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Move AudioDecoder and related stuff to the api/ directory

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BUG=webrtc:5805, webrtc:6725

Review-Url: https://codereview.webrtc.org/2668523004
Cr-Commit-Position: refs/heads/master@{#16534}
This commit is contained in:
kwiberg
2017-02-10 08:15:44 -08:00
committed by Commit bot
parent 84a3759825
commit 087bd34d23
78 changed files with 535 additions and 380 deletions
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126
webrtc/modules/audio_coding/BUILD.gn
View File

@ -39,52 +39,27 @@ audio_coding_deps = audio_codec_deps + [
"../../system_wrappers",
]
rtc_static_library("audio_format") {
sources = [
"codecs/audio_format.cc",
"codecs/audio_format.h",
]
deps = [
"../..:webrtc_common",
]
}
rtc_static_library("audio_format_conversion") {
sources = [
"codecs/audio_format_conversion.cc",
"codecs/audio_format_conversion.h",
]
deps = [
":audio_format",
"../..:webrtc_common",
"../../api/audio_codecs:audio_codecs_api",
"../../base:rtc_base_approved",
]
}
rtc_source_set("audio_decoder_factory_interface") {
rtc_static_library("builtin_audio_decoder_factory_internal") {
sources = [
"codecs/audio_decoder_factory.h",
]
deps = [
":audio_decoder_interface",
":audio_format",
# TODO(charujain): Clean this dependency when downstream projects are
# updated to properly depend on audio_format_conversion target.
":audio_format_conversion",
"../../base:rtc_base_approved",
]
}
rtc_static_library("builtin_audio_decoder_factory") {
sources = [
"codecs/builtin_audio_decoder_factory.cc",
"codecs/builtin_audio_decoder_factory.h",
"codecs/builtin_audio_decoder_factory_internal.cc",
"codecs/builtin_audio_decoder_factory_internal.h",
]
deps = [
"../..:webrtc_common",
"../../base:rtc_base_approved",
":audio_decoder_factory_interface",
"../../api/audio_codecs:audio_codecs_api",
] + audio_codec_deps
defines = audio_codec_defines
}
@ -101,7 +76,7 @@ rtc_static_library("rent_a_codec") {
"acm2/rent_a_codec.h",
]
deps = [
":audio_decoder_interface",
"../../api/audio_codecs:audio_codecs_api",
"../..:webrtc_common",
"../../base:rtc_base_approved",
] + audio_codec_deps
@ -149,9 +124,8 @@ rtc_static_library("audio_coding") {
}
deps = audio_coding_deps + [
":audio_decoder_interface",
":audio_decoder_factory_interface",
":builtin_audio_decoder_factory",
"../../api/audio_codecs:audio_codecs_api",
"../../api/audio_codecs:builtin_audio_decoder_factory",
":neteq",
":rent_a_codec",
"../../base:rtc_base_approved",
@ -160,15 +134,13 @@ rtc_static_library("audio_coding") {
defines = audio_coding_defines
}
rtc_static_library("audio_decoder_interface") {
rtc_static_library("legacy_encoded_audio_frame") {
sources = [
"codecs/audio_decoder.cc",
"codecs/audio_decoder.h",
"codecs/legacy_encoded_audio_frame.cc",
"codecs/legacy_encoded_audio_frame.h",
]
deps = [
"../..:webrtc_common",
"../../api/audio_codecs:audio_codecs_api",
"../../base:rtc_base_approved",
]
}
@ -246,9 +218,10 @@ rtc_static_library("g711") {
public_configs = [ ":g711_config" ]
deps = [
":audio_decoder_interface",
":audio_encoder_interface",
":legacy_encoded_audio_frame",
"../..:webrtc_common",
"../../api/audio_codecs:audio_codecs_api",
"../../base:rtc_base_approved",
]
public_deps = [
@ -287,9 +260,10 @@ rtc_static_library("g722") {
public_configs = [ ":g722_config" ]
deps = [
":audio_decoder_interface",
":audio_encoder_interface",
":legacy_encoded_audio_frame",
"../..:webrtc_common",
"../../api/audio_codecs:audio_codecs_api",
"../../base:rtc_base_approved",
]
public_deps = [
@ -329,9 +303,10 @@ rtc_static_library("ilbc") {
public_configs = [ ":ilbc_config" ]
deps = [
":audio_decoder_interface",
":audio_encoder_interface",
":legacy_encoded_audio_frame",
"../..:webrtc_common",
"../../api/audio_codecs:audio_codecs_api",
"../../base:rtc_base_approved",
"../../common_audio",
]
@ -487,9 +462,9 @@ rtc_source_set("ilbc_c") {
public_configs = [ ":ilbc_config" ]
deps = [
":audio_decoder_interface",
":audio_encoder_interface",
"../..:webrtc_common",
"../../api/audio_codecs:audio_codecs_api",
"../../base:rtc_base_approved",
"../../common_audio",
]
@ -525,9 +500,9 @@ rtc_static_library("isac") {
]
deps = [
":audio_decoder_interface",
":audio_encoder_interface",
":isac_common",
"../../api/audio_codecs:audio_codecs_api",
]
public_deps = [
":isac_c",
@ -619,9 +594,9 @@ rtc_static_library("isac_fix") {
public_configs = [ ":isac_fix_config" ]
deps = [
":audio_decoder_interface",
":audio_encoder_interface",
":isac_common",
"../../api/audio_codecs:audio_codecs_api",
"../../common_audio",
"../../system_wrappers",
]
@ -695,10 +670,10 @@ rtc_source_set("isac_fix_c") {
public_configs = [ ":isac_fix_config" ]
deps = [
":audio_decoder_interface",
":audio_encoder_interface",
":isac_common",
"../..:webrtc_common",
"../../api/audio_codecs:audio_codecs_api",
"../../base:rtc_base_approved",
"../../common_audio",
"../../system_wrappers",
@ -799,10 +774,11 @@ rtc_static_library("pcm16b") {
]
deps = [
":audio_decoder_interface",
":audio_encoder_interface",
":g711",
":legacy_encoded_audio_frame",
"../..:webrtc_common",
"../../api/audio_codecs:audio_codecs_api",
"../../base:rtc_base_approved",
]
public_deps = [
@ -837,10 +813,10 @@ rtc_static_library("webrtc_opus") {
]
deps = [
":audio_decoder_interface",
":audio_encoder_interface",
":audio_network_adaptor",
"../..:webrtc_common",
"../../api/audio_codecs:audio_codecs_api",
"../../base:rtc_base_approved",
"../../base:rtc_numerics",
"../../common_audio",
@ -1031,16 +1007,13 @@ rtc_static_library("neteq") {
]
deps = [
":audio_decoder_factory_interface",
":audio_decoder_interface",
":audio_format",
":builtin_audio_decoder_factory",
":cng",
":g711",
":isac_fix",
":pcm16b",
":rent_a_codec",
"../..:webrtc_common",
"../../api/audio_codecs:audio_codecs_api",
"../../base:gtest_prod",
"../../base:rtc_base_approved",
"../../common_audio",
@ -1095,9 +1068,9 @@ rtc_source_set("neteq_test_minimal") {
deps = [
":audio_encoder_interface",
":builtin_audio_decoder_factory",
":neteq",
"../..:webrtc_common",
"../../api/audio_codecs:builtin_audio_decoder_factory",
"../../base:rtc_base_approved",
]
}
@ -1161,9 +1134,9 @@ if (rtc_include_tests) {
deps = [
":audio_coding",
":audio_format_conversion",
":builtin_audio_decoder_factory",
":pcm16b_c",
"../..:webrtc_common",
"../../api/audio_codecs:builtin_audio_decoder_factory",
"../../base:rtc_base_approved",
"../../system_wrappers:system_wrappers",
"../../test:fileutils",
@ -1215,8 +1188,8 @@ if (rtc_include_tests) {
deps = audio_coding_deps + [
":audio_coding",
":audio_format_conversion",
":audio_decoder_factory_interface",
":builtin_audio_decoder_factory",
"../../api/audio_codecs:audio_codecs_api",
"../../api/audio_codecs:builtin_audio_decoder_factory",
":neteq_unittest_tools",
"../../base:rtc_base_approved",
"../../test:test_support",
@ -1235,7 +1208,7 @@ if (rtc_include_tests) {
deps = audio_coding_deps + [
":audio_coding",
":audio_decoder_interface",
"../../api/audio_codecs:audio_codecs_api",
":audio_encoder_interface",
":neteq_unittest_tools",
"../../base:rtc_base_approved",
@ -1324,12 +1297,12 @@ if (rtc_include_tests) {
deps += audio_coding_deps
deps += [
":audio_decoder_interface",
":ilbc",
":isac",
":isac_fix",
":neteq",
":neteq_unittest_tools",
"../../api/audio_codecs:audio_codecs_api",
"../../common_audio",
"../../test:test_main",
"//testing/gtest",
@ -1460,12 +1433,12 @@ if (rtc_include_tests) {
}
deps = [
":audio_decoder_interface",
":builtin_audio_decoder_factory",
":neteq",
":neteq_unittest_tools",
":pcm16b",
"../..:webrtc_common",
"../../api/audio_codecs:audio_codecs_api",
"../../api/audio_codecs:builtin_audio_decoder_factory",
"../../base:rtc_base_approved",
"../../system_wrappers",
"../../test:test_support",
@ -1486,10 +1459,10 @@ if (rtc_include_tests) {
}
deps = [
":builtin_audio_decoder_factory",
":neteq",
":neteq_unittest_tools",
"../..:webrtc_common",
"../../api/audio_codecs:builtin_audio_decoder_factory",
"../../base:rtc_base_approved",
"../../test:test_support",
"//testing/gtest",
@ -1540,10 +1513,10 @@ if (rtc_include_tests) {
}
deps = [
":audio_decoder_interface",
":audio_encoder_interface",
":pcm16b",
"../..:webrtc_common",
"../../api/audio_codecs:audio_codecs_api",
"../../base:rtc_base_approved",
"../../common_audio",
"../../test:rtp_test_utils",
@ -1999,7 +1972,7 @@ if (rtc_include_tests) {
"audio_network_adaptor/frame_length_controller_unittest.cc",
"audio_network_adaptor/mock/mock_controller.h",
"audio_network_adaptor/mock/mock_controller_manager.h",
"codecs/audio_decoder_factory_unittest.cc",
"codecs/builtin_audio_decoder_factory_unittest.cc",
"codecs/cng/audio_encoder_cng_unittest.cc",
"codecs/cng/cng_unittest.cc",
"codecs/ilbc/ilbc_unittest.cc",
@ -2063,17 +2036,16 @@ if (rtc_include_tests) {
":acm_receive_test",
":acm_send_test",
":audio_coding",
":audio_decoder_interface",
":audio_encoder_interface",
":audio_format_conversion",
":audio_network_adaptor",
":builtin_audio_decoder_factory",
":cng",
":g711",
":ilbc",
":isac",
":isac_c",
":isac_fix",
":legacy_encoded_audio_frame",
":neteq",
":neteq_test_support",
":neteq_unittest_tools",
@ -2082,6 +2054,8 @@ if (rtc_include_tests) {
":rent_a_codec",
":webrtc_opus",
"../..:webrtc_common",
"../../api/audio_codecs:audio_codecs_api",
"../../api/audio_codecs:builtin_audio_decoder_factory",
"../../base:rtc_base",
"../../base:rtc_base_approved",
"../../base:rtc_base_tests_utils",
@ -2115,3 +2089,27 @@ if (rtc_include_tests) {
}
}
}
# For backwards compatibility only! Use
# webrtc/api/audio_codecs:audio_codecs_api instead.
# TODO(kwiberg): Remove this.
rtc_source_set("audio_decoder_interface") {
sources = [
"codecs/audio_decoder.h",
]
deps = [
"../../api/audio_codecs:audio_codecs_api",
]
}
# For backwards compatibility only! Use
# webrtc/api/audio_codecs:builtin_audio_decoder_factory instead.
# TODO(kwiberg): Remove this.
rtc_source_set("builtin_audio_decoder_factory") {
sources = [
"codecs/builtin_audio_decoder_factory.h",
]
deps = [
"../../api/audio_codecs:builtin_audio_decoder_factory",
]
}

2
webrtc/modules/audio_coding/acm2/acm_receive_test.cc
View File

@ -15,8 +15,8 @@
#include <memory>
#include "webrtc/api/audio_codecs/builtin_audio_decoder_factory.h"
#include "webrtc/modules/audio_coding/codecs/audio_format_conversion.h"
#include "webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h"
#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
#include "webrtc/modules/audio_coding/neteq/tools/audio_sink.h"
#include "webrtc/modules/audio_coding/neteq/tools/packet.h"

2
webrtc/modules/audio_coding/acm2/acm_receive_test.h
View File

@ -14,9 +14,9 @@
#include <memory>
#include <string>
#include "webrtc/api/audio_codecs/audio_decoder_factory.h"
#include "webrtc/base/constructormagic.h"
#include "webrtc/base/scoped_ref_ptr.h"
#include "webrtc/modules/audio_coding/codecs/audio_decoder_factory.h"
#include "webrtc/system_wrappers/include/clock.h"
namespace webrtc {

2
webrtc/modules/audio_coding/acm2/acm_receiver.cc
View File

@ -15,13 +15,13 @@
#include <algorithm> // sort
#include <vector>
#include "webrtc/api/audio_codecs/audio_decoder.h"
#include "webrtc/base/checks.h"
#include "webrtc/base/format_macros.h"
#include "webrtc/base/logging.h"
#include "webrtc/base/safe_conversions.h"
#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
#include "webrtc/common_types.h"
#include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
#include "webrtc/modules/audio_coding/acm2/acm_resampler.h"
#include "webrtc/modules/audio_coding/acm2/call_statistics.h"
#include "webrtc/modules/audio_coding/neteq/include/neteq.h"

2
webrtc/modules/audio_coding/acm2/acm_receiver_unittest.cc
View File

@ -13,9 +13,9 @@
#include <algorithm> // std::min
#include <memory>
#include "webrtc/api/audio_codecs/builtin_audio_decoder_factory.h"
#include "webrtc/base/checks.h"
#include "webrtc/base/safe_conversions.h"
#include "webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h"
#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
#include "webrtc/modules/audio_coding/neteq/tools/rtp_generator.h"
#include "webrtc/system_wrappers/include/clock.h"

2
webrtc/modules/audio_coding/acm2/audio_coding_module.cc
View File

@ -10,13 +10,13 @@
#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
#include "webrtc/api/audio_codecs/builtin_audio_decoder_factory.h"
#include "webrtc/base/checks.h"
#include "webrtc/base/safe_conversions.h"
#include "webrtc/modules/audio_coding/acm2/acm_receiver.h"
#include "webrtc/modules/audio_coding/acm2/acm_resampler.h"
#include "webrtc/modules/audio_coding/acm2/codec_manager.h"
#include "webrtc/modules/audio_coding/acm2/rent_a_codec.h"
#include "webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h"
#include "webrtc/system_wrappers/include/metrics.h"
#include "webrtc/system_wrappers/include/trace.h"

2
webrtc/modules/audio_coding/acm2/audio_coding_module_unittest.cc
View File

@ -13,13 +13,13 @@
#include <memory>
#include <vector>
#include "webrtc/api/audio_codecs/builtin_audio_decoder_factory.h"
#include "webrtc/base/criticalsection.h"
#include "webrtc/base/md5digest.h"
#include "webrtc/base/platform_thread.h"
#include "webrtc/base/thread_annotations.h"
#include "webrtc/modules/audio_coding/acm2/acm_receive_test.h"
#include "webrtc/modules/audio_coding/acm2/acm_send_test.h"
#include "webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h"
#include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
#include "webrtc/modules/audio_coding/codecs/audio_format_conversion.h"
#include "webrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.h"

4
webrtc/modules/audio_coding/acm2/rent_a_codec.h
View File

@ -15,12 +15,12 @@
#include <map>
#include <memory>
#include "webrtc/api/audio_codecs/audio_decoder.h"
#include "webrtc/api/audio_codecs/audio_format.h"
#include "webrtc/base/array_view.h"
#include "webrtc/base/constructormagic.h"
#include "webrtc/base/optional.h"
#include "webrtc/base/scoped_ref_ptr.h"
#include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
#include "webrtc/modules/audio_coding/codecs/audio_format.h"
#include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
#include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h"
#include "webrtc/typedefs.h"

129
webrtc/modules/audio_coding/codecs/audio_decoder.cc
View File

@ -1,129 +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 "webrtc/modules/audio_coding/codecs/audio_decoder.h"
#include <assert.h>
#include <memory>
#include <utility>
#include "webrtc/base/array_view.h"
#include "webrtc/base/checks.h"
#include "webrtc/base/sanitizer.h"
#include "webrtc/base/trace_event.h"
#include "webrtc/modules/audio_coding/codecs/legacy_encoded_audio_frame.h"
namespace webrtc {
AudioDecoder::ParseResult::ParseResult() = default;
AudioDecoder::ParseResult::ParseResult(ParseResult&& b) = default;
AudioDecoder::ParseResult::ParseResult(uint32_t timestamp,
int priority,
std::unique_ptr<EncodedAudioFrame> frame)
: timestamp(timestamp), priority(priority), frame(std::move(frame)) {
RTC_DCHECK_GE(priority, 0);
}
AudioDecoder::ParseResult::~ParseResult() = default;
AudioDecoder::ParseResult& AudioDecoder::ParseResult::operator=(
ParseResult&& b) = default;
std::vector<AudioDecoder::ParseResult> AudioDecoder::ParsePayload(
rtc::Buffer&& payload,
uint32_t timestamp) {
std::vector<ParseResult> results;
std::unique_ptr<EncodedAudioFrame> frame(
new LegacyEncodedAudioFrame(this, std::move(payload)));
results.emplace_back(timestamp, 0, std::move(frame));
return results;
}
int AudioDecoder::Decode(const uint8_t* encoded, size_t encoded_len,
int sample_rate_hz, size_t max_decoded_bytes,
int16_t* decoded, SpeechType* speech_type) {
TRACE_EVENT0("webrtc", "AudioDecoder::Decode");
rtc::MsanCheckInitialized(rtc::MakeArrayView(encoded, encoded_len));
int duration = PacketDuration(encoded, encoded_len);
if (duration >= 0 &&
duration * Channels() * sizeof(int16_t) > max_decoded_bytes) {
return -1;
}
return DecodeInternal(encoded, encoded_len, sample_rate_hz, decoded,
speech_type);
}
int AudioDecoder::DecodeRedundant(const uint8_t* encoded, size_t encoded_len,
int sample_rate_hz, size_t max_decoded_bytes,
int16_t* decoded, SpeechType* speech_type) {
TRACE_EVENT0("webrtc", "AudioDecoder::DecodeRedundant");
rtc::MsanCheckInitialized(rtc::MakeArrayView(encoded, encoded_len));
int duration = PacketDurationRedundant(encoded, encoded_len);
if (duration >= 0 &&
duration * Channels() * sizeof(int16_t) > max_decoded_bytes) {
return -1;
}
return DecodeRedundantInternal(encoded, encoded_len, sample_rate_hz, decoded,
speech_type);
}
int AudioDecoder::DecodeRedundantInternal(const uint8_t* encoded,
size_t encoded_len,
int sample_rate_hz, int16_t* decoded,
SpeechType* speech_type) {
return DecodeInternal(encoded, encoded_len, sample_rate_hz, decoded,
speech_type);
}
bool AudioDecoder::HasDecodePlc() const { return false; }
size_t AudioDecoder::DecodePlc(size_t num_frames, int16_t* decoded) {
return 0;
}
int AudioDecoder::IncomingPacket(const uint8_t* payload,
size_t payload_len,
uint16_t rtp_sequence_number,
uint32_t rtp_timestamp,
uint32_t arrival_timestamp) {
return 0;
}
int AudioDecoder::ErrorCode() { return 0; }
int AudioDecoder::PacketDuration(const uint8_t* encoded,
size_t encoded_len) const {
return kNotImplemented;
}
int AudioDecoder::PacketDurationRedundant(const uint8_t* encoded,
size_t encoded_len) const {
return kNotImplemented;
}
bool AudioDecoder::PacketHasFec(const uint8_t* encoded,
size_t encoded_len) const {
return false;
}
AudioDecoder::SpeechType AudioDecoder::ConvertSpeechType(int16_t type) {
switch (type) {
case 0: // TODO(hlundin): Both iSAC and Opus return 0 for speech.
case 1:
return kSpeech;
case 2:
return kComfortNoise;
default:
assert(false);
return kSpeech;
}
}
} // namespace webrtc

169
webrtc/modules/audio_coding/codecs/audio_decoder.h
View File

@ -8,172 +8,13 @@
* be found in the AUTHORS file in the root of the source tree.
*/
// This file is for backwards compatibility only! Use
// webrtc/api/audio_codecs/audio_decoder.h instead!
// TODO(kwiberg): Remove it.
#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_DECODER_H_
#define WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_DECODER_H_
#include <memory>
#include <vector>
#include "webrtc/api/audio_codecs/audio_decoder.h"
#include "webrtc/base/array_view.h"
#include "webrtc/base/buffer.h"
#include "webrtc/base/constructormagic.h"
#include "webrtc/base/optional.h"
#include "webrtc/typedefs.h"
namespace webrtc {
// This is the interface class for decoders in NetEQ. Each codec type will have
// and implementation of this class.
class AudioDecoder {
public:
enum SpeechType {
kSpeech = 1,
kComfortNoise = 2
};
// Used by PacketDuration below. Save the value -1 for errors.
enum { kNotImplemented = -2 };
AudioDecoder() = default;
virtual ~AudioDecoder() = default;
class EncodedAudioFrame {
public:
struct DecodeResult {
size_t num_decoded_samples;
SpeechType speech_type;
};
virtual ~EncodedAudioFrame() = default;
// Returns the duration in samples-per-channel of this audio frame.
// If no duration can be ascertained, returns zero.
virtual size_t Duration() const = 0;
// Decodes this frame of audio and writes the result in |decoded|.
// |decoded| must be large enough to store as many samples as indicated by a
// call to Duration() . On success, returns an rtc::Optional containing the
// total number of samples across all channels, as well as whether the
// decoder produced comfort noise or speech. On failure, returns an empty
// rtc::Optional. Decode may be called at most once per frame object.
virtual rtc::Optional<DecodeResult> Decode(
rtc::ArrayView<int16_t> decoded) const = 0;
};
struct ParseResult {
ParseResult();
ParseResult(uint32_t timestamp,
int priority,
std::unique_ptr<EncodedAudioFrame> frame);
ParseResult(ParseResult&& b);
~ParseResult();
ParseResult& operator=(ParseResult&& b);
// The timestamp of the frame is in samples per channel.
uint32_t timestamp;
// The relative priority of the frame compared to other frames of the same
// payload and the same timeframe. A higher value means a lower priority.
// The highest priority is zero - negative values are not allowed.
int priority;
std::unique_ptr<EncodedAudioFrame> frame;
};
// Let the decoder parse this payload and prepare zero or more decodable
// frames. Each frame must be between 10 ms and 120 ms long. The caller must
// ensure that the AudioDecoder object outlives any frame objects returned by
// this call. The decoder is free to swap or move the data from the |payload|
// buffer. |timestamp| is the input timestamp, in samples, corresponding to
// the start of the payload.
virtual std::vector<ParseResult> ParsePayload(rtc::Buffer&& payload,
uint32_t timestamp);
// Decodes |encode_len| bytes from |encoded| and writes the result in
// |decoded|. The maximum bytes allowed to be written into |decoded| is
// |max_decoded_bytes|. Returns the total number of samples across all
// channels. If the decoder produced comfort noise, |speech_type|
// is set to kComfortNoise, otherwise it is kSpeech. The desired output
// sample rate is provided in |sample_rate_hz|, which must be valid for the
// codec at hand.
int Decode(const uint8_t* encoded,
size_t encoded_len,
int sample_rate_hz,
size_t max_decoded_bytes,
int16_t* decoded,
SpeechType* speech_type);
// Same as Decode(), but interfaces to the decoders redundant decode function.
// The default implementation simply calls the regular Decode() method.
int DecodeRedundant(const uint8_t* encoded,
size_t encoded_len,
int sample_rate_hz,
size_t max_decoded_bytes,
int16_t* decoded,
SpeechType* speech_type);
// Indicates if the decoder implements the DecodePlc method.
virtual bool HasDecodePlc() const;
// Calls the packet-loss concealment of the decoder to update the state after
// one or several lost packets. The caller has to make sure that the
// memory allocated in |decoded| should accommodate |num_frames| frames.
virtual size_t DecodePlc(size_t num_frames, int16_t* decoded);
// Resets the decoder state (empty buffers etc.).
virtual void Reset() = 0;
// Notifies the decoder of an incoming packet to NetEQ.
virtual int IncomingPacket(const uint8_t* payload,
size_t payload_len,
uint16_t rtp_sequence_number,
uint32_t rtp_timestamp,
uint32_t arrival_timestamp);
// Returns the last error code from the decoder.
virtual int ErrorCode();
// Returns the duration in samples-per-channel of the payload in |encoded|
// which is |encoded_len| bytes long. Returns kNotImplemented if no duration
// estimate is available, or -1 in case of an error.
virtual int PacketDuration(const uint8_t* encoded, size_t encoded_len) const;
// Returns the duration in samples-per-channel of the redandant payload in
// |encoded| which is |encoded_len| bytes long. Returns kNotImplemented if no
// duration estimate is available, or -1 in case of an error.
virtual int PacketDurationRedundant(const uint8_t* encoded,
size_t encoded_len) const;
// Detects whether a packet has forward error correction. The packet is
// comprised of the samples in |encoded| which is |encoded_len| bytes long.
// Returns true if the packet has FEC and false otherwise.
virtual bool PacketHasFec(const uint8_t* encoded, size_t encoded_len) const;
// Returns the actual sample rate of the decoder's output. This value may not
// change during the lifetime of the decoder.
virtual int SampleRateHz() const = 0;
// The number of channels in the decoder's output. This value may not change
// during the lifetime of the decoder.
virtual size_t Channels() const = 0;
protected:
static SpeechType ConvertSpeechType(int16_t type);
virtual int DecodeInternal(const uint8_t* encoded,
size_t encoded_len,
int sample_rate_hz,
int16_t* decoded,
SpeechType* speech_type) = 0;
virtual int DecodeRedundantInternal(const uint8_t* encoded,
size_t encoded_len,
int sample_rate_hz,
int16_t* decoded,
SpeechType* speech_type);
private:
RTC_DISALLOW_COPY_AND_ASSIGN(AudioDecoder);
};
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_DECODER_H_

38
webrtc/modules/audio_coding/codecs/audio_decoder_factory.h
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@ -1,38 +0,0 @@
/*
* Copyright (c) 2016 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 WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_DECODER_FACTORY_H_
#define WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_DECODER_FACTORY_H_
#include <memory>
#include <vector>
#include "webrtc/base/atomicops.h"
#include "webrtc/base/refcount.h"
#include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
#include "webrtc/modules/audio_coding/codecs/audio_format.h"
namespace webrtc {
// A factory that creates AudioDecoders.
// NOTE: This class is still under development and may change without notice.
class AudioDecoderFactory : public rtc::RefCountInterface {
public:
virtual std::vector<AudioCodecSpec> GetSupportedDecoders() = 0;
virtual bool IsSupportedDecoder(const SdpAudioFormat& format) = 0;
virtual std::unique_ptr<AudioDecoder> MakeAudioDecoder(
const SdpAudioFormat& format) = 0;
};
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_DECODER_FACTORY_H_

86
webrtc/modules/audio_coding/codecs/audio_format.cc
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@ -1,86 +0,0 @@
/*
* Copyright (c) 2016 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 "webrtc/modules/audio_coding/codecs/audio_format.h"
#include "webrtc/common_types.h"
namespace webrtc {
SdpAudioFormat::SdpAudioFormat(const SdpAudioFormat&) = default;
SdpAudioFormat::SdpAudioFormat(SdpAudioFormat&&) = default;
SdpAudioFormat::SdpAudioFormat(const char* name,
int clockrate_hz,
int num_channels)
: name(name), clockrate_hz(clockrate_hz), num_channels(num_channels) {}
SdpAudioFormat::SdpAudioFormat(const std::string& name,
int clockrate_hz,
int num_channels)
: name(name), clockrate_hz(clockrate_hz), num_channels(num_channels) {}
SdpAudioFormat::SdpAudioFormat(const char* name,
int clockrate_hz,
int num_channels,
const Parameters& param)
: name(name),
clockrate_hz(clockrate_hz),
num_channels(num_channels),
parameters(param) {}
SdpAudioFormat::SdpAudioFormat(const std::string& name,
int clockrate_hz,
int num_channels,
const Parameters& param)
: name(name),
clockrate_hz(clockrate_hz),
num_channels(num_channels),
parameters(param) {}
SdpAudioFormat::~SdpAudioFormat() = default;
SdpAudioFormat& SdpAudioFormat::operator=(const SdpAudioFormat&) = default;
SdpAudioFormat& SdpAudioFormat::operator=(SdpAudioFormat&&) = default;
bool operator==(const SdpAudioFormat& a, const SdpAudioFormat& b) {
return STR_CASE_CMP(a.name.c_str(), b.name.c_str()) == 0 &&
a.clockrate_hz == b.clockrate_hz && a.num_channels == b.num_channels &&
a.parameters == b.parameters;
}
void swap(SdpAudioFormat& a, SdpAudioFormat& b) {
using std::swap;
swap(a.name, b.name);
swap(a.clockrate_hz, b.clockrate_hz);
swap(a.num_channels, b.num_channels);
swap(a.parameters, b.parameters);
}
std::ostream& operator<<(std::ostream& os, const SdpAudioFormat& saf) {
os << "{name: " << saf.name;
os << ", clockrate_hz: " << saf.clockrate_hz;
os << ", num_channels: " << saf.num_channels;
os << ", parameters: {";
const char* sep = "";
for (const auto& kv : saf.parameters) {
os << sep << kv.first << ": " << kv.second;
sep = ", ";
}
os << "}}";
return os;
}
AudioCodecSpec::AudioCodecSpec(const SdpAudioFormat& format)
: format(format) {}
AudioCodecSpec::AudioCodecSpec(SdpAudioFormat&& format)
: format(std::move(format)) {}
} // namespace webrtc

81
webrtc/modules/audio_coding/codecs/audio_format.h
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@ -1,81 +0,0 @@
/*
* Copyright (c) 2016 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 WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_FORMAT_H_
#define WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_FORMAT_H_
#include <map>
#include <ostream>
#include <string>
#include <utility>
namespace webrtc {
// SDP specification for a single audio codec.
// NOTE: This class is still under development and may change without notice.
struct SdpAudioFormat {
using Parameters = std::map<std::string, std::string>;
SdpAudioFormat(const SdpAudioFormat&);
SdpAudioFormat(SdpAudioFormat&&);
SdpAudioFormat(const char* name, int clockrate_hz, int num_channels);
SdpAudioFormat(const std::string& name, int clockrate_hz, int num_channels);
SdpAudioFormat(const char* name,
int clockrate_hz,
int num_channels,
const Parameters& param);
SdpAudioFormat(const std::string& name,
int clockrate_hz,
int num_channels,
const Parameters& param);
~SdpAudioFormat();
SdpAudioFormat& operator=(const SdpAudioFormat&);
SdpAudioFormat& operator=(SdpAudioFormat&&);
friend bool operator==(const SdpAudioFormat& a, const SdpAudioFormat& b);
friend bool operator!=(const SdpAudioFormat& a, const SdpAudioFormat& b) {
return !(a == b);
}
std::string name;
int clockrate_hz;
int num_channels;
Parameters parameters;
};
void swap(SdpAudioFormat& a, SdpAudioFormat& b);
std::ostream& operator<<(std::ostream& os, const SdpAudioFormat& saf);
// To avoid API breakage, and make the code clearer, AudioCodecSpec should not
// be directly initializable with any flags indicating optional support. If it
// were, these initializers would break any time a new flag was added. It's also
// more difficult to understand:
// AudioCodecSpec spec{{"format", 8000, 1}, true, false, false, true, true};
// than
// AudioCodecSpec spec({"format", 8000, 1});
// spec.allow_comfort_noise = true;
// spec.future_flag_b = true;
// spec.future_flag_c = true;
struct AudioCodecSpec {
explicit AudioCodecSpec(const SdpAudioFormat& format);
explicit AudioCodecSpec(SdpAudioFormat&& format);
~AudioCodecSpec() = default;
SdpAudioFormat format;
bool allow_comfort_noise = true; // This codec can be used with an external
// comfort noise generator.
bool supports_network_adaption = false; // This codec can adapt to varying
// network conditions.
};
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_FORMAT_H_

2
webrtc/modules/audio_coding/codecs/audio_format_conversion.h
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@ -11,8 +11,8 @@
#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_FORMAT_CONVERSION_H_
#define WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_FORMAT_CONVERSION_H_
#include "webrtc/api/audio_codecs/audio_format.h"
#include "webrtc/common_types.h"
#include "webrtc/modules/audio_coding/codecs/audio_format.h"
namespace webrtc {

17
webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h
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@ -8,20 +8,13 @@
* be found in the AUTHORS file in the root of the source tree.
*/
// This file is for backwards compatibility only! Use
// webrtc/api/audio_codecs/builtin_audio_decoder_factory.h instead!
// TODO(kwiberg): Remove it.
#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_BUILTIN_AUDIO_DECODER_FACTORY_H_
#define WEBRTC_MODULES_AUDIO_CODING_CODECS_BUILTIN_AUDIO_DECODER_FACTORY_H_
#include <memory>
#include "webrtc/base/scoped_ref_ptr.h"
#include "webrtc/modules/audio_coding/codecs/audio_decoder_factory.h"
namespace webrtc {
// Creates a new factory that can create the built-in types of audio decoders.
// NOTE: This function is still under development and may change without notice.
rtc::scoped_refptr<AudioDecoderFactory> CreateBuiltinAudioDecoderFactory();
} // namespace webrtc
#include "webrtc/api/audio_codecs/builtin_audio_decoder_factory.h"
#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_BUILTIN_AUDIO_DECODER_FACTORY_H_

43
webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.cc → webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory_internal.cc
View File

@ -8,8 +8,9 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#include "webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h"
#include "webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory_internal.h"
#include <memory>
#include <vector>
#include "webrtc/base/checks.h"
@ -176,34 +177,35 @@ class BuiltinAudioDecoderFactory : public AudioDecoderFactory {
std::vector<AudioCodecSpec> GetSupportedDecoders() override {
// Although this looks a bit strange, it means specs need only be initalized
// once, and that that initialization is thread-safe.
static std::vector<AudioCodecSpec> specs =
[]{
std::vector<AudioCodecSpec> specs;
static std::vector<AudioCodecSpec> specs = [] {
std::vector<AudioCodecSpec> specs;
#ifdef WEBRTC_CODEC_OPUS
AudioCodecSpec opus({"opus", 48000, 2, {
{"minptime", "10"},
{"useinbandfec", "1"}
}});
opus.allow_comfort_noise = false;
opus.supports_network_adaption = true;
specs.push_back(opus);
// clang-format off
AudioCodecSpec opus({"opus", 48000, 2, {
{"minptime", "10"},
{"useinbandfec", "1"}
}});
// clang-format on
opus.allow_comfort_noise = false;
opus.supports_network_adaption = true;
specs.push_back(opus);
#endif
#if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX))
specs.push_back(AudioCodecSpec({"isac", 16000, 1}));
specs.push_back(AudioCodecSpec({"isac", 16000, 1}));
#endif
#if (defined(WEBRTC_CODEC_ISAC))
specs.push_back(AudioCodecSpec({"isac", 32000, 1}));
specs.push_back(AudioCodecSpec({"isac", 32000, 1}));
#endif
#ifdef WEBRTC_CODEC_G722
specs.push_back(AudioCodecSpec({"G722", 8000, 1}));
specs.push_back(AudioCodecSpec({"G722", 8000, 1}));
#endif
#ifdef WEBRTC_CODEC_ILBC
specs.push_back(AudioCodecSpec({"iLBC", 8000, 1}));
specs.push_back(AudioCodecSpec({"iLBC", 8000, 1}));
#endif
specs.push_back(AudioCodecSpec({"PCMU", 8000, 1}));
specs.push_back(AudioCodecSpec({"PCMA", 8000, 1}));
return specs;
}();
specs.push_back(AudioCodecSpec({"PCMU", 8000, 1}));
specs.push_back(AudioCodecSpec({"PCMA", 8000, 1}));
return specs;
}();
return specs;
}
@ -239,7 +241,8 @@ class BuiltinAudioDecoderFactory : public AudioDecoderFactory {
} // namespace
rtc::scoped_refptr<AudioDecoderFactory> CreateBuiltinAudioDecoderFactory() {
rtc::scoped_refptr<AudioDecoderFactory>
CreateBuiltinAudioDecoderFactoryInternal() {
return rtc::scoped_refptr<AudioDecoderFactory>(
new rtc::RefCountedObject<BuiltinAudioDecoderFactory>);
}

24
webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory_internal.h Normal file
View File

@ -0,0 +1,24 @@
/*
* Copyright (c) 2016 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 WEBRTC_MODULES_AUDIO_CODING_CODECS_BUILTIN_AUDIO_DECODER_FACTORY_INTERNAL_H_
#define WEBRTC_MODULES_AUDIO_CODING_CODECS_BUILTIN_AUDIO_DECODER_FACTORY_INTERNAL_H_
#include "webrtc/api/audio_codecs/audio_decoder_factory.h"
#include "webrtc/base/scoped_ref_ptr.h"
namespace webrtc {
rtc::scoped_refptr<AudioDecoderFactory>
CreateBuiltinAudioDecoderFactoryInternal();
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_BUILTIN_AUDIO_DECODER_FACTORY_INTERNAL_H_

2
webrtc/modules/audio_coding/codecs/audio_decoder_factory_unittest.cc → webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory_unittest.cc
View File

@ -10,7 +10,7 @@
#include <memory>
#include "webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h"
#include "webrtc/api/audio_codecs/builtin_audio_decoder_factory.h"
#include "webrtc/test/gtest.h"
namespace webrtc {

2
webrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.h
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@ -11,9 +11,9 @@
#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_G711_AUDIO_DECODER_PCM_H_
#define WEBRTC_MODULES_AUDIO_CODING_CODECS_G711_AUDIO_DECODER_PCM_H_
#include "webrtc/api/audio_codecs/audio_decoder.h"
#include "webrtc/base/checks.h"
#include "webrtc/base/constructormagic.h"
#include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
namespace webrtc {

2
webrtc/modules/audio_coding/codecs/g722/audio_decoder_g722.h
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@ -11,8 +11,8 @@
#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_G722_AUDIO_DECODER_G722_H_
#define WEBRTC_MODULES_AUDIO_CODING_CODECS_G722_AUDIO_DECODER_G722_H_
#include "webrtc/api/audio_codecs/audio_decoder.h"
#include "webrtc/base/constructormagic.h"
#include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
typedef struct WebRtcG722DecInst G722DecInst;

2
webrtc/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.h
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@ -11,8 +11,8 @@
#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_AUDIO_DECODER_ILBC_H_
#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_AUDIO_DECODER_ILBC_H_
#include "webrtc/api/audio_codecs/audio_decoder.h"
#include "webrtc/base/constructormagic.h"
#include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
typedef struct iLBC_decinst_t_ IlbcDecoderInstance;

2
webrtc/modules/audio_coding/codecs/isac/audio_decoder_isac_t.h
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@ -13,10 +13,10 @@
#include <vector>
#include "webrtc/api/audio_codecs/audio_decoder.h"
#include "webrtc/base/constructormagic.h"
#include "webrtc/base/optional.h"
#include "webrtc/base/scoped_ref_ptr.h"
#include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
#include "webrtc/modules/audio_coding/codecs/isac/locked_bandwidth_info.h"
namespace webrtc {

2
webrtc/modules/audio_coding/codecs/legacy_encoded_audio_frame.h
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@ -13,8 +13,8 @@
#include <vector>
#include "webrtc/api/audio_codecs/audio_decoder.h"
#include "webrtc/base/array_view.h"
#include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
namespace webrtc {

4
webrtc/modules/audio_coding/codecs/mock/mock_audio_decoder_factory.h
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@ -13,9 +13,9 @@
#include <vector>
#include "webrtc/api/audio_codecs/audio_decoder_factory.h"
#include "webrtc/api/audio_codecs/builtin_audio_decoder_factory.h"
#include "webrtc/base/scoped_ref_ptr.h"
#include "webrtc/modules/audio_coding/codecs/audio_decoder_factory.h"
#include "webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h"
#include "webrtc/test/gmock.h"
namespace webrtc {

2
webrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.h
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@ -11,8 +11,8 @@
#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_DECODER_OPUS_H_
#define WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_DECODER_OPUS_H_
#include "webrtc/api/audio_codecs/audio_decoder.h"
#include "webrtc/base/constructormagic.h"
#include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
#include "webrtc/modules/audio_coding/codecs/opus/opus_interface.h"
namespace webrtc {

2
webrtc/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.h
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@ -11,8 +11,8 @@
#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_PCM16B_AUDIO_DECODER_PCM16B_H_
#define WEBRTC_MODULES_AUDIO_CODING_CODECS_PCM16B_AUDIO_DECODER_PCM16B_H_
#include "webrtc/api/audio_codecs/audio_decoder.h"
#include "webrtc/base/constructormagic.h"
#include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
namespace webrtc {

2
webrtc/modules/audio_coding/include/audio_coding_module.h
View File

@ -15,11 +15,11 @@
#include <string>
#include <vector>
#include "webrtc/api/audio_codecs/audio_decoder_factory.h"
#include "webrtc/base/deprecation.h"
#include "webrtc/base/function_view.h"
#include "webrtc/base/optional.h"
#include "webrtc/common_types.h"
#include "webrtc/modules/audio_coding/codecs/audio_decoder_factory.h"
#include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h"
#include "webrtc/modules/audio_coding/neteq/include/neteq.h"
#include "webrtc/modules/include/module.h"

7
webrtc/modules/audio_coding/neteq/audio_decoder_impl.h
View File

@ -13,13 +13,14 @@
#include <assert.h>
#include "webrtc/api/audio_codecs/audio_decoder.h"
#include "webrtc/base/constructormagic.h"
#include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
#include "webrtc/modules/audio_coding/acm2/rent_a_codec.h"
#include "webrtc/typedefs.h"
#ifdef WEBRTC_CODEC_G722
#include "webrtc/modules/audio_coding/codecs/g722/g722_interface.h"
#endif
#include "webrtc/modules/audio_coding/acm2/rent_a_codec.h"
#include "webrtc/typedefs.h"
namespace webrtc {

2
webrtc/modules/audio_coding/neteq/comfort_noise.cc
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@ -12,8 +12,8 @@
#include <assert.h>
#include "webrtc/api/audio_codecs/audio_decoder.h"
#include "webrtc/base/logging.h"
#include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
#include "webrtc/modules/audio_coding/neteq/decoder_database.h"
#include "webrtc/modules/audio_coding/neteq/dsp_helper.h"
#include "webrtc/modules/audio_coding/neteq/sync_buffer.h"

2
webrtc/modules/audio_coding/neteq/decoder_database.cc
View File

@ -12,9 +12,9 @@
#include <utility> // pair
#include "webrtc/api/audio_codecs/audio_decoder.h"
#include "webrtc/base/checks.h"
#include "webrtc/base/logging.h"
#include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
namespace webrtc {

4
webrtc/modules/audio_coding/neteq/decoder_database.h
View File

@ -15,10 +15,10 @@
#include <memory>
#include <string>
#include "webrtc/api/audio_codecs/audio_decoder_factory.h"
#include "webrtc/api/audio_codecs/audio_format.h"
#include "webrtc/base/constructormagic.h"
#include "webrtc/common_types.h" // NULL
#include "webrtc/modules/audio_coding/codecs/audio_decoder_factory.h"
#include "webrtc/modules/audio_coding/codecs/audio_format.h"
#include "webrtc/modules/audio_coding/codecs/cng/webrtc_cng.h"
#include "webrtc/modules/audio_coding/neteq/audio_decoder_impl.h"
#include "webrtc/modules/audio_coding/neteq/packet.h"

2
webrtc/modules/audio_coding/neteq/decoder_database_unittest.cc
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@ -15,7 +15,7 @@
#include <string>
#include "webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h"
#include "webrtc/api/audio_codecs/builtin_audio_decoder_factory.h"
#include "webrtc/modules/audio_coding/codecs/mock/mock_audio_decoder_factory.h"
#include "webrtc/modules/audio_coding/neteq/mock/mock_audio_decoder.h"
#include "webrtc/test/gmock.h"

3
webrtc/modules/audio_coding/neteq/mock/mock_audio_decoder.h
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@ -11,8 +11,7 @@
#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_AUDIO_DECODER_H_
#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_AUDIO_DECODER_H_
#include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
#include "webrtc/api/audio_codecs/audio_decoder.h"
#include "webrtc/test/gmock.h"
namespace webrtc {

3
webrtc/modules/audio_coding/neteq/mock/mock_external_decoder_pcm16b.h
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@ -11,8 +11,7 @@
#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_EXTERNAL_DECODER_PCM16B_H_
#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_EXTERNAL_DECODER_PCM16B_H_
#include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
#include "webrtc/api/audio_codecs/audio_decoder.h"
#include "webrtc/base/constructormagic.h"
#include "webrtc/modules/audio_coding/codecs/pcm16b/pcm16b.h"
#include "webrtc/test/gmock.h"

2
webrtc/modules/audio_coding/neteq/neteq_external_decoder_unittest.cc
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@ -12,7 +12,7 @@
#include <memory>
#include "webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h"
#include "webrtc/api/audio_codecs/builtin_audio_decoder_factory.h"
#include "webrtc/modules/audio_coding/neteq/mock/mock_external_decoder_pcm16b.h"
#include "webrtc/modules/audio_coding/neteq/tools/input_audio_file.h"
#include "webrtc/modules/audio_coding/neteq/tools/neteq_external_decoder_test.h"

6
webrtc/modules/audio_coding/neteq/neteq_impl.cc
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@ -17,13 +17,13 @@
#include <utility>
#include <vector>
#include "webrtc/api/audio_codecs/audio_decoder.h"
#include "webrtc/base/checks.h"
#include "webrtc/base/logging.h"
#include "webrtc/base/safe_conversions.h"
#include "webrtc/base/sanitizer.h"
#include "webrtc/base/trace_event.h"
#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
#include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
#include "webrtc/modules/audio_coding/neteq/accelerate.h"
#include "webrtc/modules/audio_coding/neteq/background_noise.h"
#include "webrtc/modules/audio_coding/neteq/buffer_level_filter.h"
@ -39,11 +39,11 @@
#include "webrtc/modules/audio_coding/neteq/merge.h"
#include "webrtc/modules/audio_coding/neteq/nack_tracker.h"
#include "webrtc/modules/audio_coding/neteq/normal.h"
#include "webrtc/modules/audio_coding/neteq/packet_buffer.h"
#include "webrtc/modules/audio_coding/neteq/packet.h"
#include "webrtc/modules/audio_coding/neteq/red_payload_splitter.h"
#include "webrtc/modules/audio_coding/neteq/packet_buffer.h"
#include "webrtc/modules/audio_coding/neteq/post_decode_vad.h"
#include "webrtc/modules/audio_coding/neteq/preemptive_expand.h"
#include "webrtc/modules/audio_coding/neteq/red_payload_splitter.h"
#include "webrtc/modules/audio_coding/neteq/sync_buffer.h"
#include "webrtc/modules/audio_coding/neteq/tick_timer.h"
#include "webrtc/modules/audio_coding/neteq/timestamp_scaler.h"

7
webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc
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@ -10,14 +10,12 @@
#include <memory>
#include "webrtc/modules/audio_coding/neteq/include/neteq.h"
#include "webrtc/modules/audio_coding/neteq/neteq_impl.h"
#include "webrtc/api/audio_codecs/builtin_audio_decoder_factory.h"
#include "webrtc/base/safe_conversions.h"
#include "webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h"
#include "webrtc/modules/audio_coding/codecs/mock/mock_audio_decoder_factory.h"
#include "webrtc/modules/audio_coding/neteq/accelerate.h"
#include "webrtc/modules/audio_coding/neteq/expand.h"
#include "webrtc/modules/audio_coding/neteq/include/neteq.h"
#include "webrtc/modules/audio_coding/neteq/mock/mock_audio_decoder.h"
#include "webrtc/modules/audio_coding/neteq/mock/mock_buffer_level_filter.h"
#include "webrtc/modules/audio_coding/neteq/mock/mock_decoder_database.h"
@ -27,6 +25,7 @@
#include "webrtc/modules/audio_coding/neteq/mock/mock_dtmf_tone_generator.h"
#include "webrtc/modules/audio_coding/neteq/mock/mock_packet_buffer.h"
#include "webrtc/modules/audio_coding/neteq/mock/mock_red_payload_splitter.h"
#include "webrtc/modules/audio_coding/neteq/neteq_impl.h"
#include "webrtc/modules/audio_coding/neteq/preemptive_expand.h"
#include "webrtc/modules/audio_coding/neteq/sync_buffer.h"
#include "webrtc/modules/audio_coding/neteq/timestamp_scaler.h"

2
webrtc/modules/audio_coding/neteq/neteq_stereo_unittest.cc
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@ -15,7 +15,7 @@
#include <string>
#include <list>
#include "webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h"
#include "webrtc/api/audio_codecs/builtin_audio_decoder_factory.h"
#include "webrtc/modules/audio_coding/codecs/pcm16b/pcm16b.h"
#include "webrtc/modules/audio_coding/neteq/include/neteq.h"
#include "webrtc/modules/audio_coding/neteq/tools/input_audio_file.h"

2
webrtc/modules/audio_coding/neteq/neteq_unittest.cc
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@ -21,10 +21,10 @@
#include <vector>
#include "gflags/gflags.h"
#include "webrtc/api/audio_codecs/builtin_audio_decoder_factory.h"
#include "webrtc/base/ignore_wundef.h"
#include "webrtc/base/sha1digest.h"
#include "webrtc/base/stringencode.h"
#include "webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h"
#include "webrtc/modules/audio_coding/codecs/pcm16b/pcm16b.h"
#include "webrtc/modules/audio_coding/neteq/tools/audio_loop.h"
#include "webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h"

2
webrtc/modules/audio_coding/neteq/normal.cc
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@ -14,9 +14,9 @@
#include <algorithm> // min
#include "webrtc/api/audio_codecs/audio_decoder.h"
#include "webrtc/base/checks.h"
#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
#include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
#include "webrtc/modules/audio_coding/neteq/audio_multi_vector.h"
#include "webrtc/modules/audio_coding/neteq/background_noise.h"
#include "webrtc/modules/audio_coding/neteq/decoder_database.h"

2
webrtc/modules/audio_coding/neteq/packet.h
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@ -14,8 +14,8 @@
#include <list>
#include <memory>
#include "webrtc/api/audio_codecs/audio_decoder.h"
#include "webrtc/base/buffer.h"
#include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
#include "webrtc/modules/audio_coding/neteq/tick_timer.h"
#include "webrtc/typedefs.h"

2
webrtc/modules/audio_coding/neteq/packet_buffer.cc
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@ -16,8 +16,8 @@
#include <algorithm> // find_if()
#include "webrtc/api/audio_codecs/audio_decoder.h"
#include "webrtc/base/logging.h"
#include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
#include "webrtc/modules/audio_coding/neteq/decoder_database.h"
#include "webrtc/modules/audio_coding/neteq/tick_timer.h"

5
webrtc/modules/audio_coding/neteq/packet_buffer_unittest.cc
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@ -10,11 +10,10 @@
// Unit tests for PacketBuffer class.
#include "webrtc/modules/audio_coding/neteq/packet_buffer.h"
#include "webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h"
#include "webrtc/api/audio_codecs/builtin_audio_decoder_factory.h"
#include "webrtc/modules/audio_coding/neteq/mock/mock_decoder_database.h"
#include "webrtc/modules/audio_coding/neteq/packet.h"
#include "webrtc/modules/audio_coding/neteq/packet_buffer.h"
#include "webrtc/modules/audio_coding/neteq/tick_timer.h"
#include "webrtc/test/gmock.h"
#include "webrtc/test/gtest.h"

2
webrtc/modules/audio_coding/neteq/post_decode_vad.h
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@ -13,10 +13,10 @@
#include <string> // size_t
#include "webrtc/api/audio_codecs/audio_decoder.h"
#include "webrtc/base/constructormagic.h"
#include "webrtc/common_audio/vad/include/webrtc_vad.h"
#include "webrtc/common_types.h" // NULL
#include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
#include "webrtc/modules/audio_coding/neteq/defines.h"
#include "webrtc/modules/audio_coding/neteq/packet.h"
#include "webrtc/typedefs.h"

2
webrtc/modules/audio_coding/neteq/red_payload_splitter_unittest.cc
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@ -17,7 +17,7 @@
#include <memory>
#include <utility> // pair
#include "webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h"
#include "webrtc/api/audio_codecs/builtin_audio_decoder_factory.h"
#include "webrtc/modules/audio_coding/codecs/mock/mock_audio_decoder_factory.h"
#include "webrtc/modules/audio_coding/neteq/mock/mock_decoder_database.h"
#include "webrtc/modules/audio_coding/neteq/packet.h"

5
webrtc/modules/audio_coding/neteq/timestamp_scaler_unittest.cc
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@ -8,11 +8,10 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#include "webrtc/modules/audio_coding/neteq/timestamp_scaler.h"
#include "webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h"
#include "webrtc/api/audio_codecs/builtin_audio_decoder_factory.h"
#include "webrtc/modules/audio_coding/neteq/mock/mock_decoder_database.h"
#include "webrtc/modules/audio_coding/neteq/packet.h"
#include "webrtc/modules/audio_coding/neteq/timestamp_scaler.h"
#include "webrtc/test/gmock.h"
#include "webrtc/test/gtest.h"

2
webrtc/modules/audio_coding/neteq/tools/fake_decode_from_file.h
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@ -13,9 +13,9 @@
#include <memory>
#include "webrtc/api/audio_codecs/audio_decoder.h"
#include "webrtc/base/array_view.h"
#include "webrtc/base/optional.h"
#include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
#include "webrtc/modules/audio_coding/neteq/tools/input_audio_file.h"
namespace webrtc {

2
webrtc/modules/audio_coding/neteq/tools/neteq_external_decoder_test.cc
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@ -11,8 +11,8 @@
#include "webrtc/modules/audio_coding/neteq/tools/neteq_external_decoder_test.h"
#include "webrtc/api/audio_codecs/builtin_audio_decoder_factory.h"
#include "webrtc/base/format_macros.h"
#include "webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h"
#include "webrtc/test/gtest.h"
namespace webrtc {

2
webrtc/modules/audio_coding/neteq/tools/neteq_external_decoder_test.h
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@ -14,7 +14,7 @@
#include <memory>
#include <string>
#include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
#include "webrtc/api/audio_codecs/audio_decoder.h"
#include "webrtc/modules/audio_coding/neteq/include/neteq.h"
#include "webrtc/modules/include/module_common_types.h"

2
webrtc/modules/audio_coding/neteq/tools/neteq_performance_test.cc
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@ -10,8 +10,8 @@
#include "webrtc/modules/audio_coding/neteq/tools/neteq_performance_test.h"
#include "webrtc/api/audio_codecs/builtin_audio_decoder_factory.h"
#include "webrtc/base/checks.h"
#include "webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h"
#include "webrtc/modules/audio_coding/codecs/pcm16b/pcm16b.h"
#include "webrtc/modules/audio_coding/neteq/include/neteq.h"
#include "webrtc/modules/audio_coding/neteq/tools/audio_loop.h"

3
webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.cc
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@ -10,8 +10,9 @@
#include <math.h>
#include <stdio.h>
#include "webrtc/api/audio_codecs/builtin_audio_decoder_factory.h"
#include "webrtc/base/checks.h"
#include "webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h"
#include "webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.h"
#include "webrtc/modules/audio_coding/neteq/tools/output_audio_file.h"
#include "webrtc/modules/audio_coding/neteq/tools/output_wav_file.h"

2
webrtc/modules/audio_coding/neteq/tools/neteq_test.cc
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@ -12,7 +12,7 @@
#include <iostream>
#include "webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h"
#include "webrtc/api/audio_codecs/builtin_audio_decoder_factory.h"
namespace webrtc {
namespace test {

2
webrtc/modules/audio_coding/test/TwoWayCommunication.cc
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@ -20,9 +20,9 @@
#include <Windows.h>
#endif
#include "webrtc/api/audio_codecs/builtin_audio_decoder_factory.h"
#include "webrtc/common_types.h"
#include "webrtc/modules/audio_coding/codecs/audio_format_conversion.h"
#include "webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h"
#include "webrtc/modules/audio_coding/test/PCMFile.h"
#include "webrtc/modules/audio_coding/test/utility.h"
#include "webrtc/system_wrappers/include/trace.h"