Revert of L16 implementation of the Audio{En,De}coderFactoryTemplate APIs (patchset #5 id:80001 of https://codereview.webrtc.org/2995523002/ )
Reason for revert:
Breaks compilation in google3.
Original issue's description:
> L16 implementation of the Audio{En,De}coderFactoryTemplate APIs
>
> BUG=webrtc:7836, webrtc:7842
>
> Review-Url: https://codereview.webrtc.org/2995523002
> Cr-Commit-Position: refs/heads/master@{#19354}
> Committed: edff94df62
TBR=ossu@webrtc.org,kwiberg@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7836, webrtc:7842
Review-Url: https://codereview.webrtc.org/2996993002
Cr-Commit-Position: refs/heads/master@{#19356}
This commit is contained in:
@ -1,39 +0,0 @@
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# Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
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#
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# Use of this source code is governed by a BSD-style license
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# that can be found in the LICENSE file in the root of the source
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# tree. An additional intellectual property rights grant can be found
|
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# in the file PATENTS. All contributing project authors may
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# be found in the AUTHORS file in the root of the source tree.
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import("../../../webrtc.gni")
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if (is_android) {
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import("//build/config/android/config.gni")
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import("//build/config/android/rules.gni")
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}
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rtc_static_library("audio_encoder_L16") {
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sources = [
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"audio_encoder_L16.cc",
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"audio_encoder_L16.h",
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]
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deps = [
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"..:audio_codecs_api",
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"../../..:webrtc_common",
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"../../../modules/audio_coding:pcm16b",
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"../../../rtc_base:rtc_base_approved",
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]
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}
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rtc_static_library("audio_decoder_L16") {
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sources = [
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"audio_decoder_L16.cc",
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"audio_decoder_L16.h",
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]
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deps = [
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"..:audio_codecs_api",
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"../../..:webrtc_common",
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"../../../modules/audio_coding:pcm16b",
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"../../../rtc_base:rtc_base_approved",
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]
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}
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@ -1,42 +0,0 @@
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/*
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* Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
|
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* tree. An additional intellectual property rights grant can be found
|
|
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* in the file PATENTS. All contributing project authors may
|
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "webrtc/api/audio_codecs/L16/audio_decoder_L16.h"
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#include "webrtc/common_types.h"
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#include "webrtc/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.h"
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#include "webrtc/modules/audio_coding/codecs/pcm16b/pcm16b_common.h"
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#include "webrtc/rtc_base/ptr_util.h"
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namespace webrtc {
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rtc::Optional<AudioDecoderL16::Config> AudioDecoderL16::SdpToConfig(
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const SdpAudioFormat& format) {
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Config config;
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config.sample_rate_hz = format.clockrate_hz;
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config.num_channels = format.num_channels;
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return STR_CASE_CMP(format.name.c_str(), "L16") == 0 && config.IsOk()
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? rtc::Optional<Config>(config)
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: rtc::Optional<Config>();
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}
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void AudioDecoderL16::AppendSupportedDecoders(
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std::vector<AudioCodecSpec>* specs) {
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Pcm16BAppendSupportedCodecSpecs(specs);
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}
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std::unique_ptr<AudioDecoder> AudioDecoderL16::MakeAudioDecoder(
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const Config& config) {
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return config.IsOk() ? rtc::MakeUnique<AudioDecoderPcm16B>(
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config.sample_rate_hz, config.num_channels)
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: nullptr;
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}
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} // namespace webrtc
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@ -1,44 +0,0 @@
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/*
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* Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
|
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* that can be found in the LICENSE file in the root of the source
|
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* tree. An additional intellectual property rights grant can be found
|
|
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* in the file PATENTS. All contributing project authors may
|
|
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* be found in the AUTHORS file in the root of the source tree.
|
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*/
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#ifndef WEBRTC_API_AUDIO_CODECS_L16_AUDIO_DECODER_L16_H_
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#define WEBRTC_API_AUDIO_CODECS_L16_AUDIO_DECODER_L16_H_
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#include <memory>
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#include <vector>
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#include "webrtc/api/audio_codecs/audio_decoder.h"
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#include "webrtc/api/audio_codecs/audio_format.h"
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#include "webrtc/rtc_base/optional.h"
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namespace webrtc {
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// L16 decoder API for use as a template parameter to
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// CreateAudioDecoderFactory<...>().
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//
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// NOTE: This struct is still under development and may change without notice.
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struct AudioDecoderL16 {
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struct Config {
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bool IsOk() const {
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return (sample_rate_hz == 8000 || sample_rate_hz == 16000 ||
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sample_rate_hz == 32000 || sample_rate_hz == 48000) &&
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num_channels >= 1;
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}
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int sample_rate_hz = 8000;
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int num_channels = 1;
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};
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static rtc::Optional<Config> SdpToConfig(const SdpAudioFormat& audio_format);
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static void AppendSupportedDecoders(std::vector<AudioCodecSpec>* specs);
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static std::unique_ptr<AudioDecoder> MakeAudioDecoder(const Config& config);
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};
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} // namespace webrtc
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#endif // WEBRTC_API_AUDIO_CODECS_L16_AUDIO_DECODER_L16_H_
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@ -1,55 +0,0 @@
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/*
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* Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
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*
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|
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* 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.
|
|
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*/
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#include "webrtc/api/audio_codecs/L16/audio_encoder_L16.h"
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#include "webrtc/common_types.h"
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#include "webrtc/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h"
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#include "webrtc/modules/audio_coding/codecs/pcm16b/pcm16b_common.h"
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#include "webrtc/rtc_base/ptr_util.h"
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namespace webrtc {
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rtc::Optional<AudioEncoderL16::Config> AudioEncoderL16::SdpToConfig(
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const SdpAudioFormat& format) {
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Config config;
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config.sample_rate_hz = format.clockrate_hz;
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config.num_channels = format.num_channels;
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return STR_CASE_CMP(format.name.c_str(), "L16") == 0 && config.IsOk()
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? rtc::Optional<Config>(config)
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: rtc::Optional<Config>();
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}
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void AudioEncoderL16::AppendSupportedEncoders(
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std::vector<AudioCodecSpec>* specs) {
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Pcm16BAppendSupportedCodecSpecs(specs);
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}
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AudioCodecInfo AudioEncoderL16::QueryAudioEncoder(
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const AudioEncoderL16::Config& config) {
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RTC_DCHECK(config.IsOk());
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return {config.sample_rate_hz,
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rtc::dchecked_cast<size_t>(config.num_channels),
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config.sample_rate_hz * config.num_channels * 16};
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}
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std::unique_ptr<AudioEncoder> AudioEncoderL16::MakeAudioEncoder(
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const AudioEncoderL16::Config& config,
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int payload_type) {
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RTC_DCHECK(config.IsOk());
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AudioEncoderPcm16B::Config c;
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c.sample_rate_hz = config.sample_rate_hz;
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c.num_channels = config.num_channels;
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c.frame_size_ms = config.frame_size_ms;
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c.payload_type = payload_type;
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return rtc::MakeUnique<AudioEncoderPcm16B>(c);
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}
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} // namespace webrtc
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@ -1,48 +0,0 @@
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/*
|
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* 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.
|
|
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*/
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#ifndef WEBRTC_API_AUDIO_CODECS_L16_AUDIO_ENCODER_L16_H_
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#define WEBRTC_API_AUDIO_CODECS_L16_AUDIO_ENCODER_L16_H_
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#include <memory>
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#include <vector>
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#include "webrtc/api/audio_codecs/audio_encoder.h"
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#include "webrtc/api/audio_codecs/audio_format.h"
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#include "webrtc/rtc_base/optional.h"
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namespace webrtc {
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// L16 encoder API for use as a template parameter to
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// CreateAudioEncoderFactory<...>().
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//
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// NOTE: This struct is still under development and may change without notice.
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struct AudioEncoderL16 {
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struct Config {
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bool IsOk() const {
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return (sample_rate_hz == 8000 || sample_rate_hz == 16000 ||
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sample_rate_hz == 32000 || sample_rate_hz == 48000) &&
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num_channels >= 1 && frame_size_ms > 0 && frame_size_ms <= 120 &&
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frame_size_ms % 10 == 0;
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}
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int sample_rate_hz = 8000;
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int num_channels = 1;
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int frame_size_ms = 10;
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};
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static rtc::Optional<Config> SdpToConfig(const SdpAudioFormat& audio_format);
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static void AppendSupportedEncoders(std::vector<AudioCodecSpec>* specs);
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static AudioCodecInfo QueryAudioEncoder(const Config& config);
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static std::unique_ptr<AudioEncoder> MakeAudioEncoder(const Config& config,
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int payload_type);
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};
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|
|
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} // namespace webrtc
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#endif // WEBRTC_API_AUDIO_CODECS_L16_AUDIO_ENCODER_L16_H_
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@ -25,8 +25,6 @@ if (rtc_include_tests) {
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"../../../rtc_base:rtc_base_approved",
|
"../../../rtc_base:rtc_base_approved",
|
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"../../../test:audio_codec_mocks",
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"../../../test:audio_codec_mocks",
|
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"../../../test:test_support",
|
"../../../test:test_support",
|
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"../L16:audio_decoder_L16",
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|
||||||
"../L16:audio_encoder_L16",
|
|
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"../g711:audio_decoder_g711",
|
"../g711:audio_decoder_g711",
|
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"../g711:audio_encoder_g711",
|
"../g711:audio_encoder_g711",
|
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"../g722:audio_decoder_g722",
|
"../g722:audio_decoder_g722",
|
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|
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@ -9,7 +9,6 @@
|
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*/
|
*/
|
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|
|
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#include "webrtc/api/audio_codecs/audio_decoder_factory_template.h"
|
#include "webrtc/api/audio_codecs/audio_decoder_factory_template.h"
|
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#include "webrtc/api/audio_codecs/L16/audio_decoder_L16.h"
|
|
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#include "webrtc/api/audio_codecs/g711/audio_decoder_g711.h"
|
#include "webrtc/api/audio_codecs/g711/audio_decoder_g711.h"
|
||||||
#include "webrtc/api/audio_codecs/g722/audio_decoder_g722.h"
|
#include "webrtc/api/audio_codecs/g722/audio_decoder_g722.h"
|
||||||
#include "webrtc/api/audio_codecs/ilbc/audio_decoder_ilbc.h"
|
#include "webrtc/api/audio_codecs/ilbc/audio_decoder_ilbc.h"
|
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@ -166,26 +165,6 @@ TEST(AudioDecoderFactoryTemplateTest, Ilbc) {
|
|||||||
EXPECT_EQ(8000, dec->SampleRateHz());
|
EXPECT_EQ(8000, dec->SampleRateHz());
|
||||||
}
|
}
|
||||||
|
|
||||||
TEST(AudioDecoderFactoryTemplateTest, L16) {
|
|
||||||
auto factory = CreateAudioDecoderFactory<AudioDecoderL16>();
|
|
||||||
EXPECT_THAT(
|
|
||||||
factory->GetSupportedDecoders(),
|
|
||||||
testing::ElementsAre(
|
|
||||||
AudioCodecSpec{{"L16", 8000, 1}, {8000, 1, 8000 * 16}},
|
|
||||||
AudioCodecSpec{{"L16", 16000, 1}, {16000, 1, 16000 * 16}},
|
|
||||||
AudioCodecSpec{{"L16", 32000, 1}, {32000, 1, 32000 * 16}},
|
|
||||||
AudioCodecSpec{{"L16", 8000, 2}, {8000, 2, 8000 * 16 * 2}},
|
|
||||||
AudioCodecSpec{{"L16", 16000, 2}, {16000, 2, 16000 * 16 * 2}},
|
|
||||||
AudioCodecSpec{{"L16", 32000, 2}, {32000, 2, 32000 * 16 * 2}}));
|
|
||||||
EXPECT_FALSE(factory->IsSupportedDecoder({"foo", 8000, 1}));
|
|
||||||
EXPECT_TRUE(factory->IsSupportedDecoder({"L16", 48000, 1}));
|
|
||||||
EXPECT_FALSE(factory->IsSupportedDecoder({"L16", 96000, 1}));
|
|
||||||
EXPECT_EQ(nullptr, factory->MakeAudioDecoder({"L16", 8000, 0}));
|
|
||||||
auto dec = factory->MakeAudioDecoder({"L16", 48000, 2});
|
|
||||||
ASSERT_NE(nullptr, dec);
|
|
||||||
EXPECT_EQ(48000, dec->SampleRateHz());
|
|
||||||
}
|
|
||||||
|
|
||||||
TEST(AudioDecoderFactoryTemplateTest, Opus) {
|
TEST(AudioDecoderFactoryTemplateTest, Opus) {
|
||||||
auto factory = CreateAudioDecoderFactory<AudioDecoderOpus>();
|
auto factory = CreateAudioDecoderFactory<AudioDecoderOpus>();
|
||||||
AudioCodecInfo opus_info{48000, 1, 64000, 6000, 510000};
|
AudioCodecInfo opus_info{48000, 1, 64000, 6000, 510000};
|
||||||
|
|||||||
@ -9,7 +9,6 @@
|
|||||||
*/
|
*/
|
||||||
|
|
||||||
#include "webrtc/api/audio_codecs/audio_encoder_factory_template.h"
|
#include "webrtc/api/audio_codecs/audio_encoder_factory_template.h"
|
||||||
#include "webrtc/api/audio_codecs/L16/audio_encoder_L16.h"
|
|
||||||
#include "webrtc/api/audio_codecs/g711/audio_encoder_g711.h"
|
#include "webrtc/api/audio_codecs/g711/audio_encoder_g711.h"
|
||||||
#include "webrtc/api/audio_codecs/g722/audio_encoder_g722.h"
|
#include "webrtc/api/audio_codecs/g722/audio_encoder_g722.h"
|
||||||
#include "webrtc/api/audio_codecs/ilbc/audio_encoder_ilbc.h"
|
#include "webrtc/api/audio_codecs/ilbc/audio_encoder_ilbc.h"
|
||||||
@ -171,27 +170,6 @@ TEST(AudioEncoderFactoryTemplateTest, Ilbc) {
|
|||||||
EXPECT_EQ(8000, enc->SampleRateHz());
|
EXPECT_EQ(8000, enc->SampleRateHz());
|
||||||
}
|
}
|
||||||
|
|
||||||
TEST(AudioEncoderFactoryTemplateTest, L16) {
|
|
||||||
auto factory = CreateAudioEncoderFactory<AudioEncoderL16>();
|
|
||||||
EXPECT_THAT(
|
|
||||||
factory->GetSupportedEncoders(),
|
|
||||||
testing::ElementsAre(
|
|
||||||
AudioCodecSpec{{"L16", 8000, 1}, {8000, 1, 8000 * 16}},
|
|
||||||
AudioCodecSpec{{"L16", 16000, 1}, {16000, 1, 16000 * 16}},
|
|
||||||
AudioCodecSpec{{"L16", 32000, 1}, {32000, 1, 32000 * 16}},
|
|
||||||
AudioCodecSpec{{"L16", 8000, 2}, {8000, 2, 8000 * 16 * 2}},
|
|
||||||
AudioCodecSpec{{"L16", 16000, 2}, {16000, 2, 16000 * 16 * 2}},
|
|
||||||
AudioCodecSpec{{"L16", 32000, 2}, {32000, 2, 32000 * 16 * 2}}));
|
|
||||||
EXPECT_EQ(rtc::Optional<AudioCodecInfo>(),
|
|
||||||
factory->QueryAudioEncoder({"L16", 8000, 0}));
|
|
||||||
EXPECT_EQ(rtc::Optional<AudioCodecInfo>({48000, 1, 48000 * 16}),
|
|
||||||
factory->QueryAudioEncoder({"L16", 48000, 1}));
|
|
||||||
EXPECT_EQ(nullptr, factory->MakeAudioEncoder(17, {"L16", 8000, 0}));
|
|
||||||
auto enc = factory->MakeAudioEncoder(17, {"L16", 48000, 2});
|
|
||||||
ASSERT_NE(nullptr, enc);
|
|
||||||
EXPECT_EQ(48000, enc->SampleRateHz());
|
|
||||||
}
|
|
||||||
|
|
||||||
TEST(AudioEncoderFactoryTemplateTest, Opus) {
|
TEST(AudioEncoderFactoryTemplateTest, Opus) {
|
||||||
auto factory = CreateAudioEncoderFactory<AudioEncoderOpus>();
|
auto factory = CreateAudioEncoderFactory<AudioEncoderOpus>();
|
||||||
AudioCodecInfo info = {48000, 1, 32000, 6000, 510000};
|
AudioCodecInfo info = {48000, 1, 32000, 6000, 510000};
|
||||||
|
|||||||
@ -792,8 +792,6 @@ rtc_static_library("pcm16b") {
|
|||||||
"codecs/pcm16b/audio_decoder_pcm16b.h",
|
"codecs/pcm16b/audio_decoder_pcm16b.h",
|
||||||
"codecs/pcm16b/audio_encoder_pcm16b.cc",
|
"codecs/pcm16b/audio_encoder_pcm16b.cc",
|
||||||
"codecs/pcm16b/audio_encoder_pcm16b.h",
|
"codecs/pcm16b/audio_encoder_pcm16b.h",
|
||||||
"codecs/pcm16b/pcm16b_common.cc",
|
|
||||||
"codecs/pcm16b/pcm16b_common.h",
|
|
||||||
]
|
]
|
||||||
|
|
||||||
deps = [
|
deps = [
|
||||||
|
|||||||
@ -1,25 +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 "webrtc/modules/audio_coding/codecs/pcm16b/pcm16b_common.h"
|
|
||||||
|
|
||||||
namespace webrtc {
|
|
||||||
|
|
||||||
void Pcm16BAppendSupportedCodecSpecs(std::vector<AudioCodecSpec>* specs) {
|
|
||||||
for (size_t num_channels : {1, 2}) {
|
|
||||||
for (int sample_rate_hz : {8000, 16000, 32000}) {
|
|
||||||
specs->push_back(
|
|
||||||
{{"L16", sample_rate_hz, num_channels},
|
|
||||||
{sample_rate_hz, num_channels, sample_rate_hz * num_channels * 16}});
|
|
||||||
}
|
|
||||||
}
|
|
||||||
}
|
|
||||||
|
|
||||||
} // namespace webrtc
|
|
||||||
@ -1,22 +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 WEBRTC_MODULES_AUDIO_CODING_CODECS_PCM16B_PCM16B_COMMON_H_
|
|
||||||
#define WEBRTC_MODULES_AUDIO_CODING_CODECS_PCM16B_PCM16B_COMMON_H_
|
|
||||||
|
|
||||||
#include <vector>
|
|
||||||
|
|
||||||
#include "webrtc/api/audio_codecs/audio_decoder_factory.h"
|
|
||||||
|
|
||||||
namespace webrtc {
|
|
||||||
void Pcm16BAppendSupportedCodecSpecs(std::vector<AudioCodecSpec>* specs);
|
|
||||||
}
|
|
||||||
|
|
||||||
#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_PCM16B_PCM16B_COMMON_H_
|
|
||||||
Reference in New Issue
Block a user