmax_u32(vget_low_u32(max_r), vget_high_u32(max_r)); - max32x2_r = vpmax_u32(max32x2_r, max32x2_r); - uint32_t maximum = vget_lane_u32(max32x2_r, 0); -#endif - - return (int32_t)maximum; -} - -static inline void PreShiftW32toW16Neon(int32_t* inre, - int32_t* inim, - int16_t* outre, - int16_t* outim, - int32_t sh) { - int k; - int32x4_t sh32x4 = vdupq_n_s32(sh); - for (k = 0; k < FRAMESAMPLES/2; k += 16) { - int32x4x4_t inre32x4x4 = vld4q_s32(inre); - int32x4x4_t inim32x4x4 = vld4q_s32(inim); - inre += 16; - inim += 16; - inre32x4x4.val[0] = vrshlq_s32(inre32x4x4.val[0], sh32x4); - inre32x4x4.val[1] = vrshlq_s32(inre32x4x4.val[1], sh32x4); - inre32x4x4.val[2] = vrshlq_s32(inre32x4x4.val[2], sh32x4); - inre32x4x4.val[3] = vrshlq_s32(inre32x4x4.val[3], sh32x4); - inim32x4x4.val[0] = vrshlq_s32(inim32x4x4.val[0], sh32x4); - inim32x4x4.val[1] = vrshlq_s32(inim32x4x4.val[1], sh32x4); - inim32x4x4.val[2] = vrshlq_s32(inim32x4x4.val[2], sh32x4); - inim32x4x4.val[3] = vrshlq_s32(inim32x4x4.val[3], sh32x4); - int16x4x4_t outre16x4x4; - int16x4x4_t outim16x4x4; - outre16x4x4.val[0] = vmovn_s32(inre32x4x4.val[0]); - outre16x4x4.val[1] = vmovn_s32(inre32x4x4.val[1]); - outre16x4x4.val[2] = vmovn_s32(inre32x4x4.val[2]); - outre16x4x4.val[3] = vmovn_s32(inre32x4x4.val[3]); - outim16x4x4.val[0] = vmovn_s32(inim32x4x4.val[0]); - outim16x4x4.val[1] = vmovn_s32(inim32x4x4.val[1]); - outim16x4x4.val[2] = vmovn_s32(inim32x4x4.val[2]); - outim16x4x4.val[3] = vmovn_s32(inim32x4x4.val[3]); - vst4_s16(outre, outre16x4x4); - vst4_s16(outim, outim16x4x4); - outre += 16; - outim += 16; - } -} - -static inline void PostShiftAndSeparateNeon(int16_t* inre, - int16_t* inim, - int16_t* outre, - int16_t* outim, - int32_t sh) { - int k; - int16_t* inre1 = inre; - int16_t* inre2 = &inre[FRAMESAMPLES/2 - 4]; - int16_t* inim1 = inim; - int16_t* inim2 = &inim[FRAMESAMPLES/2 - 4]; - int16_t* outre1 = outre; - int16_t* outre2 = &outre[FRAMESAMPLES/2 - 4]; - int16_t* outim1 = outim; - int16_t* outim2 = &outim[FRAMESAMPLES/2 - 4]; - const int16_t* kSinTab1 = &WebRtcIsacfix_kSinTab2[0]; - const int16_t* kSinTab2 = &WebRtcIsacfix_kSinTab2[FRAMESAMPLES/4 -4]; - // By vshl, we effectively did "<< (-sh - 23)", instead of "<< (-sh)", - // ">> 14" and then ">> 9" as in the C code. - int32x4_t shift = vdupq_n_s32(-sh - 23); - - for (k = 0; k < FRAMESAMPLES/4; k += 4) { - int16x4_t tmpi = vld1_s16(kSinTab1); - kSinTab1 += 4; - int16x4_t tmpr = vld1_s16(kSinTab2); - kSinTab2 -= 4; - int16x4_t inre_0 = vld1_s16(inre1); - inre1 += 4; - int16x4_t inre_1 = vld1_s16(inre2); - inre2 -= 4; - int16x4_t inim_0 = vld1_s16(inim1); - inim1 += 4; - int16x4_t inim_1 = vld1_s16(inim2); - inim2 -= 4; - tmpr = vneg_s16(tmpr); - inre_1 = vrev64_s16(inre_1); - inim_1 = vrev64_s16(inim_1); - tmpr = vrev64_s16(tmpr); - - int16x4_t xr = vqadd_s16(inre_0, inre_1); - int16x4_t xi = vqsub_s16(inim_0, inim_1); - int16x4_t yr = vqadd_s16(inim_0, inim_1); - int16x4_t yi = vqsub_s16(inre_1, inre_0); - - int32x4_t outr0 = vmull_s16(tmpr, xr); - int32x4_t outi0 = vmull_s16(tmpi, xr); - int32x4_t outr1 = vmull_s16(tmpi, yr); - int32x4_t outi1 = vmull_s16(tmpi, yi); - outr0 = vmlsl_s16(outr0, tmpi, xi); - outi0 = vmlal_s16(outi0, tmpr, xi); - outr1 = vmlal_s16(outr1, tmpr, yi); - outi1 = vmlsl_s16(outi1, tmpr, yr); - - outr0 = vshlq_s32(outr0, shift); - outi0 = vshlq_s32(outi0, shift); - outr1 = vshlq_s32(outr1, shift); - outi1 = vshlq_s32(outi1, shift); - outr1 = vnegq_s32(outr1); - - int16x4_t outre_0 = vmovn_s32(outr0); - int16x4_t outim_0 = vmovn_s32(outi0); - int16x4_t outre_1 = vmovn_s32(outr1); - int16x4_t outim_1 = vmovn_s32(outi1); - outre_1 = vrev64_s16(outre_1); - outim_1 = vrev64_s16(outim_1); - - vst1_s16(outre1, outre_0); - outre1 += 4; - vst1_s16(outim1, outim_0); - outim1 += 4; - vst1_s16(outre2, outre_1); - outre2 -= 4; - vst1_s16(outim2, outim_1); - outim2 -= 4; - } -} - -void WebRtcIsacfix_Time2SpecNeon(int16_t* inre1Q9, - int16_t* inre2Q9, - int16_t* outreQ7, - int16_t* outimQ7) { - int32_t tmpreQ16[FRAMESAMPLES/2], tmpimQ16[FRAMESAMPLES/2]; - int32_t max; - int32_t sh; - - // Multiply with complex exponentials and combine into one complex vector. - // And find the maximum. - max = ComplexMulAndFindMaxNeon(inre1Q9, inre2Q9, tmpreQ16, tmpimQ16); - - sh = (int32_t)WebRtcSpl_NormW32(max); - sh = sh - 24; - - // If sh becomes >= 0, then we should shift sh steps to the left, - // and the domain will become Q(16 + sh). - // If sh becomes < 0, then we should shift -sh steps to the right, - // and the domain will become Q(16 + sh). - PreShiftW32toW16Neon(tmpreQ16, tmpimQ16, inre1Q9, inre2Q9, sh); - - // Get DFT. - WebRtcIsacfix_FftRadix16Fastest(inre1Q9, inre2Q9, -1); - - // If sh >= 0, shift sh steps to the right, - // If sh < 0, shift -sh steps to the left. - // Use symmetry to separate into two complex vectors - // and center frames in time around zero. - PostShiftAndSeparateNeon(inre1Q9, inre2Q9, outreQ7, outimQ7, sh); -} - -static inline int32_t TransformAndFindMaxNeon(int16_t* inre, - int16_t* inim, - int32_t* outre, - int32_t* outim) { - int k; - int16_t* inre1 = inre; - int16_t* inre2 = &inre[FRAMESAMPLES/2 - 4]; - int16_t* inim1 = inim; - int16_t* inim2 = &inim[FRAMESAMPLES/2 - 4]; - int32_t* outre1 = outre; - int32_t* outre2 = &outre[FRAMESAMPLES/2 - 4]; - int32_t* outim1 = outim; - int32_t* outim2 = &outim[FRAMESAMPLES/2 - 4]; - const int16_t* kSinTab1 = &WebRtcIsacfix_kSinTab2[0]; - const int16_t* kSinTab2 = &WebRtcIsacfix_kSinTab2[FRAMESAMPLES/4 - 4]; - uint32x4_t max_r = vdupq_n_u32(0); - uint32x4_t max_i = vdupq_n_u32(0); - - // Use ">> 5", instead of "<< 9" and then ">> 14" as in the C code. - for (k = 0; k < FRAMESAMPLES/4; k += 4) { - int16x4_t tmpi = vld1_s16(kSinTab1); - kSinTab1 += 4; - int16x4_t tmpr = vld1_s16(kSinTab2); - kSinTab2 -= 4; - int16x4_t inre_0 = vld1_s16(inre1); - inre1 += 4; - int16x4_t inre_1 = vld1_s16(inre2); - inre2 -= 4; - int16x4_t inim_0 = vld1_s16(inim1); - inim1 += 4; - int16x4_t inim_1 = vld1_s16(inim2); - inim2 -= 4; - tmpr = vneg_s16(tmpr); - inre_1 = vrev64_s16(inre_1); - inim_1 = vrev64_s16(inim_1); - tmpr = vrev64_s16(tmpr); - - int32x4_t xr = vmull_s16(tmpr, inre_0); - int32x4_t xi = vmull_s16(tmpr, inim_0); - int32x4_t yr = vmull_s16(tmpr, inim_1); - int32x4_t yi = vmull_s16(tmpi, inim_1); - xr = vmlal_s16(xr, tmpi, inim_0); - xi = vmlsl_s16(xi, tmpi, inre_0); - yr = vmlal_s16(yr, tmpi, inre_1); - yi = vmlsl_s16(yi, tmpr, inre_1); - yr = vnegq_s32(yr); - - xr = vshrq_n_s32(xr, 5); - xi = vshrq_n_s32(xi, 5); - yr = vshrq_n_s32(yr, 5); - yi = vshrq_n_s32(yi, 5); - - int32x4_t outr0 = vsubq_s32(xr, yi); - int32x4_t outr1 = vaddq_s32(xr, yi); - int32x4_t outi0 = vaddq_s32(xi, yr); - int32x4_t outi1 = vsubq_s32(yr, xi); - - // Find the absolute maximum in the vectors. - int32x4_t tmp0 = vabsq_s32(outr0); - int32x4_t tmp1 = vabsq_s32(outr1); - int32x4_t tmp2 = vabsq_s32(outi0); - int32x4_t tmp3 = vabsq_s32(outi1); - // vabs doesn't change the value of 0x80000000. - // Use u32 so we don't lose the value 0x80000000. - max_r = vmaxq_u32(max_r, vreinterpretq_u32_s32(tmp0)); - max_i = vmaxq_u32(max_i, vreinterpretq_u32_s32(tmp2)); - max_r = vmaxq_u32(max_r, vreinterpretq_u32_s32(tmp1)); - max_i = vmaxq_u32(max_i, vreinterpretq_u32_s32(tmp3)); - - // Store the vectors. - outr1 = vrev64q_s32(outr1); - outi1 = vrev64q_s32(outi1); - int32x4_t outr_1 = vcombine_s32(vget_high_s32(outr1), vget_low_s32(outr1)); - int32x4_t outi_1 = vcombine_s32(vget_high_s32(outi1), vget_low_s32(outi1)); - - vst1q_s32(outre1, outr0); - outre1 += 4; - vst1q_s32(outim1, outi0); - outim1 += 4; - vst1q_s32(outre2, outr_1); - outre2 -= 4; - vst1q_s32(outim2, outi_1); - outim2 -= 4; - } - - max_r = vmaxq_u32(max_r, max_i); -#if defined(WEBRTC_ARCH_ARM64) - uint32_t maximum = vmaxvq_u32(max_r); -#else - uint32x2_t max32x2_r = vmax_u32(vget_low_u32(max_r), vget_high_u32(max_r)); - max32x2_r = vpmax_u32(max32x2_r, max32x2_r); - uint32_t maximum = vget_lane_u32(max32x2_r, 0); -#endif - - return (int32_t)maximum; -} - -static inline void PostShiftAndDivideAndDemodulateNeon(int16_t* inre, - int16_t* inim, - int32_t* outre1, - int32_t* outre2, - int32_t sh) { - int k; - int16_t* p_inre = inre; - int16_t* p_inim = inim; - int32_t* p_outre1 = outre1; - int32_t* p_outre2 = outre2; - const int16_t* kCosTab = &WebRtcIsacfix_kCosTab1[0]; - const int16_t* kSinTab = &WebRtcIsacfix_kSinTab1[0]; - int32x4_t shift = vdupq_n_s32(-sh - 16); - // Divide through by the normalizing constant: - // scale all values with 1/240, i.e. with 273 in Q16. - // 273/65536 ~= 0.0041656 - // 1/240 ~= 0.0041666 - int16x8_t scale = vdupq_n_s16(273); - // Sqrt(240) in Q11 is round(15.49193338482967 * 2048) = 31727. - int factQ19 = 31727 << 16; - int32x4_t fact = vdupq_n_s32(factQ19); - - for (k = 0; k < FRAMESAMPLES/2; k += 8) { - int16x8_t inre16x8 = vld1q_s16(p_inre); - int16x8_t inim16x8 = vld1q_s16(p_inim); - p_inre += 8; - p_inim += 8; - int16x8_t tmpr = vld1q_s16(kCosTab); - int16x8_t tmpi = vld1q_s16(kSinTab); - kCosTab += 8; - kSinTab += 8; - // By vshl and vmull, we effectively did "<< (-sh - 16)", - // instead of "<< (-sh)" and ">> 16" as in the C code. - int32x4_t outre1_0 = vmull_s16(vget_low_s16(inre16x8), vget_low_s16(scale)); - int32x4_t outre2_0 = vmull_s16(vget_low_s16(inim16x8), vget_low_s16(scale)); -#if defined(WEBRTC_ARCH_ARM64) - int32x4_t outre1_1 = vmull_high_s16(inre16x8, scale); - int32x4_t outre2_1 = vmull_high_s16(inim16x8, scale); -#else - int32x4_t outre1_1 = vmull_s16(vget_high_s16(inre16x8), - vget_high_s16(scale)); - int32x4_t outre2_1 = vmull_s16(vget_high_s16(inim16x8), - vget_high_s16(scale)); -#endif - - outre1_0 = vshlq_s32(outre1_0, shift); - outre1_1 = vshlq_s32(outre1_1, shift); - outre2_0 = vshlq_s32(outre2_0, shift); - outre2_1 = vshlq_s32(outre2_1, shift); - - // Demodulate and separate. - int32x4_t tmpr_0 = vmovl_s16(vget_low_s16(tmpr)); - int32x4_t tmpi_0 = vmovl_s16(vget_low_s16(tmpi)); -#if defined(WEBRTC_ARCH_ARM64) - int32x4_t tmpr_1 = vmovl_high_s16(tmpr); - int32x4_t tmpi_1 = vmovl_high_s16(tmpi); -#else - int32x4_t tmpr_1 = vmovl_s16(vget_high_s16(tmpr)); - int32x4_t tmpi_1 = vmovl_s16(vget_high_s16(tmpi)); -#endif - - int64x2_t xr0 = vmull_s32(vget_low_s32(tmpr_0), vget_low_s32(outre1_0)); - int64x2_t xi0 = vmull_s32(vget_low_s32(tmpr_0), vget_low_s32(outre2_0)); - int64x2_t xr2 = vmull_s32(vget_low_s32(tmpr_1), vget_low_s32(outre1_1)); - int64x2_t xi2 = vmull_s32(vget_low_s32(tmpr_1), vget_low_s32(outre2_1)); - xr0 = vmlsl_s32(xr0, vget_low_s32(tmpi_0), vget_low_s32(outre2_0)); - xi0 = vmlal_s32(xi0, vget_low_s32(tmpi_0), vget_low_s32(outre1_0)); - xr2 = vmlsl_s32(xr2, vget_low_s32(tmpi_1), vget_low_s32(outre2_1)); - xi2 = vmlal_s32(xi2, vget_low_s32(tmpi_1), vget_low_s32(outre1_1)); - -#if defined(WEBRTC_ARCH_ARM64) - int64x2_t xr1 = vmull_high_s32(tmpr_0, outre1_0); - int64x2_t xi1 = vmull_high_s32(tmpr_0, outre2_0); - int64x2_t xr3 = vmull_high_s32(tmpr_1, outre1_1); - int64x2_t xi3 = vmull_high_s32(tmpr_1, outre2_1); - xr1 = vmlsl_high_s32(xr1, tmpi_0, outre2_0); - xi1 = vmlal_high_s32(xi1, tmpi_0, outre1_0); - xr3 = vmlsl_high_s32(xr3, tmpi_1, outre2_1); - xi3 = vmlal_high_s32(xi3, tmpi_1, outre1_1); -#else - int64x2_t xr1 = vmull_s32(vget_high_s32(tmpr_0), vget_high_s32(outre1_0)); - int64x2_t xi1 = vmull_s32(vget_high_s32(tmpr_0), vget_high_s32(outre2_0)); - int64x2_t xr3 = vmull_s32(vget_high_s32(tmpr_1), vget_high_s32(outre1_1)); - int64x2_t xi3 = vmull_s32(vget_high_s32(tmpr_1), vget_high_s32(outre2_1)); - xr1 = vmlsl_s32(xr1, vget_high_s32(tmpi_0), vget_high_s32(outre2_0)); - xi1 = vmlal_s32(xi1, vget_high_s32(tmpi_0), vget_high_s32(outre1_0)); - xr3 = vmlsl_s32(xr3, vget_high_s32(tmpi_1), vget_high_s32(outre2_1)); - xi3 = vmlal_s32(xi3, vget_high_s32(tmpi_1), vget_high_s32(outre1_1)); -#endif - - outre1_0 = vcombine_s32(vrshrn_n_s64(xr0, 10), vrshrn_n_s64(xr1, 10)); - outre2_0 = vcombine_s32(vrshrn_n_s64(xi0, 10), vrshrn_n_s64(xi1, 10)); - outre1_1 = vcombine_s32(vrshrn_n_s64(xr2, 10), vrshrn_n_s64(xr3, 10)); - outre2_1 = vcombine_s32(vrshrn_n_s64(xi2, 10), vrshrn_n_s64(xi3, 10)); - outre1_0 = vqdmulhq_s32(outre1_0, fact); - outre2_0 = vqdmulhq_s32(outre2_0, fact); - outre1_1 = vqdmulhq_s32(outre1_1, fact); - outre2_1 = vqdmulhq_s32(outre2_1, fact); - - vst1q_s32(p_outre1, outre1_0); - p_outre1 += 4; - vst1q_s32(p_outre1, outre1_1); - p_outre1 += 4; - vst1q_s32(p_outre2, outre2_0); - p_outre2 += 4; - vst1q_s32(p_outre2, outre2_1); - p_outre2 += 4; - } -} - -void WebRtcIsacfix_Spec2TimeNeon(int16_t* inreQ7, - int16_t* inimQ7, - int32_t* outre1Q16, - int32_t* outre2Q16) { - int32_t max; - int32_t sh; - - max = TransformAndFindMaxNeon(inreQ7, inimQ7, outre1Q16, outre2Q16); - - - sh = (int32_t)WebRtcSpl_NormW32(max); - sh = sh - 24; - // If sh becomes >= 0, then we should shift sh steps to the left, - // and the domain will become Q(16 + sh). - // If sh becomes < 0, then we should shift -sh steps to the right, - // and the domain will become Q(16 + sh). - - // "Fastest" vectors. - PreShiftW32toW16Neon(outre1Q16, outre2Q16, inreQ7, inimQ7, sh); - - // Get IDFT. - WebRtcIsacfix_FftRadix16Fastest(inreQ7, inimQ7, 1); - - PostShiftAndDivideAndDemodulateNeon(inreQ7, inimQ7, outre1Q16, outre2Q16, sh); -} diff --git a/modules/audio_coding/codecs/isac/fix/source/transform_tables.c b/modules/audio_coding/codecs/isac/fix/source/transform_tables.c deleted file mode 100644 index e661effdde..0000000000 --- a/modules/audio_coding/codecs/isac/fix/source/transform_tables.c +++ /dev/null @@ -1,110 +0,0 @@ -/* - * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/* - * This file contains trigonometric functions look-up tables used in - * transform functions WebRtcIsacfix_Time2Spec and WebRtcIsacfix_Spec2Time. - */ - -#include - -#include "modules/audio_coding/codecs/isac/fix/source/settings.h" - -/* Cosine table 1 in Q14. */ -const int16_t WebRtcIsacfix_kCosTab1[FRAMESAMPLES/2] = { - 16384, 16383, 16378, 16371, 16362, 16349, 16333, 16315, 16294, 16270, - 16244, 16214, 16182, 16147, 16110, 16069, 16026, 15980, 15931, 15880, - 15826, 15769, 15709, 15647, 15582, 15515, 15444, 15371, 15296, 15218, - 15137, 15053, 14968, 14879, 14788, 14694, 14598, 14500, 14399, 14295, - 14189, 14081, 13970, 13856, 13741, 13623, 13502, 13380, 13255, 13128, - 12998, 12867, 12733, 12597, 12458, 12318, 12176, 12031, 11885, 11736, - 11585, 11433, 11278, 11121, 10963, 10803, 10641, 10477, 10311, 10143, - 9974, 9803, 9630, 9456, 9280, 9102, 8923, 8743, 8561, 8377, - 8192, 8006, 7818, 7629, 7438, 7246, 7053, 6859, 6664, 6467, - 6270, 6071, 5872, 5671, 5469, 5266, 5063, 4859, 4653, 4447, - 4240, 4033, 3825, 3616, 3406, 3196, 2986, 2775, 2563, 2351, - 2139, 1926, 1713, 1499, 1285, 1072, 857, 643, 429, 214, - 0, -214, -429, -643, -857, -1072, -1285, -1499, -1713, -1926, - -2139, -2351, -2563, -2775, -2986, -3196, -3406, -3616, -3825, -4033, - -4240, -4447, -4653, -4859, -5063, -5266, -5469, -5671, -5872, -6071, - -6270, -6467, -6664, -6859, -7053, -7246, -7438, -7629, -7818, -8006, - -8192, -8377, -8561, -8743, -8923, -9102, -9280, -9456, -9630, -9803, - -9974, -10143, -10311, -10477, -10641, -10803, -10963, -11121, -11278, -11433, - -11585, -11736, -11885, -12031, -12176, -12318, -12458, -12597, -12733, - -12867, -12998, -13128, -13255, -13380, -13502, -13623, -13741, -13856, - -13970, -14081, -14189, -14295, -14399, -14500, -14598, -14694, -14788, - -14879, -14968, -15053, -15137, -15218, -15296, -15371, -15444, -15515, - -15582, -15647, -15709, -15769, -15826, -15880, -15931, -15980, -16026, - -16069, -16110, -16147, -16182, -16214, -16244, -16270, -16294, -16315, - -16333, -16349, -16362, -16371, -16378, -16383 -}; - -/* Sine table 1 in Q14. */ -const int16_t WebRtcIsacfix_kSinTab1[FRAMESAMPLES/2] = { - 0, 214, 429, 643, 857, 1072, 1285, 1499, 1713, 1926, - 2139, 2351, 2563, 2775, 2986, 3196, 3406, 3616, 3825, 4033, - 4240, 4447, 4653, 4859, 5063, 5266, 5469, 5671, 5872, 6071, - 6270, 6467, 6664, 6859, 7053, 7246, 7438, 7629, 7818, 8006, - 8192, 8377, 8561, 8743, 8923, 9102, 9280, 9456, 9630, 9803, - 9974, 10143, 10311, 10477, 10641, 10803, 10963, 11121, 11278, 11433, - 11585, 11736, 11885, 12031, 12176, 12318, 12458, 12597, 12733, 12867, - 12998, 13128, 13255, 13380, 13502, 13623, 13741, 13856, 13970, 14081, - 14189, 14295, 14399, 14500, 14598, 14694, 14788, 14879, 14968, 15053, - 15137, 15218, 15296, 15371, 15444, 15515, 15582, 15647, 15709, 15769, - 15826, 15880, 15931, 15980, 16026, 16069, 16110, 16147, 16182, 16214, - 16244, 16270, 16294, 16315, 16333, 16349, 16362, 16371, 16378, 16383, - 16384, 16383, 16378, 16371, 16362, 16349, 16333, 16315, 16294, 16270, - 16244, 16214, 16182, 16147, 16110, 16069, 16026, 15980, 15931, 15880, - 15826, 15769, 15709, 15647, 15582, 15515, 15444, 15371, 15296, 15218, - 15137, 15053, 14968, 14879, 14788, 14694, 14598, 14500, 14399, 14295, - 14189, 14081, 13970, 13856, 13741, 13623, 13502, 13380, 13255, 13128, - 12998, 12867, 12733, 12597, 12458, 12318, 12176, 12031, 11885, 11736, - 11585, 11433, 11278, 11121, 10963, 10803, 10641, 10477, 10311, 10143, - 9974, 9803, 9630, 9456, 9280, 9102, 8923, 8743, 8561, 8377, - 8192, 8006, 7818, 7629, 7438, 7246, 7053, 6859, 6664, 6467, - 6270, 6071, 5872, 5671, 5469, 5266, 5063, 4859, 4653, 4447, - 4240, 4033, 3825, 3616, 3406, 3196, 2986, 2775, 2563, 2351, - 2139, 1926, 1713, 1499, 1285, 1072, 857, 643, 429, 214 -}; - - -/* Sine table 2 in Q14. */ -const int16_t WebRtcIsacfix_kSinTab2[FRAMESAMPLES/4] = { - 16384, -16381, 16375, -16367, 16356, -16342, 16325, -16305, 16283, -16257, - 16229, -16199, 16165, -16129, 16090, -16048, 16003, -15956, 15906, -15853, - 15798, -15739, 15679, -15615, 15549, -15480, 15408, -15334, 15257, -15178, - 15095, -15011, 14924, -14834, 14741, -14647, 14549, -14449, 14347, -14242, - 14135, -14025, 13913, -13799, 13682, -13563, 13441, -13318, 13192, -13063, - 12933, -12800, 12665, -12528, 12389, -12247, 12104, -11958, 11810, -11661, - 11509, -11356, 11200, -11042, 10883, -10722, 10559, -10394, 10227, -10059, - 9889, -9717, 9543, -9368, 9191, -9013, 8833, -8652, 8469, -8285, - 8099, -7912, 7723, -7534, 7342, -7150, 6957, -6762, 6566, -6369, - 6171, -5971, 5771, -5570, 5368, -5165, 4961, -4756, 4550, -4344, - 4137, -3929, 3720, -3511, 3301, -3091, 2880, -2669, 2457, -2245, - 2032, -1819, 1606, -1392, 1179, -965, 750, -536, 322, -107 -}; - -#if defined(MIPS32_LE) -/* Cosine table 2 in Q14. Used only on MIPS platforms. */ -const int16_t WebRtcIsacfix_kCosTab2[FRAMESAMPLES/4] = { - 107, -322, 536, -750, 965, -1179, 1392, -1606, 1819, -2032, - 2245, -2457, 2669, -2880, 3091, -3301, 3511, -3720, 3929, -4137, - 4344, -4550, 4756, -4961, 5165, -5368, 5570, -5771, 5971, -6171, - 6369, -6566, 6762, -6957, 7150, -7342, 7534, -7723, 7912, -8099, - 8285, -8469, 8652, -8833, 9013, -9191, 9368, -9543, 9717, -9889, - 10059, -10227, 10394, -10559, 10722, -10883, 11042, -11200, 11356, -11509, - 11661, -11810, 11958, -12104, 12247, -12389, 12528, -12665, 12800, -12933, - 13063, -13192, 13318, -13441, 13563, -13682, 13799, -13913, 14025, -14135, - 14242, -14347, 14449, -14549, 14647, -14741, 14834, -14924, 15011, -15095, - 15178, -15257, 15334, -15408, 15480, -15549, 15615, -15679, 15739, -15798, - 15853, -15906, 15956, -16003, 16048, -16090, 16129, -16165, 16199, -16229, - 16257, -16283, 16305, -16325, 16342, -16356, 16367, -16375, 16381, -16384 -}; -#endif diff --git a/modules/audio_coding/codecs/isac/fix/source/transform_unittest.cc b/modules/audio_coding/codecs/isac/fix/source/transform_unittest.cc deleted file mode 100644 index 433ec534fe..0000000000 --- a/modules/audio_coding/codecs/isac/fix/source/transform_unittest.cc +++ /dev/null @@ -1,199 +0,0 @@ -/* - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#include "modules/audio_coding/codecs/isac/fix/source/codec.h" -#include "system_wrappers/include/cpu_features_wrapper.h" -#include "test/gtest.h" - -static const int kSamples = FRAMESAMPLES / 2; -static const int32_t spec2time_out_expected_1[kSamples] = { - -3366470, -2285227, -3415765, -2310215, -3118030, -2222470, -3030254, - -2192091, -3423170, -2216041, -3305541, -2171936, -3195767, -2095779, - -3153304, -2157560, -3071167, -2032108, -3101190, -1972016, -3103824, - -2089118, -3139811, -1898337, -3102801, -2055082, -3029665, -1854140, - -2962586, -1966454, -3071167, -1894588, -2851743, -1917315, -2848087, - -1594932, -2799242, -1462184, -2845887, -1437599, -2691776, -1329637, - -2770659, -1268491, -2625161, -1578991, -2460299, -1186385, -2365613, - -1039354, -2322608, -958518, -2271749, -789860, -2254538, -850308, - -2384436, -850959, -2133734, -587678, -2093316, -495115, -1973364, - -475177, -1801282, -173507, -1848516, -158015, -1792018, -62648, - -1643313, 214746, -1500758, 267077, -1450193, 560521, -1521579, - 675283, -1345408, 857559, -1300822, 1116332, -1294533, 1241117, - -1070027, 1263503, -983816, 1529821, -1019586, 1910421, -955420, - 2073688, -836459, 2401105, -653905, 2690474, -731425, 2930131, - -935234, 3299500, -875978, 3523432, -878906, 3924822, -1081630, - 4561267, -1203023, 5105274, -1510983, 6052762, -2294646, 7021597, - -3108053, 8826736, -4935222, 11678789, -8442713, 18725700, -21526692, - 25420577, 19589811, -28108666, 12634054, -14483066, 6263217, -9979706, - 3665661, -7909736, 2531530, -6434896, 1700772, -5525393, 1479473, - -4894262, 1231760, -4353044, 1032940, -3786590, 941152, -3331614, - 665090, -2851619, 830696, -2762201, 958007, -2483118, 788233, - -2184965, 804825, -1967306, 1007255, -1862474, 920889, -1457506, - 755406, -1405841, 890230, -1302124, 1161599, -701867, 1154163, - -1083366, 1204743, -513581, 1547264, -650636, 1493384, -285543, - 1771863, -277906, 1841343, -9078, 1751863, 230222, 1819578, - 207170, 1978972, 398137, 2106468, 552155, 1997624, 685213, - 2129520, 601078, 2238736, 944591, 2441879, 1194178, 2355280, - 986124, 2393328, 1049005, 2417944, 1208368, 2489516, 1352023, - 2572118, 1445283, 2856081, 1532997, 2742279, 1615877, 2915274, - 1808036, 2856871, 1806936, 3241747, 1622461, 2978558, 1841297, - 3010378, 1923666, 3271367, 2126700, 3070935, 1956958, 3107588, - 2128405, 3288872, 2114911, 3315952, 2406651, 3344038, 2370199, - 3368980, 2144361, 3305030, 2183803, 3401450, 2523102, 3405463, - 2452475, 3463355, 2421678, 3551968, 2431949, 3477251, 2148125, - 3244489, 2174090}; -static const int32_t spec2time_out_expected_2[kSamples] = { - 1691694, -2499988, -2035547, 1060469, 988634, -2044502, -306271, - 2041000, 201454, -2289456, 93694, 2129427, -369152, -1887834, - 860796, 2089102, -929424, -1673956, 1395291, 1785651, -1619673, - -1380109, 1963449, 1093311, -2111007, -840456, 2372786, 578119, - -2242702, 89774, 2463304, -132717, -2121480, 643634, 2277636, - -1125999, -1995858, 1543748, 2227861, -1483779, -1495491, 2102642, - 1833876, -1920568, -958378, 2485101, 772261, -2454257, -24942, - 2918714, 136838, -2500453, 816118, 3039735, -746560, -2365815, - 1586396, 2714951, -1511696, -1942334, 2571792, 2182827, -2325335, - -1311543, 3055970, 1367220, -2737182, -110626, 3889222, 631008, - -3280879, 853066, 4122279, -706638, -3334449, 2148311, 3993512, - -1846301, -3004894, 3426779, 3329522, -3165264, -2242423, 4756866, - 2557711, -4131280, -805259, 5702711, 1120592, -4852821, 743664, - 6476444, -621186, -5465828, 2815787, 6768835, -3017442, -5338409, - 5658126, 6838454, -5492288, -4682382, 8874947, 6153814, -8832561, - -2649251, 12817398, 4237692, -13000247, 1190661, 18986363, -115738, - -19693978, 9908367, 30660381, -10632635, -37962068, 47022884, 89744622, - -42087632, 40279224, -88869341, -47542383, 38572364, 10441576, -30339718, - -9926740, 19896578, 28009, -18886612, -1124047, 13232498, -4150304, - -12770551, 2637074, 9051831, -6162211, -8713972, 4557937, 5489716, - -6862312, -5532349, 5415449, 2791310, -6999367, -2790102, 5375806, - 546222, -6486452, -821261, 4994973, -1278840, -5645501, 1060484, - 3996285, -2503954, -4653629, 2220549, 3036977, -3282133, -3318585, - 2780636, 1789880, -4004589, -2041031, 3105373, 574819, -3992722, - -971004, 3001703, -676739, -3841508, 417284, 2897970, -1427018, - -3058480, 1189948, 2210960, -2268992, -2603272, 1949785, 1576172, - -2720404, -1891738, 2309456, 769178, -2975646, -707150, 2424652, - -88039, -2966660, -65452, 2320780, -957557, -2798978, 744640, - 1879794, -1672081, -2365319, 1253309, 1366383, -2204082, -1544367, - 1801452, 613828, -2531994, -983847, 2064842, 118326, -2613790, - -203220, 2219635, -730341, -2641861, 563557, 1765434, -1329916, - -2272927, 1037138, 1266725, -1939220, -1588643, 1754528, 816552, - -2376303, -1099167, 1864999, 122477, -2422762, -400027, 1889228, - -579916, -2490353, 287139, 2011318, -1176657, -2502978, 812896, - 1116502, -1940211}; -static const int16_t time2spec_out_expected_1[kSamples] = { - 20342, 23889, -10063, -9419, 3242, 7280, -2012, -5029, 332, 4478, - -97, -3244, -891, 3117, 773, -2204, -1335, 2009, 1236, -1469, - -1562, 1277, 1366, -815, -1619, 599, 1449, -177, -1507, 116, - 1294, 263, -1338, -244, 1059, 553, -1045, -549, 829, 826, - -731, -755, 516, 909, -427, -853, 189, 1004, -184, -828, - -108, 888, 72, -700, -280, 717, 342, -611, -534, 601, - 534, -374, -646, 399, 567, -171, -720, 234, 645, -11, - -712, -26, 593, 215, -643, -172, 536, 361, -527, -403, - 388, 550, -361, -480, 208, 623, -206, -585, 41, 578, - 12, -504, -182, 583, 218, -437, -339, 499, 263, -354, - -450, 347, 456, -193, -524, 212, 475, -74, -566, 94, - 511, 112, -577, -201, 408, 217, -546, -295, 338, 387, - -13, 4, -46, 2, -76, 103, -83, 108, -55, 100, - -150, 131, -156, 141, -171, 179, -190, 128, -227, 172, - -214, 215, -189, 265, -244, 322, -335, 337, -352, 358, - -368, 362, -355, 366, -381, 403, -395, 411, -392, 446, - -458, 504, -449, 507, -464, 452, -491, 481, -534, 486, - -516, 560, -535, 525, -537, 559, -554, 570, -616, 591, - -585, 627, -509, 588, -584, 547, -610, 580, -614, 635, - -620, 655, -554, 546, -591, 642, -590, 660, -656, 629, - -604, 620, -580, 617, -645, 648, -573, 612, -604, 584, - -571, 597, -562, 627, -550, 560, -606, 529, -584, 568, - -503, 532, -463, 512, -440, 399, -457, 437, -349, 278, - -317, 257, -220, 163, -8, -61, 18, -161, 367, -1306}; -static const int16_t time2spec_out_expected_2[kSamples] = { - 14283, -11552, -15335, 6626, 7554, -2150, -6309, 1307, 4523, -4, - -3908, -314, 3001, 914, -2715, -1042, 2094, 1272, -1715, -1399, - 1263, 1508, -1021, -1534, 735, 1595, -439, -1447, 155, 1433, - 22, -1325, -268, 1205, 424, -1030, -608, 950, 643, -733, - -787, 661, 861, -502, -888, 331, 852, -144, -849, 19, - 833, 99, -826, -154, 771, 368, -735, -459, 645, 513, - -491, -604, 431, 630, -314, -598, 183, 622, -78, -612, - -48, 641, 154, -645, -257, 610, 281, -529, -444, 450, - 441, -327, -506, 274, 476, -232, -570, 117, 554, -86, - -531, -21, 572, 151, -606, -221, 496, 322, -407, -388, - 407, 394, -268, -428, 280, 505, -115, -588, 19, 513, - -29, -539, -109, 468, 173, -501, -242, 442, 278, -478, - -680, 656, -659, 656, -669, 602, -688, 612, -667, 612, - -642, 627, -648, 653, -676, 596, -680, 655, -649, 678, - -672, 587, -608, 637, -645, 637, -620, 556, -580, 553, - -635, 518, -599, 583, -501, 536, -544, 473, -552, 583, - -511, 541, -532, 563, -486, 461, -453, 486, -388, 424, - -416, 432, -374, 399, -462, 364, -346, 293, -329, 331, - -313, 281, -247, 309, -337, 241, -190, 207, -194, 179, - -163, 155, -156, 117, -135, 107, -126, 29, -22, 81, - -8, 17, -61, -10, 8, -37, 80, -44, 72, -88, - 65, -89, 130, -114, 181, -215, 189, -245, 260, -288, - 294, -339, 344, -396, 407, -429, 438, -439, 485, -556, - 629, -612, 637, -645, 661, -737, 829, -830, 831, -1041}; - -class TransformTest : public ::testing::Test { - protected: - // Pass a function pointer to the Tester function. - void Time2SpecTester(Time2Spec Time2SpecFunction) { - // WebRtcIsacfix_Time2Spec functions hard coded the buffer lengths. It's a - // large buffer but we have to test it here. - int16_t data_in_1[kSamples] = {0}; - int16_t data_in_2[kSamples] = {0}; - int16_t data_out_1[kSamples] = {0}; - int16_t data_out_2[kSamples] = {0}; - - for (int i = 0; i < kSamples; i++) { - data_in_1[i] = i * i + 1777; - data_in_2[i] = WEBRTC_SPL_WORD16_MAX / (i + 1) + 17; - } - - Time2SpecFunction(data_in_1, data_in_2, data_out_1, data_out_2); - - for (int i = 0; i < kSamples; i++) { - // We don't require bit-exact for ARM assembly code. - EXPECT_LE(abs(time2spec_out_expected_1[i] - data_out_1[i]), 1); - EXPECT_LE(abs(time2spec_out_expected_2[i] - data_out_2[i]), 1); - } - } - - // Pass a function pointer to the Tester function. - void Spec2TimeTester(Spec2Time Spec2TimeFunction) { - // WebRtcIsacfix_Spec2Time functions hard coded the buffer lengths. It's a - // large buffer but we have to test it here. - int16_t data_in_1[kSamples] = {0}; - int16_t data_in_2[kSamples] = {0}; - int32_t data_out_1[kSamples] = {0}; - int32_t data_out_2[kSamples] = {0}; - for (int i = 0; i < kSamples; i++) { - data_in_1[i] = i * i + 1777; - data_in_2[i] = WEBRTC_SPL_WORD16_MAX / (i + 1) + 17; - } - - Spec2TimeFunction(data_in_1, data_in_2, data_out_1, data_out_2); - - for (int i = 0; i < kSamples; i++) { - // We don't require bit-exact for ARM assembly code. - EXPECT_LE(abs(spec2time_out_expected_1[i] - data_out_1[i]), 16); - EXPECT_LE(abs(spec2time_out_expected_2[i] - data_out_2[i]), 16); - } - } -}; - -TEST_F(TransformTest, Time2SpecTest) { - Time2SpecTester(WebRtcIsacfix_Time2SpecC); -#if defined(WEBRTC_HAS_NEON) - Time2SpecTester(WebRtcIsacfix_Time2SpecNeon); -#endif -} - -TEST_F(TransformTest, Spec2TimeTest) { - Spec2TimeTester(WebRtcIsacfix_Spec2TimeC); -#if defined(WEBRTC_HAS_NEON) - Spec2TimeTester(WebRtcIsacfix_Spec2TimeNeon); -#endif -} diff --git a/modules/audio_coding/codecs/isac/fix/test/isac_speed_test.cc b/modules/audio_coding/codecs/isac/fix/test/isac_speed_test.cc deleted file mode 100644 index 903ac64aff..0000000000 --- a/modules/audio_coding/codecs/isac/fix/test/isac_speed_test.cc +++ /dev/null @@ -1,120 +0,0 @@ -/* - * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#include "modules/audio_coding/codecs/isac/fix/include/isacfix.h" -#include "modules/audio_coding/codecs/isac/fix/source/settings.h" -#include "modules/audio_coding/codecs/tools/audio_codec_speed_test.h" -#include "rtc_base/checks.h" - -using std::string; - -namespace webrtc { - -static const int kIsacBlockDurationMs = 30; -static const int kIsacInputSamplingKhz = 16; -static const int kIsacOutputSamplingKhz = 16; - -class IsacSpeedTest : public AudioCodecSpeedTest { - protected: - IsacSpeedTest(); - void SetUp() override; - void TearDown() override; - float EncodeABlock(int16_t* in_data, - uint8_t* bit_stream, - size_t max_bytes, - size_t* encoded_bytes) override; - float DecodeABlock(const uint8_t* bit_stream, - size_t encoded_bytes, - int16_t* out_data) override; - ISACFIX_MainStruct* ISACFIX_main_inst_; -}; - -IsacSpeedTest::IsacSpeedTest() - : AudioCodecSpeedTest(kIsacBlockDurationMs, - kIsacInputSamplingKhz, - kIsacOutputSamplingKhz), - ISACFIX_main_inst_(NULL) {} - -void IsacSpeedTest::SetUp() { - AudioCodecSpeedTest::SetUp(); - - // Check whether the allocated buffer for the bit stream is large enough. - EXPECT_GE(max_bytes_, static_cast(STREAM_MAXW16_60MS)); - - // Create encoder memory. - EXPECT_EQ(0, WebRtcIsacfix_Create(&ISACFIX_main_inst_)); - EXPECT_EQ(0, WebRtcIsacfix_EncoderInit(ISACFIX_main_inst_, 1)); - WebRtcIsacfix_DecoderInit(ISACFIX_main_inst_); - // Set bitrate and block length. - EXPECT_EQ(0, WebRtcIsacfix_Control(ISACFIX_main_inst_, bit_rate_, - block_duration_ms_)); -} - -void IsacSpeedTest::TearDown() { - AudioCodecSpeedTest::TearDown(); - // Free memory. - EXPECT_EQ(0, WebRtcIsacfix_Free(ISACFIX_main_inst_)); -} - -float IsacSpeedTest::EncodeABlock(int16_t* in_data, - uint8_t* bit_stream, - size_t max_bytes, - size_t* encoded_bytes) { - // ISAC takes 10 ms everycall - const int subblocks = block_duration_ms_ / 10; - const int subblock_length = 10 * input_sampling_khz_; - int value = 0; - - clock_t clocks = clock(); - size_t pointer = 0; - for (int idx = 0; idx < subblocks; idx++, pointer += subblock_length) { - value = - WebRtcIsacfix_Encode(ISACFIX_main_inst_, &in_data[pointer], bit_stream); - if (idx == subblocks - 1) - EXPECT_GT(value, 0); - else - EXPECT_EQ(0, value); - } - clocks = clock() - clocks; - *encoded_bytes = static_cast(value); - RTC_DCHECK_LE(*encoded_bytes, max_bytes); - return 1000.0 * clocks / CLOCKS_PER_SEC; -} - -float IsacSpeedTest::DecodeABlock(const uint8_t* bit_stream, - size_t encoded_bytes, - int16_t* out_data) { - int value; - int16_t audio_type; - clock_t clocks = clock(); - value = WebRtcIsacfix_Decode(ISACFIX_main_inst_, bit_stream, encoded_bytes, - out_data, &audio_type); - clocks = clock() - clocks; - EXPECT_EQ(output_length_sample_, static_cast(value)); - return 1000.0 * clocks / CLOCKS_PER_SEC; -} - -TEST_P(IsacSpeedTest, IsacEncodeDecodeTest) { - size_t kDurationSec = 400; // Test audio length in second. - EncodeDecode(kDurationSec); -} - -const coding_param param_set[] = { - std::make_tuple(1, - 32000, - string("audio_coding/speech_mono_16kHz"), - string("pcm"), - true)}; - -INSTANTIATE_TEST_SUITE_P(AllTest, - IsacSpeedTest, - ::testing::ValuesIn(param_set)); - -} // namespace webrtc diff --git a/modules/audio_coding/codecs/isac/isac_webrtc_api_test.cc b/modules/audio_coding/codecs/isac/isac_webrtc_api_test.cc deleted file mode 100644 index cafca75e46..0000000000 --- a/modules/audio_coding/codecs/isac/isac_webrtc_api_test.cc +++ /dev/null @@ -1,346 +0,0 @@ -/* - * Copyright (c) 2020 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#include -#include -#include -#include - -#include "absl/strings/string_view.h" -#include "api/array_view.h" -#include "api/audio_codecs/isac/audio_decoder_isac_fix.h" -#include "api/audio_codecs/isac/audio_decoder_isac_float.h" -#include "api/audio_codecs/isac/audio_encoder_isac_fix.h" -#include "api/audio_codecs/isac/audio_encoder_isac_float.h" -#include "modules/audio_coding/test/PCMFile.h" -#include "rtc_base/checks.h" -#include "rtc_base/strings/string_builder.h" -#include "test/gtest.h" -#include "test/testsupport/file_utils.h" - -namespace webrtc { -namespace { - -constexpr int kPayloadType = 42; - -enum class IsacImpl { kFixed, kFloat }; - -absl::string_view IsacImplToString(IsacImpl impl) { - switch (impl) { - case IsacImpl::kFixed: - return "fixed"; - case IsacImpl::kFloat: - return "float"; - } -} - -std::unique_ptr GetPcmTestFileReader(int sample_rate_hz) { - std::string filename; - switch (sample_rate_hz) { - case 16000: - filename = test::ResourcePath("audio_coding/testfile16kHz", "pcm"); - break; - case 32000: - filename = test::ResourcePath("audio_coding/testfile32kHz", "pcm"); - break; - default: - RTC_DCHECK_NOTREACHED() - << "No test file available for " << sample_rate_hz << " Hz."; - } - auto pcm_file = std::make_unique(); - pcm_file->ReadStereo(false); - pcm_file->Open(filename, sample_rate_hz, "rb", /*auto_rewind=*/true); - pcm_file->FastForward(/*num_10ms_blocks=*/100); // Skip initial silence. - RTC_CHECK(!pcm_file->EndOfFile()); - return pcm_file; -} - -// Returns a view to the interleaved samples of an AudioFrame object. -rtc::ArrayView AudioFrameToView(const AudioFrame& audio_frame) { - return {audio_frame.data(), - audio_frame.samples_per_channel() * audio_frame.num_channels()}; -} - -std::unique_ptr CreateEncoder(IsacImpl impl, - int sample_rate_hz, - int frame_size_ms, - int bitrate_bps) { - RTC_CHECK(sample_rate_hz == 16000 || sample_rate_hz == 32000); - RTC_CHECK(frame_size_ms == 30 || frame_size_ms == 60); - RTC_CHECK_GT(bitrate_bps, 0); - switch (impl) { - case IsacImpl::kFixed: { - AudioEncoderIsacFix::Config config; - config.bit_rate = bitrate_bps; - config.frame_size_ms = frame_size_ms; - RTC_CHECK_EQ(16000, sample_rate_hz); - return AudioEncoderIsacFix::MakeAudioEncoder(config, kPayloadType); - } - case IsacImpl::kFloat: { - AudioEncoderIsacFloat::Config config; - config.bit_rate = bitrate_bps; - config.frame_size_ms = frame_size_ms; - config.sample_rate_hz = sample_rate_hz; - return AudioEncoderIsacFloat::MakeAudioEncoder(config, kPayloadType); - } - } -} - -std::unique_ptr CreateDecoder(IsacImpl impl, int sample_rate_hz) { - RTC_CHECK(sample_rate_hz == 16000 || sample_rate_hz == 32000); - switch (impl) { - case IsacImpl::kFixed: { - webrtc::AudioDecoderIsacFix::Config config; - RTC_CHECK_EQ(16000, sample_rate_hz); - return webrtc::AudioDecoderIsacFix::MakeAudioDecoder(config); - } - case IsacImpl::kFloat: { - webrtc::AudioDecoderIsacFloat::Config config; - config.sample_rate_hz = sample_rate_hz; - return webrtc::AudioDecoderIsacFloat::MakeAudioDecoder(config); - } - } -} - -struct EncoderTestParams { - IsacImpl impl; - int sample_rate_hz; - int frame_size_ms; -}; - -class EncoderTest : public testing::TestWithParam { - protected: - EncoderTest() = default; - IsacImpl GetIsacImpl() const { return GetParam().impl; } - int GetSampleRateHz() const { return GetParam().sample_rate_hz; } - int GetFrameSizeMs() const { return GetParam().frame_size_ms; } -}; - -TEST_P(EncoderTest, TestConfig) { - for (int bitrate_bps : {10000, 21000, 32000}) { - SCOPED_TRACE(bitrate_bps); - auto encoder = CreateEncoder(GetIsacImpl(), GetSampleRateHz(), - GetFrameSizeMs(), bitrate_bps); - EXPECT_EQ(GetSampleRateHz(), encoder->SampleRateHz()); - EXPECT_EQ(size_t{1}, encoder->NumChannels()); - EXPECT_EQ(bitrate_bps, encoder->GetTargetBitrate()); - } -} - -// Encodes an input audio sequence with a low and a high target bitrate and -// checks that the number of produces bytes in the first case is less than that -// of the second case. -TEST_P(EncoderTest, TestDifferentBitrates) { - auto pcm_file = GetPcmTestFileReader(GetSampleRateHz()); - constexpr int kLowBps = 20000; - constexpr int kHighBps = 25000; - auto encoder_low = CreateEncoder(GetIsacImpl(), GetSampleRateHz(), - GetFrameSizeMs(), kLowBps); - auto encoder_high = CreateEncoder(GetIsacImpl(), GetSampleRateHz(), - GetFrameSizeMs(), kHighBps); - int num_bytes_low = 0; - int num_bytes_high = 0; - constexpr int kNumFrames = 12; - for (int i = 0; i < kNumFrames; ++i) { - AudioFrame in; - pcm_file->Read10MsData(in); - rtc::Buffer low, high; - encoder_low->Encode(/*rtp_timestamp=*/0, AudioFrameToView(in), &low); - encoder_high->Encode(/*rtp_timestamp=*/0, AudioFrameToView(in), &high); - num_bytes_low += low.size(); - num_bytes_high += high.size(); - } - EXPECT_LT(num_bytes_low, num_bytes_high); -} - -// Encodes an input audio sequence first with a low, then with a high target -// bitrate *using the same encoder* and checks that the number of emitted bytes -// in the first case is less than in the second case. -TEST_P(EncoderTest, TestDynamicBitrateChange) { - constexpr int kLowBps = 20000; - constexpr int kHighBps = 25000; - constexpr int kStartBps = 30000; - auto encoder = CreateEncoder(GetIsacImpl(), GetSampleRateHz(), - GetFrameSizeMs(), kStartBps); - std::map num_bytes; - constexpr int kNumFrames = 200; // 2 seconds. - for (int bitrate_bps : {kLowBps, kHighBps}) { - auto pcm_file = GetPcmTestFileReader(GetSampleRateHz()); - encoder->OnReceivedTargetAudioBitrate(bitrate_bps); - for (int i = 0; i < kNumFrames; ++i) { - AudioFrame in; - pcm_file->Read10MsData(in); - rtc::Buffer buf; - encoder->Encode(/*rtp_timestamp=*/0, AudioFrameToView(in), &buf); - num_bytes[bitrate_bps] += buf.size(); - } - } - // kHighBps / kLowBps == 1.25, so require the high-bitrate run to produce at - // least 1.195 times the number of bytes. - EXPECT_LT(1.195 * num_bytes[kLowBps], num_bytes[kHighBps]); -} - -// Checks that, given a target bitrate, the encoder does not overshoot too much. -TEST_P(EncoderTest, DoNotOvershootTargetBitrate) { - for (int bitrate_bps : {10000, 15000, 20000, 26000, 32000}) { - SCOPED_TRACE(bitrate_bps); - auto pcm_file = GetPcmTestFileReader(GetSampleRateHz()); - auto e = CreateEncoder(GetIsacImpl(), GetSampleRateHz(), GetFrameSizeMs(), - bitrate_bps); - int num_bytes = 0; - constexpr int kNumFrames = 200; // 2 seconds. - for (int i = 0; i < kNumFrames; ++i) { - AudioFrame in; - pcm_file->Read10MsData(in); - rtc::Buffer encoded; - e->Encode(/*rtp_timestamp=*/0, AudioFrameToView(in), &encoded); - num_bytes += encoded.size(); - } - // Inverse of the duration of `kNumFrames` 10 ms frames (unit: seconds^-1). - constexpr float kAudioDurationInv = 100.f / kNumFrames; - const int measured_bitrate_bps = 8 * num_bytes * kAudioDurationInv; - EXPECT_LT(measured_bitrate_bps, bitrate_bps + 2250); // Max 2250 bps extra. - } -} - -// Creates tests for different encoder configurations and implementations. -INSTANTIATE_TEST_SUITE_P( - IsacApiTest, - EncoderTest, - ::testing::ValuesIn([] { - std::vector cases; - for (IsacImpl impl : {IsacImpl::kFloat, IsacImpl::kFixed}) { - for (int frame_size_ms : {30, 60}) { - cases.push_back({impl, 16000, frame_size_ms}); - } - } - cases.push_back({IsacImpl::kFloat, 32000, 30}); - return cases; - }()), - [](const ::testing::TestParamInfo& info) { - rtc::StringBuilder b; - const auto& p = info.param; - b << IsacImplToString(p.impl) << "_" << p.sample_rate_hz << "_" - << p.frame_size_ms; - return b.Release(); - }); - -struct DecoderTestParams { - IsacImpl impl; - int sample_rate_hz; -}; - -class DecoderTest : public testing::TestWithParam { - protected: - DecoderTest() = default; - IsacImpl GetIsacImpl() const { return GetParam().impl; } - int GetSampleRateHz() const { return GetParam().sample_rate_hz; } -}; - -TEST_P(DecoderTest, TestConfig) { - auto decoder = CreateDecoder(GetIsacImpl(), GetSampleRateHz()); - EXPECT_EQ(GetSampleRateHz(), decoder->SampleRateHz()); - EXPECT_EQ(size_t{1}, decoder->Channels()); -} - -// Creates tests for different decoder configurations and implementations. -INSTANTIATE_TEST_SUITE_P( - IsacApiTest, - DecoderTest, - ::testing::ValuesIn({DecoderTestParams{IsacImpl::kFixed, 16000}, - DecoderTestParams{IsacImpl::kFloat, 16000}, - DecoderTestParams{IsacImpl::kFloat, 32000}}), - [](const ::testing::TestParamInfo& info) { - const auto& p = info.param; - return (rtc::StringBuilder() - << IsacImplToString(p.impl) << "_" << p.sample_rate_hz) - .Release(); - }); - -struct EncoderDecoderPairTestParams { - int sample_rate_hz; - int frame_size_ms; - IsacImpl encoder_impl; - IsacImpl decoder_impl; -}; - -class EncoderDecoderPairTest - : public testing::TestWithParam { - protected: - EncoderDecoderPairTest() = default; - int GetSampleRateHz() const { return GetParam().sample_rate_hz; } - int GetEncoderFrameSizeMs() const { return GetParam().frame_size_ms; } - IsacImpl GetEncoderIsacImpl() const { return GetParam().encoder_impl; } - IsacImpl GetDecoderIsacImpl() const { return GetParam().decoder_impl; } - int GetEncoderFrameSize() const { - return GetEncoderFrameSizeMs() * GetSampleRateHz() / 1000; - } -}; - -// Checks that the number of encoded and decoded samples match. -TEST_P(EncoderDecoderPairTest, EncodeDecode) { - auto pcm_file = GetPcmTestFileReader(GetSampleRateHz()); - auto encoder = CreateEncoder(GetEncoderIsacImpl(), GetSampleRateHz(), - GetEncoderFrameSizeMs(), /*bitrate_bps=*/20000); - auto decoder = CreateDecoder(GetDecoderIsacImpl(), GetSampleRateHz()); - const int encoder_frame_size = GetEncoderFrameSize(); - std::vector out(encoder_frame_size); - size_t num_encoded_samples = 0; - size_t num_decoded_samples = 0; - constexpr int kNumFrames = 12; - for (int i = 0; i < kNumFrames; ++i) { - AudioFrame in; - pcm_file->Read10MsData(in); - rtc::Buffer encoded; - encoder->Encode(/*rtp_timestamp=*/0, AudioFrameToView(in), &encoded); - num_encoded_samples += in.samples_per_channel(); - if (encoded.empty()) { - continue; - } - // Decode. - const std::vector parse_result = - decoder->ParsePayload(std::move(encoded), /*timestamp=*/0); - EXPECT_EQ(parse_result.size(), size_t{1}); - auto decode_result = parse_result[0].frame->Decode(out); - EXPECT_TRUE(decode_result.has_value()); - EXPECT_EQ(out.size(), decode_result->num_decoded_samples); - num_decoded_samples += decode_result->num_decoded_samples; - } - EXPECT_EQ(num_encoded_samples, num_decoded_samples); -} - -// Creates tests for different encoder frame sizes and different -// encoder/decoder implementations. -INSTANTIATE_TEST_SUITE_P( - IsacApiTest, - EncoderDecoderPairTest, - ::testing::ValuesIn([] { - std::vector cases; - for (int frame_size_ms : {30, 60}) { - for (IsacImpl enc : {IsacImpl::kFloat, IsacImpl::kFixed}) { - for (IsacImpl dec : {IsacImpl::kFloat, IsacImpl::kFixed}) { - cases.push_back({16000, frame_size_ms, enc, dec}); - } - } - } - cases.push_back({32000, 30, IsacImpl::kFloat, IsacImpl::kFloat}); - return cases; - }()), - [](const ::testing::TestParamInfo& info) { - rtc::StringBuilder b; - const auto& p = info.param; - b << p.sample_rate_hz << "_" << p.frame_size_ms << "_" - << IsacImplToString(p.encoder_impl) << "_" - << IsacImplToString(p.decoder_impl); - return b.Release(); - }); - -} // namespace -} // namespace webrtc diff --git a/modules/audio_coding/codecs/isac/main/include/audio_decoder_isac.h b/modules/audio_coding/codecs/isac/main/include/audio_decoder_isac.h deleted file mode 100644 index fae2f3d4a7..0000000000 --- a/modules/audio_coding/codecs/isac/main/include/audio_decoder_isac.h +++ /dev/null @@ -1,22 +0,0 @@ -/* - * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_INCLUDE_AUDIO_DECODER_ISAC_H_ -#define MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_INCLUDE_AUDIO_DECODER_ISAC_H_ - -#include "modules/audio_coding/codecs/isac/audio_decoder_isac_t.h" -#include "modules/audio_coding/codecs/isac/main/source/isac_float_type.h" - -namespace webrtc { - -using AudioDecoderIsacFloatImpl = AudioDecoderIsacT; - -} // namespace webrtc -#endif // MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_INCLUDE_AUDIO_ENCODER_ISAC_H_ diff --git a/modules/audio_coding/codecs/isac/main/include/audio_encoder_isac.h b/modules/audio_coding/codecs/isac/main/include/audio_encoder_isac.h deleted file mode 100644 index dc32bcdde6..0000000000 --- a/modules/audio_coding/codecs/isac/main/include/audio_encoder_isac.h +++ /dev/null @@ -1,22 +0,0 @@ -/* - * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_INCLUDE_AUDIO_ENCODER_ISAC_H_ -#define MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_INCLUDE_AUDIO_ENCODER_ISAC_H_ - -#include "modules/audio_coding/codecs/isac/audio_encoder_isac_t.h" -#include "modules/audio_coding/codecs/isac/main/source/isac_float_type.h" - -namespace webrtc { - -using AudioEncoderIsacFloatImpl = AudioEncoderIsacT; - -} // namespace webrtc -#endif // MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_INCLUDE_AUDIO_ENCODER_ISAC_H_ diff --git a/modules/audio_coding/codecs/isac/main/include/isac.h b/modules/audio_coding/codecs/isac/main/include/isac.h deleted file mode 100644 index 3b05a8bcda..0000000000 --- a/modules/audio_coding/codecs/isac/main/include/isac.h +++ /dev/null @@ -1,617 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_INCLUDE_ISAC_H_ -#define MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_INCLUDE_ISAC_H_ - -#include - -#include "modules/audio_coding/codecs/isac/bandwidth_info.h" - -typedef struct WebRtcISACStruct ISACStruct; - -#if defined(__cplusplus) -extern "C" { -#endif - -/****************************************************************************** - * WebRtcIsac_Create(...) - * - * This function creates an ISAC instance, which will contain the state - * information for one coding/decoding channel. - * - * Input: - * - *ISAC_main_inst : a pointer to the coder instance. - * - * Return value : 0 - Ok - * -1 - Error - */ - -int16_t WebRtcIsac_Create(ISACStruct** ISAC_main_inst); - -/****************************************************************************** - * WebRtcIsac_Free(...) - * - * This function frees the ISAC instance created at the beginning. - * - * Input: - * - ISAC_main_inst : an ISAC instance. - * - * Return value : 0 - Ok - * -1 - Error - */ - -int16_t WebRtcIsac_Free(ISACStruct* ISAC_main_inst); - -/****************************************************************************** - * WebRtcIsac_EncoderInit(...) - * - * This function initializes an ISAC instance prior to the encoder calls. - * - * Input: - * - ISAC_main_inst : ISAC instance. - * - CodingMode : 0 -> Bit rate and frame length are - * automatically adjusted to available bandwidth - * on transmission channel, just valid if codec - * is created to work in wideband mode. - * 1 -> User sets a frame length and a target bit - * rate which is taken as the maximum - * short-term average bit rate. - * - * Return value : 0 - Ok - * -1 - Error - */ - -int16_t WebRtcIsac_EncoderInit(ISACStruct* ISAC_main_inst, int16_t CodingMode); - -/****************************************************************************** - * WebRtcIsac_Encode(...) - * - * This function encodes 10ms audio blocks and inserts it into a package. - * Input speech length has 160 samples if operating at 16 kHz sampling - * rate, or 320 if operating at 32 kHz sampling rate. The encoder buffers the - * input audio until the whole frame is buffered then proceeds with encoding. - * - * - * Input: - * - ISAC_main_inst : ISAC instance. - * - speechIn : input speech vector. - * - * Output: - * - encoded : the encoded data vector - * - * Return value: - * : >0 - Length (in bytes) of coded data - * : 0 - The buffer didn't reach the chosen - * frame-size so it keeps buffering speech - * samples. - * : -1 - Error - */ - -int WebRtcIsac_Encode(ISACStruct* ISAC_main_inst, - const int16_t* speechIn, - uint8_t* encoded); - -/****************************************************************************** - * WebRtcIsac_DecoderInit(...) - * - * This function initializes an ISAC instance prior to the decoder calls. - * - * Input: - * - ISAC_main_inst : ISAC instance. - */ - -void WebRtcIsac_DecoderInit(ISACStruct* ISAC_main_inst); - -/****************************************************************************** - * WebRtcIsac_UpdateBwEstimate(...) - * - * This function updates the estimate of the bandwidth. - * - * Input: - * - ISAC_main_inst : ISAC instance. - * - encoded : encoded ISAC frame(s). - * - packet_size : size of the packet. - * - rtp_seq_number : the RTP number of the packet. - * - send_ts : the RTP send timestamp, given in samples - * - arr_ts : the arrival time of the packet (from NetEq) - * in samples. - * - * Return value : 0 - Ok - * -1 - Error - */ - -int16_t WebRtcIsac_UpdateBwEstimate(ISACStruct* ISAC_main_inst, - const uint8_t* encoded, - size_t packet_size, - uint16_t rtp_seq_number, - uint32_t send_ts, - uint32_t arr_ts); - -/****************************************************************************** - * WebRtcIsac_Decode(...) - * - * This function decodes an ISAC frame. At 16 kHz sampling rate, the length - * of the output audio could be either 480 or 960 samples, equivalent to - * 30 or 60 ms respectively. At 32 kHz sampling rate, the length of the - * output audio is 960 samples, which is 30 ms. - * - * Input: - * - ISAC_main_inst : ISAC instance. - * - encoded : encoded ISAC frame(s). - * - len : bytes in encoded vector. - * - * Output: - * - decoded : The decoded vector. - * - * Return value : >0 - number of samples in decoded vector. - * -1 - Error. - */ - -int WebRtcIsac_Decode(ISACStruct* ISAC_main_inst, - const uint8_t* encoded, - size_t len, - int16_t* decoded, - int16_t* speechType); - -/****************************************************************************** - * WebRtcIsac_DecodePlc(...) - * - * This function conducts PLC for ISAC frame(s). Output speech length - * will be a multiple of frames, i.e. multiples of 30 ms audio. Therefore, - * the output is multiple of 480 samples if operating at 16 kHz and multiple - * of 960 if operating at 32 kHz. - * - * Input: - * - ISAC_main_inst : ISAC instance. - * - noOfLostFrames : Number of PLC frames to produce. - * - * Output: - * - decoded : The decoded vector. - * - * Return value : Number of samples in decoded PLC vector - */ - -size_t WebRtcIsac_DecodePlc(ISACStruct* ISAC_main_inst, - int16_t* decoded, - size_t noOfLostFrames); - -/****************************************************************************** - * WebRtcIsac_Control(...) - * - * This function sets the limit on the short-term average bit-rate and the - * frame length. Should be used only in Instantaneous mode. At 16 kHz sampling - * rate, an average bit-rate between 10000 to 32000 bps is valid and a - * frame-size of 30 or 60 ms is acceptable. At 32 kHz, an average bit-rate - * between 10000 to 56000 is acceptable, and the valid frame-size is 30 ms. - * - * Input: - * - ISAC_main_inst : ISAC instance. - * - rate : limit on the short-term average bit rate, - * in bits/second. - * - framesize : frame-size in millisecond. - * - * Return value : 0 - ok - * -1 - Error - */ - -int16_t WebRtcIsac_Control(ISACStruct* ISAC_main_inst, - int32_t rate, - int framesize); - -void WebRtcIsac_SetInitialBweBottleneck(ISACStruct* ISAC_main_inst, - int bottleneck_bits_per_second); - -/****************************************************************************** - * WebRtcIsac_ControlBwe(...) - * - * This function sets the initial values of bottleneck and frame-size if - * iSAC is used in channel-adaptive mode. Therefore, this API is not - * applicable if the codec is created to operate in super-wideband mode. - * - * Through this API, users can enforce a frame-size for all values of - * bottleneck. Then iSAC will not automatically change the frame-size. - * - * - * Input: - * - ISAC_main_inst : ISAC instance. - * - rateBPS : initial value of bottleneck in bits/second - * 10000 <= rateBPS <= 56000 is accepted - * For default bottleneck set rateBPS = 0 - * - frameSizeMs : number of milliseconds per frame (30 or 60) - * - enforceFrameSize : 1 to enforce the given frame-size through - * out the adaptation process, 0 to let iSAC - * change the frame-size if required. - * - * Return value : 0 - ok - * -1 - Error - */ - -int16_t WebRtcIsac_ControlBwe(ISACStruct* ISAC_main_inst, - int32_t rateBPS, - int frameSizeMs, - int16_t enforceFrameSize); - -/****************************************************************************** - * WebRtcIsac_ReadFrameLen(...) - * - * This function returns the length of the frame represented in the packet. - * - * Input: - * - encoded : Encoded bit-stream - * - * Output: - * - frameLength : Length of frame in packet (in samples) - * - */ - -int16_t WebRtcIsac_ReadFrameLen(const ISACStruct* ISAC_main_inst, - const uint8_t* encoded, - int16_t* frameLength); - -/****************************************************************************** - * WebRtcIsac_version(...) - * - * This function returns the version number. - * - * Output: - * - version : Pointer to character string - * - */ - -void WebRtcIsac_version(char* version); - -/****************************************************************************** - * WebRtcIsac_GetErrorCode(...) - * - * This function can be used to check the error code of an iSAC instance. When - * a function returns -1 a error code will be set for that instance. The - * function below extract the code of the last error that occurred in the - * specified instance. - * - * Input: - * - ISAC_main_inst : ISAC instance - * - * Return value : Error code - */ - -int16_t WebRtcIsac_GetErrorCode(ISACStruct* ISAC_main_inst); - -/**************************************************************************** - * WebRtcIsac_GetUplinkBw(...) - * - * This function outputs the target bottleneck of the codec. In - * channel-adaptive mode, the target bottleneck is specified through in-band - * signalling retreived by bandwidth estimator. - * In channel-independent, also called instantaneous mode, the target - * bottleneck is provided to the encoder by calling xxx_control(...). If - * xxx_control is never called the default values is returned. The default - * value for bottleneck at 16 kHz encoder sampling rate is 32000 bits/sec, - * and it is 56000 bits/sec for 32 kHz sampling rate. - * Note that the output is the iSAC internal operating bottleneck which might - * differ slightly from the one provided through xxx_control(). - * - * Input: - * - ISAC_main_inst : iSAC instance - * - * Output: - * - *bottleneck : bottleneck in bits/sec - * - * Return value : -1 if error happens - * 0 bit-rates computed correctly. - */ - -int16_t WebRtcIsac_GetUplinkBw(ISACStruct* ISAC_main_inst, int32_t* bottleneck); - -/****************************************************************************** - * WebRtcIsac_SetMaxPayloadSize(...) - * - * This function sets a limit for the maximum payload size of iSAC. The same - * value is used both for 30 and 60 ms packets. If the encoder sampling rate - * is 16 kHz the maximum payload size is between 120 and 400 bytes. If the - * encoder sampling rate is 32 kHz the maximum payload size is between 120 - * and 600 bytes. - * - * If an out of range limit is used, the function returns -1, but the closest - * valid value will be applied. - * - * --------------- - * IMPORTANT NOTES - * --------------- - * The size of a packet is limited to the minimum of 'max-payload-size' and - * 'max-rate.' For instance, let's assume the max-payload-size is set to - * 170 bytes, and max-rate is set to 40 kbps. Note that a limit of 40 kbps - * translates to 150 bytes for 30ms frame-size & 300 bytes for 60ms - * frame-size. Then a packet with a frame-size of 30 ms is limited to 150, - * i.e. min(170, 150), and a packet with 60 ms frame-size is limited to - * 170 bytes, i.e. min(170, 300). - * - * Input: - * - ISAC_main_inst : iSAC instance - * - maxPayloadBytes : maximum size of the payload in bytes - * valid values are between 120 and 400 bytes - * if encoder sampling rate is 16 kHz. For - * 32 kHz encoder sampling rate valid values - * are between 120 and 600 bytes. - * - * Return value : 0 if successful - * -1 if error happens - */ - -int16_t WebRtcIsac_SetMaxPayloadSize(ISACStruct* ISAC_main_inst, - int16_t maxPayloadBytes); - -/****************************************************************************** - * WebRtcIsac_SetMaxRate(...) - * - * This function sets the maximum rate which the codec may not exceed for - * any signal packet. The maximum rate is defined and payload-size per - * frame-size in bits per second. - * - * The codec has a maximum rate of 53400 bits per second (200 bytes per 30 - * ms) if the encoder sampling rate is 16kHz, and 160 kbps (600 bytes/30 ms) - * if the encoder sampling rate is 32 kHz. - * - * It is possible to set a maximum rate between 32000 and 53400 bits/sec - * in wideband mode, and 32000 to 160000 bits/sec in super-wideband mode. - * - * If an out of range limit is used, the function returns -1, but the closest - * valid value will be applied. - * - * --------------- - * IMPORTANT NOTES - * --------------- - * The size of a packet is limited to the minimum of 'max-payload-size' and - * 'max-rate.' For instance, let's assume the max-payload-size is set to - * 170 bytes, and max-rate is set to 40 kbps. Note that a limit of 40 kbps - * translates to 150 bytes for 30ms frame-size & 300 bytes for 60ms - * frame-size. Then a packet with a frame-size of 30 ms is limited to 150, - * i.e. min(170, 150), and a packet with 60 ms frame-size is limited to - * 170 bytes, min(170, 300). - * - * Input: - * - ISAC_main_inst : iSAC instance - * - maxRate : maximum rate in bits per second, - * valid values are 32000 to 53400 bits/sec in - * wideband mode, and 32000 to 160000 bits/sec in - * super-wideband mode. - * - * Return value : 0 if successful - * -1 if error happens - */ - -int16_t WebRtcIsac_SetMaxRate(ISACStruct* ISAC_main_inst, int32_t maxRate); - -/****************************************************************************** - * WebRtcIsac_DecSampRate() - * Return the sampling rate of the decoded audio. - * - * Input: - * - ISAC_main_inst : iSAC instance - * - * Return value : sampling frequency in Hertz. - * - */ - -uint16_t WebRtcIsac_DecSampRate(ISACStruct* ISAC_main_inst); - -/****************************************************************************** - * WebRtcIsac_EncSampRate() - * - * Input: - * - ISAC_main_inst : iSAC instance - * - * Return value : sampling rate in Hertz. - * - */ - -uint16_t WebRtcIsac_EncSampRate(ISACStruct* ISAC_main_inst); - -/****************************************************************************** - * WebRtcIsac_SetDecSampRate() - * Set the sampling rate of the decoder. Initialization of the decoder WILL - * NOT overwrite the sampling rate of the encoder. The default value is 16 kHz - * which is set when the instance is created. - * - * Input: - * - ISAC_main_inst : iSAC instance - * - sampRate : sampling rate in Hertz. - * - * Return value : 0 if successful - * -1 if failed. - */ - -int16_t WebRtcIsac_SetDecSampRate(ISACStruct* ISAC_main_inst, - uint16_t samp_rate_hz); - -/****************************************************************************** - * WebRtcIsac_SetEncSampRate() - * Set the sampling rate of the encoder. Initialization of the encoder WILL - * NOT overwrite the sampling rate of the encoder. The default value is 16 kHz - * which is set when the instance is created. The encoding-mode and the - * bottleneck remain unchanged by this call, however, the maximum rate and - * maximum payload-size will reset to their default value. - * - * Input: - * - ISAC_main_inst : iSAC instance - * - sampRate : sampling rate in Hertz. - * - * Return value : 0 if successful - * -1 if failed. - */ - -int16_t WebRtcIsac_SetEncSampRate(ISACStruct* ISAC_main_inst, - uint16_t sample_rate_hz); - -/****************************************************************************** - * WebRtcIsac_GetNewBitStream(...) - * - * This function returns encoded data, with the received bwe-index in the - * stream. If the rate is set to a value less than bottleneck of codec - * the new bistream will be re-encoded with the given target rate. - * It should always return a complete packet, i.e. only called once - * even for 60 msec frames. - * - * NOTE 1! This function does not write in the ISACStruct, it is not allowed. - * NOTE 2! Currently not implemented for SWB mode. - * NOTE 3! Rates larger than the bottleneck of the codec will be limited - * to the current bottleneck. - * - * Input: - * - ISAC_main_inst : ISAC instance. - * - bweIndex : Index of bandwidth estimate to put in new - * bitstream - * - rate : target rate of the transcoder is bits/sec. - * Valid values are the accepted rate in iSAC, - * i.e. 10000 to 56000. - * - isRCU : if the new bit-stream is an RCU - * stream. Note that the rate parameter always indicates the target rate of the - * main payload, regardless of 'isRCU' value. - * - * Output: - * - encoded : The encoded data vector - * - * Return value : >0 - Length (in bytes) of coded data - * -1 - Error or called in SWB mode - * NOTE! No error code is written to - * the struct since it is only allowed to read - * the struct. - */ -int16_t WebRtcIsac_GetNewBitStream(ISACStruct* ISAC_main_inst, - int16_t bweIndex, - int16_t jitterInfo, - int32_t rate, - uint8_t* encoded, - int16_t isRCU); - -/**************************************************************************** - * WebRtcIsac_GetDownLinkBwIndex(...) - * - * This function returns index representing the Bandwidth estimate from - * other side to this side. - * - * Input: - * - ISAC_main_inst : iSAC struct - * - * Output: - * - bweIndex : Bandwidth estimate to transmit to other side. - * - */ - -int16_t WebRtcIsac_GetDownLinkBwIndex(ISACStruct* ISAC_main_inst, - int16_t* bweIndex, - int16_t* jitterInfo); - -/**************************************************************************** - * WebRtcIsac_UpdateUplinkBw(...) - * - * This function takes an index representing the Bandwidth estimate from - * this side to other side and updates BWE. - * - * Input: - * - ISAC_main_inst : iSAC struct - * - bweIndex : Bandwidth estimate from other side. - * - */ - -int16_t WebRtcIsac_UpdateUplinkBw(ISACStruct* ISAC_main_inst, int16_t bweIndex); - -/**************************************************************************** - * WebRtcIsac_ReadBwIndex(...) - * - * This function returns the index of the Bandwidth estimate from the bitstream. - * - * Input: - * - encoded : Encoded bitstream - * - * Output: - * - frameLength : Length of frame in packet (in samples) - * - bweIndex : Bandwidth estimate in bitstream - * - */ - -int16_t WebRtcIsac_ReadBwIndex(const uint8_t* encoded, int16_t* bweIndex); - -/******************************************************************************* - * WebRtcIsac_GetNewFrameLen(...) - * - * returns the frame lenght (in samples) of the next packet. In the case of - * channel-adaptive mode, iSAC decides on its frame lenght based on the - * estimated bottleneck this allows a user to prepare for the next packet (at - * the encoder) - * - * The primary usage is in CE to make the iSAC works in channel-adaptive mode - * - * Input: - * - ISAC_main_inst : iSAC struct - * - * Return Value : frame lenght in samples - * - */ - -int16_t WebRtcIsac_GetNewFrameLen(ISACStruct* ISAC_main_inst); - -/**************************************************************************** - * WebRtcIsac_GetRedPayload(...) - * - * Populates "encoded" with the redundant payload of the recently encoded - * frame. This function has to be called once that WebRtcIsac_Encode(...) - * returns a positive value. Regardless of the frame-size this function will - * be called only once after encoding is completed. - * - * Input: - * - ISAC_main_inst : iSAC struct - * - * Output: - * - encoded : the encoded data vector - * - * - * Return value: - * : >0 - Length (in bytes) of coded data - * : -1 - Error - * - * - */ -int16_t WebRtcIsac_GetRedPayload(ISACStruct* ISAC_main_inst, uint8_t* encoded); - -/**************************************************************************** - * WebRtcIsac_DecodeRcu(...) - * - * This function decodes a redundant (RCU) iSAC frame. Function is called in - * NetEq with a stored RCU payload i case of packet loss. Output speech length - * will be a multiple of 480 samples: 480 or 960 samples, - * depending on the framesize (30 or 60 ms). - * - * Input: - * - ISAC_main_inst : ISAC instance. - * - encoded : encoded ISAC RCU frame(s) - * - len : bytes in encoded vector - * - * Output: - * - decoded : The decoded vector - * - * Return value : >0 - number of samples in decoded vector - * -1 - Error - */ -int WebRtcIsac_DecodeRcu(ISACStruct* ISAC_main_inst, - const uint8_t* encoded, - size_t len, - int16_t* decoded, - int16_t* speechType); - -/* If `inst` is a decoder but not an encoder: tell it what sample rate the - encoder is using, for bandwidth estimation purposes. */ -void WebRtcIsac_SetEncSampRateInDecoder(ISACStruct* inst, int sample_rate_hz); - -#if defined(__cplusplus) -} -#endif - -#endif /* MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_INCLUDE_ISAC_H_ */ diff --git a/modules/audio_coding/codecs/isac/main/source/arith_routines.c b/modules/audio_coding/codecs/isac/main/source/arith_routines.c deleted file mode 100644 index 9d5c6930b1..0000000000 --- a/modules/audio_coding/codecs/isac/main/source/arith_routines.c +++ /dev/null @@ -1,60 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#include "modules/audio_coding/codecs/isac/main/source/arith_routines.h" -#include "modules/audio_coding/codecs/isac/main/source/settings.h" - - -/* - * terminate and return byte stream; - * returns the number of bytes in the stream - */ -int WebRtcIsac_EncTerminate(Bitstr *streamdata) /* in-/output struct containing bitstream */ -{ - uint8_t *stream_ptr; - - - /* point to the right place in the stream buffer */ - stream_ptr = streamdata->stream + streamdata->stream_index; - - /* find minimum length (determined by current interval width) */ - if ( streamdata->W_upper > 0x01FFFFFF ) - { - streamdata->streamval += 0x01000000; - /* add carry to buffer */ - if (streamdata->streamval < 0x01000000) - { - /* propagate carry */ - while ( !(++(*--stream_ptr)) ); - /* put pointer back to the old value */ - stream_ptr = streamdata->stream + streamdata->stream_index; - } - /* write remaining data to bitstream */ - *stream_ptr++ = (uint8_t) (streamdata->streamval >> 24); - } - else - { - streamdata->streamval += 0x00010000; - /* add carry to buffer */ - if (streamdata->streamval < 0x00010000) - { - /* propagate carry */ - while ( !(++(*--stream_ptr)) ); - /* put pointer back to the old value */ - stream_ptr = streamdata->stream + streamdata->stream_index; - } - /* write remaining data to bitstream */ - *stream_ptr++ = (uint8_t) (streamdata->streamval >> 24); - *stream_ptr++ = (uint8_t) ((streamdata->streamval >> 16) & 0x00FF); - } - - /* calculate stream length */ - return (int)(stream_ptr - streamdata->stream); -} diff --git a/modules/audio_coding/codecs/isac/main/source/arith_routines.h b/modules/audio_coding/codecs/isac/main/source/arith_routines.h deleted file mode 100644 index 3f9f6de7bb..0000000000 --- a/modules/audio_coding/codecs/isac/main/source/arith_routines.h +++ /dev/null @@ -1,67 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/* - * arith_routines.h - * - * Functions for arithmetic coding. - * - */ - -#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_ARITH_ROUTINES_H_ -#define MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_ARITH_ROUTINES_H_ - -#include "modules/audio_coding/codecs/isac/main/source/structs.h" - -int WebRtcIsac_EncLogisticMulti2( - Bitstr* streamdata, /* in-/output struct containing bitstream */ - int16_t* dataQ7, /* input: data vector */ - const uint16_t* - env, /* input: side info vector defining the width of the pdf */ - int N, /* input: data vector length */ - int16_t isSWB12kHz); /* if the codec is working in 12kHz bandwidth */ - -/* returns the number of bytes in the stream */ -int WebRtcIsac_EncTerminate( - Bitstr* streamdata); /* in-/output struct containing bitstream */ - -/* returns the number of bytes in the stream so far */ -int WebRtcIsac_DecLogisticMulti2( - int16_t* data, /* output: data vector */ - Bitstr* streamdata, /* in-/output struct containing bitstream */ - const uint16_t* - env, /* input: side info vector defining the width of the pdf */ - const int16_t* dither, /* input: dither vector */ - int N, /* input: data vector length */ - int16_t isSWB12kHz); /* if the codec is working in 12kHz bandwidth */ - -void WebRtcIsac_EncHistMulti( - Bitstr* streamdata, /* in-/output struct containing bitstream */ - const int* data, /* input: data vector */ - const uint16_t* const* cdf, /* input: array of cdf arrays */ - int N); /* input: data vector length */ - -int WebRtcIsac_DecHistBisectMulti( - int* data, /* output: data vector */ - Bitstr* streamdata, /* in-/output struct containing bitstream */ - const uint16_t* const* cdf, /* input: array of cdf arrays */ - const uint16_t* - cdf_size, /* input: array of cdf table sizes+1 (power of two: 2^k) */ - int N); /* input: data vector length */ - -int WebRtcIsac_DecHistOneStepMulti( - int* data, /* output: data vector */ - Bitstr* streamdata, /* in-/output struct containing bitstream */ - const uint16_t* const* cdf, /* input: array of cdf arrays */ - const uint16_t* - init_index, /* input: vector of initial cdf table search entries */ - int N); /* input: data vector length */ - -#endif /* MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_ARITH_ROUTINES_H_ */ diff --git a/modules/audio_coding/codecs/isac/main/source/arith_routines_hist.c b/modules/audio_coding/codecs/isac/main/source/arith_routines_hist.c deleted file mode 100644 index e948979fd7..0000000000 --- a/modules/audio_coding/codecs/isac/main/source/arith_routines_hist.c +++ /dev/null @@ -1,291 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#include "modules/audio_coding/codecs/isac/main/source/settings.h" -#include "modules/audio_coding/codecs/isac/main/source/arith_routines.h" - - -/* - * code symbols into arithmetic bytestream - */ -void WebRtcIsac_EncHistMulti(Bitstr *streamdata, /* in-/output struct containing bitstream */ - const int *data, /* input: data vector */ - const uint16_t *const *cdf, /* input: array of cdf arrays */ - const int N) /* input: data vector length */ -{ - uint32_t W_lower, W_upper; - uint32_t W_upper_LSB, W_upper_MSB; - uint8_t *stream_ptr; - uint8_t *stream_ptr_carry; - uint32_t cdf_lo, cdf_hi; - int k; - - - /* point to beginning of stream buffer */ - stream_ptr = streamdata->stream + streamdata->stream_index; - W_upper = streamdata->W_upper; - - for (k=N; k>0; k--) - { - /* fetch cdf_lower and cdf_upper from cdf tables */ - cdf_lo = (uint32_t) *(*cdf + *data); - cdf_hi = (uint32_t) *(*cdf++ + *data++ + 1); - - /* update interval */ - W_upper_LSB = W_upper & 0x0000FFFF; - W_upper_MSB = W_upper >> 16; - W_lower = W_upper_MSB * cdf_lo; - W_lower += (W_upper_LSB * cdf_lo) >> 16; - W_upper = W_upper_MSB * cdf_hi; - W_upper += (W_upper_LSB * cdf_hi) >> 16; - - /* shift interval such that it begins at zero */ - W_upper -= ++W_lower; - - /* add integer to bitstream */ - streamdata->streamval += W_lower; - - /* handle carry */ - if (streamdata->streamval < W_lower) - { - /* propagate carry */ - stream_ptr_carry = stream_ptr; - while (!(++(*--stream_ptr_carry))); - } - - /* renormalize interval, store most significant byte of streamval and update streamval */ - while ( !(W_upper & 0xFF000000) ) /* W_upper < 2^24 */ - { - W_upper <<= 8; - *stream_ptr++ = (uint8_t) (streamdata->streamval >> 24); - streamdata->streamval <<= 8; - } - } - - /* calculate new stream_index */ - streamdata->stream_index = (int)(stream_ptr - streamdata->stream); - streamdata->W_upper = W_upper; - - return; -} - - - -/* - * function to decode more symbols from the arithmetic bytestream, using method of bisection - * cdf tables should be of size 2^k-1 (which corresponds to an alphabet size of 2^k-2) - */ -int WebRtcIsac_DecHistBisectMulti(int *data, /* output: data vector */ - Bitstr *streamdata, /* in-/output struct containing bitstream */ - const uint16_t *const *cdf, /* input: array of cdf arrays */ - const uint16_t *cdf_size, /* input: array of cdf table sizes+1 (power of two: 2^k) */ - const int N) /* input: data vector length */ -{ - uint32_t W_lower, W_upper; - uint32_t W_tmp; - uint32_t W_upper_LSB, W_upper_MSB; - uint32_t streamval; - const uint8_t *stream_ptr; - const uint16_t *cdf_ptr; - int size_tmp; - int k; - - W_lower = 0; //to remove warning -DH - stream_ptr = streamdata->stream + streamdata->stream_index; - W_upper = streamdata->W_upper; - if (W_upper == 0) - /* Should not be possible in normal operation */ - return -2; - - if (streamdata->stream_index == 0) /* first time decoder is called for this stream */ - { - /* read first word from bytestream */ - streamval = *stream_ptr << 24; - streamval |= *++stream_ptr << 16; - streamval |= *++stream_ptr << 8; - streamval |= *++stream_ptr; - } else { - streamval = streamdata->streamval; - } - - for (k=N; k>0; k--) - { - /* find the integer *data for which streamval lies in [W_lower+1, W_upper] */ - W_upper_LSB = W_upper & 0x0000FFFF; - W_upper_MSB = W_upper >> 16; - - /* start halfway the cdf range */ - size_tmp = *cdf_size++ >> 1; - cdf_ptr = *cdf + (size_tmp - 1); - - /* method of bisection */ - for ( ;; ) - { - W_tmp = W_upper_MSB * *cdf_ptr; - W_tmp += (W_upper_LSB * *cdf_ptr) >> 16; - size_tmp >>= 1; - if (size_tmp == 0) break; - if (streamval > W_tmp) - { - W_lower = W_tmp; - cdf_ptr += size_tmp; - } else { - W_upper = W_tmp; - cdf_ptr -= size_tmp; - } - } - if (streamval > W_tmp) - { - W_lower = W_tmp; - *data++ = (int)(cdf_ptr - *cdf++); - } else { - W_upper = W_tmp; - *data++ = (int)(cdf_ptr - *cdf++ - 1); - } - - /* shift interval to start at zero */ - W_upper -= ++W_lower; - - /* add integer to bitstream */ - streamval -= W_lower; - - /* renormalize interval and update streamval */ - while ( !(W_upper & 0xFF000000) ) /* W_upper < 2^24 */ - { - /* read next byte from stream */ - streamval = (streamval << 8) | *++stream_ptr; - W_upper <<= 8; - } - - if (W_upper == 0) - /* Should not be possible in normal operation */ - return -2; - - - } - - streamdata->stream_index = (int)(stream_ptr - streamdata->stream); - streamdata->W_upper = W_upper; - streamdata->streamval = streamval; - - - /* find number of bytes in original stream (determined by current interval width) */ - if ( W_upper > 0x01FFFFFF ) - return streamdata->stream_index - 2; - else - return streamdata->stream_index - 1; -} - - - -/* - * function to decode more symbols from the arithmetic bytestream, taking single step up or - * down at a time - * cdf tables can be of arbitrary size, but large tables may take a lot of iterations - */ -int WebRtcIsac_DecHistOneStepMulti(int *data, /* output: data vector */ - Bitstr *streamdata, /* in-/output struct containing bitstream */ - const uint16_t *const *cdf, /* input: array of cdf arrays */ - const uint16_t *init_index, /* input: vector of initial cdf table search entries */ - const int N) /* input: data vector length */ -{ - uint32_t W_lower, W_upper; - uint32_t W_tmp; - uint32_t W_upper_LSB, W_upper_MSB; - uint32_t streamval; - const uint8_t *stream_ptr; - const uint16_t *cdf_ptr; - int k; - - - stream_ptr = streamdata->stream + streamdata->stream_index; - W_upper = streamdata->W_upper; - if (W_upper == 0) - /* Should not be possible in normal operation */ - return -2; - - if (streamdata->stream_index == 0) /* first time decoder is called for this stream */ - { - /* read first word from bytestream */ - streamval = (uint32_t)(*stream_ptr) << 24; - streamval |= (uint32_t)(*++stream_ptr) << 16; - streamval |= (uint32_t)(*++stream_ptr) << 8; - streamval |= (uint32_t)(*++stream_ptr); - } else { - streamval = streamdata->streamval; - } - - - for (k=N; k>0; k--) - { - /* find the integer *data for which streamval lies in [W_lower+1, W_upper] */ - W_upper_LSB = W_upper & 0x0000FFFF; - W_upper_MSB = W_upper >> 16; - - /* start at the specified table entry */ - cdf_ptr = *cdf + (*init_index++); - W_tmp = W_upper_MSB * *cdf_ptr; - W_tmp += (W_upper_LSB * *cdf_ptr) >> 16; - if (streamval > W_tmp) - { - for ( ;; ) - { - W_lower = W_tmp; - if (cdf_ptr[0]==65535) - /* range check */ - return -3; - W_tmp = W_upper_MSB * *++cdf_ptr; - W_tmp += (W_upper_LSB * *cdf_ptr) >> 16; - if (streamval <= W_tmp) break; - } - W_upper = W_tmp; - *data++ = (int)(cdf_ptr - *cdf++ - 1); - } else { - for ( ;; ) - { - W_upper = W_tmp; - --cdf_ptr; - if (cdf_ptr<*cdf) { - /* range check */ - return -3; - } - W_tmp = W_upper_MSB * *cdf_ptr; - W_tmp += (W_upper_LSB * *cdf_ptr) >> 16; - if (streamval > W_tmp) break; - } - W_lower = W_tmp; - *data++ = (int)(cdf_ptr - *cdf++); - } - - /* shift interval to start at zero */ - W_upper -= ++W_lower; - /* add integer to bitstream */ - streamval -= W_lower; - - /* renormalize interval and update streamval */ - while ( !(W_upper & 0xFF000000) ) /* W_upper < 2^24 */ - { - /* read next byte from stream */ - streamval = (streamval << 8) | *++stream_ptr; - W_upper <<= 8; - } - } - - streamdata->stream_index = (int)(stream_ptr - streamdata->stream); - streamdata->W_upper = W_upper; - streamdata->streamval = streamval; - - - /* find number of bytes in original stream (determined by current interval width) */ - if ( W_upper > 0x01FFFFFF ) - return streamdata->stream_index - 2; - else - return streamdata->stream_index - 1; -} diff --git a/modules/audio_coding/codecs/isac/main/source/arith_routines_logist.c b/modules/audio_coding/codecs/isac/main/source/arith_routines_logist.c deleted file mode 100644 index 777780f54f..0000000000 --- a/modules/audio_coding/codecs/isac/main/source/arith_routines_logist.c +++ /dev/null @@ -1,303 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/* - * arith_routines.h - * - * This file contains functions for arithmatically encoding and - * decoding DFT coefficients. - * - */ - - -#include "modules/audio_coding/codecs/isac/main/source/arith_routines.h" - - - -static const int32_t kHistEdgesQ15[51] = { - -327680, -314573, -301466, -288359, -275252, -262144, -249037, -235930, -222823, -209716, - -196608, -183501, -170394, -157287, -144180, -131072, -117965, -104858, -91751, -78644, - -65536, -52429, -39322, -26215, -13108, 0, 13107, 26214, 39321, 52428, - 65536, 78643, 91750, 104857, 117964, 131072, 144179, 157286, 170393, 183500, - 196608, 209715, 222822, 235929, 249036, 262144, 275251, 288358, 301465, 314572, - 327680}; - - -static const int kCdfSlopeQ0[51] = { /* Q0 */ - 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, - 5, 5, 13, 23, 47, 87, 154, 315, 700, 1088, - 2471, 6064, 14221, 21463, 36634, 36924, 19750, 13270, 5806, 2312, - 1095, 660, 316, 145, 86, 41, 32, 5, 5, 5, - 5, 5, 5, 5, 5, 5, 5, 5, 5, 2, 0}; - - -static const int kCdfQ16[51] = { /* Q16 */ - 0, 2, 4, 6, 8, 10, 12, 14, 16, 18, - 20, 22, 24, 29, 38, 57, 92, 153, 279, 559, - 994, 1983, 4408, 10097, 18682, 33336, 48105, 56005, 61313, 63636, - 64560, 64998, 65262, 65389, 65447, 65481, 65497, 65510, 65512, 65514, - 65516, 65518, 65520, 65522, 65524, 65526, 65528, 65530, 65532, 65534, - 65535}; - - - -/* function to be converted to fixed point */ -static __inline uint32_t piecewise(int32_t xinQ15) { - - int32_t ind, qtmp1, qtmp2, qtmp3; - uint32_t tmpUW32; - - - qtmp2 = xinQ15; - - if (qtmp2 < kHistEdgesQ15[0]) { - qtmp2 = kHistEdgesQ15[0]; - } - if (qtmp2 > kHistEdgesQ15[50]) { - qtmp2 = kHistEdgesQ15[50]; - } - - qtmp1 = qtmp2 - kHistEdgesQ15[0]; /* Q15 - Q15 = Q15 */ - ind = (qtmp1 * 5) >> 16; /* 2^16 / 5 = 0.4 in Q15 */ - /* Q15 -> Q0 */ - qtmp1 = qtmp2 - kHistEdgesQ15[ind]; /* Q15 - Q15 = Q15 */ - qtmp2 = kCdfSlopeQ0[ind] * qtmp1; /* Q0 * Q15 = Q15 */ - qtmp3 = qtmp2>>15; /* Q15 -> Q0 */ - - tmpUW32 = kCdfQ16[ind] + qtmp3; /* Q0 + Q0 = Q0 */ - return tmpUW32; -} - - - -int WebRtcIsac_EncLogisticMulti2( - Bitstr *streamdata, /* in-/output struct containing bitstream */ - int16_t *dataQ7, /* input: data vector */ - const uint16_t *envQ8, /* input: side info vector defining the width of the pdf */ - const int N, /* input: data vector length / 2 */ - const int16_t isSWB12kHz) -{ - uint32_t W_lower, W_upper; - uint32_t W_upper_LSB, W_upper_MSB; - uint8_t *stream_ptr; - uint8_t *maxStreamPtr; - uint8_t *stream_ptr_carry; - uint32_t cdf_lo, cdf_hi; - int k; - - /* point to beginning of stream buffer */ - stream_ptr = streamdata->stream + streamdata->stream_index; - W_upper = streamdata->W_upper; - - maxStreamPtr = streamdata->stream + STREAM_SIZE_MAX_60 - 1; - for (k = 0; k < N; k++) - { - /* compute cdf_lower and cdf_upper by evaluating the piecewise linear cdf */ - cdf_lo = piecewise((*dataQ7 - 64) * *envQ8); - cdf_hi = piecewise((*dataQ7 + 64) * *envQ8); - - /* test and clip if probability gets too small */ - while (cdf_lo+1 >= cdf_hi) { - /* clip */ - if (*dataQ7 > 0) { - *dataQ7 -= 128; - cdf_hi = cdf_lo; - cdf_lo = piecewise((*dataQ7 - 64) * *envQ8); - } else { - *dataQ7 += 128; - cdf_lo = cdf_hi; - cdf_hi = piecewise((*dataQ7 + 64) * *envQ8); - } - } - - dataQ7++; - // increment only once per 4 iterations for SWB-16kHz or WB - // increment only once per 2 iterations for SWB-12kHz - envQ8 += (isSWB12kHz)? (k & 1):((k & 1) & (k >> 1)); - - - /* update interval */ - W_upper_LSB = W_upper & 0x0000FFFF; - W_upper_MSB = W_upper >> 16; - W_lower = W_upper_MSB * cdf_lo; - W_lower += (W_upper_LSB * cdf_lo) >> 16; - W_upper = W_upper_MSB * cdf_hi; - W_upper += (W_upper_LSB * cdf_hi) >> 16; - - /* shift interval such that it begins at zero */ - W_upper -= ++W_lower; - - /* add integer to bitstream */ - streamdata->streamval += W_lower; - - /* handle carry */ - if (streamdata->streamval < W_lower) - { - /* propagate carry */ - stream_ptr_carry = stream_ptr; - while (!(++(*--stream_ptr_carry))); - } - - /* renormalize interval, store most significant byte of streamval and update streamval */ - while ( !(W_upper & 0xFF000000) ) /* W_upper < 2^24 */ - { - W_upper <<= 8; - *stream_ptr++ = (uint8_t) (streamdata->streamval >> 24); - - if(stream_ptr > maxStreamPtr) - { - return -ISAC_DISALLOWED_BITSTREAM_LENGTH; - } - streamdata->streamval <<= 8; - } - } - - /* calculate new stream_index */ - streamdata->stream_index = (int)(stream_ptr - streamdata->stream); - streamdata->W_upper = W_upper; - - return 0; -} - - - -int WebRtcIsac_DecLogisticMulti2( - int16_t *dataQ7, /* output: data vector */ - Bitstr *streamdata, /* in-/output struct containing bitstream */ - const uint16_t *envQ8, /* input: side info vector defining the width of the pdf */ - const int16_t *ditherQ7,/* input: dither vector */ - const int N, /* input: data vector length */ - const int16_t isSWB12kHz) -{ - uint32_t W_lower, W_upper; - uint32_t W_tmp; - uint32_t W_upper_LSB, W_upper_MSB; - uint32_t streamval; - const uint8_t *stream_ptr; - uint32_t cdf_tmp; - int16_t candQ7; - int k; - - // Position just past the end of the stream. STREAM_SIZE_MAX_60 instead of - // STREAM_SIZE_MAX (which is the size of the allocated buffer) because that's - // the limit to how much data is filled in. - const uint8_t* const stream_end = streamdata->stream + STREAM_SIZE_MAX_60; - - stream_ptr = streamdata->stream + streamdata->stream_index; - W_upper = streamdata->W_upper; - if (streamdata->stream_index == 0) /* first time decoder is called for this stream */ - { - /* read first word from bytestream */ - if (stream_ptr + 3 >= stream_end) - return -1; // Would read out of bounds. Malformed input? - streamval = *stream_ptr << 24; - streamval |= *++stream_ptr << 16; - streamval |= *++stream_ptr << 8; - streamval |= *++stream_ptr; - } else { - streamval = streamdata->streamval; - } - - - for (k = 0; k < N; k++) - { - /* find the integer *data for which streamval lies in [W_lower+1, W_upper] */ - W_upper_LSB = W_upper & 0x0000FFFF; - W_upper_MSB = W_upper >> 16; - - /* find first candidate by inverting the logistic cdf */ - candQ7 = - *ditherQ7 + 64; - cdf_tmp = piecewise(candQ7 * *envQ8); - - W_tmp = W_upper_MSB * cdf_tmp; - W_tmp += (W_upper_LSB * cdf_tmp) >> 16; - if (streamval > W_tmp) - { - W_lower = W_tmp; - candQ7 += 128; - cdf_tmp = piecewise(candQ7 * *envQ8); - - W_tmp = W_upper_MSB * cdf_tmp; - W_tmp += (W_upper_LSB * cdf_tmp) >> 16; - while (streamval > W_tmp) - { - W_lower = W_tmp; - candQ7 += 128; - cdf_tmp = piecewise(candQ7 * *envQ8); - - W_tmp = W_upper_MSB * cdf_tmp; - W_tmp += (W_upper_LSB * cdf_tmp) >> 16; - - /* error check */ - if (W_lower == W_tmp) return -1; - } - W_upper = W_tmp; - - /* another sample decoded */ - *dataQ7 = candQ7 - 64; - } - else - { - W_upper = W_tmp; - candQ7 -= 128; - cdf_tmp = piecewise(candQ7 * *envQ8); - - W_tmp = W_upper_MSB * cdf_tmp; - W_tmp += (W_upper_LSB * cdf_tmp) >> 16; - while ( !(streamval > W_tmp) ) - { - W_upper = W_tmp; - candQ7 -= 128; - cdf_tmp = piecewise(candQ7 * *envQ8); - - W_tmp = W_upper_MSB * cdf_tmp; - W_tmp += (W_upper_LSB * cdf_tmp) >> 16; - - /* error check */ - if (W_upper == W_tmp) return -1; - } - W_lower = W_tmp; - - /* another sample decoded */ - *dataQ7 = candQ7 + 64; - } - ditherQ7++; - dataQ7++; - // increment only once per 4 iterations for SWB-16kHz or WB - // increment only once per 2 iterations for SWB-12kHz - envQ8 += (isSWB12kHz)? (k & 1):((k & 1) & (k >> 1)); - - /* shift interval to start at zero */ - W_upper -= ++W_lower; - - /* add integer to bitstream */ - streamval -= W_lower; - - /* renormalize interval and update streamval */ - while ( !(W_upper & 0xFF000000) ) /* W_upper < 2^24 */ - { - /* read next byte from stream */ - if (stream_ptr + 1 >= stream_end) - return -1; // Would read out of bounds. Malformed input? - streamval = (streamval << 8) | *++stream_ptr; - W_upper <<= 8; - } - } - - streamdata->stream_index = (int)(stream_ptr - streamdata->stream); - streamdata->W_upper = W_upper; - streamdata->streamval = streamval; - - /* find number of bytes in original stream (determined by current interval width) */ - if ( W_upper > 0x01FFFFFF ) - return streamdata->stream_index - 2; - else - return streamdata->stream_index - 1; -} diff --git a/modules/audio_coding/codecs/isac/main/source/audio_decoder_isac.cc b/modules/audio_coding/codecs/isac/main/source/audio_decoder_isac.cc deleted file mode 100644 index b671002e1e..0000000000 --- a/modules/audio_coding/codecs/isac/main/source/audio_decoder_isac.cc +++ /dev/null @@ -1,20 +0,0 @@ -/* - * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#include "modules/audio_coding/codecs/isac/main/include/audio_decoder_isac.h" - -#include "modules/audio_coding/codecs/isac/audio_decoder_isac_t_impl.h" - -namespace webrtc { - -// Explicit instantiation: -template class AudioDecoderIsacT; - -} // namespace webrtc diff --git a/modules/audio_coding/codecs/isac/main/source/audio_encoder_isac.cc b/modules/audio_coding/codecs/isac/main/source/audio_encoder_isac.cc deleted file mode 100644 index b7f2c0b1af..0000000000 --- a/modules/audio_coding/codecs/isac/main/source/audio_encoder_isac.cc +++ /dev/null @@ -1,20 +0,0 @@ -/* - * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#include "modules/audio_coding/codecs/isac/main/include/audio_encoder_isac.h" - -#include "modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h" - -namespace webrtc { - -// Explicit instantiation: -template class AudioEncoderIsacT; - -} // namespace webrtc diff --git a/modules/audio_coding/codecs/isac/main/source/audio_encoder_isac_unittest.cc b/modules/audio_coding/codecs/isac/main/source/audio_encoder_isac_unittest.cc deleted file mode 100644 index 07bab055e1..0000000000 --- a/modules/audio_coding/codecs/isac/main/source/audio_encoder_isac_unittest.cc +++ /dev/null @@ -1,61 +0,0 @@ -/* - * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#include "modules/audio_coding/codecs/isac/main/include/audio_encoder_isac.h" - -#include - -#include "test/gtest.h" - -namespace webrtc { - -namespace { - -void TestBadConfig(const AudioEncoderIsacFloatImpl::Config& config) { - EXPECT_FALSE(config.IsOk()); -} - -void TestGoodConfig(const AudioEncoderIsacFloatImpl::Config& config) { - EXPECT_TRUE(config.IsOk()); - AudioEncoderIsacFloatImpl aei(config); -} - -// Wrap subroutine calls that test things in this, so that the error messages -// will be accompanied by stack traces that make it possible to tell which -// subroutine invocation caused the failure. -#define S(x) \ - do { \ - SCOPED_TRACE(#x); \ - x; \ - } while (0) - -} // namespace - -TEST(AudioEncoderIsacTest, TestConfigBitrate) { - AudioEncoderIsacFloatImpl::Config config; - - // The default value is some real, positive value. - EXPECT_GT(config.bit_rate, 1); - S(TestGoodConfig(config)); - - // 0 is another way to ask for the default value. - config.bit_rate = 0; - S(TestGoodConfig(config)); - - // Try some unreasonable values and watch them fail. - config.bit_rate = -1; - S(TestBadConfig(config)); - config.bit_rate = 1; - S(TestBadConfig(config)); - config.bit_rate = std::numeric_limits::max(); - S(TestBadConfig(config)); -} - -} // namespace webrtc diff --git a/modules/audio_coding/codecs/isac/main/source/bandwidth_estimator.c b/modules/audio_coding/codecs/isac/main/source/bandwidth_estimator.c deleted file mode 100644 index 486cd95914..0000000000 --- a/modules/audio_coding/codecs/isac/main/source/bandwidth_estimator.c +++ /dev/null @@ -1,1013 +0,0 @@ -/* - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/* - * BwEstimator.c - * - * This file contains the code for the Bandwidth Estimator designed - * for iSAC. - * - */ - -#include -#include - -#include "modules/audio_coding/codecs/isac/main/source/bandwidth_estimator.h" -#include "modules/audio_coding/codecs/isac/main/source/settings.h" -#include "modules/audio_coding/codecs/isac/main/include/isac.h" -#include "rtc_base/checks.h" - -/* array of quantization levels for bottle neck info; Matlab code: */ -/* sprintf('%4.1ff, ', logspace(log10(5000), log10(40000), 12)) */ -static const float kQRateTableWb[12] = -{ - 10000.0f, 11115.3f, 12355.1f, 13733.1f, 15264.8f, 16967.3f, - 18859.8f, 20963.3f, 23301.4f, 25900.3f, 28789.0f, 32000.0f}; - - -static const float kQRateTableSwb[24] = -{ - 10000.0f, 11115.3f, 12355.1f, 13733.1f, 15264.8f, 16967.3f, - 18859.8f, 20963.3f, 23153.1f, 25342.9f, 27532.7f, 29722.5f, - 31912.3f, 34102.1f, 36291.9f, 38481.7f, 40671.4f, 42861.2f, - 45051.0f, 47240.8f, 49430.6f, 51620.4f, 53810.2f, 56000.0f, -}; - - - - -int32_t WebRtcIsac_InitBandwidthEstimator( - BwEstimatorstr* bwest_str, - enum IsacSamplingRate encoderSampRate, - enum IsacSamplingRate decoderSampRate) -{ - switch(encoderSampRate) - { - case kIsacWideband: - { - bwest_str->send_bw_avg = INIT_BN_EST_WB; - break; - } - case kIsacSuperWideband: - { - bwest_str->send_bw_avg = INIT_BN_EST_SWB; - break; - } - } - - switch(decoderSampRate) - { - case kIsacWideband: - { - bwest_str->prev_frame_length = INIT_FRAME_LEN_WB; - bwest_str->rec_bw_inv = 1.0f / - (INIT_BN_EST_WB + INIT_HDR_RATE_WB); - bwest_str->rec_bw = (int32_t)INIT_BN_EST_WB; - bwest_str->rec_bw_avg_Q = INIT_BN_EST_WB; - bwest_str->rec_bw_avg = INIT_BN_EST_WB + INIT_HDR_RATE_WB; - bwest_str->rec_header_rate = INIT_HDR_RATE_WB; - break; - } - case kIsacSuperWideband: - { - bwest_str->prev_frame_length = INIT_FRAME_LEN_SWB; - bwest_str->rec_bw_inv = 1.0f / - (INIT_BN_EST_SWB + INIT_HDR_RATE_SWB); - bwest_str->rec_bw = (int32_t)INIT_BN_EST_SWB; - bwest_str->rec_bw_avg_Q = INIT_BN_EST_SWB; - bwest_str->rec_bw_avg = INIT_BN_EST_SWB + INIT_HDR_RATE_SWB; - bwest_str->rec_header_rate = INIT_HDR_RATE_SWB; - break; - } - } - - bwest_str->prev_rec_rtp_number = 0; - bwest_str->prev_rec_arr_ts = 0; - bwest_str->prev_rec_send_ts = 0; - bwest_str->prev_rec_rtp_rate = 1.0f; - bwest_str->last_update_ts = 0; - bwest_str->last_reduction_ts = 0; - bwest_str->count_tot_updates_rec = -9; - bwest_str->rec_jitter = 10.0f; - bwest_str->rec_jitter_short_term = 0.0f; - bwest_str->rec_jitter_short_term_abs = 5.0f; - bwest_str->rec_max_delay = 10.0f; - bwest_str->rec_max_delay_avg_Q = 10.0f; - bwest_str->num_pkts_rec = 0; - - bwest_str->send_max_delay_avg = 10.0f; - - bwest_str->hsn_detect_rec = 0; - - bwest_str->num_consec_rec_pkts_over_30k = 0; - - bwest_str->hsn_detect_snd = 0; - - bwest_str->num_consec_snt_pkts_over_30k = 0; - - bwest_str->in_wait_period = 0; - - bwest_str->change_to_WB = 0; - - bwest_str->numConsecLatePkts = 0; - bwest_str->consecLatency = 0; - bwest_str->inWaitLatePkts = 0; - bwest_str->senderTimestamp = 0; - bwest_str->receiverTimestamp = 0; - - bwest_str->external_bw_info.in_use = 0; - - return 0; -} - -/* This function updates both bottle neck rates */ -/* Parameters: */ -/* rtp_number - value from RTP packet, from NetEq */ -/* frame length - length of signal frame in ms, from iSAC decoder */ -/* send_ts - value in RTP header giving send time in samples */ -/* arr_ts - value given by timeGetTime() time of arrival in samples of packet from NetEq */ -/* pksize - size of packet in bytes, from NetEq */ -/* Index - integer (range 0...23) indicating bottle neck & jitter as estimated by other side */ -/* returns 0 if everything went fine, -1 otherwise */ -int16_t WebRtcIsac_UpdateBandwidthEstimator( - BwEstimatorstr* bwest_str, - const uint16_t rtp_number, - const int32_t frame_length, - const uint32_t send_ts, - const uint32_t arr_ts, - const size_t pksize - /*, const uint16_t Index*/) -{ - float weight = 0.0f; - float curr_bw_inv = 0.0f; - float rec_rtp_rate; - float t_diff_proj; - float arr_ts_diff; - float send_ts_diff; - float arr_time_noise; - float arr_time_noise_abs; - - float delay_correction_factor = 1; - float late_diff = 0.0f; - int immediate_set = 0; - int num_pkts_expected; - - RTC_DCHECK(!bwest_str->external_bw_info.in_use); - - // We have to adjust the header-rate if the first packet has a - // frame-size different than the initialized value. - if ( frame_length != bwest_str->prev_frame_length ) - { - bwest_str->rec_header_rate = (float)HEADER_SIZE * 8.0f * - 1000.0f / (float)frame_length; /* bits/s */ - } - - /* UPDATE ESTIMATES ON THIS SIDE */ - /* compute far-side transmission rate */ - rec_rtp_rate = ((float)pksize * 8.0f * 1000.0f / (float)frame_length) + - bwest_str->rec_header_rate; - // rec_rtp_rate packet bits/s + header bits/s - - /* check for timer wrap-around */ - if (arr_ts < bwest_str->prev_rec_arr_ts) - { - bwest_str->prev_rec_arr_ts = arr_ts; - bwest_str->last_update_ts = arr_ts; - bwest_str->last_reduction_ts = arr_ts + 3*FS; - bwest_str->num_pkts_rec = 0; - - /* store frame length */ - bwest_str->prev_frame_length = frame_length; - - /* store far-side transmission rate */ - bwest_str->prev_rec_rtp_rate = rec_rtp_rate; - - /* store far-side RTP time stamp */ - bwest_str->prev_rec_rtp_number = rtp_number; - - return 0; - } - - bwest_str->num_pkts_rec++; - - /* check that it's not one of the first 9 packets */ - if ( bwest_str->count_tot_updates_rec > 0 ) - { - if(bwest_str->in_wait_period > 0 ) - { - bwest_str->in_wait_period--; - } - - bwest_str->inWaitLatePkts -= ((bwest_str->inWaitLatePkts > 0)? 1:0); - send_ts_diff = (float)(send_ts - bwest_str->prev_rec_send_ts); - - if (send_ts_diff <= (16 * frame_length)*2) - //doesn't allow for a dropped packet, not sure necessary to be - // that strict -DH - { - /* if not been updated for a long time, reduce the BN estimate */ - if((uint32_t)(arr_ts - bwest_str->last_update_ts) * - 1000.0f / FS > 3000) - { - //how many frames should have been received since the last - // update if too many have been dropped or there have been - // big delays won't allow this reduction may no longer need - // the send_ts_diff here - num_pkts_expected = (int)(((float)(arr_ts - - bwest_str->last_update_ts) * 1000.0f /(float) FS) / - (float)frame_length); - - if(((float)bwest_str->num_pkts_rec/(float)num_pkts_expected) > - 0.9) - { - float inv_bitrate = (float) pow( 0.99995, - (double)((uint32_t)(arr_ts - - bwest_str->last_reduction_ts)*1000.0f/FS) ); - - if ( inv_bitrate ) - { - bwest_str->rec_bw_inv /= inv_bitrate; - - //precautionary, likely never necessary - if (bwest_str->hsn_detect_snd && - bwest_str->hsn_detect_rec) - { - if (bwest_str->rec_bw_inv > 0.000066f) - { - bwest_str->rec_bw_inv = 0.000066f; - } - } - } - else - { - bwest_str->rec_bw_inv = 1.0f / - (INIT_BN_EST_WB + INIT_HDR_RATE_WB); - } - /* reset time-since-update counter */ - bwest_str->last_reduction_ts = arr_ts; - } - else - //reset here? - { - bwest_str->last_reduction_ts = arr_ts + 3*FS; - bwest_str->last_update_ts = arr_ts; - bwest_str->num_pkts_rec = 0; - } - } - } - else - { - bwest_str->last_reduction_ts = arr_ts + 3*FS; - bwest_str->last_update_ts = arr_ts; - bwest_str->num_pkts_rec = 0; - } - - - /* temporarily speed up adaptation if frame length has changed */ - if ( frame_length != bwest_str->prev_frame_length ) - { - bwest_str->count_tot_updates_rec = 10; - bwest_str->rec_header_rate = (float)HEADER_SIZE * 8.0f * - 1000.0f / (float)frame_length; /* bits/s */ - - bwest_str->rec_bw_inv = 1.0f /((float)bwest_str->rec_bw + - bwest_str->rec_header_rate); - } - - //////////////////////// - arr_ts_diff = (float)(arr_ts - bwest_str->prev_rec_arr_ts); - - if (send_ts_diff > 0 ) - { - late_diff = arr_ts_diff - send_ts_diff; - } - else - { - late_diff = arr_ts_diff - (float)(16 * frame_length); - } - - if((late_diff > 0) && !bwest_str->inWaitLatePkts) - { - bwest_str->numConsecLatePkts++; - bwest_str->consecLatency += late_diff; - } - else - { - bwest_str->numConsecLatePkts = 0; - bwest_str->consecLatency = 0; - } - if(bwest_str->numConsecLatePkts > 50) - { - float latencyMs = bwest_str->consecLatency/(FS/1000); - float averageLatencyMs = latencyMs / bwest_str->numConsecLatePkts; - delay_correction_factor = frame_length / (frame_length + averageLatencyMs); - immediate_set = 1; - bwest_str->inWaitLatePkts = (int16_t)((bwest_str->consecLatency/(FS/1000)) / 30);// + 150; - bwest_str->start_wait_period = arr_ts; - } - /////////////////////////////////////////////// - - - - /* update only if previous packet was not lost */ - if ( rtp_number == bwest_str->prev_rec_rtp_number + 1 ) - { - - - if (!(bwest_str->hsn_detect_snd && bwest_str->hsn_detect_rec)) - { - if ((arr_ts_diff > (float)(16 * frame_length))) - { - //1/2 second - if ((late_diff > 8000.0f) && !bwest_str->in_wait_period) - { - delay_correction_factor = 0.7f; - bwest_str->in_wait_period = 55; - bwest_str->start_wait_period = arr_ts; - immediate_set = 1; - } - //320 ms - else if (late_diff > 5120.0f && !bwest_str->in_wait_period) - { - delay_correction_factor = 0.8f; - immediate_set = 1; - bwest_str->in_wait_period = 44; - bwest_str->start_wait_period = arr_ts; - } - } - } - - - if ((bwest_str->prev_rec_rtp_rate > bwest_str->rec_bw_avg) && - (rec_rtp_rate > bwest_str->rec_bw_avg) && - !bwest_str->in_wait_period) - { - /* test if still in initiation period and increment counter */ - if (bwest_str->count_tot_updates_rec++ > 99) - { - /* constant weight after initiation part */ - weight = 0.01f; - } - else - { - /* weight decreases with number of updates */ - weight = 1.0f / (float) bwest_str->count_tot_updates_rec; - } - /* Bottle Neck Estimation */ - - /* limit outliers */ - /* if more than 25 ms too much */ - if (arr_ts_diff > frame_length * FS/1000 + 400.0f) - { - // in samples, why 25ms?? - arr_ts_diff = frame_length * FS/1000 + 400.0f; - } - if(arr_ts_diff < (frame_length * FS/1000) - 160.0f) - { - /* don't allow it to be less than frame rate - 10 ms */ - arr_ts_diff = (float)frame_length * FS/1000 - 160.0f; - } - - /* compute inverse receiving rate for last packet */ - curr_bw_inv = arr_ts_diff / ((float)(pksize + HEADER_SIZE) * - 8.0f * FS); // (180+35)*8*16000 = 27.5 Mbit.... - - - if(curr_bw_inv < - (1.0f / (MAX_ISAC_BW + bwest_str->rec_header_rate))) - { - // don't allow inv rate to be larger than MAX - curr_bw_inv = (1.0f / - (MAX_ISAC_BW + bwest_str->rec_header_rate)); - } - - /* update bottle neck rate estimate */ - bwest_str->rec_bw_inv = weight * curr_bw_inv + - (1.0f - weight) * bwest_str->rec_bw_inv; - - /* reset time-since-update counter */ - bwest_str->last_update_ts = arr_ts; - bwest_str->last_reduction_ts = arr_ts + 3 * FS; - bwest_str->num_pkts_rec = 0; - - /* Jitter Estimation */ - /* projected difference between arrival times */ - t_diff_proj = ((float)(pksize + HEADER_SIZE) * 8.0f * - 1000.0f) / bwest_str->rec_bw_avg; - - - // difference between projected and actual - // arrival time differences - arr_time_noise = (float)(arr_ts_diff*1000.0f/FS) - - t_diff_proj; - arr_time_noise_abs = (float) fabs( arr_time_noise ); - - /* long term averaged absolute jitter */ - bwest_str->rec_jitter = weight * arr_time_noise_abs + - (1.0f - weight) * bwest_str->rec_jitter; - if (bwest_str->rec_jitter > 10.0f) - { - bwest_str->rec_jitter = 10.0f; - } - /* short term averaged absolute jitter */ - bwest_str->rec_jitter_short_term_abs = 0.05f * - arr_time_noise_abs + 0.95f * - bwest_str->rec_jitter_short_term_abs; - - /* short term averaged jitter */ - bwest_str->rec_jitter_short_term = 0.05f * arr_time_noise + - 0.95f * bwest_str->rec_jitter_short_term; - } - } - } - else - { - // reset time-since-update counter when - // receiving the first 9 packets - bwest_str->last_update_ts = arr_ts; - bwest_str->last_reduction_ts = arr_ts + 3*FS; - bwest_str->num_pkts_rec = 0; - - bwest_str->count_tot_updates_rec++; - } - - /* limit minimum bottle neck rate */ - if (bwest_str->rec_bw_inv > 1.0f / ((float)MIN_ISAC_BW + - bwest_str->rec_header_rate)) - { - bwest_str->rec_bw_inv = 1.0f / ((float)MIN_ISAC_BW + - bwest_str->rec_header_rate); - } - - // limit maximum bitrate - if (bwest_str->rec_bw_inv < 1.0f / ((float)MAX_ISAC_BW + - bwest_str->rec_header_rate)) - { - bwest_str->rec_bw_inv = 1.0f / ((float)MAX_ISAC_BW + - bwest_str->rec_header_rate); - } - - /* store frame length */ - bwest_str->prev_frame_length = frame_length; - - /* store far-side transmission rate */ - bwest_str->prev_rec_rtp_rate = rec_rtp_rate; - - /* store far-side RTP time stamp */ - bwest_str->prev_rec_rtp_number = rtp_number; - - // Replace bwest_str->rec_max_delay by the new - // value (atomic operation) - bwest_str->rec_max_delay = 3.0f * bwest_str->rec_jitter; - - /* store send and arrival time stamp */ - bwest_str->prev_rec_arr_ts = arr_ts ; - bwest_str->prev_rec_send_ts = send_ts; - - /* Replace bwest_str->rec_bw by the new value (atomic operation) */ - bwest_str->rec_bw = (int32_t)(1.0f / bwest_str->rec_bw_inv - - bwest_str->rec_header_rate); - - if (immediate_set) - { - bwest_str->rec_bw = (int32_t) (delay_correction_factor * - (float) bwest_str->rec_bw); - - if (bwest_str->rec_bw < (int32_t) MIN_ISAC_BW) - { - bwest_str->rec_bw = (int32_t) MIN_ISAC_BW; - } - - bwest_str->rec_bw_avg = bwest_str->rec_bw + - bwest_str->rec_header_rate; - - bwest_str->rec_bw_avg_Q = (float) bwest_str->rec_bw; - - bwest_str->rec_jitter_short_term = 0.0f; - - bwest_str->rec_bw_inv = 1.0f / (bwest_str->rec_bw + - bwest_str->rec_header_rate); - - bwest_str->count_tot_updates_rec = 1; - - immediate_set = 0; - bwest_str->consecLatency = 0; - bwest_str->numConsecLatePkts = 0; - } - - return 0; -} - - -/* This function updates the send bottle neck rate */ -/* Index - integer (range 0...23) indicating bottle neck & jitter as estimated by other side */ -/* returns 0 if everything went fine, -1 otherwise */ -int16_t WebRtcIsac_UpdateUplinkBwImpl( - BwEstimatorstr* bwest_str, - int16_t index, - enum IsacSamplingRate encoderSamplingFreq) -{ - RTC_DCHECK(!bwest_str->external_bw_info.in_use); - - if((index < 0) || (index > 23)) - { - return -ISAC_RANGE_ERROR_BW_ESTIMATOR; - } - - /* UPDATE ESTIMATES FROM OTHER SIDE */ - if(encoderSamplingFreq == kIsacWideband) - { - if(index > 11) - { - index -= 12; - /* compute the jitter estimate as decoded on the other side */ - bwest_str->send_max_delay_avg = 0.9f * bwest_str->send_max_delay_avg + - 0.1f * (float)MAX_ISAC_MD; - } - else - { - /* compute the jitter estimate as decoded on the other side */ - bwest_str->send_max_delay_avg = 0.9f * bwest_str->send_max_delay_avg + - 0.1f * (float)MIN_ISAC_MD; - } - - /* compute the BN estimate as decoded on the other side */ - bwest_str->send_bw_avg = 0.9f * bwest_str->send_bw_avg + - 0.1f * kQRateTableWb[index]; - } - else - { - /* compute the BN estimate as decoded on the other side */ - bwest_str->send_bw_avg = 0.9f * bwest_str->send_bw_avg + - 0.1f * kQRateTableSwb[index]; - } - - if (bwest_str->send_bw_avg > (float) 28000 && !bwest_str->hsn_detect_snd) - { - bwest_str->num_consec_snt_pkts_over_30k++; - - if (bwest_str->num_consec_snt_pkts_over_30k >= 66) - { - //approx 2 seconds with 30ms frames - bwest_str->hsn_detect_snd = 1; - } - } - else if (!bwest_str->hsn_detect_snd) - { - bwest_str->num_consec_snt_pkts_over_30k = 0; - } - return 0; -} - -// called when there is upper-band bit-stream to update jitter -// statistics. -int16_t WebRtcIsac_UpdateUplinkJitter( - BwEstimatorstr* bwest_str, - int32_t index) -{ - RTC_DCHECK(!bwest_str->external_bw_info.in_use); - - if((index < 0) || (index > 23)) - { - return -ISAC_RANGE_ERROR_BW_ESTIMATOR; - } - - if(index > 0) - { - /* compute the jitter estimate as decoded on the other side */ - bwest_str->send_max_delay_avg = 0.9f * bwest_str->send_max_delay_avg + - 0.1f * (float)MAX_ISAC_MD; - } - else - { - /* compute the jitter estimate as decoded on the other side */ - bwest_str->send_max_delay_avg = 0.9f * bwest_str->send_max_delay_avg + - 0.1f * (float)MIN_ISAC_MD; - } - - return 0; -} - - - -// Returns the bandwidth/jitter estimation code (integer 0...23) -// to put in the sending iSAC payload -void -WebRtcIsac_GetDownlinkBwJitIndexImpl( - BwEstimatorstr* bwest_str, - int16_t* bottleneckIndex, - int16_t* jitterInfo, - enum IsacSamplingRate decoderSamplingFreq) -{ - float MaxDelay; - //uint16_t MaxDelayBit; - - float rate; - float r; - float e1, e2; - const float weight = 0.1f; - const float* ptrQuantizationTable; - int16_t addJitterInfo; - int16_t minInd; - int16_t maxInd; - int16_t midInd; - - if (bwest_str->external_bw_info.in_use) { - *bottleneckIndex = bwest_str->external_bw_info.bottleneck_idx; - *jitterInfo = bwest_str->external_bw_info.jitter_info; - return; - } - - /* Get Max Delay Bit */ - /* get unquantized max delay */ - MaxDelay = (float)WebRtcIsac_GetDownlinkMaxDelay(bwest_str); - - if ( ((1.f - weight) * bwest_str->rec_max_delay_avg_Q + weight * - MAX_ISAC_MD - MaxDelay) > (MaxDelay - (1.f-weight) * - bwest_str->rec_max_delay_avg_Q - weight * MIN_ISAC_MD) ) - { - jitterInfo[0] = 0; - /* update quantized average */ - bwest_str->rec_max_delay_avg_Q = - (1.f - weight) * bwest_str->rec_max_delay_avg_Q + weight * - (float)MIN_ISAC_MD; - } - else - { - jitterInfo[0] = 1; - /* update quantized average */ - bwest_str->rec_max_delay_avg_Q = - (1.f-weight) * bwest_str->rec_max_delay_avg_Q + weight * - (float)MAX_ISAC_MD; - } - - // Get unquantized rate. - rate = (float)WebRtcIsac_GetDownlinkBandwidth(bwest_str); - - /* Get Rate Index */ - if(decoderSamplingFreq == kIsacWideband) - { - ptrQuantizationTable = kQRateTableWb; - addJitterInfo = 1; - maxInd = 11; - } - else - { - ptrQuantizationTable = kQRateTableSwb; - addJitterInfo = 0; - maxInd = 23; - } - - minInd = 0; - while(maxInd > minInd + 1) - { - midInd = (maxInd + minInd) >> 1; - if(rate > ptrQuantizationTable[midInd]) - { - minInd = midInd; - } - else - { - maxInd = midInd; - } - } - // Chose the index which gives results an average which is closest - // to rate - r = (1 - weight) * bwest_str->rec_bw_avg_Q - rate; - e1 = weight * ptrQuantizationTable[minInd] + r; - e2 = weight * ptrQuantizationTable[maxInd] + r; - e1 = (e1 > 0)? e1:-e1; - e2 = (e2 > 0)? e2:-e2; - if(e1 < e2) - { - bottleneckIndex[0] = minInd; - } - else - { - bottleneckIndex[0] = maxInd; - } - - bwest_str->rec_bw_avg_Q = (1 - weight) * bwest_str->rec_bw_avg_Q + - weight * ptrQuantizationTable[bottleneckIndex[0]]; - bottleneckIndex[0] += jitterInfo[0] * 12 * addJitterInfo; - - bwest_str->rec_bw_avg = (1 - weight) * bwest_str->rec_bw_avg + weight * - (rate + bwest_str->rec_header_rate); -} - - - -/* get the bottle neck rate from far side to here, as estimated on this side */ -int32_t WebRtcIsac_GetDownlinkBandwidth( const BwEstimatorstr *bwest_str) -{ - int32_t rec_bw; - float jitter_sign; - float bw_adjust; - - RTC_DCHECK(!bwest_str->external_bw_info.in_use); - - /* create a value between -1.0 and 1.0 indicating "average sign" of jitter */ - jitter_sign = bwest_str->rec_jitter_short_term / - bwest_str->rec_jitter_short_term_abs; - - /* adjust bw proportionally to negative average jitter sign */ - bw_adjust = 1.0f - jitter_sign * (0.15f + 0.15f * jitter_sign * jitter_sign); - - /* adjust Rate if jitter sign is mostly constant */ - rec_bw = (int32_t)(bwest_str->rec_bw * bw_adjust); - - /* limit range of bottle neck rate */ - if (rec_bw < MIN_ISAC_BW) - { - rec_bw = MIN_ISAC_BW; - } - else if (rec_bw > MAX_ISAC_BW) - { - rec_bw = MAX_ISAC_BW; - } - return rec_bw; -} - -/* Returns the max delay (in ms) */ -int32_t -WebRtcIsac_GetDownlinkMaxDelay(const BwEstimatorstr *bwest_str) -{ - int32_t rec_max_delay; - - RTC_DCHECK(!bwest_str->external_bw_info.in_use); - - rec_max_delay = (int32_t)(bwest_str->rec_max_delay); - - /* limit range of jitter estimate */ - if (rec_max_delay < MIN_ISAC_MD) - { - rec_max_delay = MIN_ISAC_MD; - } - else if (rec_max_delay > MAX_ISAC_MD) - { - rec_max_delay = MAX_ISAC_MD; - } - return rec_max_delay; -} - -/* Clamp val to the closed interval [min,max]. */ -static int32_t clamp(int32_t val, int32_t min, int32_t max) { - RTC_DCHECK_LE(min, max); - return val < min ? min : (val > max ? max : val); -} - -int32_t WebRtcIsac_GetUplinkBandwidth(const BwEstimatorstr* bwest_str) { - return bwest_str->external_bw_info.in_use - ? bwest_str->external_bw_info.send_bw_avg - : clamp(bwest_str->send_bw_avg, MIN_ISAC_BW, MAX_ISAC_BW); -} - -int32_t WebRtcIsac_GetUplinkMaxDelay(const BwEstimatorstr* bwest_str) { - return bwest_str->external_bw_info.in_use - ? bwest_str->external_bw_info.send_max_delay_avg - : clamp(bwest_str->send_max_delay_avg, MIN_ISAC_MD, MAX_ISAC_MD); -} - -/* - * update long-term average bitrate and amount of data in buffer - * returns minimum payload size (bytes) - */ -int WebRtcIsac_GetMinBytes( - RateModel* State, - int StreamSize, /* bytes in bitstream */ - const int FrameSamples, /* samples per frame */ - const double BottleNeck, /* bottle neck rate; excl headers (bps) */ - const double DelayBuildUp, /* max delay from bottleneck buffering (ms) */ - enum ISACBandwidth bandwidth - /*,int16_t frequentLargePackets*/) -{ - double MinRate = 0.0; - int MinBytes; - double TransmissionTime; - int burstInterval = BURST_INTERVAL; - - // first 10 packets @ low rate, then INIT_BURST_LEN packets @ - // fixed rate of INIT_RATE bps - if (State->InitCounter > 0) - { - if (State->InitCounter-- <= INIT_BURST_LEN) - { - if(bandwidth == isac8kHz) - { - MinRate = INIT_RATE_WB; - } - else - { - MinRate = INIT_RATE_SWB; - } - } - else - { - MinRate = 0; - } - } - else - { - /* handle burst */ - if (State->BurstCounter) - { - if (State->StillBuffered < (1.0 - 1.0/BURST_LEN) * DelayBuildUp) - { - /* max bps derived from BottleNeck and DelayBuildUp values */ - MinRate = (1.0 + (FS/1000) * DelayBuildUp / - (double)(BURST_LEN * FrameSamples)) * BottleNeck; - } - else - { - // max bps derived from StillBuffered and DelayBuildUp - // values - MinRate = (1.0 + (FS/1000) * (DelayBuildUp - - State->StillBuffered) / (double)FrameSamples) * BottleNeck; - if (MinRate < 1.04 * BottleNeck) - { - MinRate = 1.04 * BottleNeck; - } - } - State->BurstCounter--; - } - } - - - /* convert rate from bits/second to bytes/packet */ - MinBytes = (int) (MinRate * FrameSamples / (8.0 * FS)); - - /* StreamSize will be adjusted if less than MinBytes */ - if (StreamSize < MinBytes) - { - StreamSize = MinBytes; - } - - /* keep track of when bottle neck was last exceeded by at least 1% */ - if (StreamSize * 8.0 * FS / FrameSamples > 1.01 * BottleNeck) { - if (State->PrevExceed) { - /* bottle_neck exceded twice in a row, decrease ExceedAgo */ - State->ExceedAgo -= /*BURST_INTERVAL*/ burstInterval / (BURST_LEN - 1); - if (State->ExceedAgo < 0) - State->ExceedAgo = 0; - } - else - { - State->ExceedAgo += (FrameSamples * 1000) / FS; /* ms */ - State->PrevExceed = 1; - } - } - else - { - State->PrevExceed = 0; - State->ExceedAgo += (FrameSamples * 1000) / FS; /* ms */ - } - - /* set burst flag if bottle neck not exceeded for long time */ - if ((State->ExceedAgo > burstInterval) && - (State->BurstCounter == 0)) - { - if (State->PrevExceed) - { - State->BurstCounter = BURST_LEN - 1; - } - else - { - State->BurstCounter = BURST_LEN; - } - } - - - /* Update buffer delay */ - TransmissionTime = StreamSize * 8.0 * 1000.0 / BottleNeck; /* ms */ - State->StillBuffered += TransmissionTime; - State->StillBuffered -= (FrameSamples * 1000) / FS; /* ms */ - if (State->StillBuffered < 0.0) - { - State->StillBuffered = 0.0; - } - - return MinBytes; -} - - -/* - * update long-term average bitrate and amount of data in buffer - */ -void WebRtcIsac_UpdateRateModel( - RateModel *State, - int StreamSize, /* bytes in bitstream */ - const int FrameSamples, /* samples per frame */ - const double BottleNeck) /* bottle neck rate; excl headers (bps) */ -{ - double TransmissionTime; - - /* avoid the initial "high-rate" burst */ - State->InitCounter = 0; - - /* Update buffer delay */ - TransmissionTime = StreamSize * 8.0 * 1000.0 / BottleNeck; /* ms */ - State->StillBuffered += TransmissionTime; - State->StillBuffered -= (FrameSamples * 1000) / FS; /* ms */ - if (State->StillBuffered < 0.0) - State->StillBuffered = 0.0; - -} - - -void WebRtcIsac_InitRateModel( - RateModel *State) -{ - State->PrevExceed = 0; /* boolean */ - State->ExceedAgo = 0; /* ms */ - State->BurstCounter = 0; /* packets */ - State->InitCounter = INIT_BURST_LEN + 10; /* packets */ - State->StillBuffered = 1.0; /* ms */ -} - -int WebRtcIsac_GetNewFrameLength( - double bottle_neck, - int current_framesamples) -{ - int new_framesamples; - - const int Thld_20_30 = 20000; - - //const int Thld_30_20 = 30000; - const int Thld_30_20 = 1000000; // disable 20 ms frames - - const int Thld_30_60 = 18000; - //const int Thld_30_60 = 0; // disable 60 ms frames - - const int Thld_60_30 = 27000; - - - new_framesamples = current_framesamples; - - /* find new framelength */ - switch(current_framesamples) { - case 320: - if (bottle_neck < Thld_20_30) - new_framesamples = 480; - break; - case 480: - if (bottle_neck < Thld_30_60) - new_framesamples = 960; - else if (bottle_neck > Thld_30_20) - new_framesamples = 320; - break; - case 960: - if (bottle_neck >= Thld_60_30) - new_framesamples = 480; - break; - } - - return new_framesamples; -} - -double WebRtcIsac_GetSnr( - double bottle_neck, - int framesamples) -{ - double s2nr; - - const double a_20 = -30.0; - const double b_20 = 0.8; - const double c_20 = 0.0; - - const double a_30 = -23.0; - const double b_30 = 0.48; - const double c_30 = 0.0; - - const double a_60 = -23.0; - const double b_60 = 0.53; - const double c_60 = 0.0; - - - /* find new SNR value */ - switch(framesamples) { - case 320: - s2nr = a_20 + b_20 * bottle_neck * 0.001 + c_20 * bottle_neck * - bottle_neck * 0.000001; - break; - case 480: - s2nr = a_30 + b_30 * bottle_neck * 0.001 + c_30 * bottle_neck * - bottle_neck * 0.000001; - break; - case 960: - s2nr = a_60 + b_60 * bottle_neck * 0.001 + c_60 * bottle_neck * - bottle_neck * 0.000001; - break; - default: - s2nr = 0; - } - - return s2nr; - -} diff --git a/modules/audio_coding/codecs/isac/main/source/bandwidth_estimator.h b/modules/audio_coding/codecs/isac/main/source/bandwidth_estimator.h deleted file mode 100644 index 5f4550a3a5..0000000000 --- a/modules/audio_coding/codecs/isac/main/source/bandwidth_estimator.h +++ /dev/null @@ -1,165 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/* - * bandwidth_estimator.h - * - * This header file contains the API for the Bandwidth Estimator - * designed for iSAC. - * - */ - -#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_BANDWIDTH_ESTIMATOR_H_ -#define MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_BANDWIDTH_ESTIMATOR_H_ - -#include - -#include "modules/audio_coding/codecs/isac/main/source/settings.h" -#include "modules/audio_coding/codecs/isac/main/source/structs.h" - -#define MIN_ISAC_BW 10000 -#define MIN_ISAC_BW_LB 10000 -#define MIN_ISAC_BW_UB 25000 - -#define MAX_ISAC_BW 56000 -#define MAX_ISAC_BW_UB 32000 -#define MAX_ISAC_BW_LB 32000 - -#define MIN_ISAC_MD 5 -#define MAX_ISAC_MD 25 - -// assumed header size, in bytes; we don't know the exact number -// (header compression may be used) -#define HEADER_SIZE 35 - -// Initial Frame-Size, in ms, for Wideband & Super-Wideband Mode -#define INIT_FRAME_LEN_WB 60 -#define INIT_FRAME_LEN_SWB 30 - -// Initial Bottleneck Estimate, in bits/sec, for -// Wideband & Super-wideband mode -#define INIT_BN_EST_WB 20e3f -#define INIT_BN_EST_SWB 56e3f - -// Initial Header rate (header rate depends on frame-size), -// in bits/sec, for Wideband & Super-Wideband mode. -#define INIT_HDR_RATE_WB \ - ((float)HEADER_SIZE * 8.0f * 1000.0f / (float)INIT_FRAME_LEN_WB) -#define INIT_HDR_RATE_SWB \ - ((float)HEADER_SIZE * 8.0f * 1000.0f / (float)INIT_FRAME_LEN_SWB) - -// number of packets in a row for a high rate burst -#define BURST_LEN 3 - -// ms, max time between two full bursts -#define BURST_INTERVAL 500 - -// number of packets in a row for initial high rate burst -#define INIT_BURST_LEN 5 - -// bits/s, rate for the first BURST_LEN packets -#define INIT_RATE_WB INIT_BN_EST_WB -#define INIT_RATE_SWB INIT_BN_EST_SWB - -#if defined(__cplusplus) -extern "C" { -#endif - -/* This function initializes the struct */ -/* to be called before using the struct for anything else */ -/* returns 0 if everything went fine, -1 otherwise */ -int32_t WebRtcIsac_InitBandwidthEstimator( - BwEstimatorstr* bwest_str, - enum IsacSamplingRate encoderSampRate, - enum IsacSamplingRate decoderSampRate); - -/* This function updates the receiving estimate */ -/* Parameters: */ -/* rtp_number - value from RTP packet, from NetEq */ -/* frame length - length of signal frame in ms, from iSAC decoder */ -/* send_ts - value in RTP header giving send time in samples */ -/* arr_ts - value given by timeGetTime() time of arrival in samples of - * packet from NetEq */ -/* pksize - size of packet in bytes, from NetEq */ -/* Index - integer (range 0...23) indicating bottle neck & jitter as - * estimated by other side */ -/* returns 0 if everything went fine, -1 otherwise */ -int16_t WebRtcIsac_UpdateBandwidthEstimator(BwEstimatorstr* bwest_str, - uint16_t rtp_number, - int32_t frame_length, - uint32_t send_ts, - uint32_t arr_ts, - size_t pksize); - -/* Update receiving estimates. Used when we only receive BWE index, no iSAC data - * packet. */ -int16_t WebRtcIsac_UpdateUplinkBwImpl( - BwEstimatorstr* bwest_str, - int16_t Index, - enum IsacSamplingRate encoderSamplingFreq); - -/* Returns the bandwidth/jitter estimation code (integer 0...23) to put in the - * sending iSAC payload */ -void WebRtcIsac_GetDownlinkBwJitIndexImpl( - BwEstimatorstr* bwest_str, - int16_t* bottleneckIndex, - int16_t* jitterInfo, - enum IsacSamplingRate decoderSamplingFreq); - -/* Returns the bandwidth estimation (in bps) */ -int32_t WebRtcIsac_GetDownlinkBandwidth(const BwEstimatorstr* bwest_str); - -/* Returns the max delay (in ms) */ -int32_t WebRtcIsac_GetDownlinkMaxDelay(const BwEstimatorstr* bwest_str); - -/* Returns the bandwidth that iSAC should send with in bps */ -int32_t WebRtcIsac_GetUplinkBandwidth(const BwEstimatorstr* bwest_str); - -/* Returns the max delay value from the other side in ms */ -int32_t WebRtcIsac_GetUplinkMaxDelay(const BwEstimatorstr* bwest_str); - -/* - * update amount of data in bottle neck buffer and burst handling - * returns minimum payload size (bytes) - */ -int WebRtcIsac_GetMinBytes( - RateModel* State, - int StreamSize, /* bytes in bitstream */ - int FrameLen, /* ms per frame */ - double BottleNeck, /* bottle neck rate; excl headers (bps) */ - double DelayBuildUp, /* max delay from bottleneck buffering (ms) */ - enum ISACBandwidth bandwidth - /*,int16_t frequentLargePackets*/); - -/* - * update long-term average bitrate and amount of data in buffer - */ -void WebRtcIsac_UpdateRateModel( - RateModel* State, - int StreamSize, /* bytes in bitstream */ - int FrameSamples, /* samples per frame */ - double BottleNeck); /* bottle neck rate; excl headers (bps) */ - -void WebRtcIsac_InitRateModel(RateModel* State); - -/* Returns the new framelength value (input argument: bottle_neck) */ -int WebRtcIsac_GetNewFrameLength(double bottle_neck, int current_framelength); - -/* Returns the new SNR value (input argument: bottle_neck) */ -double WebRtcIsac_GetSnr(double bottle_neck, int new_framelength); - -int16_t WebRtcIsac_UpdateUplinkJitter(BwEstimatorstr* bwest_str, int32_t index); - -#if defined(__cplusplus) -} -#endif - -#endif /* MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_BANDWIDTH_ESTIMATOR_H_ \ - */ diff --git a/modules/audio_coding/codecs/isac/main/source/codec.h b/modules/audio_coding/codecs/isac/main/source/codec.h deleted file mode 100644 index a7c7ddc14a..0000000000 --- a/modules/audio_coding/codecs/isac/main/source/codec.h +++ /dev/null @@ -1,223 +0,0 @@ -/* - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/* - * codec.h - * - * This header file contains the calls to the internal encoder - * and decoder functions. - * - */ - -#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_CODEC_H_ -#define MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_CODEC_H_ - -#include - -#include "modules/audio_coding/codecs/isac/main/source/structs.h" -#include "modules/third_party/fft/fft.h" - -void WebRtcIsac_ResetBitstream(Bitstr* bit_stream); - -int WebRtcIsac_EstimateBandwidth(BwEstimatorstr* bwest_str, - Bitstr* streamdata, - size_t packet_size, - uint16_t rtp_seq_number, - uint32_t send_ts, - uint32_t arr_ts, - enum IsacSamplingRate encoderSampRate, - enum IsacSamplingRate decoderSampRate); - -int WebRtcIsac_DecodeLb(const TransformTables* transform_tables, - float* signal_out, - ISACLBDecStruct* ISACdec_obj, - int16_t* current_framesamples, - int16_t isRCUPayload); - -int WebRtcIsac_DecodeRcuLb(float* signal_out, - ISACLBDecStruct* ISACdec_obj, - int16_t* current_framesamples); - -int WebRtcIsac_EncodeLb(const TransformTables* transform_tables, - float* in, - ISACLBEncStruct* ISACencLB_obj, - int16_t codingMode, - int16_t bottleneckIndex); - -int WebRtcIsac_EncodeStoredDataLb(const IsacSaveEncoderData* ISACSavedEnc_obj, - Bitstr* ISACBitStr_obj, - int BWnumber, - float scale); - -int WebRtcIsac_EncodeStoredDataUb( - const ISACUBSaveEncDataStruct* ISACSavedEnc_obj, - Bitstr* bitStream, - int32_t jitterInfo, - float scale, - enum ISACBandwidth bandwidth); - -int16_t WebRtcIsac_GetRedPayloadUb( - const ISACUBSaveEncDataStruct* ISACSavedEncObj, - Bitstr* bitStreamObj, - enum ISACBandwidth bandwidth); - -/****************************************************************************** - * WebRtcIsac_RateAllocation() - * Internal function to perform a rate-allocation for upper and lower-band, - * given a total rate. - * - * Input: - * - inRateBitPerSec : a total bit-rate in bits/sec. - * - * Output: - * - rateLBBitPerSec : a bit-rate allocated to the lower-band - * in bits/sec. - * - rateUBBitPerSec : a bit-rate allocated to the upper-band - * in bits/sec. - * - * Return value : 0 if rate allocation has been successful. - * -1 if failed to allocate rates. - */ - -int16_t WebRtcIsac_RateAllocation(int32_t inRateBitPerSec, - double* rateLBBitPerSec, - double* rateUBBitPerSec, - enum ISACBandwidth* bandwidthKHz); - -/****************************************************************************** - * WebRtcIsac_DecodeUb16() - * - * Decode the upper-band if the codec is in 0-16 kHz mode. - * - * Input/Output: - * -ISACdec_obj : pointer to the upper-band decoder object. The - * bit-stream is stored inside the decoder object. - * - * Output: - * -signal_out : decoded audio, 480 samples 30 ms. - * - * Return value : >0 number of decoded bytes. - * <0 if an error occurred. - */ -int WebRtcIsac_DecodeUb16(const TransformTables* transform_tables, - float* signal_out, - ISACUBDecStruct* ISACdec_obj, - int16_t isRCUPayload); - -/****************************************************************************** - * WebRtcIsac_DecodeUb12() - * - * Decode the upper-band if the codec is in 0-12 kHz mode. - * - * Input/Output: - * -ISACdec_obj : pointer to the upper-band decoder object. The - * bit-stream is stored inside the decoder object. - * - * Output: - * -signal_out : decoded audio, 480 samples 30 ms. - * - * Return value : >0 number of decoded bytes. - * <0 if an error occurred. - */ -int WebRtcIsac_DecodeUb12(const TransformTables* transform_tables, - float* signal_out, - ISACUBDecStruct* ISACdec_obj, - int16_t isRCUPayload); - -/****************************************************************************** - * WebRtcIsac_EncodeUb16() - * - * Encode the upper-band if the codec is in 0-16 kHz mode. - * - * Input: - * -in : upper-band audio, 160 samples (10 ms). - * - * Input/Output: - * -ISACdec_obj : pointer to the upper-band encoder object. The - * bit-stream is stored inside the encoder object. - * - * Return value : >0 number of encoded bytes. - * <0 if an error occurred. - */ -int WebRtcIsac_EncodeUb16(const TransformTables* transform_tables, - float* in, - ISACUBEncStruct* ISACenc_obj, - int32_t jitterInfo); - -/****************************************************************************** - * WebRtcIsac_EncodeUb12() - * - * Encode the upper-band if the codec is in 0-12 kHz mode. - * - * Input: - * -in : upper-band audio, 160 samples (10 ms). - * - * Input/Output: - * -ISACdec_obj : pointer to the upper-band encoder object. The - * bit-stream is stored inside the encoder object. - * - * Return value : >0 number of encoded bytes. - * <0 if an error occurred. - */ -int WebRtcIsac_EncodeUb12(const TransformTables* transform_tables, - float* in, - ISACUBEncStruct* ISACenc_obj, - int32_t jitterInfo); - -/************************** initialization functions *************************/ - -void WebRtcIsac_InitMasking(MaskFiltstr* maskdata); - -void WebRtcIsac_InitPostFilterbank(PostFiltBankstr* postfiltdata); - -/**************************** transform functions ****************************/ - -void WebRtcIsac_InitTransform(TransformTables* tables); - -void WebRtcIsac_Time2Spec(const TransformTables* tables, - double* inre1, - double* inre2, - int16_t* outre, - int16_t* outim, - FFTstr* fftstr_obj); - -void WebRtcIsac_Spec2time(const TransformTables* tables, - double* inre, - double* inim, - double* outre1, - double* outre2, - FFTstr* fftstr_obj); - -/***************************** filterbank functions **************************/ - -void WebRtcIsac_FilterAndCombineFloat(float* InLP, - float* InHP, - float* Out, - PostFiltBankstr* postfiltdata); - -/************************* normalized lattice filters ************************/ - -void WebRtcIsac_NormLatticeFilterMa(int orderCoef, - float* stateF, - float* stateG, - float* lat_in, - double* filtcoeflo, - double* lat_out); - -void WebRtcIsac_NormLatticeFilterAr(int orderCoef, - float* stateF, - float* stateG, - double* lat_in, - double* lo_filt_coef, - float* lat_out); - -void WebRtcIsac_Dir2Lat(double* a, int orderCoef, float* sth, float* cth); - -#endif /* MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_CODEC_H_ */ diff --git a/modules/audio_coding/codecs/isac/main/source/crc.c b/modules/audio_coding/codecs/isac/main/source/crc.c deleted file mode 100644 index 1bb0827031..0000000000 --- a/modules/audio_coding/codecs/isac/main/source/crc.c +++ /dev/null @@ -1,111 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#include - -#include "modules/audio_coding/codecs/isac/main/source/crc.h" -#include "common_audio/signal_processing/include/signal_processing_library.h" - -#define POLYNOMIAL 0x04c11db7L - - -static const uint32_t kCrcTable[256] = { - 0, 0x4c11db7, 0x9823b6e, 0xd4326d9, 0x130476dc, 0x17c56b6b, - 0x1a864db2, 0x1e475005, 0x2608edb8, 0x22c9f00f, 0x2f8ad6d6, 0x2b4bcb61, - 0x350c9b64, 0x31cd86d3, 0x3c8ea00a, 0x384fbdbd, 0x4c11db70, 0x48d0c6c7, - 0x4593e01e, 0x4152fda9, 0x5f15adac, 0x5bd4b01b, 0x569796c2, 0x52568b75, - 0x6a1936c8, 0x6ed82b7f, 0x639b0da6, 0x675a1011, 0x791d4014, 0x7ddc5da3, - 0x709f7b7a, 0x745e66cd, 0x9823b6e0, 0x9ce2ab57, 0x91a18d8e, 0x95609039, - 0x8b27c03c, 0x8fe6dd8b, 0x82a5fb52, 0x8664e6e5, 0xbe2b5b58, 0xbaea46ef, - 0xb7a96036, 0xb3687d81, 0xad2f2d84, 0xa9ee3033, 0xa4ad16ea, 0xa06c0b5d, - 0xd4326d90, 0xd0f37027, 0xddb056fe, 0xd9714b49, 0xc7361b4c, 0xc3f706fb, - 0xceb42022, 0xca753d95, 0xf23a8028, 0xf6fb9d9f, 0xfbb8bb46, 0xff79a6f1, - 0xe13ef6f4, 0xe5ffeb43, 0xe8bccd9a, 0xec7dd02d, 0x34867077, 0x30476dc0, - 0x3d044b19, 0x39c556ae, 0x278206ab, 0x23431b1c, 0x2e003dc5, 0x2ac12072, - 0x128e9dcf, 0x164f8078, 0x1b0ca6a1, 0x1fcdbb16, 0x18aeb13, 0x54bf6a4, - 0x808d07d, 0xcc9cdca, 0x7897ab07, 0x7c56b6b0, 0x71159069, 0x75d48dde, - 0x6b93dddb, 0x6f52c06c, 0x6211e6b5, 0x66d0fb02, 0x5e9f46bf, 0x5a5e5b08, - 0x571d7dd1, 0x53dc6066, 0x4d9b3063, 0x495a2dd4, 0x44190b0d, 0x40d816ba, - 0xaca5c697, 0xa864db20, 0xa527fdf9, 0xa1e6e04e, 0xbfa1b04b, 0xbb60adfc, - 0xb6238b25, 0xb2e29692, 0x8aad2b2f, 0x8e6c3698, 0x832f1041, 0x87ee0df6, - 0x99a95df3, 0x9d684044, 0x902b669d, 0x94ea7b2a, 0xe0b41de7, 0xe4750050, - 0xe9362689, 0xedf73b3e, 0xf3b06b3b, 0xf771768c, 0xfa325055, 0xfef34de2, - 0xc6bcf05f, 0xc27dede8, 0xcf3ecb31, 0xcbffd686, 0xd5b88683, 0xd1799b34, - 0xdc3abded, 0xd8fba05a, 0x690ce0ee, 0x6dcdfd59, 0x608edb80, 0x644fc637, - 0x7a089632, 0x7ec98b85, 0x738aad5c, 0x774bb0eb, 0x4f040d56, 0x4bc510e1, - 0x46863638, 0x42472b8f, 0x5c007b8a, 0x58c1663d, 0x558240e4, 0x51435d53, - 0x251d3b9e, 0x21dc2629, 0x2c9f00f0, 0x285e1d47, 0x36194d42, 0x32d850f5, - 0x3f9b762c, 0x3b5a6b9b, 0x315d626, 0x7d4cb91, 0xa97ed48, 0xe56f0ff, - 0x1011a0fa, 0x14d0bd4d, 0x19939b94, 0x1d528623, 0xf12f560e, 0xf5ee4bb9, - 0xf8ad6d60, 0xfc6c70d7, 0xe22b20d2, 0xe6ea3d65, 0xeba91bbc, 0xef68060b, - 0xd727bbb6, 0xd3e6a601, 0xdea580d8, 0xda649d6f, 0xc423cd6a, 0xc0e2d0dd, - 0xcda1f604, 0xc960ebb3, 0xbd3e8d7e, 0xb9ff90c9, 0xb4bcb610, 0xb07daba7, - 0xae3afba2, 0xaafbe615, 0xa7b8c0cc, 0xa379dd7b, 0x9b3660c6, 0x9ff77d71, - 0x92b45ba8, 0x9675461f, 0x8832161a, 0x8cf30bad, 0x81b02d74, 0x857130c3, - 0x5d8a9099, 0x594b8d2e, 0x5408abf7, 0x50c9b640, 0x4e8ee645, 0x4a4ffbf2, - 0x470cdd2b, 0x43cdc09c, 0x7b827d21, 0x7f436096, 0x7200464f, 0x76c15bf8, - 0x68860bfd, 0x6c47164a, 0x61043093, 0x65c52d24, 0x119b4be9, 0x155a565e, - 0x18197087, 0x1cd86d30, 0x29f3d35, 0x65e2082, 0xb1d065b, 0xfdc1bec, - 0x3793a651, 0x3352bbe6, 0x3e119d3f, 0x3ad08088, 0x2497d08d, 0x2056cd3a, - 0x2d15ebe3, 0x29d4f654, 0xc5a92679, 0xc1683bce, 0xcc2b1d17, 0xc8ea00a0, - 0xd6ad50a5, 0xd26c4d12, 0xdf2f6bcb, 0xdbee767c, 0xe3a1cbc1, 0xe760d676, - 0xea23f0af, 0xeee2ed18, 0xf0a5bd1d, 0xf464a0aa, 0xf9278673, 0xfde69bc4, - 0x89b8fd09, 0x8d79e0be, 0x803ac667, 0x84fbdbd0, 0x9abc8bd5, 0x9e7d9662, - 0x933eb0bb, 0x97ffad0c, 0xafb010b1, 0xab710d06, 0xa6322bdf, 0xa2f33668, - 0xbcb4666d, 0xb8757bda, 0xb5365d03, 0xb1f740b4 -}; - - - - -/**************************************************************************** - * WebRtcIsac_GetCrc(...) - * - * This function returns a 32 bit CRC checksum of a bit stream - * - * Input: - * - bitstream : payload bitstream - * - len_bitstream_in_bytes : number of 8-bit words in the bit stream - * - * Output: - * - crc : checksum - * - * Return value : 0 - Ok - * -1 - Error - */ - -int WebRtcIsac_GetCrc(const int16_t* bitstream, - int len_bitstream_in_bytes, - uint32_t* crc) -{ - uint8_t* bitstream_ptr_uw8; - uint32_t crc_state; - int byte_cntr; - int crc_tbl_indx; - - /* Sanity Check. */ - if (bitstream == NULL) { - return -1; - } - /* cast to UWord8 pointer */ - bitstream_ptr_uw8 = (uint8_t *)bitstream; - - /* initialize */ - crc_state = 0xFFFFFFFF; - - for (byte_cntr = 0; byte_cntr < len_bitstream_in_bytes; byte_cntr++) { - crc_tbl_indx = (WEBRTC_SPL_RSHIFT_U32(crc_state, 24) ^ - bitstream_ptr_uw8[byte_cntr]) & 0xFF; - crc_state = (crc_state << 8) ^ kCrcTable[crc_tbl_indx]; - } - - *crc = ~crc_state; - return 0; -} diff --git a/modules/audio_coding/codecs/isac/main/source/crc.h b/modules/audio_coding/codecs/isac/main/source/crc.h deleted file mode 100644 index f031019ed3..0000000000 --- a/modules/audio_coding/codecs/isac/main/source/crc.h +++ /dev/null @@ -1,41 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/* - * crc.h - * - * Checksum functions - * - */ - -#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_CRC_H_ -#define MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_CRC_H_ - -#include - -/**************************************************************************** - * WebRtcIsac_GetCrc(...) - * - * This function returns a 32 bit CRC checksum of a bit stream - * - * Input: - * - encoded : payload bit stream - * - no_of_word8s : number of 8-bit words in the bit stream - * - * Output: - * - crc : checksum - * - * Return value : 0 - Ok - * -1 - Error - */ - -int WebRtcIsac_GetCrc(const int16_t* encoded, int no_of_word8s, uint32_t* crc); - -#endif /* MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_CRC_H_ */ diff --git a/modules/audio_coding/codecs/isac/main/source/decode.c b/modules/audio_coding/codecs/isac/main/source/decode.c deleted file mode 100644 index 6e114e4a2b..0000000000 --- a/modules/audio_coding/codecs/isac/main/source/decode.c +++ /dev/null @@ -1,303 +0,0 @@ -/* - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/* - * decode_B.c - * - * This file contains definition of funtions for decoding. - * Decoding of lower-band, including normal-decoding and RCU decoding. - * Decoding of upper-band, including 8-12 kHz, when the bandwidth is - * 0-12 kHz, and 8-16 kHz, when the bandwidth is 0-16 kHz. - * - */ - -#include -#include -#include - -#include "modules/audio_coding/codecs/isac/main/source/codec.h" -#include "modules/audio_coding/codecs/isac/main/source/entropy_coding.h" -#include "modules/audio_coding/codecs/isac/main/source/pitch_estimator.h" -#include "modules/audio_coding/codecs/isac/main/source/bandwidth_estimator.h" -#include "modules/audio_coding/codecs/isac/main/source/structs.h" -#include "modules/audio_coding/codecs/isac/main/source/settings.h" -#include "modules/audio_coding/codecs/isac/main/source/pitch_filter.h" - -/* - * function to decode the bitstream - * returns the total number of bytes in the stream - */ -int WebRtcIsac_DecodeLb(const TransformTables* transform_tables, - float* signal_out, ISACLBDecStruct* ISACdecLB_obj, - int16_t* current_framesamples, - int16_t isRCUPayload) { - int k; - int len, err; - int16_t bandwidthInd; - - float LP_dec_float[FRAMESAMPLES_HALF]; - float HP_dec_float[FRAMESAMPLES_HALF]; - - double LPw[FRAMESAMPLES_HALF]; - double HPw[FRAMESAMPLES_HALF]; - double LPw_pf[FRAMESAMPLES_HALF]; - - double lo_filt_coef[(ORDERLO + 1)*SUBFRAMES]; - double hi_filt_coef[(ORDERHI + 1)*SUBFRAMES]; - - double real_f[FRAMESAMPLES_HALF]; - double imag_f[FRAMESAMPLES_HALF]; - - double PitchLags[4]; - double PitchGains[4]; - double AvgPitchGain; - int16_t PitchGains_Q12[4]; - int16_t AvgPitchGain_Q12; - - float gain; - - int frame_nb; /* counter */ - int frame_mode; /* 0 30ms, 1 for 60ms */ - /* Processed_samples: 480 (30, 60 ms). Cannot take other values. */ - - WebRtcIsac_ResetBitstream(&(ISACdecLB_obj->bitstr_obj)); - - len = 0; - - /* Decode framelength and BW estimation - not used, - only for stream pointer*/ - err = WebRtcIsac_DecodeFrameLen(&ISACdecLB_obj->bitstr_obj, - current_framesamples); - if (err < 0) { - return err; - } - - /* Frame_mode: - * 0: indicates 30 ms frame (480 samples) - * 1: indicates 60 ms frame (960 samples) */ - frame_mode = *current_framesamples / MAX_FRAMESAMPLES; - - err = WebRtcIsac_DecodeSendBW(&ISACdecLB_obj->bitstr_obj, &bandwidthInd); - if (err < 0) { - return err; - } - - /* One loop if it's one frame (20 or 30ms), 2 loops if 2 frames - bundled together (60ms). */ - for (frame_nb = 0; frame_nb <= frame_mode; frame_nb++) { - /* Decode & de-quantize pitch parameters */ - err = WebRtcIsac_DecodePitchGain(&ISACdecLB_obj->bitstr_obj, - PitchGains_Q12); - if (err < 0) { - return err; - } - - err = WebRtcIsac_DecodePitchLag(&ISACdecLB_obj->bitstr_obj, PitchGains_Q12, - PitchLags); - if (err < 0) { - return err; - } - - AvgPitchGain_Q12 = (PitchGains_Q12[0] + PitchGains_Q12[1] + - PitchGains_Q12[2] + PitchGains_Q12[3]) >> 2; - - /* Decode & de-quantize filter coefficients. */ - err = WebRtcIsac_DecodeLpc(&ISACdecLB_obj->bitstr_obj, lo_filt_coef, - hi_filt_coef); - if (err < 0) { - return err; - } - /* Decode & de-quantize spectrum. */ - len = WebRtcIsac_DecodeSpec(&ISACdecLB_obj->bitstr_obj, AvgPitchGain_Q12, - kIsacLowerBand, real_f, imag_f); - if (len < 0) { - return len; - } - - /* Inverse transform. */ - WebRtcIsac_Spec2time(transform_tables, real_f, imag_f, LPw, HPw, - &ISACdecLB_obj->fftstr_obj); - - /* Convert PitchGains back to float for pitchfilter_post */ - for (k = 0; k < 4; k++) { - PitchGains[k] = ((float)PitchGains_Q12[k]) / 4096; - } - if (isRCUPayload) { - for (k = 0; k < 240; k++) { - LPw[k] *= RCU_TRANSCODING_SCALE_INVERSE; - HPw[k] *= RCU_TRANSCODING_SCALE_INVERSE; - } - } - - /* Inverse pitch filter. */ - WebRtcIsac_PitchfilterPost(LPw, LPw_pf, &ISACdecLB_obj->pitchfiltstr_obj, - PitchLags, PitchGains); - /* Convert AvgPitchGain back to float for computation of gain. */ - AvgPitchGain = ((float)AvgPitchGain_Q12) / 4096; - gain = 1.0f - 0.45f * (float)AvgPitchGain; - - for (k = 0; k < FRAMESAMPLES_HALF; k++) { - /* Reduce gain to compensate for pitch enhancer. */ - LPw_pf[k] *= gain; - } - - if (isRCUPayload) { - for (k = 0; k < FRAMESAMPLES_HALF; k++) { - /* Compensation for transcoding gain changes. */ - LPw_pf[k] *= RCU_TRANSCODING_SCALE; - HPw[k] *= RCU_TRANSCODING_SCALE; - } - } - /* Perceptual post-filtering (using normalized lattice filter). */ - WebRtcIsac_NormLatticeFilterAr( - ORDERLO, ISACdecLB_obj->maskfiltstr_obj.PostStateLoF, - (ISACdecLB_obj->maskfiltstr_obj).PostStateLoG, LPw_pf, lo_filt_coef, - LP_dec_float); - WebRtcIsac_NormLatticeFilterAr( - ORDERHI, ISACdecLB_obj->maskfiltstr_obj.PostStateHiF, - (ISACdecLB_obj->maskfiltstr_obj).PostStateHiG, HPw, hi_filt_coef, - HP_dec_float); - - /* Recombine the 2 bands. */ - WebRtcIsac_FilterAndCombineFloat(LP_dec_float, HP_dec_float, - signal_out + frame_nb * FRAMESAMPLES, - &ISACdecLB_obj->postfiltbankstr_obj); - } - return len; -} - - -/* - * This decode function is called when the codec is operating in 16 kHz - * bandwidth to decode the upperband, i.e. 8-16 kHz. - * - * Contrary to lower-band, the upper-band (8-16 kHz) is not split in - * frequency, but split to 12 sub-frames, i.e. twice as lower-band. - */ -int WebRtcIsac_DecodeUb16(const TransformTables* transform_tables, - float* signal_out, ISACUBDecStruct* ISACdecUB_obj, - int16_t isRCUPayload) { - int len, err; - - double halfFrameFirst[FRAMESAMPLES_HALF]; - double halfFrameSecond[FRAMESAMPLES_HALF]; - - double percepFilterParam[(UB_LPC_ORDER + 1) * (SUBFRAMES << 1) + - (UB_LPC_ORDER + 1)]; - - double real_f[FRAMESAMPLES_HALF]; - double imag_f[FRAMESAMPLES_HALF]; - const int16_t kAveragePitchGain = 0; /* No pitch-gain for upper-band. */ - len = 0; - - /* Decode & de-quantize filter coefficients. */ - memset(percepFilterParam, 0, sizeof(percepFilterParam)); - err = WebRtcIsac_DecodeInterpolLpcUb(&ISACdecUB_obj->bitstr_obj, - percepFilterParam, isac16kHz); - if (err < 0) { - return err; - } - - /* Decode & de-quantize spectrum. */ - len = WebRtcIsac_DecodeSpec(&ISACdecUB_obj->bitstr_obj, kAveragePitchGain, - kIsacUpperBand16, real_f, imag_f); - if (len < 0) { - return len; - } - if (isRCUPayload) { - int n; - for (n = 0; n < 240; n++) { - real_f[n] *= RCU_TRANSCODING_SCALE_UB_INVERSE; - imag_f[n] *= RCU_TRANSCODING_SCALE_UB_INVERSE; - } - } - /* Inverse transform. */ - WebRtcIsac_Spec2time(transform_tables, - real_f, imag_f, halfFrameFirst, halfFrameSecond, - &ISACdecUB_obj->fftstr_obj); - - /* Perceptual post-filtering (using normalized lattice filter). */ - WebRtcIsac_NormLatticeFilterAr( - UB_LPC_ORDER, ISACdecUB_obj->maskfiltstr_obj.PostStateLoF, - (ISACdecUB_obj->maskfiltstr_obj).PostStateLoG, halfFrameFirst, - &percepFilterParam[(UB_LPC_ORDER + 1)], signal_out); - - WebRtcIsac_NormLatticeFilterAr( - UB_LPC_ORDER, ISACdecUB_obj->maskfiltstr_obj.PostStateLoF, - (ISACdecUB_obj->maskfiltstr_obj).PostStateLoG, halfFrameSecond, - &percepFilterParam[(UB_LPC_ORDER + 1) * SUBFRAMES + (UB_LPC_ORDER + 1)], - &signal_out[FRAMESAMPLES_HALF]); - - return len; -} - -/* - * This decode function is called when the codec operates at 0-12 kHz - * bandwidth to decode the upperband, i.e. 8-12 kHz. - * - * At the encoder the upper-band is split into two band, 8-12 kHz & 12-16 - * kHz, and only 8-12 kHz is encoded. At the decoder, 8-12 kHz band is - * reconstructed and 12-16 kHz replaced with zeros. Then two bands - * are combined, to reconstruct the upperband 8-16 kHz. - */ -int WebRtcIsac_DecodeUb12(const TransformTables* transform_tables, - float* signal_out, ISACUBDecStruct* ISACdecUB_obj, - int16_t isRCUPayload) { - int len, err; - - float LP_dec_float[FRAMESAMPLES_HALF]; - float HP_dec_float[FRAMESAMPLES_HALF]; - - double LPw[FRAMESAMPLES_HALF]; - double HPw[FRAMESAMPLES_HALF]; - - double percepFilterParam[(UB_LPC_ORDER + 1)*SUBFRAMES]; - - double real_f[FRAMESAMPLES_HALF]; - double imag_f[FRAMESAMPLES_HALF]; - const int16_t kAveragePitchGain = 0; /* No pitch-gain for upper-band. */ - len = 0; - - /* Decode & dequantize filter coefficients. */ - err = WebRtcIsac_DecodeInterpolLpcUb(&ISACdecUB_obj->bitstr_obj, - percepFilterParam, isac12kHz); - if (err < 0) { - return err; - } - - /* Decode & de-quantize spectrum. */ - len = WebRtcIsac_DecodeSpec(&ISACdecUB_obj->bitstr_obj, kAveragePitchGain, - kIsacUpperBand12, real_f, imag_f); - if (len < 0) { - return len; - } - - if (isRCUPayload) { - int n; - for (n = 0; n < 240; n++) { - real_f[n] *= RCU_TRANSCODING_SCALE_UB_INVERSE; - imag_f[n] *= RCU_TRANSCODING_SCALE_UB_INVERSE; - } - } - /* Inverse transform. */ - WebRtcIsac_Spec2time(transform_tables, - real_f, imag_f, LPw, HPw, &ISACdecUB_obj->fftstr_obj); - /* perceptual post-filtering (using normalized lattice filter) */ - WebRtcIsac_NormLatticeFilterAr(UB_LPC_ORDER, - ISACdecUB_obj->maskfiltstr_obj.PostStateLoF, - (ISACdecUB_obj->maskfiltstr_obj).PostStateLoG, - LPw, percepFilterParam, LP_dec_float); - /* Zero for 12-16 kHz. */ - memset(HP_dec_float, 0, sizeof(float) * (FRAMESAMPLES_HALF)); - /* Recombine the 2 bands. */ - WebRtcIsac_FilterAndCombineFloat(HP_dec_float, LP_dec_float, signal_out, - &ISACdecUB_obj->postfiltbankstr_obj); - return len; -} diff --git a/modules/audio_coding/codecs/isac/main/source/decode_bwe.c b/modules/audio_coding/codecs/isac/main/source/decode_bwe.c deleted file mode 100644 index 89d970fc75..0000000000 --- a/modules/audio_coding/codecs/isac/main/source/decode_bwe.c +++ /dev/null @@ -1,89 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#include "modules/audio_coding/codecs/isac/main/source/structs.h" -#include "modules/audio_coding/codecs/isac/main/source/bandwidth_estimator.h" -#include "modules/audio_coding/codecs/isac/main/source/entropy_coding.h" -#include "modules/audio_coding/codecs/isac/main/source/codec.h" - - -int -WebRtcIsac_EstimateBandwidth( - BwEstimatorstr* bwest_str, - Bitstr* streamdata, - size_t packet_size, - uint16_t rtp_seq_number, - uint32_t send_ts, - uint32_t arr_ts, - enum IsacSamplingRate encoderSampRate, - enum IsacSamplingRate decoderSampRate) -{ - int16_t index; - int16_t frame_samples; - uint32_t sendTimestampIn16kHz; - uint32_t arrivalTimestampIn16kHz; - uint32_t diffSendTime; - uint32_t diffArrivalTime; - int err; - - /* decode framelength and BW estimation */ - err = WebRtcIsac_DecodeFrameLen(streamdata, &frame_samples); - if(err < 0) // error check - { - return err; - } - err = WebRtcIsac_DecodeSendBW(streamdata, &index); - if(err < 0) // error check - { - return err; - } - - /* UPDATE ESTIMATES FROM OTHER SIDE */ - err = WebRtcIsac_UpdateUplinkBwImpl(bwest_str, index, encoderSampRate); - if(err < 0) - { - return err; - } - - // We like BWE to work at 16 kHz sampling rate, - // therefore, we have to change the timestamps accordingly. - // translate the send timestamp if required - diffSendTime = (uint32_t)((uint32_t)send_ts - - (uint32_t)bwest_str->senderTimestamp); - bwest_str->senderTimestamp = send_ts; - - diffArrivalTime = (uint32_t)((uint32_t)arr_ts - - (uint32_t)bwest_str->receiverTimestamp); - bwest_str->receiverTimestamp = arr_ts; - - if(decoderSampRate == kIsacSuperWideband) - { - diffArrivalTime = (uint32_t)diffArrivalTime >> 1; - diffSendTime = (uint32_t)diffSendTime >> 1; - } - - // arrival timestamp in 16 kHz - arrivalTimestampIn16kHz = (uint32_t)((uint32_t) - bwest_str->prev_rec_arr_ts + (uint32_t)diffArrivalTime); - // send timestamp in 16 kHz - sendTimestampIn16kHz = (uint32_t)((uint32_t) - bwest_str->prev_rec_send_ts + (uint32_t)diffSendTime); - - err = WebRtcIsac_UpdateBandwidthEstimator(bwest_str, rtp_seq_number, - (frame_samples * 1000) / FS, sendTimestampIn16kHz, - arrivalTimestampIn16kHz, packet_size); - // error check - if(err < 0) - { - return err; - } - - return 0; -} diff --git a/modules/audio_coding/codecs/isac/main/source/encode.c b/modules/audio_coding/codecs/isac/main/source/encode.c deleted file mode 100644 index bf92d02c53..0000000000 --- a/modules/audio_coding/codecs/isac/main/source/encode.c +++ /dev/null @@ -1,1260 +0,0 @@ -/* - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/* - * encode.c - * - * This file contains definition of funtions for encoding. - * Decoding of upper-band, including 8-12 kHz, when the bandwidth is - * 0-12 kHz, and 8-16 kHz, when the bandwidth is 0-16 kHz. - * - */ - -#include -#include -#include - -#include "modules/audio_coding/codecs/isac/main/source/structs.h" -#include "modules/audio_coding/codecs/isac/main/source/codec.h" -#include "modules/audio_coding/codecs/isac/main/source/pitch_estimator.h" -#include "modules/audio_coding/codecs/isac/main/source/entropy_coding.h" -#include "modules/audio_coding/codecs/isac/main/source/arith_routines.h" -#include "modules/audio_coding/codecs/isac/main/source/pitch_gain_tables.h" -#include "modules/audio_coding/codecs/isac/main/source/pitch_lag_tables.h" -#include "modules/audio_coding/codecs/isac/main/source/spectrum_ar_model_tables.h" -#include "modules/audio_coding/codecs/isac/main/source/lpc_tables.h" -#include "modules/audio_coding/codecs/isac/main/source/lpc_analysis.h" -#include "modules/audio_coding/codecs/isac/main/source/bandwidth_estimator.h" -#include "modules/audio_coding/codecs/isac/main/source/lpc_shape_swb12_tables.h" -#include "modules/audio_coding/codecs/isac/main/source/lpc_shape_swb16_tables.h" -#include "modules/audio_coding/codecs/isac/main/source/lpc_gain_swb_tables.h" -#include "modules/audio_coding/codecs/isac/main/source/isac_vad.h" -#include "modules/audio_coding/codecs/isac/main/source/pitch_filter.h" - - -#define UB_LOOKAHEAD 24 - - -/* - Rate allocation tables of lower and upper-band bottleneck for - 12kHz & 16kHz bandwidth. - - 12 kHz bandwidth - ----------------- - The overall bottleneck of the coder is between 38 kbps and 45 kbps. We have - considered 7 enteries, uniformly distributed in this interval, i.e. 38, - 39.17, 40.33, 41.5, 42.67, 43.83 and 45. For every entery, the lower-band - and the upper-band bottlenecks are specified in - 'kLowerBandBitRate12' and 'kUpperBandBitRate12' - tables, respectively. E.g. the overall rate of 41.5 kbps corresponts to a - bottleneck of 31 kbps for lower-band and 27 kbps for upper-band. Given an - overall bottleneck of the codec, we use linear interpolation to get - lower-band and upper-band bottlenecks. - - 16 kHz bandwidth - ----------------- - The overall bottleneck of the coder is between 50 kbps and 56 kbps. We have - considered 7 enteries, uniformly distributed in this interval, i.e. 50, 51.2, - 52.4, 53.6, 54.8 and 56. For every entery, the lower-band and the upper-band - bottlenecks are specified in 'kLowerBandBitRate16' and - 'kUpperBandBitRate16' tables, respectively. E.g. the overall rate - of 53.6 kbps corresponts to a bottleneck of 32 kbps for lower-band and 30 - kbps for upper-band. Given an overall bottleneck of the codec, we use linear - interpolation to get lower-band and upper-band bottlenecks. - - */ - -/* 38 39.17 40.33 41.5 42.67 43.83 45 */ -static const int16_t kLowerBandBitRate12[7] = { - 29000, 30000, 30000, 31000, 31000, 32000, 32000 }; -static const int16_t kUpperBandBitRate12[7] = { - 25000, 25000, 27000, 27000, 29000, 29000, 32000 }; - -/* 50 51.2 52.4 53.6 54.8 56 */ -static const int16_t kLowerBandBitRate16[6] = { - 31000, 31000, 32000, 32000, 32000, 32000 }; -static const int16_t kUpperBandBitRate16[6] = { - 28000, 29000, 29000, 30000, 31000, 32000 }; - -/****************************************************************************** - * WebRtcIsac_RateAllocation() - * Internal function to perform a rate-allocation for upper and lower-band, - * given a total rate. - * - * Input: - * - inRateBitPerSec : a total bottleneck in bits/sec. - * - * Output: - * - rateLBBitPerSec : a bottleneck allocated to the lower-band - * in bits/sec. - * - rateUBBitPerSec : a bottleneck allocated to the upper-band - * in bits/sec. - * - * Return value : 0 if rate allocation has been successful. - * -1 if failed to allocate rates. - */ - -int16_t WebRtcIsac_RateAllocation(int32_t inRateBitPerSec, - double* rateLBBitPerSec, - double* rateUBBitPerSec, - enum ISACBandwidth* bandwidthKHz) { - int16_t idx; - double idxD; - double idxErr; - if (inRateBitPerSec < 38000) { - /* If the given overall bottleneck is less than 38000 then - * then codec has to operate in wideband mode, i.e. 8 kHz - * bandwidth. */ - *rateLBBitPerSec = (int16_t)((inRateBitPerSec > 32000) ? - 32000 : inRateBitPerSec); - *rateUBBitPerSec = 0; - *bandwidthKHz = isac8kHz; - } else if ((inRateBitPerSec >= 38000) && (inRateBitPerSec < 50000)) { - /* At a bottleneck between 38 and 50 kbps the codec is operating - * at 12 kHz bandwidth. Using xxxBandBitRate12[] to calculates - * upper/lower bottleneck */ - - /* Find the bottlenecks by linear interpolation, - * step is (45000 - 38000)/6.0 we use the inverse of it. */ - const double stepSizeInv = 8.5714286e-4; - idxD = (inRateBitPerSec - 38000) * stepSizeInv; - idx = (idxD >= 6) ? 6 : ((int16_t)idxD); - idxErr = idxD - idx; - *rateLBBitPerSec = kLowerBandBitRate12[idx]; - *rateUBBitPerSec = kUpperBandBitRate12[idx]; - - if (idx < 6) { - *rateLBBitPerSec += (int16_t)( - idxErr * (kLowerBandBitRate12[idx + 1] - kLowerBandBitRate12[idx])); - *rateUBBitPerSec += (int16_t)( - idxErr * (kUpperBandBitRate12[idx + 1] - kUpperBandBitRate12[idx])); - } - *bandwidthKHz = isac12kHz; - } else if ((inRateBitPerSec >= 50000) && (inRateBitPerSec <= 56000)) { - /* A bottleneck between 50 and 56 kbps corresponds to bandwidth - * of 16 kHz. Using xxxBandBitRate16[] to calculates - * upper/lower bottleneck. */ - - /* Find the bottlenecks by linear interpolation - * step is (56000 - 50000)/5 we use the inverse of it. */ - const double stepSizeInv = 8.3333333e-4; - idxD = (inRateBitPerSec - 50000) * stepSizeInv; - idx = (idxD >= 5) ? 5 : ((int16_t)idxD); - idxErr = idxD - idx; - *rateLBBitPerSec = kLowerBandBitRate16[idx]; - *rateUBBitPerSec = kUpperBandBitRate16[idx]; - - if (idx < 5) { - *rateLBBitPerSec += (int16_t)(idxErr * - (kLowerBandBitRate16[idx + 1] - - kLowerBandBitRate16[idx])); - - *rateUBBitPerSec += (int16_t)(idxErr * - (kUpperBandBitRate16[idx + 1] - - kUpperBandBitRate16[idx])); - } - *bandwidthKHz = isac16kHz; - } else { - /* Out-of-range botlteneck value. */ - return -1; - } - - /* limit the values. */ - *rateLBBitPerSec = (*rateLBBitPerSec > 32000) ? 32000 : *rateLBBitPerSec; - *rateUBBitPerSec = (*rateUBBitPerSec > 32000) ? 32000 : *rateUBBitPerSec; - return 0; -} - - -void WebRtcIsac_ResetBitstream(Bitstr* bit_stream) { - bit_stream->W_upper = 0xFFFFFFFF; - bit_stream->stream_index = 0; - bit_stream->streamval = 0; -} - -int WebRtcIsac_EncodeLb(const TransformTables* transform_tables, - float* in, ISACLBEncStruct* ISACencLB_obj, - int16_t codingMode, - int16_t bottleneckIndex) { - int stream_length = 0; - int err; - int k; - int iterCntr; - - double lofilt_coef[(ORDERLO + 1)*SUBFRAMES]; - double hifilt_coef[(ORDERHI + 1)*SUBFRAMES]; - float LP[FRAMESAMPLES_HALF]; - float HP[FRAMESAMPLES_HALF]; - - double LP_lookahead[FRAMESAMPLES_HALF]; - double HP_lookahead[FRAMESAMPLES_HALF]; - double LP_lookahead_pf[FRAMESAMPLES_HALF + QLOOKAHEAD]; - double LPw[FRAMESAMPLES_HALF]; - - double HPw[FRAMESAMPLES_HALF]; - double LPw_pf[FRAMESAMPLES_HALF]; - int16_t fre[FRAMESAMPLES_HALF]; /* Q7 */ - int16_t fim[FRAMESAMPLES_HALF]; /* Q7 */ - - double PitchLags[4]; - double PitchGains[4]; - int16_t PitchGains_Q12[4]; - int16_t AvgPitchGain_Q12; - - int frame_mode; /* 0 for 30ms, 1 for 60ms */ - int status = 0; - int my_index; - transcode_obj transcodingParam; - double bytesLeftSpecCoding; - uint16_t payloadLimitBytes; - - /* Copy new frame-length and bottleneck rate only for the first 10 ms data */ - if (ISACencLB_obj->buffer_index == 0) { - /* Set the framelength for the next packet. */ - ISACencLB_obj->current_framesamples = ISACencLB_obj->new_framelength; - } - /* 'frame_mode' is 0 (30 ms) or 1 (60 ms). */ - frame_mode = ISACencLB_obj->current_framesamples / MAX_FRAMESAMPLES; - - /* buffer speech samples (by 10ms packet) until the frame-length */ - /* is reached (30 or 60 ms). */ - /*****************************************************************/ - - /* fill the buffer with 10ms input data */ - for (k = 0; k < FRAMESAMPLES_10ms; k++) { - ISACencLB_obj->data_buffer_float[k + ISACencLB_obj->buffer_index] = in[k]; - } - - /* If buffersize is not equal to current framesize then increase index - * and return. We do no encoding untill we have enough audio. */ - if (ISACencLB_obj->buffer_index + FRAMESAMPLES_10ms != FRAMESAMPLES) { - ISACencLB_obj->buffer_index += FRAMESAMPLES_10ms; - return 0; - } - /* If buffer reached the right size, reset index and continue with - * encoding the frame. */ - ISACencLB_obj->buffer_index = 0; - - /* End of buffer function. */ - /**************************/ - - /* Encoding */ - /************/ - - if (frame_mode == 0 || ISACencLB_obj->frame_nb == 0) { - /* This is to avoid Linux warnings until we change 'int' to 'Word32' - * at all places. */ - int intVar; - /* reset bitstream */ - WebRtcIsac_ResetBitstream(&(ISACencLB_obj->bitstr_obj)); - - if ((codingMode == 0) && (frame_mode == 0) && - (ISACencLB_obj->enforceFrameSize == 0)) { - ISACencLB_obj->new_framelength = WebRtcIsac_GetNewFrameLength( - ISACencLB_obj->bottleneck, ISACencLB_obj->current_framesamples); - } - - ISACencLB_obj->s2nr = WebRtcIsac_GetSnr( - ISACencLB_obj->bottleneck, ISACencLB_obj->current_framesamples); - - /* Encode frame length. */ - status = WebRtcIsac_EncodeFrameLen( - ISACencLB_obj->current_framesamples, &ISACencLB_obj->bitstr_obj); - if (status < 0) { - /* Wrong frame size. */ - return status; - } - /* Save framelength for multiple packets memory. */ - ISACencLB_obj->SaveEnc_obj.framelength = - ISACencLB_obj->current_framesamples; - - /* To be used for Redundant Coding. */ - ISACencLB_obj->lastBWIdx = bottleneckIndex; - intVar = (int)bottleneckIndex; - WebRtcIsac_EncodeReceiveBw(&intVar, &ISACencLB_obj->bitstr_obj); - } - - /* Split signal in two bands. */ - WebRtcIsac_SplitAndFilterFloat(ISACencLB_obj->data_buffer_float, LP, HP, - LP_lookahead, HP_lookahead, - &ISACencLB_obj->prefiltbankstr_obj); - - /* estimate pitch parameters and pitch-filter lookahead signal */ - WebRtcIsac_PitchAnalysis(LP_lookahead, LP_lookahead_pf, - &ISACencLB_obj->pitchanalysisstr_obj, PitchLags, - PitchGains); - - /* Encode in FIX Q12. */ - - /* Convert PitchGain to Fixed point. */ - for (k = 0; k < PITCH_SUBFRAMES; k++) { - PitchGains_Q12[k] = (int16_t)(PitchGains[k] * 4096.0); - } - - /* Set where to store data in multiple packets memory. */ - if (frame_mode == 0 || ISACencLB_obj->frame_nb == 0) { - ISACencLB_obj->SaveEnc_obj.startIdx = 0; - } else { - ISACencLB_obj->SaveEnc_obj.startIdx = 1; - } - - /* Quantize & encode pitch parameters. */ - WebRtcIsac_EncodePitchGain(PitchGains_Q12, &ISACencLB_obj->bitstr_obj, - &ISACencLB_obj->SaveEnc_obj); - WebRtcIsac_EncodePitchLag(PitchLags, PitchGains_Q12, - &ISACencLB_obj->bitstr_obj, - &ISACencLB_obj->SaveEnc_obj); - - AvgPitchGain_Q12 = (PitchGains_Q12[0] + PitchGains_Q12[1] + - PitchGains_Q12[2] + PitchGains_Q12[3]) >> 2; - - /* Find coefficients for perceptual pre-filters. */ - WebRtcIsac_GetLpcCoefLb(LP_lookahead_pf, HP_lookahead, - &ISACencLB_obj->maskfiltstr_obj, ISACencLB_obj->s2nr, - PitchGains_Q12, lofilt_coef, hifilt_coef); - - /* Code LPC model and shape - gains not quantized yet. */ - WebRtcIsac_EncodeLpcLb(lofilt_coef, hifilt_coef, &ISACencLB_obj->bitstr_obj, - &ISACencLB_obj->SaveEnc_obj); - - /* Convert PitchGains back to FLOAT for pitchfilter_pre. */ - for (k = 0; k < 4; k++) { - PitchGains[k] = ((float)PitchGains_Q12[k]) / 4096; - } - - /* Store the state of arithmetic coder before coding LPC gains. */ - transcodingParam.W_upper = ISACencLB_obj->bitstr_obj.W_upper; - transcodingParam.stream_index = ISACencLB_obj->bitstr_obj.stream_index; - transcodingParam.streamval = ISACencLB_obj->bitstr_obj.streamval; - transcodingParam.stream[0] = - ISACencLB_obj->bitstr_obj.stream[ISACencLB_obj->bitstr_obj.stream_index - - 2]; - transcodingParam.stream[1] = - ISACencLB_obj->bitstr_obj.stream[ISACencLB_obj->bitstr_obj.stream_index - - 1]; - transcodingParam.stream[2] = - ISACencLB_obj->bitstr_obj.stream[ISACencLB_obj->bitstr_obj.stream_index]; - - /* Store LPC Gains before encoding them. */ - for (k = 0; k < SUBFRAMES; k++) { - transcodingParam.loFiltGain[k] = lofilt_coef[(LPC_LOBAND_ORDER + 1) * k]; - transcodingParam.hiFiltGain[k] = hifilt_coef[(LPC_HIBAND_ORDER + 1) * k]; - } - - /* Code gains */ - WebRtcIsac_EncodeLpcGainLb(lofilt_coef, hifilt_coef, - &ISACencLB_obj->bitstr_obj, - &ISACencLB_obj->SaveEnc_obj); - - /* Get the correct value for the payload limit and calculate the - * number of bytes left for coding the spectrum. */ - if ((frame_mode == 1) && (ISACencLB_obj->frame_nb == 0)) { - /* It is a 60ms and we are in the first 30ms then the limit at - * this point should be half of the assigned value. */ - payloadLimitBytes = ISACencLB_obj->payloadLimitBytes60 >> 1; - } else if (frame_mode == 0) { - /* It is a 30ms frame */ - /* Subract 3 because termination process may add 3 bytes. */ - payloadLimitBytes = ISACencLB_obj->payloadLimitBytes30 - 3; - } else { - /* This is the second half of a 60ms frame. */ - /* Subract 3 because termination process may add 3 bytes. */ - payloadLimitBytes = ISACencLB_obj->payloadLimitBytes60 - 3; - } - bytesLeftSpecCoding = payloadLimitBytes - transcodingParam.stream_index; - - /* Perceptual pre-filtering (using normalized lattice filter). */ - /* Low-band filtering. */ - WebRtcIsac_NormLatticeFilterMa(ORDERLO, - ISACencLB_obj->maskfiltstr_obj.PreStateLoF, - ISACencLB_obj->maskfiltstr_obj.PreStateLoG, - LP, lofilt_coef, LPw); - /* High-band filtering. */ - WebRtcIsac_NormLatticeFilterMa(ORDERHI, - ISACencLB_obj->maskfiltstr_obj.PreStateHiF, - ISACencLB_obj->maskfiltstr_obj.PreStateHiG, - HP, hifilt_coef, HPw); - /* Pitch filter. */ - WebRtcIsac_PitchfilterPre(LPw, LPw_pf, &ISACencLB_obj->pitchfiltstr_obj, - PitchLags, PitchGains); - /* Transform */ - WebRtcIsac_Time2Spec(transform_tables, - LPw_pf, HPw, fre, fim, &ISACencLB_obj->fftstr_obj); - - /* Save data for multiple packets memory. */ - my_index = ISACencLB_obj->SaveEnc_obj.startIdx * FRAMESAMPLES_HALF; - memcpy(&ISACencLB_obj->SaveEnc_obj.fre[my_index], fre, sizeof(fre)); - memcpy(&ISACencLB_obj->SaveEnc_obj.fim[my_index], fim, sizeof(fim)); - - ISACencLB_obj->SaveEnc_obj.AvgPitchGain[ISACencLB_obj->SaveEnc_obj.startIdx] = - AvgPitchGain_Q12; - - /* Quantization and loss-less coding. */ - err = WebRtcIsac_EncodeSpec(fre, fim, AvgPitchGain_Q12, kIsacLowerBand, - &ISACencLB_obj->bitstr_obj); - if ((err < 0) && (err != -ISAC_DISALLOWED_BITSTREAM_LENGTH)) { - /* There has been an error but it was not too large payload - (we can cure too large payload). */ - if (frame_mode == 1 && ISACencLB_obj->frame_nb == 1) { - /* If this is the second 30ms of a 60ms frame reset - this such that in the next call encoder starts fresh. */ - ISACencLB_obj->frame_nb = 0; - } - return err; - } - iterCntr = 0; - while ((ISACencLB_obj->bitstr_obj.stream_index > payloadLimitBytes) || - (err == -ISAC_DISALLOWED_BITSTREAM_LENGTH)) { - double bytesSpecCoderUsed; - double transcodeScale; - - if (iterCntr >= MAX_PAYLOAD_LIMIT_ITERATION) { - /* We were not able to limit the payload size */ - if ((frame_mode == 1) && (ISACencLB_obj->frame_nb == 0)) { - /* This was the first 30ms of a 60ms frame. Although - the payload is larger than it should be but we let - the second 30ms be encoded. Maybe together we - won't exceed the limit. */ - ISACencLB_obj->frame_nb = 1; - return 0; - } else if ((frame_mode == 1) && (ISACencLB_obj->frame_nb == 1)) { - ISACencLB_obj->frame_nb = 0; - } - - if (err != -ISAC_DISALLOWED_BITSTREAM_LENGTH) { - return -ISAC_PAYLOAD_LARGER_THAN_LIMIT; - } else { - return status; - } - } - - if (err == -ISAC_DISALLOWED_BITSTREAM_LENGTH) { - bytesSpecCoderUsed = STREAM_SIZE_MAX; - /* Being conservative */ - transcodeScale = bytesLeftSpecCoding / bytesSpecCoderUsed * 0.5; - } else { - bytesSpecCoderUsed = ISACencLB_obj->bitstr_obj.stream_index - - transcodingParam.stream_index; - transcodeScale = bytesLeftSpecCoding / bytesSpecCoderUsed; - } - - /* To be safe, we reduce the scale depending on - the number of iterations. */ - transcodeScale *= (1.0 - (0.9 * (double)iterCntr / - (double)MAX_PAYLOAD_LIMIT_ITERATION)); - - /* Scale the LPC Gains. */ - for (k = 0; k < SUBFRAMES; k++) { - lofilt_coef[(LPC_LOBAND_ORDER + 1) * k] = - transcodingParam.loFiltGain[k] * transcodeScale; - hifilt_coef[(LPC_HIBAND_ORDER + 1) * k] = - transcodingParam.hiFiltGain[k] * transcodeScale; - transcodingParam.loFiltGain[k] = lofilt_coef[(LPC_LOBAND_ORDER + 1) * k]; - transcodingParam.hiFiltGain[k] = hifilt_coef[(LPC_HIBAND_ORDER + 1) * k]; - } - - /* Scale DFT coefficients. */ - for (k = 0; k < FRAMESAMPLES_HALF; k++) { - fre[k] = (int16_t)(fre[k] * transcodeScale); - fim[k] = (int16_t)(fim[k] * transcodeScale); - } - - /* Save data for multiple packets memory. */ - my_index = ISACencLB_obj->SaveEnc_obj.startIdx * FRAMESAMPLES_HALF; - memcpy(&ISACencLB_obj->SaveEnc_obj.fre[my_index], fre, sizeof(fre)); - memcpy(&ISACencLB_obj->SaveEnc_obj.fim[my_index], fim, sizeof(fim)); - - /* Re-store the state of arithmetic coder before coding LPC gains. */ - ISACencLB_obj->bitstr_obj.W_upper = transcodingParam.W_upper; - ISACencLB_obj->bitstr_obj.stream_index = transcodingParam.stream_index; - ISACencLB_obj->bitstr_obj.streamval = transcodingParam.streamval; - ISACencLB_obj->bitstr_obj.stream[transcodingParam.stream_index - 2] = - transcodingParam.stream[0]; - ISACencLB_obj->bitstr_obj.stream[transcodingParam.stream_index - 1] = - transcodingParam.stream[1]; - ISACencLB_obj->bitstr_obj.stream[transcodingParam.stream_index] = - transcodingParam.stream[2]; - - /* Code gains. */ - WebRtcIsac_EncodeLpcGainLb(lofilt_coef, hifilt_coef, - &ISACencLB_obj->bitstr_obj, - &ISACencLB_obj->SaveEnc_obj); - - /* Update the number of bytes left for encoding the spectrum. */ - bytesLeftSpecCoding = payloadLimitBytes - transcodingParam.stream_index; - - /* Encode the spectrum. */ - err = WebRtcIsac_EncodeSpec(fre, fim, AvgPitchGain_Q12, kIsacLowerBand, - &ISACencLB_obj->bitstr_obj); - - if ((err < 0) && (err != -ISAC_DISALLOWED_BITSTREAM_LENGTH)) { - /* There has been an error but it was not too large - payload (we can cure too large payload). */ - if (frame_mode == 1 && ISACencLB_obj->frame_nb == 1) { - /* If this is the second 30 ms of a 60 ms frame reset - this such that in the next call encoder starts fresh. */ - ISACencLB_obj->frame_nb = 0; - } - return err; - } - iterCntr++; - } - - /* If 60 ms frame-size and just processed the first 30 ms, */ - /* go back to main function to buffer the other 30 ms speech frame. */ - if (frame_mode == 1) { - if (ISACencLB_obj->frame_nb == 0) { - ISACencLB_obj->frame_nb = 1; - return 0; - } else if (ISACencLB_obj->frame_nb == 1) { - ISACencLB_obj->frame_nb = 0; - /* Also update the frame-length for next packet, - in Adaptive mode only. */ - if (codingMode == 0 && (ISACencLB_obj->enforceFrameSize == 0)) { - ISACencLB_obj->new_framelength = - WebRtcIsac_GetNewFrameLength(ISACencLB_obj->bottleneck, - ISACencLB_obj->current_framesamples); - } - } - } else { - ISACencLB_obj->frame_nb = 0; - } - - /* Complete arithmetic coding. */ - stream_length = WebRtcIsac_EncTerminate(&ISACencLB_obj->bitstr_obj); - return stream_length; -} - - - -static int LimitPayloadUb(ISACUBEncStruct* ISACencUB_obj, - uint16_t payloadLimitBytes, - double bytesLeftSpecCoding, - transcode_obj* transcodingParam, - int16_t* fre, int16_t* fim, - double* lpcGains, enum ISACBand band, int status) { - - int iterCntr = 0; - int k; - double bytesSpecCoderUsed; - double transcodeScale; - const int16_t kAveragePitchGain = 0.0; - - do { - if (iterCntr >= MAX_PAYLOAD_LIMIT_ITERATION) { - /* We were not able to limit the payload size. */ - return -ISAC_PAYLOAD_LARGER_THAN_LIMIT; - } - - if (status == -ISAC_DISALLOWED_BITSTREAM_LENGTH) { - bytesSpecCoderUsed = STREAM_SIZE_MAX; - /* Being conservative. */ - transcodeScale = bytesLeftSpecCoding / bytesSpecCoderUsed * 0.5; - } else { - bytesSpecCoderUsed = ISACencUB_obj->bitstr_obj.stream_index - - transcodingParam->stream_index; - transcodeScale = bytesLeftSpecCoding / bytesSpecCoderUsed; - } - - /* To be safe, we reduce the scale depending on the - number of iterations. */ - transcodeScale *= (1.0 - (0.9 * (double)iterCntr / - (double)MAX_PAYLOAD_LIMIT_ITERATION)); - - /* Scale the LPC Gains. */ - if (band == kIsacUpperBand16) { - /* Two sets of coefficients if 16 kHz. */ - for (k = 0; k < SUBFRAMES; k++) { - transcodingParam->loFiltGain[k] *= transcodeScale; - transcodingParam->hiFiltGain[k] *= transcodeScale; - } - } else { - /* One sets of coefficients if 12 kHz. */ - for (k = 0; k < SUBFRAMES; k++) { - transcodingParam->loFiltGain[k] *= transcodeScale; - } - } - - /* Scale DFT coefficients. */ - for (k = 0; k < FRAMESAMPLES_HALF; k++) { - fre[k] = (int16_t)(fre[k] * transcodeScale + 0.5); - fim[k] = (int16_t)(fim[k] * transcodeScale + 0.5); - } - /* Store FFT coefficients for multiple encoding. */ - memcpy(ISACencUB_obj->SaveEnc_obj.realFFT, fre, - sizeof(ISACencUB_obj->SaveEnc_obj.realFFT)); - memcpy(ISACencUB_obj->SaveEnc_obj.imagFFT, fim, - sizeof(ISACencUB_obj->SaveEnc_obj.imagFFT)); - - /* Store the state of arithmetic coder before coding LPC gains */ - ISACencUB_obj->bitstr_obj.W_upper = transcodingParam->W_upper; - ISACencUB_obj->bitstr_obj.stream_index = transcodingParam->stream_index; - ISACencUB_obj->bitstr_obj.streamval = transcodingParam->streamval; - ISACencUB_obj->bitstr_obj.stream[transcodingParam->stream_index - 2] = - transcodingParam->stream[0]; - ISACencUB_obj->bitstr_obj.stream[transcodingParam->stream_index - 1] = - transcodingParam->stream[1]; - ISACencUB_obj->bitstr_obj.stream[transcodingParam->stream_index] = - transcodingParam->stream[2]; - - /* Store the gains for multiple encoding. */ - memcpy(ISACencUB_obj->SaveEnc_obj.lpcGain, lpcGains, - SUBFRAMES * sizeof(double)); - /* Entropy Code lpc-gains, indices are stored for a later use.*/ - WebRtcIsac_EncodeLpcGainUb(transcodingParam->loFiltGain, - &ISACencUB_obj->bitstr_obj, - ISACencUB_obj->SaveEnc_obj.lpcGainIndex); - - /* If 16kHz should do one more set. */ - if (band == kIsacUpperBand16) { - /* Store the gains for multiple encoding. */ - memcpy(&ISACencUB_obj->SaveEnc_obj.lpcGain[SUBFRAMES], - &lpcGains[SUBFRAMES], SUBFRAMES * sizeof(double)); - /* Entropy Code lpc-gains, indices are stored for a later use.*/ - WebRtcIsac_EncodeLpcGainUb( - transcodingParam->hiFiltGain, &ISACencUB_obj->bitstr_obj, - &ISACencUB_obj->SaveEnc_obj.lpcGainIndex[SUBFRAMES]); - } - - /* Update the number of bytes left for encoding the spectrum. */ - bytesLeftSpecCoding = payloadLimitBytes - - ISACencUB_obj->bitstr_obj.stream_index; - - /* Save the bit-stream object at this point for FEC. */ - memcpy(&ISACencUB_obj->SaveEnc_obj.bitStreamObj, - &ISACencUB_obj->bitstr_obj, sizeof(Bitstr)); - - /* Encode the spectrum. */ - status = WebRtcIsac_EncodeSpec(fre, fim, kAveragePitchGain, - band, &ISACencUB_obj->bitstr_obj); - if ((status < 0) && (status != -ISAC_DISALLOWED_BITSTREAM_LENGTH)) { - /* There has been an error but it was not too large payload - (we can cure too large payload). */ - return status; - } - iterCntr++; - } while ((ISACencUB_obj->bitstr_obj.stream_index > payloadLimitBytes) || - (status == -ISAC_DISALLOWED_BITSTREAM_LENGTH)); - return 0; -} - -int WebRtcIsac_EncodeUb16(const TransformTables* transform_tables, - float* in, ISACUBEncStruct* ISACencUB_obj, - int32_t jitterInfo) { - int err; - int k; - - double lpcVecs[UB_LPC_ORDER * UB16_LPC_VEC_PER_FRAME]; - double percepFilterParams[(1 + UB_LPC_ORDER) * (SUBFRAMES << 1) + - (1 + UB_LPC_ORDER)]; - - double LP_lookahead[FRAMESAMPLES]; - int16_t fre[FRAMESAMPLES_HALF]; /* Q7 */ - int16_t fim[FRAMESAMPLES_HALF]; /* Q7 */ - - int status = 0; - - double varscale[2]; - double corr[SUBFRAMES << 1][UB_LPC_ORDER + 1]; - double lpcGains[SUBFRAMES << 1]; - transcode_obj transcodingParam; - uint16_t payloadLimitBytes; - double s2nr; - const int16_t kAveragePitchGain = 0.0; - int bytesLeftSpecCoding; - - /* Buffer speech samples (by 10ms packet) until the frame-length is */ - /* reached (30 ms). */ - /*********************************************************************/ - - /* fill the buffer with 10ms input data */ - memcpy(&ISACencUB_obj->data_buffer_float[ISACencUB_obj->buffer_index], in, - FRAMESAMPLES_10ms * sizeof(float)); - - /* If buffer size is not equal to current frame-size, and end of file is - * not reached yet, we don't do encoding unless we have the whole frame. */ - if (ISACencUB_obj->buffer_index + FRAMESAMPLES_10ms < FRAMESAMPLES) { - ISACencUB_obj->buffer_index += FRAMESAMPLES_10ms; - return 0; - } - - /* End of buffer function. */ - /**************************/ - - /* Encoding */ - /************/ - - /* Reset bit-stream */ - WebRtcIsac_ResetBitstream(&(ISACencUB_obj->bitstr_obj)); - - /* Encoding of bandwidth information. */ - WebRtcIsac_EncodeJitterInfo(jitterInfo, &ISACencUB_obj->bitstr_obj); - - status = WebRtcIsac_EncodeBandwidth(isac16kHz, &ISACencUB_obj->bitstr_obj); - if (status < 0) { - return status; - } - - s2nr = WebRtcIsac_GetSnr(ISACencUB_obj->bottleneck, FRAMESAMPLES); - - memcpy(lpcVecs, ISACencUB_obj->lastLPCVec, UB_LPC_ORDER * sizeof(double)); - - for (k = 0; k < FRAMESAMPLES; k++) { - LP_lookahead[k] = ISACencUB_obj->data_buffer_float[UB_LOOKAHEAD + k]; - } - - /* Find coefficients for perceptual pre-filters. */ - WebRtcIsac_GetLpcCoefUb(LP_lookahead, &ISACencUB_obj->maskfiltstr_obj, - &lpcVecs[UB_LPC_ORDER], corr, varscale, isac16kHz); - - memcpy(ISACencUB_obj->lastLPCVec, - &lpcVecs[(UB16_LPC_VEC_PER_FRAME - 1) * (UB_LPC_ORDER)], - sizeof(double) * UB_LPC_ORDER); - - /* Code LPC model and shape - gains not quantized yet. */ - WebRtcIsac_EncodeLpcUB(lpcVecs, &ISACencUB_obj->bitstr_obj, - percepFilterParams, isac16kHz, - &ISACencUB_obj->SaveEnc_obj); - - /* the first set of lpc parameters are from the last sub-frame of - * the previous frame. so we don't care about them. */ - WebRtcIsac_GetLpcGain(s2nr, &percepFilterParams[UB_LPC_ORDER + 1], - (SUBFRAMES << 1), lpcGains, corr, varscale); - - /* Store the state of arithmetic coder before coding LPC gains */ - transcodingParam.stream_index = ISACencUB_obj->bitstr_obj.stream_index; - transcodingParam.W_upper = ISACencUB_obj->bitstr_obj.W_upper; - transcodingParam.streamval = ISACencUB_obj->bitstr_obj.streamval; - transcodingParam.stream[0] = - ISACencUB_obj->bitstr_obj.stream[ISACencUB_obj->bitstr_obj.stream_index - - 2]; - transcodingParam.stream[1] = - ISACencUB_obj->bitstr_obj.stream[ISACencUB_obj->bitstr_obj.stream_index - - 1]; - transcodingParam.stream[2] = - ISACencUB_obj->bitstr_obj.stream[ISACencUB_obj->bitstr_obj.stream_index]; - - /* Store LPC Gains before encoding them. */ - for (k = 0; k < SUBFRAMES; k++) { - transcodingParam.loFiltGain[k] = lpcGains[k]; - transcodingParam.hiFiltGain[k] = lpcGains[SUBFRAMES + k]; - } - - /* Store the gains for multiple encoding. */ - memcpy(ISACencUB_obj->SaveEnc_obj.lpcGain, lpcGains, - (SUBFRAMES << 1) * sizeof(double)); - - WebRtcIsac_EncodeLpcGainUb(lpcGains, &ISACencUB_obj->bitstr_obj, - ISACencUB_obj->SaveEnc_obj.lpcGainIndex); - WebRtcIsac_EncodeLpcGainUb( - &lpcGains[SUBFRAMES], &ISACencUB_obj->bitstr_obj, - &ISACencUB_obj->SaveEnc_obj.lpcGainIndex[SUBFRAMES]); - - /* Get the correct value for the payload limit and calculate the number of - bytes left for coding the spectrum. It is a 30ms frame - Subract 3 because termination process may add 3 bytes */ - payloadLimitBytes = ISACencUB_obj->maxPayloadSizeBytes - - ISACencUB_obj->numBytesUsed - 3; - bytesLeftSpecCoding = payloadLimitBytes - - ISACencUB_obj->bitstr_obj.stream_index; - - for (k = 0; k < (SUBFRAMES << 1); k++) { - percepFilterParams[k * (UB_LPC_ORDER + 1) + (UB_LPC_ORDER + 1)] = - lpcGains[k]; - } - - /* LPC filtering (using normalized lattice filter), */ - /* first half-frame. */ - WebRtcIsac_NormLatticeFilterMa(UB_LPC_ORDER, - ISACencUB_obj->maskfiltstr_obj.PreStateLoF, - ISACencUB_obj->maskfiltstr_obj.PreStateLoG, - &ISACencUB_obj->data_buffer_float[0], - &percepFilterParams[UB_LPC_ORDER + 1], - &LP_lookahead[0]); - - /* Second half-frame filtering. */ - WebRtcIsac_NormLatticeFilterMa( - UB_LPC_ORDER, ISACencUB_obj->maskfiltstr_obj.PreStateLoF, - ISACencUB_obj->maskfiltstr_obj.PreStateLoG, - &ISACencUB_obj->data_buffer_float[FRAMESAMPLES_HALF], - &percepFilterParams[(UB_LPC_ORDER + 1) + SUBFRAMES * (UB_LPC_ORDER + 1)], - &LP_lookahead[FRAMESAMPLES_HALF]); - - WebRtcIsac_Time2Spec(transform_tables, - &LP_lookahead[0], &LP_lookahead[FRAMESAMPLES_HALF], - fre, fim, &ISACencUB_obj->fftstr_obj); - - /* Store FFT coefficients for multiple encoding. */ - memcpy(ISACencUB_obj->SaveEnc_obj.realFFT, fre, sizeof(fre)); - memcpy(ISACencUB_obj->SaveEnc_obj.imagFFT, fim, sizeof(fim)); - - /* Prepare the audio buffer for the next packet - * move the last 3 ms to the beginning of the buffer. */ - memcpy(ISACencUB_obj->data_buffer_float, - &ISACencUB_obj->data_buffer_float[FRAMESAMPLES], - LB_TOTAL_DELAY_SAMPLES * sizeof(float)); - /* start writing with 3 ms delay to compensate for the delay - * of the lower-band. */ - ISACencUB_obj->buffer_index = LB_TOTAL_DELAY_SAMPLES; - - /* Save the bit-stream object at this point for FEC. */ - memcpy(&ISACencUB_obj->SaveEnc_obj.bitStreamObj, &ISACencUB_obj->bitstr_obj, - sizeof(Bitstr)); - - /* Qantization and lossless coding */ - /* Note that there is no pitch-gain for this band so kAveragePitchGain = 0 - * is passed to the function. In fact, the function ignores the 3rd parameter - * for this band. */ - err = WebRtcIsac_EncodeSpec(fre, fim, kAveragePitchGain, kIsacUpperBand16, - &ISACencUB_obj->bitstr_obj); - if ((err < 0) && (err != -ISAC_DISALLOWED_BITSTREAM_LENGTH)) { - return err; - } - - if ((ISACencUB_obj->bitstr_obj.stream_index > payloadLimitBytes) || - (err == -ISAC_DISALLOWED_BITSTREAM_LENGTH)) { - err = LimitPayloadUb(ISACencUB_obj, payloadLimitBytes, bytesLeftSpecCoding, - &transcodingParam, fre, fim, lpcGains, - kIsacUpperBand16, err); - } - if (err < 0) { - return err; - } - /* Complete arithmetic coding. */ - return WebRtcIsac_EncTerminate(&ISACencUB_obj->bitstr_obj); -} - - -int WebRtcIsac_EncodeUb12(const TransformTables* transform_tables, - float* in, ISACUBEncStruct* ISACencUB_obj, - int32_t jitterInfo) { - int err; - int k; - - double lpcVecs[UB_LPC_ORDER * UB_LPC_VEC_PER_FRAME]; - - double percepFilterParams[(1 + UB_LPC_ORDER) * SUBFRAMES]; - float LP[FRAMESAMPLES_HALF]; - float HP[FRAMESAMPLES_HALF]; - - double LP_lookahead[FRAMESAMPLES_HALF]; - double HP_lookahead[FRAMESAMPLES_HALF]; - double LPw[FRAMESAMPLES_HALF]; - - double HPw[FRAMESAMPLES_HALF]; - int16_t fre[FRAMESAMPLES_HALF]; /* Q7 */ - int16_t fim[FRAMESAMPLES_HALF]; /* Q7 */ - - int status = 0; - - double varscale[1]; - - double corr[UB_LPC_GAIN_DIM][UB_LPC_ORDER + 1]; - double lpcGains[SUBFRAMES]; - transcode_obj transcodingParam; - uint16_t payloadLimitBytes; - double s2nr; - const int16_t kAveragePitchGain = 0.0; - double bytesLeftSpecCoding; - - /* Buffer speech samples (by 10ms packet) until the framelength is */ - /* reached (30 ms). */ - /********************************************************************/ - - /* Fill the buffer with 10ms input data. */ - memcpy(&ISACencUB_obj->data_buffer_float[ISACencUB_obj->buffer_index], in, - FRAMESAMPLES_10ms * sizeof(float)); - - /* if buffer-size is not equal to current frame-size then increase the - index and return. We do the encoding when we have enough audio. */ - if (ISACencUB_obj->buffer_index + FRAMESAMPLES_10ms < FRAMESAMPLES) { - ISACencUB_obj->buffer_index += FRAMESAMPLES_10ms; - return 0; - } - /* If buffer reached the right size, reset index and continue - with encoding the frame */ - ISACencUB_obj->buffer_index = 0; - - /* End of buffer function */ - /**************************/ - - /* Encoding */ - /************/ - - /* Reset bit-stream. */ - WebRtcIsac_ResetBitstream(&(ISACencUB_obj->bitstr_obj)); - - /* Encoding bandwidth information. */ - WebRtcIsac_EncodeJitterInfo(jitterInfo, &ISACencUB_obj->bitstr_obj); - status = WebRtcIsac_EncodeBandwidth(isac12kHz, &ISACencUB_obj->bitstr_obj); - if (status < 0) { - return status; - } - - s2nr = WebRtcIsac_GetSnr(ISACencUB_obj->bottleneck, FRAMESAMPLES); - - /* Split signal in two bands. */ - WebRtcIsac_SplitAndFilterFloat(ISACencUB_obj->data_buffer_float, HP, LP, - HP_lookahead, LP_lookahead, - &ISACencUB_obj->prefiltbankstr_obj); - - /* Find coefficients for perceptual pre-filters. */ - WebRtcIsac_GetLpcCoefUb(LP_lookahead, &ISACencUB_obj->maskfiltstr_obj, - lpcVecs, corr, varscale, isac12kHz); - - /* Code LPC model and shape - gains not quantized yet. */ - WebRtcIsac_EncodeLpcUB(lpcVecs, &ISACencUB_obj->bitstr_obj, - percepFilterParams, isac12kHz, - &ISACencUB_obj->SaveEnc_obj); - - WebRtcIsac_GetLpcGain(s2nr, percepFilterParams, SUBFRAMES, lpcGains, corr, - varscale); - - /* Store the state of arithmetic coder before coding LPC gains. */ - transcodingParam.W_upper = ISACencUB_obj->bitstr_obj.W_upper; - transcodingParam.stream_index = ISACencUB_obj->bitstr_obj.stream_index; - transcodingParam.streamval = ISACencUB_obj->bitstr_obj.streamval; - transcodingParam.stream[0] = - ISACencUB_obj->bitstr_obj.stream[ISACencUB_obj->bitstr_obj.stream_index - - 2]; - transcodingParam.stream[1] = - ISACencUB_obj->bitstr_obj.stream[ISACencUB_obj->bitstr_obj.stream_index - - 1]; - transcodingParam.stream[2] = - ISACencUB_obj->bitstr_obj.stream[ISACencUB_obj->bitstr_obj.stream_index]; - - /* Store LPC Gains before encoding them. */ - for (k = 0; k < SUBFRAMES; k++) { - transcodingParam.loFiltGain[k] = lpcGains[k]; - } - - /* Store the gains for multiple encoding. */ - memcpy(ISACencUB_obj->SaveEnc_obj.lpcGain, lpcGains, SUBFRAMES * - sizeof(double)); - - WebRtcIsac_EncodeLpcGainUb(lpcGains, &ISACencUB_obj->bitstr_obj, - ISACencUB_obj->SaveEnc_obj.lpcGainIndex); - - for (k = 0; k < SUBFRAMES; k++) { - percepFilterParams[k * (UB_LPC_ORDER + 1)] = lpcGains[k]; - } - - /* perceptual pre-filtering (using normalized lattice filter) */ - /* low-band filtering */ - WebRtcIsac_NormLatticeFilterMa(UB_LPC_ORDER, - ISACencUB_obj->maskfiltstr_obj.PreStateLoF, - ISACencUB_obj->maskfiltstr_obj.PreStateLoG, LP, - percepFilterParams, LPw); - - /* Get the correct value for the payload limit and calculate the number - of bytes left for coding the spectrum. It is a 30ms frame Subract 3 - because termination process may add 3 bytes */ - payloadLimitBytes = ISACencUB_obj->maxPayloadSizeBytes - - ISACencUB_obj->numBytesUsed - 3; - bytesLeftSpecCoding = payloadLimitBytes - - ISACencUB_obj->bitstr_obj.stream_index; - - memset(HPw, 0, sizeof(HPw)); - - /* Transform */ - WebRtcIsac_Time2Spec(transform_tables, - LPw, HPw, fre, fim, &ISACencUB_obj->fftstr_obj); - - /* Store FFT coefficients for multiple encoding. */ - memcpy(ISACencUB_obj->SaveEnc_obj.realFFT, fre, - sizeof(ISACencUB_obj->SaveEnc_obj.realFFT)); - memcpy(ISACencUB_obj->SaveEnc_obj.imagFFT, fim, - sizeof(ISACencUB_obj->SaveEnc_obj.imagFFT)); - - /* Save the bit-stream object at this point for FEC. */ - memcpy(&ISACencUB_obj->SaveEnc_obj.bitStreamObj, - &ISACencUB_obj->bitstr_obj, sizeof(Bitstr)); - - /* Quantization and loss-less coding */ - /* The 4th parameter to this function is pitch-gain, which is only used - * when encoding 0-8 kHz band, and irrelevant in this function, therefore, - * we insert zero here. */ - err = WebRtcIsac_EncodeSpec(fre, fim, kAveragePitchGain, kIsacUpperBand12, - &ISACencUB_obj->bitstr_obj); - if ((err < 0) && (err != -ISAC_DISALLOWED_BITSTREAM_LENGTH)) { - /* There has been an error but it was not too large - payload (we can cure too large payload) */ - return err; - } - - if ((ISACencUB_obj->bitstr_obj.stream_index > payloadLimitBytes) || - (err == -ISAC_DISALLOWED_BITSTREAM_LENGTH)) { - err = LimitPayloadUb(ISACencUB_obj, payloadLimitBytes, bytesLeftSpecCoding, - &transcodingParam, fre, fim, lpcGains, - kIsacUpperBand12, err); - } - if (err < 0) { - return err; - } - /* Complete arithmetic coding. */ - return WebRtcIsac_EncTerminate(&ISACencUB_obj->bitstr_obj); -} - - - - - - -/* This function is used to create a new bit-stream with new BWE. - The same data as previously encoded with the function WebRtcIsac_Encoder(). - The data needed is taken from the structure, where it was stored - when calling the encoder. */ - -int WebRtcIsac_EncodeStoredDataLb(const IsacSaveEncoderData* ISACSavedEnc_obj, - Bitstr* ISACBitStr_obj, int BWnumber, - float scale) { - int ii; - int status; - int BWno = BWnumber; - - const uint16_t* WebRtcIsac_kQPitchGainCdf_ptr[1]; - const uint16_t** cdf; - - double tmpLPCcoeffs_lo[(ORDERLO + 1)*SUBFRAMES * 2]; - double tmpLPCcoeffs_hi[(ORDERHI + 1)*SUBFRAMES * 2]; - int tmpLPCindex_g[12 * 2]; - int16_t tmp_fre[FRAMESAMPLES], tmp_fim[FRAMESAMPLES]; - const int kModel = 0; - - /* Sanity Check - possible values for BWnumber is 0 - 23. */ - if ((BWnumber < 0) || (BWnumber > 23)) { - return -ISAC_RANGE_ERROR_BW_ESTIMATOR; - } - - /* Reset bit-stream. */ - WebRtcIsac_ResetBitstream(ISACBitStr_obj); - - /* Encode frame length */ - status = WebRtcIsac_EncodeFrameLen(ISACSavedEnc_obj->framelength, - ISACBitStr_obj); - if (status < 0) { - /* Wrong frame size. */ - return status; - } - - /* Transcoding */ - if ((scale > 0.0) && (scale < 1.0)) { - /* Compensate LPC gain. */ - for (ii = 0; - ii < ((ORDERLO + 1)* SUBFRAMES * (1 + ISACSavedEnc_obj->startIdx)); - ii++) { - tmpLPCcoeffs_lo[ii] = scale * ISACSavedEnc_obj->LPCcoeffs_lo[ii]; - } - for (ii = 0; - ii < ((ORDERHI + 1) * SUBFRAMES * (1 + ISACSavedEnc_obj->startIdx)); - ii++) { - tmpLPCcoeffs_hi[ii] = scale * ISACSavedEnc_obj->LPCcoeffs_hi[ii]; - } - /* Scale DFT. */ - for (ii = 0; - ii < (FRAMESAMPLES_HALF * (1 + ISACSavedEnc_obj->startIdx)); - ii++) { - tmp_fre[ii] = (int16_t)((scale) * (float)ISACSavedEnc_obj->fre[ii]); - tmp_fim[ii] = (int16_t)((scale) * (float)ISACSavedEnc_obj->fim[ii]); - } - } else { - for (ii = 0; - ii < (KLT_ORDER_GAIN * (1 + ISACSavedEnc_obj->startIdx)); - ii++) { - tmpLPCindex_g[ii] = ISACSavedEnc_obj->LPCindex_g[ii]; - } - for (ii = 0; - ii < (FRAMESAMPLES_HALF * (1 + ISACSavedEnc_obj->startIdx)); - ii++) { - tmp_fre[ii] = ISACSavedEnc_obj->fre[ii]; - tmp_fim[ii] = ISACSavedEnc_obj->fim[ii]; - } - } - - /* Encode bandwidth estimate. */ - WebRtcIsac_EncodeReceiveBw(&BWno, ISACBitStr_obj); - - /* Loop over number of 30 msec */ - for (ii = 0; ii <= ISACSavedEnc_obj->startIdx; ii++) { - /* Encode pitch gains. */ - *WebRtcIsac_kQPitchGainCdf_ptr = WebRtcIsac_kQPitchGainCdf; - WebRtcIsac_EncHistMulti(ISACBitStr_obj, - &ISACSavedEnc_obj->pitchGain_index[ii], - WebRtcIsac_kQPitchGainCdf_ptr, 1); - - /* Entropy coding of quantization pitch lags */ - /* Voicing classification. */ - if (ISACSavedEnc_obj->meanGain[ii] < 0.2) { - cdf = WebRtcIsac_kQPitchLagCdfPtrLo; - } else if (ISACSavedEnc_obj->meanGain[ii] < 0.4) { - cdf = WebRtcIsac_kQPitchLagCdfPtrMid; - } else { - cdf = WebRtcIsac_kQPitchLagCdfPtrHi; - } - WebRtcIsac_EncHistMulti(ISACBitStr_obj, - &ISACSavedEnc_obj->pitchIndex[PITCH_SUBFRAMES * ii], - cdf, PITCH_SUBFRAMES); - - /* LPC */ - /* Only one model exists. The entropy coding is done only for backward - * compatibility. */ - WebRtcIsac_EncHistMulti(ISACBitStr_obj, &kModel, - WebRtcIsac_kQKltModelCdfPtr, 1); - /* Entropy coding of quantization indices - LPC shape only. */ - WebRtcIsac_EncHistMulti(ISACBitStr_obj, - &ISACSavedEnc_obj->LPCindex_s[KLT_ORDER_SHAPE * ii], - WebRtcIsac_kQKltCdfPtrShape, - KLT_ORDER_SHAPE); - - /* If transcoding, get new LPC gain indices */ - if (scale < 1.0) { - WebRtcIsac_TranscodeLPCCoef( - &tmpLPCcoeffs_lo[(ORDERLO + 1) * SUBFRAMES * ii], - &tmpLPCcoeffs_hi[(ORDERHI + 1)*SUBFRAMES * ii], - &tmpLPCindex_g[KLT_ORDER_GAIN * ii]); - } - - /* Entropy coding of quantization indices - LPC gain. */ - WebRtcIsac_EncHistMulti(ISACBitStr_obj, &tmpLPCindex_g[KLT_ORDER_GAIN * ii], - WebRtcIsac_kQKltCdfPtrGain, KLT_ORDER_GAIN); - - /* Quantization and loss-less coding. */ - status = WebRtcIsac_EncodeSpec(&tmp_fre[ii * FRAMESAMPLES_HALF], - &tmp_fim[ii * FRAMESAMPLES_HALF], - ISACSavedEnc_obj->AvgPitchGain[ii], - kIsacLowerBand, ISACBitStr_obj); - if (status < 0) { - return status; - } - } - /* Complete arithmetic coding. */ - return WebRtcIsac_EncTerminate(ISACBitStr_obj); -} - - -int WebRtcIsac_EncodeStoredDataUb( - const ISACUBSaveEncDataStruct* ISACSavedEnc_obj, - Bitstr* bitStream, - int32_t jitterInfo, - float scale, - enum ISACBandwidth bandwidth) { - int n; - int err; - double lpcGain[SUBFRAMES]; - int16_t realFFT[FRAMESAMPLES_HALF]; - int16_t imagFFT[FRAMESAMPLES_HALF]; - const uint16_t** shape_cdf; - int shape_len; - const int16_t kAveragePitchGain = 0.0; - enum ISACBand band; - /* Reset bitstream. */ - WebRtcIsac_ResetBitstream(bitStream); - - /* Encode jitter index. */ - WebRtcIsac_EncodeJitterInfo(jitterInfo, bitStream); - - err = WebRtcIsac_EncodeBandwidth(bandwidth, bitStream); - if (err < 0) { - return err; - } - - /* Encode LPC-shape. */ - if (bandwidth == isac12kHz) { - shape_cdf = WebRtcIsac_kLpcShapeCdfMatUb12; - shape_len = UB_LPC_ORDER * UB_LPC_VEC_PER_FRAME; - band = kIsacUpperBand12; - } else { - shape_cdf = WebRtcIsac_kLpcShapeCdfMatUb16; - shape_len = UB_LPC_ORDER * UB16_LPC_VEC_PER_FRAME; - band = kIsacUpperBand16; - } - WebRtcIsac_EncHistMulti(bitStream, ISACSavedEnc_obj->indexLPCShape, - shape_cdf, shape_len); - - if ((scale <= 0.0) || (scale >= 1.0)) { - /* We only consider scales between zero and one. */ - WebRtcIsac_EncHistMulti(bitStream, ISACSavedEnc_obj->lpcGainIndex, - WebRtcIsac_kLpcGainCdfMat, UB_LPC_GAIN_DIM); - if (bandwidth == isac16kHz) { - /* Store gain indices of the second half. */ - WebRtcIsac_EncHistMulti(bitStream, - &ISACSavedEnc_obj->lpcGainIndex[SUBFRAMES], - WebRtcIsac_kLpcGainCdfMat, UB_LPC_GAIN_DIM); - } - /* Store FFT coefficients. */ - err = WebRtcIsac_EncodeSpec(ISACSavedEnc_obj->realFFT, - ISACSavedEnc_obj->imagFFT, kAveragePitchGain, - band, bitStream); - } else { - /* Scale LPC gain and FFT coefficients. */ - for (n = 0; n < SUBFRAMES; n++) { - lpcGain[n] = scale * ISACSavedEnc_obj->lpcGain[n]; - } - /* Store LPC gains. */ - WebRtcIsac_StoreLpcGainUb(lpcGain, bitStream); - - if (bandwidth == isac16kHz) { - /* Scale and code the gains of the second half of the frame, if 16kHz. */ - for (n = 0; n < SUBFRAMES; n++) { - lpcGain[n] = scale * ISACSavedEnc_obj->lpcGain[n + SUBFRAMES]; - } - WebRtcIsac_StoreLpcGainUb(lpcGain, bitStream); - } - - for (n = 0; n < FRAMESAMPLES_HALF; n++) { - realFFT[n] = (int16_t)(scale * (float)ISACSavedEnc_obj->realFFT[n] + - 0.5f); - imagFFT[n] = (int16_t)(scale * (float)ISACSavedEnc_obj->imagFFT[n] + - 0.5f); - } - /* Store FFT coefficients. */ - err = WebRtcIsac_EncodeSpec(realFFT, imagFFT, kAveragePitchGain, - band, bitStream); - } - if (err < 0) { - /* Error happened while encoding FFT coefficients. */ - return err; - } - - /* Complete arithmetic coding. */ - return WebRtcIsac_EncTerminate(bitStream); -} - -int16_t WebRtcIsac_GetRedPayloadUb( - const ISACUBSaveEncDataStruct* ISACSavedEncObj, - Bitstr* bitStreamObj, - enum ISACBandwidth bandwidth) { - int n; - int16_t status; - int16_t realFFT[FRAMESAMPLES_HALF]; - int16_t imagFFT[FRAMESAMPLES_HALF]; - enum ISACBand band; - const int16_t kAveragePitchGain = 0.0; - /* Store bit-stream object. */ - memcpy(bitStreamObj, &ISACSavedEncObj->bitStreamObj, sizeof(Bitstr)); - - /* Scale FFT coefficients. */ - for (n = 0; n < FRAMESAMPLES_HALF; n++) { - realFFT[n] = (int16_t)((float)ISACSavedEncObj->realFFT[n] * - RCU_TRANSCODING_SCALE_UB + 0.5); - imagFFT[n] = (int16_t)((float)ISACSavedEncObj->imagFFT[n] * - RCU_TRANSCODING_SCALE_UB + 0.5); - } - - band = (bandwidth == isac12kHz) ? kIsacUpperBand12 : kIsacUpperBand16; - status = WebRtcIsac_EncodeSpec(realFFT, imagFFT, kAveragePitchGain, band, - bitStreamObj); - if (status < 0) { - return status; - } else { - /* Terminate entropy coding */ - return WebRtcIsac_EncTerminate(bitStreamObj); - } -} diff --git a/modules/audio_coding/codecs/isac/main/source/encode_lpc_swb.c b/modules/audio_coding/codecs/isac/main/source/encode_lpc_swb.c deleted file mode 100644 index 7b02e64a01..0000000000 --- a/modules/audio_coding/codecs/isac/main/source/encode_lpc_swb.c +++ /dev/null @@ -1,706 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/* - * code_LPC_UB.c - * - * This file contains definition of functions used to - * encode LPC parameters (Shape & gain) of the upper band. - * - */ - -#include -#include -#include - -#include "modules/audio_coding/codecs/isac/main/source/encode_lpc_swb.h" -#include "modules/audio_coding/codecs/isac/main/source/lpc_gain_swb_tables.h" -#include "modules/audio_coding/codecs/isac/main/source/lpc_shape_swb12_tables.h" -#include "modules/audio_coding/codecs/isac/main/source/lpc_shape_swb16_tables.h" -#include "modules/audio_coding/codecs/isac/main/source/settings.h" - -/****************************************************************************** - * WebRtcIsac_RemoveLarMean() - * - * Remove the means from LAR coefficients. - * - * Input: - * -lar : pointer to lar vectors. LAR vectors are - * concatenated. - * -bandwidth : indicates if the given LAR vectors belong - * to SWB-12kHz or SWB-16kHz. - * - * Output: - * -lar : pointer to mean-removed LAR:s. - * - * - */ -int16_t -WebRtcIsac_RemoveLarMean( - double* lar, - int16_t bandwidth) -{ - int16_t coeffCntr; - int16_t vecCntr; - int16_t numVec; - const double* meanLAR; - switch(bandwidth) - { - case isac12kHz: - { - numVec = UB_LPC_VEC_PER_FRAME; - meanLAR = WebRtcIsac_kMeanLarUb12; - break; - } - case isac16kHz: - { - numVec = UB16_LPC_VEC_PER_FRAME; - meanLAR = WebRtcIsac_kMeanLarUb16; - break; - } - default: - return -1; - } - - for(vecCntr = 0; vecCntr < numVec; vecCntr++) - { - for(coeffCntr = 0; coeffCntr < UB_LPC_ORDER; coeffCntr++) - { - // REMOVE MEAN - *lar++ -= meanLAR[coeffCntr]; - } - } - return 0; -} - -/****************************************************************************** - * WebRtcIsac_DecorrelateIntraVec() - * - * Remove the correlation amonge the components of LAR vectors. If LAR vectors - * of one frame are put in a matrix where each column is a LAR vector of a - * sub-frame, then this is equivalent to multiplying the LAR matrix with - * a decorrelting mtrix from left. - * - * Input: - * -inLar : pointer to mean-removed LAR vecrtors. - * -bandwidth : indicates if the given LAR vectors belong - * to SWB-12kHz or SWB-16kHz. - * - * Output: - * -out : decorrelated LAR vectors. - */ -int16_t -WebRtcIsac_DecorrelateIntraVec( - const double* data, - double* out, - int16_t bandwidth) -{ - const double* ptrData; - const double* ptrRow; - int16_t rowCntr; - int16_t colCntr; - int16_t larVecCntr; - int16_t numVec; - const double* decorrMat; - switch(bandwidth) - { - case isac12kHz: - { - decorrMat = &WebRtcIsac_kIntraVecDecorrMatUb12[0][0]; - numVec = UB_LPC_VEC_PER_FRAME; - break; - } - case isac16kHz: - { - decorrMat = &WebRtcIsac_kIintraVecDecorrMatUb16[0][0]; - numVec = UB16_LPC_VEC_PER_FRAME; - break; - } - default: - return -1; - } - - // - // decorrMat * data - // - // data is assumed to contain 'numVec' of LAR - // vectors (mean removed) each of dimension 'UB_LPC_ORDER' - // concatenated one after the other. - // - - ptrData = data; - for(larVecCntr = 0; larVecCntr < numVec; larVecCntr++) - { - for(rowCntr = 0; rowCntr < UB_LPC_ORDER; rowCntr++) - { - ptrRow = &decorrMat[rowCntr * UB_LPC_ORDER]; - *out = 0; - for(colCntr = 0; colCntr < UB_LPC_ORDER; colCntr++) - { - *out += ptrData[colCntr] * ptrRow[colCntr]; - } - out++; - } - ptrData += UB_LPC_ORDER; - } - return 0; -} - -/****************************************************************************** - * WebRtcIsac_DecorrelateInterVec() - * - * Remover the correlation among mean-removed LAR vectors. If LAR vectors - * of one frame are put in a matrix where each column is a LAR vector of a - * sub-frame, then this is equivalent to multiplying the LAR matrix with - * a decorrelting mtrix from right. - * - * Input: - * -data : pointer to matrix of LAR vectors. The matrix - * is stored column-wise. - * -bandwidth : indicates if the given LAR vectors belong - * to SWB-12kHz or SWB-16kHz. - * - * Output: - * -out : decorrelated LAR vectors. - */ -int16_t -WebRtcIsac_DecorrelateInterVec( - const double* data, - double* out, - int16_t bandwidth) -{ - int16_t coeffCntr; - int16_t rowCntr; - int16_t colCntr; - const double* decorrMat; - int16_t interVecDim; - - switch(bandwidth) - { - case isac12kHz: - { - decorrMat = &WebRtcIsac_kInterVecDecorrMatUb12[0][0]; - interVecDim = UB_LPC_VEC_PER_FRAME; - break; - } - case isac16kHz: - { - decorrMat = &WebRtcIsac_kInterVecDecorrMatUb16[0][0]; - interVecDim = UB16_LPC_VEC_PER_FRAME; - break; - } - default: - return -1; - } - - // - // data * decorrMat - // - // data is of size 'interVecDim' * 'UB_LPC_ORDER' - // That is 'interVecDim' of LAR vectors (mean removed) - // in columns each of dimension 'UB_LPC_ORDER'. - // matrix is stored column-wise. - // - - for(coeffCntr = 0; coeffCntr < UB_LPC_ORDER; coeffCntr++) - { - for(colCntr = 0; colCntr < interVecDim; colCntr++) - { - out[coeffCntr + colCntr * UB_LPC_ORDER] = 0; - for(rowCntr = 0; rowCntr < interVecDim; rowCntr++) - { - out[coeffCntr + colCntr * UB_LPC_ORDER] += - data[coeffCntr + rowCntr * UB_LPC_ORDER] * - decorrMat[rowCntr * interVecDim + colCntr]; - } - } - } - return 0; -} - -/****************************************************************************** - * WebRtcIsac_QuantizeUncorrLar() - * - * Quantize the uncorrelated parameters. - * - * Input: - * -data : uncorrelated LAR vectors. - * -bandwidth : indicates if the given LAR vectors belong - * to SWB-12kHz or SWB-16kHz. - * - * Output: - * -data : quantized version of the input. - * -idx : pointer to quantization indices. - */ -double -WebRtcIsac_QuantizeUncorrLar( - double* data, - int* recIdx, - int16_t bandwidth) -{ - int16_t cntr; - int32_t idx; - int16_t interVecDim; - const double* leftRecPoint; - double quantizationStepSize; - const int16_t* numQuantCell; - switch(bandwidth) - { - case isac12kHz: - { - leftRecPoint = WebRtcIsac_kLpcShapeLeftRecPointUb12; - quantizationStepSize = WebRtcIsac_kLpcShapeQStepSizeUb12; - numQuantCell = WebRtcIsac_kLpcShapeNumRecPointUb12; - interVecDim = UB_LPC_VEC_PER_FRAME; - break; - } - case isac16kHz: - { - leftRecPoint = WebRtcIsac_kLpcShapeLeftRecPointUb16; - quantizationStepSize = WebRtcIsac_kLpcShapeQStepSizeUb16; - numQuantCell = WebRtcIsac_kLpcShapeNumRecPointUb16; - interVecDim = UB16_LPC_VEC_PER_FRAME; - break; - } - default: - return -1; - } - - // - // Quantize the parametrs. - // - for(cntr = 0; cntr < UB_LPC_ORDER * interVecDim; cntr++) - { - idx = (int32_t)floor((*data - leftRecPoint[cntr]) / - quantizationStepSize + 0.5); - if(idx < 0) - { - idx = 0; - } - else if(idx >= numQuantCell[cntr]) - { - idx = numQuantCell[cntr] - 1; - } - - *data++ = leftRecPoint[cntr] + idx * quantizationStepSize; - *recIdx++ = idx; - } - return 0; -} - - -/****************************************************************************** - * WebRtcIsac_DequantizeLpcParam() - * - * Get the quantized value of uncorrelated LARs given the quantization indices. - * - * Input: - * -idx : pointer to quantiztion indices. - * -bandwidth : indicates if the given LAR vectors belong - * to SWB-12kHz or SWB-16kHz. - * - * Output: - * -out : pointer to quantized values. - */ -int16_t -WebRtcIsac_DequantizeLpcParam( - const int* idx, - double* out, - int16_t bandwidth) -{ - int16_t cntr; - int16_t interVecDim; - const double* leftRecPoint; - double quantizationStepSize; - - switch(bandwidth) - { - case isac12kHz: - { - leftRecPoint = WebRtcIsac_kLpcShapeLeftRecPointUb12; - quantizationStepSize = WebRtcIsac_kLpcShapeQStepSizeUb12; - interVecDim = UB_LPC_VEC_PER_FRAME; - break; - } - case isac16kHz: - { - leftRecPoint = WebRtcIsac_kLpcShapeLeftRecPointUb16; - quantizationStepSize = WebRtcIsac_kLpcShapeQStepSizeUb16; - interVecDim = UB16_LPC_VEC_PER_FRAME; - break; - } - default: - return -1; - } - - // - // Dequantize given the quantization indices - // - - for(cntr = 0; cntr < UB_LPC_ORDER * interVecDim; cntr++) - { - *out++ = leftRecPoint[cntr] + *idx++ * quantizationStepSize; - } - return 0; -} - - -/****************************************************************************** - * WebRtcIsac_CorrelateIntraVec() - * - * This is the inverse of WebRtcIsac_DecorrelateIntraVec(). - * - * Input: - * -data : uncorrelated parameters. - * -bandwidth : indicates if the given LAR vectors belong - * to SWB-12kHz or SWB-16kHz. - * - * Output: - * -out : correlated parametrs. - */ -int16_t -WebRtcIsac_CorrelateIntraVec( - const double* data, - double* out, - int16_t bandwidth) -{ - int16_t vecCntr; - int16_t rowCntr; - int16_t colCntr; - int16_t numVec; - const double* ptrData; - const double* intraVecDecorrMat; - - switch(bandwidth) - { - case isac12kHz: - { - numVec = UB_LPC_VEC_PER_FRAME; - intraVecDecorrMat = &WebRtcIsac_kIntraVecDecorrMatUb12[0][0]; - break; - } - case isac16kHz: - { - numVec = UB16_LPC_VEC_PER_FRAME; - intraVecDecorrMat = &WebRtcIsac_kIintraVecDecorrMatUb16[0][0]; - break; - } - default: - return -1; - } - - - ptrData = data; - for(vecCntr = 0; vecCntr < numVec; vecCntr++) - { - for(colCntr = 0; colCntr < UB_LPC_ORDER; colCntr++) - { - *out = 0; - for(rowCntr = 0; rowCntr < UB_LPC_ORDER; rowCntr++) - { - *out += ptrData[rowCntr] * - intraVecDecorrMat[rowCntr * UB_LPC_ORDER + colCntr]; - } - out++; - } - ptrData += UB_LPC_ORDER; - } - return 0; -} - -/****************************************************************************** - * WebRtcIsac_CorrelateInterVec() - * - * This is the inverse of WebRtcIsac_DecorrelateInterVec(). - * - * Input: - * -data - * -bandwidth : indicates if the given LAR vectors belong - * to SWB-12kHz or SWB-16kHz. - * - * Output: - * -out : correlated parametrs. - */ -int16_t -WebRtcIsac_CorrelateInterVec( - const double* data, - double* out, - int16_t bandwidth) -{ - int16_t coeffCntr; - int16_t rowCntr; - int16_t colCntr; - int16_t interVecDim; - double myVec[UB16_LPC_VEC_PER_FRAME] = {0.0}; - const double* interVecDecorrMat; - - switch(bandwidth) - { - case isac12kHz: - { - interVecDim = UB_LPC_VEC_PER_FRAME; - interVecDecorrMat = &WebRtcIsac_kInterVecDecorrMatUb12[0][0]; - break; - } - case isac16kHz: - { - interVecDim = UB16_LPC_VEC_PER_FRAME; - interVecDecorrMat = &WebRtcIsac_kInterVecDecorrMatUb16[0][0]; - break; - } - default: - return -1; - } - - for(coeffCntr = 0; coeffCntr < UB_LPC_ORDER; coeffCntr++) - { - for(rowCntr = 0; rowCntr < interVecDim; rowCntr++) - { - myVec[rowCntr] = 0; - for(colCntr = 0; colCntr < interVecDim; colCntr++) - { - myVec[rowCntr] += data[coeffCntr + colCntr * UB_LPC_ORDER] * //*ptrData * - interVecDecorrMat[rowCntr * interVecDim + colCntr]; - //ptrData += UB_LPC_ORDER; - } - } - - for(rowCntr = 0; rowCntr < interVecDim; rowCntr++) - { - out[coeffCntr + rowCntr * UB_LPC_ORDER] = myVec[rowCntr]; - } - } - return 0; -} - -/****************************************************************************** - * WebRtcIsac_AddLarMean() - * - * This is the inverse of WebRtcIsac_RemoveLarMean() - * - * Input: - * -data : pointer to mean-removed LAR:s. - * -bandwidth : indicates if the given LAR vectors belong - * to SWB-12kHz or SWB-16kHz. - * - * Output: - * -data : pointer to LARs. - */ -int16_t -WebRtcIsac_AddLarMean( - double* data, - int16_t bandwidth) -{ - int16_t coeffCntr; - int16_t vecCntr; - int16_t numVec; - const double* meanLAR; - - switch(bandwidth) - { - case isac12kHz: - { - numVec = UB_LPC_VEC_PER_FRAME; - meanLAR = WebRtcIsac_kMeanLarUb12; - break; - } - case isac16kHz: - { - numVec = UB16_LPC_VEC_PER_FRAME; - meanLAR = WebRtcIsac_kMeanLarUb16; - break; - } - default: - return -1; - } - - for(vecCntr = 0; vecCntr < numVec; vecCntr++) - { - for(coeffCntr = 0; coeffCntr < UB_LPC_ORDER; coeffCntr++) - { - *data++ += meanLAR[coeffCntr]; - } - } - return 0; -} - -/****************************************************************************** - * WebRtcIsac_ToLogDomainRemoveMean() - * - * Transform the LPC gain to log domain then remove the mean value. - * - * Input: - * -lpcGain : pointer to LPC Gain, expecting 6 LPC gains - * - * Output: - * -lpcGain : mean-removed in log domain. - */ -int16_t -WebRtcIsac_ToLogDomainRemoveMean( - double* data) -{ - int16_t coeffCntr; - for(coeffCntr = 0; coeffCntr < UB_LPC_GAIN_DIM; coeffCntr++) - { - data[coeffCntr] = log(data[coeffCntr]) - WebRtcIsac_kMeanLpcGain; - } - return 0; -} - - -/****************************************************************************** - * WebRtcIsac_DecorrelateLPGain() - * - * Decorrelate LPC gains. There are 6 LPC Gains per frame. This is like - * multiplying gain vector with decorrelating matrix. - * - * Input: - * -data : LPC gain in log-domain with mean removed. - * - * Output: - * -out : decorrelated parameters. - */ -int16_t WebRtcIsac_DecorrelateLPGain( - const double* data, - double* out) -{ - int16_t rowCntr; - int16_t colCntr; - - for(colCntr = 0; colCntr < UB_LPC_GAIN_DIM; colCntr++) - { - *out = 0; - for(rowCntr = 0; rowCntr < UB_LPC_GAIN_DIM; rowCntr++) - { - *out += data[rowCntr] * WebRtcIsac_kLpcGainDecorrMat[rowCntr][colCntr]; - } - out++; - } - return 0; -} - -/****************************************************************************** - * WebRtcIsac_QuantizeLpcGain() - * - * Quantize the decorrelated log-domain gains. - * - * Input: - * -lpcGain : uncorrelated LPC gains. - * - * Output: - * -idx : quantization indices - * -lpcGain : quantized value of the inpt. - */ -double WebRtcIsac_QuantizeLpcGain( - double* data, - int* idx) -{ - int16_t coeffCntr; - for(coeffCntr = 0; coeffCntr < UB_LPC_GAIN_DIM; coeffCntr++) - { - *idx = (int)floor((*data - WebRtcIsac_kLeftRecPointLpcGain[coeffCntr]) / - WebRtcIsac_kQSizeLpcGain + 0.5); - - if(*idx < 0) - { - *idx = 0; - } - else if(*idx >= WebRtcIsac_kNumQCellLpcGain[coeffCntr]) - { - *idx = WebRtcIsac_kNumQCellLpcGain[coeffCntr] - 1; - } - *data = WebRtcIsac_kLeftRecPointLpcGain[coeffCntr] + *idx * - WebRtcIsac_kQSizeLpcGain; - - data++; - idx++; - } - return 0; -} - -/****************************************************************************** - * WebRtcIsac_DequantizeLpcGain() - * - * Get the quantized values given the quantization indices. - * - * Input: - * -idx : pointer to quantization indices. - * - * Output: - * -lpcGains : quantized values of the given parametes. - */ -int16_t WebRtcIsac_DequantizeLpcGain( - const int* idx, - double* out) -{ - int16_t coeffCntr; - for(coeffCntr = 0; coeffCntr < UB_LPC_GAIN_DIM; coeffCntr++) - { - *out = WebRtcIsac_kLeftRecPointLpcGain[coeffCntr] + *idx * - WebRtcIsac_kQSizeLpcGain; - out++; - idx++; - } - return 0; -} - -/****************************************************************************** - * WebRtcIsac_CorrelateLpcGain() - * - * This is the inverse of WebRtcIsac_DecorrelateLPGain(). - * - * Input: - * -data : decorrelated parameters. - * - * Output: - * -out : correlated parameters. - */ -int16_t WebRtcIsac_CorrelateLpcGain( - const double* data, - double* out) -{ - int16_t rowCntr; - int16_t colCntr; - - for(rowCntr = 0; rowCntr < UB_LPC_GAIN_DIM; rowCntr++) - { - *out = 0; - for(colCntr = 0; colCntr < UB_LPC_GAIN_DIM; colCntr++) - { - *out += WebRtcIsac_kLpcGainDecorrMat[rowCntr][colCntr] * data[colCntr]; - } - out++; - } - - return 0; -} - - -/****************************************************************************** - * WebRtcIsac_AddMeanToLinearDomain() - * - * This is the inverse of WebRtcIsac_ToLogDomainRemoveMean(). - * - * Input: - * -lpcGain : LPC gain in log-domain & mean removed - * - * Output: - * -lpcGain : LPC gain in normal domain. - */ -int16_t WebRtcIsac_AddMeanToLinearDomain( - double* lpcGains) -{ - int16_t coeffCntr; - for(coeffCntr = 0; coeffCntr < UB_LPC_GAIN_DIM; coeffCntr++) - { - lpcGains[coeffCntr] = exp(lpcGains[coeffCntr] + WebRtcIsac_kMeanLpcGain); - } - return 0; -} diff --git a/modules/audio_coding/codecs/isac/main/source/encode_lpc_swb.h b/modules/audio_coding/codecs/isac/main/source/encode_lpc_swb.h deleted file mode 100644 index 8bc3d752c3..0000000000 --- a/modules/audio_coding/codecs/isac/main/source/encode_lpc_swb.h +++ /dev/null @@ -1,246 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/* - * encode_lpc_swb.h - * - * This file contains declaration of functions used to - * encode LPC parameters (Shape & gain) of the upper band. - * - */ - -#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_ENCODE_LPC_SWB_H_ -#define MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_ENCODE_LPC_SWB_H_ - -#include "modules/audio_coding/codecs/isac/main/source/settings.h" -#include "modules/audio_coding/codecs/isac/main/source/structs.h" - -/****************************************************************************** - * WebRtcIsac_RemoveLarMean() - * - * Remove the means from LAR coefficients. - * - * Input: - * -lar : pointer to lar vectors. LAR vectors are - * concatenated. - * -bandwidth : indicates if the given LAR vectors belong - * to SWB-12kHz or SWB-16kHz. - * - * Output: - * -lar : pointer to mean-removed LAR:s. - * - * - */ -int16_t WebRtcIsac_RemoveLarMean(double* lar, int16_t bandwidth); - -/****************************************************************************** - * WebRtcIsac_DecorrelateIntraVec() - * - * Remove the correlation amonge the components of LAR vectors. If LAR vectors - * of one frame are put in a matrix where each column is a LAR vector of a - * sub-frame, then this is equivalent to multiplying the LAR matrix with - * a decorrelting mtrix from left. - * - * Input: - * -inLar : pointer to mean-removed LAR vecrtors. - * -bandwidth : indicates if the given LAR vectors belong - * to SWB-12kHz or SWB-16kHz. - * - * Output: - * -out : decorrelated LAR vectors. - */ -int16_t WebRtcIsac_DecorrelateIntraVec(const double* inLAR, - double* out, - int16_t bandwidth); - -/****************************************************************************** - * WebRtcIsac_DecorrelateInterVec() - * - * Remover the correlation among mean-removed LAR vectors. If LAR vectors - * of one frame are put in a matrix where each column is a LAR vector of a - * sub-frame, then this is equivalent to multiplying the LAR matrix with - * a decorrelting mtrix from right. - * - * Input: - * -data : pointer to matrix of LAR vectors. The matrix - * is stored column-wise. - * -bandwidth : indicates if the given LAR vectors belong - * to SWB-12kHz or SWB-16kHz. - * - * Output: - * -out : decorrelated LAR vectors. - */ -int16_t WebRtcIsac_DecorrelateInterVec(const double* data, - double* out, - int16_t bandwidth); - -/****************************************************************************** - * WebRtcIsac_QuantizeUncorrLar() - * - * Quantize the uncorrelated parameters. - * - * Input: - * -data : uncorrelated LAR vectors. - * -bandwidth : indicates if the given LAR vectors belong - * to SWB-12kHz or SWB-16kHz. - * - * Output: - * -data : quantized version of the input. - * -idx : pointer to quantization indices. - */ -double WebRtcIsac_QuantizeUncorrLar(double* data, int* idx, int16_t bandwidth); - -/****************************************************************************** - * WebRtcIsac_CorrelateIntraVec() - * - * This is the inverse of WebRtcIsac_DecorrelateIntraVec(). - * - * Input: - * -data : uncorrelated parameters. - * -bandwidth : indicates if the given LAR vectors belong - * to SWB-12kHz or SWB-16kHz. - * - * Output: - * -out : correlated parametrs. - */ -int16_t WebRtcIsac_CorrelateIntraVec(const double* data, - double* out, - int16_t bandwidth); - -/****************************************************************************** - * WebRtcIsac_CorrelateInterVec() - * - * This is the inverse of WebRtcIsac_DecorrelateInterVec(). - * - * Input: - * -data - * -bandwidth : indicates if the given LAR vectors belong - * to SWB-12kHz or SWB-16kHz. - * - * Output: - * -out : correlated parametrs. - */ -int16_t WebRtcIsac_CorrelateInterVec(const double* data, - double* out, - int16_t bandwidth); - -/****************************************************************************** - * WebRtcIsac_AddLarMean() - * - * This is the inverse of WebRtcIsac_RemoveLarMean() - * - * Input: - * -data : pointer to mean-removed LAR:s. - * -bandwidth : indicates if the given LAR vectors belong - * to SWB-12kHz or SWB-16kHz. - * - * Output: - * -data : pointer to LARs. - */ -int16_t WebRtcIsac_AddLarMean(double* data, int16_t bandwidth); - -/****************************************************************************** - * WebRtcIsac_DequantizeLpcParam() - * - * Get the quantized value of uncorrelated LARs given the quantization indices. - * - * Input: - * -idx : pointer to quantiztion indices. - * -bandwidth : indicates if the given LAR vectors belong - * to SWB-12kHz or SWB-16kHz. - * - * Output: - * -out : pointer to quantized values. - */ -int16_t WebRtcIsac_DequantizeLpcParam(const int* idx, - double* out, - int16_t bandwidth); - -/****************************************************************************** - * WebRtcIsac_ToLogDomainRemoveMean() - * - * Transform the LPC gain to log domain then remove the mean value. - * - * Input: - * -lpcGain : pointer to LPC Gain, expecting 6 LPC gains - * - * Output: - * -lpcGain : mean-removed in log domain. - */ -int16_t WebRtcIsac_ToLogDomainRemoveMean(double* lpGains); - -/****************************************************************************** - * WebRtcIsac_DecorrelateLPGain() - * - * Decorrelate LPC gains. There are 6 LPC Gains per frame. This is like - * multiplying gain vector with decorrelating matrix. - * - * Input: - * -data : LPC gain in log-domain with mean removed. - * - * Output: - * -out : decorrelated parameters. - */ -int16_t WebRtcIsac_DecorrelateLPGain(const double* data, double* out); - -/****************************************************************************** - * WebRtcIsac_QuantizeLpcGain() - * - * Quantize the decorrelated log-domain gains. - * - * Input: - * -lpcGain : uncorrelated LPC gains. - * - * Output: - * -idx : quantization indices - * -lpcGain : quantized value of the inpt. - */ -double WebRtcIsac_QuantizeLpcGain(double* lpGains, int* idx); - -/****************************************************************************** - * WebRtcIsac_DequantizeLpcGain() - * - * Get the quantized values given the quantization indices. - * - * Input: - * -idx : pointer to quantization indices. - * - * Output: - * -lpcGains : quantized values of the given parametes. - */ -int16_t WebRtcIsac_DequantizeLpcGain(const int* idx, double* lpGains); - -/****************************************************************************** - * WebRtcIsac_CorrelateLpcGain() - * - * This is the inverse of WebRtcIsac_DecorrelateLPGain(). - * - * Input: - * -data : decorrelated parameters. - * - * Output: - * -out : correlated parameters. - */ -int16_t WebRtcIsac_CorrelateLpcGain(const double* data, double* out); - -/****************************************************************************** - * WebRtcIsac_AddMeanToLinearDomain() - * - * This is the inverse of WebRtcIsac_ToLogDomainRemoveMean(). - * - * Input: - * -lpcGain : LPC gain in log-domain & mean removed - * - * Output: - * -lpcGain : LPC gain in normal domain. - */ -int16_t WebRtcIsac_AddMeanToLinearDomain(double* lpcGains); - -#endif // MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_ENCODE_LPC_SWB_H_ diff --git a/modules/audio_coding/codecs/isac/main/source/entropy_coding.c b/modules/audio_coding/codecs/isac/main/source/entropy_coding.c deleted file mode 100644 index 188c8f6b86..0000000000 --- a/modules/audio_coding/codecs/isac/main/source/entropy_coding.c +++ /dev/null @@ -1,2066 +0,0 @@ -/* - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/* - * entropy_coding.c - * - * This header file defines all of the functions used to arithmetically - * encode the iSAC bistream - * - */ - - -#include "common_audio/signal_processing/include/signal_processing_library.h" -#include "modules/audio_coding/codecs/isac/main/source/entropy_coding.h" -#include "modules/audio_coding/codecs/isac/main/source/settings.h" -#include "modules/audio_coding/codecs/isac/main/source/arith_routines.h" -#include "modules/audio_coding/codecs/isac/main/source/spectrum_ar_model_tables.h" -#include "modules/audio_coding/codecs/isac/main/source/lpc_tables.h" -#include "modules/audio_coding/codecs/isac/main/source/pitch_gain_tables.h" -#include "modules/audio_coding/codecs/isac/main/source/pitch_lag_tables.h" -#include "modules/audio_coding/codecs/isac/main/source/encode_lpc_swb.h" -#include "modules/audio_coding/codecs/isac/main/source/lpc_shape_swb12_tables.h" -#include "modules/audio_coding/codecs/isac/main/source/lpc_shape_swb16_tables.h" -#include "modules/audio_coding/codecs/isac/main/source/lpc_gain_swb_tables.h" -#include "modules/audio_coding/codecs/isac/main/source/os_specific_inline.h" - -#include -#include - -static const uint16_t kLpcVecPerSegmentUb12 = 5; -static const uint16_t kLpcVecPerSegmentUb16 = 4; - -/* CDF array for encoder bandwidth (12 vs 16 kHz) indicator. */ -static const uint16_t kOneBitEqualProbCdf[3] = { - 0, 32768, 65535 }; - -/* Pointer to cdf array for encoder bandwidth (12 vs 16 kHz) indicator. */ -static const uint16_t* const kOneBitEqualProbCdf_ptr[1] = { - kOneBitEqualProbCdf }; - -/* - * Initial cdf index for decoder of encoded bandwidth - * (12 vs 16 kHz) indicator. - */ -static const uint16_t kOneBitEqualProbInitIndex[1] = { 1 }; - - -static const int kIsSWB12 = 1; - -/* compute correlation from power spectrum */ -static void FindCorrelation(int32_t* PSpecQ12, int32_t* CorrQ7) { - int32_t summ[FRAMESAMPLES / 8]; - int32_t diff[FRAMESAMPLES / 8]; - const int16_t* CS_ptrQ9; - int32_t sum; - int k, n; - - for (k = 0; k < FRAMESAMPLES / 8; k++) { - summ[k] = (PSpecQ12[k] + PSpecQ12[FRAMESAMPLES_QUARTER - 1 - k] + 16) >> 5; - diff[k] = (PSpecQ12[k] - PSpecQ12[FRAMESAMPLES_QUARTER - 1 - k] + 16) >> 5; - } - - sum = 2; - for (n = 0; n < FRAMESAMPLES / 8; n++) { - sum += summ[n]; - } - CorrQ7[0] = sum; - - for (k = 0; k < AR_ORDER; k += 2) { - sum = 0; - CS_ptrQ9 = WebRtcIsac_kCos[k]; - for (n = 0; n < FRAMESAMPLES / 8; n++) - sum += (CS_ptrQ9[n] * diff[n] + 256) >> 9; - CorrQ7[k + 1] = sum; - } - - for (k = 1; k < AR_ORDER; k += 2) { - sum = 0; - CS_ptrQ9 = WebRtcIsac_kCos[k]; - for (n = 0; n < FRAMESAMPLES / 8; n++) - sum += (CS_ptrQ9[n] * summ[n] + 256) >> 9; - CorrQ7[k + 1] = sum; - } -} - -/* compute inverse AR power spectrum */ -/* Changed to the function used in iSAC FIX for compatibility reasons */ -static void FindInvArSpec(const int16_t* ARCoefQ12, - const int32_t gainQ10, - int32_t* CurveQ16) { - int32_t CorrQ11[AR_ORDER + 1]; - int64_t sum, tmpGain; - int32_t diffQ16[FRAMESAMPLES / 8]; - const int16_t* CS_ptrQ9; - int k, n; - int16_t round, shftVal = 0, sh; - - sum = 0; - for (n = 0; n < AR_ORDER + 1; n++) { - sum += WEBRTC_SPL_MUL(ARCoefQ12[n], ARCoefQ12[n]); /* Q24 */ - } - sum = ((sum >> 6) * 65 + 32768) >> 16; /* Q8 */ - CorrQ11[0] = (sum * gainQ10 + 256) >> 9; - - /* To avoid overflow, we shift down gainQ10 if it is large. - * We will not lose any precision */ - if (gainQ10 > 400000) { - tmpGain = gainQ10 >> 3; - round = 32; - shftVal = 6; - } else { - tmpGain = gainQ10; - round = 256; - shftVal = 9; - } - - for (k = 1; k < AR_ORDER + 1; k++) { - sum = 16384; - for (n = k; n < AR_ORDER + 1; n++) - sum += WEBRTC_SPL_MUL(ARCoefQ12[n - k], ARCoefQ12[n]); /* Q24 */ - sum >>= 15; - CorrQ11[k] = (sum * tmpGain + round) >> shftVal; - } - sum = CorrQ11[0] << 7; - for (n = 0; n < FRAMESAMPLES / 8; n++) { - CurveQ16[n] = sum; - } - for (k = 1; k < AR_ORDER; k += 2) { - for (n = 0; n < FRAMESAMPLES / 8; n++) { - CurveQ16[n] += (WebRtcIsac_kCos[k][n] * CorrQ11[k + 1] + 2) >> 2; - } - } - - CS_ptrQ9 = WebRtcIsac_kCos[0]; - - /* If CorrQ11[1] too large we avoid getting overflow in the - * calculation by shifting */ - sh = WebRtcSpl_NormW32(CorrQ11[1]); - if (CorrQ11[1] == 0) { /* Use next correlation */ - sh = WebRtcSpl_NormW32(CorrQ11[2]); - } - if (sh < 9) { - shftVal = 9 - sh; - } else { - shftVal = 0; - } - for (n = 0; n < FRAMESAMPLES / 8; n++) { - diffQ16[n] = (CS_ptrQ9[n] * (CorrQ11[1] >> shftVal) + 2) >> 2; - } - for (k = 2; k < AR_ORDER; k += 2) { - CS_ptrQ9 = WebRtcIsac_kCos[k]; - for (n = 0; n < FRAMESAMPLES / 8; n++) { - diffQ16[n] += (CS_ptrQ9[n] * (CorrQ11[k + 1] >> shftVal) + 2) >> 2; - } - } - - for (k = 0; k < FRAMESAMPLES / 8; k++) { - int32_t diff_q16_shifted = (int32_t)((uint32_t)(diffQ16[k]) << shftVal); - CurveQ16[FRAMESAMPLES_QUARTER - 1 - k] = CurveQ16[k] - diff_q16_shifted; - CurveQ16[k] += diff_q16_shifted; - } -} - -/* Generate array of dither samples in Q7. */ -static void GenerateDitherQ7Lb(int16_t* bufQ7, uint32_t seed, - int length, int16_t AvgPitchGain_Q12) { - int k, shft; - int16_t dither1_Q7, dither2_Q7, dither_gain_Q14; - - /* This threshold should be equal to that in decode_spec(). */ - if (AvgPitchGain_Q12 < 614) { - for (k = 0; k < length - 2; k += 3) { - /* New random unsigned int. */ - seed = (seed * 196314165) + 907633515; - - /* Fixed-point dither sample between -64 and 64 (Q7). */ - /* dither = seed * 128 / 4294967295 */ - dither1_Q7 = (int16_t)(((int32_t)(seed + 16777216)) >> 25); - - /* New random unsigned int. */ - seed = (seed * 196314165) + 907633515; - - /* Fixed-point dither sample between -64 and 64. */ - dither2_Q7 = (int16_t)(((int32_t)(seed + 16777216)) >> 25); - - shft = (seed >> 25) & 15; - if (shft < 5) { - bufQ7[k] = dither1_Q7; - bufQ7[k + 1] = dither2_Q7; - bufQ7[k + 2] = 0; - } else if (shft < 10) { - bufQ7[k] = dither1_Q7; - bufQ7[k + 1] = 0; - bufQ7[k + 2] = dither2_Q7; - } else { - bufQ7[k] = 0; - bufQ7[k + 1] = dither1_Q7; - bufQ7[k + 2] = dither2_Q7; - } - } - } else { - dither_gain_Q14 = (int16_t)(22528 - 10 * AvgPitchGain_Q12); - - /* Dither on half of the coefficients. */ - for (k = 0; k < length - 1; k += 2) { - /* New random unsigned int */ - seed = (seed * 196314165) + 907633515; - - /* Fixed-point dither sample between -64 and 64. */ - dither1_Q7 = (int16_t)(((int32_t)(seed + 16777216)) >> 25); - - /* Dither sample is placed in either even or odd index. */ - shft = (seed >> 25) & 1; /* Either 0 or 1 */ - - bufQ7[k + shft] = (((dither_gain_Q14 * dither1_Q7) + 8192) >> 14); - bufQ7[k + 1 - shft] = 0; - } - } -} - - - -/****************************************************************************** - * GenerateDitherQ7LbUB() - * - * generate array of dither samples in Q7 There are less zeros in dither - * vector compared to GenerateDitherQ7Lb. - * - * A uniform random number generator with the range of [-64 64] is employed - * but the generated dithers are scaled by 0.35, a heuristic scaling. - * - * Input: - * -seed : the initial seed for the random number generator. - * -length : the number of dither values to be generated. - * - * Output: - * -bufQ7 : pointer to a buffer where dithers are written to. - */ -static void GenerateDitherQ7LbUB( - int16_t* bufQ7, - uint32_t seed, - int length) { - int k; - for (k = 0; k < length; k++) { - /* new random unsigned int */ - seed = (seed * 196314165) + 907633515; - - /* Fixed-point dither sample between -64 and 64 (Q7). */ - /* bufQ7 = seed * 128 / 4294967295 */ - bufQ7[k] = (int16_t)(((int32_t)(seed + 16777216)) >> 25); - - /* Scale by 0.35. */ - bufQ7[k] = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(bufQ7[k], 2048, 13); - } -} - -/* - * Function to decode the complex spectrum from the bit stream - * returns the total number of bytes in the stream. - */ -int WebRtcIsac_DecodeSpec(Bitstr* streamdata, int16_t AvgPitchGain_Q12, - enum ISACBand band, double* fr, double* fi) { - int16_t DitherQ7[FRAMESAMPLES]; - int16_t data[FRAMESAMPLES]; - int32_t invARSpec2_Q16[FRAMESAMPLES_QUARTER]; - uint16_t invARSpecQ8[FRAMESAMPLES_QUARTER]; - int16_t ARCoefQ12[AR_ORDER + 1]; - int16_t RCQ15[AR_ORDER]; - int16_t gainQ10; - int32_t gain2_Q10, res; - int32_t in_sqrt; - int32_t newRes; - int k, len, i; - int is_12khz = !kIsSWB12; - int num_dft_coeff = FRAMESAMPLES; - /* Create dither signal. */ - if (band == kIsacLowerBand) { - GenerateDitherQ7Lb(DitherQ7, streamdata->W_upper, FRAMESAMPLES, - AvgPitchGain_Q12); - } else { - GenerateDitherQ7LbUB(DitherQ7, streamdata->W_upper, FRAMESAMPLES); - if (band == kIsacUpperBand12) { - is_12khz = kIsSWB12; - num_dft_coeff = FRAMESAMPLES_HALF; - } - } - - /* Decode model parameters. */ - if (WebRtcIsac_DecodeRc(streamdata, RCQ15) < 0) - return -ISAC_RANGE_ERROR_DECODE_SPECTRUM; - - WebRtcSpl_ReflCoefToLpc(RCQ15, AR_ORDER, ARCoefQ12); - - if (WebRtcIsac_DecodeGain2(streamdata, &gain2_Q10) < 0) - return -ISAC_RANGE_ERROR_DECODE_SPECTRUM; - - /* Compute inverse AR power spectrum. */ - FindInvArSpec(ARCoefQ12, gain2_Q10, invARSpec2_Q16); - - /* Convert to magnitude spectrum, - * by doing square-roots (modified from SPLIB). */ - res = 1 << (WebRtcSpl_GetSizeInBits(invARSpec2_Q16[0]) >> 1); - for (k = 0; k < FRAMESAMPLES_QUARTER; k++) { - in_sqrt = invARSpec2_Q16[k]; - i = 10; - - /* Negative values make no sense for a real sqrt-function. */ - if (in_sqrt < 0) - in_sqrt = -in_sqrt; - - newRes = (in_sqrt / res + res) >> 1; - do { - res = newRes; - newRes = (in_sqrt / res + res) >> 1; - } while (newRes != res && i-- > 0); - - invARSpecQ8[k] = (int16_t)newRes; - } - - len = WebRtcIsac_DecLogisticMulti2(data, streamdata, invARSpecQ8, DitherQ7, - num_dft_coeff, is_12khz); - /* Arithmetic decoding of spectrum. */ - if (len < 1) { - return -ISAC_RANGE_ERROR_DECODE_SPECTRUM; - } - - switch (band) { - case kIsacLowerBand: { - /* Scale down spectral samples with low SNR. */ - int32_t p1; - int32_t p2; - if (AvgPitchGain_Q12 <= 614) { - p1 = 30 << 10; - p2 = 32768 + (33 << 16); - } else { - p1 = 36 << 10; - p2 = 32768 + (40 << 16); - } - for (k = 0; k < FRAMESAMPLES; k += 4) { - gainQ10 = WebRtcSpl_DivW32W16ResW16(p1, (int16_t)( - (invARSpec2_Q16[k >> 2] + p2) >> 16)); - *fr++ = (double)((data[ k ] * gainQ10 + 512) >> 10) / 128.0; - *fi++ = (double)((data[k + 1] * gainQ10 + 512) >> 10) / 128.0; - *fr++ = (double)((data[k + 2] * gainQ10 + 512) >> 10) / 128.0; - *fi++ = (double)((data[k + 3] * gainQ10 + 512) >> 10) / 128.0; - } - break; - } - case kIsacUpperBand12: { - for (k = 0, i = 0; k < FRAMESAMPLES_HALF; k += 4) { - fr[i] = (double)data[ k ] / 128.0; - fi[i] = (double)data[k + 1] / 128.0; - i++; - fr[i] = (double)data[k + 2] / 128.0; - fi[i] = (double)data[k + 3] / 128.0; - i++; - } - /* The second half of real and imaginary coefficients is zero. This is - * due to using the old FFT module which requires two signals as input - * while in 0-12 kHz mode we only have 8-12 kHz band, and the second - * signal is set to zero. */ - memset(&fr[FRAMESAMPLES_QUARTER], 0, FRAMESAMPLES_QUARTER * - sizeof(double)); - memset(&fi[FRAMESAMPLES_QUARTER], 0, FRAMESAMPLES_QUARTER * - sizeof(double)); - break; - } - case kIsacUpperBand16: { - for (i = 0, k = 0; k < FRAMESAMPLES; k += 4, i++) { - fr[i] = (double)data[ k ] / 128.0; - fi[i] = (double)data[k + 1] / 128.0; - fr[(FRAMESAMPLES_HALF) - 1 - i] = (double)data[k + 2] / 128.0; - fi[(FRAMESAMPLES_HALF) - 1 - i] = (double)data[k + 3] / 128.0; - } - break; - } - } - return len; -} - - -int WebRtcIsac_EncodeSpec(const int16_t* fr, const int16_t* fi, - int16_t AvgPitchGain_Q12, enum ISACBand band, - Bitstr* streamdata) { - int16_t ditherQ7[FRAMESAMPLES]; - int16_t dataQ7[FRAMESAMPLES]; - int32_t PSpec[FRAMESAMPLES_QUARTER]; - int32_t invARSpec2_Q16[FRAMESAMPLES_QUARTER]; - uint16_t invARSpecQ8[FRAMESAMPLES_QUARTER]; - int32_t CorrQ7[AR_ORDER + 1]; - int32_t CorrQ7_norm[AR_ORDER + 1]; - int16_t RCQ15[AR_ORDER]; - int16_t ARCoefQ12[AR_ORDER + 1]; - int32_t gain2_Q10; - int16_t val; - int32_t nrg, res; - uint32_t sum; - int32_t in_sqrt; - int32_t newRes; - int16_t err; - uint32_t nrg_u32; - int shift_var; - int k, n, j, i; - int is_12khz = !kIsSWB12; - int num_dft_coeff = FRAMESAMPLES; - - /* Create dither signal. */ - if (band == kIsacLowerBand) { - GenerateDitherQ7Lb(ditherQ7, streamdata->W_upper, FRAMESAMPLES, - AvgPitchGain_Q12); - } else { - GenerateDitherQ7LbUB(ditherQ7, streamdata->W_upper, FRAMESAMPLES); - if (band == kIsacUpperBand12) { - is_12khz = kIsSWB12; - num_dft_coeff = FRAMESAMPLES_HALF; - } - } - - /* add dither and quantize, and compute power spectrum */ - switch (band) { - case kIsacLowerBand: { - for (k = 0; k < FRAMESAMPLES; k += 4) { - val = ((*fr++ + ditherQ7[k] + 64) & 0xFF80) - ditherQ7[k]; - dataQ7[k] = val; - sum = val * val; - - val = ((*fi++ + ditherQ7[k + 1] + 64) & 0xFF80) - ditherQ7[k + 1]; - dataQ7[k + 1] = val; - sum += val * val; - - val = ((*fr++ + ditherQ7[k + 2] + 64) & 0xFF80) - ditherQ7[k + 2]; - dataQ7[k + 2] = val; - sum += val * val; - - val = ((*fi++ + ditherQ7[k + 3] + 64) & 0xFF80) - ditherQ7[k + 3]; - dataQ7[k + 3] = val; - sum += val * val; - - PSpec[k >> 2] = sum >> 2; - } - break; - } - case kIsacUpperBand12: { - for (k = 0, j = 0; k < FRAMESAMPLES_HALF; k += 4) { - val = ((*fr++ + ditherQ7[k] + 64) & 0xFF80) - ditherQ7[k]; - dataQ7[k] = val; - sum = val * val; - - val = ((*fi++ + ditherQ7[k + 1] + 64) & 0xFF80) - ditherQ7[k + 1]; - dataQ7[k + 1] = val; - sum += val * val; - - PSpec[j++] = sum >> 1; - - val = ((*fr++ + ditherQ7[k + 2] + 64) & 0xFF80) - ditherQ7[k + 2]; - dataQ7[k + 2] = val; - sum = val * val; - - val = ((*fi++ + ditherQ7[k + 3] + 64) & 0xFF80) - ditherQ7[k + 3]; - dataQ7[k + 3] = val; - sum += val * val; - - PSpec[j++] = sum >> 1; - } - break; - } - case kIsacUpperBand16: { - for (j = 0, k = 0; k < FRAMESAMPLES; k += 4, j++) { - val = ((fr[j] + ditherQ7[k] + 64) & 0xFF80) - ditherQ7[k]; - dataQ7[k] = val; - sum = val * val; - - val = ((fi[j] + ditherQ7[k + 1] + 64) & 0xFF80) - ditherQ7[k + 1]; - dataQ7[k + 1] = val; - sum += val * val; - - val = ((fr[(FRAMESAMPLES_HALF) - 1 - j] + ditherQ7[k + 2] + 64) & - 0xFF80) - ditherQ7[k + 2]; - dataQ7[k + 2] = val; - sum += val * val; - - val = ((fi[(FRAMESAMPLES_HALF) - 1 - j] + ditherQ7[k + 3] + 64) & - 0xFF80) - ditherQ7[k + 3]; - dataQ7[k + 3] = val; - sum += val * val; - - PSpec[k >> 2] = sum >> 2; - } - break; - } - } - - /* compute correlation from power spectrum */ - FindCorrelation(PSpec, CorrQ7); - - /* Find AR coefficients */ - /* Aumber of bit shifts to 14-bit normalize CorrQ7[0] - * (leaving room for sign) */ - shift_var = WebRtcSpl_NormW32(CorrQ7[0]) - 18; - - if (shift_var > 0) { - for (k = 0; k < AR_ORDER + 1; k++) { - CorrQ7_norm[k] = CorrQ7[k] << shift_var; - } - } else { - for (k = 0; k < AR_ORDER + 1; k++) { - CorrQ7_norm[k] = CorrQ7[k] >> (-shift_var); - } - } - - /* Find RC coefficients. */ - WebRtcSpl_AutoCorrToReflCoef(CorrQ7_norm, AR_ORDER, RCQ15); - - /* Quantize & code RC Coefficient. */ - WebRtcIsac_EncodeRc(RCQ15, streamdata); - - /* RC -> AR coefficients */ - WebRtcSpl_ReflCoefToLpc(RCQ15, AR_ORDER, ARCoefQ12); - - /* Compute ARCoef' * Corr * ARCoef in Q19. */ - nrg = 0; - for (j = 0; j <= AR_ORDER; j++) { - for (n = 0; n <= j; n++) { - nrg += (ARCoefQ12[j] * ((CorrQ7_norm[j - n] * ARCoefQ12[n] + 256) >> 9) + - 4) >> 3; - } - for (n = j + 1; n <= AR_ORDER; n++) { - nrg += (ARCoefQ12[j] * ((CorrQ7_norm[n - j] * ARCoefQ12[n] + 256) >> 9) + - 4) >> 3; - } - } - - nrg_u32 = (uint32_t)nrg; - if (shift_var > 0) { - nrg_u32 = nrg_u32 >> shift_var; - } else { - nrg_u32 = nrg_u32 << (-shift_var); - } - if (nrg_u32 > 0x7FFFFFFF) { - nrg = 0x7FFFFFFF; - } else { - nrg = (int32_t)nrg_u32; - } - /* Also shifts 31 bits to the left! */ - gain2_Q10 = WebRtcSpl_DivResultInQ31(FRAMESAMPLES_QUARTER, nrg); - - /* Quantize & code gain2_Q10. */ - if (WebRtcIsac_EncodeGain2(&gain2_Q10, streamdata)) { - return -1; - } - - /* Compute inverse AR power spectrum. */ - FindInvArSpec(ARCoefQ12, gain2_Q10, invARSpec2_Q16); - /* Convert to magnitude spectrum, by doing square-roots - * (modified from SPLIB). */ - res = 1 << (WebRtcSpl_GetSizeInBits(invARSpec2_Q16[0]) >> 1); - for (k = 0; k < FRAMESAMPLES_QUARTER; k++) { - in_sqrt = invARSpec2_Q16[k]; - i = 10; - /* Negative values make no sense for a real sqrt-function. */ - if (in_sqrt < 0) { - in_sqrt = -in_sqrt; - } - newRes = (in_sqrt / res + res) >> 1; - do { - res = newRes; - newRes = (in_sqrt / res + res) >> 1; - } while (newRes != res && i-- > 0); - - invARSpecQ8[k] = (int16_t)newRes; - } - /* arithmetic coding of spectrum */ - err = WebRtcIsac_EncLogisticMulti2(streamdata, dataQ7, invARSpecQ8, - num_dft_coeff, is_12khz); - if (err < 0) { - return (err); - } - return 0; -} - - -/* step-up */ -void WebRtcIsac_Rc2Poly(double* RC, int N, double* a) { - int m, k; - double tmp[MAX_AR_MODEL_ORDER]; - - a[0] = 1.0; - tmp[0] = 1.0; - for (m = 1; m <= N; m++) { - /* copy */ - memcpy(&tmp[1], &a[1], (m - 1) * sizeof(double)); - a[m] = RC[m - 1]; - for (k = 1; k < m; k++) { - a[k] += RC[m - 1] * tmp[m - k]; - } - } - return; -} - -/* step-down */ -void WebRtcIsac_Poly2Rc(double* a, int N, double* RC) { - int m, k; - double tmp[MAX_AR_MODEL_ORDER]; - double tmp_inv; - - RC[N - 1] = a[N]; - for (m = N - 1; m > 0; m--) { - tmp_inv = 1.0 / (1.0 - RC[m] * RC[m]); - for (k = 1; k <= m; k++) { - tmp[k] = (a[k] - RC[m] * a[m - k + 1]) * tmp_inv; - } - - memcpy(&a[1], &tmp[1], (m - 1) * sizeof(double)); - RC[m - 1] = tmp[m]; - } - return; -} - - -#define MAX_ORDER 100 - -/* Matlab's LAR definition */ -void WebRtcIsac_Rc2Lar(const double* refc, double* lar, int order) { - int k; - for (k = 0; k < order; k++) { - lar[k] = log((1 + refc[k]) / (1 - refc[k])); - } -} - - -void WebRtcIsac_Lar2Rc(const double* lar, double* refc, int order) { - int k; - double tmp; - - for (k = 0; k < order; k++) { - tmp = exp(lar[k]); - refc[k] = (tmp - 1) / (tmp + 1); - } -} - -void WebRtcIsac_Poly2Lar(double* lowband, int orderLo, double* hiband, - int orderHi, int Nsub, double* lars) { - int k; - double rc[MAX_ORDER], *inpl, *inph, *outp; - - inpl = lowband; - inph = hiband; - outp = lars; - for (k = 0; k < Nsub; k++) { - /* gains */ - outp[0] = inpl[0]; - outp[1] = inph[0]; - outp += 2; - - /* Low band */ - inpl[0] = 1.0; - WebRtcIsac_Poly2Rc(inpl, orderLo, rc); - WebRtcIsac_Rc2Lar(rc, outp, orderLo); - outp += orderLo; - - /* High band */ - inph[0] = 1.0; - WebRtcIsac_Poly2Rc(inph, orderHi, rc); - WebRtcIsac_Rc2Lar(rc, outp, orderHi); - outp += orderHi; - - inpl += orderLo + 1; - inph += orderHi + 1; - } -} - - -int16_t WebRtcIsac_Poly2LarUB(double* lpcVecs, int16_t bandwidth) { - double poly[MAX_ORDER]; - double rc[MAX_ORDER]; - double* ptrIO; - int16_t vecCntr; - int16_t vecSize; - int16_t numVec; - - vecSize = UB_LPC_ORDER; - switch (bandwidth) { - case isac12kHz: { - numVec = UB_LPC_VEC_PER_FRAME; - break; - } - case isac16kHz: { - numVec = UB16_LPC_VEC_PER_FRAME; - break; - } - default: - return -1; - } - - ptrIO = lpcVecs; - poly[0] = 1.0; - for (vecCntr = 0; vecCntr < numVec; vecCntr++) { - memcpy(&poly[1], ptrIO, sizeof(double) * vecSize); - WebRtcIsac_Poly2Rc(poly, vecSize, rc); - WebRtcIsac_Rc2Lar(rc, ptrIO, vecSize); - ptrIO += vecSize; - } - return 0; -} - - -void WebRtcIsac_Lar2Poly(double* lars, double* lowband, int orderLo, - double* hiband, int orderHi, int Nsub) { - int k, orderTot; - double rc[MAX_ORDER], *outpl, *outph, *inp; - - orderTot = (orderLo + orderHi + 2); - outpl = lowband; - outph = hiband; - /* First two elements of 'inp' store gains*/ - inp = lars; - for (k = 0; k < Nsub; k++) { - /* Low band */ - WebRtcIsac_Lar2Rc(&inp[2], rc, orderLo); - WebRtcIsac_Rc2Poly(rc, orderLo, outpl); - - /* High band */ - WebRtcIsac_Lar2Rc(&inp[orderLo + 2], rc, orderHi); - WebRtcIsac_Rc2Poly(rc, orderHi, outph); - - /* gains */ - outpl[0] = inp[0]; - outph[0] = inp[1]; - - outpl += orderLo + 1; - outph += orderHi + 1; - inp += orderTot; - } -} - -/* - * assumes 2 LAR vectors interpolates to 'numPolyVec' A-polynomials - * Note: 'numPolyVecs' includes the first and the last point of the interval - */ -void WebRtcIsac_Lar2PolyInterpolUB(double* larVecs, double* percepFilterParams, - int numPolyVecs) { - int polyCntr, coeffCntr; - double larInterpol[UB_LPC_ORDER]; - double rc[UB_LPC_ORDER]; - double delta[UB_LPC_ORDER]; - - /* calculate the step-size for linear interpolation coefficients */ - for (coeffCntr = 0; coeffCntr < UB_LPC_ORDER; coeffCntr++) { - delta[coeffCntr] = (larVecs[UB_LPC_ORDER + coeffCntr] - - larVecs[coeffCntr]) / (numPolyVecs - 1); - } - - for (polyCntr = 0; polyCntr < numPolyVecs; polyCntr++) { - for (coeffCntr = 0; coeffCntr < UB_LPC_ORDER; coeffCntr++) { - larInterpol[coeffCntr] = larVecs[coeffCntr] + - delta[coeffCntr] * polyCntr; - } - WebRtcIsac_Lar2Rc(larInterpol, rc, UB_LPC_ORDER); - - /* convert to A-polynomial, the following function returns A[0] = 1; - * which is written where gains had to be written. Then we write the - * gain (outside this function). This way we say a memcpy. */ - WebRtcIsac_Rc2Poly(rc, UB_LPC_ORDER, percepFilterParams); - percepFilterParams += (UB_LPC_ORDER + 1); - } -} - -int WebRtcIsac_DecodeLpc(Bitstr* streamdata, double* LPCCoef_lo, - double* LPCCoef_hi) { - double lars[KLT_ORDER_GAIN + KLT_ORDER_SHAPE]; - int err; - - err = WebRtcIsac_DecodeLpcCoef(streamdata, lars); - if (err < 0) { - return -ISAC_RANGE_ERROR_DECODE_LPC; - } - WebRtcIsac_Lar2Poly(lars, LPCCoef_lo, ORDERLO, LPCCoef_hi, ORDERHI, - SUBFRAMES); - return 0; -} - -int16_t WebRtcIsac_DecodeInterpolLpcUb(Bitstr* streamdata, - double* percepFilterParams, - int16_t bandwidth) { - double lpcCoeff[UB_LPC_ORDER * UB16_LPC_VEC_PER_FRAME]; - int err; - int interpolCntr; - int subframeCntr; - int16_t numSegments; - int16_t numVecPerSegment; - int16_t numGains; - - double percepFilterGains[SUBFRAMES << 1]; - double* ptrOutParam = percepFilterParams; - - err = WebRtcIsac_DecodeLpcCoefUB(streamdata, lpcCoeff, percepFilterGains, - bandwidth); - if (err < 0) { - return -ISAC_RANGE_ERROR_DECODE_LPC; - } - - switch (bandwidth) { - case isac12kHz: { - numGains = SUBFRAMES; - numSegments = UB_LPC_VEC_PER_FRAME - 1; - numVecPerSegment = kLpcVecPerSegmentUb12; - break; - } - case isac16kHz: { - numGains = SUBFRAMES << 1; - numSegments = UB16_LPC_VEC_PER_FRAME - 1; - numVecPerSegment = kLpcVecPerSegmentUb16; - break; - } - default: - return -1; - } - - for (interpolCntr = 0; interpolCntr < numSegments; interpolCntr++) { - WebRtcIsac_Lar2PolyInterpolUB(&lpcCoeff[interpolCntr * UB_LPC_ORDER], - ptrOutParam, numVecPerSegment + 1); - ptrOutParam += (numVecPerSegment * (UB_LPC_ORDER + 1)); - } - - ptrOutParam = percepFilterParams; - - if (bandwidth == isac16kHz) { - ptrOutParam += (1 + UB_LPC_ORDER); - } - - for (subframeCntr = 0; subframeCntr < numGains; subframeCntr++) { - *ptrOutParam = percepFilterGains[subframeCntr]; - ptrOutParam += (1 + UB_LPC_ORDER); - } - return 0; -} - - -/* decode & dequantize LPC Coef */ -int WebRtcIsac_DecodeLpcCoef(Bitstr* streamdata, double* LPCCoef) { - int j, k, n, pos, pos2, posg, poss, offsg, offss, offs2; - int index_g[KLT_ORDER_GAIN], index_s[KLT_ORDER_SHAPE]; - double tmpcoeffs_g[KLT_ORDER_GAIN], tmpcoeffs_s[KLT_ORDER_SHAPE]; - double tmpcoeffs2_g[KLT_ORDER_GAIN], tmpcoeffs2_s[KLT_ORDER_SHAPE]; - double sum; - int err; - int model = 1; - - /* entropy decoding of model number */ - /* We are keeping this for backward compatibility of bit-streams. */ - err = WebRtcIsac_DecHistOneStepMulti(&model, streamdata, - WebRtcIsac_kQKltModelCdfPtr, - WebRtcIsac_kQKltModelInitIndex, 1); - if (err < 0) { - return err; - } - /* Only accepted value of model is 0. It is kept in bit-stream for backward - * compatibility. */ - if (model != 0) { - return -ISAC_DISALLOWED_LPC_MODEL; - } - - /* entropy decoding of quantization indices */ - err = WebRtcIsac_DecHistOneStepMulti( - index_s, streamdata, WebRtcIsac_kQKltCdfPtrShape, - WebRtcIsac_kQKltInitIndexShape, KLT_ORDER_SHAPE); - if (err < 0) { - return err; - } - err = WebRtcIsac_DecHistOneStepMulti( - index_g, streamdata, WebRtcIsac_kQKltCdfPtrGain, - WebRtcIsac_kQKltInitIndexGain, KLT_ORDER_GAIN); - if (err < 0) { - return err; - } - - /* find quantization levels for coefficients */ - for (k = 0; k < KLT_ORDER_SHAPE; k++) { - tmpcoeffs_s[k] = - WebRtcIsac_kQKltLevelsShape[WebRtcIsac_kQKltOffsetShape[k] + - index_s[k]]; - } - for (k = 0; k < KLT_ORDER_GAIN; k++) { - tmpcoeffs_g[k] = WebRtcIsac_kQKltLevelsGain[WebRtcIsac_kQKltOffsetGain[k] + - index_g[k]]; - } - - /* Inverse KLT */ - - /* Left transform, transpose matrix! */ - offsg = 0; - offss = 0; - posg = 0; - poss = 0; - for (j = 0; j < SUBFRAMES; j++) { - offs2 = 0; - for (k = 0; k < LPC_GAIN_ORDER; k++) { - sum = 0; - pos = offsg; - pos2 = offs2; - for (n = 0; n < LPC_GAIN_ORDER; n++) { - sum += tmpcoeffs_g[pos++] * WebRtcIsac_kKltT1Gain[pos2++]; - } - tmpcoeffs2_g[posg++] = sum; - offs2 += LPC_GAIN_ORDER; - } - offs2 = 0; - for (k = 0; k < LPC_SHAPE_ORDER; k++) { - sum = 0; - pos = offss; - pos2 = offs2; - for (n = 0; n < LPC_SHAPE_ORDER; n++) { - sum += tmpcoeffs_s[pos++] * WebRtcIsac_kKltT1Shape[pos2++]; - } - tmpcoeffs2_s[poss++] = sum; - offs2 += LPC_SHAPE_ORDER; - } - offsg += LPC_GAIN_ORDER; - offss += LPC_SHAPE_ORDER; - } - - /* Right transform, transpose matrix */ - offsg = 0; - offss = 0; - posg = 0; - poss = 0; - for (j = 0; j < SUBFRAMES; j++) { - posg = offsg; - for (k = 0; k < LPC_GAIN_ORDER; k++) { - sum = 0; - pos = k; - pos2 = j; - for (n = 0; n < SUBFRAMES; n++) { - sum += tmpcoeffs2_g[pos] * WebRtcIsac_kKltT2Gain[pos2]; - pos += LPC_GAIN_ORDER; - pos2 += SUBFRAMES; - - } - tmpcoeffs_g[posg++] = sum; - } - poss = offss; - for (k = 0; k < LPC_SHAPE_ORDER; k++) { - sum = 0; - pos = k; - pos2 = j; - for (n = 0; n < SUBFRAMES; n++) { - sum += tmpcoeffs2_s[pos] * WebRtcIsac_kKltT2Shape[pos2]; - pos += LPC_SHAPE_ORDER; - pos2 += SUBFRAMES; - } - tmpcoeffs_s[poss++] = sum; - } - offsg += LPC_GAIN_ORDER; - offss += LPC_SHAPE_ORDER; - } - - /* scaling, mean addition, and gain restoration */ - posg = 0; - poss = 0; - pos = 0; - for (k = 0; k < SUBFRAMES; k++) { - /* log gains */ - LPCCoef[pos] = tmpcoeffs_g[posg] / LPC_GAIN_SCALE; - LPCCoef[pos] += WebRtcIsac_kLpcMeansGain[posg]; - LPCCoef[pos] = exp(LPCCoef[pos]); - pos++; - posg++; - LPCCoef[pos] = tmpcoeffs_g[posg] / LPC_GAIN_SCALE; - LPCCoef[pos] += WebRtcIsac_kLpcMeansGain[posg]; - LPCCoef[pos] = exp(LPCCoef[pos]); - pos++; - posg++; - - /* Low-band LAR coefficients. */ - for (n = 0; n < LPC_LOBAND_ORDER; n++, pos++, poss++) { - LPCCoef[pos] = tmpcoeffs_s[poss] / LPC_LOBAND_SCALE; - LPCCoef[pos] += WebRtcIsac_kLpcMeansShape[poss]; - } - - /* High-band LAR coefficients. */ - for (n = 0; n < LPC_HIBAND_ORDER; n++, pos++, poss++) { - LPCCoef[pos] = tmpcoeffs_s[poss] / LPC_HIBAND_SCALE; - LPCCoef[pos] += WebRtcIsac_kLpcMeansShape[poss]; - } - } - return 0; -} - -/* Encode LPC in LAR domain. */ -void WebRtcIsac_EncodeLar(double* LPCCoef, Bitstr* streamdata, - IsacSaveEncoderData* encData) { - int j, k, n, pos, pos2, poss, offss, offs2; - int index_s[KLT_ORDER_SHAPE]; - int index_ovr_s[KLT_ORDER_SHAPE]; - double tmpcoeffs_s[KLT_ORDER_SHAPE]; - double tmpcoeffs2_s[KLT_ORDER_SHAPE]; - double sum; - const int kModel = 0; - - /* Mean removal and scaling. */ - poss = 0; - pos = 0; - for (k = 0; k < SUBFRAMES; k++) { - /* First two element are gains, move over them. */ - pos += 2; - - /* Low-band LAR coefficients. */ - for (n = 0; n < LPC_LOBAND_ORDER; n++, poss++, pos++) { - tmpcoeffs_s[poss] = LPCCoef[pos] - WebRtcIsac_kLpcMeansShape[poss]; - tmpcoeffs_s[poss] *= LPC_LOBAND_SCALE; - } - - /* High-band LAR coefficients. */ - for (n = 0; n < LPC_HIBAND_ORDER; n++, poss++, pos++) { - tmpcoeffs_s[poss] = LPCCoef[pos] - WebRtcIsac_kLpcMeansShape[poss]; - tmpcoeffs_s[poss] *= LPC_HIBAND_SCALE; - } - } - - /* KLT */ - - /* Left transform. */ - offss = 0; - for (j = 0; j < SUBFRAMES; j++) { - poss = offss; - for (k = 0; k < LPC_SHAPE_ORDER; k++) { - sum = 0; - pos = offss; - pos2 = k; - for (n = 0; n < LPC_SHAPE_ORDER; n++) { - sum += tmpcoeffs_s[pos++] * WebRtcIsac_kKltT1Shape[pos2]; - pos2 += LPC_SHAPE_ORDER; - } - tmpcoeffs2_s[poss++] = sum; - } - offss += LPC_SHAPE_ORDER; - } - - /* Right transform. */ - offss = 0; - offs2 = 0; - for (j = 0; j < SUBFRAMES; j++) { - poss = offss; - for (k = 0; k < LPC_SHAPE_ORDER; k++) { - sum = 0; - pos = k; - pos2 = offs2; - for (n = 0; n < SUBFRAMES; n++) { - sum += tmpcoeffs2_s[pos] * WebRtcIsac_kKltT2Shape[pos2++]; - pos += LPC_SHAPE_ORDER; - } - tmpcoeffs_s[poss++] = sum; - } - offs2 += SUBFRAMES; - offss += LPC_SHAPE_ORDER; - } - - /* Quantize coefficients. */ - for (k = 0; k < KLT_ORDER_SHAPE; k++) { - index_s[k] = (WebRtcIsac_lrint(tmpcoeffs_s[k] / KLT_STEPSIZE)) + - WebRtcIsac_kQKltQuantMinShape[k]; - if (index_s[k] < 0) { - index_s[k] = 0; - } else if (index_s[k] > WebRtcIsac_kQKltMaxIndShape[k]) { - index_s[k] = WebRtcIsac_kQKltMaxIndShape[k]; - } - index_ovr_s[k] = WebRtcIsac_kQKltOffsetShape[k] + index_s[k]; - } - - - /* Only one model remains in this version of the code, kModel = 0. We - * are keeping for bit-streams to be backward compatible. */ - /* entropy coding of model number */ - WebRtcIsac_EncHistMulti(streamdata, &kModel, WebRtcIsac_kQKltModelCdfPtr, 1); - - /* Save data for creation of multiple bit streams */ - /* Entropy coding of quantization indices - shape only. */ - WebRtcIsac_EncHistMulti(streamdata, index_s, WebRtcIsac_kQKltCdfPtrShape, - KLT_ORDER_SHAPE); - - /* Save data for creation of multiple bit streams. */ - for (k = 0; k < KLT_ORDER_SHAPE; k++) { - encData->LPCindex_s[KLT_ORDER_SHAPE * encData->startIdx + k] = index_s[k]; - } - - /* Find quantization levels for shape coefficients. */ - for (k = 0; k < KLT_ORDER_SHAPE; k++) { - tmpcoeffs_s[k] = WebRtcIsac_kQKltLevelsShape[index_ovr_s[k]]; - } - /* Inverse KLT. */ - /* Left transform, transpose matrix.! */ - offss = 0; - poss = 0; - for (j = 0; j < SUBFRAMES; j++) { - offs2 = 0; - for (k = 0; k < LPC_SHAPE_ORDER; k++) { - sum = 0; - pos = offss; - pos2 = offs2; - for (n = 0; n < LPC_SHAPE_ORDER; n++) { - sum += tmpcoeffs_s[pos++] * WebRtcIsac_kKltT1Shape[pos2++]; - } - tmpcoeffs2_s[poss++] = sum; - offs2 += LPC_SHAPE_ORDER; - } - offss += LPC_SHAPE_ORDER; - } - - /* Right transform, Transpose matrix */ - offss = 0; - poss = 0; - for (j = 0; j < SUBFRAMES; j++) { - poss = offss; - for (k = 0; k < LPC_SHAPE_ORDER; k++) { - sum = 0; - pos = k; - pos2 = j; - for (n = 0; n < SUBFRAMES; n++) { - sum += tmpcoeffs2_s[pos] * WebRtcIsac_kKltT2Shape[pos2]; - pos += LPC_SHAPE_ORDER; - pos2 += SUBFRAMES; - } - tmpcoeffs_s[poss++] = sum; - } - offss += LPC_SHAPE_ORDER; - } - - /* Scaling, mean addition, and gain restoration. */ - poss = 0; - pos = 0; - for (k = 0; k < SUBFRAMES; k++) { - /* Ignore gains. */ - pos += 2; - - /* Low band LAR coefficients. */ - for (n = 0; n < LPC_LOBAND_ORDER; n++, pos++, poss++) { - LPCCoef[pos] = tmpcoeffs_s[poss] / LPC_LOBAND_SCALE; - LPCCoef[pos] += WebRtcIsac_kLpcMeansShape[poss]; - } - - /* High band LAR coefficients. */ - for (n = 0; n < LPC_HIBAND_ORDER; n++, pos++, poss++) { - LPCCoef[pos] = tmpcoeffs_s[poss] / LPC_HIBAND_SCALE; - LPCCoef[pos] += WebRtcIsac_kLpcMeansShape[poss]; - } - } -} - - -void WebRtcIsac_EncodeLpcLb(double* LPCCoef_lo, double* LPCCoef_hi, - Bitstr* streamdata, IsacSaveEncoderData* encData) { - double lars[KLT_ORDER_GAIN + KLT_ORDER_SHAPE]; - int k; - - WebRtcIsac_Poly2Lar(LPCCoef_lo, ORDERLO, LPCCoef_hi, ORDERHI, SUBFRAMES, - lars); - WebRtcIsac_EncodeLar(lars, streamdata, encData); - WebRtcIsac_Lar2Poly(lars, LPCCoef_lo, ORDERLO, LPCCoef_hi, ORDERHI, - SUBFRAMES); - /* Save data for creation of multiple bit streams (and transcoding). */ - for (k = 0; k < (ORDERLO + 1)*SUBFRAMES; k++) { - encData->LPCcoeffs_lo[(ORDERLO + 1)*SUBFRAMES * encData->startIdx + k] = - LPCCoef_lo[k]; - } - for (k = 0; k < (ORDERHI + 1)*SUBFRAMES; k++) { - encData->LPCcoeffs_hi[(ORDERHI + 1)*SUBFRAMES * encData->startIdx + k] = - LPCCoef_hi[k]; - } -} - - -int16_t WebRtcIsac_EncodeLpcUB(double* lpcVecs, Bitstr* streamdata, - double* interpolLPCCoeff, - int16_t bandwidth, - ISACUBSaveEncDataStruct* encData) { - double U[UB_LPC_ORDER * UB16_LPC_VEC_PER_FRAME]; - int idx[UB_LPC_ORDER * UB16_LPC_VEC_PER_FRAME]; - int interpolCntr; - - WebRtcIsac_Poly2LarUB(lpcVecs, bandwidth); - WebRtcIsac_RemoveLarMean(lpcVecs, bandwidth); - WebRtcIsac_DecorrelateIntraVec(lpcVecs, U, bandwidth); - WebRtcIsac_DecorrelateInterVec(U, lpcVecs, bandwidth); - WebRtcIsac_QuantizeUncorrLar(lpcVecs, idx, bandwidth); - - WebRtcIsac_CorrelateInterVec(lpcVecs, U, bandwidth); - WebRtcIsac_CorrelateIntraVec(U, lpcVecs, bandwidth); - WebRtcIsac_AddLarMean(lpcVecs, bandwidth); - - switch (bandwidth) { - case isac12kHz: { - /* Store the indices to be used for multiple encoding. */ - memcpy(encData->indexLPCShape, idx, UB_LPC_ORDER * - UB_LPC_VEC_PER_FRAME * sizeof(int)); - WebRtcIsac_EncHistMulti(streamdata, idx, WebRtcIsac_kLpcShapeCdfMatUb12, - UB_LPC_ORDER * UB_LPC_VEC_PER_FRAME); - for (interpolCntr = 0; interpolCntr < UB_INTERPOL_SEGMENTS; - interpolCntr++) { - WebRtcIsac_Lar2PolyInterpolUB(lpcVecs, interpolLPCCoeff, - kLpcVecPerSegmentUb12 + 1); - lpcVecs += UB_LPC_ORDER; - interpolLPCCoeff += (kLpcVecPerSegmentUb12 * (UB_LPC_ORDER + 1)); - } - break; - } - case isac16kHz: { - /* Store the indices to be used for multiple encoding. */ - memcpy(encData->indexLPCShape, idx, UB_LPC_ORDER * - UB16_LPC_VEC_PER_FRAME * sizeof(int)); - WebRtcIsac_EncHistMulti(streamdata, idx, WebRtcIsac_kLpcShapeCdfMatUb16, - UB_LPC_ORDER * UB16_LPC_VEC_PER_FRAME); - for (interpolCntr = 0; interpolCntr < UB16_INTERPOL_SEGMENTS; - interpolCntr++) { - WebRtcIsac_Lar2PolyInterpolUB(lpcVecs, interpolLPCCoeff, - kLpcVecPerSegmentUb16 + 1); - lpcVecs += UB_LPC_ORDER; - interpolLPCCoeff += (kLpcVecPerSegmentUb16 * (UB_LPC_ORDER + 1)); - } - break; - } - default: - return -1; - } - return 0; -} - -void WebRtcIsac_EncodeLpcGainLb(double* LPCCoef_lo, double* LPCCoef_hi, - Bitstr* streamdata, - IsacSaveEncoderData* encData) { - int j, k, n, pos, pos2, posg, offsg, offs2; - int index_g[KLT_ORDER_GAIN]; - int index_ovr_g[KLT_ORDER_GAIN]; - double tmpcoeffs_g[KLT_ORDER_GAIN]; - double tmpcoeffs2_g[KLT_ORDER_GAIN]; - double sum; - /* log gains, mean removal and scaling */ - posg = 0; - for (k = 0; k < SUBFRAMES; k++) { - tmpcoeffs_g[posg] = log(LPCCoef_lo[(LPC_LOBAND_ORDER + 1) * k]); - tmpcoeffs_g[posg] -= WebRtcIsac_kLpcMeansGain[posg]; - tmpcoeffs_g[posg] *= LPC_GAIN_SCALE; - posg++; - tmpcoeffs_g[posg] = log(LPCCoef_hi[(LPC_HIBAND_ORDER + 1) * k]); - tmpcoeffs_g[posg] -= WebRtcIsac_kLpcMeansGain[posg]; - tmpcoeffs_g[posg] *= LPC_GAIN_SCALE; - posg++; - } - - /* KLT */ - - /* Left transform. */ - offsg = 0; - for (j = 0; j < SUBFRAMES; j++) { - posg = offsg; - for (k = 0; k < LPC_GAIN_ORDER; k++) { - sum = 0; - pos = offsg; - pos2 = k; - for (n = 0; n < LPC_GAIN_ORDER; n++) { - sum += tmpcoeffs_g[pos++] * WebRtcIsac_kKltT1Gain[pos2]; - pos2 += LPC_GAIN_ORDER; - } - tmpcoeffs2_g[posg++] = sum; - } - offsg += LPC_GAIN_ORDER; - } - - /* Right transform. */ - offsg = 0; - offs2 = 0; - for (j = 0; j < SUBFRAMES; j++) { - posg = offsg; - for (k = 0; k < LPC_GAIN_ORDER; k++) { - sum = 0; - pos = k; - pos2 = offs2; - for (n = 0; n < SUBFRAMES; n++) { - sum += tmpcoeffs2_g[pos] * WebRtcIsac_kKltT2Gain[pos2++]; - pos += LPC_GAIN_ORDER; - } - tmpcoeffs_g[posg++] = sum; - } - offs2 += SUBFRAMES; - offsg += LPC_GAIN_ORDER; - } - - /* Quantize coefficients. */ - for (k = 0; k < KLT_ORDER_GAIN; k++) { - /* Get index. */ - pos2 = WebRtcIsac_lrint(tmpcoeffs_g[k] / KLT_STEPSIZE); - index_g[k] = (pos2) + WebRtcIsac_kQKltQuantMinGain[k]; - if (index_g[k] < 0) { - index_g[k] = 0; - } else if (index_g[k] > WebRtcIsac_kQKltMaxIndGain[k]) { - index_g[k] = WebRtcIsac_kQKltMaxIndGain[k]; - } - index_ovr_g[k] = WebRtcIsac_kQKltOffsetGain[k] + index_g[k]; - - /* Find quantization levels for coefficients. */ - tmpcoeffs_g[k] = WebRtcIsac_kQKltLevelsGain[index_ovr_g[k]]; - - /* Save data for creation of multiple bit streams. */ - encData->LPCindex_g[KLT_ORDER_GAIN * encData->startIdx + k] = index_g[k]; - } - - /* Entropy coding of quantization indices - gain. */ - WebRtcIsac_EncHistMulti(streamdata, index_g, WebRtcIsac_kQKltCdfPtrGain, - KLT_ORDER_GAIN); - - /* Find quantization levels for coefficients. */ - /* Left transform. */ - offsg = 0; - posg = 0; - for (j = 0; j < SUBFRAMES; j++) { - offs2 = 0; - for (k = 0; k < LPC_GAIN_ORDER; k++) { - sum = 0; - pos = offsg; - pos2 = offs2; - for (n = 0; n < LPC_GAIN_ORDER; n++) - sum += tmpcoeffs_g[pos++] * WebRtcIsac_kKltT1Gain[pos2++]; - tmpcoeffs2_g[posg++] = sum; - offs2 += LPC_GAIN_ORDER; - } - offsg += LPC_GAIN_ORDER; - } - - /* Right transform, transpose matrix. */ - offsg = 0; - posg = 0; - for (j = 0; j < SUBFRAMES; j++) { - posg = offsg; - for (k = 0; k < LPC_GAIN_ORDER; k++) { - sum = 0; - pos = k; - pos2 = j; - for (n = 0; n < SUBFRAMES; n++) { - sum += tmpcoeffs2_g[pos] * WebRtcIsac_kKltT2Gain[pos2]; - pos += LPC_GAIN_ORDER; - pos2 += SUBFRAMES; - } - tmpcoeffs_g[posg++] = sum; - } - offsg += LPC_GAIN_ORDER; - } - - - /* Scaling, mean addition, and gain restoration. */ - posg = 0; - for (k = 0; k < SUBFRAMES; k++) { - sum = tmpcoeffs_g[posg] / LPC_GAIN_SCALE; - sum += WebRtcIsac_kLpcMeansGain[posg]; - LPCCoef_lo[k * (LPC_LOBAND_ORDER + 1)] = exp(sum); - pos++; - posg++; - sum = tmpcoeffs_g[posg] / LPC_GAIN_SCALE; - sum += WebRtcIsac_kLpcMeansGain[posg]; - LPCCoef_hi[k * (LPC_HIBAND_ORDER + 1)] = exp(sum); - pos++; - posg++; - } - -} - -void WebRtcIsac_EncodeLpcGainUb(double* lpGains, Bitstr* streamdata, - int* lpcGainIndex) { - double U[UB_LPC_GAIN_DIM]; - int idx[UB_LPC_GAIN_DIM]; - WebRtcIsac_ToLogDomainRemoveMean(lpGains); - WebRtcIsac_DecorrelateLPGain(lpGains, U); - WebRtcIsac_QuantizeLpcGain(U, idx); - /* Store the index for re-encoding for FEC. */ - memcpy(lpcGainIndex, idx, UB_LPC_GAIN_DIM * sizeof(int)); - WebRtcIsac_CorrelateLpcGain(U, lpGains); - WebRtcIsac_AddMeanToLinearDomain(lpGains); - WebRtcIsac_EncHistMulti(streamdata, idx, WebRtcIsac_kLpcGainCdfMat, - UB_LPC_GAIN_DIM); -} - - -void WebRtcIsac_StoreLpcGainUb(double* lpGains, Bitstr* streamdata) { - double U[UB_LPC_GAIN_DIM]; - int idx[UB_LPC_GAIN_DIM]; - WebRtcIsac_ToLogDomainRemoveMean(lpGains); - WebRtcIsac_DecorrelateLPGain(lpGains, U); - WebRtcIsac_QuantizeLpcGain(U, idx); - WebRtcIsac_EncHistMulti(streamdata, idx, WebRtcIsac_kLpcGainCdfMat, - UB_LPC_GAIN_DIM); -} - - - -int16_t WebRtcIsac_DecodeLpcGainUb(double* lpGains, Bitstr* streamdata) { - double U[UB_LPC_GAIN_DIM]; - int idx[UB_LPC_GAIN_DIM]; - int err; - err = WebRtcIsac_DecHistOneStepMulti(idx, streamdata, - WebRtcIsac_kLpcGainCdfMat, - WebRtcIsac_kLpcGainEntropySearch, - UB_LPC_GAIN_DIM); - if (err < 0) { - return -1; - } - WebRtcIsac_DequantizeLpcGain(idx, U); - WebRtcIsac_CorrelateLpcGain(U, lpGains); - WebRtcIsac_AddMeanToLinearDomain(lpGains); - return 0; -} - - - -/* decode & dequantize RC */ -int WebRtcIsac_DecodeRc(Bitstr* streamdata, int16_t* RCQ15) { - int k, err; - int index[AR_ORDER]; - - /* entropy decoding of quantization indices */ - err = WebRtcIsac_DecHistOneStepMulti(index, streamdata, - WebRtcIsac_kQArRcCdfPtr, - WebRtcIsac_kQArRcInitIndex, AR_ORDER); - if (err < 0) - return err; - - /* find quantization levels for reflection coefficients */ - for (k = 0; k < AR_ORDER; k++) { - RCQ15[k] = *(WebRtcIsac_kQArRcLevelsPtr[k] + index[k]); - } - return 0; -} - - -/* quantize & code RC */ -void WebRtcIsac_EncodeRc(int16_t* RCQ15, Bitstr* streamdata) { - int k; - int index[AR_ORDER]; - - /* quantize reflection coefficients (add noise feedback?) */ - for (k = 0; k < AR_ORDER; k++) { - index[k] = WebRtcIsac_kQArRcInitIndex[k]; - // The safe-guards in following while conditions are to suppress gcc 4.8.3 - // warnings, Issue 2888. Otherwise, first and last elements of - // `WebRtcIsac_kQArBoundaryLevels` are such that the following search - // *never* cause an out-of-boundary read. - if (RCQ15[k] > WebRtcIsac_kQArBoundaryLevels[index[k]]) { - while (index[k] + 1 < NUM_AR_RC_QUANT_BAUNDARY && - RCQ15[k] > WebRtcIsac_kQArBoundaryLevels[index[k] + 1]) { - index[k]++; - } - } else { - while (index[k] > 0 && - RCQ15[k] < WebRtcIsac_kQArBoundaryLevels[--index[k]]) ; - } - RCQ15[k] = *(WebRtcIsac_kQArRcLevelsPtr[k] + index[k]); - } - - /* entropy coding of quantization indices */ - WebRtcIsac_EncHistMulti(streamdata, index, WebRtcIsac_kQArRcCdfPtr, AR_ORDER); -} - - -/* decode & dequantize squared Gain */ -int WebRtcIsac_DecodeGain2(Bitstr* streamdata, int32_t* gainQ10) { - int index, err; - - /* entropy decoding of quantization index */ - err = WebRtcIsac_DecHistOneStepMulti(&index, streamdata, - WebRtcIsac_kQGainCdf_ptr, - WebRtcIsac_kQGainInitIndex, 1); - if (err < 0) { - return err; - } - /* find quantization level */ - *gainQ10 = WebRtcIsac_kQGain2Levels[index]; - return 0; -} - - -/* quantize & code squared Gain */ -int WebRtcIsac_EncodeGain2(int32_t* gainQ10, Bitstr* streamdata) { - int index; - - /* find quantization index */ - index = WebRtcIsac_kQGainInitIndex[0]; - if (*gainQ10 > WebRtcIsac_kQGain2BoundaryLevels[index]) { - while (*gainQ10 > WebRtcIsac_kQGain2BoundaryLevels[index + 1]) { - index++; - } - } else { - while (*gainQ10 < WebRtcIsac_kQGain2BoundaryLevels[--index]) ; - } - /* De-quantize */ - *gainQ10 = WebRtcIsac_kQGain2Levels[index]; - - /* entropy coding of quantization index */ - WebRtcIsac_EncHistMulti(streamdata, &index, WebRtcIsac_kQGainCdf_ptr, 1); - return 0; -} - - -/* code and decode Pitch Gains and Lags functions */ - -/* decode & dequantize Pitch Gains */ -int WebRtcIsac_DecodePitchGain(Bitstr* streamdata, - int16_t* PitchGains_Q12) { - int index_comb, err; - const uint16_t* WebRtcIsac_kQPitchGainCdf_ptr[1]; - - /* Entropy decoding of quantization indices */ - *WebRtcIsac_kQPitchGainCdf_ptr = WebRtcIsac_kQPitchGainCdf; - err = WebRtcIsac_DecHistBisectMulti(&index_comb, streamdata, - WebRtcIsac_kQPitchGainCdf_ptr, - WebRtcIsac_kQCdfTableSizeGain, 1); - /* Error check, Q_mean_Gain.. tables are of size 144 */ - if ((err < 0) || (index_comb < 0) || (index_comb >= 144)) { - return -ISAC_RANGE_ERROR_DECODE_PITCH_GAIN; - } - /* De-quantize back to pitch gains by table look-up. */ - PitchGains_Q12[0] = WebRtcIsac_kQMeanGain1Q12[index_comb]; - PitchGains_Q12[1] = WebRtcIsac_kQMeanGain2Q12[index_comb]; - PitchGains_Q12[2] = WebRtcIsac_kQMeanGain3Q12[index_comb]; - PitchGains_Q12[3] = WebRtcIsac_kQMeanGain4Q12[index_comb]; - return 0; -} - - -/* Quantize & code Pitch Gains. */ -void WebRtcIsac_EncodePitchGain(int16_t* PitchGains_Q12, - Bitstr* streamdata, - IsacSaveEncoderData* encData) { - int k, j; - double C; - double S[PITCH_SUBFRAMES]; - int index[3]; - int index_comb; - const uint16_t* WebRtcIsac_kQPitchGainCdf_ptr[1]; - double PitchGains[PITCH_SUBFRAMES] = {0, 0, 0, 0}; - - /* Take the asin. */ - for (k = 0; k < PITCH_SUBFRAMES; k++) { - PitchGains[k] = ((float)PitchGains_Q12[k]) / 4096; - S[k] = asin(PitchGains[k]); - } - - /* Find quantization index; only for the first three - * transform coefficients. */ - for (k = 0; k < 3; k++) { - /* transform */ - C = 0.0; - for (j = 0; j < PITCH_SUBFRAMES; j++) { - C += WebRtcIsac_kTransform[k][j] * S[j]; - } - /* Quantize */ - index[k] = WebRtcIsac_lrint(C / PITCH_GAIN_STEPSIZE); - - /* Check that the index is not outside the boundaries of the table. */ - if (index[k] < WebRtcIsac_kIndexLowerLimitGain[k]) { - index[k] = WebRtcIsac_kIndexLowerLimitGain[k]; - } else if (index[k] > WebRtcIsac_kIndexUpperLimitGain[k]) { - index[k] = WebRtcIsac_kIndexUpperLimitGain[k]; - } - index[k] -= WebRtcIsac_kIndexLowerLimitGain[k]; - } - - /* Calculate unique overall index. */ - index_comb = WebRtcIsac_kIndexMultsGain[0] * index[0] + - WebRtcIsac_kIndexMultsGain[1] * index[1] + index[2]; - - /* unquantize back to pitch gains by table look-up */ - PitchGains_Q12[0] = WebRtcIsac_kQMeanGain1Q12[index_comb]; - PitchGains_Q12[1] = WebRtcIsac_kQMeanGain2Q12[index_comb]; - PitchGains_Q12[2] = WebRtcIsac_kQMeanGain3Q12[index_comb]; - PitchGains_Q12[3] = WebRtcIsac_kQMeanGain4Q12[index_comb]; - - /* entropy coding of quantization pitch gains */ - *WebRtcIsac_kQPitchGainCdf_ptr = WebRtcIsac_kQPitchGainCdf; - WebRtcIsac_EncHistMulti(streamdata, &index_comb, - WebRtcIsac_kQPitchGainCdf_ptr, 1); - encData->pitchGain_index[encData->startIdx] = index_comb; -} - - - -/* Pitch LAG */ -/* Decode & de-quantize Pitch Lags. */ -int WebRtcIsac_DecodePitchLag(Bitstr* streamdata, int16_t* PitchGain_Q12, - double* PitchLags) { - int k, err; - double StepSize; - double C; - int index[PITCH_SUBFRAMES]; - double mean_gain; - const double* mean_val2, *mean_val3, *mean_val4; - const int16_t* lower_limit; - const uint16_t* init_index; - const uint16_t* cdf_size; - const uint16_t** cdf; - double PitchGain[4] = {0, 0, 0, 0}; - - /* compute mean pitch gain */ - mean_gain = 0.0; - for (k = 0; k < 4; k++) { - PitchGain[k] = ((float)PitchGain_Q12[k]) / 4096; - mean_gain += PitchGain[k]; - } - mean_gain /= 4.0; - - /* voicing classification. */ - if (mean_gain < 0.2) { - StepSize = WebRtcIsac_kQPitchLagStepsizeLo; - cdf = WebRtcIsac_kQPitchLagCdfPtrLo; - cdf_size = WebRtcIsac_kQPitchLagCdfSizeLo; - mean_val2 = WebRtcIsac_kQMeanLag2Lo; - mean_val3 = WebRtcIsac_kQMeanLag3Lo; - mean_val4 = WebRtcIsac_kQMeanLag4Lo; - lower_limit = WebRtcIsac_kQIndexLowerLimitLagLo; - init_index = WebRtcIsac_kQInitIndexLagLo; - } else if (mean_gain < 0.4) { - StepSize = WebRtcIsac_kQPitchLagStepsizeMid; - cdf = WebRtcIsac_kQPitchLagCdfPtrMid; - cdf_size = WebRtcIsac_kQPitchLagCdfSizeMid; - mean_val2 = WebRtcIsac_kQMeanLag2Mid; - mean_val3 = WebRtcIsac_kQMeanLag3Mid; - mean_val4 = WebRtcIsac_kQMeanLag4Mid; - lower_limit = WebRtcIsac_kQIndexLowerLimitLagMid; - init_index = WebRtcIsac_kQInitIndexLagMid; - } else { - StepSize = WebRtcIsac_kQPitchLagStepsizeHi; - cdf = WebRtcIsac_kQPitchLagCdfPtrHi; - cdf_size = WebRtcIsac_kQPitchLagCdfSizeHi; - mean_val2 = WebRtcIsac_kQMeanLag2Hi; - mean_val3 = WebRtcIsac_kQMeanLag3Hi; - mean_val4 = WebRtcIsac_kQMeanLag4Hi; - lower_limit = WebRtcIsac_kQindexLowerLimitLagHi; - init_index = WebRtcIsac_kQInitIndexLagHi; - } - - /* Entropy decoding of quantization indices. */ - err = WebRtcIsac_DecHistBisectMulti(index, streamdata, cdf, cdf_size, 1); - if ((err < 0) || (index[0] < 0)) { - return -ISAC_RANGE_ERROR_DECODE_PITCH_LAG; - } - err = WebRtcIsac_DecHistOneStepMulti(index + 1, streamdata, cdf + 1, - init_index, 3); - if (err < 0) { - return -ISAC_RANGE_ERROR_DECODE_PITCH_LAG; - } - - /* Unquantize back to transform coefficients and do the inverse transform: - * S = T'*C. */ - C = (index[0] + lower_limit[0]) * StepSize; - for (k = 0; k < PITCH_SUBFRAMES; k++) { - PitchLags[k] = WebRtcIsac_kTransformTranspose[k][0] * C; - } - C = mean_val2[index[1]]; - for (k = 0; k < PITCH_SUBFRAMES; k++) { - PitchLags[k] += WebRtcIsac_kTransformTranspose[k][1] * C; - } - C = mean_val3[index[2]]; - for (k = 0; k < PITCH_SUBFRAMES; k++) { - PitchLags[k] += WebRtcIsac_kTransformTranspose[k][2] * C; - } - C = mean_val4[index[3]]; - for (k = 0; k < PITCH_SUBFRAMES; k++) { - PitchLags[k] += WebRtcIsac_kTransformTranspose[k][3] * C; - } - return 0; -} - - - -/* Quantize & code pitch lags. */ -void WebRtcIsac_EncodePitchLag(double* PitchLags, int16_t* PitchGain_Q12, - Bitstr* streamdata, - IsacSaveEncoderData* encData) { - int k, j; - double StepSize; - double C; - int index[PITCH_SUBFRAMES]; - double mean_gain; - const double* mean_val2, *mean_val3, *mean_val4; - const int16_t* lower_limit, *upper_limit; - const uint16_t** cdf; - double PitchGain[4] = {0, 0, 0, 0}; - - /* compute mean pitch gain */ - mean_gain = 0.0; - for (k = 0; k < 4; k++) { - PitchGain[k] = ((float)PitchGain_Q12[k]) / 4096; - mean_gain += PitchGain[k]; - } - mean_gain /= 4.0; - - /* Save data for creation of multiple bit streams */ - encData->meanGain[encData->startIdx] = mean_gain; - - /* Voicing classification. */ - if (mean_gain < 0.2) { - StepSize = WebRtcIsac_kQPitchLagStepsizeLo; - cdf = WebRtcIsac_kQPitchLagCdfPtrLo; - mean_val2 = WebRtcIsac_kQMeanLag2Lo; - mean_val3 = WebRtcIsac_kQMeanLag3Lo; - mean_val4 = WebRtcIsac_kQMeanLag4Lo; - lower_limit = WebRtcIsac_kQIndexLowerLimitLagLo; - upper_limit = WebRtcIsac_kQIndexUpperLimitLagLo; - } else if (mean_gain < 0.4) { - StepSize = WebRtcIsac_kQPitchLagStepsizeMid; - cdf = WebRtcIsac_kQPitchLagCdfPtrMid; - mean_val2 = WebRtcIsac_kQMeanLag2Mid; - mean_val3 = WebRtcIsac_kQMeanLag3Mid; - mean_val4 = WebRtcIsac_kQMeanLag4Mid; - lower_limit = WebRtcIsac_kQIndexLowerLimitLagMid; - upper_limit = WebRtcIsac_kQIndexUpperLimitLagMid; - } else { - StepSize = WebRtcIsac_kQPitchLagStepsizeHi; - cdf = WebRtcIsac_kQPitchLagCdfPtrHi; - mean_val2 = WebRtcIsac_kQMeanLag2Hi; - mean_val3 = WebRtcIsac_kQMeanLag3Hi; - mean_val4 = WebRtcIsac_kQMeanLag4Hi; - lower_limit = WebRtcIsac_kQindexLowerLimitLagHi; - upper_limit = WebRtcIsac_kQindexUpperLimitLagHi; - } - - /* find quantization index */ - for (k = 0; k < 4; k++) { - /* transform */ - C = 0.0; - for (j = 0; j < PITCH_SUBFRAMES; j++) { - C += WebRtcIsac_kTransform[k][j] * PitchLags[j]; - } - /* quantize */ - index[k] = WebRtcIsac_lrint(C / StepSize); - - /* check that the index is not outside the boundaries of the table */ - if (index[k] < lower_limit[k]) { - index[k] = lower_limit[k]; - } else if (index[k] > upper_limit[k]) index[k] = upper_limit[k]; { - index[k] -= lower_limit[k]; - } - /* Save data for creation of multiple bit streams */ - encData->pitchIndex[PITCH_SUBFRAMES * encData->startIdx + k] = index[k]; - } - - /* Un-quantize back to transform coefficients and do the inverse transform: - * S = T'*C */ - C = (index[0] + lower_limit[0]) * StepSize; - for (k = 0; k < PITCH_SUBFRAMES; k++) { - PitchLags[k] = WebRtcIsac_kTransformTranspose[k][0] * C; - } - C = mean_val2[index[1]]; - for (k = 0; k < PITCH_SUBFRAMES; k++) { - PitchLags[k] += WebRtcIsac_kTransformTranspose[k][1] * C; - } - C = mean_val3[index[2]]; - for (k = 0; k < PITCH_SUBFRAMES; k++) { - PitchLags[k] += WebRtcIsac_kTransformTranspose[k][2] * C; - } - C = mean_val4[index[3]]; - for (k = 0; k < PITCH_SUBFRAMES; k++) { - PitchLags[k] += WebRtcIsac_kTransformTranspose[k][3] * C; - } - /* entropy coding of quantization pitch lags */ - WebRtcIsac_EncHistMulti(streamdata, index, cdf, PITCH_SUBFRAMES); -} - - - -/* Routines for in-band signaling of bandwidth estimation */ -/* Histograms based on uniform distribution of indices */ -/* Move global variables later! */ - - -/* cdf array for frame length indicator */ -const uint16_t WebRtcIsac_kFrameLengthCdf[4] = { - 0, 21845, 43690, 65535 }; - -/* pointer to cdf array for frame length indicator */ -const uint16_t* WebRtcIsac_kFrameLengthCdf_ptr[1] = { - WebRtcIsac_kFrameLengthCdf }; - -/* initial cdf index for decoder of frame length indicator */ -const uint16_t WebRtcIsac_kFrameLengthInitIndex[1] = { 1 }; - - -int WebRtcIsac_DecodeFrameLen(Bitstr* streamdata, int16_t* framesamples) { - int frame_mode, err; - err = 0; - /* entropy decoding of frame length [1:30ms,2:60ms] */ - err = WebRtcIsac_DecHistOneStepMulti(&frame_mode, streamdata, - WebRtcIsac_kFrameLengthCdf_ptr, - WebRtcIsac_kFrameLengthInitIndex, 1); - if (err < 0) - return -ISAC_RANGE_ERROR_DECODE_FRAME_LENGTH; - - switch (frame_mode) { - case 1: - *framesamples = 480; /* 30ms */ - break; - case 2: - *framesamples = 960; /* 60ms */ - break; - default: - err = -ISAC_DISALLOWED_FRAME_MODE_DECODER; - } - return err; -} - -int WebRtcIsac_EncodeFrameLen(int16_t framesamples, Bitstr* streamdata) { - int frame_mode, status; - - status = 0; - frame_mode = 0; - /* entropy coding of frame length [1:480 samples,2:960 samples] */ - switch (framesamples) { - case 480: - frame_mode = 1; - break; - case 960: - frame_mode = 2; - break; - default: - status = - ISAC_DISALLOWED_FRAME_MODE_ENCODER; - } - - if (status < 0) - return status; - - WebRtcIsac_EncHistMulti(streamdata, &frame_mode, - WebRtcIsac_kFrameLengthCdf_ptr, 1); - return status; -} - -/* cdf array for estimated bandwidth */ -static const uint16_t kBwCdf[25] = { - 0, 2731, 5461, 8192, 10923, 13653, 16384, 19114, 21845, 24576, 27306, 30037, - 32768, 35498, 38229, 40959, 43690, 46421, 49151, 51882, 54613, 57343, 60074, - 62804, 65535 }; - -/* pointer to cdf array for estimated bandwidth */ -static const uint16_t* const kBwCdfPtr[1] = { kBwCdf }; - -/* initial cdf index for decoder of estimated bandwidth*/ -static const uint16_t kBwInitIndex[1] = { 7 }; - - -int WebRtcIsac_DecodeSendBW(Bitstr* streamdata, int16_t* BWno) { - int BWno32, err; - - /* entropy decoding of sender's BW estimation [0..23] */ - err = WebRtcIsac_DecHistOneStepMulti(&BWno32, streamdata, kBwCdfPtr, - kBwInitIndex, 1); - if (err < 0) { - return -ISAC_RANGE_ERROR_DECODE_BANDWIDTH; - } - *BWno = (int16_t)BWno32; - return err; -} - -void WebRtcIsac_EncodeReceiveBw(int* BWno, Bitstr* streamdata) { - /* entropy encoding of receiver's BW estimation [0..23] */ - WebRtcIsac_EncHistMulti(streamdata, BWno, kBwCdfPtr, 1); -} - - -/* estimate code length of LPC Coef */ -void WebRtcIsac_TranscodeLPCCoef(double* LPCCoef_lo, double* LPCCoef_hi, - int* index_g) { - int j, k, n, pos, pos2, posg, offsg, offs2; - int index_ovr_g[KLT_ORDER_GAIN]; - double tmpcoeffs_g[KLT_ORDER_GAIN]; - double tmpcoeffs2_g[KLT_ORDER_GAIN]; - double sum; - - /* log gains, mean removal and scaling */ - posg = 0; - for (k = 0; k < SUBFRAMES; k++) { - tmpcoeffs_g[posg] = log(LPCCoef_lo[(LPC_LOBAND_ORDER + 1) * k]); - tmpcoeffs_g[posg] -= WebRtcIsac_kLpcMeansGain[posg]; - tmpcoeffs_g[posg] *= LPC_GAIN_SCALE; - posg++; - tmpcoeffs_g[posg] = log(LPCCoef_hi[(LPC_HIBAND_ORDER + 1) * k]); - tmpcoeffs_g[posg] -= WebRtcIsac_kLpcMeansGain[posg]; - tmpcoeffs_g[posg] *= LPC_GAIN_SCALE; - posg++; - } - - /* KLT */ - - /* Left transform. */ - offsg = 0; - for (j = 0; j < SUBFRAMES; j++) { - posg = offsg; - for (k = 0; k < LPC_GAIN_ORDER; k++) { - sum = 0; - pos = offsg; - pos2 = k; - for (n = 0; n < LPC_GAIN_ORDER; n++) { - sum += tmpcoeffs_g[pos++] * WebRtcIsac_kKltT1Gain[pos2]; - pos2 += LPC_GAIN_ORDER; - } - tmpcoeffs2_g[posg++] = sum; - } - offsg += LPC_GAIN_ORDER; - } - - /* Right transform. */ - offsg = 0; - offs2 = 0; - for (j = 0; j < SUBFRAMES; j++) { - posg = offsg; - for (k = 0; k < LPC_GAIN_ORDER; k++) { - sum = 0; - pos = k; - pos2 = offs2; - for (n = 0; n < SUBFRAMES; n++) { - sum += tmpcoeffs2_g[pos] * WebRtcIsac_kKltT2Gain[pos2++]; - pos += LPC_GAIN_ORDER; - } - tmpcoeffs_g[posg++] = sum; - } - offs2 += SUBFRAMES; - offsg += LPC_GAIN_ORDER; - } - - - /* quantize coefficients */ - for (k = 0; k < KLT_ORDER_GAIN; k++) { - /* Get index. */ - pos2 = WebRtcIsac_lrint(tmpcoeffs_g[k] / KLT_STEPSIZE); - index_g[k] = (pos2) + WebRtcIsac_kQKltQuantMinGain[k]; - if (index_g[k] < 0) { - index_g[k] = 0; - } else if (index_g[k] > WebRtcIsac_kQKltMaxIndGain[k]) { - index_g[k] = WebRtcIsac_kQKltMaxIndGain[k]; - } - index_ovr_g[k] = WebRtcIsac_kQKltOffsetGain[k] + index_g[k]; - - /* find quantization levels for coefficients */ - tmpcoeffs_g[k] = WebRtcIsac_kQKltLevelsGain[index_ovr_g[k]]; - } -} - - -/* Decode & de-quantize LPC Coefficients. */ -int WebRtcIsac_DecodeLpcCoefUB(Bitstr* streamdata, double* lpcVecs, - double* percepFilterGains, - int16_t bandwidth) { - int index_s[KLT_ORDER_SHAPE]; - - double U[UB_LPC_ORDER * UB16_LPC_VEC_PER_FRAME]; - int err; - - /* Entropy decoding of quantization indices. */ - switch (bandwidth) { - case isac12kHz: { - err = WebRtcIsac_DecHistOneStepMulti( - index_s, streamdata, WebRtcIsac_kLpcShapeCdfMatUb12, - WebRtcIsac_kLpcShapeEntropySearchUb12, UB_LPC_ORDER * - UB_LPC_VEC_PER_FRAME); - break; - } - case isac16kHz: { - err = WebRtcIsac_DecHistOneStepMulti( - index_s, streamdata, WebRtcIsac_kLpcShapeCdfMatUb16, - WebRtcIsac_kLpcShapeEntropySearchUb16, UB_LPC_ORDER * - UB16_LPC_VEC_PER_FRAME); - break; - } - default: - return -1; - } - - if (err < 0) { - return err; - } - - WebRtcIsac_DequantizeLpcParam(index_s, lpcVecs, bandwidth); - WebRtcIsac_CorrelateInterVec(lpcVecs, U, bandwidth); - WebRtcIsac_CorrelateIntraVec(U, lpcVecs, bandwidth); - WebRtcIsac_AddLarMean(lpcVecs, bandwidth); - WebRtcIsac_DecodeLpcGainUb(percepFilterGains, streamdata); - - if (bandwidth == isac16kHz) { - /* Decode another set of Gains. */ - WebRtcIsac_DecodeLpcGainUb(&percepFilterGains[SUBFRAMES], streamdata); - } - return 0; -} - -int16_t WebRtcIsac_EncodeBandwidth(enum ISACBandwidth bandwidth, - Bitstr* streamData) { - int bandwidthMode; - switch (bandwidth) { - case isac12kHz: { - bandwidthMode = 0; - break; - } - case isac16kHz: { - bandwidthMode = 1; - break; - } - default: - return -ISAC_DISALLOWED_ENCODER_BANDWIDTH; - } - WebRtcIsac_EncHistMulti(streamData, &bandwidthMode, kOneBitEqualProbCdf_ptr, - 1); - return 0; -} - -int16_t WebRtcIsac_DecodeBandwidth(Bitstr* streamData, - enum ISACBandwidth* bandwidth) { - int bandwidthMode; - if (WebRtcIsac_DecHistOneStepMulti(&bandwidthMode, streamData, - kOneBitEqualProbCdf_ptr, - kOneBitEqualProbInitIndex, 1) < 0) { - return -ISAC_RANGE_ERROR_DECODE_BANDWITH; - } - switch (bandwidthMode) { - case 0: { - *bandwidth = isac12kHz; - break; - } - case 1: { - *bandwidth = isac16kHz; - break; - } - default: - return -ISAC_DISALLOWED_BANDWIDTH_MODE_DECODER; - } - return 0; -} - -int16_t WebRtcIsac_EncodeJitterInfo(int32_t jitterIndex, - Bitstr* streamData) { - /* This is to avoid LINUX warning until we change 'int' to 'Word32'. */ - int intVar; - - if ((jitterIndex < 0) || (jitterIndex > 1)) { - return -1; - } - intVar = (int)(jitterIndex); - /* Use the same CDF table as for bandwidth - * both take two values with equal probability.*/ - WebRtcIsac_EncHistMulti(streamData, &intVar, kOneBitEqualProbCdf_ptr, 1); - return 0; -} - -int16_t WebRtcIsac_DecodeJitterInfo(Bitstr* streamData, - int32_t* jitterInfo) { - int intVar; - /* Use the same CDF table as for bandwidth - * both take two values with equal probability. */ - if (WebRtcIsac_DecHistOneStepMulti(&intVar, streamData, - kOneBitEqualProbCdf_ptr, - kOneBitEqualProbInitIndex, 1) < 0) { - return -ISAC_RANGE_ERROR_DECODE_BANDWITH; - } - *jitterInfo = (int16_t)(intVar); - return 0; -} diff --git a/modules/audio_coding/codecs/isac/main/source/entropy_coding.h b/modules/audio_coding/codecs/isac/main/source/entropy_coding.h deleted file mode 100644 index 6c2b8d3cc1..0000000000 --- a/modules/audio_coding/codecs/isac/main/source/entropy_coding.h +++ /dev/null @@ -1,347 +0,0 @@ -/* - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/* - * entropy_coding.h - * - * This header file declares all of the functions used to arithmetically - * encode the iSAC bistream - * - */ - -#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_ENTROPY_CODING_H_ -#define MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_ENTROPY_CODING_H_ - -#include "modules/audio_coding/codecs/isac/main/source/settings.h" -#include "modules/audio_coding/codecs/isac/main/source/structs.h" - -/****************************************************************************** - * WebRtcIsac_DecodeSpec() - * Decode real and imaginary part of the DFT coefficients, given a bit-stream. - * The decoded DFT coefficient can be transformed to time domain by - * WebRtcIsac_Time2Spec(). - * - * Input: - * - streamdata : pointer to a stucture containg the encoded - * data and theparameters needed for entropy - * coding. - * - AvgPitchGain_Q12 : average pitch-gain of the frame. This is only - * relevant for 0-4 kHz band, and the input value is - * not used in other bands. - * - band : specifies which band's DFT should be decoded. - * - * Output: - * - *fr : pointer to a buffer where the real part of DFT - * coefficients are written to. - * - *fi : pointer to a buffer where the imaginary part - * of DFT coefficients are written to. - * - * Return value : < 0 if an error occures - * 0 if succeeded. - */ -int WebRtcIsac_DecodeSpec(Bitstr* streamdata, - int16_t AvgPitchGain_Q12, - enum ISACBand band, - double* fr, - double* fi); - -/****************************************************************************** - * WebRtcIsac_EncodeSpec() - * Encode real and imaginary part of the DFT coefficients into the given - * bit-stream. - * - * Input: - * - *fr : pointer to a buffer where the real part of DFT - * coefficients are written to. - * - *fi : pointer to a buffer where the imaginary part - * of DFT coefficients are written to. - * - AvgPitchGain_Q12 : average pitch-gain of the frame. This is only - * relevant for 0-4 kHz band, and the input value is - * not used in other bands. - * - band : specifies which band's DFT should be decoded. - * - * Output: - * - streamdata : pointer to a stucture containg the encoded - * data and theparameters needed for entropy - * coding. - * - * Return value : < 0 if an error occures - * 0 if succeeded. - */ -int WebRtcIsac_EncodeSpec(const int16_t* fr, - const int16_t* fi, - int16_t AvgPitchGain_Q12, - enum ISACBand band, - Bitstr* streamdata); - -/* decode & dequantize LPC Coef */ -int WebRtcIsac_DecodeLpcCoef(Bitstr* streamdata, double* LPCCoef); -int WebRtcIsac_DecodeLpcCoefUB(Bitstr* streamdata, - double* lpcVecs, - double* percepFilterGains, - int16_t bandwidth); - -int WebRtcIsac_DecodeLpc(Bitstr* streamdata, - double* LPCCoef_lo, - double* LPCCoef_hi); - -/* quantize & code LPC Coef */ -void WebRtcIsac_EncodeLpcLb(double* LPCCoef_lo, - double* LPCCoef_hi, - Bitstr* streamdata, - IsacSaveEncoderData* encData); - -void WebRtcIsac_EncodeLpcGainLb(double* LPCCoef_lo, - double* LPCCoef_hi, - Bitstr* streamdata, - IsacSaveEncoderData* encData); - -/****************************************************************************** - * WebRtcIsac_EncodeLpcUB() - * Encode LPC parameters, given as A-polynomial, of upper-band. The encoding - * is performed in LAR domain. - * For the upper-band, we compute and encode LPC of some sub-frames, LPC of - * other sub-frames are computed by linear interpolation, in LAR domain. This - * function performs the interpolation and returns the LPC of all sub-frames. - * - * Inputs: - * - lpcCoef : a buffer containing A-polynomials of sub-frames - * (excluding first coefficient that is 1). - * - bandwidth : specifies if the codec is operating at 0-12 kHz - * or 0-16 kHz mode. - * - * Input/output: - * - streamdata : pointer to a structure containing the encoded - * data and the parameters needed for entropy - * coding. - * - * Output: - * - interpolLPCCoeff : Decoded and interpolated LPC (A-polynomial) - * of all sub-frames. - * If LP analysis is of order K, and there are N - * sub-frames then this is a buffer of size - * (k + 1) * N, each vector starts with the LPC gain - * of the corresponding sub-frame. The LPC gains - * are encoded and inserted after this function is - * called. The first A-coefficient which is 1 is not - * included. - * - * Return value : 0 if encoding is successful, - * <0 if failed to encode. - */ -int16_t WebRtcIsac_EncodeLpcUB(double* lpcCoeff, - Bitstr* streamdata, - double* interpolLPCCoeff, - int16_t bandwidth, - ISACUBSaveEncDataStruct* encData); - -/****************************************************************************** - * WebRtcIsac_DecodeInterpolLpcUb() - * Decode LPC coefficients and interpolate to get the coefficients fo all - * sub-frmaes. - * - * Inputs: - * - bandwidth : spepecifies if the codec is in 0-12 kHz or - * 0-16 kHz mode. - * - * Input/output: - * - streamdata : pointer to a stucture containg the encoded - * data and theparameters needed for entropy - * coding. - * - * Output: - * - percepFilterParam : Decoded and interpolated LPC (A-polynomial) of - * all sub-frames. - * If LP analysis is of order K, and there are N - * sub-frames then this is a buffer of size - * (k + 1) * N, each vector starts with the LPC gain - * of the corresponding sub-frame. The LPC gains - * are encoded and inserted after this function is - * called. The first A-coefficient which is 1 is not - * included. - * - * Return value : 0 if encoding is successful, - * <0 if failed to encode. - */ -int16_t WebRtcIsac_DecodeInterpolLpcUb(Bitstr* streamdata, - double* percepFilterParam, - int16_t bandwidth); - -/* Decode & dequantize RC */ -int WebRtcIsac_DecodeRc(Bitstr* streamdata, int16_t* RCQ15); - -/* Quantize & code RC */ -void WebRtcIsac_EncodeRc(int16_t* RCQ15, Bitstr* streamdata); - -/* Decode & dequantize squared Gain */ -int WebRtcIsac_DecodeGain2(Bitstr* streamdata, int32_t* Gain2); - -/* Quantize & code squared Gain (input is squared gain) */ -int WebRtcIsac_EncodeGain2(int32_t* gain2, Bitstr* streamdata); - -void WebRtcIsac_EncodePitchGain(int16_t* PitchGains_Q12, - Bitstr* streamdata, - IsacSaveEncoderData* encData); - -void WebRtcIsac_EncodePitchLag(double* PitchLags, - int16_t* PitchGain_Q12, - Bitstr* streamdata, - IsacSaveEncoderData* encData); - -int WebRtcIsac_DecodePitchGain(Bitstr* streamdata, int16_t* PitchGain_Q12); -int WebRtcIsac_DecodePitchLag(Bitstr* streamdata, - int16_t* PitchGain_Q12, - double* PitchLag); - -int WebRtcIsac_DecodeFrameLen(Bitstr* streamdata, int16_t* framelength); -int WebRtcIsac_EncodeFrameLen(int16_t framelength, Bitstr* streamdata); -int WebRtcIsac_DecodeSendBW(Bitstr* streamdata, int16_t* BWno); -void WebRtcIsac_EncodeReceiveBw(int* BWno, Bitstr* streamdata); - -/* Step-down */ -void WebRtcIsac_Poly2Rc(double* a, int N, double* RC); - -/* Step-up */ -void WebRtcIsac_Rc2Poly(double* RC, int N, double* a); - -void WebRtcIsac_TranscodeLPCCoef(double* LPCCoef_lo, - double* LPCCoef_hi, - int* index_g); - -/****************************************************************************** - * WebRtcIsac_EncodeLpcGainUb() - * Encode LPC gains of sub-Frames. - * - * Input/outputs: - * - lpGains : a buffer which contains 'SUBFRAME' number of - * LP gains to be encoded. The input values are - * overwritten by the quantized values. - * - streamdata : pointer to a stucture containg the encoded - * data and theparameters needed for entropy - * coding. - * - * Output: - * - lpcGainIndex : quantization indices for lpc gains, these will - * be stored to be used for FEC. - */ -void WebRtcIsac_EncodeLpcGainUb(double* lpGains, - Bitstr* streamdata, - int* lpcGainIndex); - -/****************************************************************************** - * WebRtcIsac_EncodeLpcGainUb() - * Store LPC gains of sub-Frames in 'streamdata'. - * - * Input: - * - lpGains : a buffer which contains 'SUBFRAME' number of - * LP gains to be encoded. - * Input/outputs: - * - streamdata : pointer to a stucture containg the encoded - * data and theparameters needed for entropy - * coding. - * - */ -void WebRtcIsac_StoreLpcGainUb(double* lpGains, Bitstr* streamdata); - -/****************************************************************************** - * WebRtcIsac_DecodeLpcGainUb() - * Decode the LPC gain of sub-frames. - * - * Input/output: - * - streamdata : pointer to a stucture containg the encoded - * data and theparameters needed for entropy - * coding. - * - * Output: - * - lpGains : a buffer where decoded LPC gians will be stored. - * - * Return value : 0 if succeeded. - * <0 if failed. - */ -int16_t WebRtcIsac_DecodeLpcGainUb(double* lpGains, Bitstr* streamdata); - -/****************************************************************************** - * WebRtcIsac_EncodeBandwidth() - * Encode if the bandwidth of encoded audio is 0-12 kHz or 0-16 kHz. - * - * Input: - * - bandwidth : an enumerator specifying if the codec in is - * 0-12 kHz or 0-16 kHz mode. - * - * Input/output: - * - streamdata : pointer to a stucture containg the encoded - * data and theparameters needed for entropy - * coding. - * - * Return value : 0 if succeeded. - * <0 if failed. - */ -int16_t WebRtcIsac_EncodeBandwidth(enum ISACBandwidth bandwidth, - Bitstr* streamData); - -/****************************************************************************** - * WebRtcIsac_DecodeBandwidth() - * Decode the bandwidth of the encoded audio, i.e. if the bandwidth is 0-12 kHz - * or 0-16 kHz. - * - * Input/output: - * - streamdata : pointer to a stucture containg the encoded - * data and theparameters needed for entropy - * coding. - * - * Output: - * - bandwidth : an enumerator specifying if the codec is in - * 0-12 kHz or 0-16 kHz mode. - * - * Return value : 0 if succeeded. - * <0 if failed. - */ -int16_t WebRtcIsac_DecodeBandwidth(Bitstr* streamData, - enum ISACBandwidth* bandwidth); - -/****************************************************************************** - * WebRtcIsac_EncodeJitterInfo() - * Decode the jitter information. - * - * Input/output: - * - streamdata : pointer to a stucture containg the encoded - * data and theparameters needed for entropy - * coding. - * - * Input: - * - jitterInfo : one bit of info specifying if the channel is - * in high/low jitter. Zero indicates low jitter - * and one indicates high jitter. - * - * Return value : 0 if succeeded. - * <0 if failed. - */ -int16_t WebRtcIsac_EncodeJitterInfo(int32_t jitterIndex, Bitstr* streamData); - -/****************************************************************************** - * WebRtcIsac_DecodeJitterInfo() - * Decode the jitter information. - * - * Input/output: - * - streamdata : pointer to a stucture containg the encoded - * data and theparameters needed for entropy - * coding. - * - * Output: - * - jitterInfo : one bit of info specifying if the channel is - * in high/low jitter. Zero indicates low jitter - * and one indicates high jitter. - * - * Return value : 0 if succeeded. - * <0 if failed. - */ -int16_t WebRtcIsac_DecodeJitterInfo(Bitstr* streamData, int32_t* jitterInfo); - -#endif /* MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_ENTROPY_CODING_H_ */ diff --git a/modules/audio_coding/codecs/isac/main/source/filterbanks.c b/modules/audio_coding/codecs/isac/main/source/filterbanks.c deleted file mode 100644 index d57b55022d..0000000000 --- a/modules/audio_coding/codecs/isac/main/source/filterbanks.c +++ /dev/null @@ -1,114 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/* - * filterbanks.c - * - * This file contains function WebRtcIsac_AllPassFilter2Float, - * WebRtcIsac_SplitAndFilter, and WebRtcIsac_FilterAndCombine - * which implement filterbanks that produce decimated lowpass and - * highpass versions of a signal, and performs reconstruction. - * - */ - -#include "modules/audio_coding/codecs/isac/main/source/settings.h" -#include "modules/audio_coding/codecs/isac/main/source/codec.h" -#include "modules/audio_coding/codecs/isac/main/source/isac_vad.h" - -/* Combining */ - -/* HPstcoeff_out_1 = {a1, a2, b1 - b0 * a1, b2 - b0 * a2}; */ -static const float kHpStCoefOut1Float[4] = -{-1.99701049409000f, 0.99714204490000f, 0.01701049409000f, -0.01704204490000f}; - -/* HPstcoeff_out_2 = {a1, a2, b1 - b0 * a1, b2 - b0 * a2}; */ -static const float kHpStCoefOut2Float[4] = -{-1.98645294509837f, 0.98672435560000f, 0.00645294509837f, -0.00662435560000f}; - - -/* Function WebRtcIsac_FilterAndCombine */ -/* This is a decoder function that takes the decimated - length FRAMESAMPLES_HALF input low-pass and - high-pass signals and creates a reconstructed fullband - output signal of length FRAMESAMPLES. WebRtcIsac_FilterAndCombine - is the sibling function of WebRtcIsac_SplitAndFilter */ -/* INPUTS: - inLP: a length FRAMESAMPLES_HALF array of input low-pass - samples. - inHP: a length FRAMESAMPLES_HALF array of input high-pass - samples. - postfiltdata: input data structure containing the filterbank - states from the previous decoding iteration. - OUTPUTS: - Out: a length FRAMESAMPLES array of output reconstructed - samples (fullband) based on the input low-pass and - high-pass signals. - postfiltdata: the input data structure containing the filterbank - states is updated for the next decoding iteration */ -void WebRtcIsac_FilterAndCombineFloat(float *InLP, - float *InHP, - float *Out, - PostFiltBankstr *postfiltdata) -{ - int k; - float tempin_ch1[FRAMESAMPLES+MAX_AR_MODEL_ORDER]; - float tempin_ch2[FRAMESAMPLES+MAX_AR_MODEL_ORDER]; - float ftmp, ftmp2; - - /* Form the polyphase signals*/ - for (k=0;kSTATE_0_UPPER_float); - - /* Now, all-pass filter the new lower channel signal. But since all-pass filter factors - at the decoder are swapped from the ones at the encoder, the 'upper' channel - all-pass filter factors (WebRtcIsac_kUpperApFactorsFloat) are used to filter this new - lower channel signal */ - WebRtcIsac_AllPassFilter2Float(tempin_ch2, WebRtcIsac_kUpperApFactorsFloat, - FRAMESAMPLES_HALF, NUMBEROFCHANNELAPSECTIONS,postfiltdata->STATE_0_LOWER_float); - - - /* Merge outputs to form the full length output signal.*/ - for (k=0;kHPstates1_float[0] + - kHpStCoefOut1Float[3] * postfiltdata->HPstates1_float[1]; - ftmp = Out[k] - kHpStCoefOut1Float[0] * postfiltdata->HPstates1_float[0] - - kHpStCoefOut1Float[1] * postfiltdata->HPstates1_float[1]; - postfiltdata->HPstates1_float[1] = postfiltdata->HPstates1_float[0]; - postfiltdata->HPstates1_float[0] = ftmp; - Out[k] = ftmp2; - } - - for (k=0;kHPstates2_float[0] + - kHpStCoefOut2Float[3] * postfiltdata->HPstates2_float[1]; - ftmp = Out[k] - kHpStCoefOut2Float[0] * postfiltdata->HPstates2_float[0] - - kHpStCoefOut2Float[1] * postfiltdata->HPstates2_float[1]; - postfiltdata->HPstates2_float[1] = postfiltdata->HPstates2_float[0]; - postfiltdata->HPstates2_float[0] = ftmp; - Out[k] = ftmp2; - } -} diff --git a/modules/audio_coding/codecs/isac/main/source/intialize.c b/modules/audio_coding/codecs/isac/main/source/intialize.c deleted file mode 100644 index 5c951f6e9d..0000000000 --- a/modules/audio_coding/codecs/isac/main/source/intialize.c +++ /dev/null @@ -1,72 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/* encode.c - Encoding function for the iSAC coder */ - -#include - -#include "modules/audio_coding/codecs/isac/main/source/structs.h" -#include "modules/audio_coding/codecs/isac/main/source/codec.h" -#include "modules/audio_coding/codecs/isac/main/source/pitch_estimator.h" - -void WebRtcIsac_InitMasking(MaskFiltstr *maskdata) { - - int k; - - for (k = 0; k < WINLEN; k++) { - maskdata->DataBufferLo[k] = 0.0; - maskdata->DataBufferHi[k] = 0.0; - } - for (k = 0; k < ORDERLO+1; k++) { - maskdata->CorrBufLo[k] = 0.0; - maskdata->PreStateLoF[k] = 0.0; - maskdata->PreStateLoG[k] = 0.0; - maskdata->PostStateLoF[k] = 0.0; - maskdata->PostStateLoG[k] = 0.0; - } - for (k = 0; k < ORDERHI+1; k++) { - maskdata->CorrBufHi[k] = 0.0; - maskdata->PreStateHiF[k] = 0.0; - maskdata->PreStateHiG[k] = 0.0; - maskdata->PostStateHiF[k] = 0.0; - maskdata->PostStateHiG[k] = 0.0; - } - - maskdata->OldEnergy = 10.0; - return; -} - -void WebRtcIsac_InitPostFilterbank(PostFiltBankstr *postfiltdata) -{ - int k; - - for (k = 0; k < 2*POSTQORDER; k++) { - postfiltdata->STATE_0_LOWER[k] = 0; - postfiltdata->STATE_0_UPPER[k] = 0; - - postfiltdata->STATE_0_LOWER_float[k] = 0; - postfiltdata->STATE_0_UPPER_float[k] = 0; - } - - /* High pass filter states */ - postfiltdata->HPstates1[0] = 0.0; - postfiltdata->HPstates1[1] = 0.0; - - postfiltdata->HPstates2[0] = 0.0; - postfiltdata->HPstates2[1] = 0.0; - - postfiltdata->HPstates1_float[0] = 0.0f; - postfiltdata->HPstates1_float[1] = 0.0f; - - postfiltdata->HPstates2_float[0] = 0.0f; - postfiltdata->HPstates2_float[1] = 0.0f; - - return; -} diff --git a/modules/audio_coding/codecs/isac/main/source/isac.c b/modules/audio_coding/codecs/isac/main/source/isac.c deleted file mode 100644 index 456f447d9a..0000000000 --- a/modules/audio_coding/codecs/isac/main/source/isac.c +++ /dev/null @@ -1,2307 +0,0 @@ -/* - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/* - * isac.c - * - * This C file contains the functions for the ISAC API - * - */ - -#include "modules/audio_coding/codecs/isac/main/include/isac.h" - -#include -#include -#include -#include - -#include "rtc_base/checks.h" -#include "common_audio/signal_processing/include/signal_processing_library.h" -#include "modules/audio_coding/codecs/isac/main/source/bandwidth_estimator.h" -#include "modules/audio_coding/codecs/isac/main/source/codec.h" -#include "modules/audio_coding/codecs/isac/main/source/crc.h" -#include "modules/audio_coding/codecs/isac/main/source/entropy_coding.h" -#include "modules/audio_coding/codecs/isac/main/source/lpc_shape_swb16_tables.h" -#include "modules/audio_coding/codecs/isac/main/source/os_specific_inline.h" -#include "modules/audio_coding/codecs/isac/main/source/structs.h" -#include "modules/audio_coding/codecs/isac/main/source/isac_vad.h" -#include "rtc_base/system/arch.h" - -#define BIT_MASK_DEC_INIT 0x0001 -#define BIT_MASK_ENC_INIT 0x0002 - -#define LEN_CHECK_SUM_WORD8 4 -#define MAX_NUM_LAYERS 10 - - -/**************************************************************************** - * UpdatePayloadSizeLimit(...) - * - * Call this function to update the limit on the payload size. The limit on - * payload size might change i) if a user ''directly changes the limit by - * calling xxx_setMaxPayloadSize() or xxx_setMaxRate(), or ii) indirectly - * when bandwidth is changing. The latter might be the result of bandwidth - * adaptation, or direct change of the bottleneck in instantaneous mode. - * - * This function takes the current overall limit on payload, and translates it - * to the limits on lower and upper-band. If the codec is in wideband mode, - * then the overall limit and the limit on the lower-band is the same. - * Otherwise, a fraction of the limit should be allocated to lower-band - * leaving some room for the upper-band bit-stream. That is why an update - * of limit is required every time that the bandwidth is changing. - * - */ -static void UpdatePayloadSizeLimit(ISACMainStruct* instISAC) { - int16_t lim30MsPayloadBytes = WEBRTC_SPL_MIN( - (instISAC->maxPayloadSizeBytes), - (instISAC->maxRateBytesPer30Ms)); - int16_t lim60MsPayloadBytes = WEBRTC_SPL_MIN( - (instISAC->maxPayloadSizeBytes), - (instISAC->maxRateBytesPer30Ms << 1)); - - /* The only time that iSAC will have 60 ms - * frame-size is when operating in wideband, so - * there is no upper-band bit-stream. */ - - if (instISAC->bandwidthKHz == isac8kHz) { - /* At 8 kHz there is no upper-band bit-stream, - * therefore, the lower-band limit is the overall limit. */ - instISAC->instLB.ISACencLB_obj.payloadLimitBytes60 = - lim60MsPayloadBytes; - instISAC->instLB.ISACencLB_obj.payloadLimitBytes30 = - lim30MsPayloadBytes; - } else { - /* When in super-wideband, we only have 30 ms frames. - * Do a rate allocation for the given limit. */ - if (lim30MsPayloadBytes > 250) { - /* 4/5 to lower-band the rest for upper-band. */ - instISAC->instLB.ISACencLB_obj.payloadLimitBytes30 = - (lim30MsPayloadBytes << 2) / 5; - } else if (lim30MsPayloadBytes > 200) { - /* For the interval of 200 to 250 the share of - * upper-band linearly grows from 20 to 50. */ - instISAC->instLB.ISACencLB_obj.payloadLimitBytes30 = - (lim30MsPayloadBytes << 1) / 5 + 100; - } else { - /* Allocate only 20 for upper-band. */ - instISAC->instLB.ISACencLB_obj.payloadLimitBytes30 = - lim30MsPayloadBytes - 20; - } - instISAC->instUB.ISACencUB_obj.maxPayloadSizeBytes = - lim30MsPayloadBytes; - } -} - - -/**************************************************************************** - * UpdateBottleneck(...) - * - * This function updates the bottleneck only if the codec is operating in - * channel-adaptive mode. Furthermore, as the update of bottleneck might - * result in an update of bandwidth, therefore, the bottlenech should be - * updated just right before the first 10ms of a frame is pushed into encoder. - * - */ -static void UpdateBottleneck(ISACMainStruct* instISAC) { - /* Read the bottleneck from bandwidth estimator for the - * first 10 ms audio. This way, if there is a change - * in bandwidth, upper and lower-band will be in sync. */ - if ((instISAC->codingMode == 0) && - (instISAC->instLB.ISACencLB_obj.buffer_index == 0) && - (instISAC->instLB.ISACencLB_obj.frame_nb == 0)) { - int32_t bottleneck = - WebRtcIsac_GetUplinkBandwidth(&instISAC->bwestimator_obj); - - /* Adding hysteresis when increasing signal bandwidth. */ - if ((instISAC->bandwidthKHz == isac8kHz) - && (bottleneck > 37000) - && (bottleneck < 41000)) { - bottleneck = 37000; - } - - /* Switching from 12 kHz to 16 kHz is not allowed at this revision. - * If we let this happen, we have to take care of buffer_index and - * the last LPC vector. */ - if ((instISAC->bandwidthKHz != isac16kHz) && - (bottleneck > 46000)) { - bottleneck = 46000; - } - - /* We might need a rate allocation. */ - if (instISAC->encoderSamplingRateKHz == kIsacWideband) { - /* Wideband is the only choice we have here. */ - instISAC->instLB.ISACencLB_obj.bottleneck = - (bottleneck > 32000) ? 32000 : bottleneck; - instISAC->bandwidthKHz = isac8kHz; - } else { - /* Do the rate-allocation and get the new bandwidth. */ - enum ISACBandwidth bandwidth; - WebRtcIsac_RateAllocation(bottleneck, - &(instISAC->instLB.ISACencLB_obj.bottleneck), - &(instISAC->instUB.ISACencUB_obj.bottleneck), - &bandwidth); - if (bandwidth != isac8kHz) { - instISAC->instLB.ISACencLB_obj.new_framelength = 480; - } - if (bandwidth != instISAC->bandwidthKHz) { - /* Bandwidth is changing. */ - instISAC->bandwidthKHz = bandwidth; - UpdatePayloadSizeLimit(instISAC); - if (bandwidth == isac12kHz) { - instISAC->instLB.ISACencLB_obj.buffer_index = 0; - } - /* Currently we don't let the bandwidth to switch to 16 kHz - * if in adaptive mode. If we let this happen, we have to take - * care of buffer_index and the last LPC vector. */ - } - } - } -} - - -/**************************************************************************** - * GetSendBandwidthInfo(...) - * - * This is called to get the bandwidth info. This info is the bandwidth and - * the jitter of 'there-to-here' channel, estimated 'here.' These info - * is signaled in an in-band fashion to the other side. - * - * The call to the bandwidth estimator triggers a recursive averaging which - * has to be synchronized between encoder & decoder, therefore, the call to - * BWE should be once per packet. As the BWE info is inserted into bit-stream - * We need a valid info right before the encodeLB function is going to - * generate a bit-stream. That is when lower-band buffer has already 20ms - * of audio, and the 3rd block of 10ms is going to be injected into encoder. - * - * Inputs: - * - instISAC : iSAC instance. - * - * Outputs: - * - bandwidthIndex : an index which has to be encoded in - * lower-band bit-stream, indicating the - * bandwidth of there-to-here channel. - * - jitterInfo : this indicates if the jitter is high - * or low and it is encoded in upper-band - * bit-stream. - * - */ -static void GetSendBandwidthInfo(ISACMainStruct* instISAC, - int16_t* bandwidthIndex, - int16_t* jitterInfo) { - if ((instISAC->instLB.ISACencLB_obj.buffer_index == - (FRAMESAMPLES_10ms << 1)) && - (instISAC->instLB.ISACencLB_obj.frame_nb == 0)) { - /* Bandwidth estimation and coding. */ - WebRtcIsac_GetDownlinkBwJitIndexImpl(&(instISAC->bwestimator_obj), - bandwidthIndex, jitterInfo, - instISAC->decoderSamplingRateKHz); - } -} - - -/**************************************************************************** - * WebRtcIsac_Create(...) - * - * This function creates an ISAC instance, which will contain the state - * information for one coding/decoding channel. - * - * Input: - * - ISAC_main_inst : address of the pointer to the coder instance. - * - * Return value : 0 - Ok - * -1 - Error - */ -int16_t WebRtcIsac_Create(ISACStruct** ISAC_main_inst) { - ISACMainStruct* instISAC; - - if (ISAC_main_inst != NULL) { - instISAC = (ISACMainStruct*)malloc(sizeof(ISACMainStruct)); - *ISAC_main_inst = (ISACStruct*)instISAC; - if (*ISAC_main_inst != NULL) { - instISAC->errorCode = 0; - instISAC->initFlag = 0; - /* Default is wideband. */ - instISAC->bandwidthKHz = isac8kHz; - instISAC->encoderSamplingRateKHz = kIsacWideband; - instISAC->decoderSamplingRateKHz = kIsacWideband; - instISAC->in_sample_rate_hz = 16000; - - WebRtcIsac_InitTransform(&instISAC->transform_tables); - return 0; - } else { - return -1; - } - } else { - return -1; - } -} - - -/**************************************************************************** - * WebRtcIsac_Free(...) - * - * This function frees the ISAC instance created at the beginning. - * - * Input: - * - ISAC_main_inst : a ISAC instance. - * - * Return value : 0 - Ok - * -1 - Error - */ -int16_t WebRtcIsac_Free(ISACStruct* ISAC_main_inst) { - ISACMainStruct* instISAC = (ISACMainStruct*)ISAC_main_inst; - free(instISAC); - return 0; -} - - -/**************************************************************************** - * EncoderInitLb(...) - internal function for initialization of - * Lower Band - * EncoderInitUb(...) - internal function for initialization of - * Upper Band - * WebRtcIsac_EncoderInit(...) - API function - * - * This function initializes a ISAC instance prior to the encoder calls. - * - * Input: - * - ISAC_main_inst : ISAC instance. - * - CodingMode : 0 -> Bit rate and frame length are automatically - * adjusted to available bandwidth on - * transmission channel, applicable just to - * wideband mode. - * 1 -> User sets a frame length and a target bit - * rate which is taken as the maximum - * short-term average bit rate. - * - * Return value : 0 - Ok - * -1 - Error - */ -static int16_t EncoderInitLb(ISACLBStruct* instLB, - int16_t codingMode, - enum IsacSamplingRate sampRate) { - int16_t statusInit = 0; - int k; - - /* Init stream vector to zero */ - for (k = 0; k < STREAM_SIZE_MAX_60; k++) { - instLB->ISACencLB_obj.bitstr_obj.stream[k] = 0; - } - - if ((codingMode == 1) || (sampRate == kIsacSuperWideband)) { - /* 30 ms frame-size if either in super-wideband or - * instantaneous mode (I-mode). */ - instLB->ISACencLB_obj.new_framelength = 480; - } else { - instLB->ISACencLB_obj.new_framelength = INITIAL_FRAMESAMPLES; - } - - WebRtcIsac_InitMasking(&instLB->ISACencLB_obj.maskfiltstr_obj); - WebRtcIsac_InitPreFilterbank(&instLB->ISACencLB_obj.prefiltbankstr_obj); - WebRtcIsac_InitPitchFilter(&instLB->ISACencLB_obj.pitchfiltstr_obj); - WebRtcIsac_InitPitchAnalysis( - &instLB->ISACencLB_obj.pitchanalysisstr_obj); - - instLB->ISACencLB_obj.buffer_index = 0; - instLB->ISACencLB_obj.frame_nb = 0; - /* Default for I-mode. */ - instLB->ISACencLB_obj.bottleneck = 32000; - instLB->ISACencLB_obj.current_framesamples = 0; - instLB->ISACencLB_obj.s2nr = 0; - instLB->ISACencLB_obj.payloadLimitBytes30 = STREAM_SIZE_MAX_30; - instLB->ISACencLB_obj.payloadLimitBytes60 = STREAM_SIZE_MAX_60; - instLB->ISACencLB_obj.maxPayloadBytes = STREAM_SIZE_MAX_60; - instLB->ISACencLB_obj.maxRateInBytes = STREAM_SIZE_MAX_30; - instLB->ISACencLB_obj.enforceFrameSize = 0; - /* Invalid value prevents getRedPayload to - run before encoder is called. */ - instLB->ISACencLB_obj.lastBWIdx = -1; - return statusInit; -} - -static int16_t EncoderInitUb(ISACUBStruct* instUB, - int16_t bandwidth) { - int16_t statusInit = 0; - int k; - - /* Init stream vector to zero. */ - for (k = 0; k < STREAM_SIZE_MAX_60; k++) { - instUB->ISACencUB_obj.bitstr_obj.stream[k] = 0; - } - - WebRtcIsac_InitMasking(&instUB->ISACencUB_obj.maskfiltstr_obj); - WebRtcIsac_InitPreFilterbank(&instUB->ISACencUB_obj.prefiltbankstr_obj); - - if (bandwidth == isac16kHz) { - instUB->ISACencUB_obj.buffer_index = LB_TOTAL_DELAY_SAMPLES; - } else { - instUB->ISACencUB_obj.buffer_index = 0; - } - /* Default for I-mode. */ - instUB->ISACencUB_obj.bottleneck = 32000; - /* These store the limits for the wideband + super-wideband bit-stream. */ - instUB->ISACencUB_obj.maxPayloadSizeBytes = STREAM_SIZE_MAX_30 << 1; - /* This has to be updated after each lower-band encoding to guarantee - * a correct payload-limitation. */ - instUB->ISACencUB_obj.numBytesUsed = 0; - memset(instUB->ISACencUB_obj.data_buffer_float, 0, - (MAX_FRAMESAMPLES + LB_TOTAL_DELAY_SAMPLES) * sizeof(float)); - - memcpy(&(instUB->ISACencUB_obj.lastLPCVec), - WebRtcIsac_kMeanLarUb16, sizeof(double) * UB_LPC_ORDER); - - return statusInit; -} - - -int16_t WebRtcIsac_EncoderInit(ISACStruct* ISAC_main_inst, - int16_t codingMode) { - ISACMainStruct* instISAC = (ISACMainStruct*)ISAC_main_inst; - int16_t status; - - if ((codingMode != 0) && (codingMode != 1)) { - instISAC->errorCode = ISAC_DISALLOWED_CODING_MODE; - return -1; - } - /* Default bottleneck. */ - instISAC->bottleneck = MAX_ISAC_BW; - - if (instISAC->encoderSamplingRateKHz == kIsacWideband) { - instISAC->bandwidthKHz = isac8kHz; - instISAC->maxPayloadSizeBytes = STREAM_SIZE_MAX_60; - instISAC->maxRateBytesPer30Ms = STREAM_SIZE_MAX_30; - } else { - instISAC->bandwidthKHz = isac16kHz; - instISAC->maxPayloadSizeBytes = STREAM_SIZE_MAX; - instISAC->maxRateBytesPer30Ms = STREAM_SIZE_MAX; - } - - /* Channel-adaptive = 0; Instantaneous (Channel-independent) = 1. */ - instISAC->codingMode = codingMode; - - WebRtcIsac_InitBandwidthEstimator(&instISAC->bwestimator_obj, - instISAC->encoderSamplingRateKHz, - instISAC->decoderSamplingRateKHz); - - WebRtcIsac_InitRateModel(&instISAC->rate_data_obj); - /* Default for I-mode. */ - instISAC->MaxDelay = 10.0; - - status = EncoderInitLb(&instISAC->instLB, codingMode, - instISAC->encoderSamplingRateKHz); - if (status < 0) { - instISAC->errorCode = -status; - return -1; - } - - if (instISAC->encoderSamplingRateKHz == kIsacSuperWideband) { - /* Initialize encoder filter-bank. */ - memset(instISAC->analysisFBState1, 0, - FB_STATE_SIZE_WORD32 * sizeof(int32_t)); - memset(instISAC->analysisFBState2, 0, - FB_STATE_SIZE_WORD32 * sizeof(int32_t)); - - status = EncoderInitUb(&(instISAC->instUB), - instISAC->bandwidthKHz); - if (status < 0) { - instISAC->errorCode = -status; - return -1; - } - } - /* Initialization is successful, set the flag. */ - instISAC->initFlag |= BIT_MASK_ENC_INIT; - return 0; -} - - -/**************************************************************************** - * WebRtcIsac_Encode(...) - * - * This function encodes 10ms frame(s) and inserts it into a package. - * Input speech length has to be 160 samples (10ms). The encoder buffers those - * 10ms frames until it reaches the chosen Framesize (480 or 960 samples - * corresponding to 30 or 60 ms frames), and then proceeds to the encoding. - * - * Input: - * - ISAC_main_inst : ISAC instance. - * - speechIn : input speech vector. - * - * Output: - * - encoded : the encoded data vector - * - * Return value: - * : >0 - Length (in bytes) of coded data - * : 0 - The buffer didn't reach the chosen - * frameSize so it keeps buffering speech - * samples. - * : -1 - Error - */ -int WebRtcIsac_Encode(ISACStruct* ISAC_main_inst, - const int16_t* speechIn, - uint8_t* encoded) { - float inFrame[FRAMESAMPLES_10ms]; - int16_t speechInLB[FRAMESAMPLES_10ms]; - int16_t speechInUB[FRAMESAMPLES_10ms]; - int streamLenLB = 0; - int streamLenUB = 0; - int streamLen = 0; - size_t k = 0; - uint8_t garbageLen = 0; - int32_t bottleneck = 0; - int16_t bottleneckIdx = 0; - int16_t jitterInfo = 0; - - ISACMainStruct* instISAC = (ISACMainStruct*)ISAC_main_inst; - ISACLBStruct* instLB = &(instISAC->instLB); - ISACUBStruct* instUB = &(instISAC->instUB); - - /* Check if encoder initiated. */ - if ((instISAC->initFlag & BIT_MASK_ENC_INIT) != - BIT_MASK_ENC_INIT) { - instISAC->errorCode = ISAC_ENCODER_NOT_INITIATED; - return -1; - } - - if (instISAC->encoderSamplingRateKHz == kIsacSuperWideband) { - WebRtcSpl_AnalysisQMF(speechIn, SWBFRAMESAMPLES_10ms, speechInLB, - speechInUB, instISAC->analysisFBState1, - instISAC->analysisFBState2); - - /* Convert from fixed to floating point. */ - for (k = 0; k < FRAMESAMPLES_10ms; k++) { - inFrame[k] = (float)speechInLB[k]; - } - } else { - for (k = 0; k < FRAMESAMPLES_10ms; k++) { - inFrame[k] = (float) speechIn[k]; - } - } - - /* Add some noise to avoid denormal numbers. */ - inFrame[0] += (float)1.23455334e-3; - inFrame[1] -= (float)2.04324239e-3; - inFrame[2] += (float)1.90854954e-3; - inFrame[9] += (float)1.84854878e-3; - - /* This function will update the bottleneck if required. */ - UpdateBottleneck(instISAC); - - /* Get the bandwith information which has to be sent to the other side. */ - GetSendBandwidthInfo(instISAC, &bottleneckIdx, &jitterInfo); - - /* Encode lower-band. */ - streamLenLB = WebRtcIsac_EncodeLb(&instISAC->transform_tables, - inFrame, &instLB->ISACencLB_obj, - instISAC->codingMode, bottleneckIdx); - if (streamLenLB < 0) { - return -1; - } - - if (instISAC->encoderSamplingRateKHz == kIsacSuperWideband) { - instUB = &(instISAC->instUB); - - /* Convert to float. */ - for (k = 0; k < FRAMESAMPLES_10ms; k++) { - inFrame[k] = (float) speechInUB[k]; - } - - /* Add some noise to avoid denormal numbers. */ - inFrame[0] += (float)1.23455334e-3; - inFrame[1] -= (float)2.04324239e-3; - inFrame[2] += (float)1.90854954e-3; - inFrame[9] += (float)1.84854878e-3; - - /* Tell to upper-band the number of bytes used so far. - * This is for payload limitation. */ - instUB->ISACencUB_obj.numBytesUsed = - (int16_t)(streamLenLB + 1 + LEN_CHECK_SUM_WORD8); - /* Encode upper-band. */ - switch (instISAC->bandwidthKHz) { - case isac12kHz: { - streamLenUB = WebRtcIsac_EncodeUb12(&instISAC->transform_tables, - inFrame, &instUB->ISACencUB_obj, - jitterInfo); - break; - } - case isac16kHz: { - streamLenUB = WebRtcIsac_EncodeUb16(&instISAC->transform_tables, - inFrame, &instUB->ISACencUB_obj, - jitterInfo); - break; - } - case isac8kHz: { - streamLenUB = 0; - break; - } - } - - if ((streamLenUB < 0) && (streamLenUB != -ISAC_PAYLOAD_LARGER_THAN_LIMIT)) { - /* An error has happened but this is not the error due to a - * bit-stream larger than the limit. */ - return -1; - } - - if (streamLenLB == 0) { - return 0; - } - - /* One byte is allocated for the length. According to older decoders - so the length bit-stream plus one byte for size and - LEN_CHECK_SUM_WORD8 for the checksum should be less than or equal - to 255. */ - if ((streamLenUB > (255 - (LEN_CHECK_SUM_WORD8 + 1))) || - (streamLenUB == -ISAC_PAYLOAD_LARGER_THAN_LIMIT)) { - /* We have got a too long bit-stream we skip the upper-band - * bit-stream for this frame. */ - streamLenUB = 0; - } - - memcpy(encoded, instLB->ISACencLB_obj.bitstr_obj.stream, streamLenLB); - streamLen = streamLenLB; - if (streamLenUB > 0) { - encoded[streamLenLB] = (uint8_t)(streamLenUB + 1 + LEN_CHECK_SUM_WORD8); - memcpy(&encoded[streamLenLB + 1], - instUB->ISACencUB_obj.bitstr_obj.stream, - streamLenUB); - streamLen += encoded[streamLenLB]; - } else { - encoded[streamLenLB] = 0; - } - } else { - if (streamLenLB == 0) { - return 0; - } - memcpy(encoded, instLB->ISACencLB_obj.bitstr_obj.stream, streamLenLB); - streamLenUB = 0; - streamLen = streamLenLB; - } - - /* Add Garbage if required. */ - bottleneck = WebRtcIsac_GetUplinkBandwidth(&instISAC->bwestimator_obj); - if (instISAC->codingMode == 0) { - int minBytes; - int limit; - uint8_t* ptrGarbage; - - instISAC->MaxDelay = (double)WebRtcIsac_GetUplinkMaxDelay( - &instISAC->bwestimator_obj); - - /* Update rate model and get minimum number of bytes in this packet. */ - minBytes = WebRtcIsac_GetMinBytes( - &(instISAC->rate_data_obj), streamLen, - instISAC->instLB.ISACencLB_obj.current_framesamples, bottleneck, - instISAC->MaxDelay, instISAC->bandwidthKHz); - - /* Make sure MinBytes does not exceed packet size limit. */ - if (instISAC->bandwidthKHz == isac8kHz) { - if (instLB->ISACencLB_obj.current_framesamples == FRAMESAMPLES) { - limit = instLB->ISACencLB_obj.payloadLimitBytes30; - } else { - limit = instLB->ISACencLB_obj.payloadLimitBytes60; - } - } else { - limit = instUB->ISACencUB_obj.maxPayloadSizeBytes; - } - minBytes = (minBytes > limit) ? limit : minBytes; - - /* Make sure we don't allow more than 255 bytes of garbage data. - * We store the length of the garbage data in 8 bits in the bitstream, - * 255 is the max garbage length we can signal using 8 bits. */ - if ((instISAC->bandwidthKHz == isac8kHz) || - (streamLenUB == 0)) { - ptrGarbage = &encoded[streamLenLB]; - limit = streamLen + 255; - } else { - ptrGarbage = &encoded[streamLenLB + 1 + streamLenUB]; - limit = streamLen + (255 - encoded[streamLenLB]); - } - minBytes = (minBytes > limit) ? limit : minBytes; - - garbageLen = (minBytes > streamLen) ? (uint8_t)(minBytes - streamLen) : 0; - - /* Save data for creation of multiple bit-streams. */ - /* If bit-stream too short then add garbage at the end. */ - if (garbageLen > 0) { - /* Overwrite the garbage area to avoid leaking possibly sensitive data - over the network. This also makes the output deterministic. */ - memset(ptrGarbage, 0, garbageLen); - - /* For a correct length of the upper-band bit-stream together - * with the garbage. Garbage is embeded in upper-band bit-stream. - * That is the only way to preserve backward compatibility. */ - if ((instISAC->bandwidthKHz == isac8kHz) || - (streamLenUB == 0)) { - encoded[streamLenLB] = garbageLen; - } else { - encoded[streamLenLB] += garbageLen; - /* Write the length of the garbage at the end of the upper-band - * bit-stream, if exists. This helps for sanity check. */ - encoded[streamLenLB + 1 + streamLenUB] = garbageLen; - - } - streamLen += garbageLen; - } - } else { - /* update rate model */ - WebRtcIsac_UpdateRateModel( - &instISAC->rate_data_obj, streamLen, - instISAC->instLB.ISACencLB_obj.current_framesamples, bottleneck); - garbageLen = 0; - } - - /* Generate CRC if required. */ - if ((instISAC->bandwidthKHz != isac8kHz) && (streamLenUB > 0)) { - uint32_t crc; - - WebRtcIsac_GetCrc((int16_t*)(&(encoded[streamLenLB + 1])), - streamLenUB + garbageLen, &crc); -#ifndef WEBRTC_ARCH_BIG_ENDIAN - for (k = 0; k < LEN_CHECK_SUM_WORD8; k++) { - encoded[streamLen - LEN_CHECK_SUM_WORD8 + k] = - (uint8_t)(crc >> (24 - k * 8)); - } -#else - memcpy(&encoded[streamLenLB + streamLenUB + 1], &crc, LEN_CHECK_SUM_WORD8); -#endif - } - return streamLen; -} - - -/****************************************************************************** - * WebRtcIsac_GetNewBitStream(...) - * - * This function returns encoded data, with the received bwe-index in the - * stream. If the rate is set to a value less than bottleneck of codec - * the new bistream will be re-encoded with the given target rate. - * It should always return a complete packet, i.e. only called once - * even for 60 msec frames. - * - * NOTE 1! This function does not write in the ISACStruct, it is not allowed. - * NOTE 2! Rates larger than the bottleneck of the codec will be limited - * to the current bottleneck. - * - * Input: - * - ISAC_main_inst : ISAC instance. - * - bweIndex : Index of bandwidth estimate to put in new - * bitstream - * - rate : target rate of the transcoder is bits/sec. - * Valid values are the accepted rate in iSAC, - * i.e. 10000 to 56000. - * - * Output: - * - encoded : The encoded data vector - * - * Return value : >0 - Length (in bytes) of coded data - * -1 - Error or called in SWB mode - * NOTE! No error code is written to - * the struct since it is only allowed to read - * the struct. - */ -int16_t WebRtcIsac_GetNewBitStream(ISACStruct* ISAC_main_inst, - int16_t bweIndex, - int16_t jitterInfo, - int32_t rate, - uint8_t* encoded, - int16_t isRCU) { - Bitstr iSACBitStreamInst; /* Local struct for bitstream handling */ - int16_t streamLenLB; - int16_t streamLenUB; - int16_t totalStreamLen; - double gain2; - double gain1; - float scale; - enum ISACBandwidth bandwidthKHz; - double rateLB; - double rateUB; - int32_t currentBN; - uint32_t crc; -#ifndef WEBRTC_ARCH_BIG_ENDIAN - int16_t k; -#endif - ISACMainStruct* instISAC = (ISACMainStruct*)ISAC_main_inst; - - if ((instISAC->initFlag & BIT_MASK_ENC_INIT) != - BIT_MASK_ENC_INIT) { - return -1; - } - - /* Get the bottleneck of this iSAC and limit the - * given rate to the current bottleneck. */ - WebRtcIsac_GetUplinkBw(ISAC_main_inst, ¤tBN); - if (rate > currentBN) { - rate = currentBN; - } - - if (WebRtcIsac_RateAllocation(rate, &rateLB, &rateUB, &bandwidthKHz) < 0) { - return -1; - } - - /* Cannot transcode from 16 kHz to 12 kHz. */ - if ((bandwidthKHz == isac12kHz) && - (instISAC->bandwidthKHz == isac16kHz)) { - return -1; - } - - /* A gain [dB] for the given rate. */ - gain1 = WebRtcIsac_GetSnr( - rateLB, instISAC->instLB.ISACencLB_obj.current_framesamples); - /* The gain [dB] of this iSAC. */ - gain2 = WebRtcIsac_GetSnr( - instISAC->instLB.ISACencLB_obj.bottleneck, - instISAC->instLB.ISACencLB_obj.current_framesamples); - - /* Scale is the ratio of two gains in normal domain. */ - scale = (float)pow(10, (gain1 - gain2) / 20.0); - /* Change the scale if this is a RCU bit-stream. */ - scale = (isRCU) ? (scale * RCU_TRANSCODING_SCALE) : scale; - - streamLenLB = WebRtcIsac_EncodeStoredDataLb( - &instISAC->instLB.ISACencLB_obj.SaveEnc_obj, - &iSACBitStreamInst, bweIndex, scale); - - if (streamLenLB < 0) { - return -1; - } - - /* Convert from bytes to int16_t. */ - memcpy(encoded, iSACBitStreamInst.stream, streamLenLB); - - if (bandwidthKHz == isac8kHz) { - return streamLenLB; - } - - totalStreamLen = streamLenLB; - /* super-wideband is always at 30ms. - * These gains are in dB. - * Gain for the given rate. */ - gain1 = WebRtcIsac_GetSnr(rateUB, FRAMESAMPLES); - /* Gain of this iSAC */ - gain2 = WebRtcIsac_GetSnr(instISAC->instUB.ISACencUB_obj.bottleneck, - FRAMESAMPLES); - - /* Scale is the ratio of two gains in normal domain. */ - scale = (float)pow(10, (gain1 - gain2) / 20.0); - - /* Change the scale if this is a RCU bit-stream. */ - scale = (isRCU)? (scale * RCU_TRANSCODING_SCALE_UB) : scale; - - streamLenUB = WebRtcIsac_EncodeStoredDataUb( - &(instISAC->instUB.ISACencUB_obj.SaveEnc_obj), - &iSACBitStreamInst, jitterInfo, scale, - instISAC->bandwidthKHz); - - if (streamLenUB < 0) { - return -1; - } - - if (streamLenUB + 1 + LEN_CHECK_SUM_WORD8 > 255) { - return streamLenLB; - } - - totalStreamLen = streamLenLB + streamLenUB + 1 + LEN_CHECK_SUM_WORD8; - encoded[streamLenLB] = streamLenUB + 1 + LEN_CHECK_SUM_WORD8; - - memcpy(&encoded[streamLenLB + 1], iSACBitStreamInst.stream, - streamLenUB); - - WebRtcIsac_GetCrc((int16_t*)(&(encoded[streamLenLB + 1])), - streamLenUB, &crc); -#ifndef WEBRTC_ARCH_BIG_ENDIAN - for (k = 0; k < LEN_CHECK_SUM_WORD8; k++) { - encoded[totalStreamLen - LEN_CHECK_SUM_WORD8 + k] = - (uint8_t)((crc >> (24 - k * 8)) & 0xFF); - } -#else - memcpy(&encoded[streamLenLB + streamLenUB + 1], &crc, - LEN_CHECK_SUM_WORD8); -#endif - return totalStreamLen; -} - - -/**************************************************************************** - * DecoderInitLb(...) - internal function for initialization of - * Lower Band - * DecoderInitUb(...) - internal function for initialization of - * Upper Band - * WebRtcIsac_DecoderInit(...) - API function - * - * This function initializes a ISAC instance prior to the decoder calls. - * - * Input: - * - ISAC_main_inst : ISAC instance. - */ -static void DecoderInitLb(ISACLBStruct* instISAC) { - int i; - /* Initialize stream vector to zero. */ - for (i = 0; i < STREAM_SIZE_MAX_60; i++) { - instISAC->ISACdecLB_obj.bitstr_obj.stream[i] = 0; - } - - WebRtcIsac_InitMasking(&instISAC->ISACdecLB_obj.maskfiltstr_obj); - WebRtcIsac_InitPostFilterbank( - &instISAC->ISACdecLB_obj.postfiltbankstr_obj); - WebRtcIsac_InitPitchFilter(&instISAC->ISACdecLB_obj.pitchfiltstr_obj); -} - -static void DecoderInitUb(ISACUBStruct* instISAC) { - int i; - /* Init stream vector to zero */ - for (i = 0; i < STREAM_SIZE_MAX_60; i++) { - instISAC->ISACdecUB_obj.bitstr_obj.stream[i] = 0; - } - - WebRtcIsac_InitMasking(&instISAC->ISACdecUB_obj.maskfiltstr_obj); - WebRtcIsac_InitPostFilterbank( - &instISAC->ISACdecUB_obj.postfiltbankstr_obj); -} - -void WebRtcIsac_DecoderInit(ISACStruct* ISAC_main_inst) { - ISACMainStruct* instISAC = (ISACMainStruct*)ISAC_main_inst; - - DecoderInitLb(&instISAC->instLB); - if (instISAC->decoderSamplingRateKHz == kIsacSuperWideband) { - memset(instISAC->synthesisFBState1, 0, - FB_STATE_SIZE_WORD32 * sizeof(int32_t)); - memset(instISAC->synthesisFBState2, 0, - FB_STATE_SIZE_WORD32 * sizeof(int32_t)); - DecoderInitUb(&(instISAC->instUB)); - } - if ((instISAC->initFlag & BIT_MASK_ENC_INIT) != BIT_MASK_ENC_INIT) { - WebRtcIsac_InitBandwidthEstimator(&instISAC->bwestimator_obj, - instISAC->encoderSamplingRateKHz, - instISAC->decoderSamplingRateKHz); - } - instISAC->initFlag |= BIT_MASK_DEC_INIT; - instISAC->resetFlag_8kHz = 0; -} - - -/**************************************************************************** - * WebRtcIsac_UpdateBwEstimate(...) - * - * This function updates the estimate of the bandwidth. - * - * NOTE: - * The estimates of bandwidth is not valid if the sample rate of the far-end - * encoder is set to 48 kHz and send timestamps are increamented according to - * 48 kHz sampling rate. - * - * Input: - * - ISAC_main_inst : ISAC instance. - * - encoded : encoded ISAC frame(s). - * - packet_size : size of the packet. - * - rtp_seq_number : the RTP number of the packet. - * - arr_ts : the arrival time of the packet (from NetEq) - * in samples. - * - * Return value : 0 - Ok - * -1 - Error - */ -int16_t WebRtcIsac_UpdateBwEstimate(ISACStruct* ISAC_main_inst, - const uint8_t* encoded, - size_t packet_size, - uint16_t rtp_seq_number, - uint32_t send_ts, - uint32_t arr_ts) { - ISACMainStruct* instISAC = (ISACMainStruct*)ISAC_main_inst; - Bitstr streamdata; -#ifndef WEBRTC_ARCH_BIG_ENDIAN - int k; -#endif - int16_t err; - - /* Check if decoder initiated. */ - if ((instISAC->initFlag & BIT_MASK_DEC_INIT) != BIT_MASK_DEC_INIT) { - instISAC->errorCode = ISAC_DECODER_NOT_INITIATED; - return -1; - } - - /* Check that the size of the packet is valid, and if not return without - * updating the bandwidth estimate. A valid size is at least 10 bytes. */ - if (packet_size < 10) { - /* Return error code if the packet length is null. */ - instISAC->errorCode = ISAC_EMPTY_PACKET; - return -1; - } - - WebRtcIsac_ResetBitstream(&(streamdata)); - -#ifndef WEBRTC_ARCH_BIG_ENDIAN - for (k = 0; k < 10; k++) { - uint16_t ek = ((const uint16_t*)encoded)[k >> 1]; - streamdata.stream[k] = (uint8_t)((ek >> ((k & 1) << 3)) & 0xff); - } -#else - memcpy(streamdata.stream, encoded, 10); -#endif - - err = WebRtcIsac_EstimateBandwidth(&instISAC->bwestimator_obj, &streamdata, - packet_size, rtp_seq_number, send_ts, - arr_ts, instISAC->encoderSamplingRateKHz, - instISAC->decoderSamplingRateKHz); - if (err < 0) { - /* Return error code if something went wrong. */ - instISAC->errorCode = -err; - return -1; - } - return 0; -} - -static int Decode(ISACStruct* ISAC_main_inst, - const uint8_t* encoded, - size_t lenEncodedBytes, - int16_t* decoded, - int16_t* speechType, - int16_t isRCUPayload) { - /* Number of samples (480 or 960), output from decoder - that were actually used in the encoder/decoder - (determined on the fly). */ - int16_t numSamplesLB; - int16_t numSamplesUB; - int16_t speechIdx; - float outFrame[MAX_FRAMESAMPLES]; - int16_t outFrameLB[MAX_FRAMESAMPLES]; - int16_t outFrameUB[MAX_FRAMESAMPLES]; - int numDecodedBytesLBint; - size_t numDecodedBytesLB; - int numDecodedBytesUB; - size_t lenEncodedLBBytes; - int16_t validChecksum = 1; - int16_t k; - uint16_t numLayer; - size_t totSizeBytes; - int16_t err; - - ISACMainStruct* instISAC = (ISACMainStruct*)ISAC_main_inst; - ISACUBDecStruct* decInstUB = &(instISAC->instUB.ISACdecUB_obj); - ISACLBDecStruct* decInstLB = &(instISAC->instLB.ISACdecLB_obj); - - /* Check if decoder initiated. */ - if ((instISAC->initFlag & BIT_MASK_DEC_INIT) != - BIT_MASK_DEC_INIT) { - instISAC->errorCode = ISAC_DECODER_NOT_INITIATED; - return -1; - } - - if (lenEncodedBytes == 0) { - /* return error code if the packet length is null. */ - instISAC->errorCode = ISAC_EMPTY_PACKET; - return -1; - } - - /* The size of the encoded lower-band is bounded by - * STREAM_SIZE_MAX. If a payload with the size larger than STREAM_SIZE_MAX - * is received, it is not considered erroneous. */ - lenEncodedLBBytes = (lenEncodedBytes > STREAM_SIZE_MAX) ? - STREAM_SIZE_MAX : lenEncodedBytes; - - /* Copy to lower-band bit-stream structure. */ - memcpy(instISAC->instLB.ISACdecLB_obj.bitstr_obj.stream, encoded, - lenEncodedLBBytes); - - /* We need to initialize numSamplesLB to something; otherwise, in the test - for whether we should return -1 below, the compiler might generate code - that fools Memcheck (Valgrind) into thinking that the control flow depends - on the uninitialized value in numSamplesLB (since WebRtcIsac_DecodeLb will - not fill it in if it fails and returns -1). */ - numSamplesLB = 0; - - /* Regardless of that the current codec is setup to work in - * wideband or super-wideband, the decoding of the lower-band - * has to be performed. */ - numDecodedBytesLBint = WebRtcIsac_DecodeLb(&instISAC->transform_tables, - outFrame, decInstLB, - &numSamplesLB, isRCUPayload); - numDecodedBytesLB = (size_t)numDecodedBytesLBint; - if ((numDecodedBytesLBint < 0) || - (numDecodedBytesLB > lenEncodedLBBytes) || - (numSamplesLB > MAX_FRAMESAMPLES)) { - instISAC->errorCode = ISAC_LENGTH_MISMATCH; - return -1; - } - - /* Error Check, we accept multi-layer bit-stream This will limit number - * of iterations of the while loop. Even without this the number - * of iterations is limited. */ - numLayer = 1; - totSizeBytes = numDecodedBytesLB; - while (totSizeBytes != lenEncodedBytes) { - if ((totSizeBytes > lenEncodedBytes) || - (encoded[totSizeBytes] == 0) || - (numLayer > MAX_NUM_LAYERS)) { - instISAC->errorCode = ISAC_LENGTH_MISMATCH; - return -1; - } - totSizeBytes += encoded[totSizeBytes]; - numLayer++; - } - - if (instISAC->decoderSamplingRateKHz == kIsacWideband) { - for (k = 0; k < numSamplesLB; k++) { - if (outFrame[k] > 32767) { - decoded[k] = 32767; - } else if (outFrame[k] < -32768) { - decoded[k] = -32768; - } else { - decoded[k] = (int16_t)WebRtcIsac_lrint(outFrame[k]); - } - } - numSamplesUB = 0; - } else { - uint32_t crc; - /* We don't accept larger than 30ms (480 samples at lower-band) - * frame-size. */ - for (k = 0; k < numSamplesLB; k++) { - if (outFrame[k] > 32767) { - outFrameLB[k] = 32767; - } else if (outFrame[k] < -32768) { - outFrameLB[k] = -32768; - } else { - outFrameLB[k] = (int16_t)WebRtcIsac_lrint(outFrame[k]); - } - } - - /* Check for possible error, and if upper-band stream exists. */ - if (numDecodedBytesLB == lenEncodedBytes) { - /* Decoding was successful. No super-wideband bit-stream exists. */ - numSamplesUB = numSamplesLB; - memset(outFrameUB, 0, sizeof(int16_t) * numSamplesUB); - - /* Prepare for the potential increase of signal bandwidth. */ - instISAC->resetFlag_8kHz = 2; - } else { - /* This includes the checksum and the bytes that stores the length. */ - int16_t lenNextStream = encoded[numDecodedBytesLB]; - - /* Is this garbage or valid super-wideband bit-stream? - * Check if checksum is valid. */ - if (lenNextStream <= (LEN_CHECK_SUM_WORD8 + 1)) { - /* Such a small second layer cannot be super-wideband layer. - * It must be a short garbage. */ - validChecksum = 0; - } else { - /* Run CRC to see if the checksum match. */ - WebRtcIsac_GetCrc((int16_t*)(&encoded[numDecodedBytesLB + 1]), - lenNextStream - LEN_CHECK_SUM_WORD8 - 1, &crc); - - validChecksum = 1; - for (k = 0; k < LEN_CHECK_SUM_WORD8; k++) { - validChecksum &= (((crc >> (24 - k * 8)) & 0xFF) == - encoded[numDecodedBytesLB + lenNextStream - - LEN_CHECK_SUM_WORD8 + k]); - } - } - - if (!validChecksum) { - /* This is a garbage, we have received a wideband - * bit-stream with garbage. */ - numSamplesUB = numSamplesLB; - memset(outFrameUB, 0, sizeof(int16_t) * numSamplesUB); - } else { - /* A valid super-wideband biststream exists. */ - enum ISACBandwidth bandwidthKHz; - int32_t maxDelayBit; - - /* If we have super-wideband bit-stream, we cannot - * have 60 ms frame-size. */ - if (numSamplesLB > FRAMESAMPLES) { - instISAC->errorCode = ISAC_LENGTH_MISMATCH; - return -1; - } - - /* The rest of the bit-stream contains the upper-band - * bit-stream curently this is the only thing there, - * however, we might add more layers. */ - - /* Have to exclude one byte where the length is stored - * and last 'LEN_CHECK_SUM_WORD8' bytes where the - * checksum is stored. */ - lenNextStream -= (LEN_CHECK_SUM_WORD8 + 1); - - memcpy(decInstUB->bitstr_obj.stream, - &encoded[numDecodedBytesLB + 1], lenNextStream); - - /* Reset bit-stream object, this is the first decoding. */ - WebRtcIsac_ResetBitstream(&(decInstUB->bitstr_obj)); - - /* Decode jitter information. */ - err = WebRtcIsac_DecodeJitterInfo(&decInstUB->bitstr_obj, &maxDelayBit); - if (err < 0) { - instISAC->errorCode = -err; - return -1; - } - - /* Update jitter info which is in the upper-band bit-stream - * only if the encoder is in super-wideband. Otherwise, - * the jitter info is already embedded in bandwidth index - * and has been updated. */ - if (instISAC->encoderSamplingRateKHz == kIsacSuperWideband) { - err = WebRtcIsac_UpdateUplinkJitter( - &(instISAC->bwestimator_obj), maxDelayBit); - if (err < 0) { - instISAC->errorCode = -err; - return -1; - } - } - - /* Decode bandwidth information. */ - err = WebRtcIsac_DecodeBandwidth(&decInstUB->bitstr_obj, - &bandwidthKHz); - if (err < 0) { - instISAC->errorCode = -err; - return -1; - } - - switch (bandwidthKHz) { - case isac12kHz: { - numDecodedBytesUB = WebRtcIsac_DecodeUb12( - &instISAC->transform_tables, outFrame, decInstUB, isRCUPayload); - - /* Hang-over for transient alleviation - - * wait two frames to add the upper band going up from 8 kHz. */ - if (instISAC->resetFlag_8kHz > 0) { - if (instISAC->resetFlag_8kHz == 2) { - /* Silence first and a half frame. */ - memset(outFrame, 0, MAX_FRAMESAMPLES * - sizeof(float)); - } else { - const float rampStep = 2.0f / MAX_FRAMESAMPLES; - float rampVal = 0; - memset(outFrame, 0, (MAX_FRAMESAMPLES >> 1) * - sizeof(float)); - - /* Ramp up second half of second frame. */ - for (k = MAX_FRAMESAMPLES / 2; k < MAX_FRAMESAMPLES; k++) { - outFrame[k] *= rampVal; - rampVal += rampStep; - } - } - instISAC->resetFlag_8kHz -= 1; - } - - break; - } - case isac16kHz: { - numDecodedBytesUB = WebRtcIsac_DecodeUb16( - &instISAC->transform_tables, outFrame, decInstUB, isRCUPayload); - break; - } - default: - return -1; - } - - if (numDecodedBytesUB < 0) { - instISAC->errorCode = numDecodedBytesUB; - return -1; - } - if (numDecodedBytesLB + numDecodedBytesUB > lenEncodedBytes) { - // We have supposedly decoded more bytes than we were given. Likely - // caused by bad input data. - instISAC->errorCode = ISAC_LENGTH_MISMATCH; - return -1; - } - - /* It might be less due to garbage. */ - if ((numDecodedBytesUB != lenNextStream) && - (numDecodedBytesLB + 1 + numDecodedBytesUB >= lenEncodedBytes || - numDecodedBytesUB != - (lenNextStream - - encoded[numDecodedBytesLB + 1 + numDecodedBytesUB]))) { - instISAC->errorCode = ISAC_LENGTH_MISMATCH; - return -1; - } - - /* If there is no error Upper-band always decodes - * 30 ms (480 samples). */ - numSamplesUB = FRAMESAMPLES; - - /* Convert to W16. */ - for (k = 0; k < numSamplesUB; k++) { - if (outFrame[k] > 32767) { - outFrameUB[k] = 32767; - } else if (outFrame[k] < -32768) { - outFrameUB[k] = -32768; - } else { - outFrameUB[k] = (int16_t)WebRtcIsac_lrint( - outFrame[k]); - } - } - } - } - - speechIdx = 0; - while (speechIdx < numSamplesLB) { - WebRtcSpl_SynthesisQMF(&outFrameLB[speechIdx], &outFrameUB[speechIdx], - FRAMESAMPLES_10ms, &decoded[(speechIdx << 1)], - instISAC->synthesisFBState1, - instISAC->synthesisFBState2); - - speechIdx += FRAMESAMPLES_10ms; - } - } - *speechType = 0; - return (numSamplesLB + numSamplesUB); -} - - - - - - - -/**************************************************************************** - * WebRtcIsac_Decode(...) - * - * This function decodes a ISAC frame. Output speech length - * will be a multiple of 480 samples: 480 or 960 samples, - * depending on the frameSize (30 or 60 ms). - * - * Input: - * - ISAC_main_inst : ISAC instance. - * - encoded : encoded ISAC frame(s) - * - len : bytes in encoded vector - * - * Output: - * - decoded : The decoded vector - * - * Return value : >0 - number of samples in decoded vector - * -1 - Error - */ - -int WebRtcIsac_Decode(ISACStruct* ISAC_main_inst, - const uint8_t* encoded, - size_t lenEncodedBytes, - int16_t* decoded, - int16_t* speechType) { - int16_t isRCUPayload = 0; - return Decode(ISAC_main_inst, encoded, lenEncodedBytes, decoded, - speechType, isRCUPayload); -} - -/**************************************************************************** - * WebRtcIsac_DecodeRcu(...) - * - * This function decodes a redundant (RCU) iSAC frame. Function is called in - * NetEq with a stored RCU payload in case of packet loss. Output speech length - * will be a multiple of 480 samples: 480 or 960 samples, - * depending on the framesize (30 or 60 ms). - * - * Input: - * - ISAC_main_inst : ISAC instance. - * - encoded : encoded ISAC RCU frame(s) - * - len : bytes in encoded vector - * - * Output: - * - decoded : The decoded vector - * - * Return value : >0 - number of samples in decoded vector - * -1 - Error - */ - - - -int WebRtcIsac_DecodeRcu(ISACStruct* ISAC_main_inst, - const uint8_t* encoded, - size_t lenEncodedBytes, - int16_t* decoded, - int16_t* speechType) { - int16_t isRCUPayload = 1; - return Decode(ISAC_main_inst, encoded, lenEncodedBytes, decoded, - speechType, isRCUPayload); -} - - -/**************************************************************************** - * WebRtcIsac_DecodePlc(...) - * - * This function conducts PLC for ISAC frame(s). Output speech length - * will be a multiple of 480 samples: 480 or 960 samples, - * depending on the frameSize (30 or 60 ms). - * - * Input: - * - ISAC_main_inst : ISAC instance. - * - noOfLostFrames : Number of PLC frames to produce - * - * Output: - * - decoded : The decoded vector - * - * Return value : Number of samples in decoded PLC vector - */ -size_t WebRtcIsac_DecodePlc(ISACStruct* ISAC_main_inst, - int16_t* decoded, - size_t noOfLostFrames) { - size_t numSamples = 0; - ISACMainStruct* instISAC = (ISACMainStruct*)ISAC_main_inst; - - /* Limit number of frames to two = 60 millisecond. - * Otherwise we exceed data vectors. */ - if (noOfLostFrames > 2) { - noOfLostFrames = 2; - } - - /* Get the number of samples per frame */ - switch (instISAC->decoderSamplingRateKHz) { - case kIsacWideband: { - numSamples = 480 * noOfLostFrames; - break; - } - case kIsacSuperWideband: { - numSamples = 960 * noOfLostFrames; - break; - } - } - - /* Set output samples to zero. */ - memset(decoded, 0, numSamples * sizeof(int16_t)); - return numSamples; -} - - -/**************************************************************************** - * ControlLb(...) - Internal function for controlling Lower Band - * ControlUb(...) - Internal function for controlling Upper Band - * WebRtcIsac_Control(...) - API function - * - * This function sets the limit on the short-term average bit rate and the - * frame length. Should be used only in Instantaneous mode. - * - * Input: - * - ISAC_main_inst : ISAC instance. - * - rate : limit on the short-term average bit rate, - * in bits/second (between 10000 and 32000) - * - frameSize : number of milliseconds per frame (30 or 60) - * - * Return value : 0 - ok - * -1 - Error - */ -static int16_t ControlLb(ISACLBStruct* instISAC, double rate, - int16_t frameSize) { - if ((rate >= 10000) && (rate <= 32000)) { - instISAC->ISACencLB_obj.bottleneck = rate; - } else { - return -ISAC_DISALLOWED_BOTTLENECK; - } - - if ((frameSize == 30) || (frameSize == 60)) { - instISAC->ISACencLB_obj.new_framelength = (FS / 1000) * frameSize; - } else { - return -ISAC_DISALLOWED_FRAME_LENGTH; - } - - return 0; -} - -static int16_t ControlUb(ISACUBStruct* instISAC, double rate) { - if ((rate >= 10000) && (rate <= 32000)) { - instISAC->ISACencUB_obj.bottleneck = rate; - } else { - return -ISAC_DISALLOWED_BOTTLENECK; - } - return 0; -} - -int16_t WebRtcIsac_Control(ISACStruct* ISAC_main_inst, - int32_t bottleneckBPS, - int frameSize) { - ISACMainStruct* instISAC = (ISACMainStruct*)ISAC_main_inst; - int16_t status; - double rateLB; - double rateUB; - enum ISACBandwidth bandwidthKHz; - - if (instISAC->codingMode == 0) { - /* In adaptive mode. */ - instISAC->errorCode = ISAC_MODE_MISMATCH; - return -1; - } - - /* Check if encoder initiated */ - if ((instISAC->initFlag & BIT_MASK_ENC_INIT) != - BIT_MASK_ENC_INIT) { - instISAC->errorCode = ISAC_ENCODER_NOT_INITIATED; - return -1; - } - - if (instISAC->encoderSamplingRateKHz == kIsacWideband) { - /* If the sampling rate is 16kHz then bandwith should be 8kHz, - * regardless of bottleneck. */ - bandwidthKHz = isac8kHz; - rateLB = (bottleneckBPS > 32000) ? 32000 : bottleneckBPS; - rateUB = 0; - } else { - if (WebRtcIsac_RateAllocation(bottleneckBPS, &rateLB, &rateUB, - &bandwidthKHz) < 0) { - return -1; - } - } - - if ((instISAC->encoderSamplingRateKHz == kIsacSuperWideband) && - (frameSize != 30) && - (bandwidthKHz != isac8kHz)) { - /* Cannot have 60 ms in super-wideband. */ - instISAC->errorCode = ISAC_DISALLOWED_FRAME_LENGTH; - return -1; - } - - status = ControlLb(&instISAC->instLB, rateLB, (int16_t)frameSize); - if (status < 0) { - instISAC->errorCode = -status; - return -1; - } - if (bandwidthKHz != isac8kHz) { - status = ControlUb(&(instISAC->instUB), rateUB); - if (status < 0) { - instISAC->errorCode = -status; - return -1; - } - } - - - /* Check if bandwidth is changing from wideband to super-wideband - * then we have to synch data buffer of lower & upper-band. Also - * clean up the upper-band data buffer. */ - - if ((instISAC->bandwidthKHz == isac8kHz) && (bandwidthKHz != isac8kHz)) { - memset(instISAC->instUB.ISACencUB_obj.data_buffer_float, 0, - sizeof(float) * (MAX_FRAMESAMPLES + LB_TOTAL_DELAY_SAMPLES)); - - if (bandwidthKHz == isac12kHz) { - instISAC->instUB.ISACencUB_obj.buffer_index = - instISAC->instLB.ISACencLB_obj.buffer_index; - } else { - instISAC->instUB.ISACencUB_obj.buffer_index = - LB_TOTAL_DELAY_SAMPLES + instISAC->instLB.ISACencLB_obj.buffer_index; - - memcpy(&(instISAC->instUB.ISACencUB_obj.lastLPCVec), - WebRtcIsac_kMeanLarUb16, sizeof(double) * UB_LPC_ORDER); - } - } - - /* Update the payload limit if the bandwidth is changing. */ - if (instISAC->bandwidthKHz != bandwidthKHz) { - instISAC->bandwidthKHz = bandwidthKHz; - UpdatePayloadSizeLimit(instISAC); - } - instISAC->bottleneck = bottleneckBPS; - return 0; -} - -void WebRtcIsac_SetInitialBweBottleneck(ISACStruct* ISAC_main_inst, - int bottleneck_bits_per_second) { - ISACMainStruct* instISAC = (ISACMainStruct*)ISAC_main_inst; - RTC_DCHECK_GE(bottleneck_bits_per_second, 10000); - RTC_DCHECK_LE(bottleneck_bits_per_second, 32000); - instISAC->bwestimator_obj.send_bw_avg = (float)bottleneck_bits_per_second; -} - -/**************************************************************************** - * WebRtcIsac_ControlBwe(...) - * - * This function sets the initial values of bottleneck and frame-size if - * iSAC is used in channel-adaptive mode. Through this API, users can - * enforce a frame-size for all values of bottleneck. Then iSAC will not - * automatically change the frame-size. - * - * - * Input: - * - ISAC_main_inst : ISAC instance. - * - rateBPS : initial value of bottleneck in bits/second - * 10000 <= rateBPS <= 32000 is accepted - * For default bottleneck set rateBPS = 0 - * - frameSizeMs : number of milliseconds per frame (30 or 60) - * - enforceFrameSize : 1 to enforce the given frame-size through out - * the adaptation process, 0 to let iSAC change - * the frame-size if required. - * - * Return value : 0 - ok - * -1 - Error - */ -int16_t WebRtcIsac_ControlBwe(ISACStruct* ISAC_main_inst, - int32_t bottleneckBPS, - int frameSizeMs, - int16_t enforceFrameSize) { - ISACMainStruct* instISAC = (ISACMainStruct*)ISAC_main_inst; - enum ISACBandwidth bandwidth; - - /* Check if encoder initiated */ - if ((instISAC->initFlag & BIT_MASK_ENC_INIT) != - BIT_MASK_ENC_INIT) { - instISAC->errorCode = ISAC_ENCODER_NOT_INITIATED; - return -1; - } - - /* Check that we are in channel-adaptive mode, otherwise, return (-1) */ - if (instISAC->codingMode != 0) { - instISAC->errorCode = ISAC_MODE_MISMATCH; - return -1; - } - if ((frameSizeMs != 30) && - (instISAC->encoderSamplingRateKHz == kIsacSuperWideband)) { - return -1; - } - - /* Set structure variable if enforceFrameSize is set. ISAC will then - * keep the chosen frame size. */ - if (enforceFrameSize != 0) { - instISAC->instLB.ISACencLB_obj.enforceFrameSize = 1; - } else { - instISAC->instLB.ISACencLB_obj.enforceFrameSize = 0; - } - - /* Set the initial rate. If the input value is zero then the default intial - * rate is used. Otehrwise, values between 10 to 32 kbps are accepted. */ - if (bottleneckBPS != 0) { - double rateLB; - double rateUB; - if (WebRtcIsac_RateAllocation(bottleneckBPS, &rateLB, &rateUB, - &bandwidth) < 0) { - return -1; - } - instISAC->bwestimator_obj.send_bw_avg = (float)bottleneckBPS; - instISAC->bandwidthKHz = bandwidth; - } - - /* Set the initial frame-size. If 'enforceFrameSize' is set, the frame-size - * will not change */ - if (frameSizeMs != 0) { - if ((frameSizeMs == 30) || (frameSizeMs == 60)) { - instISAC->instLB.ISACencLB_obj.new_framelength = - (int16_t)((FS / 1000) * frameSizeMs); - } else { - instISAC->errorCode = ISAC_DISALLOWED_FRAME_LENGTH; - return -1; - } - } - return 0; -} - - -/**************************************************************************** - * WebRtcIsac_GetDownLinkBwIndex(...) - * - * This function returns index representing the Bandwidth estimate from - * the other side to this side. - * - * Input: - * - ISAC_main_inst : iSAC structure - * - * Output: - * - bweIndex : Bandwidth estimate to transmit to other side. - * - */ -int16_t WebRtcIsac_GetDownLinkBwIndex(ISACStruct* ISAC_main_inst, - int16_t* bweIndex, - int16_t* jitterInfo) { - ISACMainStruct* instISAC = (ISACMainStruct*)ISAC_main_inst; - - /* Check if encoder initialized. */ - if ((instISAC->initFlag & BIT_MASK_DEC_INIT) != - BIT_MASK_DEC_INIT) { - instISAC->errorCode = ISAC_ENCODER_NOT_INITIATED; - return -1; - } - - /* Call function to get Bandwidth Estimate. */ - WebRtcIsac_GetDownlinkBwJitIndexImpl(&(instISAC->bwestimator_obj), bweIndex, - jitterInfo, - instISAC->decoderSamplingRateKHz); - return 0; -} - - -/**************************************************************************** - * WebRtcIsac_UpdateUplinkBw(...) - * - * This function takes an index representing the Bandwidth estimate from - * this side to other side and updates BWE. - * - * Input: - * - ISAC_main_inst : iSAC structure - * - rateIndex : Bandwidth estimate from other side. - * - * Return value : 0 - ok - * -1 - index out of range - */ -int16_t WebRtcIsac_UpdateUplinkBw(ISACStruct* ISAC_main_inst, - int16_t bweIndex) { - ISACMainStruct* instISAC = (ISACMainStruct*)ISAC_main_inst; - int16_t returnVal; - - /* Check if encoder initiated. */ - if ((instISAC->initFlag & BIT_MASK_ENC_INIT) != - BIT_MASK_ENC_INIT) { - instISAC->errorCode = ISAC_ENCODER_NOT_INITIATED; - return -1; - } - - /* Call function to get Bandwidth Estimate. */ - returnVal = WebRtcIsac_UpdateUplinkBwImpl( - &(instISAC->bwestimator_obj), bweIndex, - instISAC->encoderSamplingRateKHz); - - if (returnVal < 0) { - instISAC->errorCode = -returnVal; - return -1; - } else { - return 0; - } -} - - -/**************************************************************************** - * WebRtcIsac_ReadBwIndex(...) - * - * This function returns the index of the Bandwidth estimate from the - * bit-stream. - * - * Input: - * - encoded : Encoded bit-stream - * - * Output: - * - frameLength : Length of frame in packet (in samples) - * - bweIndex : Bandwidth estimate in bit-stream - * - */ -int16_t WebRtcIsac_ReadBwIndex(const uint8_t* encoded, - int16_t* bweIndex) { - Bitstr streamdata; -#ifndef WEBRTC_ARCH_BIG_ENDIAN - int k; -#endif - int16_t err; - - WebRtcIsac_ResetBitstream(&(streamdata)); - -#ifndef WEBRTC_ARCH_BIG_ENDIAN - for (k = 0; k < 10; k++) { - int16_t ek2 = ((const int16_t*)encoded)[k >> 1]; - streamdata.stream[k] = (uint8_t)((ek2 >> ((k & 1) << 3)) & 0xff); - } -#else - memcpy(streamdata.stream, encoded, 10); -#endif - - /* Decode frame length. */ - err = WebRtcIsac_DecodeFrameLen(&streamdata, bweIndex); - if (err < 0) { - return err; - } - - /* Decode BW estimation. */ - err = WebRtcIsac_DecodeSendBW(&streamdata, bweIndex); - if (err < 0) { - return err; - } - - return 0; -} - - -/**************************************************************************** - * WebRtcIsac_ReadFrameLen(...) - * - * This function returns the number of samples the decoder will generate if - * the given payload is decoded. - * - * Input: - * - encoded : Encoded bitstream - * - * Output: - * - frameLength : Length of frame in packet (in samples) - * - */ -int16_t WebRtcIsac_ReadFrameLen(const ISACStruct* ISAC_main_inst, - const uint8_t* encoded, - int16_t* frameLength) { - Bitstr streamdata; -#ifndef WEBRTC_ARCH_BIG_ENDIAN - int k; -#endif - int16_t err; - ISACMainStruct* instISAC; - - WebRtcIsac_ResetBitstream(&(streamdata)); - -#ifndef WEBRTC_ARCH_BIG_ENDIAN - for (k = 0; k < 10; k++) { - int16_t ek2 = ((const int16_t*)encoded)[k >> 1]; - streamdata.stream[k] = (uint8_t)((ek2 >> ((k & 1) << 3)) & 0xff); - } -#else - memcpy(streamdata.stream, encoded, 10); -#endif - - /* Decode frame length. */ - err = WebRtcIsac_DecodeFrameLen(&streamdata, frameLength); - if (err < 0) { - return -1; - } - instISAC = (ISACMainStruct*)ISAC_main_inst; - - if (instISAC->decoderSamplingRateKHz == kIsacSuperWideband) { - /* The decoded frame length indicates the number of samples in - * lower-band in this case, multiply by 2 to get the total number - * of samples. */ - *frameLength <<= 1; - } - return 0; -} - - -/******************************************************************************* - * WebRtcIsac_GetNewFrameLen(...) - * - * This function returns the frame length (in samples) of the next packet. - * In the case of channel-adaptive mode, iSAC decides on its frame length based - * on the estimated bottleneck, this AOI allows a user to prepare for the next - * packet (at the encoder). - * - * The primary usage is in CE to make the iSAC works in channel-adaptive mode - * - * Input: - * - ISAC_main_inst : iSAC struct - * - * Return Value : frame lenght in samples - * - */ -int16_t WebRtcIsac_GetNewFrameLen(ISACStruct* ISAC_main_inst) { - ISACMainStruct* instISAC = (ISACMainStruct*)ISAC_main_inst; - - /* Return new frame length. */ - if (instISAC->in_sample_rate_hz == 16000) - return (instISAC->instLB.ISACencLB_obj.new_framelength); - else /* 32000 Hz */ - return ((instISAC->instLB.ISACencLB_obj.new_framelength) * 2); -} - - -/**************************************************************************** - * WebRtcIsac_GetErrorCode(...) - * - * This function can be used to check the error code of an iSAC instance. - * When a function returns -1 an error code will be set for that instance. - * The function below extracts the code of the last error that occurred in - * the specified instance. - * - * Input: - * - ISAC_main_inst : ISAC instance - * - * Return value : Error code - */ -int16_t WebRtcIsac_GetErrorCode(ISACStruct* ISAC_main_inst) { - return ((ISACMainStruct*)ISAC_main_inst)->errorCode; -} - - -/**************************************************************************** - * WebRtcIsac_GetUplinkBw(...) - * - * This function outputs the target bottleneck of the codec. In - * channel-adaptive mode, the target bottleneck is specified through an in-band - * signalling retrieved by bandwidth estimator. - * In channel-independent, also called instantaneous mode, the target - * bottleneck is provided to the encoder by calling xxx_control(...) (if - * xxx_control is never called, the default values are used.). - * Note that the output is the iSAC internal operating bottleneck which might - * differ slightly from the one provided through xxx_control(). - * - * Input: - * - ISAC_main_inst : iSAC instance - * - * Output: - * - *bottleneck : bottleneck in bits/sec - * - * Return value : -1 if error happens - * 0 bit-rates computed correctly. - */ -int16_t WebRtcIsac_GetUplinkBw(ISACStruct* ISAC_main_inst, - int32_t* bottleneck) { - ISACMainStruct* instISAC = (ISACMainStruct*)ISAC_main_inst; - - if (instISAC->codingMode == 0) { - /* We are in adaptive mode then get the bottleneck from BWE. */ - *bottleneck = (int32_t)instISAC->bwestimator_obj.send_bw_avg; - } else { - *bottleneck = instISAC->bottleneck; - } - - if ((*bottleneck > 32000) && (*bottleneck < 38000)) { - *bottleneck = 32000; - } else if ((*bottleneck > 45000) && (*bottleneck < 50000)) { - *bottleneck = 45000; - } else if (*bottleneck > 56000) { - *bottleneck = 56000; - } - return 0; -} - - -/****************************************************************************** - * WebRtcIsac_SetMaxPayloadSize(...) - * - * This function sets a limit for the maximum payload size of iSAC. The same - * value is used both for 30 and 60 ms packets. If the encoder sampling rate - * is 16 kHz the maximum payload size is between 120 and 400 bytes. If the - * encoder sampling rate is 32 kHz the maximum payload size is between 120 - * and 600 bytes. - * - * --------------- - * IMPORTANT NOTES - * --------------- - * The size of a packet is limited to the minimum of 'max-payload-size' and - * 'max-rate.' For instance, let's assume the max-payload-size is set to - * 170 bytes, and max-rate is set to 40 kbps. Note that a limit of 40 kbps - * translates to 150 bytes for 30ms frame-size & 300 bytes for 60ms - * frame-size. Then a packet with a frame-size of 30 ms is limited to 150, - * i.e. min(170, 150), and a packet with 60 ms frame-size is limited to - * 170 bytes, i.e. min(170, 300). - * - * Input: - * - ISAC_main_inst : iSAC instance - * - maxPayloadBytes : maximum size of the payload in bytes - * valid values are between 100 and 400 bytes - * if encoder sampling rate is 16 kHz. For - * 32 kHz encoder sampling rate valid values - * are between 100 and 600 bytes. - * - * Return value : 0 if successful - * -1 if error happens - */ -int16_t WebRtcIsac_SetMaxPayloadSize(ISACStruct* ISAC_main_inst, - int16_t maxPayloadBytes) { - ISACMainStruct* instISAC = (ISACMainStruct*)ISAC_main_inst; - int16_t status = 0; - - /* Check if encoder initiated */ - if ((instISAC->initFlag & BIT_MASK_ENC_INIT) != - BIT_MASK_ENC_INIT) { - instISAC->errorCode = ISAC_ENCODER_NOT_INITIATED; - return -1; - } - - if (instISAC->encoderSamplingRateKHz == kIsacSuperWideband) { - /* Sanity check. */ - if (maxPayloadBytes < 120) { - /* 'maxRate' is out of valid range - * set to the acceptable value and return -1. */ - maxPayloadBytes = 120; - status = -1; - } - - /* sanity check */ - if (maxPayloadBytes > STREAM_SIZE_MAX) { - /* maxRate is out of valid range, - * set to the acceptable value and return -1. */ - maxPayloadBytes = STREAM_SIZE_MAX; - status = -1; - } - } else { - if (maxPayloadBytes < 120) { - /* Max payload-size is out of valid range - * set to the acceptable value and return -1. */ - maxPayloadBytes = 120; - status = -1; - } - if (maxPayloadBytes > STREAM_SIZE_MAX_60) { - /* Max payload-size is out of valid range - * set to the acceptable value and return -1. */ - maxPayloadBytes = STREAM_SIZE_MAX_60; - status = -1; - } - } - instISAC->maxPayloadSizeBytes = maxPayloadBytes; - UpdatePayloadSizeLimit(instISAC); - return status; -} - - -/****************************************************************************** - * WebRtcIsac_SetMaxRate(...) - * - * This function sets the maximum rate which the codec may not exceed for - * any signal packet. The maximum rate is defined and payload-size per - * frame-size in bits per second. - * - * The codec has a maximum rate of 53400 bits per second (200 bytes per 30 - * ms) if the encoder sampling rate is 16kHz, and 160 kbps (600 bytes/30 ms) - * if the encoder sampling rate is 32 kHz. - * - * It is possible to set a maximum rate between 32000 and 53400 bits/sec - * in wideband mode, and 32000 to 160000 bits/sec in super-wideband mode. - * - * --------------- - * IMPORTANT NOTES - * --------------- - * The size of a packet is limited to the minimum of 'max-payload-size' and - * 'max-rate.' For instance, let's assume the max-payload-size is set to - * 170 bytes, and max-rate is set to 40 kbps. Note that a limit of 40 kbps - * translates to 150 bytes for 30ms frame-size & 300 bytes for 60ms - * frame-size. Then a packet with a frame-size of 30 ms is limited to 150, - * i.e. min(170, 150), and a packet with 60 ms frame-size is limited to - * 170 bytes, min(170, 300). - * - * Input: - * - ISAC_main_inst : iSAC instance - * - maxRate : maximum rate in bits per second, - * valid values are 32000 to 53400 bits/sec in - * wideband mode, and 32000 to 160000 bits/sec in - * super-wideband mode. - * - * Return value : 0 if successful - * -1 if error happens - */ -int16_t WebRtcIsac_SetMaxRate(ISACStruct* ISAC_main_inst, - int32_t maxRate) { - ISACMainStruct* instISAC = (ISACMainStruct*)ISAC_main_inst; - int16_t maxRateInBytesPer30Ms; - int16_t status = 0; - - /* check if encoder initiated */ - if ((instISAC->initFlag & BIT_MASK_ENC_INIT) != BIT_MASK_ENC_INIT) { - instISAC->errorCode = ISAC_ENCODER_NOT_INITIATED; - return -1; - } - /* Calculate maximum number of bytes per 30 msec packets for the - given maximum rate. Multiply with 30/1000 to get number of - bits per 30 ms, divide by 8 to get number of bytes per 30 ms: - maxRateInBytes = floor((maxRate * 30/1000) / 8); */ - maxRateInBytesPer30Ms = (int16_t)(maxRate * 3 / 800); - - if (instISAC->encoderSamplingRateKHz == kIsacWideband) { - if (maxRate < 32000) { - /* 'maxRate' is out of valid range. - * Set to the acceptable value and return -1. */ - maxRateInBytesPer30Ms = 120; - status = -1; - } - - if (maxRate > 53400) { - /* 'maxRate' is out of valid range. - * Set to the acceptable value and return -1. */ - maxRateInBytesPer30Ms = 200; - status = -1; - } - } else { - if (maxRateInBytesPer30Ms < 120) { - /* 'maxRate' is out of valid range - * Set to the acceptable value and return -1. */ - maxRateInBytesPer30Ms = 120; - status = -1; - } - - if (maxRateInBytesPer30Ms > STREAM_SIZE_MAX) { - /* 'maxRate' is out of valid range. - * Set to the acceptable value and return -1. */ - maxRateInBytesPer30Ms = STREAM_SIZE_MAX; - status = -1; - } - } - instISAC->maxRateBytesPer30Ms = maxRateInBytesPer30Ms; - UpdatePayloadSizeLimit(instISAC); - return status; -} - - -/**************************************************************************** - * WebRtcIsac_GetRedPayload(...) - * - * This function populates "encoded" with the redundant payload of the recently - * encodedframe. This function has to be called once that WebRtcIsac_Encode(...) - * returns a positive value. Regardless of the frame-size this function will - * be called only once after encoding is completed. The bit-stream is - * targeted for 16000 bit/sec. - * - * Input: - * - ISAC_main_inst : iSAC struct - * - * Output: - * - encoded : the encoded data vector - * - * - * Return value : >0 - Length (in bytes) of coded data - * : -1 - Error - */ -int16_t WebRtcIsac_GetRedPayload(ISACStruct* ISAC_main_inst, - uint8_t* encoded) { - Bitstr iSACBitStreamInst; - int16_t streamLenLB; - int16_t streamLenUB; - int16_t streamLen; - int16_t totalLenUB; - ISACMainStruct* instISAC = (ISACMainStruct*)ISAC_main_inst; -#ifndef WEBRTC_ARCH_BIG_ENDIAN - int k; -#endif - - if ((instISAC->initFlag & BIT_MASK_ENC_INIT) != - BIT_MASK_ENC_INIT) { - instISAC->errorCode = ISAC_ENCODER_NOT_INITIATED; - } - - WebRtcIsac_ResetBitstream(&(iSACBitStreamInst)); - - streamLenLB = WebRtcIsac_EncodeStoredDataLb( - &instISAC->instLB.ISACencLB_obj.SaveEnc_obj, - &iSACBitStreamInst, - instISAC->instLB.ISACencLB_obj.lastBWIdx, - RCU_TRANSCODING_SCALE); - if (streamLenLB < 0) { - return -1; - } - - /* convert from bytes to int16_t. */ - memcpy(encoded, iSACBitStreamInst.stream, streamLenLB); - streamLen = streamLenLB; - if (instISAC->bandwidthKHz == isac8kHz) { - return streamLenLB; - } - - streamLenUB = WebRtcIsac_GetRedPayloadUb( - &instISAC->instUB.ISACencUB_obj.SaveEnc_obj, - &iSACBitStreamInst, instISAC->bandwidthKHz); - if (streamLenUB < 0) { - /* An error has happened but this is not the error due to a - * bit-stream larger than the limit. */ - return -1; - } - - /* We have one byte to write the total length of the upper-band. - * The length includes the bit-stream length, check-sum and the - * single byte where the length is written to. This is according to - * iSAC wideband and how the "garbage" is dealt. */ - totalLenUB = streamLenUB + 1 + LEN_CHECK_SUM_WORD8; - if (totalLenUB > 255) { - streamLenUB = 0; - } - - /* Generate CRC if required. */ - if ((instISAC->bandwidthKHz != isac8kHz) && - (streamLenUB > 0)) { - uint32_t crc; - streamLen += totalLenUB; - encoded[streamLenLB] = (uint8_t)totalLenUB; - memcpy(&encoded[streamLenLB + 1], iSACBitStreamInst.stream, - streamLenUB); - - WebRtcIsac_GetCrc((int16_t*)(&(encoded[streamLenLB + 1])), - streamLenUB, &crc); -#ifndef WEBRTC_ARCH_BIG_ENDIAN - for (k = 0; k < LEN_CHECK_SUM_WORD8; k++) { - encoded[streamLen - LEN_CHECK_SUM_WORD8 + k] = - (uint8_t)((crc >> (24 - k * 8)) & 0xFF); - } -#else - memcpy(&encoded[streamLenLB + streamLenUB + 1], &crc, - LEN_CHECK_SUM_WORD8); -#endif - } - return streamLen; -} - - -/**************************************************************************** - * WebRtcIsac_version(...) - * - * This function returns the version number. - * - * Output: - * - version : Pointer to character string - * - */ -void WebRtcIsac_version(char* version) { - strcpy(version, "4.3.0"); -} - - -/****************************************************************************** - * WebRtcIsac_SetEncSampRate() - * This function sets the sampling rate of the encoder. Initialization of the - * encoder WILL NOT overwrite the sampling rate of the encoder. The default - * value is 16 kHz which is set when the instance is created. The encoding-mode - * and the bottleneck remain unchanged by this call, however, the maximum rate - * and maximum payload-size will be reset to their default values. - * - * Input: - * - ISAC_main_inst : iSAC instance - * - sample_rate_hz : sampling rate in Hertz, valid values are 16000 - * and 32000. - * - * Return value : 0 if successful - * -1 if failed. - */ -int16_t WebRtcIsac_SetEncSampRate(ISACStruct* ISAC_main_inst, - uint16_t sample_rate_hz) { - ISACMainStruct* instISAC = (ISACMainStruct*)ISAC_main_inst; - enum IsacSamplingRate encoder_operational_rate; - - if ((sample_rate_hz != 16000) && (sample_rate_hz != 32000)) { - /* Sampling Frequency is not supported. */ - instISAC->errorCode = ISAC_UNSUPPORTED_SAMPLING_FREQUENCY; - return -1; - } - if (sample_rate_hz == 16000) { - encoder_operational_rate = kIsacWideband; - } else { - encoder_operational_rate = kIsacSuperWideband; - } - - if ((instISAC->initFlag & BIT_MASK_ENC_INIT) != - BIT_MASK_ENC_INIT) { - if (encoder_operational_rate == kIsacWideband) { - instISAC->bandwidthKHz = isac8kHz; - } else { - instISAC->bandwidthKHz = isac16kHz; - } - } else { - ISACUBStruct* instUB = &(instISAC->instUB); - ISACLBStruct* instLB = &(instISAC->instLB); - int32_t bottleneck = instISAC->bottleneck; - int16_t codingMode = instISAC->codingMode; - int16_t frameSizeMs = instLB->ISACencLB_obj.new_framelength / - (FS / 1000); - - if ((encoder_operational_rate == kIsacWideband) && - (instISAC->encoderSamplingRateKHz == kIsacSuperWideband)) { - /* Changing from super-wideband to wideband. - * we don't need to re-initialize the encoder of the lower-band. */ - instISAC->bandwidthKHz = isac8kHz; - if (codingMode == 1) { - ControlLb(instLB, - (bottleneck > 32000) ? 32000 : bottleneck, FRAMESIZE); - } - instISAC->maxPayloadSizeBytes = STREAM_SIZE_MAX_60; - instISAC->maxRateBytesPer30Ms = STREAM_SIZE_MAX_30; - } else if ((encoder_operational_rate == kIsacSuperWideband) && - (instISAC->encoderSamplingRateKHz == kIsacWideband)) { - double bottleneckLB = 0; - double bottleneckUB = 0; - if (codingMode == 1) { - WebRtcIsac_RateAllocation(bottleneck, &bottleneckLB, &bottleneckUB, - &(instISAC->bandwidthKHz)); - } - - instISAC->bandwidthKHz = isac16kHz; - instISAC->maxPayloadSizeBytes = STREAM_SIZE_MAX; - instISAC->maxRateBytesPer30Ms = STREAM_SIZE_MAX; - - EncoderInitLb(instLB, codingMode, encoder_operational_rate); - EncoderInitUb(instUB, instISAC->bandwidthKHz); - - memset(instISAC->analysisFBState1, 0, - FB_STATE_SIZE_WORD32 * sizeof(int32_t)); - memset(instISAC->analysisFBState2, 0, - FB_STATE_SIZE_WORD32 * sizeof(int32_t)); - - if (codingMode == 1) { - instISAC->bottleneck = bottleneck; - ControlLb(instLB, bottleneckLB, - (instISAC->bandwidthKHz == isac8kHz) ? frameSizeMs:FRAMESIZE); - if (instISAC->bandwidthKHz > isac8kHz) { - ControlUb(instUB, bottleneckUB); - } - } else { - instLB->ISACencLB_obj.enforceFrameSize = 0; - instLB->ISACencLB_obj.new_framelength = FRAMESAMPLES; - } - } - } - instISAC->encoderSamplingRateKHz = encoder_operational_rate; - instISAC->in_sample_rate_hz = sample_rate_hz; - return 0; -} - - -/****************************************************************************** - * WebRtcIsac_SetDecSampRate() - * This function sets the sampling rate of the decoder. Initialization of the - * decoder WILL NOT overwrite the sampling rate of the encoder. The default - * value is 16 kHz which is set when the instance is created. - * - * Input: - * - ISAC_main_inst : iSAC instance - * - sample_rate_hz : sampling rate in Hertz, valid values are 16000 - * and 32000. - * - * Return value : 0 if successful - * -1 if failed. - */ -int16_t WebRtcIsac_SetDecSampRate(ISACStruct* ISAC_main_inst, - uint16_t sample_rate_hz) { - ISACMainStruct* instISAC = (ISACMainStruct*)ISAC_main_inst; - enum IsacSamplingRate decoder_operational_rate; - - if (sample_rate_hz == 16000) { - decoder_operational_rate = kIsacWideband; - } else if (sample_rate_hz == 32000) { - decoder_operational_rate = kIsacSuperWideband; - } else { - /* Sampling Frequency is not supported. */ - instISAC->errorCode = ISAC_UNSUPPORTED_SAMPLING_FREQUENCY; - return -1; - } - - if ((instISAC->decoderSamplingRateKHz == kIsacWideband) && - (decoder_operational_rate == kIsacSuperWideband)) { - /* Switching from wideband to super-wideband at the decoder - * we need to reset the filter-bank and initialize upper-band decoder. */ - memset(instISAC->synthesisFBState1, 0, - FB_STATE_SIZE_WORD32 * sizeof(int32_t)); - memset(instISAC->synthesisFBState2, 0, - FB_STATE_SIZE_WORD32 * sizeof(int32_t)); - - DecoderInitUb(&instISAC->instUB); - } - instISAC->decoderSamplingRateKHz = decoder_operational_rate; - return 0; -} - - -/****************************************************************************** - * WebRtcIsac_EncSampRate() - * - * Input: - * - ISAC_main_inst : iSAC instance - * - * Return value : sampling rate in Hertz. The input to encoder - * is expected to be sampled in this rate. - * - */ -uint16_t WebRtcIsac_EncSampRate(ISACStruct* ISAC_main_inst) { - ISACMainStruct* instISAC = (ISACMainStruct*)ISAC_main_inst; - return instISAC->in_sample_rate_hz; -} - - -/****************************************************************************** - * WebRtcIsac_DecSampRate() - * Return the sampling rate of the decoded audio. - * - * Input: - * - ISAC_main_inst : iSAC instance - * - * Return value : sampling rate in Hertz. Decoder output is - * sampled at this rate. - * - */ -uint16_t WebRtcIsac_DecSampRate(ISACStruct* ISAC_main_inst) { - ISACMainStruct* instISAC = (ISACMainStruct*)ISAC_main_inst; - return instISAC->decoderSamplingRateKHz == kIsacWideband ? 16000 : 32000; -} - -void WebRtcIsac_SetEncSampRateInDecoder(ISACStruct* inst, - int sample_rate_hz) { - ISACMainStruct* instISAC = (ISACMainStruct*)inst; - RTC_DCHECK_NE(0, instISAC->initFlag & BIT_MASK_DEC_INIT); - RTC_DCHECK(!(instISAC->initFlag & BIT_MASK_ENC_INIT)); - RTC_DCHECK(sample_rate_hz == 16000 || sample_rate_hz == 32000); - instISAC->encoderSamplingRateKHz = sample_rate_hz / 1000; -} diff --git a/modules/audio_coding/codecs/isac/main/source/isac_float_type.h b/modules/audio_coding/codecs/isac/main/source/isac_float_type.h deleted file mode 100644 index 511bc97ee6..0000000000 --- a/modules/audio_coding/codecs/isac/main/source/isac_float_type.h +++ /dev/null @@ -1,100 +0,0 @@ -/* - * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_ISAC_FLOAT_TYPE_H_ -#define MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_ISAC_FLOAT_TYPE_H_ - -#include "modules/audio_coding/codecs/isac/main/include/isac.h" - -namespace webrtc { - -struct IsacFloat { - using instance_type = ISACStruct; - static const bool has_swb = true; - static inline int16_t Control(instance_type* inst, - int32_t rate, - int framesize) { - return WebRtcIsac_Control(inst, rate, framesize); - } - static inline int16_t ControlBwe(instance_type* inst, - int32_t rate_bps, - int frame_size_ms, - int16_t enforce_frame_size) { - return WebRtcIsac_ControlBwe(inst, rate_bps, frame_size_ms, - enforce_frame_size); - } - static inline int16_t Create(instance_type** inst) { - return WebRtcIsac_Create(inst); - } - static inline int DecodeInternal(instance_type* inst, - const uint8_t* encoded, - size_t len, - int16_t* decoded, - int16_t* speech_type) { - return WebRtcIsac_Decode(inst, encoded, len, decoded, speech_type); - } - static inline size_t DecodePlc(instance_type* inst, - int16_t* decoded, - size_t num_lost_frames) { - return WebRtcIsac_DecodePlc(inst, decoded, num_lost_frames); - } - - static inline void DecoderInit(instance_type* inst) { - WebRtcIsac_DecoderInit(inst); - } - static inline int Encode(instance_type* inst, - const int16_t* speech_in, - uint8_t* encoded) { - return WebRtcIsac_Encode(inst, speech_in, encoded); - } - static inline int16_t EncoderInit(instance_type* inst, int16_t coding_mode) { - return WebRtcIsac_EncoderInit(inst, coding_mode); - } - static inline uint16_t EncSampRate(instance_type* inst) { - return WebRtcIsac_EncSampRate(inst); - } - - static inline int16_t Free(instance_type* inst) { - return WebRtcIsac_Free(inst); - } - static inline int16_t GetErrorCode(instance_type* inst) { - return WebRtcIsac_GetErrorCode(inst); - } - - static inline int16_t GetNewFrameLen(instance_type* inst) { - return WebRtcIsac_GetNewFrameLen(inst); - } - static inline int16_t SetDecSampRate(instance_type* inst, - uint16_t sample_rate_hz) { - return WebRtcIsac_SetDecSampRate(inst, sample_rate_hz); - } - static inline int16_t SetEncSampRate(instance_type* inst, - uint16_t sample_rate_hz) { - return WebRtcIsac_SetEncSampRate(inst, sample_rate_hz); - } - static inline void SetEncSampRateInDecoder(instance_type* inst, - uint16_t sample_rate_hz) { - WebRtcIsac_SetEncSampRateInDecoder(inst, sample_rate_hz); - } - static inline void SetInitialBweBottleneck(instance_type* inst, - int bottleneck_bits_per_second) { - WebRtcIsac_SetInitialBweBottleneck(inst, bottleneck_bits_per_second); - } - static inline int16_t SetMaxPayloadSize(instance_type* inst, - int16_t max_payload_size_bytes) { - return WebRtcIsac_SetMaxPayloadSize(inst, max_payload_size_bytes); - } - static inline int16_t SetMaxRate(instance_type* inst, int32_t max_bit_rate) { - return WebRtcIsac_SetMaxRate(inst, max_bit_rate); - } -}; - -} // namespace webrtc -#endif // MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_ISAC_FLOAT_TYPE_H_ diff --git a/modules/audio_coding/codecs/isac/main/source/isac_unittest.cc b/modules/audio_coding/codecs/isac/main/source/isac_unittest.cc deleted file mode 100644 index c98b21d86f..0000000000 --- a/modules/audio_coding/codecs/isac/main/source/isac_unittest.cc +++ /dev/null @@ -1,111 +0,0 @@ -/* - * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ -#include "modules/audio_coding/codecs/isac/main/include/isac.h" - -#include - -#include "test/gtest.h" -#include "test/testsupport/file_utils.h" - -struct WebRtcISACStruct; - -namespace webrtc { - -// Number of samples in a 60 ms, sampled at 32 kHz. -const int kIsacNumberOfSamples = 320 * 6; -// Maximum number of bytes in output bitstream. -const size_t kMaxBytes = 1000; - -class IsacTest : public ::testing::Test { - protected: - IsacTest(); - virtual void SetUp(); - - WebRtcISACStruct* isac_codec_; - - int16_t speech_data_[kIsacNumberOfSamples]; - int16_t output_data_[kIsacNumberOfSamples]; - uint8_t bitstream_[kMaxBytes]; - uint8_t bitstream_small_[7]; // Simulate sync packets. -}; - -IsacTest::IsacTest() : isac_codec_(NULL) {} - -void IsacTest::SetUp() { - // Read some samples from a speech file, to be used in the encode test. - FILE* input_file; - const std::string file_name = - webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm"); - input_file = fopen(file_name.c_str(), "rb"); - ASSERT_TRUE(input_file != NULL); - ASSERT_EQ(kIsacNumberOfSamples, - static_cast(fread(speech_data_, sizeof(int16_t), - kIsacNumberOfSamples, input_file))); - fclose(input_file); - input_file = NULL; -} - -// Test failing Create. -TEST_F(IsacTest, IsacCreateFail) { - // Test to see that an invalid pointer is caught. - EXPECT_EQ(-1, WebRtcIsac_Create(NULL)); -} - -// Test failing Free. -TEST_F(IsacTest, IsacFreeFail) { - // Test to see that free function doesn't crash. - EXPECT_EQ(0, WebRtcIsac_Free(NULL)); -} - -// Test normal Create and Free. -TEST_F(IsacTest, IsacCreateFree) { - EXPECT_EQ(0, WebRtcIsac_Create(&isac_codec_)); - EXPECT_TRUE(isac_codec_ != NULL); - EXPECT_EQ(0, WebRtcIsac_Free(isac_codec_)); -} - -TEST_F(IsacTest, IsacUpdateBWE) { - // Create encoder memory. - EXPECT_EQ(0, WebRtcIsac_Create(&isac_codec_)); - - // Init encoder (adaptive mode) and decoder. - WebRtcIsac_EncoderInit(isac_codec_, 0); - WebRtcIsac_DecoderInit(isac_codec_); - - int encoded_bytes; - - // Test with call with a small packet (sync packet). - EXPECT_EQ(-1, WebRtcIsac_UpdateBwEstimate(isac_codec_, bitstream_small_, 7, 1, - 12345, 56789)); - - // Encode 60 ms of data (needed to create a first packet). - encoded_bytes = WebRtcIsac_Encode(isac_codec_, speech_data_, bitstream_); - EXPECT_EQ(0, encoded_bytes); - encoded_bytes = WebRtcIsac_Encode(isac_codec_, speech_data_, bitstream_); - EXPECT_EQ(0, encoded_bytes); - encoded_bytes = WebRtcIsac_Encode(isac_codec_, speech_data_, bitstream_); - EXPECT_EQ(0, encoded_bytes); - encoded_bytes = WebRtcIsac_Encode(isac_codec_, speech_data_, bitstream_); - EXPECT_EQ(0, encoded_bytes); - encoded_bytes = WebRtcIsac_Encode(isac_codec_, speech_data_, bitstream_); - EXPECT_EQ(0, encoded_bytes); - encoded_bytes = WebRtcIsac_Encode(isac_codec_, speech_data_, bitstream_); - EXPECT_GT(encoded_bytes, 0); - - // Call to update bandwidth estimator with real data. - EXPECT_EQ(0, WebRtcIsac_UpdateBwEstimate(isac_codec_, bitstream_, - static_cast(encoded_bytes), - 1, 12345, 56789)); - - // Free memory. - EXPECT_EQ(0, WebRtcIsac_Free(isac_codec_)); -} - -} // namespace webrtc diff --git a/modules/audio_coding/codecs/isac/main/source/lattice.c b/modules/audio_coding/codecs/isac/main/source/lattice.c deleted file mode 100644 index d9d2d65665..0000000000 --- a/modules/audio_coding/codecs/isac/main/source/lattice.c +++ /dev/null @@ -1,219 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/* - * lattice.c - * - * contains the normalized lattice filter routines (MA and AR) for iSAC codec - * - */ - -#include -#include -#include -#ifdef WEBRTC_ANDROID -#include -#endif - -#include "modules/audio_coding/codecs/isac/main/source/settings.h" -#include "modules/audio_coding/codecs/isac/main/source/codec.h" - -/* filter the signal using normalized lattice filter */ -/* MA filter */ -void WebRtcIsac_NormLatticeFilterMa(int orderCoef, - float *stateF, - float *stateG, - float *lat_in, - double *filtcoeflo, - double *lat_out) -{ - int n,k,i,u,temp1; - int ord_1 = orderCoef+1; - float sth[MAX_AR_MODEL_ORDER]; - float cth[MAX_AR_MODEL_ORDER]; - float inv_cth[MAX_AR_MODEL_ORDER]; - double a[MAX_AR_MODEL_ORDER+1]; - float f[MAX_AR_MODEL_ORDER+1][HALF_SUBFRAMELEN], g[MAX_AR_MODEL_ORDER+1][HALF_SUBFRAMELEN]; - float gain1; - - for (u=0;u=0;i--) //get the state of f&g for the first input, for all orders - { - ARf[i][0] = cth[i]*ARf[i+1][0] - sth[i]*stateG[i]; - ARg[i+1][0] = sth[i]*ARf[i+1][0] + cth[i]* stateG[i]; - } - ARg[0][0] = ARf[0][0]; - - for(n=0;n<(HALF_SUBFRAMELEN-1);n++) - { - for(k=orderCoef-1;k>=0;k--) - { - ARf[k][n+1] = cth[k]*ARf[k+1][n+1] - sth[k]*ARg[k][n]; - ARg[k+1][n+1] = sth[k]*ARf[k+1][n+1] + cth[k]* ARg[k][n]; - } - ARg[0][n+1] = ARf[0][n+1]; - } - - memcpy(lat_out+u * HALF_SUBFRAMELEN, &(ARf[0][0]), sizeof(float) * HALF_SUBFRAMELEN); - - /* cannot use memcpy in the following */ - for (i=0;i0; m--) - { - tmp_inv = 1.0f / cth2; - for (k=1; k<=m; k++) - { - tmp[k] = ((float)a[k] - sth[m] * (float)a[m-k+1]) * tmp_inv; - } - - for (k=1; k -#include - -#include "modules/audio_coding/codecs/isac/main/source/lpc_analysis.h" -#include "modules/audio_coding/codecs/isac/main/source/settings.h" -#include "modules/audio_coding/codecs/isac/main/source/codec.h" -#include "modules/audio_coding/codecs/isac/main/source/entropy_coding.h" -#include "modules/audio_coding/codecs/isac/main/source/filter_functions.h" -#include "modules/audio_coding/codecs/isac/main/source/isac_vad.h" - -/* window */ -/* Matlab generation code: - * t = (1:256)/257; r = 1-(1-t).^.45; w = sin(r*pi).^3; w = w/sum(w); plot((1:256)/8, w); grid; - * for k=1:16, fprintf(1, '%.8f, ', w(k*16 + (-15:0))); fprintf(1, '\n'); end - */ -static const double kLpcCorrWindow[WINLEN] = { - 0.00000000, 0.00000001, 0.00000004, 0.00000010, 0.00000020, - 0.00000035, 0.00000055, 0.00000083, 0.00000118, 0.00000163, - 0.00000218, 0.00000283, 0.00000361, 0.00000453, 0.00000558, 0.00000679, - 0.00000817, 0.00000973, 0.00001147, 0.00001342, 0.00001558, - 0.00001796, 0.00002058, 0.00002344, 0.00002657, 0.00002997, - 0.00003365, 0.00003762, 0.00004190, 0.00004651, 0.00005144, 0.00005673, - 0.00006236, 0.00006837, 0.00007476, 0.00008155, 0.00008875, - 0.00009636, 0.00010441, 0.00011290, 0.00012186, 0.00013128, - 0.00014119, 0.00015160, 0.00016252, 0.00017396, 0.00018594, 0.00019846, - 0.00021155, 0.00022521, 0.00023946, 0.00025432, 0.00026978, - 0.00028587, 0.00030260, 0.00031998, 0.00033802, 0.00035674, - 0.00037615, 0.00039626, 0.00041708, 0.00043863, 0.00046092, 0.00048396, - 0.00050775, 0.00053233, 0.00055768, 0.00058384, 0.00061080, - 0.00063858, 0.00066720, 0.00069665, 0.00072696, 0.00075813, - 0.00079017, 0.00082310, 0.00085692, 0.00089164, 0.00092728, 0.00096384, - 0.00100133, 0.00103976, 0.00107914, 0.00111947, 0.00116077, - 0.00120304, 0.00124630, 0.00129053, 0.00133577, 0.00138200, - 0.00142924, 0.00147749, 0.00152676, 0.00157705, 0.00162836, 0.00168070, - 0.00173408, 0.00178850, 0.00184395, 0.00190045, 0.00195799, - 0.00201658, 0.00207621, 0.00213688, 0.00219860, 0.00226137, - 0.00232518, 0.00239003, 0.00245591, 0.00252284, 0.00259079, 0.00265977, - 0.00272977, 0.00280078, 0.00287280, 0.00294582, 0.00301984, - 0.00309484, 0.00317081, 0.00324774, 0.00332563, 0.00340446, - 0.00348421, 0.00356488, 0.00364644, 0.00372889, 0.00381220, 0.00389636, - 0.00398135, 0.00406715, 0.00415374, 0.00424109, 0.00432920, - 0.00441802, 0.00450754, 0.00459773, 0.00468857, 0.00478001, - 0.00487205, 0.00496464, 0.00505775, 0.00515136, 0.00524542, 0.00533990, - 0.00543476, 0.00552997, 0.00562548, 0.00572125, 0.00581725, - 0.00591342, 0.00600973, 0.00610612, 0.00620254, 0.00629895, - 0.00639530, 0.00649153, 0.00658758, 0.00668341, 0.00677894, 0.00687413, - 0.00696891, 0.00706322, 0.00715699, 0.00725016, 0.00734266, - 0.00743441, 0.00752535, 0.00761540, 0.00770449, 0.00779254, - 0.00787947, 0.00796519, 0.00804963, 0.00813270, 0.00821431, 0.00829437, - 0.00837280, 0.00844949, 0.00852436, 0.00859730, 0.00866822, - 0.00873701, 0.00880358, 0.00886781, 0.00892960, 0.00898884, - 0.00904542, 0.00909923, 0.00915014, 0.00919805, 0.00924283, 0.00928436, - 0.00932252, 0.00935718, 0.00938821, 0.00941550, 0.00943890, - 0.00945828, 0.00947351, 0.00948446, 0.00949098, 0.00949294, - 0.00949020, 0.00948262, 0.00947005, 0.00945235, 0.00942938, 0.00940099, - 0.00936704, 0.00932738, 0.00928186, 0.00923034, 0.00917268, - 0.00910872, 0.00903832, 0.00896134, 0.00887763, 0.00878706, - 0.00868949, 0.00858478, 0.00847280, 0.00835343, 0.00822653, 0.00809199, - 0.00794970, 0.00779956, 0.00764145, 0.00747530, 0.00730103, - 0.00711857, 0.00692787, 0.00672888, 0.00652158, 0.00630597, - 0.00608208, 0.00584994, 0.00560962, 0.00536124, 0.00510493, 0.00484089, - 0.00456935, 0.00429062, 0.00400505, 0.00371310, 0.00341532, - 0.00311238, 0.00280511, 0.00249452, 0.00218184, 0.00186864, - 0.00155690, 0.00124918, 0.00094895, 0.00066112, 0.00039320, 0.00015881 -}; - -static void WebRtcIsac_GetVars(const double* input, - const int16_t* pitchGains_Q12, - double* oldEnergy, - double* varscale) { - double nrg[4], chng, pg; - int k; - - double pitchGains[4]={0,0,0,0};; - - /* Calculate energies of first and second frame halfs */ - nrg[0] = 0.0001; - for (k = QLOOKAHEAD/2; k < (FRAMESAMPLES_QUARTER + QLOOKAHEAD) / 2; k++) { - nrg[0] += input[k]*input[k]; - } - nrg[1] = 0.0001; - for ( ; k < (FRAMESAMPLES_HALF + QLOOKAHEAD) / 2; k++) { - nrg[1] += input[k]*input[k]; - } - nrg[2] = 0.0001; - for ( ; k < (FRAMESAMPLES*3/4 + QLOOKAHEAD) / 2; k++) { - nrg[2] += input[k]*input[k]; - } - nrg[3] = 0.0001; - for ( ; k < (FRAMESAMPLES + QLOOKAHEAD) / 2; k++) { - nrg[3] += input[k]*input[k]; - } - - /* Calculate average level change */ - chng = 0.25 * (fabs(10.0 * log10(nrg[3] / nrg[2])) + - fabs(10.0 * log10(nrg[2] / nrg[1])) + - fabs(10.0 * log10(nrg[1] / nrg[0])) + - fabs(10.0 * log10(nrg[0] / *oldEnergy))); - - - /* Find average pitch gain */ - pg = 0.0; - for (k=0; k<4; k++) - { - pitchGains[k] = ((float)pitchGains_Q12[k])/4096; - pg += pitchGains[k]; - } - pg *= 0.25; - - /* If pitch gain is low and energy constant - increase noise level*/ - /* Matlab code: - pg = 0:.01:.45; plot(pg, 0.0 + 1.0 * exp( -1.0 * exp(-200.0 * pg.*pg.*pg) / (1.0 + 0.4 * 0) )) - */ - *varscale = 0.0 + 1.0 * exp( -1.4 * exp(-200.0 * pg*pg*pg) / (1.0 + 0.4 * chng) ); - - *oldEnergy = nrg[3]; -} - -static void WebRtcIsac_GetVarsUB(const double* input, - double* oldEnergy, - double* varscale) { - double nrg[4], chng; - int k; - - /* Calculate energies of first and second frame halfs */ - nrg[0] = 0.0001; - for (k = 0; k < (FRAMESAMPLES_QUARTER) / 2; k++) { - nrg[0] += input[k]*input[k]; - } - nrg[1] = 0.0001; - for ( ; k < (FRAMESAMPLES_HALF) / 2; k++) { - nrg[1] += input[k]*input[k]; - } - nrg[2] = 0.0001; - for ( ; k < (FRAMESAMPLES*3/4) / 2; k++) { - nrg[2] += input[k]*input[k]; - } - nrg[3] = 0.0001; - for ( ; k < (FRAMESAMPLES) / 2; k++) { - nrg[3] += input[k]*input[k]; - } - - /* Calculate average level change */ - chng = 0.25 * (fabs(10.0 * log10(nrg[3] / nrg[2])) + - fabs(10.0 * log10(nrg[2] / nrg[1])) + - fabs(10.0 * log10(nrg[1] / nrg[0])) + - fabs(10.0 * log10(nrg[0] / *oldEnergy))); - - - /* If pitch gain is low and energy constant - increase noise level*/ - /* Matlab code: - pg = 0:.01:.45; plot(pg, 0.0 + 1.0 * exp( -1.0 * exp(-200.0 * pg.*pg.*pg) / (1.0 + 0.4 * 0) )) - */ - *varscale = exp( -1.4 / (1.0 + 0.4 * chng) ); - - *oldEnergy = nrg[3]; -} - -void WebRtcIsac_GetLpcCoefLb(double *inLo, double *inHi, MaskFiltstr *maskdata, - double signal_noise_ratio, const int16_t *pitchGains_Q12, - double *lo_coeff, double *hi_coeff) -{ - int k, n, j, pos1, pos2; - double varscale; - - double DataLo[WINLEN], DataHi[WINLEN]; - double corrlo[ORDERLO+2], corrlo2[ORDERLO+1]; - double corrhi[ORDERHI+1]; - double k_veclo[ORDERLO], k_vechi[ORDERHI]; - - double a_LO[ORDERLO+1], a_HI[ORDERHI+1]; - double tmp, res_nrg; - - double FwdA, FwdB; - - /* hearing threshold level in dB; higher value gives more noise */ - const double HearThresOffset = -28.0; - - /* bandwdith expansion factors for low- and high band */ - const double gammaLo = 0.9; - const double gammaHi = 0.8; - - /* less-noise-at-low-frequencies factor */ - double aa; - - - /* convert from dB to signal level */ - const double H_T_H = pow(10.0, 0.05 * HearThresOffset); - double S_N_R = pow(10.0, 0.05 * signal_noise_ratio) / 3.46; /* divide by sqrt(12) */ - - /* change quallevel depending on pitch gains and level fluctuations */ - WebRtcIsac_GetVars(inLo, pitchGains_Q12, &(maskdata->OldEnergy), &varscale); - - /* less-noise-at-low-frequencies factor */ - aa = 0.35 * (0.5 + 0.5 * varscale); - - /* replace data in buffer by new look-ahead data */ - for (pos1 = 0; pos1 < QLOOKAHEAD; pos1++) - maskdata->DataBufferLo[pos1 + WINLEN - QLOOKAHEAD] = inLo[pos1]; - - for (k = 0; k < SUBFRAMES; k++) { - - /* Update input buffer and multiply signal with window */ - for (pos1 = 0; pos1 < WINLEN - UPDATE/2; pos1++) { - maskdata->DataBufferLo[pos1] = maskdata->DataBufferLo[pos1 + UPDATE/2]; - maskdata->DataBufferHi[pos1] = maskdata->DataBufferHi[pos1 + UPDATE/2]; - DataLo[pos1] = maskdata->DataBufferLo[pos1] * kLpcCorrWindow[pos1]; - DataHi[pos1] = maskdata->DataBufferHi[pos1] * kLpcCorrWindow[pos1]; - } - pos2 = k * UPDATE/2; - for (n = 0; n < UPDATE/2; n++, pos1++) { - maskdata->DataBufferLo[pos1] = inLo[QLOOKAHEAD + pos2]; - maskdata->DataBufferHi[pos1] = inHi[pos2++]; - DataLo[pos1] = maskdata->DataBufferLo[pos1] * kLpcCorrWindow[pos1]; - DataHi[pos1] = maskdata->DataBufferHi[pos1] * kLpcCorrWindow[pos1]; - } - - /* Get correlation coefficients */ - WebRtcIsac_AutoCorr(corrlo, DataLo, WINLEN, ORDERLO+1); /* computing autocorrelation */ - WebRtcIsac_AutoCorr(corrhi, DataHi, WINLEN, ORDERHI); - - - /* less noise for lower frequencies, by filtering/scaling autocorrelation sequences */ - corrlo2[0] = (1.0+aa*aa) * corrlo[0] - 2.0*aa * corrlo[1]; - tmp = (1.0 + aa*aa); - for (n = 1; n <= ORDERLO; n++) { - corrlo2[n] = tmp * corrlo[n] - aa * (corrlo[n-1] + corrlo[n+1]); - } - tmp = (1.0+aa) * (1.0+aa); - for (n = 0; n <= ORDERHI; n++) { - corrhi[n] = tmp * corrhi[n]; - } - - /* add white noise floor */ - corrlo2[0] += 1e-6; - corrhi[0] += 1e-6; - - - FwdA = 0.01; - FwdB = 0.01; - - /* recursive filtering of correlation over subframes */ - for (n = 0; n <= ORDERLO; n++) { - maskdata->CorrBufLo[n] = FwdA * maskdata->CorrBufLo[n] + corrlo2[n]; - corrlo2[n] = ((1.0-FwdA)*FwdB) * maskdata->CorrBufLo[n] + (1.0-FwdB) * corrlo2[n]; - } - for (n = 0; n <= ORDERHI; n++) { - maskdata->CorrBufHi[n] = FwdA * maskdata->CorrBufHi[n] + corrhi[n]; - corrhi[n] = ((1.0-FwdA)*FwdB) * maskdata->CorrBufHi[n] + (1.0-FwdB) * corrhi[n]; - } - - /* compute prediction coefficients */ - WebRtcIsac_LevDurb(a_LO, k_veclo, corrlo2, ORDERLO); - WebRtcIsac_LevDurb(a_HI, k_vechi, corrhi, ORDERHI); - - /* bandwidth expansion */ - tmp = gammaLo; - for (n = 1; n <= ORDERLO; n++) { - a_LO[n] *= tmp; - tmp *= gammaLo; - } - - /* residual energy */ - res_nrg = 0.0; - for (j = 0; j <= ORDERLO; j++) { - for (n = 0; n <= j; n++) { - res_nrg += a_LO[j] * corrlo2[j-n] * a_LO[n]; - } - for (n = j+1; n <= ORDERLO; n++) { - res_nrg += a_LO[j] * corrlo2[n-j] * a_LO[n]; - } - } - - /* add hearing threshold and compute the gain */ - *lo_coeff++ = S_N_R / (sqrt(res_nrg) / varscale + H_T_H); - - /* copy coefficients to output array */ - for (n = 1; n <= ORDERLO; n++) { - *lo_coeff++ = a_LO[n]; - } - - - /* bandwidth expansion */ - tmp = gammaHi; - for (n = 1; n <= ORDERHI; n++) { - a_HI[n] *= tmp; - tmp *= gammaHi; - } - - /* residual energy */ - res_nrg = 0.0; - for (j = 0; j <= ORDERHI; j++) { - for (n = 0; n <= j; n++) { - res_nrg += a_HI[j] * corrhi[j-n] * a_HI[n]; - } - for (n = j+1; n <= ORDERHI; n++) { - res_nrg += a_HI[j] * corrhi[n-j] * a_HI[n]; - } - } - - /* add hearing threshold and compute of the gain */ - *hi_coeff++ = S_N_R / (sqrt(res_nrg) / varscale + H_T_H); - - /* copy coefficients to output array */ - for (n = 1; n <= ORDERHI; n++) { - *hi_coeff++ = a_HI[n]; - } - } -} - - - -/****************************************************************************** - * WebRtcIsac_GetLpcCoefUb() - * - * Compute LP coefficients and correlation coefficients. At 12 kHz LP - * coefficients of the first and the last sub-frame is computed. At 16 kHz - * LP coefficients of 4th, 8th and 12th sub-frames are computed. We always - * compute correlation coefficients of all sub-frames. - * - * Inputs: - * -inSignal : Input signal - * -maskdata : a structure keeping signal from previous frame. - * -bandwidth : specifies if the codec is in 0-16 kHz mode or - * 0-12 kHz mode. - * - * Outputs: - * -lpCoeff : pointer to a buffer where A-polynomials are - * written to (first coeff is 1 and it is not - * written) - * -corrMat : a matrix where correlation coefficients of each - * sub-frame are written to one row. - * -varscale : a scale used to compute LPC gains. - */ -void -WebRtcIsac_GetLpcCoefUb( - double* inSignal, - MaskFiltstr* maskdata, - double* lpCoeff, - double corrMat[][UB_LPC_ORDER + 1], - double* varscale, - int16_t bandwidth) -{ - int frameCntr, activeFrameCntr, n, pos1, pos2; - int16_t criterion1; - int16_t criterion2; - int16_t numSubFrames = SUBFRAMES * (1 + (bandwidth == isac16kHz)); - double data[WINLEN]; - double corrSubFrame[UB_LPC_ORDER+2]; - double reflecCoeff[UB_LPC_ORDER]; - - double aPolynom[UB_LPC_ORDER+1]; - double tmp; - - /* bandwdith expansion factors */ - const double gamma = 0.9; - - /* change quallevel depending on pitch gains and level fluctuations */ - WebRtcIsac_GetVarsUB(inSignal, &(maskdata->OldEnergy), varscale); - - /* replace data in buffer by new look-ahead data */ - for(frameCntr = 0, activeFrameCntr = 0; frameCntr < numSubFrames; - frameCntr++) - { - if(frameCntr == SUBFRAMES) - { - // we are in 16 kHz - varscale++; - WebRtcIsac_GetVarsUB(&inSignal[FRAMESAMPLES_HALF], - &(maskdata->OldEnergy), varscale); - } - /* Update input buffer and multiply signal with window */ - for(pos1 = 0; pos1 < WINLEN - UPDATE/2; pos1++) - { - maskdata->DataBufferLo[pos1] = maskdata->DataBufferLo[pos1 + - UPDATE/2]; - data[pos1] = maskdata->DataBufferLo[pos1] * kLpcCorrWindow[pos1]; - } - pos2 = frameCntr * UPDATE/2; - for(n = 0; n < UPDATE/2; n++, pos1++, pos2++) - { - maskdata->DataBufferLo[pos1] = inSignal[pos2]; - data[pos1] = maskdata->DataBufferLo[pos1] * kLpcCorrWindow[pos1]; - } - - /* Get correlation coefficients */ - /* computing autocorrelation */ - WebRtcIsac_AutoCorr(corrSubFrame, data, WINLEN, UB_LPC_ORDER+1); - memcpy(corrMat[frameCntr], corrSubFrame, - (UB_LPC_ORDER+1)*sizeof(double)); - - criterion1 = ((frameCntr == 0) || (frameCntr == (SUBFRAMES - 1))) && - (bandwidth == isac12kHz); - criterion2 = (((frameCntr+1) % 4) == 0) && - (bandwidth == isac16kHz); - if(criterion1 || criterion2) - { - /* add noise */ - corrSubFrame[0] += 1e-6; - /* compute prediction coefficients */ - WebRtcIsac_LevDurb(aPolynom, reflecCoeff, corrSubFrame, - UB_LPC_ORDER); - - /* bandwidth expansion */ - tmp = gamma; - for (n = 1; n <= UB_LPC_ORDER; n++) - { - *lpCoeff++ = aPolynom[n] * tmp; - tmp *= gamma; - } - activeFrameCntr++; - } - } -} - - - -/****************************************************************************** - * WebRtcIsac_GetLpcGain() - * - * Compute the LPC gains for each sub-frame, given the LPC of each sub-frame - * and the corresponding correlation coefficients. - * - * Inputs: - * -signal_noise_ratio : the desired SNR in dB. - * -numVecs : number of sub-frames - * -corrMat : a matrix of correlation coefficients where - * each row is a set of correlation coefficients of - * one sub-frame. - * -varscale : a scale computed when WebRtcIsac_GetLpcCoefUb() - * is called. - * - * Outputs: - * -gain : pointer to a buffer where LP gains are written. - * - */ -void -WebRtcIsac_GetLpcGain( - double signal_noise_ratio, - const double* filtCoeffVecs, - int numVecs, - double* gain, - double corrMat[][UB_LPC_ORDER + 1], - const double* varscale) -{ - int16_t j, n; - int16_t subFrameCntr; - double aPolynom[ORDERLO + 1]; - double res_nrg; - - const double HearThresOffset = -28.0; - const double H_T_H = pow(10.0, 0.05 * HearThresOffset); - /* divide by sqrt(12) = 3.46 */ - const double S_N_R = pow(10.0, 0.05 * signal_noise_ratio) / 3.46; - - aPolynom[0] = 1; - for(subFrameCntr = 0; subFrameCntr < numVecs; subFrameCntr++) - { - if(subFrameCntr == SUBFRAMES) - { - // we are in second half of a SWB frame. use new varscale - varscale++; - } - memcpy(&aPolynom[1], &filtCoeffVecs[(subFrameCntr * (UB_LPC_ORDER + 1)) + - 1], sizeof(double) * UB_LPC_ORDER); - - /* residual energy */ - res_nrg = 0.0; - for(j = 0; j <= UB_LPC_ORDER; j++) - { - for(n = 0; n <= j; n++) - { - res_nrg += aPolynom[j] * corrMat[subFrameCntr][j-n] * - aPolynom[n]; - } - for(n = j+1; n <= UB_LPC_ORDER; n++) - { - res_nrg += aPolynom[j] * corrMat[subFrameCntr][n-j] * - aPolynom[n]; - } - } - - /* add hearing threshold and compute the gain */ - gain[subFrameCntr] = S_N_R / (sqrt(res_nrg) / *varscale + H_T_H); - } -} diff --git a/modules/audio_coding/codecs/isac/main/source/lpc_analysis.h b/modules/audio_coding/codecs/isac/main/source/lpc_analysis.h deleted file mode 100644 index 5503e2d49b..0000000000 --- a/modules/audio_coding/codecs/isac/main/source/lpc_analysis.h +++ /dev/null @@ -1,46 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/* - * lpc_analysis.h - * - * LPC functions - * - */ - -#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_LPC_ANALYSIS_H_ -#define MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_LPC_ANALYSIS_H_ - -#include "modules/audio_coding/codecs/isac/main/source/settings.h" -#include "modules/audio_coding/codecs/isac/main/source/structs.h" - -void WebRtcIsac_GetLpcCoefLb(double* inLo, - double* inHi, - MaskFiltstr* maskdata, - double signal_noise_ratio, - const int16_t* pitchGains_Q12, - double* lo_coeff, - double* hi_coeff); - -void WebRtcIsac_GetLpcGain(double signal_noise_ratio, - const double* filtCoeffVecs, - int numVecs, - double* gain, - double corrLo[][UB_LPC_ORDER + 1], - const double* varscale); - -void WebRtcIsac_GetLpcCoefUb(double* inSignal, - MaskFiltstr* maskdata, - double* lpCoeff, - double corr[][UB_LPC_ORDER + 1], - double* varscale, - int16_t bandwidth); - -#endif /* MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_LPC_ANALYIS_H_ */ diff --git a/modules/audio_coding/codecs/isac/main/source/lpc_gain_swb_tables.c b/modules/audio_coding/codecs/isac/main/source/lpc_gain_swb_tables.c deleted file mode 100644 index 670754065f..0000000000 --- a/modules/audio_coding/codecs/isac/main/source/lpc_gain_swb_tables.c +++ /dev/null @@ -1,136 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/* - * SWB_KLT_Tables_LPCGain.c - * - * This file defines tables used for entropy coding of LPC Gain - * of upper-band. - * - */ - -#include "modules/audio_coding/codecs/isac/main/source/lpc_gain_swb_tables.h" -#include "modules/audio_coding/codecs/isac/main/source/settings.h" - -const double WebRtcIsac_kQSizeLpcGain = 0.100000; - -const double WebRtcIsac_kMeanLpcGain = -3.3822; - -/* -* The smallest reconstruction points for quantiztion of -* LPC gains. -*/ -const double WebRtcIsac_kLeftRecPointLpcGain[SUBFRAMES] = -{ - -0.800000, -1.000000, -1.200000, -2.200000, -3.000000, -12.700000 -}; - -/* -* Number of reconstruction points of quantizers for LPC Gains. -*/ -const int16_t WebRtcIsac_kNumQCellLpcGain[SUBFRAMES] = -{ - 17, 20, 25, 45, 77, 170 -}; -/* -* Starting index for entropy decoder to search for the right interval, -* one entry per LAR coefficient -*/ -const uint16_t WebRtcIsac_kLpcGainEntropySearch[SUBFRAMES] = -{ - 8, 10, 12, 22, 38, 85 -}; - -/* -* The following 6 vectors define CDF of 6 decorrelated LPC -* gains. -*/ -const uint16_t WebRtcIsac_kLpcGainCdfVec0[18] = -{ - 0, 10, 27, 83, 234, 568, 1601, 4683, 16830, 57534, 63437, - 64767, 65229, 65408, 65483, 65514, 65527, 65535 -}; - -const uint16_t WebRtcIsac_kLpcGainCdfVec1[21] = -{ - 0, 15, 33, 84, 185, 385, 807, 1619, 3529, 7850, 19488, - 51365, 62437, 64548, 65088, 65304, 65409, 65484, 65507, 65522, 65535 -}; - -const uint16_t WebRtcIsac_kLpcGainCdfVec2[26] = -{ - 0, 15, 29, 54, 89, 145, 228, 380, 652, 1493, 4260, - 12359, 34133, 50749, 57224, 60814, 62927, 64078, 64742, 65103, 65311, 65418, - 65473, 65509, 65521, 65535 -}; - -const uint16_t WebRtcIsac_kLpcGainCdfVec3[46] = -{ - 0, 8, 12, 16, 26, 42, 56, 76, 111, 164, 247, - 366, 508, 693, 1000, 1442, 2155, 3188, 4854, 7387, 11249, 17617, - 30079, 46711, 56291, 60127, 62140, 63258, 63954, 64384, 64690, 64891, 65031, - 65139, 65227, 65293, 65351, 65399, 65438, 65467, 65492, 65504, 65510, 65518, - 65523, 65535 -}; - -const uint16_t WebRtcIsac_kLpcGainCdfVec4[78] = -{ - 0, 17, 29, 39, 51, 70, 104, 154, 234, 324, 443, - 590, 760, 971, 1202, 1494, 1845, 2274, 2797, 3366, 4088, 4905, - 5899, 7142, 8683, 10625, 12983, 16095, 20637, 28216, 38859, 47237, 51537, - 54150, 56066, 57583, 58756, 59685, 60458, 61103, 61659, 62144, 62550, 62886, - 63186, 63480, 63743, 63954, 64148, 64320, 64467, 64600, 64719, 64837, 64939, - 65014, 65098, 65160, 65211, 65250, 65290, 65325, 65344, 65366, 65391, 65410, - 65430, 65447, 65460, 65474, 65487, 65494, 65501, 65509, 65513, 65518, 65520, - 65535 -}; - -const uint16_t WebRtcIsac_kLpcGainCdfVec5[171] = -{ - 0, 10, 12, 14, 16, 18, 23, 29, 35, 42, 51, - 58, 65, 72, 78, 87, 96, 103, 111, 122, 134, 150, - 167, 184, 202, 223, 244, 265, 289, 315, 346, 379, 414, - 450, 491, 532, 572, 613, 656, 700, 751, 802, 853, 905, - 957, 1021, 1098, 1174, 1250, 1331, 1413, 1490, 1565, 1647, 1730, - 1821, 1913, 2004, 2100, 2207, 2314, 2420, 2532, 2652, 2783, 2921, - 3056, 3189, 3327, 3468, 3640, 3817, 3993, 4171, 4362, 4554, 4751, - 4948, 5142, 5346, 5566, 5799, 6044, 6301, 6565, 6852, 7150, 7470, - 7797, 8143, 8492, 8835, 9181, 9547, 9919, 10315, 10718, 11136, 11566, - 12015, 12482, 12967, 13458, 13953, 14432, 14903, 15416, 15936, 16452, 16967, - 17492, 18024, 18600, 19173, 19736, 20311, 20911, 21490, 22041, 22597, 23157, - 23768, 24405, 25034, 25660, 26280, 26899, 27614, 28331, 29015, 29702, 30403, - 31107, 31817, 32566, 33381, 34224, 35099, 36112, 37222, 38375, 39549, 40801, - 42074, 43350, 44626, 45982, 47354, 48860, 50361, 51845, 53312, 54739, 56026, - 57116, 58104, 58996, 59842, 60658, 61488, 62324, 63057, 63769, 64285, 64779, - 65076, 65344, 65430, 65500, 65517, 65535 -}; - -/* -* An array of pointers to CDFs of decorrelated LPC Gains -*/ -const uint16_t* WebRtcIsac_kLpcGainCdfMat[SUBFRAMES] = -{ - WebRtcIsac_kLpcGainCdfVec0, WebRtcIsac_kLpcGainCdfVec1, - WebRtcIsac_kLpcGainCdfVec2, WebRtcIsac_kLpcGainCdfVec3, - WebRtcIsac_kLpcGainCdfVec4, WebRtcIsac_kLpcGainCdfVec5 -}; - -/* -* A matrix to decorrellate LPC gains of subframes. -*/ -const double WebRtcIsac_kLpcGainDecorrMat[SUBFRAMES][SUBFRAMES] = -{ - {-0.150860, 0.327872, 0.367220, 0.504613, 0.559270, 0.409234}, - { 0.457128, -0.613591, -0.289283, -0.029734, 0.393760, 0.418240}, - {-0.626043, 0.136489, -0.439118, -0.448323, 0.135987, 0.420869}, - { 0.526617, 0.480187, 0.242552, -0.488754, -0.158713, 0.411331}, - {-0.302587, -0.494953, 0.588112, -0.063035, -0.404290, 0.387510}, - { 0.086378, 0.147714, -0.428875, 0.548300, -0.570121, 0.401391} -}; diff --git a/modules/audio_coding/codecs/isac/main/source/lpc_gain_swb_tables.h b/modules/audio_coding/codecs/isac/main/source/lpc_gain_swb_tables.h deleted file mode 100644 index 39c4a24ef4..0000000000 --- a/modules/audio_coding/codecs/isac/main/source/lpc_gain_swb_tables.h +++ /dev/null @@ -1,50 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/* - * SWB_KLT_Tables_LPCGain.h - * - * This file declares tables used for entropy coding of LPC Gain - * of upper-band. - * - */ - -#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_LPC_GAIN_SWB_TABLES_H_ -#define MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_LPC_GAIN_SWB_TABLES_H_ - -#include - -#include "modules/audio_coding/codecs/isac/main/source/settings.h" - -extern const double WebRtcIsac_kQSizeLpcGain; - -extern const double WebRtcIsac_kLeftRecPointLpcGain[SUBFRAMES]; - -extern const int16_t WebRtcIsac_kNumQCellLpcGain[SUBFRAMES]; - -extern const uint16_t WebRtcIsac_kLpcGainEntropySearch[SUBFRAMES]; - -extern const uint16_t WebRtcIsac_kLpcGainCdfVec0[18]; - -extern const uint16_t WebRtcIsac_kLpcGainCdfVec1[21]; - -extern const uint16_t WebRtcIsac_kLpcGainCdfVec2[26]; - -extern const uint16_t WebRtcIsac_kLpcGainCdfVec3[46]; - -extern const uint16_t WebRtcIsac_kLpcGainCdfVec4[78]; - -extern const uint16_t WebRtcIsac_kLpcGainCdfVec5[171]; - -extern const uint16_t* WebRtcIsac_kLpcGainCdfMat[SUBFRAMES]; - -extern const double WebRtcIsac_kLpcGainDecorrMat[SUBFRAMES][SUBFRAMES]; - -#endif // MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_LPC_GAIN_SWB_TABLES_H_ diff --git a/modules/audio_coding/codecs/isac/main/source/lpc_shape_swb12_tables.c b/modules/audio_coding/codecs/isac/main/source/lpc_shape_swb12_tables.c deleted file mode 100644 index e3600a7fab..0000000000 --- a/modules/audio_coding/codecs/isac/main/source/lpc_shape_swb12_tables.c +++ /dev/null @@ -1,158 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/* - * SWB_KLT_Tables.c - * - * This file defines tables used for entropy coding of LPC shape of - * upper-band signal if the bandwidth is 12 kHz. - * - */ - -#include "modules/audio_coding/codecs/isac/main/source/lpc_shape_swb12_tables.h" -#include "modules/audio_coding/codecs/isac/main/source/settings.h" - -/* -* Mean value of LAR -*/ -const double WebRtcIsac_kMeanLarUb12[UB_LPC_ORDER] = -{ - 0.03748928306641, 0.09453441192543, -0.01112522344398, 0.03800237516842 -}; - -/* -* A rotation matrix to decorrelate intra-vector correlation, -* i.e. correlation among components of LAR vector. -*/ -const double WebRtcIsac_kIntraVecDecorrMatUb12[UB_LPC_ORDER][UB_LPC_ORDER] = -{ - {-0.00075365493856, -0.05809964887743, -0.23397966154116, 0.97050367376411}, - { 0.00625021257734, -0.17299965610679, 0.95977735920651, 0.22104179375008}, - { 0.20543384258374, -0.96202143495696, -0.15301870801552, -0.09432375099565}, - {-0.97865075648479, -0.20300322280841, -0.02581111653779, -0.01913568980258} -}; - -/* -* A rotation matrix to remove correlation among LAR coefficients -* of different LAR vectors. One might guess that decorrelation matrix -* for the first component should differ from the second component -* but we haven't observed a significant benefit of having different -* decorrelation matrices for different components. -*/ -const double WebRtcIsac_kInterVecDecorrMatUb12 -[UB_LPC_VEC_PER_FRAME][UB_LPC_VEC_PER_FRAME] = -{ - { 0.70650597970460, -0.70770707262373}, - {-0.70770707262373, -0.70650597970460} -}; - -/* -* LAR quantization step-size. -*/ -const double WebRtcIsac_kLpcShapeQStepSizeUb12 = 0.150000; - -/* -* The smallest reconstruction points for quantiztion of LAR coefficients. -*/ -const double WebRtcIsac_kLpcShapeLeftRecPointUb12 -[UB_LPC_ORDER*UB_LPC_VEC_PER_FRAME] = -{ - -0.900000, -1.050000, -1.350000, -1.800000, -1.350000, -1.650000, - -2.250000, -3.450000 -}; - -/* -* Number of reconstruction points of quantizers for LAR coefficients. -*/ -const int16_t WebRtcIsac_kLpcShapeNumRecPointUb12 -[UB_LPC_ORDER * UB_LPC_VEC_PER_FRAME] = -{ - 13, 15, 19, 27, 19, 24, 32, 48 -}; - -/* -* Starting index for entropy decoder to search for the right interval, -* one entry per LAR coefficient -*/ -const uint16_t WebRtcIsac_kLpcShapeEntropySearchUb12 -[UB_LPC_ORDER * UB_LPC_VEC_PER_FRAME] = -{ - 6, 7, 9, 13, 9, 12, 16, 24 -}; - -/* -* The following 8 vectors define CDF of 8 decorrelated LAR -* coefficients. -*/ -const uint16_t WebRtcIsac_kLpcShapeCdfVec0Ub12[14] = -{ - 0, 13, 95, 418, 1687, 6498, 21317, 44200, 59029, 63849, 65147, - 65449, 65525, 65535 -}; - -const uint16_t WebRtcIsac_kLpcShapeCdfVec1Ub12[16] = -{ - 0, 10, 59, 255, 858, 2667, 8200, 22609, 42988, 57202, 62947, - 64743, 65308, 65476, 65522, 65535 -}; - -const uint16_t WebRtcIsac_kLpcShapeCdfVec2Ub12[20] = -{ - 0, 18, 40, 118, 332, 857, 2017, 4822, 11321, 24330, 41279, - 54342, 60637, 63394, 64659, 65184, 65398, 65482, 65518, 65535 -}; - -const uint16_t WebRtcIsac_kLpcShapeCdfVec3Ub12[28] = -{ - 0, 21, 38, 90, 196, 398, 770, 1400, 2589, 4650, 8211, - 14933, 26044, 39592, 50814, 57452, 60971, 62884, 63995, 64621, 65019, 65273, - 65410, 65480, 65514, 65522, 65531, 65535 -}; - -const uint16_t WebRtcIsac_kLpcShapeCdfVec4Ub12[20] = -{ - 0, 7, 46, 141, 403, 969, 2132, 4649, 10633, 24902, 43254, - 54665, 59928, 62674, 64173, 64938, 65293, 65464, 65523, 65535 -}; - -const uint16_t WebRtcIsac_kLpcShapeCdfVec5Ub12[25] = -{ - 0, 7, 22, 72, 174, 411, 854, 1737, 3545, 6774, 13165, - 25221, 40980, 52821, 58714, 61706, 63472, 64437, 64989, 65287, 65430, 65503, - 65525, 65529, 65535 -}; - -const uint16_t WebRtcIsac_kLpcShapeCdfVec6Ub12[33] = -{ - 0, 11, 21, 36, 65, 128, 228, 401, 707, 1241, 2126, - 3589, 6060, 10517, 18853, 31114, 42477, 49770, 54271, 57467, 59838, 61569, - 62831, 63772, 64433, 64833, 65123, 65306, 65419, 65466, 65499, 65519, 65535 -}; - -const uint16_t WebRtcIsac_kLpcShapeCdfVec7Ub12[49] = -{ - 0, 14, 34, 67, 107, 167, 245, 326, 449, 645, 861, - 1155, 1508, 2003, 2669, 3544, 4592, 5961, 7583, 9887, 13256, 18765, - 26519, 34077, 40034, 44349, 47795, 50663, 53262, 55473, 57458, 59122, 60592, - 61742, 62690, 63391, 63997, 64463, 64794, 65045, 65207, 65309, 65394, 65443, - 65478, 65504, 65514, 65523, 65535 -}; - -/* -* An array of pointers to CDFs of decorrelated LARs -*/ -const uint16_t* WebRtcIsac_kLpcShapeCdfMatUb12 -[UB_LPC_ORDER * UB_LPC_VEC_PER_FRAME] = -{ - WebRtcIsac_kLpcShapeCdfVec0Ub12, WebRtcIsac_kLpcShapeCdfVec1Ub12, - WebRtcIsac_kLpcShapeCdfVec2Ub12, WebRtcIsac_kLpcShapeCdfVec3Ub12, - WebRtcIsac_kLpcShapeCdfVec4Ub12, WebRtcIsac_kLpcShapeCdfVec5Ub12, - WebRtcIsac_kLpcShapeCdfVec6Ub12, WebRtcIsac_kLpcShapeCdfVec7Ub12 -}; diff --git a/modules/audio_coding/codecs/isac/main/source/lpc_shape_swb12_tables.h b/modules/audio_coding/codecs/isac/main/source/lpc_shape_swb12_tables.h deleted file mode 100644 index 7448a1e76b..0000000000 --- a/modules/audio_coding/codecs/isac/main/source/lpc_shape_swb12_tables.h +++ /dev/null @@ -1,66 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/* - * lpc_shape_swb12_tables.h - * - * This file declares tables used for entropy coding of LPC shape of - * upper-band signal if the bandwidth is 12 kHz. - * - */ - -#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_LPC_SHAPE_SWB12_TABLES_H_ -#define MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_LPC_SHAPE_SWB12_TABLES_H_ - -#include - -#include "modules/audio_coding/codecs/isac/main/source/settings.h" - -extern const double WebRtcIsac_kMeanLarUb12[UB_LPC_ORDER]; - -extern const double WebRtcIsac_kMeanLpcGain; - -extern const double WebRtcIsac_kIntraVecDecorrMatUb12[UB_LPC_ORDER] - [UB_LPC_ORDER]; - -extern const double WebRtcIsac_kInterVecDecorrMatUb12[UB_LPC_VEC_PER_FRAME] - [UB_LPC_VEC_PER_FRAME]; - -extern const double WebRtcIsac_kLpcShapeQStepSizeUb12; - -extern const double - WebRtcIsac_kLpcShapeLeftRecPointUb12[UB_LPC_ORDER * UB_LPC_VEC_PER_FRAME]; - -extern const int16_t - WebRtcIsac_kLpcShapeNumRecPointUb12[UB_LPC_ORDER * UB_LPC_VEC_PER_FRAME]; - -extern const uint16_t - WebRtcIsac_kLpcShapeEntropySearchUb12[UB_LPC_ORDER * UB_LPC_VEC_PER_FRAME]; - -extern const uint16_t WebRtcIsac_kLpcShapeCdfVec0Ub12[14]; - -extern const uint16_t WebRtcIsac_kLpcShapeCdfVec1Ub12[16]; - -extern const uint16_t WebRtcIsac_kLpcShapeCdfVec2Ub12[20]; - -extern const uint16_t WebRtcIsac_kLpcShapeCdfVec3Ub12[28]; - -extern const uint16_t WebRtcIsac_kLpcShapeCdfVec4Ub12[20]; - -extern const uint16_t WebRtcIsac_kLpcShapeCdfVec5Ub12[25]; - -extern const uint16_t WebRtcIsac_kLpcShapeCdfVec6Ub12[33]; - -extern const uint16_t WebRtcIsac_kLpcShapeCdfVec7Ub12[49]; - -extern const uint16_t* - WebRtcIsac_kLpcShapeCdfMatUb12[UB_LPC_ORDER * UB_LPC_VEC_PER_FRAME]; - -#endif // MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_LPC_SHAPE_SWB12_TABLES_H_ diff --git a/modules/audio_coding/codecs/isac/main/source/lpc_shape_swb16_tables.c b/modules/audio_coding/codecs/isac/main/source/lpc_shape_swb16_tables.c deleted file mode 100644 index 59617fd274..0000000000 --- a/modules/audio_coding/codecs/isac/main/source/lpc_shape_swb16_tables.c +++ /dev/null @@ -1,247 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/* - * SWB16_KLT_Tables.c - * - * This file defines tables used for entropy coding of LPC shape of - * upper-band signal if the bandwidth is 16 kHz. - * - */ - -#include "modules/audio_coding/codecs/isac/main/source/lpc_shape_swb16_tables.h" -#include "modules/audio_coding/codecs/isac/main/source/settings.h" - -/* -* Mean value of LAR -*/ -const double WebRtcIsac_kMeanLarUb16[UB_LPC_ORDER] = -{ -0.454978, 0.364747, 0.102999, 0.104523 -}; - -/* -* A rotation matrix to decorrelate intra-vector correlation, -* i.e. correlation among components of LAR vector. -*/ -const double WebRtcIsac_kIintraVecDecorrMatUb16[UB_LPC_ORDER][UB_LPC_ORDER] = -{ - {-0.020528, -0.085858, -0.002431, 0.996093}, - {-0.033155, 0.036102, 0.998786, 0.004866}, - { 0.202627, 0.974853, -0.028940, 0.088132}, - {-0.978479, 0.202454, -0.039785, -0.002811} -}; - -/* -* A rotation matrix to remove correlation among LAR coefficients -* of different LAR vectors. One might guess that decorrelation matrix -* for the first component should differ from the second component -* but we haven't observed a significant benefit of having different -* decorrelation matrices for different components. -*/ -const double WebRtcIsac_kInterVecDecorrMatUb16 -[UB16_LPC_VEC_PER_FRAME][UB16_LPC_VEC_PER_FRAME] = -{ - { 0.291675, -0.515786, 0.644927, 0.482658}, - {-0.647220, 0.479712, 0.289556, 0.516856}, - { 0.643084, 0.485489, -0.289307, 0.516763}, - {-0.287185, -0.517823, -0.645389, 0.482553} -}; - -/* -* The following 16 vectors define CDF of 16 decorrelated LAR -* coefficients. -*/ -const uint16_t WebRtcIsac_kLpcShapeCdfVec01Ub16[14] = -{ - 0, 2, 20, 159, 1034, 5688, 20892, 44653, - 59849, 64485, 65383, 65518, 65534, 65535 -}; - -const uint16_t WebRtcIsac_kLpcShapeCdfVec1Ub16[16] = -{ - 0, 1, 7, 43, 276, 1496, 6681, 21653, - 43891, 58859, 64022, 65248, 65489, 65529, 65534, 65535 -}; - -const uint16_t WebRtcIsac_kLpcShapeCdfVec2Ub16[18] = -{ - 0, 1, 9, 54, 238, 933, 3192, 9461, - 23226, 42146, 56138, 62413, 64623, 65300, 65473, 65521, - 65533, 65535 -}; - -const uint16_t WebRtcIsac_kLpcShapeCdfVec3Ub16[30] = -{ - 0, 2, 4, 8, 17, 36, 75, 155, - 329, 683, 1376, 2662, 5047, 9508, 17526, 29027, - 40363, 48997, 55096, 59180, 61789, 63407, 64400, 64967, - 65273, 65429, 65497, 65526, 65534, 65535 -}; - -const uint16_t WebRtcIsac_kLpcShapeCdfVec4Ub16[16] = -{ - 0, 1, 10, 63, 361, 1785, 7407, 22242, - 43337, 58125, 63729, 65181, 65472, 65527, 65534, 65535 -}; - -const uint16_t WebRtcIsac_kLpcShapeCdfVec5Ub16[17] = -{ - 0, 1, 7, 29, 134, 599, 2443, 8590, - 22962, 42635, 56911, 63060, 64940, 65408, 65513, 65531, - 65535 -}; - -const uint16_t WebRtcIsac_kLpcShapeCdfVec6Ub16[21] = -{ - 0, 1, 5, 16, 57, 191, 611, 1808, - 4847, 11755, 24612, 40910, 53789, 60698, 63729, 64924, - 65346, 65486, 65523, 65532, 65535 -}; - -const uint16_t WebRtcIsac_kLpcShapeCdfVec7Ub16[36] = -{ - 0, 1, 4, 12, 25, 55, 104, 184, - 314, 539, 926, 1550, 2479, 3861, 5892, 8845, - 13281, 20018, 29019, 38029, 45581, 51557, 56057, 59284, - 61517, 63047, 64030, 64648, 65031, 65261, 65402, 65480, - 65518, 65530, 65534, 65535 -}; - -const uint16_t WebRtcIsac_kLpcShapeCdfVec8Ub16[21] = -{ - 0, 1, 2, 7, 26, 103, 351, 1149, - 3583, 10204, 23846, 41711, 55361, 61917, 64382, 65186, - 65433, 65506, 65528, 65534, 65535 -}; - -const uint16_t WebRtcIsac_kLpcShapeCdfVec01Ub160[21] = -{ - 0, 6, 19, 63, 205, 638, 1799, 4784, - 11721, 24494, 40803, 53805, 60886, 63822, 64931, 65333, - 65472, 65517, 65530, 65533, 65535 -}; - -const uint16_t WebRtcIsac_kLpcShapeCdfVec01Ub161[28] = -{ - 0, 1, 3, 11, 31, 86, 221, 506, - 1101, 2296, 4486, 8477, 15356, 26079, 38941, 49952, - 57165, 61257, 63426, 64549, 65097, 65351, 65463, 65510, - 65526, 65532, 65534, 65535 -}; - -const uint16_t WebRtcIsac_kLpcShapeCdfVec01Ub162[55] = -{ - 0, 3, 12, 23, 42, 65, 89, 115, - 150, 195, 248, 327, 430, 580, 784, 1099, - 1586, 2358, 3651, 5899, 9568, 14312, 19158, 23776, - 28267, 32663, 36991, 41153, 45098, 48680, 51870, 54729, - 57141, 59158, 60772, 62029, 63000, 63761, 64322, 64728, - 65000, 65192, 65321, 65411, 65463, 65496, 65514, 65523, - 65527, 65529, 65531, 65532, 65533, 65534, 65535 -}; - -const uint16_t WebRtcIsac_kLpcShapeCdfVec01Ub163[26] = -{ - 0, 2, 4, 10, 21, 48, 114, 280, - 701, 1765, 4555, 11270, 24267, 41213, 54285, 61003, - 63767, 64840, 65254, 65421, 65489, 65514, 65526, 65532, - 65534, 65535 -}; - -const uint16_t WebRtcIsac_kLpcShapeCdfVec01Ub164[28] = -{ - 0, 1, 3, 6, 15, 36, 82, 196, - 453, 1087, 2557, 5923, 13016, 25366, 40449, 52582, - 59539, 62896, 64389, 65033, 65316, 65442, 65494, 65519, - 65529, 65533, 65534, 65535 -}; - -const uint16_t WebRtcIsac_kLpcShapeCdfVec01Ub165[34] = -{ - 0, 2, 4, 8, 18, 35, 73, 146, - 279, 524, 980, 1789, 3235, 5784, 10040, 16998, - 27070, 38543, 48499, 55421, 59712, 62257, 63748, 64591, - 65041, 65278, 65410, 65474, 65508, 65522, 65530, 65533, - 65534, 65535 -}; - -const uint16_t WebRtcIsac_kLpcShapeCdfVec01Ub166[71] = -{ - 0, 1, 2, 6, 13, 26, 55, 92, - 141, 191, 242, 296, 355, 429, 522, 636, - 777, 947, 1162, 1428, 1753, 2137, 2605, 3140, - 3743, 4409, 5164, 6016, 6982, 8118, 9451, 10993, - 12754, 14810, 17130, 19780, 22864, 26424, 30547, 35222, - 40140, 44716, 48698, 52056, 54850, 57162, 59068, 60643, - 61877, 62827, 63561, 64113, 64519, 64807, 65019, 65167, - 65272, 65343, 65399, 65440, 65471, 65487, 65500, 65509, - 65518, 65524, 65527, 65531, 65533, 65534, 65535 -}; - -/* -* An array of pointers to CDFs of decorrelated LARs -*/ -const uint16_t* WebRtcIsac_kLpcShapeCdfMatUb16 -[UB_LPC_ORDER * UB16_LPC_VEC_PER_FRAME] = { - WebRtcIsac_kLpcShapeCdfVec01Ub16, - WebRtcIsac_kLpcShapeCdfVec1Ub16, - WebRtcIsac_kLpcShapeCdfVec2Ub16, - WebRtcIsac_kLpcShapeCdfVec3Ub16, - WebRtcIsac_kLpcShapeCdfVec4Ub16, - WebRtcIsac_kLpcShapeCdfVec5Ub16, - WebRtcIsac_kLpcShapeCdfVec6Ub16, - WebRtcIsac_kLpcShapeCdfVec7Ub16, - WebRtcIsac_kLpcShapeCdfVec8Ub16, - WebRtcIsac_kLpcShapeCdfVec01Ub160, - WebRtcIsac_kLpcShapeCdfVec01Ub161, - WebRtcIsac_kLpcShapeCdfVec01Ub162, - WebRtcIsac_kLpcShapeCdfVec01Ub163, - WebRtcIsac_kLpcShapeCdfVec01Ub164, - WebRtcIsac_kLpcShapeCdfVec01Ub165, - WebRtcIsac_kLpcShapeCdfVec01Ub166 -}; - -/* -* The smallest reconstruction points for quantiztion of LAR coefficients. -*/ -const double WebRtcIsac_kLpcShapeLeftRecPointUb16 -[UB_LPC_ORDER * UB16_LPC_VEC_PER_FRAME] = -{ - -0.8250, -0.9750, -1.1250, -2.1750, -0.9750, -1.1250, -1.4250, - -2.6250, -1.4250, -1.2750, -1.8750, -3.6750, -1.7250, -1.8750, - -2.3250, -5.4750 -}; - -/* -* Number of reconstruction points of quantizers for LAR coefficients. -*/ -const int16_t WebRtcIsac_kLpcShapeNumRecPointUb16 -[UB_LPC_ORDER * UB16_LPC_VEC_PER_FRAME] = -{ - 13, 15, 17, 29, 15, 16, 20, 35, 20, - 20, 27, 54, 25, 27, 33, 70 -}; - -/* -* Starting index for entropy decoder to search for the right interval, -* one entry per LAR coefficient -*/ -const uint16_t WebRtcIsac_kLpcShapeEntropySearchUb16 -[UB_LPC_ORDER * UB16_LPC_VEC_PER_FRAME] = -{ - 6, 7, 8, 14, 7, 8, 10, 17, 10, - 10, 13, 27, 12, 13, 16, 35 -}; - -/* -* LAR quantization step-size. -*/ -const double WebRtcIsac_kLpcShapeQStepSizeUb16 = 0.150000; diff --git a/modules/audio_coding/codecs/isac/main/source/lpc_shape_swb16_tables.h b/modules/audio_coding/codecs/isac/main/source/lpc_shape_swb16_tables.h deleted file mode 100644 index 51101db936..0000000000 --- a/modules/audio_coding/codecs/isac/main/source/lpc_shape_swb16_tables.h +++ /dev/null @@ -1,81 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/* - * lpc_shape_swb16_tables.h - * - * This file declares tables used for entropy coding of LPC shape of - * upper-band signal if the bandwidth is 16 kHz. - * - */ - -#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_LPC_SHAPE_SWB16_TABLES_H_ -#define MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_LPC_SHAPE_SWB16_TABLES_H_ - -#include - -#include "modules/audio_coding/codecs/isac/main/source/settings.h" - -extern const double WebRtcIsac_kMeanLarUb16[UB_LPC_ORDER]; - -extern const double WebRtcIsac_kIintraVecDecorrMatUb16[UB_LPC_ORDER] - [UB_LPC_ORDER]; - -extern const double WebRtcIsac_kInterVecDecorrMatUb16[UB16_LPC_VEC_PER_FRAME] - [UB16_LPC_VEC_PER_FRAME]; - -extern const uint16_t WebRtcIsac_kLpcShapeCdfVec01Ub16[14]; - -extern const uint16_t WebRtcIsac_kLpcShapeCdfVec1Ub16[16]; - -extern const uint16_t WebRtcIsac_kLpcShapeCdfVec2Ub16[18]; - -extern const uint16_t WebRtcIsac_kLpcShapeCdfVec3Ub16[30]; - -extern const uint16_t WebRtcIsac_kLpcShapeCdfVec4Ub16[16]; - -extern const uint16_t WebRtcIsac_kLpcShapeCdfVec5Ub16[17]; - -extern const uint16_t WebRtcIsac_kLpcShapeCdfVec6Ub16[21]; - -extern const uint16_t WebRtcIsac_kLpcShapeCdfVec7Ub16[36]; - -extern const uint16_t WebRtcIsac_kLpcShapeCdfVec8Ub16[21]; - -extern const uint16_t WebRtcIsac_kLpcShapeCdfVec01Ub160[21]; - -extern const uint16_t WebRtcIsac_kLpcShapeCdfVec01Ub161[28]; - -extern const uint16_t WebRtcIsac_kLpcShapeCdfVec01Ub162[55]; - -extern const uint16_t WebRtcIsac_kLpcShapeCdfVec01Ub163[26]; - -extern const uint16_t WebRtcIsac_kLpcShapeCdfVec01Ub164[28]; - -extern const uint16_t WebRtcIsac_kLpcShapeCdfVec01Ub165[34]; - -extern const uint16_t WebRtcIsac_kLpcShapeCdfVec01Ub166[71]; - -extern const uint16_t* - WebRtcIsac_kLpcShapeCdfMatUb16[UB_LPC_ORDER * UB16_LPC_VEC_PER_FRAME]; - -extern const double - WebRtcIsac_kLpcShapeLeftRecPointUb16[UB_LPC_ORDER * UB16_LPC_VEC_PER_FRAME]; - -extern const int16_t - WebRtcIsac_kLpcShapeNumRecPointUb16[UB_LPC_ORDER * UB16_LPC_VEC_PER_FRAME]; - -extern const uint16_t - WebRtcIsac_kLpcShapeEntropySearchUb16[UB_LPC_ORDER * - UB16_LPC_VEC_PER_FRAME]; - -extern const double WebRtcIsac_kLpcShapeQStepSizeUb16; - -#endif // MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_LPC_SHAPE_SWB16_TABLES_H_ diff --git a/modules/audio_coding/codecs/isac/main/source/lpc_tables.c b/modules/audio_coding/codecs/isac/main/source/lpc_tables.c deleted file mode 100644 index 461b92eb8a..0000000000 --- a/modules/audio_coding/codecs/isac/main/source/lpc_tables.c +++ /dev/null @@ -1,601 +0,0 @@ -/* - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/* coding tables for the KLT coefficients */ - -#include "modules/audio_coding/codecs/isac/main/source/lpc_tables.h" -#include "modules/audio_coding/codecs/isac/main/source/settings.h" - -/* cdf array for model indicator */ -const uint16_t WebRtcIsac_kQKltModelCdf[4] = { - 0, 15434, 37548, 65535 }; - -/* pointer to cdf array for model indicator */ -const uint16_t *WebRtcIsac_kQKltModelCdfPtr[1] = { - WebRtcIsac_kQKltModelCdf }; - -/* initial cdf index for decoder of model indicator */ -const uint16_t WebRtcIsac_kQKltModelInitIndex[1] = { 1 }; - -/* offset to go from rounded value to quantization index */ -const short WebRtcIsac_kQKltQuantMinGain[12] = { - 3, 6, 4, 6, 6, 9, 5, 16, 11, 34, 32, 47 }; - - -const short WebRtcIsac_kQKltQuantMinShape[108] = { - 0, 0, 0, 0, 0, 0, 0, 0, 0, 1, - 1, 1, 1, 1, 2, 2, 2, 3, 0, 0, - 0, 0, 1, 0, 0, 0, 0, 1, 1, 1, - 1, 1, 1, 2, 2, 3, 0, 0, 0, 0, - 1, 0, 1, 1, 1, 1, 1, 1, 1, 2, - 2, 4, 3, 5, 0, 0, 0, 0, 1, 1, - 1, 1, 1, 1, 2, 1, 2, 2, 3, 4, - 4, 7, 0, 0, 1, 1, 1, 1, 1, 1, - 1, 2, 3, 2, 3, 4, 4, 5, 7, 13, - 0, 1, 1, 2, 3, 2, 2, 2, 4, 4, - 5, 6, 7, 11, 9, 13, 12, 26 }; - -/* maximum quantization index */ -const uint16_t WebRtcIsac_kQKltMaxIndGain[12] = { - 6, 12, 8, 14, 10, 19, 12, 31, 22, 56, 52, 138 }; - -const uint16_t WebRtcIsac_kQKltMaxIndShape[108] = { - 0, 0, 0, 0, 0, 0, 0, 0, 0, 1, - 2, 2, 2, 2, 4, 4, 5, 6, 0, 0, - 0, 0, 1, 0, 0, 0, 0, 1, 2, 2, - 2, 2, 3, 4, 5, 7, 0, 0, 0, 0, - 2, 0, 2, 2, 2, 2, 3, 2, 2, 4, - 4, 6, 6, 9, 0, 0, 0, 0, 2, 2, - 2, 2, 2, 2, 3, 2, 4, 4, 7, 7, - 9, 13, 0, 0, 2, 2, 2, 2, 2, 2, - 3, 4, 5, 4, 6, 8, 8, 10, 16, 25, - 0, 2, 2, 4, 5, 4, 4, 4, 7, 8, - 9, 10, 13, 19, 17, 23, 25, 49 }; - -/* index offset */ -const uint16_t WebRtcIsac_kQKltOffsetGain[12] = { - 0, 7, 20, 29, 44, 55, 75, 88, 120, 143, 200, 253 }; - -const uint16_t WebRtcIsac_kQKltOffsetShape[108] = { - 0, 1, 2, 3, 4, 5, 6, 7, 8, 9, - 11, 14, 17, 20, 23, 28, 33, 39, 46, 47, - 48, 49, 50, 52, 53, 54, 55, 56, 58, 61, - 64, 67, 70, 74, 79, 85, 93, 94, 95, 96, - 97, 100, 101, 104, 107, 110, 113, 117, 120, 123, - 128, 133, 140, 147, 157, 158, 159, 160, 161, 164, - 167, 170, 173, 176, 179, 183, 186, 191, 196, 204, - 212, 222, 236, 237, 238, 241, 244, 247, 250, 253, - 256, 260, 265, 271, 276, 283, 292, 301, 312, 329, - 355, 356, 359, 362, 367, 373, 378, 383, 388, 396, - 405, 415, 426, 440, 460, 478, 502, 528 }; - -/* initial cdf index for KLT coefficients */ -const uint16_t WebRtcIsac_kQKltInitIndexGain[12] = { - 3, 6, 4, 7, 5, 10, 6, 16, 11, 28, 26, 69}; - -const uint16_t WebRtcIsac_kQKltInitIndexShape[108] = { - 0, 0, 0, 0, 0, 0, 0, 0, 0, 1, - 1, 1, 1, 1, 2, 2, 3, 3, 0, 0, - 0, 0, 1, 0, 0, 0, 0, 1, 1, 1, - 1, 1, 2, 2, 3, 4, 0, 0, 0, 0, - 1, 0, 1, 1, 1, 1, 2, 1, 1, 2, - 2, 3, 3, 5, 0, 0, 0, 0, 1, 1, - 1, 1, 1, 1, 2, 1, 2, 2, 4, 4, - 5, 7, 0, 0, 1, 1, 1, 1, 1, 1, - 2, 2, 3, 2, 3, 4, 4, 5, 8, 13, - 0, 1, 1, 2, 3, 2, 2, 2, 4, 4, - 5, 5, 7, 10, 9, 12, 13, 25 }; - - -/* quantizer representation levels */ -const double WebRtcIsac_kQKltLevelsGain[392] = { - -2.78127126, -1.76745590, -0.77913790, -0.00437329, 0.79961206, - 1.81775776, 2.81389782, -5.78753143, -4.88384084, -3.89320940, - -2.88133610, -1.92859977, -0.86347396, 0.02003888, 0.86140400, - 1.89667156, 2.97134967, 3.98781964, 4.91727277, 5.82865898, - -4.11195874, -2.80898424, -1.87547977, -0.80943825, -0.00679084, - 0.79573851, 1.83953397, 2.67586037, 3.76274082, -6.10933968, - -4.93034581, -3.89281296, -2.91530625, -1.89684163, -0.85319130, - -0.02275767, 0.86862017, 1.91578276, 2.96107339, 3.96543056, - 4.91369908, 5.91058154, 6.83848343, 8.07136925, -5.87470395, - -4.84703049, -3.84284597, -2.86168446, -1.89290192, -0.82798145, - -0.00080013, 0.82594974, 1.85754329, 2.88351798, 3.96172628, - -8.85684885, -7.87387461, -6.97811862, -5.93256270, -4.94301439, - -3.95513701, -2.96041544, -1.94031192, -0.87961478, -0.00456201, - 0.89911505, 1.91723376, 2.94011511, 3.93302540, 4.97990967, - 5.93133404, 7.02181199, 7.92407762, 8.80155440, 10.04665814, - -4.82396678, -3.85612158, -2.89482244, -1.89558408, -0.90036978, - -0.00677823, 0.90607989, 1.90937981, 2.91175777, 3.91637730, - 4.97565723, 5.84771228, 7.11145863, -16.07879840, -15.03776309, - -13.93905670, -12.95671800, -11.89171202, -10.95820934, -9.95923714, - -8.94357334, -7.99068299, -6.97481009, -5.94826231, -4.96673988, - -3.97490466, -2.97846970, -1.95130435, -0.94215262, -0.01444043, - 0.96770704, 1.95848598, 2.94107862, 3.95666119, 4.97253085, - 5.97191122, 6.93277360, 7.96608727, 8.87958779, 10.00264269, - 10.86560820, 12.07449071, 13.04491775, 13.97507061, 14.91845261, - -10.85696295, -9.83365357, -9.01245635, -7.95915145, -6.95625003, - -5.95362618, -4.93468444, -3.98760978, -2.95044407, -1.97041277, - -0.97701799, -0.00840234, 0.97834289, 1.98361415, 2.97802439, - 3.96415871, 4.95369042, 5.94101770, 6.92756798, 7.94063998, - 8.85951828, 9.97077022, 11.00068503, -33.92030406, -32.81426422, - -32.00000000, -31.13243639, -30.11886909, -29.06017570, -28.12598824, - -27.22045482, -25.81215858, -25.07849962, -23.93018013, -23.02097643, - -21.89529725, -20.99091085, -19.98889048, -18.94327044, -17.96562071, - -16.96126218, -15.95054062, -14.98516200, -13.97101012, -13.02106500, - -11.98438006, -11.03216748, -9.95930286, -8.97043946, -7.98085082, - -6.98360995, -5.98998802, -4.98668173, -4.00032906, -3.00420619, - -1.98701132, -0.99324682, -0.00609324, 0.98297834, 1.99483076, - 3.00305044, 3.97142097, 4.97525759, 5.98612258, 6.97448236, - 7.97575900, 9.01086211, 9.98665542, 11.00541438, 11.98078628, - 12.92352471, 14.06849675, 14.99949430, 15.94904834, 16.97440321, - 18.04040916, 18.88987609, 20.05312391, 21.00000000, 21.79443341, - -31.98578825, -31.00000000, -29.89060567, -28.98555686, -27.97114102, - -26.84935410, -26.02402230, -24.94195278, -23.92336849, -22.95552382, - -21.97932836, -20.96055470, -19.99649553, -19.03436122, -17.96706525, - -17.01139515, -16.01363516, -14.99154248, -14.00298333, -12.99630613, - -11.99955519, -10.99000421, -10.00819092, -8.99763648, -7.98431793, - -7.01769025, -5.99604690, -4.99980697, -3.99334671, -3.01748192, - -2.02051217, -1.00848371, -0.01942358, 1.00477757, 1.95477872, - 2.98593031, 3.98779079, 4.96862849, 6.02694771, 6.93983733, - 7.89874717, 8.99615862, 10.02367921, 10.96293452, 11.84351528, - 12.92207187, 13.85122329, 15.05146877, 15.99371264, 17.00000000, - 18.00000000, 19.00000000, 19.82763573, -47.00000000, -46.00000000, - -44.87138498, -44.00000000, -43.00000000, -42.00000000, -41.00000000, - -39.88966612, -38.98913239, -37.80306486, -37.23584325, -35.94200288, - -34.99881301, -34.11361858, -33.06507360, -32.13129135, -30.90891364, - -29.81511907, -28.99250380, -28.04535391, -26.99767800, -26.04418164, - -24.95687851, -24.04865595, -23.03392645, -21.89366707, -20.93517364, - -19.99388660, -18.91620943, -18.03749683, -16.99532379, -15.98683813, - -15.06421479, -13.99359211, -12.99714098, -11.97022520, -10.98500279, - -9.98834422, -8.95729330, -8.01232284, -7.00253661, -5.99681626, - -5.01207817, -3.95914904, -3.01232178, -1.96615919, -0.97687670, - 0.01228030, 0.98412288, 2.01753544, 3.00580570, 3.97783510, - 4.98846894, 6.01321400, 7.00867732, 8.00416375, 9.01771966, - 9.98637729, 10.98255180, 11.99194163, 13.01807333, 14.00999545, - 15.00118556, 16.00089224, 17.00584148, 17.98251763, 18.99942091, - 19.96917690, 20.97839265, 21.98207297, 23.00171271, 23.99930737, - 24.99746061, 26.00936304, 26.98240132, 28.01126868, 29.01395915, - 29.98153507, 31.01376711, 31.99876818, 33.00475317, 33.99753994, - 34.99493913, 35.98933585, 36.95620160, 37.98428461, 38.99317544, - 40.01832073, 40.98048133, 41.95999283, 42.98232091, 43.96523612, - 44.99574268, 45.99524194, 47.05464025, 48.03821548, 48.99354366, - 49.96400411, 50.98017973, 51.95184408, 52.96291806, 54.00194392, - 54.96603783, 55.95623778, 57.03076595, 58.05889901, 58.99081551, - 59.97928121, 61.05071612, 62.03971580, 63.01286038, 64.01290338, - 65.02074503, 65.99454594, 67.00399425, 67.96571257, 68.95305727, - 69.92030664, 70.95594862, 71.98088567, 73.04764124, 74.00285480, - 75.02696330, 75.89837673, 76.93459997, 78.16266309, 78.83317543, - 80.00000000, 80.87251574, 82.09803524, 83.10671664, 84.00000000, - 84.77023523, 86.00000000, 87.00000000, 87.92946897, 88.69159118, - 90.00000000, 90.90535270 }; - -const double WebRtcIsac_kQKltLevelsShape[578] = { - 0.00032397, 0.00008053, -0.00061202, -0.00012620, 0.00030437, - 0.00054764, -0.00027902, 0.00069360, 0.00029449, -0.80219239, - 0.00091089, -0.74514927, -0.00094283, 0.64030631, -0.60509119, - 0.00035575, 0.61851665, -0.62129957, 0.00375219, 0.60054900, - -0.61554359, 0.00054977, 0.63362016, -1.73118727, -0.65422341, - 0.00524568, 0.66165298, 1.76785515, -1.83182018, -0.65997434, - -0.00011887, 0.67524299, 1.79933938, -1.76344480, -0.72547708, - -0.00133017, 0.73104704, 1.75305377, 2.85164534, -2.80423916, - -1.71959639, -0.75419722, -0.00329945, 0.77196760, 1.72211069, - 2.87339653, 0.00031089, -0.00015311, 0.00018201, -0.00035035, - -0.77357251, 0.00154647, -0.00047625, -0.00045299, 0.00086590, - 0.00044762, -0.83383829, 0.00024787, -0.68526258, -0.00122472, - 0.64643255, -0.60904942, -0.00448987, 0.62309184, -0.59626442, - -0.00574132, 0.62296546, -0.63222115, 0.00013441, 0.63609545, - -0.66911055, -0.00369971, 0.66346095, 2.07281301, -1.77184694, - -0.67640425, -0.00010145, 0.64818392, 1.74948973, -1.69420224, - -0.71943894, -0.00004680, 0.75303493, 1.81075983, 2.80610041, - -2.80005755, -1.79866753, -0.77409777, -0.00084220, 0.80141293, - 1.78291081, 2.73954236, 3.82994169, 0.00015140, -0.00012766, - -0.00034241, -0.00119125, -0.76113497, 0.00069246, 0.76722027, - 0.00132862, -0.69107530, 0.00010656, 0.77061578, -0.78012970, - 0.00095947, 0.77828502, -0.64787758, 0.00217168, 0.63050167, - -0.58601125, 0.00306596, 0.59466308, -0.58603410, 0.00059779, - 0.64257970, 1.76512766, -0.61193600, -0.00259517, 0.59767574, - -0.61026273, 0.00315811, 0.61725479, -1.69169719, -0.65816029, - 0.00067575, 0.65576890, 2.00000000, -1.72689193, -0.69780808, - -0.00040990, 0.70668487, 1.74198458, -3.79028154, -3.00000000, - -1.73194459, -0.70179341, -0.00106695, 0.71302629, 1.76849782, - -2.89332364, -1.78585007, -0.78731491, -0.00132610, 0.79692976, - 1.75247009, 2.97828682, -5.26238694, -3.69559829, -2.87286122, - -1.84908818, -0.84434577, -0.01167975, 0.84641753, 1.84087672, - 2.87628156, 3.83556679, -0.00190204, 0.00092642, 0.00354385, - -0.00012982, -0.67742785, 0.00229509, 0.64935672, -0.58444751, - 0.00470733, 0.57299534, -0.58456202, -0.00097715, 0.64593607, - -0.64060330, -0.00638534, 0.59680157, -0.59287537, 0.00490772, - 0.58919707, -0.60306173, -0.00417464, 0.60562100, -1.75218757, - -0.63018569, -0.00225922, 0.63863300, -0.63949939, -0.00126421, - 0.64268914, -1.75851182, -0.68318060, 0.00510418, 0.69049211, - 1.88178506, -1.71136148, -0.72710534, -0.00815559, 0.73412917, - 1.79996711, -2.77111145, -1.73940498, -0.78212945, 0.01074476, - 0.77688916, 1.76873972, 2.87281379, 3.77554698, -3.75832725, - -2.95463235, -1.80451491, -0.80017226, 0.00149902, 0.80729206, - 1.78265046, 2.89391793, -3.78236148, -2.83640598, -1.82532067, - -0.88844327, -0.00620952, 0.88208030, 1.85757631, 2.81712391, - 3.88430176, 5.16179367, -7.00000000, -5.93805408, -4.87172597, - -3.87524433, -2.89399744, -1.92359563, -0.92136341, -0.00172725, - 0.93087018, 1.90528280, 2.89809686, 3.88085708, 4.89147740, - 5.89078692, -0.00239502, 0.00312564, -1.00000000, 0.00178325, - 1.00000000, -0.62198029, 0.00143254, 0.65344051, -0.59851220, - -0.00676987, 0.61510140, -0.58894151, 0.00385055, 0.59794203, - -0.59808568, -0.00038214, 0.57625703, -0.63009713, -0.01107985, - 0.61278758, -0.64206758, -0.00154369, 0.65480598, 1.80604162, - -1.80909286, -0.67810514, 0.00205762, 0.68571097, 1.79453891, - -3.22682422, -1.73808453, -0.71870305, -0.00738594, 0.71486172, - 1.73005326, -1.66891897, -0.73689615, -0.00616203, 0.74262409, - 1.73807899, -2.92417482, -1.73866741, -0.78133871, 0.00764425, - 0.80027264, 1.78668732, 2.74992588, -4.00000000, -2.75578740, - -1.83697516, -0.83117035, -0.00355191, 0.83527172, 1.82814700, - 2.77377675, 3.80718693, -3.81667698, -2.83575471, -1.83372350, - -0.86579471, 0.00547578, 0.87582281, 1.82858793, 2.87265007, - 3.91405377, -4.87521600, -3.78999094, -2.86437014, -1.86964365, - -0.90618018, 0.00128243, 0.91497811, 1.87374952, 2.83199819, - 3.91519130, 4.76632822, -6.68713448, -6.01252467, -4.94587936, - -3.88795368, -2.91299088, -1.92592211, -0.95504570, -0.00089980, - 0.94565200, 1.93239633, 2.91832808, 3.91363475, 4.88920034, - 5.96471415, 6.83905252, 7.86195009, 8.81571018,-12.96141759, - -11.73039516,-10.96459719, -9.97382433, -9.04414433, -7.89460619, - -6.96628608, -5.93236595, -4.93337924, -3.95479990, -2.96451499, - -1.96635876, -0.97271229, -0.00402238, 0.98343930, 1.98348291, - 2.96641164, 3.95456471, 4.95517089, 5.98975714, 6.90322073, - 7.90468849, 8.85639467, 9.97255498, 10.79006309, 11.81988596, - 0.04950500, -1.00000000, -0.01226628, 1.00000000, -0.59479469, - -0.10438305, 0.59822144, -2.00000000, -0.67109149, -0.09256692, - 0.65171621, 2.00000000, -3.00000000, -1.68391999, -0.76681039, - -0.03354151, 0.71509146, 1.77615472, -2.00000000, -0.68661511, - -0.02497881, 0.66478398, 2.00000000, -2.00000000, -0.67032784, - -0.00920582, 0.64892756, 2.00000000, -2.00000000, -0.68561894, - 0.03641869, 0.73021611, 1.68293863, -4.00000000, -2.72024184, - -1.80096059, -0.81696185, 0.03604685, 0.79232033, 1.70070730, - 3.00000000, -4.00000000, -2.71795670, -1.80482986, -0.86001162, - 0.03764903, 0.87723968, 1.79970771, 2.72685932, 3.67589143, - -5.00000000, -4.00000000, -2.85492548, -1.78996365, -0.83250358, - -0.01376828, 0.84195506, 1.78161105, 2.76754458, 4.00000000, - -6.00000000, -5.00000000, -3.82268811, -2.77563624, -1.82608163, - -0.86486114, -0.02671886, 0.86693165, 1.88422879, 2.86248347, - 3.95632216, -7.00000000, -6.00000000, -5.00000000, -3.77533988, - -2.86391432, -1.87052039, -0.90513658, 0.06271236, 0.91083620, - 1.85734756, 2.86031688, 3.82019418, 4.94420394, 6.00000000, - -11.00000000,-10.00000000, -9.00000000, -8.00000000, -6.91952415, - -6.00000000, -4.92044374, -3.87845165, -2.87392362, -1.88413020, - -0.91915740, 0.00318517, 0.91602800, 1.89664838, 2.88925058, - 3.84123856, 4.78988651, 5.94526812, 6.81953917, 8.00000000, - -9.00000000, -8.00000000, -7.03319143, -5.94530963, -4.86669720, - -3.92438007, -2.88620396, -1.92848070, -0.94365985, 0.01671855, - 0.97349410, 1.93419878, 2.89740109, 3.89662823, 4.83235583, - 5.88106535, 6.80328232, 8.00000000,-13.00000000,-12.00000000, - -11.00000000,-10.00000000, -9.00000000, -7.86033489, -6.83344055, - -5.89844215, -4.90811454, -3.94841298, -2.95820490, -1.98627966, - -0.99161468, -0.02286136, 0.96055651, 1.95052433, 2.93969396, - 3.94304346, 4.88522624, 5.87434241, 6.78309433, 7.87244101, - 9.00000000, 10.00000000,-12.09117356,-11.00000000,-10.00000000, - -8.84766108, -7.86934236, -6.98544896, -5.94233429, -4.95583292, - -3.95575986, -2.97085529, -1.98955811, -0.99359873, -0.00485413, - 0.98298870, 1.98093258, 2.96430203, 3.95540216, 4.96915010, - 5.96775124, 6.99236918, 7.96503302, 8.99864542, 9.85857723, - 10.96541926, 11.91647197, 12.71060069,-26.00000000,-25.00000000, - -24.00585596,-23.11642573,-22.14271284,-20.89800711,-19.87815799, - -19.05036354,-17.88555651,-16.86471209,-15.97711073,-14.94012359, - -14.02661226,-12.98243228,-11.97489256,-10.97402777, -9.96425624, - -9.01085220, -7.97372506, -6.98795002, -5.97271328, -5.00191694, - -3.98055849, -2.98458048, -1.99470442, -0.99656768, -0.00825666, - 1.00272004, 1.99922218, 2.99357669, 4.01407905, 5.01003897, - 5.98115528, 7.00018958, 8.00338125, 8.98981046, 9.98990318, - 10.96341479, 11.96866930, 12.99175139, 13.94580443, 14.95745083, - 15.98992869, 16.97484646, 17.99630043, 18.93396897, 19.88347741, - 20.96532482, 21.92191032, 23.22314702 }; - - -/* cdf tables for quantizer indices */ -const uint16_t WebRtcIsac_kQKltCdfGain[404] = { - 0, 13, 301, 3730, 61784, 65167, 65489, 65535, 0, 17, - 142, 314, 929, 2466, 7678, 56450, 63463, 64740, 65204, 65426, - 65527, 65535, 0, 8, 100, 724, 6301, 60105, 65125, 65510, - 65531, 65535, 0, 13, 117, 368, 1068, 3010, 11928, 53603, - 61177, 63404, 64505, 65108, 65422, 65502, 65531, 65535, 0, 4, - 17, 96, 410, 1859, 12125, 54361, 64103, 65305, 65497, 65535, - 0, 4, 88, 230, 469, 950, 1746, 3228, 6092, 16592, - 44756, 56848, 61256, 63308, 64325, 64920, 65309, 65460, 65502, - 65522, 65535, 0, 88, 352, 1675, 6339, 20749, 46686, 59284, 63525, - 64949, 65359, 65502, 65527, 65535, 0, 13, 38, 63, 117, - 234, 381, 641, 929, 1407, 2043, 2809, 4032, 5753, 8792, - 14407, 24308, 38941, 48947, 55403, 59293, 61411, 62688, 63630, - 64329, 64840, 65188, 65376, 65472, 65506, 65527, 65531, 65535, - 0, 8, 29, 75, 222, 615, 1327, 2801, 5623, 9931, 16094, 24966, - 34419, 43458, 50676, 56186, 60055, 62500, 63936, 64765, 65225, - 65435, 65514, 65535, 0, 8, 13, 15, 17, 21, 33, 59, - 71, 92, 151, 243, 360, 456, 674, 934, 1223, 1583, - 1989, 2504, 3031, 3617, 4354, 5154, 6163, 7411, 8780, 10747, - 12874, 15591, 18974, 23027, 27436, 32020, 36948, 41830, 46205, - 49797, 53042, 56094, 58418, 60360, 61763, 62818, 63559, 64103, - 64509, 64798, 65045, 65162, 65288, 65363, 65447, 65506, 65522, - 65531, 65533, 65535, 0, 4, 6, 25, 38, 71, 138, 264, 519, 808, - 1227, 1825, 2516, 3408, 4279, 5560, 7092, 9197, 11420, 14108, - 16947, 20300, 23926, 27459, 31164, 34827, 38575, 42178, 45540, - 48747, 51444, 54090, 56426, 58460, 60080, 61595, 62734, 63668, - 64275, 64673, 64936, 65112, 65217, 65334, 65426, 65464, 65477, - 65489, 65518, 65527, 65529, 65531, 65533, 65535, 0, 2, 4, 8, 10, - 12, 14, 16, 21, 33, 50, 71, 84, 92, 105, 138, 180, 255, 318, - 377, 435, 473, 511, 590, 682, 758, 913, 1097, 1256, 1449, 1671, - 1884, 2169, 2445, 2772, 3157, 3563, 3944, 4375, 4848, 5334, 5820, - 6448, 7101, 7716, 8378, 9102, 9956, 10752, 11648, 12707, 13670, - 14758, 15910, 17187, 18472, 19627, 20649, 21951, 23169, 24283, - 25552, 26862, 28227, 29391, 30764, 31882, 33213, 34432, 35600, - 36910, 38116, 39464, 40729, 41872, 43144, 44371, 45514, 46762, - 47813, 48968, 50069, 51032, 51974, 52908, 53737, 54603, 55445, - 56282, 56990, 57572, 58191, 58840, 59410, 59887, 60264, 60607, - 60946, 61269, 61516, 61771, 61960, 62198, 62408, 62558, 62776, - 62985, 63207, 63408, 63546, 63739, 63906, 64070, 64237, 64371, - 64551, 64677, 64836, 64999, 65095, 65213, 65284, 65338, 65380, - 65426, 65447, 65472, 65485, 65487, 65489, 65502, 65510, 65512, - 65514, 65516, 65518, 65522, 65531, 65533, 65535 }; - - -const uint16_t WebRtcIsac_kQKltCdfShape[686] = { - 0, 65535, 0, 65535, 0, 65535, 0, 65535, 0, 65535, - 0, 65535, 0, 65535, 0, 65535, 0, 65535, 0, 4, - 65535, 0, 8, 65514, 65535, 0, 29, 65481, 65535, 0, - 121, 65439, 65535, 0, 239, 65284, 65535, 0, 8, 779, - 64999, 65527, 65535, 0, 8, 888, 64693, 65522, 65535, 0, - 29, 2604, 62843, 65497, 65531, 65535, 0, 25, 176, 4576, - 61164, 65275, 65527, 65535, 0, 65535, 0, 65535, 0, 65535, - 0, 65535, 0, 4, 65535, 0, 65535, 0, 65535, 0, - 65535, 0, 65535, 0, 4, 65535, 0, 33, 65502, 65535, - 0, 54, 65481, 65535, 0, 251, 65309, 65535, 0, 611, - 65074, 65535, 0, 1273, 64292, 65527, 65535, 0, 4, 1809, - 63940, 65518, 65535, 0, 88, 4392, 60603, 65426, 65531, 65535, - 0, 25, 419, 7046, 57756, 64961, 65514, 65531, 65535, 0, - 65535, 0, 65535, 0, 65535, 0, 65535, 0, 4, 65531, - 65535, 0, 65535, 0, 8, 65531, 65535, 0, 4, 65527, - 65535, 0, 17, 65510, 65535, 0, 42, 65481, 65535, 0, - 197, 65342, 65531, 65535, 0, 385, 65154, 65535, 0, 1005, - 64522, 65535, 0, 8, 1985, 63469, 65533, 65535, 0, 38, - 3119, 61884, 65514, 65535, 0, 4, 6, 67, 4961, 60804, - 65472, 65535, 0, 17, 565, 9182, 56538, 65087, 65514, 65535, - 0, 8, 63, 327, 2118, 14490, 52774, 63839, 65376, 65522, - 65535, 0, 65535, 0, 65535, 0, 65535, 0, 65535, 0, - 17, 65522, 65535, 0, 59, 65489, 65535, 0, 50, 65522, - 65535, 0, 54, 65489, 65535, 0, 310, 65179, 65535, 0, - 615, 64836, 65535, 0, 4, 1503, 63965, 65535, 0, 2780, - 63383, 65535, 0, 21, 3919, 61051, 65527, 65535, 0, 84, - 6674, 59929, 65435, 65535, 0, 4, 255, 7976, 55784, 65150, - 65518, 65531, 65535, 0, 4, 8, 582, 10726, 53465, 64949, - 65518, 65535, 0, 29, 339, 3006, 17555, 49517, 62956, 65200, - 65497, 65531, 65535, 0, 2, 33, 138, 565, 2324, 7670, - 22089, 45966, 58949, 63479, 64966, 65380, 65518, 65535, 0, 65535, - 0, 65535, 0, 2, 65533, 65535, 0, 46, 65514, 65535, - 0, 414, 65091, 65535, 0, 540, 64911, 65535, 0, 419, - 65162, 65535, 0, 976, 64790, 65535, 0, 2977, 62495, 65531, - 65535, 0, 4, 3852, 61034, 65527, 65535, 0, 4, 29, - 6021, 60243, 65468, 65535, 0, 84, 6711, 58066, 65418, 65535, - 0, 13, 281, 9550, 54917, 65125, 65506, 65535, 0, 2, - 63, 984, 12108, 52644, 64342, 65435, 65527, 65535, 0, 29, - 251, 2014, 14871, 47553, 62881, 65229, 65518, 65535, 0, 13, - 142, 749, 4220, 18497, 45200, 60913, 64823, 65426, 65527, 65535, - 0, 13, 71, 264, 1176, 3789, 10500, 24480, 43488, 56324, - 62315, 64493, 65242, 65464, 65514, 65522, 65531, 65535, 0, 4, - 13, 38, 109, 205, 448, 850, 1708, 3429, 6276, 11371, - 19221, 29734, 40955, 49391, 55411, 59460, 62102, 63793, 64656, - 65150, 65401, 65485, 65522, 65531, 65535, 0, 65535, 0, 2, 65533, - 65535, 0, 1160, 65476, 65535, 0, 2, 6640, 64763, 65533, - 65535, 0, 2, 38, 9923, 61009, 65527, 65535, 0, 2, - 4949, 63092, 65533, 65535, 0, 2, 3090, 63398, 65533, 65535, - 0, 2, 2520, 58744, 65510, 65535, 0, 2, 13, 544, - 8784, 51403, 65148, 65533, 65535, 0, 2, 25, 1017, 10412, - 43550, 63651, 65489, 65527, 65535, 0, 2, 4, 29, 783, - 13377, 52462, 64524, 65495, 65533, 65535, 0, 2, 4, 6, - 100, 1817, 18451, 52590, 63559, 65376, 65531, 65535, 0, 2, - 4, 6, 46, 385, 2562, 11225, 37416, 60488, 65026, 65487, - 65529, 65533, 65535, 0, 2, 4, 6, 8, 10, 12, - 42, 222, 971, 5221, 19811, 45048, 60312, 64486, 65294, 65474, - 65525, 65529, 65533, 65535, 0, 2, 4, 8, 71, 167, - 666, 2533, 7875, 19622, 38082, 54359, 62108, 64633, 65290, 65495, - 65529, 65533, 65535, 0, 2, 4, 6, 8, 10, 13, - 109, 586, 1930, 4949, 11600, 22641, 36125, 48312, 56899, 61495, - 63927, 64932, 65389, 65489, 65518, 65531, 65533, 65535, 0, 4, - 6, 8, 67, 209, 712, 1838, 4195, 8432, 14432, 22834, - 31723, 40523, 48139, 53929, 57865, 60657, 62403, 63584, 64363, - 64907, 65167, 65372, 65472, 65514, 65535, 0, 2, 4, 13, 25, - 42, 46, 50, 75, 113, 147, 281, 448, 657, 909, - 1185, 1591, 1976, 2600, 3676, 5317, 7398, 9914, 12941, 16169, - 19477, 22885, 26464, 29851, 33360, 37228, 41139, 44802, 48654, - 52058, 55181, 57676, 59581, 61022, 62190, 63107, 63676, 64199, - 64547, 64924, 65158, 65313, 65430, 65481, 65518, 65535 }; - - -/* pointers to cdf tables for quantizer indices */ -const uint16_t *WebRtcIsac_kQKltCdfPtrGain[12] = { - WebRtcIsac_kQKltCdfGain +0 +0, WebRtcIsac_kQKltCdfGain +0 +8, - WebRtcIsac_kQKltCdfGain +0 +22, WebRtcIsac_kQKltCdfGain +0 +32, - WebRtcIsac_kQKltCdfGain +0 +48, WebRtcIsac_kQKltCdfGain +0 +60, - WebRtcIsac_kQKltCdfGain +0 +81, WebRtcIsac_kQKltCdfGain +0 +95, - WebRtcIsac_kQKltCdfGain +0 +128, WebRtcIsac_kQKltCdfGain +0 +152, - WebRtcIsac_kQKltCdfGain +0 +210, WebRtcIsac_kQKltCdfGain +0 +264 }; - -const uint16_t *WebRtcIsac_kQKltCdfPtrShape[108] = { - WebRtcIsac_kQKltCdfShape +0 +0, WebRtcIsac_kQKltCdfShape +0 +2, - WebRtcIsac_kQKltCdfShape +0 +4, WebRtcIsac_kQKltCdfShape +0 +6, - WebRtcIsac_kQKltCdfShape +0 +8, WebRtcIsac_kQKltCdfShape +0 +10, - WebRtcIsac_kQKltCdfShape +0 +12, WebRtcIsac_kQKltCdfShape +0 +14, - WebRtcIsac_kQKltCdfShape +0 +16, WebRtcIsac_kQKltCdfShape +0 +18, - WebRtcIsac_kQKltCdfShape +0 +21, WebRtcIsac_kQKltCdfShape +0 +25, - WebRtcIsac_kQKltCdfShape +0 +29, WebRtcIsac_kQKltCdfShape +0 +33, - WebRtcIsac_kQKltCdfShape +0 +37, WebRtcIsac_kQKltCdfShape +0 +43, - WebRtcIsac_kQKltCdfShape +0 +49, WebRtcIsac_kQKltCdfShape +0 +56, - WebRtcIsac_kQKltCdfShape +0 +64, WebRtcIsac_kQKltCdfShape +0 +66, - WebRtcIsac_kQKltCdfShape +0 +68, WebRtcIsac_kQKltCdfShape +0 +70, - WebRtcIsac_kQKltCdfShape +0 +72, WebRtcIsac_kQKltCdfShape +0 +75, - WebRtcIsac_kQKltCdfShape +0 +77, WebRtcIsac_kQKltCdfShape +0 +79, - WebRtcIsac_kQKltCdfShape +0 +81, WebRtcIsac_kQKltCdfShape +0 +83, - WebRtcIsac_kQKltCdfShape +0 +86, WebRtcIsac_kQKltCdfShape +0 +90, - WebRtcIsac_kQKltCdfShape +0 +94, WebRtcIsac_kQKltCdfShape +0 +98, - WebRtcIsac_kQKltCdfShape +0 +102, WebRtcIsac_kQKltCdfShape +0 +107, - WebRtcIsac_kQKltCdfShape +0 +113, WebRtcIsac_kQKltCdfShape +0 +120, - WebRtcIsac_kQKltCdfShape +0 +129, WebRtcIsac_kQKltCdfShape +0 +131, - WebRtcIsac_kQKltCdfShape +0 +133, WebRtcIsac_kQKltCdfShape +0 +135, - WebRtcIsac_kQKltCdfShape +0 +137, WebRtcIsac_kQKltCdfShape +0 +141, - WebRtcIsac_kQKltCdfShape +0 +143, WebRtcIsac_kQKltCdfShape +0 +147, - WebRtcIsac_kQKltCdfShape +0 +151, WebRtcIsac_kQKltCdfShape +0 +155, - WebRtcIsac_kQKltCdfShape +0 +159, WebRtcIsac_kQKltCdfShape +0 +164, - WebRtcIsac_kQKltCdfShape +0 +168, WebRtcIsac_kQKltCdfShape +0 +172, - WebRtcIsac_kQKltCdfShape +0 +178, WebRtcIsac_kQKltCdfShape +0 +184, - WebRtcIsac_kQKltCdfShape +0 +192, WebRtcIsac_kQKltCdfShape +0 +200, - WebRtcIsac_kQKltCdfShape +0 +211, WebRtcIsac_kQKltCdfShape +0 +213, - WebRtcIsac_kQKltCdfShape +0 +215, WebRtcIsac_kQKltCdfShape +0 +217, - WebRtcIsac_kQKltCdfShape +0 +219, WebRtcIsac_kQKltCdfShape +0 +223, - WebRtcIsac_kQKltCdfShape +0 +227, WebRtcIsac_kQKltCdfShape +0 +231, - WebRtcIsac_kQKltCdfShape +0 +235, WebRtcIsac_kQKltCdfShape +0 +239, - WebRtcIsac_kQKltCdfShape +0 +243, WebRtcIsac_kQKltCdfShape +0 +248, - WebRtcIsac_kQKltCdfShape +0 +252, WebRtcIsac_kQKltCdfShape +0 +258, - WebRtcIsac_kQKltCdfShape +0 +264, WebRtcIsac_kQKltCdfShape +0 +273, - WebRtcIsac_kQKltCdfShape +0 +282, WebRtcIsac_kQKltCdfShape +0 +293, - WebRtcIsac_kQKltCdfShape +0 +308, WebRtcIsac_kQKltCdfShape +0 +310, - WebRtcIsac_kQKltCdfShape +0 +312, WebRtcIsac_kQKltCdfShape +0 +316, - WebRtcIsac_kQKltCdfShape +0 +320, WebRtcIsac_kQKltCdfShape +0 +324, - WebRtcIsac_kQKltCdfShape +0 +328, WebRtcIsac_kQKltCdfShape +0 +332, - WebRtcIsac_kQKltCdfShape +0 +336, WebRtcIsac_kQKltCdfShape +0 +341, - WebRtcIsac_kQKltCdfShape +0 +347, WebRtcIsac_kQKltCdfShape +0 +354, - WebRtcIsac_kQKltCdfShape +0 +360, WebRtcIsac_kQKltCdfShape +0 +368, - WebRtcIsac_kQKltCdfShape +0 +378, WebRtcIsac_kQKltCdfShape +0 +388, - WebRtcIsac_kQKltCdfShape +0 +400, WebRtcIsac_kQKltCdfShape +0 +418, - WebRtcIsac_kQKltCdfShape +0 +445, WebRtcIsac_kQKltCdfShape +0 +447, - WebRtcIsac_kQKltCdfShape +0 +451, WebRtcIsac_kQKltCdfShape +0 +455, - WebRtcIsac_kQKltCdfShape +0 +461, WebRtcIsac_kQKltCdfShape +0 +468, - WebRtcIsac_kQKltCdfShape +0 +474, WebRtcIsac_kQKltCdfShape +0 +480, - WebRtcIsac_kQKltCdfShape +0 +486, WebRtcIsac_kQKltCdfShape +0 +495, - WebRtcIsac_kQKltCdfShape +0 +505, WebRtcIsac_kQKltCdfShape +0 +516, - WebRtcIsac_kQKltCdfShape +0 +528, WebRtcIsac_kQKltCdfShape +0 +543, - WebRtcIsac_kQKltCdfShape +0 +564, WebRtcIsac_kQKltCdfShape +0 +583, - WebRtcIsac_kQKltCdfShape +0 +608, WebRtcIsac_kQKltCdfShape +0 +635 }; - - -/* left KLT transforms */ -const double WebRtcIsac_kKltT1Gain[4] = { - -0.79742827, 0.60341375, 0.60341375, 0.79742827 }; - -const double WebRtcIsac_kKltT1Shape[324] = { - 0.00159597, 0.00049320, 0.00513821, 0.00021066, 0.01338581, - -0.00422367, -0.00272072, 0.00935107, 0.02047622, 0.02691189, - 0.00478236, 0.03969702, 0.00886698, 0.04877604, -0.10898362, - -0.05930891, -0.03415047, 0.98889721, 0.00293558, -0.00035282, - 0.01156321, -0.00195341, -0.00937631, 0.01052213, -0.02551163, - 0.01644059, 0.03189927, 0.07754773, -0.08742313, -0.03026338, - 0.05136248, -0.14395974, 0.17725040, 0.22664856, 0.93380230, - 0.07076411, 0.00557890, -0.00222834, 0.01377569, 0.01466808, - 0.02847361, -0.00603178, 0.02382480, -0.01210452, 0.03797267, - -0.02371480, 0.11260335, -0.07366682, 0.00453436, -0.04136941, - -0.07912843, -0.95031418, 0.25295337, -0.05302216, -0.00617554, - -0.00044040, -0.00653778, 0.01097838, 0.01529174, 0.01374431, - -0.00748512, -0.00020034, 0.02432713, 0.11101570, -0.08556891, - 0.09282249, -0.01029446, 0.67556443, -0.67454300, 0.06910063, - 0.20866865, -0.10318050, 0.00932175, 0.00524058, 0.00803610, - -0.00594676, -0.01082578, 0.01069906, 0.00546768, 0.01565291, - 0.06816200, 0.10201227, 0.16812734, 0.22984074, 0.58213170, - -0.54138651, -0.51379962, 0.06847390, -0.01920037, -0.04592324, - -0.00467394, 0.00328858, 0.00377424, -0.00987448, 0.08222096, - -0.00377301, 0.04551941, -0.02592517, 0.16317082, 0.13077530, - 0.22702921, -0.31215289, -0.69645962, -0.38047101, -0.39339411, - 0.11124777, 0.02508035, -0.00708074, 0.00400344, 0.00040331, - 0.01142402, 0.01725406, 0.01635170, 0.14285366, 0.03949233, - -0.05905676, 0.05877154, -0.17497577, -0.32479440, 0.80754464, - -0.38085603, -0.17055430, -0.03168622, -0.07531451, 0.02942002, - -0.02148095, -0.00754114, -0.00322372, 0.00567812, -0.01701521, - -0.12358320, 0.11473564, 0.09070136, 0.06533068, -0.22560802, - 0.19209022, 0.81605094, 0.36592275, -0.09919829, 0.16667122, - 0.16300725, 0.04803807, 0.06739263, -0.00156752, -0.01685302, - -0.00905240, -0.02297836, -0.00469939, 0.06310613, -0.16391930, - 0.10919511, 0.12529293, 0.85581322, -0.32145522, 0.24539076, - 0.07181839, 0.07289591, 0.14066759, 0.10406711, 0.05815518, - 0.01072680, -0.00759339, 0.00053486, -0.00044865, 0.03407361, - 0.01645348, 0.08758579, 0.27722240, 0.53665485, -0.74853376, - -0.01118192, -0.19805430, 0.06130619, -0.09675299, 0.08978480, - 0.03405255, -0.00706867, 0.05102045, 0.03250746, 0.01849966, - -0.01216314, -0.01184187, -0.01579288, 0.00114807, 0.11376166, - 0.88342114, -0.36425379, 0.13863190, 0.12524180, -0.13553892, - 0.04715856, -0.12341103, 0.04531568, 0.01899360, -0.00206897, - 0.00567768, -0.01444163, 0.00411946, -0.00855896, 0.00381663, - -0.01664861, -0.05534280, 0.21328278, 0.20161162, 0.72360394, - 0.59130708, -0.08043791, 0.08757349, -0.13893918, -0.05147377, - 0.02680690, -0.01144070, 0.00625162, -0.00634215, -0.01248947, - -0.00329455, -0.00609625, -0.00136305, -0.05097048, -0.01029851, - 0.25065384, -0.16856837, -0.07123372, 0.15992623, -0.39487617, - -0.79972301, 0.18118185, -0.04826639, -0.01805578, -0.02927253, - -0.16400618, 0.07472763, 0.10376449, 0.01705406, 0.01065801, - -0.01500498, 0.02039914, 0.37776349, -0.84484186, 0.10434286, - 0.15616990, 0.13474456, -0.00906238, -0.25238368, -0.03820885, - -0.10650905, -0.03880833, -0.03660028, -0.09640894, 0.00583314, - 0.01922097, 0.01489911, -0.02431117, -0.09372217, 0.39404721, - -0.84786223, -0.31277121, 0.03193850, 0.01974060, 0.01887901, - 0.00337911, -0.11359599, -0.02792521, -0.03220184, -0.01533311, - 0.00015962, -0.04225043, -0.00933965, 0.00675311, 0.00206060, - 0.15926771, 0.40199829, -0.80792558, -0.35591604, -0.17169764, - 0.02830436, 0.02459982, -0.03438589, 0.00718705, -0.01798329, - -0.01594508, -0.00702430, -0.00952419, -0.00962701, -0.01307212, - -0.01749740, 0.01299602, 0.00587270, -0.36103108, -0.82039266, - -0.43092844, -0.08500097, -0.04361674, -0.00333482, 0.01250434, - -0.02538295, -0.00921797, 0.01645071, -0.01400872, 0.00317607, - 0.00003277, -0.01617646, -0.00616863, -0.00882661, 0.00466157, - 0.00353237, 0.91803104, -0.39503305, -0.02048964, 0.00060125, - 0.01980634, 0.00300109, 0.00313880, 0.00657337, 0.00715163, - 0.00000261, 0.00854276, -0.00154825, -0.00516128, 0.00909527, - 0.00095609, 0.00701196, -0.00221867, -0.00156741 }; - -/* right KLT transforms */ -const double WebRtcIsac_kKltT2Gain[36] = { - 0.14572837, -0.45446306, 0.61990621, -0.52197033, 0.32145074, - -0.11026900, -0.20698282, 0.48962182, -0.27127933, -0.33627476, - 0.65094037, -0.32715751, 0.40262573, -0.47844405, -0.33876075, - 0.44130653, 0.37383966, -0.39964662, -0.51730480, 0.06611973, - 0.49030187, 0.47512886, -0.02141226, -0.51129451, -0.58578569, - -0.39132064, -0.13187771, 0.15649421, 0.40735596, 0.54396897, - 0.40381276, 0.40904942, 0.41179766, 0.41167576, 0.40840251, - 0.40468132 }; - -const double WebRtcIsac_kKltT2Shape[36] = { - 0.13427386, -0.35132558, 0.52506528, -0.59419077, 0.45075085, - -0.16312057, 0.29857439, -0.58660147, 0.34265431, 0.20879510, - -0.56063262, 0.30238345, 0.43308283, -0.41186999, -0.35288681, - 0.42768996, 0.36094634, -0.45284910, -0.47116680, 0.02893449, - 0.54326135, 0.45249040, -0.06264420, -0.52283830, 0.57137758, - 0.44298139, 0.12617554, -0.20819946, -0.42324603, -0.48876443, - 0.39597050, 0.40713935, 0.41389880, 0.41512486, 0.41130400, - 0.40575001 }; - -/* means of log gains and LAR coefficients*/ -const double WebRtcIsac_kLpcMeansGain[12] = { - -6.86881911, -5.35075273, -6.86792680, -5.36200897, -6.86401538, - -5.36921533, -6.86802969, -5.36893966, -6.86538097, -5.36315063, - -6.85535304, -5.35155315 }; - -const double WebRtcIsac_kLpcMeansShape[108] = { - -0.91232981, 0.26258634, -0.33716701, 0.08477430, -0.03378426, - 0.14423909, 0.07036185, 0.06155019, 0.01490385, 0.04138740, - 0.01427317, 0.01288970, 0.83872106, 0.25750199, 0.07988929, - -0.01957923, 0.00831390, 0.01770300, -0.90957164, 0.25732216, - -0.33385344, 0.08735740, -0.03715332, 0.14584917, 0.06998990, - 0.06131968, 0.01504379, 0.04067339, 0.01428039, 0.01406460, - 0.83846243, 0.26169862, 0.08109025, -0.01767055, 0.00970539, - 0.01954310, -0.90490803, 0.24656405, -0.33578607, 0.08843286, - -0.03749139, 0.14443959, 0.07214669, 0.06170993, 0.01449947, - 0.04134309, 0.01314762, 0.01413471, 0.83895203, 0.26748062, - 0.08197507, -0.01781298, 0.00885967, 0.01922394, -0.90922472, - 0.24495889, -0.33921540, 0.08877169, -0.03581332, 0.14199172, - 0.07444032, 0.06185940, 0.01502054, 0.04185113, 0.01276579, - 0.01355457, 0.83645358, 0.26631720, 0.08119697, -0.01835449, - 0.00788512, 0.01846446, -0.90482253, 0.24658310, -0.34019734, - 0.08281090, -0.03486038, 0.14359248, 0.07401336, 0.06001471, - 0.01528421, 0.04254560, 0.01321472, 0.01240799, 0.83857127, - 0.26281654, 0.08174380, -0.02099842, 0.00755176, 0.01699448, - -0.90132307, 0.25174308, -0.33838268, 0.07883863, -0.02877906, - 0.14105407, 0.07220290, 0.06000352, 0.01684879, 0.04226844, - 0.01331331, 0.01269244, 0.83832138, 0.25467485, 0.08118028, - -0.02120528, 0.00747832, 0.01567212 }; diff --git a/modules/audio_coding/codecs/isac/main/source/lpc_tables.h b/modules/audio_coding/codecs/isac/main/source/lpc_tables.h deleted file mode 100644 index 56ff22c06c..0000000000 --- a/modules/audio_coding/codecs/isac/main/source/lpc_tables.h +++ /dev/null @@ -1,99 +0,0 @@ -/* - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/* - * lpc_tables.h - * - * header file for coding tables for the LPC coefficients - * - */ - -#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_LPC_TABLES_H_ -#define MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_LPC_TABLES_H_ - -#include "modules/audio_coding/codecs/isac/main/source/settings.h" -#include "modules/audio_coding/codecs/isac/main/source/structs.h" - -#define KLT_STEPSIZE 1.00000000 -#define KLT_NUM_AVG_GAIN 0 -#define KLT_NUM_AVG_SHAPE 0 -#define KLT_NUM_MODELS 3 -#define LPC_GAIN_SCALE 4.000f -#define LPC_LOBAND_SCALE 2.100f -#define LPC_LOBAND_ORDER ORDERLO -#define LPC_HIBAND_SCALE 0.450f -#define LPC_HIBAND_ORDER ORDERHI -#define LPC_GAIN_ORDER 2 - -#define LPC_SHAPE_ORDER (LPC_LOBAND_ORDER + LPC_HIBAND_ORDER) - -#define KLT_ORDER_GAIN (LPC_GAIN_ORDER * SUBFRAMES) -#define KLT_ORDER_SHAPE (LPC_SHAPE_ORDER * SUBFRAMES) - -/* cdf array for model indicator */ -extern const uint16_t WebRtcIsac_kQKltModelCdf[KLT_NUM_MODELS + 1]; - -/* pointer to cdf array for model indicator */ -extern const uint16_t* WebRtcIsac_kQKltModelCdfPtr[1]; - -/* initial cdf index for decoder of model indicator */ -extern const uint16_t WebRtcIsac_kQKltModelInitIndex[1]; - -/* offset to go from rounded value to quantization index */ -extern const short WebRtcIsac_kQKltQuantMinGain[12]; - -extern const short WebRtcIsac_kQKltQuantMinShape[108]; - -/* maximum quantization index */ -extern const uint16_t WebRtcIsac_kQKltMaxIndGain[12]; - -extern const uint16_t WebRtcIsac_kQKltMaxIndShape[108]; - -/* index offset */ -extern const uint16_t WebRtcIsac_kQKltOffsetGain[12]; - -extern const uint16_t WebRtcIsac_kQKltOffsetShape[108]; - -/* initial cdf index for KLT coefficients */ -extern const uint16_t WebRtcIsac_kQKltInitIndexGain[12]; - -extern const uint16_t WebRtcIsac_kQKltInitIndexShape[108]; - -/* quantizer representation levels */ -extern const double WebRtcIsac_kQKltLevelsGain[392]; - -extern const double WebRtcIsac_kQKltLevelsShape[578]; - -/* cdf tables for quantizer indices */ -extern const uint16_t WebRtcIsac_kQKltCdfGain[404]; - -extern const uint16_t WebRtcIsac_kQKltCdfShape[686]; - -/* pointers to cdf tables for quantizer indices */ -extern const uint16_t* WebRtcIsac_kQKltCdfPtrGain[12]; - -extern const uint16_t* WebRtcIsac_kQKltCdfPtrShape[108]; - -/* left KLT transforms */ -extern const double WebRtcIsac_kKltT1Gain[4]; - -extern const double WebRtcIsac_kKltT1Shape[324]; - -/* right KLT transforms */ -extern const double WebRtcIsac_kKltT2Gain[36]; - -extern const double WebRtcIsac_kKltT2Shape[36]; - -/* means of log gains and LAR coefficients */ -extern const double WebRtcIsac_kLpcMeansGain[12]; - -extern const double WebRtcIsac_kLpcMeansShape[108]; - -#endif /* MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_LPC_TABLES_H_ */ diff --git a/modules/audio_coding/codecs/isac/main/source/pitch_gain_tables.c b/modules/audio_coding/codecs/isac/main/source/pitch_gain_tables.c deleted file mode 100644 index 080432c3a5..0000000000 --- a/modules/audio_coding/codecs/isac/main/source/pitch_gain_tables.c +++ /dev/null @@ -1,104 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#include "modules/audio_coding/codecs/isac/main/source/pitch_gain_tables.h" -#include "modules/audio_coding/codecs/isac/main/source/settings.h" - -/* header file for coding tables for the pitch filter side-info in the entropy coder */ -/********************* Pitch Filter Gain Coefficient Tables ************************/ -/* cdf for quantized pitch filter gains */ -const uint16_t WebRtcIsac_kQPitchGainCdf[255] = { - 0, 2, 4, 6, 64, 901, 903, 905, 16954, 16956, - 16961, 17360, 17362, 17364, 17366, 17368, 17370, 17372, 17374, 17411, - 17514, 17516, 17583, 18790, 18796, 18802, 20760, 20777, 20782, 21722, - 21724, 21728, 21738, 21740, 21742, 21744, 21746, 21748, 22224, 22227, - 22230, 23214, 23229, 23239, 25086, 25108, 25120, 26088, 26094, 26098, - 26175, 26177, 26179, 26181, 26183, 26185, 26484, 26507, 26522, 27705, - 27731, 27750, 29767, 29799, 29817, 30866, 30883, 30885, 31025, 31029, - 31031, 31033, 31035, 31037, 31114, 31126, 31134, 32687, 32722, 32767, - 35718, 35742, 35757, 36943, 36952, 36954, 37115, 37128, 37130, 37132, - 37134, 37136, 37143, 37145, 37152, 38843, 38863, 38897, 47458, 47467, - 47474, 49040, 49061, 49063, 49145, 49157, 49159, 49161, 49163, 49165, - 49167, 49169, 49171, 49757, 49770, 49782, 61333, 61344, 61346, 62860, - 62883, 62885, 62887, 62889, 62891, 62893, 62895, 62897, 62899, 62901, - 62903, 62905, 62907, 62909, 65496, 65498, 65500, 65521, 65523, 65525, - 65527, 65529, 65531, 65533, 65535, 65535, 65535, 65535, 65535, 65535, - 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, - 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, - 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, - 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, - 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, - 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, - 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, - 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, - 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, - 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, - 65535, 65535, 65535, 65535, 65535}; - -/* index limits and ranges */ -const int16_t WebRtcIsac_kIndexLowerLimitGain[3] = { - -7, -2, -1}; - -const int16_t WebRtcIsac_kIndexUpperLimitGain[3] = { - 0, 3, 1}; - -const uint16_t WebRtcIsac_kIndexMultsGain[2] = { - 18, 3}; - -/* size of cdf table */ -const uint16_t WebRtcIsac_kQCdfTableSizeGain[1] = { - 256}; - -///////////////////////////FIXED POINT -/* mean values of pitch filter gains in FIXED point */ -const int16_t WebRtcIsac_kQMeanGain1Q12[144] = { - 843, 1092, 1336, 1222, 1405, 1656, 1500, 1815, 1843, 1838, 1839, 1843, 1843, 1843, 1843, 1843, - 1843, 1843, 814, 846, 1092, 1013, 1174, 1383, 1391, 1511, 1584, 1734, 1753, 1843, 1843, 1843, - 1843, 1843, 1843, 1843, 524, 689, 777, 845, 947, 1069, 1090, 1263, 1380, 1447, 1559, 1676, - 1645, 1749, 1843, 1843, 1843, 1843, 81, 477, 563, 611, 706, 806, 849, 1012, 1192, 1128, - 1330, 1489, 1425, 1576, 1826, 1741, 1843, 1843, 0, 290, 305, 356, 488, 575, 602, 741, - 890, 835, 1079, 1196, 1182, 1376, 1519, 1506, 1680, 1843, 0, 47, 97, 69, 289, 381, - 385, 474, 617, 664, 803, 1079, 935, 1160, 1269, 1265, 1506, 1741, 0, 0, 0, 0, - 112, 120, 190, 283, 442, 343, 526, 809, 684, 935, 1134, 1020, 1265, 1506, 0, 0, - 0, 0, 0, 0, 0, 111, 256, 87, 373, 597, 430, 684, 935, 770, 1020, 1265}; - -const int16_t WebRtcIsac_kQMeanGain2Q12[144] = { - 1760, 1525, 1285, 1747, 1671, 1393, 1843, 1826, 1555, 1843, 1784, 1606, 1843, 1843, 1711, 1843, - 1843, 1814, 1389, 1275, 1040, 1564, 1414, 1252, 1610, 1495, 1343, 1753, 1592, 1405, 1804, 1720, - 1475, 1843, 1814, 1581, 1208, 1061, 856, 1349, 1148, 994, 1390, 1253, 1111, 1495, 1343, 1178, - 1770, 1465, 1234, 1814, 1581, 1342, 1040, 793, 713, 1053, 895, 737, 1128, 1003, 861, 1277, - 1094, 981, 1475, 1192, 1019, 1581, 1342, 1098, 855, 570, 483, 833, 648, 540, 948, 744, - 572, 1009, 844, 636, 1234, 934, 685, 1342, 1217, 984, 537, 318, 124, 603, 423, 350, - 687, 479, 322, 791, 581, 430, 987, 671, 488, 1098, 849, 597, 283, 27, 0, 397, - 222, 38, 513, 271, 124, 624, 325, 157, 737, 484, 233, 849, 597, 343, 27, 0, - 0, 141, 0, 0, 256, 69, 0, 370, 87, 0, 484, 229, 0, 597, 343, 87}; - -const int16_t WebRtcIsac_kQMeanGain3Q12[144] = { - 1843, 1843, 1711, 1843, 1818, 1606, 1843, 1827, 1511, 1814, 1639, 1393, 1760, 1525, 1285, 1656, - 1419, 1176, 1835, 1718, 1475, 1841, 1650, 1387, 1648, 1498, 1287, 1600, 1411, 1176, 1522, 1299, - 1040, 1419, 1176, 928, 1773, 1461, 1128, 1532, 1355, 1202, 1429, 1260, 1115, 1398, 1151, 1025, - 1172, 1080, 790, 1176, 928, 677, 1475, 1147, 1019, 1276, 1096, 922, 1214, 1010, 901, 1057, - 893, 800, 1040, 796, 734, 928, 677, 424, 1137, 897, 753, 1120, 830, 710, 875, 751, - 601, 795, 642, 583, 790, 544, 475, 677, 474, 140, 987, 750, 482, 697, 573, 450, - 691, 487, 303, 661, 394, 332, 537, 303, 220, 424, 168, 0, 737, 484, 229, 624, - 348, 153, 441, 261, 136, 397, 166, 51, 283, 27, 0, 168, 0, 0, 484, 229, - 0, 370, 57, 0, 256, 43, 0, 141, 0, 0, 27, 0, 0, 0, 0, 0}; - - -const int16_t WebRtcIsac_kQMeanGain4Q12[144] = { - 1843, 1843, 1843, 1843, 1841, 1843, 1500, 1821, 1843, 1222, 1434, 1656, 843, 1092, 1336, 504, - 757, 1007, 1843, 1843, 1843, 1838, 1791, 1843, 1265, 1505, 1599, 965, 1219, 1425, 730, 821, - 1092, 249, 504, 757, 1783, 1819, 1843, 1351, 1567, 1727, 1096, 1268, 1409, 805, 961, 1131, - 444, 670, 843, 0, 249, 504, 1425, 1655, 1743, 1096, 1324, 1448, 822, 1019, 1199, 490, - 704, 867, 81, 450, 555, 0, 0, 249, 1247, 1428, 1530, 881, 1073, 1283, 610, 759, - 939, 278, 464, 645, 0, 200, 270, 0, 0, 0, 935, 1163, 1410, 528, 790, 1068, - 377, 499, 717, 173, 240, 274, 0, 43, 62, 0, 0, 0, 684, 935, 1182, 343, - 551, 735, 161, 262, 423, 0, 55, 27, 0, 0, 0, 0, 0, 0, 430, 684, - 935, 87, 377, 597, 0, 46, 256, 0, 0, 0, 0, 0, 0, 0, 0, 0}; diff --git a/modules/audio_coding/codecs/isac/main/source/pitch_gain_tables.h b/modules/audio_coding/codecs/isac/main/source/pitch_gain_tables.h deleted file mode 100644 index 145fd4e6aa..0000000000 --- a/modules/audio_coding/codecs/isac/main/source/pitch_gain_tables.h +++ /dev/null @@ -1,48 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/* - * pitch_gain_tables.h - * - * This file contains tables for the pitch filter side-info in the entropy - * coder. - * - */ - -#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_PITCH_GAIN_TABLES_H_ -#define MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_PITCH_GAIN_TABLES_H_ - -#include - -/* header file for coding tables for the pitch filter side-info in the entropy - * coder */ -/********************* Pitch Filter Gain Coefficient Tables - * ************************/ -/* cdf for quantized pitch filter gains */ -extern const uint16_t WebRtcIsac_kQPitchGainCdf[255]; - -/* index limits and ranges */ -extern const int16_t WebRtcIsac_kIndexLowerLimitGain[3]; - -extern const int16_t WebRtcIsac_kIndexUpperLimitGain[3]; -extern const uint16_t WebRtcIsac_kIndexMultsGain[2]; - -/* mean values of pitch filter gains */ -//(Y) -extern const int16_t WebRtcIsac_kQMeanGain1Q12[144]; -extern const int16_t WebRtcIsac_kQMeanGain2Q12[144]; -extern const int16_t WebRtcIsac_kQMeanGain3Q12[144]; -extern const int16_t WebRtcIsac_kQMeanGain4Q12[144]; -//(Y) - -/* size of cdf table */ -extern const uint16_t WebRtcIsac_kQCdfTableSizeGain[1]; - -#endif /* MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_PITCH_GAIN_TABLES_H_ */ diff --git a/modules/audio_coding/codecs/isac/main/source/pitch_lag_tables.c b/modules/audio_coding/codecs/isac/main/source/pitch_lag_tables.c deleted file mode 100644 index 57d12021ac..0000000000 --- a/modules/audio_coding/codecs/isac/main/source/pitch_lag_tables.c +++ /dev/null @@ -1,277 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#include "modules/audio_coding/codecs/isac/main/source/pitch_lag_tables.h" -#include "modules/audio_coding/codecs/isac/main/source/settings.h" - -/* header file for coding tables for the pitch filter side-info in the entropy coder */ -/********************* Pitch Filter Gain Coefficient Tables ************************/ - -/* tables for use with small pitch gain */ - -/* cdf for quantized pitch filter lags */ -const uint16_t WebRtcIsac_kQPitchLagCdf1Lo[127] = { - 0, 134, 336, 549, 778, 998, 1264, 1512, 1777, 2070, - 2423, 2794, 3051, 3361, 3708, 3979, 4315, 4610, 4933, 5269, - 5575, 5896, 6155, 6480, 6816, 7129, 7477, 7764, 8061, 8358, - 8718, 9020, 9390, 9783, 10177, 10543, 10885, 11342, 11795, 12213, - 12680, 13096, 13524, 13919, 14436, 14903, 15349, 15795, 16267, 16734, - 17266, 17697, 18130, 18632, 19080, 19447, 19884, 20315, 20735, 21288, - 21764, 22264, 22723, 23193, 23680, 24111, 24557, 25022, 25537, 26082, - 26543, 27090, 27620, 28139, 28652, 29149, 29634, 30175, 30692, 31273, - 31866, 32506, 33059, 33650, 34296, 34955, 35629, 36295, 36967, 37726, - 38559, 39458, 40364, 41293, 42256, 43215, 44231, 45253, 46274, 47359, - 48482, 49678, 50810, 51853, 53016, 54148, 55235, 56263, 57282, 58363, - 59288, 60179, 61076, 61806, 62474, 63129, 63656, 64160, 64533, 64856, - 65152, 65535, 65535, 65535, 65535, 65535, 65535}; - -const uint16_t WebRtcIsac_kQPitchLagCdf2Lo[20] = { - 0, 429, 3558, 5861, 8558, 11639, 15210, 19502, 24773, 31983, - 42602, 48567, 52601, 55676, 58160, 60172, 61889, 63235, 65383, 65535}; - -const uint16_t WebRtcIsac_kQPitchLagCdf3Lo[2] = { - 0, 65535}; - -const uint16_t WebRtcIsac_kQPitchLagCdf4Lo[10] = { - 0, 2966, 6368, 11182, 19431, 37793, 48532, 55353, 60626, 65535}; - -const uint16_t *WebRtcIsac_kQPitchLagCdfPtrLo[4] = {WebRtcIsac_kQPitchLagCdf1Lo, WebRtcIsac_kQPitchLagCdf2Lo, WebRtcIsac_kQPitchLagCdf3Lo, WebRtcIsac_kQPitchLagCdf4Lo}; - -/* size of first cdf table */ -const uint16_t WebRtcIsac_kQPitchLagCdfSizeLo[1] = {128}; - -/* index limits and ranges */ -const int16_t WebRtcIsac_kQIndexLowerLimitLagLo[4] = { --140, -9, 0, -4}; - -const int16_t WebRtcIsac_kQIndexUpperLimitLagLo[4] = { --20, 9, 0, 4}; - -/* initial index for arithmetic decoder */ -const uint16_t WebRtcIsac_kQInitIndexLagLo[3] = { - 10, 1, 5}; - -/* mean values of pitch filter lags */ -const double WebRtcIsac_kQMeanLag2Lo[19] = { --17.21385070, -15.82678944, -14.07123081, -12.03003877, -10.01311864, -8.00794627, -5.91162987, -3.89231876, -1.90220980, -0.01879275, - 1.89144232, 3.88123171, 5.92146992, 7.96435361, 9.98923648, 11.98266347, 13.96101002, 15.74855713, 17.10976611}; - -const double WebRtcIsac_kQMeanLag3Lo[1] = { - 0.00000000}; - -const double WebRtcIsac_kQMeanLag4Lo[9] = { --7.76246496, -5.92083980, -3.94095226, -1.89502305, 0.03724681, 1.93054221, 3.96443467, 5.91726366, 7.78434291}; - -const double WebRtcIsac_kQPitchLagStepsizeLo = 2.000000; - - -/* tables for use with medium pitch gain */ - -/* cdf for quantized pitch filter lags */ -const uint16_t WebRtcIsac_kQPitchLagCdf1Mid[255] = { - 0, 28, 61, 88, 121, 149, 233, 331, 475, 559, - 624, 661, 689, 712, 745, 791, 815, 843, 866, 922, - 959, 1024, 1061, 1117, 1178, 1238, 1280, 1350, 1453, 1513, - 1564, 1625, 1671, 1741, 1788, 1904, 2072, 2421, 2626, 2770, - 2840, 2900, 2942, 3012, 3068, 3115, 3147, 3194, 3254, 3319, - 3366, 3520, 3678, 3780, 3850, 3911, 3957, 4032, 4106, 4185, - 4292, 4474, 4683, 4842, 5019, 5191, 5321, 5428, 5540, 5675, - 5763, 5847, 5959, 6127, 6304, 6564, 6839, 7090, 7263, 7421, - 7556, 7728, 7872, 7984, 8142, 8361, 8580, 8743, 8938, 9227, - 9409, 9539, 9674, 9795, 9930, 10060, 10177, 10382, 10614, 10861, - 11038, 11271, 11415, 11629, 11792, 12044, 12193, 12416, 12574, 12821, - 13007, 13235, 13445, 13654, 13901, 14134, 14488, 15000, 15703, 16285, - 16504, 16797, 17086, 17328, 17579, 17807, 17998, 18268, 18538, 18836, - 19087, 19274, 19474, 19716, 19935, 20270, 20833, 21303, 21532, 21741, - 21978, 22207, 22523, 22770, 23054, 23613, 23943, 24204, 24399, 24651, - 24832, 25074, 25270, 25549, 25759, 26015, 26150, 26424, 26713, 27048, - 27342, 27504, 27681, 27854, 28021, 28207, 28412, 28664, 28859, 29064, - 29278, 29548, 29748, 30107, 30377, 30656, 30856, 31164, 31452, 31755, - 32011, 32328, 32626, 32919, 33319, 33789, 34329, 34925, 35396, 35973, - 36443, 36964, 37551, 38156, 38724, 39357, 40023, 40908, 41587, 42602, - 43924, 45037, 45810, 46597, 47421, 48291, 49092, 50051, 51448, 52719, - 53440, 54241, 54944, 55977, 56676, 57299, 57872, 58389, 59059, 59688, - 60237, 60782, 61094, 61573, 61890, 62290, 62658, 63030, 63217, 63454, - 63622, 63882, 64003, 64273, 64427, 64529, 64581, 64697, 64758, 64902, - 65414, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, - 65535, 65535, 65535, 65535, 65535}; - -const uint16_t WebRtcIsac_kQPitchLagCdf2Mid[36] = { - 0, 71, 335, 581, 836, 1039, 1323, 1795, 2258, 2608, - 3005, 3591, 4243, 5344, 7163, 10583, 16848, 28078, 49448, 57007, - 60357, 61850, 62837, 63437, 63872, 64188, 64377, 64614, 64774, 64949, - 65039, 65115, 65223, 65360, 65474, 65535}; - -const uint16_t WebRtcIsac_kQPitchLagCdf3Mid[2] = { - 0, 65535}; - -const uint16_t WebRtcIsac_kQPitchLagCdf4Mid[20] = { - 0, 28, 246, 459, 667, 1045, 1523, 2337, 4337, 11347, - 44231, 56709, 60781, 62243, 63161, 63969, 64608, 65062, 65502, 65535}; - -const uint16_t *WebRtcIsac_kQPitchLagCdfPtrMid[4] = {WebRtcIsac_kQPitchLagCdf1Mid, WebRtcIsac_kQPitchLagCdf2Mid, WebRtcIsac_kQPitchLagCdf3Mid, WebRtcIsac_kQPitchLagCdf4Mid}; - -/* size of first cdf table */ -const uint16_t WebRtcIsac_kQPitchLagCdfSizeMid[1] = {256}; - -/* index limits and ranges */ -const int16_t WebRtcIsac_kQIndexLowerLimitLagMid[4] = { --280, -17, 0, -9}; - -const int16_t WebRtcIsac_kQIndexUpperLimitLagMid[4] = { --40, 17, 0, 9}; - -/* initial index for arithmetic decoder */ -const uint16_t WebRtcIsac_kQInitIndexLagMid[3] = { - 18, 1, 10}; - -/* mean values of pitch filter lags */ -const double WebRtcIsac_kQMeanLag2Mid[35] = { --16.89183900, -15.86949778, -15.05476653, -14.00664348, -13.02793036, -12.07324237, -11.00542532, -10.11250602, -8.90792971, -8.02474753, --7.00426767, -5.94055287, -4.98251338, -3.91053158, -2.98820425, -1.93524245, -0.92978085, -0.01722509, 0.91317387, 1.92973955, - 2.96908851, 3.93728974, 4.96308471, 5.92244151, 7.08673497, 8.00993708, 9.04656316, 9.98538742, 10.97851694, 11.94772884, - 13.02426166, 14.00039951, 15.01347042, 15.80758023, 16.94086895}; - -const double WebRtcIsac_kQMeanLag3Mid[1] = { - 0.00000000}; - -const double WebRtcIsac_kQMeanLag4Mid[19] = { --8.60409403, -7.89198395, -7.03450280, -5.86260421, -4.93822322, -3.93078706, -2.91302322, -1.91824007, -0.87003282, 0.02822649, - 0.89951758, 1.87495484, 2.91802604, 3.96874074, 5.06571703, 5.93618227, 7.00520185, 7.88497726, 8.64160364}; - -const double WebRtcIsac_kQPitchLagStepsizeMid = 1.000000; - - -/* tables for use with large pitch gain */ - -/* cdf for quantized pitch filter lags */ -const uint16_t WebRtcIsac_kQPitchLagCdf1Hi[511] = { - 0, 7, 18, 33, 69, 105, 156, 228, 315, 612, - 680, 691, 709, 724, 735, 738, 742, 746, 749, 753, - 756, 760, 764, 774, 782, 785, 789, 796, 800, 803, - 807, 814, 818, 822, 829, 832, 847, 854, 858, 869, - 876, 883, 898, 908, 934, 977, 1010, 1050, 1060, 1064, - 1075, 1078, 1086, 1089, 1093, 1104, 1111, 1122, 1133, 1136, - 1151, 1162, 1183, 1209, 1252, 1281, 1339, 1364, 1386, 1401, - 1411, 1415, 1426, 1430, 1433, 1440, 1448, 1455, 1462, 1477, - 1487, 1495, 1502, 1506, 1509, 1516, 1524, 1531, 1535, 1542, - 1553, 1556, 1578, 1589, 1611, 1625, 1639, 1643, 1654, 1665, - 1672, 1687, 1694, 1705, 1708, 1719, 1730, 1744, 1752, 1759, - 1791, 1795, 1820, 1867, 1886, 1915, 1936, 1943, 1965, 1987, - 2041, 2099, 2161, 2175, 2200, 2211, 2226, 2233, 2244, 2251, - 2266, 2280, 2287, 2298, 2309, 2316, 2331, 2342, 2356, 2378, - 2403, 2418, 2447, 2497, 2544, 2602, 2863, 2895, 2903, 2935, - 2950, 2971, 3004, 3011, 3018, 3029, 3040, 3062, 3087, 3127, - 3152, 3170, 3199, 3243, 3293, 3322, 3340, 3377, 3402, 3427, - 3474, 3518, 3543, 3579, 3601, 3637, 3659, 3706, 3731, 3760, - 3818, 3847, 3869, 3901, 3920, 3952, 4068, 4169, 4220, 4271, - 4524, 4571, 4604, 4632, 4672, 4730, 4777, 4806, 4857, 4904, - 4951, 5002, 5031, 5060, 5107, 5150, 5212, 5266, 5331, 5382, - 5432, 5490, 5544, 5610, 5700, 5762, 5812, 5874, 5972, 6022, - 6091, 6163, 6232, 6305, 6402, 6540, 6685, 6880, 7090, 7271, - 7379, 7452, 7542, 7625, 7687, 7770, 7843, 7911, 7966, 8024, - 8096, 8190, 8252, 8320, 8411, 8501, 8585, 8639, 8751, 8842, - 8918, 8986, 9066, 9127, 9203, 9269, 9345, 9406, 9464, 9536, - 9612, 9667, 9735, 9844, 9931, 10036, 10119, 10199, 10260, 10358, - 10441, 10514, 10666, 10734, 10872, 10951, 11053, 11125, 11223, 11324, - 11516, 11664, 11737, 11816, 11892, 12008, 12120, 12200, 12280, 12392, - 12490, 12576, 12685, 12812, 12917, 13003, 13108, 13210, 13300, 13384, - 13470, 13579, 13673, 13771, 13879, 13999, 14136, 14201, 14368, 14614, - 14759, 14867, 14958, 15030, 15121, 15189, 15280, 15385, 15461, 15555, - 15653, 15768, 15884, 15971, 16069, 16145, 16210, 16279, 16380, 16463, - 16539, 16615, 16688, 16818, 16919, 17017, 18041, 18338, 18523, 18649, - 18790, 18917, 19047, 19167, 19315, 19460, 19601, 19731, 19858, 20068, - 20173, 20318, 20466, 20625, 20741, 20911, 21045, 21201, 21396, 21588, - 21816, 22022, 22305, 22547, 22786, 23072, 23322, 23600, 23879, 24168, - 24433, 24769, 25120, 25511, 25895, 26289, 26792, 27219, 27683, 28077, - 28566, 29094, 29546, 29977, 30491, 30991, 31573, 32105, 32594, 33173, - 33788, 34497, 35181, 35833, 36488, 37255, 37921, 38645, 39275, 39894, - 40505, 41167, 41790, 42431, 43096, 43723, 44385, 45134, 45858, 46607, - 47349, 48091, 48768, 49405, 49955, 50555, 51167, 51985, 52611, 53078, - 53494, 53965, 54435, 54996, 55601, 56125, 56563, 56838, 57244, 57566, - 57967, 58297, 58771, 59093, 59419, 59647, 59886, 60143, 60461, 60693, - 60917, 61170, 61416, 61634, 61891, 62122, 62310, 62455, 62632, 62839, - 63103, 63436, 63639, 63805, 63906, 64015, 64192, 64355, 64475, 64558, - 64663, 64742, 64811, 64865, 64916, 64956, 64981, 65025, 65068, 65115, - 65195, 65314, 65419, 65535, 65535, 65535, 65535, 65535, 65535, 65535, - 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, - 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, - 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, - 65535}; - -const uint16_t WebRtcIsac_kQPitchLagCdf2Hi[68] = { - 0, 7, 11, 22, 37, 52, 56, 59, 81, 85, - 89, 96, 115, 130, 137, 152, 170, 181, 193, 200, - 207, 233, 237, 259, 289, 318, 363, 433, 592, 992, - 1607, 3062, 6149, 12206, 25522, 48368, 58223, 61918, 63640, 64584, - 64943, 65098, 65206, 65268, 65294, 65335, 65350, 65372, 65387, 65402, - 65413, 65420, 65428, 65435, 65439, 65450, 65454, 65468, 65472, 65476, - 65483, 65491, 65498, 65505, 65516, 65520, 65528, 65535}; - -const uint16_t WebRtcIsac_kQPitchLagCdf3Hi[2] = { - 0, 65535}; - -const uint16_t WebRtcIsac_kQPitchLagCdf4Hi[35] = { - 0, 7, 19, 30, 41, 48, 63, 74, 82, 96, - 122, 152, 215, 330, 701, 2611, 10931, 48106, 61177, 64341, - 65112, 65238, 65309, 65338, 65364, 65379, 65401, 65427, 65453, 65465, - 65476, 65490, 65509, 65528, 65535}; - -const uint16_t *WebRtcIsac_kQPitchLagCdfPtrHi[4] = {WebRtcIsac_kQPitchLagCdf1Hi, WebRtcIsac_kQPitchLagCdf2Hi, WebRtcIsac_kQPitchLagCdf3Hi, WebRtcIsac_kQPitchLagCdf4Hi}; - -/* size of first cdf table */ -const uint16_t WebRtcIsac_kQPitchLagCdfSizeHi[1] = {512}; - -/* index limits and ranges */ -const int16_t WebRtcIsac_kQindexLowerLimitLagHi[4] = { --552, -34, 0, -16}; - -const int16_t WebRtcIsac_kQindexUpperLimitLagHi[4] = { --80, 32, 0, 17}; - -/* initial index for arithmetic decoder */ -const uint16_t WebRtcIsac_kQInitIndexLagHi[3] = { - 34, 1, 18}; - -/* mean values of pitch filter lags */ -const double WebRtcIsac_kQMeanLag2Hi[67] = { --17.07263295, -16.50000000, -15.83966081, -15.55613708, -14.96948007, -14.50000000, -14.00000000, -13.48377986, -13.00000000, -12.50000000, --11.93199636, -11.44530414, -11.04197641, -10.39910301, -10.15202337, -9.51322461, -8.93357741, -8.46456632, -8.10270672, -7.53751847, --6.98686404, -6.50000000, -6.08463150, -5.46872991, -5.00864717, -4.50163760, -4.01382410, -3.43856708, -2.96898001, -2.46554810, --1.96861004, -1.47106701, -0.97197237, -0.46561654, -0.00531409, 0.45767857, 0.96777907, 1.47507903, 1.97740425, 2.46695420, - 3.00695774, 3.47167185, 4.02712538, 4.49280007, 5.01087640, 5.48191963, 6.04916550, 6.51511058, 6.97297819, 7.46565499, - 8.01489405, 8.39912001, 8.91819757, 9.50000000, 10.11654065, 10.50000000, 11.03712583, 11.50000000, 12.00000000, 12.38964346, - 12.89466127, 13.43657881, 13.96013840, 14.46279912, 15.00000000, 15.39412269, 15.96662441}; - -const double WebRtcIsac_kQMeanLag3Hi[1] = { - 0.00000000}; - -const double WebRtcIsac_kQMeanLag4Hi[34] = { --7.98331221, -7.47988769, -7.03626557, -6.52708003, -6.06982173, -5.51856292, -5.05827033, -4.45909878, -3.99125864, -3.45308135, --3.02328139, -2.47297273, -1.94341995, -1.44699056, -0.93612243, -0.43012406, 0.01120357, 0.44054812, 0.93199883, 1.45669587, - 1.97218322, 2.50187419, 2.98748690, 3.49343202, 4.01660147, 4.50984306, 5.01402683, 5.58936797, 5.91787793, 6.59998900, - 6.85034315, 7.53503316, 7.87711194, 8.53631648}; - -const double WebRtcIsac_kQPitchLagStepsizeHi = 0.500000; - -/* transform matrix */ -const double WebRtcIsac_kTransform[4][4] = { -{-0.50000000, -0.50000000, -0.50000000, -0.50000000}, -{ 0.67082039, 0.22360680, -0.22360680, -0.67082039}, -{ 0.50000000, -0.50000000, -0.50000000, 0.50000000}, -{ 0.22360680, -0.67082039, 0.67082039, -0.22360680}}; - -/* transpose transform matrix */ -const double WebRtcIsac_kTransformTranspose[4][4] = { -{-0.50000000, 0.67082039, 0.50000000, 0.22360680}, -{-0.50000000, 0.22360680, -0.50000000, -0.67082039}, -{-0.50000000, -0.22360680, -0.50000000, 0.67082039}, -{-0.50000000, -0.67082039, 0.50000000, -0.22360680}}; - diff --git a/modules/audio_coding/codecs/isac/main/source/pitch_lag_tables.h b/modules/audio_coding/codecs/isac/main/source/pitch_lag_tables.h deleted file mode 100644 index b48e358a5a..0000000000 --- a/modules/audio_coding/codecs/isac/main/source/pitch_lag_tables.h +++ /dev/null @@ -1,116 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/* - * pitch_lag_tables.h - * - * This file contains tables for the pitch filter side-info in the entropy - * coder. - * - */ - -#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_PITCH_LAG_TABLES_H_ -#define MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_PITCH_LAG_TABLES_H_ - -#include - -/* header file for coding tables for the pitch filter side-info in the entropy - * coder */ -/********************* Pitch Filter Lag Coefficient Tables - * ************************/ - -/* tables for use with small pitch gain */ - -/* cdfs for quantized pitch lags */ -extern const uint16_t WebRtcIsac_kQPitchLagCdf1Lo[127]; -extern const uint16_t WebRtcIsac_kQPitchLagCdf2Lo[20]; -extern const uint16_t WebRtcIsac_kQPitchLagCdf3Lo[2]; -extern const uint16_t WebRtcIsac_kQPitchLagCdf4Lo[10]; - -extern const uint16_t* WebRtcIsac_kQPitchLagCdfPtrLo[4]; - -/* size of first cdf table */ -extern const uint16_t WebRtcIsac_kQPitchLagCdfSizeLo[1]; - -/* index limits and ranges */ -extern const int16_t WebRtcIsac_kQIndexLowerLimitLagLo[4]; -extern const int16_t WebRtcIsac_kQIndexUpperLimitLagLo[4]; - -/* initial index for arithmetic decoder */ -extern const uint16_t WebRtcIsac_kQInitIndexLagLo[3]; - -/* mean values of pitch filter lags */ -extern const double WebRtcIsac_kQMeanLag2Lo[19]; -extern const double WebRtcIsac_kQMeanLag3Lo[1]; -extern const double WebRtcIsac_kQMeanLag4Lo[9]; - -extern const double WebRtcIsac_kQPitchLagStepsizeLo; - -/* tables for use with medium pitch gain */ - -/* cdfs for quantized pitch lags */ -extern const uint16_t WebRtcIsac_kQPitchLagCdf1Mid[255]; -extern const uint16_t WebRtcIsac_kQPitchLagCdf2Mid[36]; -extern const uint16_t WebRtcIsac_kQPitchLagCdf3Mid[2]; -extern const uint16_t WebRtcIsac_kQPitchLagCdf4Mid[20]; - -extern const uint16_t* WebRtcIsac_kQPitchLagCdfPtrMid[4]; - -/* size of first cdf table */ -extern const uint16_t WebRtcIsac_kQPitchLagCdfSizeMid[1]; - -/* index limits and ranges */ -extern const int16_t WebRtcIsac_kQIndexLowerLimitLagMid[4]; -extern const int16_t WebRtcIsac_kQIndexUpperLimitLagMid[4]; - -/* initial index for arithmetic decoder */ -extern const uint16_t WebRtcIsac_kQInitIndexLagMid[3]; - -/* mean values of pitch filter lags */ -extern const double WebRtcIsac_kQMeanLag2Mid[35]; -extern const double WebRtcIsac_kQMeanLag3Mid[1]; -extern const double WebRtcIsac_kQMeanLag4Mid[19]; - -extern const double WebRtcIsac_kQPitchLagStepsizeMid; - -/* tables for use with large pitch gain */ - -/* cdfs for quantized pitch lags */ -extern const uint16_t WebRtcIsac_kQPitchLagCdf1Hi[511]; -extern const uint16_t WebRtcIsac_kQPitchLagCdf2Hi[68]; -extern const uint16_t WebRtcIsac_kQPitchLagCdf3Hi[2]; -extern const uint16_t WebRtcIsac_kQPitchLagCdf4Hi[35]; - -extern const uint16_t* WebRtcIsac_kQPitchLagCdfPtrHi[4]; - -/* size of first cdf table */ -extern const uint16_t WebRtcIsac_kQPitchLagCdfSizeHi[1]; - -/* index limits and ranges */ -extern const int16_t WebRtcIsac_kQindexLowerLimitLagHi[4]; -extern const int16_t WebRtcIsac_kQindexUpperLimitLagHi[4]; - -/* initial index for arithmetic decoder */ -extern const uint16_t WebRtcIsac_kQInitIndexLagHi[3]; - -/* mean values of pitch filter lags */ -extern const double WebRtcIsac_kQMeanLag2Hi[67]; -extern const double WebRtcIsac_kQMeanLag3Hi[1]; -extern const double WebRtcIsac_kQMeanLag4Hi[34]; - -extern const double WebRtcIsac_kQPitchLagStepsizeHi; - -/* transform matrix */ -extern const double WebRtcIsac_kTransform[4][4]; - -/* transpose transform matrix */ -extern const double WebRtcIsac_kTransformTranspose[4][4]; - -#endif /* MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_PITCH_LAG_TABLES_H_ */ diff --git a/modules/audio_coding/codecs/isac/main/source/spectrum_ar_model_tables.c b/modules/audio_coding/codecs/isac/main/source/spectrum_ar_model_tables.c deleted file mode 100644 index 839d5d4586..0000000000 --- a/modules/audio_coding/codecs/isac/main/source/spectrum_ar_model_tables.c +++ /dev/null @@ -1,139 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#include "modules/audio_coding/codecs/isac/main/source/spectrum_ar_model_tables.h" -#include "modules/audio_coding/codecs/isac/main/source/settings.h" - -/********************* AR Coefficient Tables ************************/ -/* cdf for quantized reflection coefficient 1 */ -const uint16_t WebRtcIsac_kQArRc1Cdf[NUM_AR_RC_QUANT_BAUNDARY] = { - 0, 2, 4, 129, 7707, 57485, 65495, 65527, 65529, 65531, - 65533, 65535}; - -/* cdf for quantized reflection coefficient 2 */ -const uint16_t WebRtcIsac_kQArRc2Cdf[NUM_AR_RC_QUANT_BAUNDARY] = { - 0, 2, 4, 7, 531, 25298, 64525, 65526, 65529, 65531, - 65533, 65535}; - -/* cdf for quantized reflection coefficient 3 */ -const uint16_t WebRtcIsac_kQArRc3Cdf[NUM_AR_RC_QUANT_BAUNDARY] = { - 0, 2, 4, 6, 620, 22898, 64843, 65527, 65529, 65531, - 65533, 65535}; - -/* cdf for quantized reflection coefficient 4 */ -const uint16_t WebRtcIsac_kQArRc4Cdf[NUM_AR_RC_QUANT_BAUNDARY] = { - 0, 2, 4, 6, 35, 10034, 60733, 65506, 65529, 65531, - 65533, 65535}; - -/* cdf for quantized reflection coefficient 5 */ -const uint16_t WebRtcIsac_kQArRc5Cdf[NUM_AR_RC_QUANT_BAUNDARY] = { - 0, 2, 4, 6, 36, 7567, 56727, 65385, 65529, 65531, - 65533, 65535}; - -/* cdf for quantized reflection coefficient 6 */ -const uint16_t WebRtcIsac_kQArRc6Cdf[NUM_AR_RC_QUANT_BAUNDARY] = { - 0, 2, 4, 6, 14, 6579, 57360, 65409, 65529, 65531, - 65533, 65535}; - -/* representation levels for quantized reflection coefficient 1 */ -const int16_t WebRtcIsac_kQArRc1Levels[NUM_AR_RC_QUANT_BAUNDARY - 1] = { - -32104, -29007, -23202, -15496, -9279, -2577, 5934, 17535, 24512, 29503, 32104 -}; - -/* representation levels for quantized reflection coefficient 2 */ -const int16_t WebRtcIsac_kQArRc2Levels[NUM_AR_RC_QUANT_BAUNDARY - 1] = { - -32104, -29503, -23494, -15261, -7309, -1399, 6158, 16381, 24512, 29503, 32104 -}; - -/* representation levels for quantized reflection coefficient 3 */ -const int16_t WebRtcIsac_kQArRc3Levels[NUM_AR_RC_QUANT_BAUNDARY - 1] = { --32104, -29503, -23157, -15186, -7347, -1359, 5829, 17535, 24512, 29503, 32104 -}; - -/* representation levels for quantized reflection coefficient 4 */ -const int16_t WebRtcIsac_kQArRc4Levels[NUM_AR_RC_QUANT_BAUNDARY - 1] = { --32104, -29503, -24512, -15362, -6665, -342, 6596, 14585, 24512, 29503, 32104 -}; - -/* representation levels for quantized reflection coefficient 5 */ -const int16_t WebRtcIsac_kQArRc5Levels[NUM_AR_RC_QUANT_BAUNDARY - 1] = { --32104, -29503, -24512, -15005, -6564, -106, 7123, 14920, 24512, 29503, 32104 -}; - -/* representation levels for quantized reflection coefficient 6 */ -const int16_t WebRtcIsac_kQArRc6Levels[NUM_AR_RC_QUANT_BAUNDARY - 1] = { --32104, -29503, -24512, -15096, -6656, -37, 7036, 14847, 24512, 29503, 32104 -}; - -/* quantization boundary levels for reflection coefficients */ -const int16_t WebRtcIsac_kQArBoundaryLevels[NUM_AR_RC_QUANT_BAUNDARY] = { --32768, -31441, -27566, -21458, -13612, -4663, 4663, 13612, 21458, 27566, 31441, -32767 -}; - -/* initial index for AR reflection coefficient quantizer and cdf table search */ -const uint16_t WebRtcIsac_kQArRcInitIndex[6] = { - 5, 5, 5, 5, 5, 5}; - -/* pointers to AR cdf tables */ -const uint16_t *WebRtcIsac_kQArRcCdfPtr[AR_ORDER] = { - WebRtcIsac_kQArRc1Cdf, WebRtcIsac_kQArRc2Cdf, WebRtcIsac_kQArRc3Cdf, - WebRtcIsac_kQArRc4Cdf, WebRtcIsac_kQArRc5Cdf, WebRtcIsac_kQArRc6Cdf -}; - -/* pointers to AR representation levels tables */ -const int16_t *WebRtcIsac_kQArRcLevelsPtr[AR_ORDER] = { - WebRtcIsac_kQArRc1Levels, WebRtcIsac_kQArRc2Levels, WebRtcIsac_kQArRc3Levels, - WebRtcIsac_kQArRc4Levels, WebRtcIsac_kQArRc5Levels, WebRtcIsac_kQArRc6Levels -}; - - -/******************** GAIN Coefficient Tables ***********************/ -/* cdf for Gain coefficient */ -const uint16_t WebRtcIsac_kQGainCdf[19] = { - 0, 2, 4, 6, 8, 10, 12, 14, 16, 1172, - 11119, 29411, 51699, 64445, 65527, 65529, 65531, 65533, 65535}; - -/* representation levels for quantized squared Gain coefficient */ -const int32_t WebRtcIsac_kQGain2Levels[18] = { -// 17, 28, 46, 76, 128, 215, 364, 709, 1268, 1960, 3405, 6078, 11286, 17827, 51918, 134498, 487432, 2048000}; - 128, 128, 128, 128, 128, 215, 364, 709, 1268, 1960, 3405, 6078, 11286, 17827, 51918, 134498, 487432, 2048000}; -/* quantization boundary levels for squared Gain coefficient */ -const int32_t WebRtcIsac_kQGain2BoundaryLevels[19] = { -0, 21, 35, 59, 99, 166, 280, 475, 815, 1414, 2495, 4505, 8397, 16405, 34431, 81359, 240497, 921600, 0x7FFFFFFF}; - -/* pointers to Gain cdf table */ -const uint16_t *WebRtcIsac_kQGainCdf_ptr[1] = {WebRtcIsac_kQGainCdf}; - -/* Gain initial index for gain quantizer and cdf table search */ -const uint16_t WebRtcIsac_kQGainInitIndex[1] = {11}; - -/************************* Cosine Tables ****************************/ -/* Cosine table */ -const int16_t WebRtcIsac_kCos[6][60] = { -{512, 512, 511, 510, 508, 507, 505, 502, 499, 496, 493, 489, 485, 480, 476, 470, 465, 459, 453, 447, -440, 433, 426, 418, 410, 402, 394, 385, 376, 367, 357, 348, 338, 327, 317, 306, 295, 284, 273, 262, -250, 238, 226, 214, 202, 190, 177, 165, 152, 139, 126, 113, 100, 87, 73, 60, 47, 33, 20, 7}, -{512, 510, 508, 503, 498, 491, 483, 473, 462, 450, 437, 422, 406, 389, 371, 352, 333, 312, 290, 268, -244, 220, 196, 171, 145, 120, 93, 67, 40, 13, -13, -40, -67, -93, -120, -145, -171, -196, -220, -244, --268, -290, -312, -333, -352, -371, -389, -406, -422, -437, -450, -462, -473, -483, -491, -498, -503, -508, -510, -512}, -{512, 508, 502, 493, 480, 465, 447, 426, 402, 376, 348, 317, 284, 250, 214, 177, 139, 100, 60, 20, --20, -60, -100, -139, -177, -214, -250, -284, -317, -348, -376, -402, -426, -447, -465, -480, -493, -502, -508, -512, --512, -508, -502, -493, -480, -465, -447, -426, -402, -376, -348, -317, -284, -250, -214, -177, -139, -100, -60, -20}, -{511, 506, 495, 478, 456, 429, 398, 362, 322, 279, 232, 183, 133, 80, 27, -27, -80, -133, -183, -232, --279, -322, -362, -398, -429, -456, -478, -495, -506, -511, -511, -506, -495, -478, -456, -429, -398, -362, -322, -279, --232, -183, -133, -80, -27, 27, 80, 133, 183, 232, 279, 322, 362, 398, 429, 456, 478, 495, 506, 511}, -{511, 502, 485, 459, 426, 385, 338, 284, 226, 165, 100, 33, -33, -100, -165, -226, -284, -338, -385, -426, --459, -485, -502, -511, -511, -502, -485, -459, -426, -385, -338, -284, -226, -165, -100, -33, 33, 100, 165, 226, -284, 338, 385, 426, 459, 485, 502, 511, 511, 502, 485, 459, 426, 385, 338, 284, 226, 165, 100, 33}, -{510, 498, 473, 437, 389, 333, 268, 196, 120, 40, -40, -120, -196, -268, -333, -389, -437, -473, -498, -510, --510, -498, -473, -437, -389, -333, -268, -196, -120, -40, 40, 120, 196, 268, 333, 389, 437, 473, 498, 510, -510, 498, 473, 437, 389, 333, 268, 196, 120, 40, -40, -120, -196, -268, -333, -389, -437, -473, -498, -510} -}; diff --git a/modules/audio_coding/codecs/isac/main/source/spectrum_ar_model_tables.h b/modules/audio_coding/codecs/isac/main/source/spectrum_ar_model_tables.h deleted file mode 100644 index d272be0dc3..0000000000 --- a/modules/audio_coding/codecs/isac/main/source/spectrum_ar_model_tables.h +++ /dev/null @@ -1,79 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/* - * spectrum_ar_model_tables.h - * - * This file contains definitions of tables with AR coefficients, - * Gain coefficients and cosine tables. - * - */ - -#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_SPECTRUM_AR_MODEL_TABLES_H_ -#define MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_SPECTRUM_AR_MODEL_TABLES_H_ - -#include "modules/audio_coding/codecs/isac/main/source/structs.h" - -#define NUM_AR_RC_QUANT_BAUNDARY 12 - -/********************* AR Coefficient Tables ************************/ -/* cdf for quantized reflection coefficient 1 */ -extern const uint16_t WebRtcIsac_kQArRc1Cdf[NUM_AR_RC_QUANT_BAUNDARY]; - -/* cdf for quantized reflection coefficient 2 */ -extern const uint16_t WebRtcIsac_kQArRc2Cdf[NUM_AR_RC_QUANT_BAUNDARY]; - -/* cdf for quantized reflection coefficient 3 */ -extern const uint16_t WebRtcIsac_kQArRc3Cdf[NUM_AR_RC_QUANT_BAUNDARY]; - -/* cdf for quantized reflection coefficient 4 */ -extern const uint16_t WebRtcIsac_kQArRc4Cdf[NUM_AR_RC_QUANT_BAUNDARY]; - -/* cdf for quantized reflection coefficient 5 */ -extern const uint16_t WebRtcIsac_kQArRc5Cdf[NUM_AR_RC_QUANT_BAUNDARY]; - -/* cdf for quantized reflection coefficient 6 */ -extern const uint16_t WebRtcIsac_kQArRc6Cdf[NUM_AR_RC_QUANT_BAUNDARY]; - -/* quantization boundary levels for reflection coefficients */ -extern const int16_t WebRtcIsac_kQArBoundaryLevels[NUM_AR_RC_QUANT_BAUNDARY]; - -/* initial indices for AR reflection coefficient quantizer and cdf table search - */ -extern const uint16_t WebRtcIsac_kQArRcInitIndex[AR_ORDER]; - -/* pointers to AR cdf tables */ -extern const uint16_t* WebRtcIsac_kQArRcCdfPtr[AR_ORDER]; - -/* pointers to AR representation levels tables */ -extern const int16_t* WebRtcIsac_kQArRcLevelsPtr[AR_ORDER]; - -/******************** GAIN Coefficient Tables ***********************/ -/* cdf for Gain coefficient */ -extern const uint16_t WebRtcIsac_kQGainCdf[19]; - -/* representation levels for quantized Gain coefficient */ -extern const int32_t WebRtcIsac_kQGain2Levels[18]; - -/* squared quantization boundary levels for Gain coefficient */ -extern const int32_t WebRtcIsac_kQGain2BoundaryLevels[19]; - -/* pointer to Gain cdf table */ -extern const uint16_t* WebRtcIsac_kQGainCdf_ptr[1]; - -/* Gain initial index for gain quantizer and cdf table search */ -extern const uint16_t WebRtcIsac_kQGainInitIndex[1]; - -/************************* Cosine Tables ****************************/ -/* Cosine table */ -extern const int16_t WebRtcIsac_kCos[6][60]; - -#endif /* MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_SPECTRUM_AR_MODEL_TABLES_H_ \ - */ diff --git a/modules/audio_coding/codecs/isac/main/source/transform.c b/modules/audio_coding/codecs/isac/main/source/transform.c deleted file mode 100644 index 082ad941c4..0000000000 --- a/modules/audio_coding/codecs/isac/main/source/transform.c +++ /dev/null @@ -1,126 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#include - -#include "modules/audio_coding/codecs/isac/main/source/settings.h" -#include "modules/audio_coding/codecs/isac/main/source/codec.h" -#include "modules/audio_coding/codecs/isac/main/source/os_specific_inline.h" -#include "modules/third_party/fft/fft.h" - -void WebRtcIsac_InitTransform(TransformTables* tables) { - int k; - double fact, phase; - - fact = PI / (FRAMESAMPLES_HALF); - phase = 0.0; - for (k = 0; k < FRAMESAMPLES_HALF; k++) { - tables->costab1[k] = cos(phase); - tables->sintab1[k] = sin(phase); - phase += fact; - } - - fact = PI * ((double) (FRAMESAMPLES_HALF - 1)) / ((double) FRAMESAMPLES_HALF); - phase = 0.5 * fact; - for (k = 0; k < FRAMESAMPLES_QUARTER; k++) { - tables->costab2[k] = cos(phase); - tables->sintab2[k] = sin(phase); - phase += fact; - } -} - -void WebRtcIsac_Time2Spec(const TransformTables* tables, - double* inre1, - double* inre2, - int16_t* outreQ7, - int16_t* outimQ7, - FFTstr* fftstr_obj) { - int k; - int dims[1]; - double tmp1r, tmp1i, xr, xi, yr, yi, fact; - double tmpre[FRAMESAMPLES_HALF], tmpim[FRAMESAMPLES_HALF]; - - - dims[0] = FRAMESAMPLES_HALF; - - - /* Multiply with complex exponentials and combine into one complex vector */ - fact = 0.5 / sqrt(FRAMESAMPLES_HALF); - for (k = 0; k < FRAMESAMPLES_HALF; k++) { - tmp1r = tables->costab1[k]; - tmp1i = tables->sintab1[k]; - tmpre[k] = (inre1[k] * tmp1r + inre2[k] * tmp1i) * fact; - tmpim[k] = (inre2[k] * tmp1r - inre1[k] * tmp1i) * fact; - } - - - /* Get DFT */ - WebRtcIsac_Fftns(1, dims, tmpre, tmpim, -1, 1.0, fftstr_obj); - - /* Use symmetry to separate into two complex vectors and center frames in time around zero */ - for (k = 0; k < FRAMESAMPLES_QUARTER; k++) { - xr = tmpre[k] + tmpre[FRAMESAMPLES_HALF - 1 - k]; - yi = -tmpre[k] + tmpre[FRAMESAMPLES_HALF - 1 - k]; - xi = tmpim[k] - tmpim[FRAMESAMPLES_HALF - 1 - k]; - yr = tmpim[k] + tmpim[FRAMESAMPLES_HALF - 1 - k]; - - tmp1r = tables->costab2[k]; - tmp1i = tables->sintab2[k]; - outreQ7[k] = (int16_t)WebRtcIsac_lrint((xr * tmp1r - xi * tmp1i) * 128.0); - outimQ7[k] = (int16_t)WebRtcIsac_lrint((xr * tmp1i + xi * tmp1r) * 128.0); - outreQ7[FRAMESAMPLES_HALF - 1 - k] = (int16_t)WebRtcIsac_lrint((-yr * tmp1i - yi * tmp1r) * 128.0); - outimQ7[FRAMESAMPLES_HALF - 1 - k] = (int16_t)WebRtcIsac_lrint((-yr * tmp1r + yi * tmp1i) * 128.0); - } -} - -void WebRtcIsac_Spec2time(const TransformTables* tables, - double* inre, - double* inim, - double* outre1, - double* outre2, - FFTstr* fftstr_obj) { - int k; - double tmp1r, tmp1i, xr, xi, yr, yi, fact; - - int dims; - - dims = FRAMESAMPLES_HALF; - - for (k = 0; k < FRAMESAMPLES_QUARTER; k++) { - /* Move zero in time to beginning of frames */ - tmp1r = tables->costab2[k]; - tmp1i = tables->sintab2[k]; - xr = inre[k] * tmp1r + inim[k] * tmp1i; - xi = inim[k] * tmp1r - inre[k] * tmp1i; - yr = -inim[FRAMESAMPLES_HALF - 1 - k] * tmp1r - inre[FRAMESAMPLES_HALF - 1 - k] * tmp1i; - yi = -inre[FRAMESAMPLES_HALF - 1 - k] * tmp1r + inim[FRAMESAMPLES_HALF - 1 - k] * tmp1i; - - /* Combine into one vector, z = x + j * y */ - outre1[k] = xr - yi; - outre1[FRAMESAMPLES_HALF - 1 - k] = xr + yi; - outre2[k] = xi + yr; - outre2[FRAMESAMPLES_HALF - 1 - k] = -xi + yr; - } - - - /* Get IDFT */ - WebRtcIsac_Fftns(1, &dims, outre1, outre2, 1, FRAMESAMPLES_HALF, fftstr_obj); - - - /* Demodulate and separate */ - fact = sqrt(FRAMESAMPLES_HALF); - for (k = 0; k < FRAMESAMPLES_HALF; k++) { - tmp1r = tables->costab1[k]; - tmp1i = tables->sintab1[k]; - xr = (outre1[k] * tmp1r - outre2[k] * tmp1i) * fact; - outre2[k] = (outre2[k] * tmp1r + outre1[k] * tmp1i) * fact; - outre1[k] = xr; - } -} diff --git a/modules/audio_coding/codecs/isac/main/test/ReleaseTest-API/ReleaseTest-API.cc b/modules/audio_coding/codecs/isac/main/test/ReleaseTest-API/ReleaseTest-API.cc deleted file mode 100644 index ee72b07dc3..0000000000 --- a/modules/audio_coding/codecs/isac/main/test/ReleaseTest-API/ReleaseTest-API.cc +++ /dev/null @@ -1,942 +0,0 @@ -/* - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -// ReleaseTest-API.cpp : Defines the entry point for the console application. -// - -#include -#include -#include -#include -#include - -#include - -/* include API */ -#include "modules/audio_coding/codecs/isac/main/include/isac.h" -#include "modules/audio_coding/codecs/isac/main/util/utility.h" - -/* Defines */ -#define SEED_FILE \ - "randseed.txt" /* Used when running decoder on garbage data \ - */ -#define MAX_FRAMESAMPLES \ - 960 /* max number of samples per frame \ - (= 60 ms frame & 16 kHz) or \ - (= 30 ms frame & 32 kHz) */ -#define FRAMESAMPLES_10ms 160 /* number of samples per 10ms frame */ -#define SWBFRAMESAMPLES_10ms 320 -//#define FS 16000 /* sampling frequency (Hz) */ - -#ifdef WIN32 -#ifndef CLOCKS_PER_SEC -#define CLOCKS_PER_SEC 1000 /* Runtime statistics */ -#endif -#endif - -int main(int argc, char* argv[]) { - char inname[100], outname[100], bottleneck_file[100], vadfile[100]; - FILE *inp, *outp, *f_bn = NULL, *vadp = NULL, *bandwidthp; - int framecnt, endfile; - - size_t i; - int errtype, VADusage = 0, packetLossPercent = 0; - int16_t CodingMode; - int32_t bottleneck = 0; - int framesize = 30; /* ms */ - int cur_framesmpls, err; - - /* Runtime statistics */ - double starttime, runtime, length_file; - - size_t stream_len = 0; - int declen = 0, declenTC = 0; - bool lostFrame = false; - - int16_t shortdata[SWBFRAMESAMPLES_10ms]; - int16_t vaddata[SWBFRAMESAMPLES_10ms * 3]; - int16_t decoded[MAX_FRAMESAMPLES << 1]; - int16_t decodedTC[MAX_FRAMESAMPLES << 1]; - uint16_t streamdata[500]; - int16_t speechType[1]; - int16_t rateBPS = 0; - int16_t fixedFL = 0; - int16_t payloadSize = 0; - int32_t payloadRate = 0; - int setControlBWE = 0; - short FL, testNum; - char version_number[20]; - FILE* plFile; - int32_t sendBN; - -#if !defined(NDEBUG) - FILE* fy; - double kbps; -#endif - size_t totalbits = 0; - int totalsmpls = 0; - - /* If use GNS file */ - FILE* fp_gns = NULL; - char gns_file[100]; - size_t maxStreamLen30 = 0; - size_t maxStreamLen60 = 0; - short sampFreqKHz = 32; - short samplesIn10Ms; - // FILE logFile; - bool doTransCoding = false; - int32_t rateTransCoding = 0; - uint8_t streamDataTransCoding[1200]; - size_t streamLenTransCoding = 0; - FILE* transCodingFile = NULL; - FILE* transcodingBitstream = NULL; - size_t numTransCodingBytes = 0; - - /* only one structure used for ISAC encoder */ - ISACStruct* ISAC_main_inst = NULL; - ISACStruct* decoderTransCoding = NULL; - - BottleNeckModel BN_data; - -#if !defined(NDEBUG) - fy = fopen("bit_rate.dat", "w"); - fclose(fy); - fy = fopen("bytes_frames.dat", "w"); - fclose(fy); -#endif - - /* Handling wrong input arguments in the command line */ - if ((argc < 3) || (argc > 17)) { - printf("\n\nWrong number of arguments or flag values.\n\n"); - - printf("\n"); - WebRtcIsac_version(version_number); - printf("iSAC-swb version %s \n\n", version_number); - - printf("Usage:\n\n"); - printf("%s [-I] bottleneck_value infile outfile \n\n", argv[0]); - printf("with:\n"); - printf("[-FS num] : sampling frequency in kHz, valid values are\n"); - printf(" 16 & 32, with 16 as default.\n"); - printf("[-I] : if -I option is specified, the coder will use\n"); - printf(" an instantaneous Bottleneck value. If not, it\n"); - printf(" will be an adaptive Bottleneck value.\n"); - printf("[-assign] : Use Assign API.\n"); - printf("[-B num] : the value of the bottleneck provided either\n"); - printf(" as a fixed value in bits/sec (e.g. 25000) or\n"); - printf(" read from a file (e.g. bottleneck.txt)\n"); - printf("[-INITRATE num] : Set a new value for initial rate. Note! Only\n"); - printf(" used in adaptive mode.\n"); - printf("[-FL num] : Set (initial) frame length in msec. Valid\n"); - printf(" lengths are 30 and 60 msec.\n"); - printf("[-FIXED_FL] : Frame length will be fixed to initial value.\n"); - printf("[-MAX num] : Set the limit for the payload size of iSAC\n"); - printf(" in bytes. Minimum 100 maximum 400.\n"); - printf("[-MAXRATE num] : Set the maxrate for iSAC in bits per second.\n"); - printf(" Minimum 32000, maximum 53400.\n"); - printf("[-F num] : if -F option is specified, the test function\n"); - printf(" will run the iSAC API fault scenario\n"); - printf(" specified by the supplied number.\n"); - printf(" F 1 - Call encoder prior to init encoder call\n"); - printf(" F 2 - Call decoder prior to init decoder call\n"); - printf(" F 3 - Call decoder prior to encoder call\n"); - printf(" F 4 - Call decoder with a too short coded\n"); - printf(" sequence\n"); - printf(" F 5 - Call decoder with a too long coded\n"); - printf(" sequence\n"); - printf(" F 6 - Call decoder with random bit stream\n"); - printf(" F 7 - Call init encoder/decoder at random\n"); - printf(" during a call\n"); - printf(" F 8 - Call encoder/decoder without having\n"); - printf(" allocated memory for encoder/decoder\n"); - printf(" instance\n"); - printf(" F 9 - Call decodeB without calling decodeA\n"); - printf(" F 10 - Call decodeB with garbage data\n"); - printf("[-PL num] : if -PL option is specified \n"); - printf("[-T rate file] : test trans-coding with target bottleneck\n"); - printf(" 'rate' bits/sec\n"); - printf(" the output file is written to 'file'\n"); - printf("[-LOOP num] : number of times to repeat coding the input\n"); - printf(" file for stress testing\n"); - // printf("[-CE num] : Test of APIs used by Conference Engine.\n"); - // printf(" CE 1 - getNewBitstream, getBWE \n"); - // printf(" (CE 2 - RESERVED for transcoding)\n"); - // printf(" CE 3 - getSendBWE, setSendBWE. \n"); - // printf("-L filename : write the logging info into file - // (appending)\n"); - printf("infile : Normal speech input file\n"); - printf("outfile : Speech output file\n"); - exit(0); - } - - /* Print version number */ - printf("-------------------------------------------------\n"); - WebRtcIsac_version(version_number); - printf("iSAC version %s \n\n", version_number); - - /* Loop over all command line arguments */ - CodingMode = 0; - testNum = 0; - // logFile = NULL; - char transCodingFileName[500]; - int16_t totFileLoop = 0; - int16_t numFileLoop = 0; - for (i = 1; i + 2 < static_cast(argc); i++) { - if (!strcmp("-LOOP", argv[i])) { - i++; - totFileLoop = (int16_t)atol(argv[i]); - if (totFileLoop <= 0) { - fprintf(stderr, "Invalid number of runs for the given input file, %d.", - totFileLoop); - exit(0); - } - } - - if (!strcmp("-T", argv[i])) { - doTransCoding = true; - i++; - rateTransCoding = atoi(argv[i]); - i++; - strcpy(transCodingFileName, argv[i]); - } - - /* Set Sampling Rate */ - if (!strcmp("-FS", argv[i])) { - i++; - sampFreqKHz = atoi(argv[i]); - } - - /* Instantaneous mode */ - if (!strcmp("-I", argv[i])) { - printf("Instantaneous BottleNeck\n"); - CodingMode = 1; - } - - /* Set (initial) bottleneck value */ - if (!strcmp("-INITRATE", argv[i])) { - rateBPS = atoi(argv[i + 1]); - setControlBWE = 1; - if ((rateBPS < 10000) || (rateBPS > 32000)) { - printf( - "\n%d is not a initial rate. Valid values are in the range " - "10000 to 32000.\n", - rateBPS); - exit(0); - } - printf("New initial rate: %d\n", rateBPS); - i++; - } - - /* Set (initial) framelength */ - if (!strcmp("-FL", argv[i])) { - framesize = atoi(argv[i + 1]); - if ((framesize != 30) && (framesize != 60)) { - printf( - "\n%d is not a valid frame length. Valid length are 30 and 60 " - "msec.\n", - framesize); - exit(0); - } - setControlBWE = 1; - printf("Frame Length: %d\n", framesize); - i++; - } - - /* Fixed frame length */ - if (!strcmp("-FIXED_FL", argv[i])) { - fixedFL = 1; - setControlBWE = 1; - printf("Fixed Frame Length\n"); - } - - /* Set maximum allowed payload size in bytes */ - if (!strcmp("-MAX", argv[i])) { - payloadSize = atoi(argv[i + 1]); - printf("Maximum Payload Size: %d\n", payloadSize); - i++; - } - - /* Set maximum rate in bytes */ - if (!strcmp("-MAXRATE", argv[i])) { - payloadRate = atoi(argv[i + 1]); - printf("Maximum Rate in kbps: %d\n", payloadRate); - i++; - } - - /* Test of fault scenarious */ - if (!strcmp("-F", argv[i])) { - testNum = atoi(argv[i + 1]); - printf("Fault test: %d\n", testNum); - if (testNum < 1 || testNum > 10) { - printf( - "\n%d is not a valid Fault Scenario number. Valid Fault " - "Scenarios are numbered 1-10.\n", - testNum); - exit(0); - } - i++; - } - - /* Packet loss test */ - if (!strcmp("-PL", argv[i])) { - if (isdigit(static_cast(*argv[i + 1]))) { - packetLossPercent = atoi(argv[i + 1]); - if ((packetLossPercent < 0) | (packetLossPercent > 100)) { - printf("\nInvalid packet loss perentage \n"); - exit(0); - } - if (packetLossPercent > 0) { - printf("Simulating %d %% of independent packet loss\n", - packetLossPercent); - } else { - printf("\nNo Packet Loss Is Simulated \n"); - } - } else { - plFile = fopen(argv[i + 1], "rb"); - if (plFile == NULL) { - printf("\n couldn't open the frameloss file: %s\n", argv[i + 1]); - exit(0); - } - printf("Simulating packet loss through the given channel file: %s\n", - argv[i + 1]); - } - i++; - } - - /* Random packetlosses */ - if (!strcmp("-rnd", argv[i])) { - srand((unsigned int)time(NULL)); - printf("Random pattern in lossed packets \n"); - } - - /* Use gns file */ - if (!strcmp("-G", argv[i])) { - sscanf(argv[i + 1], "%s", gns_file); - fp_gns = fopen(gns_file, "rb"); - if (fp_gns == NULL) { - printf("Cannot read file %s.\n", gns_file); - exit(0); - } - i++; - } - - // make it with '-B' - /* Get Bottleneck value */ - if (!strcmp("-B", argv[i])) { - i++; - bottleneck = atoi(argv[i]); - if (bottleneck == 0) { - sscanf(argv[i], "%s", bottleneck_file); - f_bn = fopen(bottleneck_file, "rb"); - if (f_bn == NULL) { - printf( - "Error No value provided for BottleNeck and cannot read file " - "%s.\n", - bottleneck_file); - exit(0); - } else { - printf("reading bottleneck rates from file %s\n\n", bottleneck_file); - if (fscanf(f_bn, "%d", &bottleneck) == EOF) { - /* Set pointer to beginning of file */ - fseek(f_bn, 0L, SEEK_SET); - if (fscanf(f_bn, "%d", &bottleneck) == EOF) { - exit(0); - } - } - - /* Bottleneck is a cosine function - * Matlab code for writing the bottleneck file: - * BottleNeck_10ms = 20e3 + 10e3 * cos((0:5999)/5999*2*pi); - * fid = fopen('bottleneck.txt', 'wb'); - * fprintf(fid, '%d\n', BottleNeck_10ms); fclose(fid); - */ - } - } else { - printf("\nfixed bottleneck rate of %d bits/s\n\n", bottleneck); - } - } - /* Run Conference Engine APIs */ - // Do not test it in the first release - // - // if(!strcmp ("-CE", argv[i])) - // { - // testCE = atoi(argv[i + 1]); - // if(testCE==1) - // { - // i++; - // scale = (float)atof( argv[i+1] ); - // } - // else if(testCE == 2) - // { - // printf("\nCE-test 2 (transcoding) not implemented.\n"); - // exit(0); - // } - // else if(testCE < 1 || testCE > 3) - // { - // printf("\n%d is not a valid CE-test number. Valid CE tests - // are 1-3.\n", testCE); - // exit(0); - // } - // printf("CE-test number: %d\n", testCE); - // i++; - // } - } - - if (CodingMode == 0) { - printf("\nAdaptive BottleNeck\n"); - } - - switch (sampFreqKHz) { - case 16: { - printf("iSAC Wideband.\n"); - samplesIn10Ms = FRAMESAMPLES_10ms; - break; - } - case 32: { - printf("iSAC Supper-Wideband.\n"); - samplesIn10Ms = SWBFRAMESAMPLES_10ms; - break; - } - default: - printf("Unsupported sampling frequency %d kHz", sampFreqKHz); - exit(0); - } - - /* Get Input and Output files */ - sscanf(argv[argc - 2], "%s", inname); - sscanf(argv[argc - 1], "%s", outname); - printf("\nInput file: %s\n", inname); - printf("Output file: %s\n\n", outname); - if ((inp = fopen(inname, "rb")) == NULL) { - printf(" Error iSAC Cannot read file %s.\n", inname); - std::cout << std::flush; - exit(1); - } - - if ((outp = fopen(outname, "wb")) == NULL) { - printf(" Error iSAC Cannot write file %s.\n", outname); - std::cout << std::flush; - getc(stdin); - exit(1); - } - if (VADusage) { - if ((vadp = fopen(vadfile, "rb")) == NULL) { - printf(" Error iSAC Cannot read file %s.\n", vadfile); - std::cout << std::flush; - exit(1); - } - } - - if ((bandwidthp = fopen("bwe.pcm", "wb")) == NULL) { - printf(" Error iSAC Cannot read file %s.\n", "bwe.pcm"); - std::cout << std::flush; - exit(1); - } - - starttime = clock() / (double)CLOCKS_PER_SEC; /* Runtime statistics */ - - /* Initialize the ISAC and BN structs */ - if (testNum != 8) { - err = WebRtcIsac_Create(&ISAC_main_inst); - WebRtcIsac_SetEncSampRate(ISAC_main_inst, sampFreqKHz * 1000); - WebRtcIsac_SetDecSampRate(ISAC_main_inst, - sampFreqKHz >= 32 ? 32000 : 16000); - /* Error check */ - if (err < 0) { - printf("\n\n Error in create.\n\n"); - std::cout << std::flush; - exit(EXIT_FAILURE); - } - } - BN_data.arrival_time = 0; - BN_data.sample_count = 0; - BN_data.rtp_number = 0; - - /* Initialize encoder and decoder */ - framecnt = 0; - endfile = 0; - - if (doTransCoding) { - WebRtcIsac_Create(&decoderTransCoding); - WebRtcIsac_SetEncSampRate(decoderTransCoding, sampFreqKHz * 1000); - WebRtcIsac_SetDecSampRate(decoderTransCoding, - sampFreqKHz >= 32 ? 32000 : 16000); - WebRtcIsac_DecoderInit(decoderTransCoding); - transCodingFile = fopen(transCodingFileName, "wb"); - if (transCodingFile == NULL) { - printf("Could not open %s to output trans-coding.\n", - transCodingFileName); - exit(0); - } - strcat(transCodingFileName, ".bit"); - transcodingBitstream = fopen(transCodingFileName, "wb"); - if (transcodingBitstream == NULL) { - printf("Could not open %s to write the bit-stream of transcoder.\n", - transCodingFileName); - exit(0); - } - } - - if (testNum != 1) { - if (WebRtcIsac_EncoderInit(ISAC_main_inst, CodingMode) < 0) { - printf("Error could not initialize the encoder \n"); - std::cout << std::flush; - return 0; - } - } - if (testNum != 2) - WebRtcIsac_DecoderInit(ISAC_main_inst); - if (CodingMode == 1) { - err = WebRtcIsac_Control(ISAC_main_inst, bottleneck, framesize); - if (err < 0) { - /* exit if returned with error */ - errtype = WebRtcIsac_GetErrorCode(ISAC_main_inst); - printf("\n\n Error in initialization (control): %d.\n\n", errtype); - std::cout << std::flush; - if (testNum == 0) { - exit(EXIT_FAILURE); - } - } - } - - if ((setControlBWE) && (CodingMode == 0)) { - err = WebRtcIsac_ControlBwe(ISAC_main_inst, rateBPS, framesize, fixedFL); - if (err < 0) { - /* exit if returned with error */ - errtype = WebRtcIsac_GetErrorCode(ISAC_main_inst); - - printf("\n\n Error in Control BWE: %d.\n\n", errtype); - std::cout << std::flush; - exit(EXIT_FAILURE); - } - } - - if (payloadSize != 0) { - err = WebRtcIsac_SetMaxPayloadSize(ISAC_main_inst, payloadSize); - if (err < 0) { - /* exit if returned with error */ - errtype = WebRtcIsac_GetErrorCode(ISAC_main_inst); - printf("\n\n Error in SetMaxPayloadSize: %d.\n\n", errtype); - std::cout << std::flush; - exit(EXIT_FAILURE); - } - } - if (payloadRate != 0) { - err = WebRtcIsac_SetMaxRate(ISAC_main_inst, payloadRate); - if (err < 0) { - /* exit if returned with error */ - errtype = WebRtcIsac_GetErrorCode(ISAC_main_inst); - printf("\n\n Error in SetMaxRateInBytes: %d.\n\n", errtype); - std::cout << std::flush; - exit(EXIT_FAILURE); - } - } - - *speechType = 1; - - std::cout << "\n" << std::flush; - - length_file = 0; - int16_t bnIdxTC = 0; - int16_t jitterInfoTC = 0; - while (endfile == 0) { - /* Call init functions at random, fault test number 7 */ - if (testNum == 7 && (rand() % 2 == 0)) { - err = WebRtcIsac_EncoderInit(ISAC_main_inst, CodingMode); - /* Error check */ - if (err < 0) { - errtype = WebRtcIsac_GetErrorCode(ISAC_main_inst); - printf("\n\n Error in encoderinit: %d.\n\n", errtype); - std::cout << std::flush; - } - - WebRtcIsac_DecoderInit(ISAC_main_inst); - } - - cur_framesmpls = 0; - while (1) { - int stream_len_int = 0; - - /* Read 10 ms speech block */ - endfile = readframe(shortdata, inp, samplesIn10Ms); - - if (endfile) { - numFileLoop++; - if (numFileLoop < totFileLoop) { - rewind(inp); - framecnt = 0; - fprintf(stderr, "\n"); - endfile = readframe(shortdata, inp, samplesIn10Ms); - } - } - - if (testNum == 7) { - srand((unsigned int)time(NULL)); - } - - /* iSAC encoding */ - if (!(testNum == 3 && framecnt == 0)) { - stream_len_int = - WebRtcIsac_Encode(ISAC_main_inst, shortdata, (uint8_t*)streamdata); - if ((payloadSize != 0) && (stream_len_int > payloadSize)) { - if (testNum == 0) { - printf("\n\n"); - } - - printf("\nError: Streamsize out of range %d\n", - stream_len_int - payloadSize); - std::cout << std::flush; - } - - WebRtcIsac_GetUplinkBw(ISAC_main_inst, &sendBN); - - if (stream_len_int > 0) { - if (doTransCoding) { - int16_t indexStream; - uint8_t auxUW8; - - /******************** Main Transcoding stream ********************/ - WebRtcIsac_GetDownLinkBwIndex(ISAC_main_inst, &bnIdxTC, - &jitterInfoTC); - int streamLenTransCoding_int = WebRtcIsac_GetNewBitStream( - ISAC_main_inst, bnIdxTC, jitterInfoTC, rateTransCoding, - streamDataTransCoding, false); - if (streamLenTransCoding_int < 0) { - fprintf(stderr, "Error in trans-coding\n"); - exit(0); - } - streamLenTransCoding = - static_cast(streamLenTransCoding_int); - auxUW8 = (uint8_t)(((streamLenTransCoding & 0xFF00) >> 8) & 0x00FF); - if (fwrite(&auxUW8, sizeof(uint8_t), 1, transcodingBitstream) != - 1) { - return -1; - } - - auxUW8 = (uint8_t)(streamLenTransCoding & 0x00FF); - if (fwrite(&auxUW8, sizeof(uint8_t), 1, transcodingBitstream) != - 1) { - return -1; - } - - if (fwrite(streamDataTransCoding, sizeof(uint8_t), - streamLenTransCoding, - transcodingBitstream) != streamLenTransCoding) { - return -1; - } - - WebRtcIsac_ReadBwIndex(streamDataTransCoding, &indexStream); - if (indexStream != bnIdxTC) { - fprintf(stderr, - "Error in inserting Bandwidth index into transcoding " - "stream.\n"); - exit(0); - } - numTransCodingBytes += streamLenTransCoding; - } - } - } else { - break; - } - - if (stream_len_int < 0) { - /* exit if returned with error */ - errtype = WebRtcIsac_GetErrorCode(ISAC_main_inst); - fprintf(stderr, "Error in encoder: %d.\n", errtype); - std::cout << std::flush; - exit(0); - } - stream_len = static_cast(stream_len_int); - - cur_framesmpls += samplesIn10Ms; - /* exit encoder loop if the encoder returned a bitstream */ - if (stream_len != 0) - break; - } - - /* read next bottleneck rate */ - if (f_bn != NULL) { - if (fscanf(f_bn, "%d", &bottleneck) == EOF) { - /* Set pointer to beginning of file */ - fseek(f_bn, 0L, SEEK_SET); - if (fscanf(f_bn, "%d", &bottleneck) == EOF) { - exit(0); - } - } - if (CodingMode == 1) { - WebRtcIsac_Control(ISAC_main_inst, bottleneck, framesize); - } - } - - length_file += cur_framesmpls; - if (cur_framesmpls == (3 * samplesIn10Ms)) { - maxStreamLen30 = - (stream_len > maxStreamLen30) ? stream_len : maxStreamLen30; - } else { - maxStreamLen60 = - (stream_len > maxStreamLen60) ? stream_len : maxStreamLen60; - } - - if (!lostFrame) { - lostFrame = ((rand() % 100) < packetLossPercent); - } else { - lostFrame = false; - } - - // RED. - if (lostFrame) { - int stream_len_int = WebRtcIsac_GetRedPayload( - ISAC_main_inst, reinterpret_cast(streamdata)); - if (stream_len_int < 0) { - fprintf(stderr, "Error getting RED payload\n"); - exit(0); - } - stream_len = static_cast(stream_len_int); - - if (doTransCoding) { - int streamLenTransCoding_int = WebRtcIsac_GetNewBitStream( - ISAC_main_inst, bnIdxTC, jitterInfoTC, rateTransCoding, - streamDataTransCoding, true); - if (streamLenTransCoding_int < 0) { - fprintf(stderr, "Error in RED trans-coding\n"); - exit(0); - } - streamLenTransCoding = static_cast(streamLenTransCoding_int); - } - } - - /* make coded sequence to short be inreasing */ - /* the length the decoder expects */ - if (testNum == 4) { - stream_len += 10; - } - - /* make coded sequence to long be decreasing */ - /* the length the decoder expects */ - if (testNum == 5) { - stream_len -= 10; - } - - if (testNum == 6) { - srand((unsigned int)time(NULL)); - for (i = 0; i < stream_len; i++) { - streamdata[i] = rand(); - } - } - - if (VADusage) { - readframe(vaddata, vadp, samplesIn10Ms * 3); - } - - /* simulate packet handling through NetEq and the modem */ - if (!(testNum == 3 && framecnt == 0)) { - get_arrival_time(cur_framesmpls, stream_len, bottleneck, &BN_data, - sampFreqKHz * 1000, sampFreqKHz * 1000); - } - - if (VADusage && (framecnt > 10 && vaddata[0] == 0)) { - BN_data.rtp_number--; - } else { - /* Error test number 10, garbage data */ - if (testNum == 10) { - /* Test to run decoder with garbage data */ - for (i = 0; i < stream_len; i++) { - streamdata[i] = (short)(streamdata[i]) + (short)rand(); - } - } - - if (testNum != 9) { - err = WebRtcIsac_UpdateBwEstimate( - ISAC_main_inst, reinterpret_cast(streamdata), - stream_len, BN_data.rtp_number, BN_data.sample_count, - BN_data.arrival_time); - - if (err < 0) { - /* exit if returned with error */ - errtype = WebRtcIsac_GetErrorCode(ISAC_main_inst); - if (testNum == 0) { - printf("\n\n"); - } - - printf("Error: in decoder: %d.", errtype); - std::cout << std::flush; - if (testNum == 0) { - printf("\n\n"); - } - } - } - - /* Call getFramelen, only used here for function test */ - err = WebRtcIsac_ReadFrameLen( - ISAC_main_inst, reinterpret_cast(streamdata), &FL); - if (err < 0) { - /* exit if returned with error */ - errtype = WebRtcIsac_GetErrorCode(ISAC_main_inst); - if (testNum == 0) { - printf("\n\n"); - } - printf(" Error: in getFrameLen %d.", errtype); - std::cout << std::flush; - if (testNum == 0) { - printf("\n\n"); - } - } - - // iSAC decoding - - if (lostFrame) { - declen = WebRtcIsac_DecodeRcu( - ISAC_main_inst, reinterpret_cast(streamdata), - stream_len, decoded, speechType); - - if (doTransCoding) { - declenTC = - WebRtcIsac_DecodeRcu(decoderTransCoding, streamDataTransCoding, - streamLenTransCoding, decodedTC, speechType); - } - } else { - declen = WebRtcIsac_Decode(ISAC_main_inst, - reinterpret_cast(streamdata), - stream_len, decoded, speechType); - if (doTransCoding) { - declenTC = - WebRtcIsac_Decode(decoderTransCoding, streamDataTransCoding, - streamLenTransCoding, decodedTC, speechType); - } - } - - if (declen < 0) { - /* exit if returned with error */ - errtype = WebRtcIsac_GetErrorCode(ISAC_main_inst); - if (testNum == 0) { - printf("\n\n"); - } - printf(" Error: in decoder %d.", errtype); - std::cout << std::flush; - if (testNum == 0) { - printf("\n\n"); - } - } - - if (declenTC < 0) { - if (testNum == 0) { - printf("\n\n"); - } - printf(" Error: in decoding the transcoded stream"); - std::cout << std::flush; - if (testNum == 0) { - printf("\n\n"); - } - } - } - /* Write decoded speech frame to file */ - if ((declen > 0) && (numFileLoop == 0)) { - if (fwrite(decoded, sizeof(int16_t), declen, outp) != - static_cast(declen)) { - return -1; - } - } - - if ((declenTC > 0) && (numFileLoop == 0)) { - if (fwrite(decodedTC, sizeof(int16_t), declen, transCodingFile) != - static_cast(declen)) { - return -1; - } - } - - fprintf(stderr, "\rframe = %5d ", framecnt); - fflush(stderr); - framecnt++; - - /* Error test number 10, garbage data */ - // if (testNum == 10) - // { - // /* Test to run decoder with garbage data */ - // if ((seedfile = fopen(SEED_FILE, "a+t")) == NULL) { - // fprintf(stderr, "Error: Could not open file %s\n", SEED_FILE); - // } else { - // fprintf(seedfile, "ok\n\n"); - // fclose(seedfile); - // } - // } - /* Error test number 10, garbage data */ - // if (testNum == 10) { - // /* Test to run decoder with garbage data */ - // for (i = 0; i < stream_len; i++) { - // streamdata[i] = (short) (streamdata[i] + (short) rand()); - // } - // } - - totalsmpls += declen; - totalbits += 8 * stream_len; -#if !defined(NDEBUG) - kbps = ((double)sampFreqKHz * 1000.) / ((double)cur_framesmpls) * 8.0 * - stream_len / 1000.0; // kbits/s - fy = fopen("bit_rate.dat", "a"); - fprintf(fy, "Frame %i = %0.14f\n", framecnt, kbps); - fclose(fy); - -#endif - } - printf("\n"); - printf("total bits = %zu bits\n", totalbits); - printf("measured average bitrate = %0.3f kbits/s\n", - (double)totalbits * (sampFreqKHz) / totalsmpls); - if (doTransCoding) { - printf("Transcoding average bit-rate = %0.3f kbps\n", - (double)numTransCodingBytes * 8.0 * (sampFreqKHz) / totalsmpls); - fclose(transCodingFile); - } - printf("\n"); - - /* Runtime statistics */ - runtime = (double)(clock() / (double)CLOCKS_PER_SEC - starttime); - length_file = length_file / (sampFreqKHz * 1000.); - - printf("\n\nLength of speech file: %.1f s\n", length_file); - printf("Time to run iSAC: %.2f s (%.2f %% of realtime)\n\n", runtime, - (100 * runtime / length_file)); - - if (maxStreamLen30 != 0) { - printf( - "Maximum payload size 30ms Frames %zu" - " bytes (%0.3f kbps)\n", - maxStreamLen30, maxStreamLen30 * 8 / 30.); - } - if (maxStreamLen60 != 0) { - printf( - "Maximum payload size 60ms Frames %zu" - " bytes (%0.3f kbps)\n", - maxStreamLen60, maxStreamLen60 * 8 / 60.); - } - // fprintf(stderr, "\n"); - - fprintf(stderr, " %.1f s", length_file); - fprintf(stderr, " %0.1f kbps", - (double)totalbits * (sampFreqKHz) / totalsmpls); - if (maxStreamLen30 != 0) { - fprintf(stderr, " plmax-30ms %zu bytes (%0.0f kbps)", maxStreamLen30, - maxStreamLen30 * 8 / 30.); - } - if (maxStreamLen60 != 0) { - fprintf(stderr, " plmax-60ms %zu bytes (%0.0f kbps)", maxStreamLen60, - maxStreamLen60 * 8 / 60.); - } - if (doTransCoding) { - fprintf(stderr, " transcoding rate %.0f kbps", - (double)numTransCodingBytes * 8.0 * (sampFreqKHz) / totalsmpls); - } - - fclose(inp); - fclose(outp); - WebRtcIsac_Free(ISAC_main_inst); - - exit(0); -} diff --git a/modules/audio_coding/codecs/isac/main/test/SwitchingSampRate/SwitchingSampRate.cc b/modules/audio_coding/codecs/isac/main/test/SwitchingSampRate/SwitchingSampRate.cc deleted file mode 100644 index 549163fc44..0000000000 --- a/modules/audio_coding/codecs/isac/main/test/SwitchingSampRate/SwitchingSampRate.cc +++ /dev/null @@ -1,425 +0,0 @@ -/* - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -// SwitchingSampRate.cpp : Defines the entry point for the console -// application. -// - -#include - -#include "common_audio/signal_processing/include/signal_processing_library.h" -#include "modules/audio_coding/codecs/isac/main/include/isac.h" -#include "modules/audio_coding/codecs/isac/main/util/utility.h" - -#define MAX_FILE_NAME 500 -#define MAX_NUM_CLIENTS 2 - -#define NUM_CLIENTS 2 - -int main(int argc, char* argv[]) { - char fileNameWB[MAX_FILE_NAME]; - char fileNameSWB[MAX_FILE_NAME]; - - char outFileName[MAX_NUM_CLIENTS][MAX_FILE_NAME]; - - FILE* inFile[MAX_NUM_CLIENTS]; - FILE* outFile[MAX_NUM_CLIENTS]; - - ISACStruct* codecInstance[MAX_NUM_CLIENTS]; - int32_t resamplerState[MAX_NUM_CLIENTS][8]; - - int encoderSampRate[MAX_NUM_CLIENTS]; - - int minBn = 16000; - int maxBn = 56000; - - int bnWB = 32000; - int bnSWB = 56000; - - strcpy(outFileName[0], "switchSampRate_out1.pcm"); - strcpy(outFileName[1], "switchSampRate_out2.pcm"); - - short clientCntr; - - size_t lenEncodedInBytes[MAX_NUM_CLIENTS]; - unsigned int lenAudioIn10ms[MAX_NUM_CLIENTS]; - size_t lenEncodedInBytesTmp[MAX_NUM_CLIENTS]; - unsigned int lenAudioIn10msTmp[MAX_NUM_CLIENTS]; - BottleNeckModel* packetData[MAX_NUM_CLIENTS]; - - char versionNumber[100]; - short samplesIn10ms[MAX_NUM_CLIENTS]; - int bottleneck[MAX_NUM_CLIENTS]; - - printf("\n\n"); - printf("____________________________________________\n\n"); - WebRtcIsac_version(versionNumber); - printf(" iSAC-swb version %s\n", versionNumber); - printf("____________________________________________\n"); - - fileNameWB[0] = '\0'; - fileNameSWB[0] = '\0'; - - char myFlag[20]; - strcpy(myFlag, "-wb"); - // READ THE WIDEBAND AND SUPER-WIDEBAND FILE NAMES - if (readParamString(argc, argv, myFlag, fileNameWB, MAX_FILE_NAME) <= 0) { - printf("No wideband file is specified"); - } - - strcpy(myFlag, "-swb"); - if (readParamString(argc, argv, myFlag, fileNameSWB, MAX_FILE_NAME) <= 0) { - printf("No super-wideband file is specified"); - } - - // THE FIRST CLIENT STARTS IN WIDEBAND - encoderSampRate[0] = 16000; - OPEN_FILE_RB(inFile[0], fileNameWB); - - // THE SECOND CLIENT STARTS IN SUPER-WIDEBAND - encoderSampRate[1] = 32000; - OPEN_FILE_RB(inFile[1], fileNameSWB); - - strcpy(myFlag, "-I"); - short codingMode = readSwitch(argc, argv, myFlag); - - for (clientCntr = 0; clientCntr < NUM_CLIENTS; clientCntr++) { - codecInstance[clientCntr] = NULL; - - printf("\n"); - printf("Client %d\n", clientCntr + 1); - printf("---------\n"); - printf("Starting %s", (encoderSampRate[clientCntr] == 16000) - ? "wideband" - : "super-wideband"); - - // Open output File Name - OPEN_FILE_WB(outFile[clientCntr], outFileName[clientCntr]); - printf("Output File...................... %s\n", outFileName[clientCntr]); - - samplesIn10ms[clientCntr] = encoderSampRate[clientCntr] * 10; - - if (codingMode == 1) { - bottleneck[clientCntr] = (clientCntr) ? bnSWB : bnWB; - } else { - bottleneck[clientCntr] = (clientCntr) ? minBn : maxBn; - } - - printf("Bottleneck....................... %0.3f kbits/sec \n", - bottleneck[clientCntr] / 1000.0); - - // coding-mode - printf( - "Encoding Mode.................... %s\n", - (codingMode == 1) ? "Channel-Independent (Instantaneous)" : "Adaptive"); - - lenEncodedInBytes[clientCntr] = 0; - lenAudioIn10ms[clientCntr] = 0; - lenEncodedInBytesTmp[clientCntr] = 0; - lenAudioIn10msTmp[clientCntr] = 0; - - packetData[clientCntr] = (BottleNeckModel*)new (BottleNeckModel); - if (packetData[clientCntr] == NULL) { - printf("Could not allocate memory for packetData \n"); - return -1; - } - memset(packetData[clientCntr], 0, sizeof(BottleNeckModel)); - memset(resamplerState[clientCntr], 0, sizeof(int32_t) * 8); - } - - for (clientCntr = 0; clientCntr < NUM_CLIENTS; clientCntr++) { - // Create - if (WebRtcIsac_Create(&codecInstance[clientCntr])) { - printf("Could not creat client %d\n", clientCntr + 1); - return -1; - } - - WebRtcIsac_SetEncSampRate(codecInstance[clientCntr], - encoderSampRate[clientCntr]); - - WebRtcIsac_SetDecSampRate( - codecInstance[clientCntr], - encoderSampRate[clientCntr + (1 - ((clientCntr & 1) << 1))]); - - // Initialize Encoder - if (WebRtcIsac_EncoderInit(codecInstance[clientCntr], codingMode) < 0) { - printf("Could not initialize client, %d\n", clientCntr + 1); - return -1; - } - - WebRtcIsac_DecoderInit(codecInstance[clientCntr]); - - // setup Rate if in Instantaneous mode - if (codingMode != 0) { - // ONLY Clients who are not in Adaptive mode - if (WebRtcIsac_Control(codecInstance[clientCntr], bottleneck[clientCntr], - 30) < 0) { - printf("Could not setup bottleneck and frame-size for client %d\n", - clientCntr + 1); - return -1; - } - } - } - - size_t streamLen; - short numSamplesRead; - size_t lenDecodedAudio; - short senderIdx; - short receiverIdx; - - printf("\n"); - short num10ms[MAX_NUM_CLIENTS]; - memset(num10ms, 0, sizeof(short) * MAX_NUM_CLIENTS); - FILE* arrivalTimeFile1 = fopen("arrivalTime1.dat", "wb"); - FILE* arrivalTimeFile2 = fopen("arrivalTime2.dat", "wb"); - short numPrint[MAX_NUM_CLIENTS]; - memset(numPrint, 0, sizeof(short) * MAX_NUM_CLIENTS); - - // Audio Buffers - short silence10ms[10 * 32]; - memset(silence10ms, 0, 320 * sizeof(short)); - short audioBuff10ms[10 * 32]; - short audioBuff60ms[60 * 32]; - short resampledAudio60ms[60 * 32]; - - unsigned short bitStream[600 + 600]; - short speechType[1]; - - short numSampFreqChanged = 0; - while (numSampFreqChanged < 10) { - for (clientCntr = 0; clientCntr < NUM_CLIENTS; clientCntr++) { - // Encoding/decoding for this pair of clients, if there is - // audio for any of them - // if(audioLeft[clientCntr] || audioLeft[clientCntr + 1]) - //{ - // for(pairCntr = 0; pairCntr < 2; pairCntr++) - //{ - senderIdx = clientCntr; // + pairCntr; - receiverIdx = 1 - clientCntr; // + (1 - pairCntr); - - // if(num10ms[senderIdx] > 6) - //{ - // printf("Too many frames read for client %d", - // senderIdx + 1); - // return -1; - //} - - numSamplesRead = - (short)fread(audioBuff10ms, sizeof(short), samplesIn10ms[senderIdx], - inFile[senderIdx]); - if (numSamplesRead != samplesIn10ms[senderIdx]) { - // file finished switch encoder sampling frequency. - printf("Changing Encoder Sampling frequency in client %d to ", - senderIdx + 1); - fclose(inFile[senderIdx]); - numSampFreqChanged++; - if (encoderSampRate[senderIdx] == 16000) { - printf("super-wideband.\n"); - OPEN_FILE_RB(inFile[senderIdx], fileNameSWB); - encoderSampRate[senderIdx] = 32000; - } else { - printf("wideband.\n"); - OPEN_FILE_RB(inFile[senderIdx], fileNameWB); - encoderSampRate[senderIdx] = 16000; - } - WebRtcIsac_SetEncSampRate(codecInstance[senderIdx], - encoderSampRate[senderIdx]); - WebRtcIsac_SetDecSampRate(codecInstance[receiverIdx], - encoderSampRate[senderIdx]); - - samplesIn10ms[clientCntr] = encoderSampRate[clientCntr] * 10; - - numSamplesRead = - (short)fread(audioBuff10ms, sizeof(short), samplesIn10ms[senderIdx], - inFile[senderIdx]); - if (numSamplesRead != samplesIn10ms[senderIdx]) { - printf(" File %s for client %d has not enough audio\n", - (encoderSampRate[senderIdx] == 16000) ? "wideband" - : "super-wideband", - senderIdx + 1); - return -1; - } - } - num10ms[senderIdx]++; - - // sanity check - // if(num10ms[senderIdx] > 6) - //{ - // printf("Client %d has got more than 60 ms audio and encoded no - // packet.\n", - // senderIdx); - // return -1; - //} - - // Encode - - int streamLen_int = WebRtcIsac_Encode(codecInstance[senderIdx], - audioBuff10ms, (uint8_t*)bitStream); - int16_t ggg; - if (streamLen_int > 0) { - if ((WebRtcIsac_ReadFrameLen( - codecInstance[receiverIdx], - reinterpret_cast(bitStream), &ggg)) < 0) - printf("ERROR\n"); - } - - // Sanity check - if (streamLen_int < 0) { - printf(" Encoder error in client %d \n", senderIdx + 1); - return -1; - } - streamLen = static_cast(streamLen_int); - - if (streamLen > 0) { - // Packet generated; model sending through a channel, do bandwidth - // estimation at the receiver and decode. - lenEncodedInBytes[senderIdx] += streamLen; - lenAudioIn10ms[senderIdx] += (unsigned int)num10ms[senderIdx]; - lenEncodedInBytesTmp[senderIdx] += streamLen; - lenAudioIn10msTmp[senderIdx] += (unsigned int)num10ms[senderIdx]; - - // Print after ~5 sec. - if (lenAudioIn10msTmp[senderIdx] >= 100) { - numPrint[senderIdx]++; - printf(" %d, %6.3f => %6.3f ", senderIdx + 1, - bottleneck[senderIdx] / 1000.0, - lenEncodedInBytesTmp[senderIdx] * 0.8 / - lenAudioIn10msTmp[senderIdx]); - - if (codingMode == 0) { - int32_t bn; - WebRtcIsac_GetUplinkBw(codecInstance[senderIdx], &bn); - printf("[%d] ", bn); - } - // int16_t rateIndexLB; - // int16_t rateIndexUB; - // WebRtcIsac_GetDownLinkBwIndex(codecInstance[receiverIdx], - // &rateIndexLB, &rateIndexUB); - // printf(" (%2d, %2d) ", rateIndexLB, rateIndexUB); - - std::cout << std::flush; - lenEncodedInBytesTmp[senderIdx] = 0; - lenAudioIn10msTmp[senderIdx] = 0; - // if(senderIdx == (NUM_CLIENTS - 1)) - //{ - printf(" %0.1f \n", lenAudioIn10ms[senderIdx] * 10. / 1000); - //} - - // After ~20 sec change the bottleneck. - // if((numPrint[senderIdx] == 4) && (codingMode == 0)) - // { - // numPrint[senderIdx] = 0; - // if(codingMode == 0) - // { - // int newBottleneck = bottleneck[senderIdx] + - // (bottleneckChange[senderIdx] * 1000); - - // if(bottleneckChange[senderIdx] > 0) - // { - // if(newBottleneck >maxBn) - // { - // bottleneckChange[senderIdx] = -1; - // newBottleneck = bottleneck[senderIdx] + - // (bottleneckChange[senderIdx] * 1000); - // if(newBottleneck > minBn) - // { - // bottleneck[senderIdx] = newBottleneck; - // } - // } - // else - // { - // bottleneck[senderIdx] = newBottleneck; - // } - // } - // else - // { - // if(newBottleneck < minBn) - // { - // bottleneckChange[senderIdx] = 1; - // newBottleneck = bottleneck[senderIdx] + - // (bottleneckChange[senderIdx] * 1000); - // if(newBottleneck < maxBn) - // { - // bottleneck[senderIdx] = newBottleneck; - // } - // } - // else - // { - // bottleneck[senderIdx] = newBottleneck; - // } - // } - // } - // } - } - - // model a channel of given bottleneck, to get the receive timestamp - get_arrival_time(num10ms[senderIdx] * samplesIn10ms[senderIdx], - streamLen, bottleneck[senderIdx], - packetData[senderIdx], - encoderSampRate[senderIdx] * 1000, - encoderSampRate[senderIdx] * 1000); - - // Write the arrival time. - if (senderIdx == 0) { - if (fwrite(&(packetData[senderIdx]->arrival_time), - sizeof(unsigned int), 1, arrivalTimeFile1) != 1) { - return -1; - } - } else { - if (fwrite(&(packetData[senderIdx]->arrival_time), - sizeof(unsigned int), 1, arrivalTimeFile2) != 1) { - return -1; - } - } - - // BWE - if (WebRtcIsac_UpdateBwEstimate( - codecInstance[receiverIdx], - reinterpret_cast(bitStream), streamLen, - packetData[senderIdx]->rtp_number, - packetData[senderIdx]->sample_count, - packetData[senderIdx]->arrival_time) < 0) { - printf(" BWE Error at client %d \n", receiverIdx + 1); - return -1; - } - /**/ - // Decode - int lenDecodedAudio_int = - WebRtcIsac_Decode(codecInstance[receiverIdx], - reinterpret_cast(bitStream), - streamLen, audioBuff60ms, speechType); - if (lenDecodedAudio_int < 0) { - printf(" Decoder error in client %d \n", receiverIdx + 1); - return -1; - } - lenDecodedAudio = static_cast(lenDecodedAudio_int); - - if (encoderSampRate[senderIdx] == 16000) { - WebRtcSpl_UpsampleBy2(audioBuff60ms, lenDecodedAudio, - resampledAudio60ms, - resamplerState[receiverIdx]); - if (fwrite(resampledAudio60ms, sizeof(short), lenDecodedAudio << 1, - outFile[receiverIdx]) != lenDecodedAudio << 1) { - return -1; - } - } else { - if (fwrite(audioBuff60ms, sizeof(short), lenDecodedAudio, - outFile[receiverIdx]) != lenDecodedAudio) { - return -1; - } - } - num10ms[senderIdx] = 0; - } - //} - //} - } - } - return 0; -} diff --git a/modules/audio_coding/codecs/isac/main/test/simpleKenny.c b/modules/audio_coding/codecs/isac/main/test/simpleKenny.c deleted file mode 100644 index 4446ff7806..0000000000 --- a/modules/audio_coding/codecs/isac/main/test/simpleKenny.c +++ /dev/null @@ -1,461 +0,0 @@ -/* - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -/* kenny.c - Main function for the iSAC coder */ - -#include -#include -#include -#include - -#ifdef WIN32 -#include "windows.h" -#ifndef CLOCKS_PER_SEC -#define CLOCKS_PER_SEC 1000 -#endif -#endif - -#include - -/* include API */ -#include "modules/audio_coding/codecs/isac/main/include/isac.h" -#include "modules/audio_coding/codecs/isac/main/util/utility.h" - -/* max number of samples per frame (= 60 ms frame) */ -#define MAX_FRAMESAMPLES_SWB 1920 -/* number of samples per 10ms frame */ -#define FRAMESAMPLES_SWB_10ms 320 -#define FRAMESAMPLES_WB_10ms 160 - -/* sampling frequency (Hz) */ -#define FS_SWB 32000 -#define FS_WB 16000 - -unsigned long framecnt = 0; - -int main(int argc, char* argv[]) { - //--- File IO ---- - FILE* inp; - FILE* outp; - char inname[500]; - char outname[500]; - - /* Runtime statistics */ - double rate; - double rateRCU; - size_t totalbits = 0; - unsigned long totalBitsRCU = 0; - unsigned long totalsmpls = 0; - - int32_t bottleneck = 39; - int frameSize = 30; /* ms */ - int16_t codingMode = 1; - int16_t shortdata[FRAMESAMPLES_SWB_10ms]; - int16_t decoded[MAX_FRAMESAMPLES_SWB]; - int16_t speechType[1]; - int16_t payloadLimit; - int32_t rateLimit; - ISACStruct* ISAC_main_inst; - - size_t stream_len = 0; - int declen = 0; - int16_t err; - int cur_framesmpls; - int endfile; -#ifdef WIN32 - double length_file; - double runtime; - char outDrive[10]; - char outPath[500]; - char outPrefix[500]; - char outSuffix[500]; - char bitrateFileName[500]; - FILE* bitrateFile; - double starttime; - double rateLB = 0; - double rateUB = 0; -#endif - FILE* histFile; - FILE* averageFile; - int sampFreqKHz; - int samplesIn10Ms; - size_t maxStreamLen = 0; - char histFileName[500]; - char averageFileName[500]; - unsigned int hist[600]; - double tmpSumStreamLen = 0; - unsigned int packetCntr = 0; - unsigned int lostPacketCntr = 0; - uint8_t payload[1200]; - uint8_t payloadRCU[1200]; - uint16_t packetLossPercent = 0; - int16_t rcuStreamLen = 0; - int onlyEncode; - int onlyDecode; - - BottleNeckModel packetData; - packetData.arrival_time = 0; - packetData.sample_count = 0; - packetData.rtp_number = 0; - memset(hist, 0, sizeof(hist)); - - /* handling wrong input arguments in the command line */ - if (argc < 5) { - printf("\n\nWrong number of arguments or flag values.\n\n"); - - printf("Usage:\n\n"); - printf("%s infile outfile -bn bottleneck [options]\n\n", argv[0]); - printf("with:\n"); - printf("-I.............. indicates encoding in instantaneous mode.\n"); - printf("-bn bottleneck.. the value of the bottleneck in bit/sec, e.g.\n"); - printf(" 39742, in instantaneous (channel-independent)\n"); - printf(" mode.\n\n"); - printf("infile.......... Normal speech input file\n\n"); - printf("outfile......... Speech output file\n\n"); - printf("OPTIONS\n"); - printf("-------\n"); - printf("-fs sampFreq.... sampling frequency of codec 16 or 32 (default)\n"); - printf(" kHz.\n"); - printf("-plim payloadLim payload limit in bytes, default is the maximum\n"); - printf(" possible.\n"); - printf("-rlim rateLim... rate limit in bits/sec, default is the maximum\n"); - printf(" possible.\n"); - printf("-h file......... record histogram and *append* to 'file'.\n"); - printf("-ave file....... record average rate of 3 sec intervales and\n"); - printf(" *append* to 'file'.\n"); - printf("-ploss.......... packet-loss percentage.\n"); - printf("-enc............ do only encoding and store the bit-stream\n"); - printf("-dec............ the input file is a bit-stream, decode it.\n\n"); - printf("Example usage:\n\n"); - printf("%s speechIn.pcm speechOut.pcm -B 40000 -fs 32\n\n", argv[0]); - - exit(0); - } - - /* Get Bottleneck value */ - bottleneck = readParamInt(argc, argv, "-bn", 50000); - fprintf(stderr, "\nfixed bottleneck rate of %d bits/s\n\n", bottleneck); - - /* Get Input and Output files */ - sscanf(argv[1], "%s", inname); - sscanf(argv[2], "%s", outname); - codingMode = readSwitch(argc, argv, "-I"); - sampFreqKHz = (int16_t)readParamInt(argc, argv, "-fs", 32); - if (readParamString(argc, argv, "-h", histFileName, 500) > 0) { - histFile = fopen(histFileName, "a"); - if (histFile == NULL) { - printf("cannot open hist file %s", histFileName); - exit(0); - } - } else { - // NO recording of hitstogram - histFile = NULL; - } - - packetLossPercent = readParamInt(argc, argv, "-ploss", 0); - - if (readParamString(argc, argv, "-ave", averageFileName, 500) > 0) { - averageFile = fopen(averageFileName, "a"); - if (averageFile == NULL) { - printf("cannot open file to write rate %s", averageFileName); - exit(0); - } - } else { - averageFile = NULL; - } - - onlyEncode = readSwitch(argc, argv, "-enc"); - onlyDecode = readSwitch(argc, argv, "-dec"); - - switch (sampFreqKHz) { - case 16: { - samplesIn10Ms = 160; - break; - } - case 32: { - samplesIn10Ms = 320; - break; - } - default: - printf("A sampling frequency of %d kHz is not supported, valid values are" - " 8 and 16.\n", sampFreqKHz); - exit(-1); - } - payloadLimit = (int16_t)readParamInt(argc, argv, "-plim", 400); - rateLimit = readParamInt(argc, argv, "-rlim", 106800); - - if ((inp = fopen(inname, "rb")) == NULL) { - printf(" iSAC: Cannot read file %s.\n", inname); - exit(1); - } - if ((outp = fopen(outname, "wb")) == NULL) { - printf(" iSAC: Cannot write file %s.\n", outname); - exit(1); - } - -#ifdef WIN32 - _splitpath(outname, outDrive, outPath, outPrefix, outSuffix); - _makepath(bitrateFileName, outDrive, outPath, "bitrate", ".txt"); - - bitrateFile = fopen(bitrateFileName, "a"); - fprintf(bitrateFile, "%% %s \n", inname); -#endif - - printf("\n"); - printf("Input.................... %s\n", inname); - printf("Output................... %s\n", outname); - printf("Encoding Mode............ %s\n", - (codingMode == 1) ? "Channel-Independent" : "Channel-Adaptive"); - printf("Bottleneck............... %d bits/sec\n", bottleneck); - printf("Packet-loss Percentage... %d\n", packetLossPercent); - printf("\n"); - -#ifdef WIN32 - starttime = clock() / (double)CLOCKS_PER_SEC; /* Runtime statistics */ -#endif - - /* Initialize the ISAC and BN structs */ - err = WebRtcIsac_Create(&ISAC_main_inst); - - WebRtcIsac_SetEncSampRate(ISAC_main_inst, sampFreqKHz * 1000); - WebRtcIsac_SetDecSampRate(ISAC_main_inst, sampFreqKHz >= 32 ? 32000 : 16000); - /* Error check */ - if (err < 0) { - fprintf(stderr, "\n\n Error in create.\n\n"); - exit(EXIT_FAILURE); - } - - framecnt = 0; - endfile = 0; - - /* Initialize encoder and decoder */ - if (WebRtcIsac_EncoderInit(ISAC_main_inst, codingMode) < 0) { - printf("cannot initialize encoder\n"); - return -1; - } - WebRtcIsac_DecoderInit(ISAC_main_inst); - - if (codingMode == 1) { - if (WebRtcIsac_Control(ISAC_main_inst, bottleneck, frameSize) < 0) { - printf("cannot set bottleneck\n"); - return -1; - } - } else { - if (WebRtcIsac_ControlBwe(ISAC_main_inst, 15000, 30, 1) < 0) { - printf("cannot configure BWE\n"); - return -1; - } - } - - if (WebRtcIsac_SetMaxPayloadSize(ISAC_main_inst, payloadLimit) < 0) { - printf("cannot set maximum payload size %d.\n", payloadLimit); - return -1; - } - - if (rateLimit < 106800) { - if (WebRtcIsac_SetMaxRate(ISAC_main_inst, rateLimit) < 0) { - printf("cannot set the maximum rate %d.\n", rateLimit); - return -1; - } - } - - while (endfile == 0) { - fprintf(stderr, " \rframe = %7li", framecnt); - - //============== Readind from the file and encoding ================= - cur_framesmpls = 0; - stream_len = 0; - - if (onlyDecode) { - uint8_t auxUW8; - if (fread(&auxUW8, sizeof(uint8_t), 1, inp) < 1) { - break; - } - stream_len = auxUW8 << 8; - if (fread(&auxUW8, sizeof(uint8_t), 1, inp) < 1) { - break; - } - stream_len |= auxUW8; - if (fread(payload, 1, stream_len, inp) < stream_len) { - printf("last payload is corrupted\n"); - break; - } - } else { - while (stream_len == 0) { - int stream_len_int; - - // Read 10 ms speech block - endfile = readframe(shortdata, inp, samplesIn10Ms); - if (endfile) { - break; - } - cur_framesmpls += samplesIn10Ms; - - //-------- iSAC encoding --------- - stream_len_int = WebRtcIsac_Encode(ISAC_main_inst, shortdata, payload); - - if (stream_len_int < 0) { - // exit if returned with error - // errType=WebRtcIsac_GetErrorCode(ISAC_main_inst); - fprintf(stderr, "\nError in encoder\n"); - getc(stdin); - exit(EXIT_FAILURE); - } - stream_len = (size_t)stream_len_int; - } - //=================================================================== - if (endfile) { - break; - } - - rcuStreamLen = WebRtcIsac_GetRedPayload(ISAC_main_inst, payloadRCU); - if (rcuStreamLen < 0) { - fprintf(stderr, "\nError getting RED payload\n"); - getc(stdin); - exit(EXIT_FAILURE); - } - - get_arrival_time(cur_framesmpls, stream_len, bottleneck, &packetData, - sampFreqKHz * 1000, sampFreqKHz * 1000); - if (WebRtcIsac_UpdateBwEstimate( - ISAC_main_inst, payload, stream_len, packetData.rtp_number, - packetData.sample_count, packetData.arrival_time) < 0) { - printf(" BWE Error at client\n"); - return -1; - } - } - - if (endfile) { - break; - } - - maxStreamLen = (stream_len > maxStreamLen) ? stream_len : maxStreamLen; - packetCntr++; - - hist[stream_len]++; - if (averageFile != NULL) { - tmpSumStreamLen += stream_len; - if (packetCntr == 100) { - // kbps - fprintf(averageFile, "%8.3f ", - tmpSumStreamLen * 8.0 / (30.0 * packetCntr)); - packetCntr = 0; - tmpSumStreamLen = 0; - } - } - - if (onlyEncode) { - uint8_t auxUW8; - auxUW8 = (uint8_t)(((stream_len & 0x7F00) >> 8) & 0xFF); - if (fwrite(&auxUW8, sizeof(uint8_t), 1, outp) != 1) { - return -1; - } - - auxUW8 = (uint8_t)(stream_len & 0xFF); - if (fwrite(&auxUW8, sizeof(uint8_t), 1, outp) != 1) { - return -1; - } - if (fwrite(payload, 1, stream_len, outp) != stream_len) { - return -1; - } - } else { - //======================= iSAC decoding =========================== - - if ((rand() % 100) < packetLossPercent) { - declen = WebRtcIsac_DecodeRcu(ISAC_main_inst, payloadRCU, - (size_t)rcuStreamLen, decoded, - speechType); - lostPacketCntr++; - } else { - declen = WebRtcIsac_Decode(ISAC_main_inst, payload, stream_len, decoded, - speechType); - } - if (declen <= 0) { - // errType=WebRtcIsac_GetErrorCode(ISAC_main_inst); - fprintf(stderr, "\nError in decoder.\n"); - getc(stdin); - exit(1); - } - - // Write decoded speech frame to file - if (fwrite(decoded, sizeof(int16_t), declen, outp) != (size_t)declen) { - return -1; - } - cur_framesmpls = declen; - } - // Update Statistics - framecnt++; - totalsmpls += cur_framesmpls; - if (stream_len > 0) { - totalbits += 8 * stream_len; - } - if (rcuStreamLen > 0) { - totalBitsRCU += 8 * rcuStreamLen; - } - } - - rate = ((double)totalbits * (sampFreqKHz)) / (double)totalsmpls; - rateRCU = ((double)totalBitsRCU * (sampFreqKHz)) / (double)totalsmpls; - - printf("\n\n"); - printf("Sampling Rate............... %d kHz\n", sampFreqKHz); - printf("Payload Limit............... %d bytes \n", payloadLimit); - printf("Rate Limit.................. %d bits/sec \n", rateLimit); - -#ifdef WIN32 - fprintf(bitrateFile, "%d %10lu %d %6.3f %6.3f %6.3f\n", - sampFreqKHz, framecnt, bottleneck, rateLB, rateUB, rate); - fclose(bitrateFile); -#endif // WIN32 - - printf("\n"); - printf("Measured bit-rate........... %0.3f kbps\n", rate); - printf("Measured RCU bit-ratre...... %0.3f kbps\n", rateRCU); - printf("Maximum bit-rate/payloadsize %0.3f / %zu\n", - maxStreamLen * 8 / 0.03, maxStreamLen); - printf("Measured packet-loss........ %0.1f%% \n", - 100.0f * (float)lostPacketCntr / (float)packetCntr); - - printf("\n"); - -/* Runtime statistics */ -#ifdef WIN32 - runtime = (double)(clock() / (double)CLOCKS_PER_SEC - starttime); - length_file = ((double)framecnt * (double)declen / (sampFreqKHz * 1000)); - printf("Length of speech file....... %.1f s\n", length_file); - printf("Time to run iSAC............ %.2f s (%.2f %% of realtime)\n\n", - runtime, (100 * runtime / length_file)); -#endif - printf("\n\n_______________________________________________\n"); - - if (histFile != NULL) { - int n; - for (n = 0; n < 600; n++) { - fprintf(histFile, "%6d ", hist[n]); - } - fprintf(histFile, "\n"); - fclose(histFile); - } - if (averageFile != NULL) { - if (packetCntr > 0) { - fprintf(averageFile, "%8.3f ", - tmpSumStreamLen * 8.0 / (30.0 * packetCntr)); - } - fprintf(averageFile, "\n"); - fclose(averageFile); - } - - fclose(inp); - fclose(outp); - - WebRtcIsac_Free(ISAC_main_inst); - - exit(0); -} diff --git a/modules/audio_coding/codecs/isac/main/util/utility.c b/modules/audio_coding/codecs/isac/main/util/utility.c deleted file mode 100644 index 56547b11c2..0000000000 --- a/modules/audio_coding/codecs/isac/main/util/utility.c +++ /dev/null @@ -1,179 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#include -#include -#include -#include - -#include "modules/audio_coding/codecs/isac/main/util/utility.h" - -/* function for reading audio data from PCM file */ -int -readframe( - short* data, - FILE* inp, - int length) -{ - short k, rlen, status = 0; - unsigned char* ptrUChar; - ptrUChar = (unsigned char*)data; - - rlen = (short)fread(data, sizeof(short), length, inp); - if (rlen < length) { - for (k = rlen; k < length; k++) - data[k] = 0; - status = 1; - } - - // Assuming that our PCM files are written in Intel machines - for(k = 0; k < length; k++) - { - data[k] = (short)ptrUChar[k<<1] | ((((short)ptrUChar[(k<<1) + 1]) << 8) & 0xFF00); - } - - return status; -} - -short -readSwitch( - int argc, - char* argv[], - char* strID) -{ - short n; - for(n = 0; n < argc; n++) - { - if(strcmp(argv[n], strID) == 0) - { - return 1; - } - } - return 0; -} - -double -readParamDouble( - int argc, - char* argv[], - char* strID, - double defaultVal) -{ - double returnVal = defaultVal; - short n; - for(n = 0; n < argc; n++) - { - if(strcmp(argv[n], strID) == 0) - { - n++; - if(n < argc) - { - returnVal = atof(argv[n]); - } - break; - } - } - return returnVal; -} - -int -readParamInt( - int argc, - char* argv[], - char* strID, - int defaultVal) -{ - int returnVal = defaultVal; - short n; - for(n = 0; n < argc; n++) - { - if(strcmp(argv[n], strID) == 0) - { - n++; - if(n < argc) - { - returnVal = atoi(argv[n]); - } - break; - } - } - return returnVal; -} - -int -readParamString( - int argc, - char* argv[], - char* strID, - char* stringParam, - int maxSize) -{ - int paramLenght = 0; - short n; - for(n = 0; n < argc; n++) - { - if(strcmp(argv[n], strID) == 0) - { - n++; - if(n < argc) - { - strncpy(stringParam, argv[n], maxSize); - paramLenght = (int)strlen(argv[n]); - } - break; - } - } - return paramLenght; -} - -void -get_arrival_time( - int current_framesamples, /* samples */ - size_t packet_size, /* bytes */ - int bottleneck, /* excluding headers; bits/s */ - BottleNeckModel* BN_data, - short senderSampFreqHz, - short receiverSampFreqHz) -{ - unsigned int travelTimeMs; - const int headerSizeByte = 35; - - int headerRate; - - BN_data->whenPackGeneratedMs += (current_framesamples / (senderSampFreqHz / 1000)); - - headerRate = headerSizeByte * 8 * senderSampFreqHz / current_framesamples; /* bits/s */ - - /* everything in samples */ - BN_data->sample_count = BN_data->sample_count + current_framesamples; - - //travelTimeMs = ((packet_size + HeaderSize) * 8 * sampFreqHz) / - // (bottleneck + HeaderRate) - travelTimeMs = (unsigned int)floor((double)((packet_size + headerSizeByte) * 8 * 1000) - / (double)(bottleneck + headerRate) + 0.5); - - if(BN_data->whenPrevPackLeftMs > BN_data->whenPackGeneratedMs) - { - BN_data->whenPrevPackLeftMs += travelTimeMs; - } - else - { - BN_data->whenPrevPackLeftMs = BN_data->whenPackGeneratedMs + - travelTimeMs; - } - - BN_data->arrival_time = (BN_data->whenPrevPackLeftMs * - (receiverSampFreqHz / 1000)); - -// if (BN_data->arrival_time < BN_data->sample_count) -// BN_data->arrival_time = BN_data->sample_count; - - BN_data->rtp_number++; -} diff --git a/modules/audio_coding/codecs/isac/main/util/utility.h b/modules/audio_coding/codecs/isac/main/util/utility.h deleted file mode 100644 index 1acc54251b..0000000000 --- a/modules/audio_coding/codecs/isac/main/util/utility.h +++ /dev/null @@ -1,108 +0,0 @@ -/* - * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_UTIL_UTILITY_H_ -#define MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_UTIL_UTILITY_H_ - -#include -#include - -#if defined(__cplusplus) -extern "C" { -#endif - -#define OPEN_FILE_WB(filePtr, fullPath) \ - do { \ - if (fullPath != NULL) { \ - filePtr = fopen(fullPath, "wb"); \ - if (filePtr == NULL) { \ - printf("could not open %s to write to.", fullPath); \ - return -1; \ - } \ - } else { \ - filePtr = NULL; \ - } \ - } while (0) - -#define OPEN_FILE_AB(filePtr, fullPath) \ - do { \ - if (fullPath != NULL) { \ - filePtr = fopen(fullPath, "ab"); \ - if (filePtr == NULL) { \ - printf("could not open %s to write to.", fullPath); \ - return -1; \ - } \ - } else { \ - filePtr = NULL; \ - } \ - } while (0) - -#define OPEN_FILE_RB(filePtr, fullPath) \ - do { \ - if (fullPath != NULL) { \ - filePtr = fopen(fullPath, "rb"); \ - if (filePtr == NULL) { \ - printf("could not open %s to read from.", fullPath); \ - return -1; \ - } \ - } else { \ - filePtr = NULL; \ - } \ - } while (0) - -#define WRITE_FILE_D(bufferPtr, len, filePtr) \ - do { \ - if (filePtr != NULL) { \ - double dummy[1000]; \ - int cntr; \ - for (cntr = 0; cntr < (len); cntr++) { \ - dummy[cntr] = (double)bufferPtr[cntr]; \ - } \ - fwrite(dummy, sizeof(double), len, filePtr); \ - fflush(filePtr); \ - } \ - } while (0) - -typedef struct { - unsigned int whenPackGeneratedMs; - unsigned int whenPrevPackLeftMs; - unsigned int sendTimeMs; /* milisecond */ - unsigned int arrival_time; /* samples */ - unsigned int sample_count; /* samples, also used as "send time stamp" */ - unsigned int rtp_number; -} BottleNeckModel; - -void get_arrival_time(int current_framesamples, /* samples */ - size_t packet_size, /* bytes */ - int bottleneck, /* excluding headers; bits/s */ - BottleNeckModel* BN_data, - short senderSampFreqHz, - short receiverSampFreqHz); - -/* function for reading audio data from PCM file */ -int readframe(short* data, FILE* inp, int length); - -short readSwitch(int argc, char* argv[], char* strID); - -double readParamDouble(int argc, char* argv[], char* strID, double defaultVal); - -int readParamInt(int argc, char* argv[], char* strID, int defaultVal); - -int readParamString(int argc, - char* argv[], - char* strID, - char* stringParam, - int maxSize); - -#if defined(__cplusplus) -} -#endif - -#endif diff --git a/modules/audio_coding/neteq/audio_decoder_unittest.cc b/modules/audio_coding/neteq/audio_decoder_unittest.cc index bb5c6d167b..fef3c3c1e4 100644 --- a/modules/audio_coding/neteq/audio_decoder_unittest.cc +++ b/modules/audio_coding/neteq/audio_decoder_unittest.cc @@ -22,10 +22,6 @@ #include "modules/audio_coding/codecs/g722/audio_encoder_g722.h" #include "modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.h" #include "modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h" -#include "modules/audio_coding/codecs/isac/fix/include/audio_decoder_isacfix.h" -#include "modules/audio_coding/codecs/isac/fix/include/audio_encoder_isacfix.h" -#include "modules/audio_coding/codecs/isac/main/include/audio_decoder_isac.h" -#include "modules/audio_coding/codecs/isac/main/include/audio_encoder_isac.h" #include "modules/audio_coding/codecs/opus/audio_decoder_opus.h" #include "modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.h" #include "modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h" @@ -195,8 +191,8 @@ class AudioDecoderTest : public ::testing::Test { processed_samples += frame_size_; } // For some codecs it doesn't make sense to check expected number of bytes, - // since the number can vary for different platforms. Opus and iSAC are - // such codecs. In this case expected_bytes is set to 0. + // since the number can vary for different platforms. Opus is such a codec. + // In this case expected_bytes is set to 0. if (expected_bytes) { EXPECT_EQ(expected_bytes, encoded_bytes); } @@ -347,66 +343,6 @@ class AudioDecoderIlbcTest : public AudioDecoderTest { } }; -class AudioDecoderIsacFloatTest : public AudioDecoderTest { - protected: - AudioDecoderIsacFloatTest() : AudioDecoderTest() { - codec_input_rate_hz_ = 16000; - frame_size_ = 480; - data_length_ = 10 * frame_size_; - AudioEncoderIsacFloatImpl::Config config; - config.payload_type = payload_type_; - config.sample_rate_hz = codec_input_rate_hz_; - config.frame_size_ms = - 1000 * static_cast(frame_size_) / codec_input_rate_hz_; - audio_encoder_.reset(new AudioEncoderIsacFloatImpl(config)); - audio_encoder_->OnReceivedOverhead(kOverheadBytesPerPacket); - - AudioDecoderIsacFloatImpl::Config decoder_config; - decoder_config.sample_rate_hz = codec_input_rate_hz_; - decoder_ = new AudioDecoderIsacFloatImpl(decoder_config); - } -}; - -class AudioDecoderIsacSwbTest : public AudioDecoderTest { - protected: - AudioDecoderIsacSwbTest() : AudioDecoderTest() { - codec_input_rate_hz_ = 32000; - frame_size_ = 960; - data_length_ = 10 * frame_size_; - AudioEncoderIsacFloatImpl::Config config; - config.payload_type = payload_type_; - config.sample_rate_hz = codec_input_rate_hz_; - config.frame_size_ms = - 1000 * static_cast(frame_size_) / codec_input_rate_hz_; - audio_encoder_.reset(new AudioEncoderIsacFloatImpl(config)); - audio_encoder_->OnReceivedOverhead(kOverheadBytesPerPacket); - - AudioDecoderIsacFloatImpl::Config decoder_config; - decoder_config.sample_rate_hz = codec_input_rate_hz_; - decoder_ = new AudioDecoderIsacFloatImpl(decoder_config); - } -}; - -class AudioDecoderIsacFixTest : public AudioDecoderTest { - protected: - AudioDecoderIsacFixTest() : AudioDecoderTest() { - codec_input_rate_hz_ = 16000; - frame_size_ = 480; - data_length_ = 10 * frame_size_; - AudioEncoderIsacFixImpl::Config config; - config.payload_type = payload_type_; - config.sample_rate_hz = codec_input_rate_hz_; - config.frame_size_ms = - 1000 * static_cast(frame_size_) / codec_input_rate_hz_; - audio_encoder_.reset(new AudioEncoderIsacFixImpl(config)); - audio_encoder_->OnReceivedOverhead(kOverheadBytesPerPacket); - - AudioDecoderIsacFixImpl::Config decoder_config; - decoder_config.sample_rate_hz = codec_input_rate_hz_; - decoder_ = new AudioDecoderIsacFixImpl(decoder_config); - } -}; - class AudioDecoderG722Test : public AudioDecoderTest { protected: AudioDecoderG722Test() : AudioDecoderTest() { @@ -533,94 +469,6 @@ TEST_F(AudioDecoderIlbcTest, SetTargetBitrate) { TestSetAndGetTargetBitratesWithFixedCodec(audio_encoder_.get(), 13333); } -TEST_F(AudioDecoderIsacFloatTest, EncodeDecode) { - int tolerance = 3399; - double mse = 434951.0; - int delay = 48; // Delay from input to output. - EncodeDecodeTest(0, tolerance, mse, delay); - ReInitTest(); - EXPECT_FALSE(decoder_->HasDecodePlc()); -} - -TEST_F(AudioDecoderIsacFloatTest, SetTargetBitrate) { - const int overhead_rate = - 8 * kOverheadBytesPerPacket * codec_input_rate_hz_ / frame_size_; - EXPECT_EQ(10000, - SetAndGetTargetBitrate(audio_encoder_.get(), 9999 + overhead_rate)); - EXPECT_EQ(10000, SetAndGetTargetBitrate(audio_encoder_.get(), - 10000 + overhead_rate)); - EXPECT_EQ(23456, SetAndGetTargetBitrate(audio_encoder_.get(), - 23456 + overhead_rate)); - EXPECT_EQ(32000, SetAndGetTargetBitrate(audio_encoder_.get(), - 32000 + overhead_rate)); - EXPECT_EQ(32000, SetAndGetTargetBitrate(audio_encoder_.get(), - 32001 + overhead_rate)); -} - -TEST_F(AudioDecoderIsacSwbTest, EncodeDecode) { - int tolerance = 19757; - double mse = 8.18e6; - int delay = 160; // Delay from input to output. - EncodeDecodeTest(0, tolerance, mse, delay); - ReInitTest(); - EXPECT_FALSE(decoder_->HasDecodePlc()); -} - -TEST_F(AudioDecoderIsacSwbTest, SetTargetBitrate) { - const int overhead_rate = - 8 * kOverheadBytesPerPacket * codec_input_rate_hz_ / frame_size_; - EXPECT_EQ(10000, - SetAndGetTargetBitrate(audio_encoder_.get(), 9999 + overhead_rate)); - EXPECT_EQ(10000, SetAndGetTargetBitrate(audio_encoder_.get(), - 10000 + overhead_rate)); - EXPECT_EQ(23456, SetAndGetTargetBitrate(audio_encoder_.get(), - 23456 + overhead_rate)); - EXPECT_EQ(56000, SetAndGetTargetBitrate(audio_encoder_.get(), - 56000 + overhead_rate)); - EXPECT_EQ(56000, SetAndGetTargetBitrate(audio_encoder_.get(), - 56001 + overhead_rate)); -} - -// Run bit exactness test only for release builds. -#if defined(NDEBUG) -TEST_F(AudioDecoderIsacFixTest, EncodeDecode) { - int tolerance = 11034; - double mse = 3.46e6; - int delay = 54; // Delay from input to output. -#if defined(WEBRTC_ANDROID) && defined(WEBRTC_ARCH_ARM) - static const int kEncodedBytes = 685; -#elif defined(WEBRTC_MAC) && defined(WEBRTC_ARCH_ARM64) // M1 Mac - static const int kEncodedBytes = 673; -#elif defined(WEBRTC_ARCH_ARM64) - static const int kEncodedBytes = 673; -#elif defined(WEBRTC_WIN) && defined(_MSC_VER) && !defined(__clang__) - static const int kEncodedBytes = 671; -#elif defined(WEBRTC_IOS) && defined(WEBRTC_ARCH_X86_64) - static const int kEncodedBytes = 671; -#else - static const int kEncodedBytes = 671; -#endif - EncodeDecodeTest(kEncodedBytes, tolerance, mse, delay); - ReInitTest(); - EXPECT_FALSE(decoder_->HasDecodePlc()); -} -#endif - -TEST_F(AudioDecoderIsacFixTest, SetTargetBitrate) { - const int overhead_rate = - 8 * kOverheadBytesPerPacket * codec_input_rate_hz_ / frame_size_; - EXPECT_EQ(10000, - SetAndGetTargetBitrate(audio_encoder_.get(), 9999 + overhead_rate)); - EXPECT_EQ(10000, SetAndGetTargetBitrate(audio_encoder_.get(), - 10000 + overhead_rate)); - EXPECT_EQ(23456, SetAndGetTargetBitrate(audio_encoder_.get(), - 23456 + overhead_rate)); - EXPECT_EQ(32000, SetAndGetTargetBitrate(audio_encoder_.get(), - 32000 + overhead_rate)); - EXPECT_EQ(32000, SetAndGetTargetBitrate(audio_encoder_.get(), - 32001 + overhead_rate)); -} - TEST_F(AudioDecoderG722Test, EncodeDecode) { int tolerance = 6176; double mse = 238630.0; diff --git a/modules/audio_coding/neteq/test/neteq_isac_quality_test.cc b/modules/audio_coding/neteq/test/neteq_isac_quality_test.cc deleted file mode 100644 index 6a096c307c..0000000000 --- a/modules/audio_coding/neteq/test/neteq_isac_quality_test.cc +++ /dev/null @@ -1,102 +0,0 @@ -/* - * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#include "absl/flags/flag.h" -#include "modules/audio_coding/codecs/isac/fix/include/isacfix.h" -#include "modules/audio_coding/neteq/tools/neteq_quality_test.h" - -ABSL_FLAG(int, bit_rate_kbps, 32, "Target bit rate (kbps)."); - -using ::testing::InitGoogleTest; - -namespace webrtc { -namespace test { -namespace { -static const int kIsacBlockDurationMs = 30; -static const int kIsacInputSamplingKhz = 16; -static const int kIsacOutputSamplingKhz = 16; -} // namespace - -class NetEqIsacQualityTest : public NetEqQualityTest { - protected: - NetEqIsacQualityTest(); - void SetUp() override; - void TearDown() override; - int EncodeBlock(int16_t* in_data, - size_t block_size_samples, - rtc::Buffer* payload, - size_t max_bytes) override; - - private: - ISACFIX_MainStruct* isac_encoder_; - int bit_rate_kbps_; -}; - -NetEqIsacQualityTest::NetEqIsacQualityTest() - : NetEqQualityTest(kIsacBlockDurationMs, - kIsacInputSamplingKhz, - kIsacOutputSamplingKhz, - SdpAudioFormat("isac", 16000, 1)), - isac_encoder_(NULL), - bit_rate_kbps_(absl::GetFlag(FLAGS_bit_rate_kbps)) { - // Flag validation - RTC_CHECK(absl::GetFlag(FLAGS_bit_rate_kbps) >= 10 && - absl::GetFlag(FLAGS_bit_rate_kbps) <= 32) - << "Invalid bit rate, should be between 10 and 32 kbps."; -} - -void NetEqIsacQualityTest::SetUp() { - ASSERT_EQ(1u, channels_) << "iSAC supports only mono audio."; - // Create encoder memory. - WebRtcIsacfix_Create(&isac_encoder_); - ASSERT_TRUE(isac_encoder_ != NULL); - EXPECT_EQ(0, WebRtcIsacfix_EncoderInit(isac_encoder_, 1)); - // Set bitrate and block length. - EXPECT_EQ(0, WebRtcIsacfix_Control(isac_encoder_, bit_rate_kbps_ * 1000, - kIsacBlockDurationMs)); - NetEqQualityTest::SetUp(); -} - -void NetEqIsacQualityTest::TearDown() { - // Free memory. - EXPECT_EQ(0, WebRtcIsacfix_Free(isac_encoder_)); - NetEqQualityTest::TearDown(); -} - -int NetEqIsacQualityTest::EncodeBlock(int16_t* in_data, - size_t block_size_samples, - rtc::Buffer* payload, - size_t max_bytes) { - // ISAC takes 10 ms for every call. - const int subblocks = kIsacBlockDurationMs / 10; - const int subblock_length = 10 * kIsacInputSamplingKhz; - int value = 0; - - int pointer = 0; - for (int idx = 0; idx < subblocks; idx++, pointer += subblock_length) { - // The Isac encoder does not perform encoding (and returns 0) until it - // receives a sequence of sub-blocks that amount to the frame duration. - EXPECT_EQ(0, value); - payload->AppendData(max_bytes, [&](rtc::ArrayView payload) { - value = WebRtcIsacfix_Encode(isac_encoder_, &in_data[pointer], - payload.data()); - return (value >= 0) ? static_cast(value) : 0; - }); - } - EXPECT_GT(value, 0); - return value; -} - -TEST_F(NetEqIsacQualityTest, Test) { - Simulate(); -} - -} // namespace test -} // namespace webrtc diff --git a/modules/audio_coding/neteq/tools/rtp_encode.cc b/modules/audio_coding/neteq/tools/rtp_encode.cc index 6aeeb6d129..8adca927f0 100644 --- a/modules/audio_coding/neteq/tools/rtp_encode.cc +++ b/modules/audio_coding/neteq/tools/rtp_encode.cc @@ -30,7 +30,6 @@ #include "api/audio_codecs/g711/audio_encoder_g711.h" #include "api/audio_codecs/g722/audio_encoder_g722.h" #include "api/audio_codecs/ilbc/audio_encoder_ilbc.h" -#include "api/audio_codecs/isac/audio_encoder_isac.h" #include "api/audio_codecs/opus/audio_encoder_opus.h" #include "modules/audio_coding/codecs/cng/audio_encoder_cng.h" #include "modules/audio_coding/include/audio_coding_module.h" @@ -71,7 +70,6 @@ enum class CodecType { kPcm16b32, kPcm16b48, kIlbc, - kIsac }; struct CodecTypeAndInfo { @@ -94,8 +92,7 @@ const std::map& CodecList() { {"pcm16b_16", {CodecType::kPcm16b16, 94, false}}, {"pcm16b_32", {CodecType::kPcm16b32, 95, false}}, {"pcm16b_48", {CodecType::kPcm16b48, 96, false}}, - {"ilbc", {CodecType::kIlbc, 102, false}}, - {"isac", {CodecType::kIsac, 103, false}}}; + {"ilbc", {CodecType::kIlbc, 102, false}}}; return *codec_list; } @@ -236,11 +233,6 @@ std::unique_ptr CreateEncoder(CodecType codec_type, return AudioEncoderIlbc::MakeAudioEncoder( GetCodecConfig(), payload_type); } - - case CodecType::kIsac: { - return AudioEncoderIsac::MakeAudioEncoder( - GetCodecConfig(), payload_type); - } } RTC_DCHECK_NOTREACHED(); return nullptr; diff --git a/modules/audio_coding/test/EncodeDecodeTest.cc b/modules/audio_coding/test/EncodeDecodeTest.cc index 8d4bcce8df..9f9c4aa74c 100644 --- a/modules/audio_coding/test/EncodeDecodeTest.cc +++ b/modules/audio_coding/test/EncodeDecodeTest.cc @@ -110,9 +110,7 @@ void Receiver::Setup(AudioCodingModule* acm, EXPECT_EQ(0, acm->InitializeReceiver()); if (channels == 1) { - acm->SetReceiveCodecs({{103, {"ISAC", 16000, 1}}, - {104, {"ISAC", 32000, 1}}, - {107, {"L16", 8000, 1}}, + acm->SetReceiveCodecs({{107, {"L16", 8000, 1}}, {108, {"L16", 16000, 1}}, {109, {"L16", 32000, 1}}, {0, {"PCMU", 8000, 1}}, @@ -232,7 +230,6 @@ EncodeDecodeTest::EncodeDecodeTest() = default; void EncodeDecodeTest::Perform() { const std::map send_codecs = { - {103, {"ISAC", 16000, 1}}, {104, {"ISAC", 32000, 1}}, {107, {"L16", 8000, 1}}, {108, {"L16", 16000, 1}}, {109, {"L16", 32000, 1}}, {0, {"PCMU", 8000, 1}}, {8, {"PCMA", 8000, 1}}, diff --git a/modules/audio_coding/test/TestAllCodecs.cc b/modules/audio_coding/test/TestAllCodecs.cc index e93df346f1..b44037d732 100644 --- a/modules/audio_coding/test/TestAllCodecs.cc +++ b/modules/audio_coding/test/TestAllCodecs.cc @@ -130,9 +130,7 @@ void TestAllCodecs::Perform() { acm_a_->InitializeReceiver(); acm_b_->InitializeReceiver(); - acm_b_->SetReceiveCodecs({{103, {"ISAC", 16000, 1}}, - {104, {"ISAC", 32000, 1}}, - {107, {"L16", 8000, 1}}, + acm_b_->SetReceiveCodecs({{107, {"L16", 8000, 1}}, {108, {"L16", 16000, 1}}, {109, {"L16", 32000, 1}}, {111, {"L16", 8000, 2}}, @@ -186,33 +184,6 @@ void TestAllCodecs::Perform() { RegisterSendCodec('A', codec_ilbc, 8000, 15200, 320, 0); Run(channel_a_to_b_); outfile_b_.Close(); -#endif -#if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)) - test_count_++; - OpenOutFile(test_count_); - char codec_isac[] = "ISAC"; - RegisterSendCodec('A', codec_isac, 16000, -1, 480, kVariableSize); - Run(channel_a_to_b_); - RegisterSendCodec('A', codec_isac, 16000, -1, 960, kVariableSize); - Run(channel_a_to_b_); - RegisterSendCodec('A', codec_isac, 16000, 15000, 480, kVariableSize); - Run(channel_a_to_b_); - RegisterSendCodec('A', codec_isac, 16000, 32000, 960, kVariableSize); - Run(channel_a_to_b_); - outfile_b_.Close(); -#endif -#ifdef WEBRTC_CODEC_ISAC - test_count_++; - OpenOutFile(test_count_); - RegisterSendCodec('A', codec_isac, 32000, -1, 960, kVariableSize); - Run(channel_a_to_b_); - RegisterSendCodec('A', codec_isac, 32000, 56000, 960, kVariableSize); - Run(channel_a_to_b_); - RegisterSendCodec('A', codec_isac, 32000, 37000, 960, kVariableSize); - Run(channel_a_to_b_); - RegisterSendCodec('A', codec_isac, 32000, 32000, 960, kVariableSize); - Run(channel_a_to_b_); - outfile_b_.Close(); #endif test_count_++; OpenOutFile(test_count_); @@ -319,15 +290,11 @@ void TestAllCodecs::RegisterSendCodec(char side, // Store packet-size in samples, used to validate the received packet. // If G.722, store half the size to compensate for the timestamp bug in the // RFC for G.722. - // If iSAC runs in adaptive mode, packet size in samples can change on the - // fly, so we exclude this test by setting `packet_size_samples_` to -1. int clockrate_hz = sampling_freq_hz; size_t num_channels = 1; if (absl::EqualsIgnoreCase(codec_name, "G722")) { packet_size_samples_ = packet_size / 2; clockrate_hz = sampling_freq_hz / 2; - } else if (absl::EqualsIgnoreCase(codec_name, "ISAC") && (rate == -1)) { - packet_size_samples_ = -1; } else if (absl::EqualsIgnoreCase(codec_name, "OPUS")) { packet_size_samples_ = packet_size; num_channels = 2; diff --git a/modules/audio_coding/test/TestRedFec.cc b/modules/audio_coding/test/TestRedFec.cc index 892fbc83d6..fff48b27bc 100644 --- a/modules/audio_coding/test/TestRedFec.cc +++ b/modules/audio_coding/test/TestRedFec.cc @@ -22,8 +22,6 @@ #include "api/audio_codecs/g711/audio_encoder_g711.h" #include "api/audio_codecs/g722/audio_decoder_g722.h" #include "api/audio_codecs/g722/audio_encoder_g722.h" -#include "api/audio_codecs/isac/audio_decoder_isac_float.h" -#include "api/audio_codecs/isac/audio_encoder_isac_float.h" #include "api/audio_codecs/opus/audio_decoder_opus.h" #include "api/audio_codecs/opus/audio_encoder_opus.h" #include "modules/audio_coding/codecs/cng/audio_encoder_cng.h" @@ -38,12 +36,10 @@ namespace webrtc { TestRedFec::TestRedFec() : encoder_factory_(CreateAudioEncoderFactory()), decoder_factory_(CreateAudioDecoderFactory()), _acmA(AudioCodingModule::Create( @@ -95,19 +91,6 @@ void TestRedFec::Perform() { Run(); _outFileB.Close(); - RegisterSendCodec(_acmA, {"ISAC", 16000, 1}, Vad::kVadVeryAggressive, false); - OpenOutFile(_testCntr); - Run(); - _outFileB.Close(); - - // Switch to a 32 kHz codec; RED should be switched off. - RegisterSendCodec(_acmA, {"ISAC", 32000, 1}, Vad::kVadVeryAggressive, false); - OpenOutFile(_testCntr); - Run(); - _outFileB.Close(); - - RegisterSendCodec(_acmA, {"ISAC", 32000, 1}, absl::nullopt, false); - _channelA2B->SetFECTestWithPacketLoss(true); // Following tests are under packet losses. @@ -118,22 +101,6 @@ void TestRedFec::Perform() { Run(); _outFileB.Close(); - // Switch to a 16 kHz codec, RED should have been switched off. - RegisterSendCodec(_acmA, {"ISAC", 16000, 1}, Vad::kVadVeryAggressive, false); - - OpenOutFile(_testCntr); - Run(); - _outFileB.Close(); - - // Switch to a 32 kHz codec, RED should have been switched off. - RegisterSendCodec(_acmA, {"ISAC", 32000, 1}, Vad::kVadVeryAggressive, false); - - OpenOutFile(_testCntr); - Run(); - _outFileB.Close(); - - RegisterSendCodec(_acmA, {"ISAC", 32000, 1}, absl::nullopt, false); - RegisterSendCodec(_acmA, {"opus", 48000, 2}, absl::nullopt, false); // _channelA2B imposes 25% packet loss rate. diff --git a/modules/audio_coding/test/TestVADDTX.cc b/modules/audio_coding/test/TestVADDTX.cc index cb05deb92a..19367d9bde 100644 --- a/modules/audio_coding/test/TestVADDTX.cc +++ b/modules/audio_coding/test/TestVADDTX.cc @@ -18,8 +18,6 @@ #include "api/audio_codecs/audio_encoder_factory_template.h" #include "api/audio_codecs/ilbc/audio_decoder_ilbc.h" #include "api/audio_codecs/ilbc/audio_encoder_ilbc.h" -#include "api/audio_codecs/isac/audio_decoder_isac_float.h" -#include "api/audio_codecs/isac/audio_encoder_isac_float.h" #include "api/audio_codecs/opus/audio_decoder_opus.h" #include "api/audio_codecs/opus/audio_encoder_opus.h" #include "modules/audio_coding/codecs/cng/audio_encoder_cng.h" @@ -68,12 +66,10 @@ void MonitoringAudioPacketizationCallback::GetStatistics(uint32_t* counter) { } TestVadDtx::TestVadDtx() - : encoder_factory_(CreateAudioEncoderFactory()), - decoder_factory_(CreateAudioDecoderFactory()), + : encoder_factory_( + CreateAudioEncoderFactory()), + decoder_factory_( + CreateAudioDecoderFactory()), acm_send_(AudioCodingModule::Create( AudioCodingModule::Config(decoder_factory_))), acm_receive_(AudioCodingModule::Create( @@ -182,8 +178,6 @@ void TestVadDtx::Run(absl::string_view in_filename, TestWebRtcVadDtx::TestWebRtcVadDtx() : output_file_num_(0) {} void TestWebRtcVadDtx::Perform() { - RunTestCases({"ISAC", 16000, 1}); - RunTestCases({"ISAC", 32000, 1}); RunTestCases({"ILBC", 8000, 1}); RunTestCases({"opus", 48000, 2}); } diff --git a/modules/audio_coding/test/Tester.cc b/modules/audio_coding/test/Tester.cc index 113dbe059e..7612aa43a3 100644 --- a/modules/audio_coding/test/Tester.cc +++ b/modules/audio_coding/test/Tester.cc @@ -21,7 +21,6 @@ #include "modules/audio_coding/test/TestStereo.h" #include "modules/audio_coding/test/TestVADDTX.h" #include "modules/audio_coding/test/TwoWayCommunication.h" -#include "modules/audio_coding/test/iSACTest.h" #include "modules/audio_coding/test/opus_test.h" #include "test/gtest.h" #include "test/testsupport/file_utils.h" @@ -42,25 +41,6 @@ TEST(AudioCodingModuleTest, TestRedFec) { webrtc::TestRedFec().Perform(); } -#if defined(WEBRTC_ANDROID) -TEST(AudioCodingModuleTest, DISABLED_TestIsac) { -#else -TEST(AudioCodingModuleTest, TestIsac) { -#endif - webrtc::ISACTest().Perform(); -} - -#if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)) && \ - defined(WEBRTC_CODEC_ILBC) -#if defined(WEBRTC_ANDROID) -TEST(AudioCodingModuleTest, DISABLED_TwoWayCommunication) { -#else -TEST(AudioCodingModuleTest, TwoWayCommunication) { -#endif - webrtc::TwoWayCommunication().Perform(); -} -#endif - // Disabled on ios as flaky, see https://crbug.com/webrtc/7057 #if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS) TEST(AudioCodingModuleTest, DISABLED_TestStereo) { diff --git a/modules/audio_coding/test/iSACTest.cc b/modules/audio_coding/test/iSACTest.cc deleted file mode 100644 index 246c485afe..0000000000 --- a/modules/audio_coding/test/iSACTest.cc +++ /dev/null @@ -1,273 +0,0 @@ -/* - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#include "modules/audio_coding/test/iSACTest.h" - -#include -#include - -#include "absl/strings/match.h" -#include "api/audio_codecs/builtin_audio_decoder_factory.h" -#include "api/audio_codecs/isac/audio_encoder_isac_float.h" -#include "rtc_base/strings/string_builder.h" -#include "rtc_base/time_utils.h" -#include "test/gmock.h" -#include "test/gtest.h" -#include "test/testsupport/file_utils.h" - -namespace webrtc { - -using ::testing::AnyOf; -using ::testing::Eq; -using ::testing::StrCaseEq; - -namespace { - -constexpr int kISAC16kPayloadType = 103; -constexpr int kISAC32kPayloadType = 104; -const SdpAudioFormat kISAC16kFormat = {"ISAC", 16000, 1}; -const SdpAudioFormat kISAC32kFormat = {"ISAC", 32000, 1}; - -AudioEncoderIsacFloat::Config TweakConfig( - AudioEncoderIsacFloat::Config config, - const ACMTestISACConfig& test_config) { - if (test_config.currentRateBitPerSec > 0) { - config.bit_rate = test_config.currentRateBitPerSec; - } - if (test_config.currentFrameSizeMsec != 0) { - config.frame_size_ms = test_config.currentFrameSizeMsec; - } - EXPECT_THAT(config.IsOk(), Eq(true)); - return config; -} - -void SetISACConfigDefault(ACMTestISACConfig& isacConfig) { - isacConfig.currentRateBitPerSec = 0; - isacConfig.currentFrameSizeMsec = 0; - isacConfig.encodingMode = -1; - isacConfig.initRateBitPerSec = 0; - isacConfig.initFrameSizeInMsec = 0; - isacConfig.enforceFrameSize = false; -} - -} // namespace - -ISACTest::ISACTest() - : _acmA(AudioCodingModule::Create( - AudioCodingModule::Config(CreateBuiltinAudioDecoderFactory()))), - _acmB(AudioCodingModule::Create( - AudioCodingModule::Config(CreateBuiltinAudioDecoderFactory()))) {} - -ISACTest::~ISACTest() {} - -void ISACTest::Setup() { - // Register both iSAC-wb & iSAC-swb in both sides as receiver codecs. - std::map receive_codecs = { - {kISAC16kPayloadType, kISAC16kFormat}, - {kISAC32kPayloadType, kISAC32kFormat}}; - _acmA->SetReceiveCodecs(receive_codecs); - _acmB->SetReceiveCodecs(receive_codecs); - - //--- Set A-to-B channel - _channel_A2B.reset(new Channel); - EXPECT_EQ(0, _acmA->RegisterTransportCallback(_channel_A2B.get())); - _channel_A2B->RegisterReceiverACM(_acmB.get()); - - //--- Set B-to-A channel - _channel_B2A.reset(new Channel); - EXPECT_EQ(0, _acmB->RegisterTransportCallback(_channel_B2A.get())); - _channel_B2A->RegisterReceiverACM(_acmA.get()); - - file_name_swb_ = - webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm"); - - _acmB->SetEncoder(AudioEncoderIsacFloat::MakeAudioEncoder( - *AudioEncoderIsacFloat::SdpToConfig(kISAC16kFormat), - kISAC16kPayloadType)); - _acmA->SetEncoder(AudioEncoderIsacFloat::MakeAudioEncoder( - *AudioEncoderIsacFloat::SdpToConfig(kISAC32kFormat), - kISAC32kPayloadType)); - - _inFileA.Open(file_name_swb_, 32000, "rb"); - // Set test length to 500 ms (50 blocks of 10 ms each). - _inFileA.SetNum10MsBlocksToRead(50); - // Fast-forward 1 second (100 blocks) since the files start with silence. - _inFileA.FastForward(100); - std::string fileNameA = webrtc::test::OutputPath() + "testisac_a.pcm"; - std::string fileNameB = webrtc::test::OutputPath() + "testisac_b.pcm"; - _outFileA.Open(fileNameA, 32000, "wb"); - _outFileB.Open(fileNameB, 32000, "wb"); - - while (!_inFileA.EndOfFile()) { - Run10ms(); - } - - _inFileA.Close(); - _outFileA.Close(); - _outFileB.Close(); -} - -void ISACTest::Perform() { - Setup(); - - int16_t testNr = 0; - ACMTestISACConfig wbISACConfig; - ACMTestISACConfig swbISACConfig; - - SetISACConfigDefault(wbISACConfig); - SetISACConfigDefault(swbISACConfig); - - wbISACConfig.currentRateBitPerSec = -1; - swbISACConfig.currentRateBitPerSec = -1; - testNr++; - EncodeDecode(testNr, wbISACConfig, swbISACConfig); - - SetISACConfigDefault(wbISACConfig); - SetISACConfigDefault(swbISACConfig); - testNr++; - EncodeDecode(testNr, wbISACConfig, swbISACConfig); - - testNr++; - SwitchingSamplingRate(testNr, 4); -} - -void ISACTest::Run10ms() { - AudioFrame audioFrame; - EXPECT_GT(_inFileA.Read10MsData(audioFrame), 0); - EXPECT_GE(_acmA->Add10MsData(audioFrame), 0); - EXPECT_GE(_acmB->Add10MsData(audioFrame), 0); - bool muted; - EXPECT_EQ(0, _acmA->PlayoutData10Ms(32000, &audioFrame, &muted)); - ASSERT_FALSE(muted); - _outFileA.Write10MsData(audioFrame); - EXPECT_EQ(0, _acmB->PlayoutData10Ms(32000, &audioFrame, &muted)); - ASSERT_FALSE(muted); - _outFileB.Write10MsData(audioFrame); -} - -void ISACTest::EncodeDecode(int testNr, - ACMTestISACConfig& wbISACConfig, - ACMTestISACConfig& swbISACConfig) { - // Files in Side A and B - _inFileA.Open(file_name_swb_, 32000, "rb", true); - _inFileB.Open(file_name_swb_, 32000, "rb", true); - - std::string file_name_out; - rtc::StringBuilder file_stream_a; - rtc::StringBuilder file_stream_b; - file_stream_a << webrtc::test::OutputPath(); - file_stream_b << webrtc::test::OutputPath(); - file_stream_a << "out_iSACTest_A_" << testNr << ".pcm"; - file_stream_b << "out_iSACTest_B_" << testNr << ".pcm"; - file_name_out = file_stream_a.str(); - _outFileA.Open(file_name_out, 32000, "wb"); - file_name_out = file_stream_b.str(); - _outFileB.Open(file_name_out, 32000, "wb"); - - // Side A is sending super-wideband, and side B is sending wideband. - _acmA->SetEncoder(AudioEncoderIsacFloat::MakeAudioEncoder( - TweakConfig(*AudioEncoderIsacFloat::SdpToConfig(kISAC32kFormat), - swbISACConfig), - kISAC32kPayloadType)); - _acmB->SetEncoder(AudioEncoderIsacFloat::MakeAudioEncoder( - TweakConfig(*AudioEncoderIsacFloat::SdpToConfig(kISAC16kFormat), - wbISACConfig), - kISAC16kPayloadType)); - - _channel_A2B->ResetStats(); - _channel_B2A->ResetStats(); - - while (!(_inFileA.EndOfFile() || _inFileA.Rewinded())) { - Run10ms(); - } - - _channel_A2B->ResetStats(); - _channel_B2A->ResetStats(); - - _outFileA.Close(); - _outFileB.Close(); - _inFileA.Close(); - _inFileB.Close(); -} - -void ISACTest::SwitchingSamplingRate(int testNr, int maxSampRateChange) { - // Files in Side A - _inFileA.Open(file_name_swb_, 32000, "rb"); - _inFileB.Open(file_name_swb_, 32000, "rb"); - - std::string file_name_out; - rtc::StringBuilder file_stream_a; - rtc::StringBuilder file_stream_b; - file_stream_a << webrtc::test::OutputPath(); - file_stream_b << webrtc::test::OutputPath(); - file_stream_a << "out_iSACTest_A_" << testNr << ".pcm"; - file_stream_b << "out_iSACTest_B_" << testNr << ".pcm"; - file_name_out = file_stream_a.str(); - _outFileA.Open(file_name_out, 32000, "wb"); - file_name_out = file_stream_b.str(); - _outFileB.Open(file_name_out, 32000, "wb"); - - // Start with side A sending super-wideband and side B seding wideband. - // Toggle sending wideband/super-wideband in this test. - _acmA->SetEncoder(AudioEncoderIsacFloat::MakeAudioEncoder( - *AudioEncoderIsacFloat::SdpToConfig(kISAC32kFormat), - kISAC32kPayloadType)); - _acmB->SetEncoder(AudioEncoderIsacFloat::MakeAudioEncoder( - *AudioEncoderIsacFloat::SdpToConfig(kISAC16kFormat), - kISAC16kPayloadType)); - - int numSendCodecChanged = 0; - while (numSendCodecChanged < (maxSampRateChange << 1)) { - Run10ms(); - if (_inFileA.EndOfFile()) { - if (_inFileA.SamplingFrequency() == 16000) { - // Switch side A to send super-wideband. - _inFileA.Close(); - _inFileA.Open(file_name_swb_, 32000, "rb"); - _acmA->SetEncoder(AudioEncoderIsacFloat::MakeAudioEncoder( - *AudioEncoderIsacFloat::SdpToConfig(kISAC32kFormat), - kISAC32kPayloadType)); - } else { - // Switch side A to send wideband. - _inFileA.Close(); - _inFileA.Open(file_name_swb_, 32000, "rb"); - _acmA->SetEncoder(AudioEncoderIsacFloat::MakeAudioEncoder( - *AudioEncoderIsacFloat::SdpToConfig(kISAC16kFormat), - kISAC16kPayloadType)); - } - numSendCodecChanged++; - } - - if (_inFileB.EndOfFile()) { - if (_inFileB.SamplingFrequency() == 16000) { - // Switch side B to send super-wideband. - _inFileB.Close(); - _inFileB.Open(file_name_swb_, 32000, "rb"); - _acmB->SetEncoder(AudioEncoderIsacFloat::MakeAudioEncoder( - *AudioEncoderIsacFloat::SdpToConfig(kISAC32kFormat), - kISAC32kPayloadType)); - } else { - // Switch side B to send wideband. - _inFileB.Close(); - _inFileB.Open(file_name_swb_, 32000, "rb"); - _acmB->SetEncoder(AudioEncoderIsacFloat::MakeAudioEncoder( - *AudioEncoderIsacFloat::SdpToConfig(kISAC16kFormat), - kISAC16kPayloadType)); - } - numSendCodecChanged++; - } - } - _outFileA.Close(); - _outFileB.Close(); - _inFileA.Close(); - _inFileB.Close(); -} - -} // namespace webrtc diff --git a/modules/audio_coding/test/iSACTest.h b/modules/audio_coding/test/iSACTest.h deleted file mode 100644 index f6efeeac1c..0000000000 --- a/modules/audio_coding/test/iSACTest.h +++ /dev/null @@ -1,68 +0,0 @@ -/* - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef MODULES_AUDIO_CODING_TEST_ISACTEST_H_ -#define MODULES_AUDIO_CODING_TEST_ISACTEST_H_ - -#include - -#include - -#include "modules/audio_coding/include/audio_coding_module.h" -#include "modules/audio_coding/test/Channel.h" -#include "modules/audio_coding/test/PCMFile.h" - -namespace webrtc { - -struct ACMTestISACConfig { - int32_t currentRateBitPerSec; - int16_t currentFrameSizeMsec; - int16_t encodingMode; - uint32_t initRateBitPerSec; - int16_t initFrameSizeInMsec; - bool enforceFrameSize; -}; - -class ISACTest { - public: - ISACTest(); - ~ISACTest(); - - void Perform(); - - private: - void Setup(); - - void Run10ms(); - - void EncodeDecode(int testNr, - ACMTestISACConfig& wbISACConfig, - ACMTestISACConfig& swbISACConfig); - - void SwitchingSamplingRate(int testNr, int maxSampRateChange); - - std::unique_ptr _acmA; - std::unique_ptr _acmB; - - std::unique_ptr _channel_A2B; - std::unique_ptr _channel_B2A; - - PCMFile _inFileA; - PCMFile _inFileB; - - PCMFile _outFileA; - PCMFile _outFileB; - - std::string file_name_swb_; -}; - -} // namespace webrtc - -#endif // MODULES_AUDIO_CODING_TEST_ISACTEST_H_ diff --git a/pc/peer_connection_interface_unittest.cc b/pc/peer_connection_interface_unittest.cc index e4c1ce25df..71d6f0c7e1 100644 --- a/pc/peer_connection_interface_unittest.cc +++ b/pc/peer_connection_interface_unittest.cc @@ -115,7 +115,7 @@ static const char kSdpStringWithStream1PlanB[] = "o=- 0 0 IN IP4 127.0.0.1\r\n" "s=-\r\n" "t=0 0\r\n" - "m=audio 1 RTP/AVPF 103\r\n" + "m=audio 1 RTP/AVPF 111\r\n" "a=ice-ufrag:e5785931\r\n" "a=ice-pwd:36fb7878390db89481c1d46daa4278d8\r\n" "a=fingerprint:sha-256 58:AB:6E:F5:F1:E4:57:B7:E9:46:F4:86:04:28:F9:A7:ED:" @@ -123,7 +123,7 @@ static const char kSdpStringWithStream1PlanB[] = "a=mid:audio\r\n" "a=sendrecv\r\n" "a=rtcp-mux\r\n" - "a=rtpmap:103 ISAC/16000\r\n" + "a=rtpmap:111 OPUS/48000/2\r\n" "a=ssrc:1 cname:stream1\r\n" "a=ssrc:1 msid:stream1 audiotrack0\r\n" "m=video 1 RTP/AVPF 120\r\n" @@ -145,7 +145,7 @@ static const char kSdpStringWithStream1UnifiedPlan[] = "o=- 0 0 IN IP4 127.0.0.1\r\n" "s=-\r\n" "t=0 0\r\n" - "m=audio 1 RTP/AVPF 103\r\n" + "m=audio 1 RTP/AVPF 111\r\n" "a=ice-ufrag:e5785931\r\n" "a=ice-pwd:36fb7878390db89481c1d46daa4278d8\r\n" "a=fingerprint:sha-256 58:AB:6E:F5:F1:E4:57:B7:E9:46:F4:86:04:28:F9:A7:ED:" @@ -153,7 +153,7 @@ static const char kSdpStringWithStream1UnifiedPlan[] = "a=mid:0\r\n" "a=sendrecv\r\n" "a=rtcp-mux\r\n" - "a=rtpmap:103 ISAC/16000\r\n" + "a=rtpmap:111 OPUS/48000/2\r\n" "a=msid:stream1 audiotrack0\r\n" "a=ssrc:1 cname:stream1\r\n" "m=video 1 RTP/AVPF 120\r\n" @@ -175,14 +175,14 @@ static const char kSdpStringWithStream1AudioTrackOnly[] = "o=- 0 0 IN IP4 127.0.0.1\r\n" "s=-\r\n" "t=0 0\r\n" - "m=audio 1 RTP/AVPF 103\r\n" + "m=audio 1 RTP/AVPF 111\r\n" "a=ice-ufrag:e5785931\r\n" "a=ice-pwd:36fb7878390db89481c1d46daa4278d8\r\n" "a=fingerprint:sha-256 58:AB:6E:F5:F1:E4:57:B7:E9:46:F4:86:04:28:F9:A7:ED:" "BD:AB:AE:40:EF:CE:9A:51:2C:2A:B1:9B:8B:78:84\r\n" "a=mid:audio\r\n" "a=sendrecv\r\n" - "a=rtpmap:103 ISAC/16000\r\n" + "a=rtpmap:111 OPUS/48000/2\r\n" "a=ssrc:1 cname:stream1\r\n" "a=ssrc:1 msid:stream1 audiotrack0\r\n" "a=rtcp-mux\r\n"; @@ -196,7 +196,7 @@ static const char kSdpStringWithStream1And2PlanB[] = "s=-\r\n" "t=0 0\r\n" "a=msid-semantic: WMS stream1 stream2\r\n" - "m=audio 1 RTP/AVPF 103\r\n" + "m=audio 1 RTP/AVPF 111\r\n" "a=ice-ufrag:e5785931\r\n" "a=ice-pwd:36fb7878390db89481c1d46daa4278d8\r\n" "a=fingerprint:sha-256 58:AB:6E:F5:F1:E4:57:B7:E9:46:F4:86:04:28:F9:A7:ED:" @@ -204,7 +204,7 @@ static const char kSdpStringWithStream1And2PlanB[] = "a=mid:audio\r\n" "a=sendrecv\r\n" "a=rtcp-mux\r\n" - "a=rtpmap:103 ISAC/16000\r\n" + "a=rtpmap:111 OPUS/48000/2\r\n" "a=ssrc:1 cname:stream1\r\n" "a=ssrc:1 msid:stream1 audiotrack0\r\n" "a=ssrc:3 cname:stream2\r\n" @@ -228,7 +228,7 @@ static const char kSdpStringWithStream1And2UnifiedPlan[] = "s=-\r\n" "t=0 0\r\n" "a=msid-semantic: WMS stream1 stream2\r\n" - "m=audio 1 RTP/AVPF 103\r\n" + "m=audio 1 RTP/AVPF 111\r\n" "a=ice-ufrag:e5785931\r\n" "a=ice-pwd:36fb7878390db89481c1d46daa4278d8\r\n" "a=fingerprint:sha-256 58:AB:6E:F5:F1:E4:57:B7:E9:46:F4:86:04:28:F9:A7:ED:" @@ -236,7 +236,7 @@ static const char kSdpStringWithStream1And2UnifiedPlan[] = "a=mid:0\r\n" "a=sendrecv\r\n" "a=rtcp-mux\r\n" - "a=rtpmap:103 ISAC/16000\r\n" + "a=rtpmap:111 OPUS/48000/2\r\n" "a=ssrc:1 cname:stream1\r\n" "a=ssrc:1 msid:stream1 audiotrack0\r\n" "m=video 1 RTP/AVPF 120\r\n" @@ -250,7 +250,7 @@ static const char kSdpStringWithStream1And2UnifiedPlan[] = "a=rtpmap:120 VP8/0\r\n" "a=ssrc:2 cname:stream1\r\n" "a=ssrc:2 msid:stream1 videotrack0\r\n" - "m=audio 1 RTP/AVPF 103\r\n" + "m=audio 1 RTP/AVPF 111\r\n" "a=ice-ufrag:e5785931\r\n" "a=ice-pwd:36fb7878390db89481c1d46daa4278d8\r\n" "a=fingerprint:sha-256 58:AB:6E:F5:F1:E4:57:B7:E9:46:F4:86:04:28:F9:A7:ED:" @@ -258,7 +258,7 @@ static const char kSdpStringWithStream1And2UnifiedPlan[] = "a=mid:2\r\n" "a=sendrecv\r\n" "a=rtcp-mux\r\n" - "a=rtpmap:103 ISAC/16000\r\n" + "a=rtpmap:111 OPUS/48000/2\r\n" "a=ssrc:3 cname:stream2\r\n" "a=ssrc:3 msid:stream2 audiotrack1\r\n" "m=video 1 RTP/AVPF 120\r\n" @@ -279,7 +279,7 @@ static const char kSdpStringWithoutStreams[] = "o=- 0 0 IN IP4 127.0.0.1\r\n" "s=-\r\n" "t=0 0\r\n" - "m=audio 1 RTP/AVPF 103\r\n" + "m=audio 1 RTP/AVPF 111\r\n" "a=ice-ufrag:e5785931\r\n" "a=ice-pwd:36fb7878390db89481c1d46daa4278d8\r\n" "a=fingerprint:sha-256 58:AB:6E:F5:F1:E4:57:B7:E9:46:F4:86:04:28:F9:A7:ED:" @@ -287,7 +287,7 @@ static const char kSdpStringWithoutStreams[] = "a=mid:audio\r\n" "a=sendrecv\r\n" "a=rtcp-mux\r\n" - "a=rtpmap:103 ISAC/16000\r\n" + "a=rtpmap:111 OPUS/48000/2\r\n" "m=video 1 RTP/AVPF 120\r\n" "a=ice-ufrag:e5785931\r\n" "a=ice-pwd:36fb7878390db89481c1d46daa4278d8\r\n" @@ -305,7 +305,7 @@ static const char kSdpStringWithMsidWithoutStreams[] = "s=-\r\n" "t=0 0\r\n" "a=msid-semantic: WMS\r\n" - "m=audio 1 RTP/AVPF 103\r\n" + "m=audio 1 RTP/AVPF 111\r\n" "a=ice-ufrag:e5785931\r\n" "a=ice-pwd:36fb7878390db89481c1d46daa4278d8\r\n" "a=fingerprint:sha-256 58:AB:6E:F5:F1:E4:57:B7:E9:46:F4:86:04:28:F9:A7:ED:" @@ -313,7 +313,7 @@ static const char kSdpStringWithMsidWithoutStreams[] = "a=mid:audio\r\n" "a=sendrecv\r\n" "a=rtcp-mux\r\n" - "a=rtpmap:103 ISAC/16000\r\n" + "a=rtpmap:111 OPUS/48000/2\r\n" "m=video 1 RTP/AVPF 120\r\n" "a=ice-ufrag:e5785931\r\n" "a=ice-pwd:36fb7878390db89481c1d46daa4278d8\r\n" @@ -330,7 +330,7 @@ static const char kSdpStringWithoutStreamsAudioOnly[] = "o=- 0 0 IN IP4 127.0.0.1\r\n" "s=-\r\n" "t=0 0\r\n" - "m=audio 1 RTP/AVPF 103\r\n" + "m=audio 1 RTP/AVPF 111\r\n" "a=ice-ufrag:e5785931\r\n" "a=ice-pwd:36fb7878390db89481c1d46daa4278d8\r\n" "a=fingerprint:sha-256 58:AB:6E:F5:F1:E4:57:B7:E9:46:F4:86:04:28:F9:A7:ED:" @@ -338,7 +338,7 @@ static const char kSdpStringWithoutStreamsAudioOnly[] = "a=mid:audio\r\n" "a=sendrecv\r\n" "a=rtcp-mux\r\n" - "a=rtpmap:103 ISAC/16000\r\n"; + "a=rtpmap:111 OPUS/48000/2\r\n"; // Reference SENDONLY SDP without MediaStreams. Msid is not supported. static const char kSdpStringSendOnlyWithoutStreams[] = @@ -346,7 +346,7 @@ static const char kSdpStringSendOnlyWithoutStreams[] = "o=- 0 0 IN IP4 127.0.0.1\r\n" "s=-\r\n" "t=0 0\r\n" - "m=audio 1 RTP/AVPF 103\r\n" + "m=audio 1 RTP/AVPF 111\r\n" "a=ice-ufrag:e5785931\r\n" "a=ice-pwd:36fb7878390db89481c1d46daa4278d8\r\n" "a=fingerprint:sha-256 58:AB:6E:F5:F1:E4:57:B7:E9:46:F4:86:04:28:F9:A7:ED:" @@ -355,7 +355,7 @@ static const char kSdpStringSendOnlyWithoutStreams[] = "a=sendrecv\r\n" "a=sendonly\r\n" "a=rtcp-mux\r\n" - "a=rtpmap:103 ISAC/16000\r\n" + "a=rtpmap:111 OPUS/48000/2\r\n" "m=video 1 RTP/AVPF 120\r\n" "a=ice-ufrag:e5785931\r\n" "a=ice-pwd:36fb7878390db89481c1d46daa4278d8\r\n" @@ -375,7 +375,7 @@ static const char kSdpStringInit[] = "a=msid-semantic: WMS\r\n"; static const char kSdpStringAudio[] = - "m=audio 1 RTP/AVPF 103\r\n" + "m=audio 1 RTP/AVPF 111\r\n" "a=ice-ufrag:e5785931\r\n" "a=ice-pwd:36fb7878390db89481c1d46daa4278d8\r\n" "a=fingerprint:sha-256 58:AB:6E:F5:F1:E4:57:B7:E9:46:F4:86:04:28:F9:A7:ED:" @@ -383,7 +383,7 @@ static const char kSdpStringAudio[] = "a=mid:audio\r\n" "a=sendrecv\r\n" "a=rtcp-mux\r\n" - "a=rtpmap:103 ISAC/16000\r\n"; + "a=rtpmap:111 OPUS/48000/2\r\n"; static const char kSdpStringVideo[] = "m=video 1 RTP/AVPF 120\r\n" diff --git a/test/fuzzers/BUILD.gn b/test/fuzzers/BUILD.gn index 9896e07563..fd67372506 100644 --- a/test/fuzzers/BUILD.gn +++ b/test/fuzzers/BUILD.gn @@ -269,22 +269,6 @@ webrtc_fuzzer_test("audio_decoder_ilbc_fuzzer") { ] } -webrtc_fuzzer_test("audio_decoder_isac_fuzzer") { - sources = [ "audio_decoder_isac_fuzzer.cc" ] - deps = [ - ":audio_decoder_fuzzer", - "../../modules/audio_coding:isac", - ] -} - -webrtc_fuzzer_test("audio_decoder_isacfix_fuzzer") { - sources = [ "audio_decoder_isacfix_fuzzer.cc" ] - deps = [ - ":audio_decoder_fuzzer", - "../../modules/audio_coding:isac_fix", - ] -} - webrtc_fuzzer_test("audio_decoder_opus_fuzzer") { sources = [ "audio_decoder_opus_fuzzer.cc" ] deps = [ @@ -350,24 +334,6 @@ webrtc_fuzzer_test("audio_encoder_opus_fuzzer") { ] } -webrtc_fuzzer_test("audio_encoder_isac_fixed_fuzzer") { - sources = [ "audio_encoder_isac_fixed_fuzzer.cc" ] - deps = [ - ":audio_encoder_fuzzer", - "../../api/audio_codecs/isac:audio_encoder_isac_fix", - "../../rtc_base:checks", - ] -} - -webrtc_fuzzer_test("audio_encoder_isac_float_fuzzer") { - sources = [ "audio_encoder_isac_float_fuzzer.cc" ] - deps = [ - ":audio_encoder_fuzzer", - "../../api/audio_codecs/isac:audio_encoder_isac_float", - "../../rtc_base:checks", - ] -} - webrtc_fuzzer_test("turn_unwrap_fuzzer") { sources = [ "turn_unwrap_fuzzer.cc" ] deps = [ diff --git a/test/fuzzers/audio_decoder_isac_fuzzer.cc b/test/fuzzers/audio_decoder_isac_fuzzer.cc deleted file mode 100644 index 96fa75d6ce..0000000000 --- a/test/fuzzers/audio_decoder_isac_fuzzer.cc +++ /dev/null @@ -1,28 +0,0 @@ -/* - * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#include "modules/audio_coding/codecs/isac/main/include/audio_decoder_isac.h" -#include "test/fuzzers/audio_decoder_fuzzer.h" - -namespace webrtc { -void FuzzOneInput(const uint8_t* data, size_t size) { - if (size > 20000) { - return; - } - const int sample_rate_hz = size % 2 == 0 ? 16000 : 32000; // 16 or 32 kHz. - static const size_t kAllocatedOuputSizeSamples = 32000 / 10; // 100 ms. - int16_t output[kAllocatedOuputSizeSamples]; - AudioDecoderIsacFloatImpl::Config c; - c.sample_rate_hz = sample_rate_hz; - AudioDecoderIsacFloatImpl dec(c); - FuzzAudioDecoder(DecoderFunctionType::kNormalDecode, data, size, &dec, - sample_rate_hz, sizeof(output), output); -} -} // namespace webrtc diff --git a/test/fuzzers/audio_decoder_isacfix_fuzzer.cc b/test/fuzzers/audio_decoder_isacfix_fuzzer.cc deleted file mode 100644 index 08aa69feb4..0000000000 --- a/test/fuzzers/audio_decoder_isacfix_fuzzer.cc +++ /dev/null @@ -1,28 +0,0 @@ -/* - * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#include "modules/audio_coding/codecs/isac/fix/include/audio_decoder_isacfix.h" -#include "test/fuzzers/audio_decoder_fuzzer.h" - -namespace webrtc { -void FuzzOneInput(const uint8_t* data, size_t size) { - if (size > 20000) { - return; - } - static const int kSampleRateHz = 16000; - static const size_t kAllocatedOuputSizeSamples = 16000 / 10; // 100 ms. - int16_t output[kAllocatedOuputSizeSamples]; - AudioDecoderIsacFixImpl::Config c; - c.sample_rate_hz = kSampleRateHz; - AudioDecoderIsacFixImpl dec(c); - FuzzAudioDecoder(DecoderFunctionType::kNormalDecode, data, size, &dec, - kSampleRateHz, sizeof(output), output); -} -} // namespace webrtc diff --git a/test/fuzzers/audio_encoder_isac_fixed_fuzzer.cc b/test/fuzzers/audio_encoder_isac_fixed_fuzzer.cc deleted file mode 100644 index 5357dc1b3e..0000000000 --- a/test/fuzzers/audio_encoder_isac_fixed_fuzzer.cc +++ /dev/null @@ -1,26 +0,0 @@ -/* - * Copyright (c) 2018 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#include "api/audio_codecs/isac/audio_encoder_isac_fix.h" -#include "rtc_base/checks.h" -#include "test/fuzzers/audio_encoder_fuzzer.h" - -namespace webrtc { - -void FuzzOneInput(const uint8_t* data, size_t size) { - AudioEncoderIsacFix::Config config; - RTC_CHECK(config.IsOk()); - constexpr int kPayloadType = 100; - FuzzAudioEncoder( - /*data_view=*/{data, size}, - /*encoder=*/AudioEncoderIsacFix::MakeAudioEncoder(config, kPayloadType)); -} - -} // namespace webrtc diff --git a/test/fuzzers/audio_encoder_isac_float_fuzzer.cc b/test/fuzzers/audio_encoder_isac_float_fuzzer.cc deleted file mode 100644 index f9e2e0206d..0000000000 --- a/test/fuzzers/audio_encoder_isac_float_fuzzer.cc +++ /dev/null @@ -1,27 +0,0 @@ -/* - * Copyright (c) 2018 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#include "api/audio_codecs/isac/audio_encoder_isac_float.h" -#include "rtc_base/checks.h" -#include "test/fuzzers/audio_encoder_fuzzer.h" - -namespace webrtc { - -void FuzzOneInput(const uint8_t* data, size_t size) { - AudioEncoderIsacFloat::Config config; - config.sample_rate_hz = 16000; - RTC_CHECK(config.IsOk()); - constexpr int kPayloadType = 100; - FuzzAudioEncoder(/*data_view=*/{data, size}, - /*encoder=*/AudioEncoderIsacFloat::MakeAudioEncoder( - config, kPayloadType)); -} - -} // namespace webrtc