New interface (AudioDecoderFactory), with an implementation
This is a first draft of what we're hoping to use to create all AudioDecoder instances. Follow-up CLs will start using this internally in NetEq instead of calling constructors manually. BUG=webrtc:5801 Review-Url: https://codereview.webrtc.org/1917163002 Cr-Commit-Position: refs/heads/master@{#12548}
This commit is contained in:
kwiberg
@ -39,6 +39,33 @@ if (!build_with_mozilla && !build_with_chromium) {
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audio_codec_deps += [ ":red" ]
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audio_codec_deps += [ ":red" ]
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}
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}
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|
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source_set("audio_decoder_factory_interface") {
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sources = [
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"codecs/audio_decoder_factory.h",
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"codecs/audio_format.cc",
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"codecs/audio_format.h",
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]
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configs += [ "../..:common_config" ]
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|
public_configs = [ "../..:common_inherited_config" ]
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|
deps = [
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"../..:webrtc_common",
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]
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}
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|
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source_set("builtin_audio_decoder_factory") {
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sources = [
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"codecs/builtin_audio_decoder_factory.cc",
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|
"codecs/builtin_audio_decoder_factory.h",
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]
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|
configs += [ "../..:common_config" ]
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|
public_configs = [ "../..:common_inherited_config" ]
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|
deps = [
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|
"../..:webrtc_common",
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":audio_decoder_factory_interface",
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|
] + audio_codec_deps
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|
defines = audio_codec_defines
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}
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|
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source_set("rent_a_codec") {
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source_set("rent_a_codec") {
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sources = [
|
sources = [
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"acm2/acm_codec_database.cc",
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"acm2/acm_codec_database.cc",
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@ -68,6 +68,50 @@
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'audio_coding_defines': '<(audio_codec_defines)',
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'audio_coding_defines': '<(audio_codec_defines)',
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},
|
},
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'targets': [
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'targets': [
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{
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|
'target_name': 'audio_decoder_factory_interface',
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|
'type': 'static_library',
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|
'dependencies': [
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'<(webrtc_root)/common.gyp:webrtc_common',
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|
],
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|
'include_dirs': [
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|
'<(webrtc_root)',
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|
],
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|
'direct_dependent_settings': {
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|
'include_dirs': [
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|
'<(webrtc_root)',
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|
],
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|
},
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|
'sources': [
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|
'codecs/audio_decoder_factory.h',
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'codecs/audio_format.cc',
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|
'codecs/audio_format.h',
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|
],
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|
},
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{
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|
'target_name': 'builtin_audio_decoder_factory',
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|
'type': 'static_library',
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|
'defines': [
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'<@(audio_codec_defines)',
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|
],
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|
'dependencies': [
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|
'<(webrtc_root)/common.gyp:webrtc_common',
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|
'<@(audio_codec_dependencies)',
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|
'audio_decoder_factory_interface',
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|
],
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|
'include_dirs': [
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|
'<(webrtc_root)',
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|
],
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|
'direct_dependent_settings': {
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|
'include_dirs': [
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|
'<(webrtc_root)',
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|
],
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|
},
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|
'sources': [
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|
'codecs/builtin_audio_decoder_factory.cc',
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|
'codecs/builtin_audio_decoder_factory.h',
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|
],
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|
},
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{
|
{
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'target_name': 'rent_a_codec',
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'target_name': 'rent_a_codec',
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'type': 'static_library',
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'type': 'static_library',
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36
webrtc/modules/audio_coding/codecs/audio_decoder_factory.h
Normal file
36
webrtc/modules/audio_coding/codecs/audio_decoder_factory.h
Normal file
@ -0,0 +1,36 @@
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|
/*
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|
* Copyright (c) 2016 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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|
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#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_DECODER_FACTORY_H_
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#define WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_DECODER_FACTORY_H_
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#include <memory>
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#include <vector>
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#include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
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#include "webrtc/modules/audio_coding/codecs/audio_format.h"
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namespace webrtc {
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// A factory that creates AudioDecoders.
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// NOTE: This class is still under development and may change without notice.
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class AudioDecoderFactory {
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public:
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virtual ~AudioDecoderFactory() = default;
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virtual std::vector<SdpAudioFormat> GetSupportedFormats() = 0;
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virtual std::unique_ptr<AudioDecoder> MakeAudioDecoder(
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const SdpAudioFormat& format) = 0;
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_DECODER_FACTORY_H_
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@ -0,0 +1,117 @@
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|
/*
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|
* Copyright (c) 2016 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 "testing/gtest/include/gtest/gtest.h"
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#include "webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h"
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namespace webrtc {
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TEST(AudioDecoderFactoryTest, CreateUnknownDecoder) {
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std::unique_ptr<AudioDecoderFactory> adf = CreateBuiltinAudioDecoderFactory();
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ASSERT_TRUE(adf);
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EXPECT_FALSE(adf->MakeAudioDecoder(SdpAudioFormat("rey", 8000, 1)));
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}
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TEST(AudioDecoderFactoryTest, CreatePcmu) {
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std::unique_ptr<AudioDecoderFactory> adf = CreateBuiltinAudioDecoderFactory();
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ASSERT_TRUE(adf);
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// PCMu supports 8 kHz, and any number of channels.
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EXPECT_FALSE(adf->MakeAudioDecoder(SdpAudioFormat("pcmu", 8000, 0)));
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EXPECT_TRUE(adf->MakeAudioDecoder(SdpAudioFormat("pcmu", 8000, 1)));
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EXPECT_TRUE(adf->MakeAudioDecoder(SdpAudioFormat("pcmu", 8000, 2)));
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EXPECT_TRUE(adf->MakeAudioDecoder(SdpAudioFormat("pcmu", 8000, 3)));
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EXPECT_FALSE(adf->MakeAudioDecoder(SdpAudioFormat("pcmu", 16000, 1)));
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}
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TEST(AudioDecoderFactoryTest, CreatePcma) {
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std::unique_ptr<AudioDecoderFactory> adf = CreateBuiltinAudioDecoderFactory();
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ASSERT_TRUE(adf);
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// PCMa supports 8 kHz, and any number of channels.
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EXPECT_FALSE(adf->MakeAudioDecoder(SdpAudioFormat("pcma", 8000, 0)));
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EXPECT_TRUE(adf->MakeAudioDecoder(SdpAudioFormat("pcma", 8000, 1)));
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EXPECT_TRUE(adf->MakeAudioDecoder(SdpAudioFormat("pcma", 8000, 2)));
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EXPECT_TRUE(adf->MakeAudioDecoder(SdpAudioFormat("pcma", 8000, 3)));
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EXPECT_FALSE(adf->MakeAudioDecoder(SdpAudioFormat("pcma", 16000, 1)));
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|
}
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|
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|
TEST(AudioDecoderFactoryTest, CreateIlbc) {
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|
std::unique_ptr<AudioDecoderFactory> adf = CreateBuiltinAudioDecoderFactory();
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|
ASSERT_TRUE(adf);
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|
// iLBC supports 8 kHz, 1 channel.
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|
EXPECT_FALSE(adf->MakeAudioDecoder(SdpAudioFormat("ilbc", 8000, 0)));
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EXPECT_TRUE(adf->MakeAudioDecoder(SdpAudioFormat("ilbc", 8000, 1)));
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EXPECT_FALSE(adf->MakeAudioDecoder(SdpAudioFormat("ilbc", 8000, 2)));
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|
EXPECT_FALSE(adf->MakeAudioDecoder(SdpAudioFormat("ilbc", 16000, 1)));
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|
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|
// iLBC actually uses a 16 kHz sample rate instead of the nominal 8 kHz.
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|
// TODO(kwiberg): Uncomment this once AudioDecoder has a SampleRateHz method.
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|
// std::unique_ptr<AudioDecoder> dec =
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|
// adf->MakeAudioDecoder(SdpAudioFormat("ilbc", 8000, 1));
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|
// EXPECT_EQ(16000, dec->SampleRateHz());
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|
}
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|
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|
TEST(AudioDecoderFactoryTest, CreateIsac) {
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|
std::unique_ptr<AudioDecoderFactory> adf = CreateBuiltinAudioDecoderFactory();
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|
ASSERT_TRUE(adf);
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|
// iSAC supports 16 kHz, 1 channel. The float implementation additionally
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|
// supports 32 kHz, 1 channel.
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|
EXPECT_FALSE(adf->MakeAudioDecoder(SdpAudioFormat("isac", 16000, 0)));
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|
EXPECT_TRUE(adf->MakeAudioDecoder(SdpAudioFormat("isac", 16000, 1)));
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|
EXPECT_FALSE(adf->MakeAudioDecoder(SdpAudioFormat("isac", 16000, 2)));
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|
EXPECT_FALSE(adf->MakeAudioDecoder(SdpAudioFormat("isac", 8000, 1)));
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|
EXPECT_FALSE(adf->MakeAudioDecoder(SdpAudioFormat("isac", 48000, 1)));
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|
#ifdef WEBRTC_ARCH_ARM
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|
EXPECT_FALSE(adf->MakeAudioDecoder(SdpAudioFormat("isac", 32000, 1)));
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|
#else
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|
EXPECT_TRUE(adf->MakeAudioDecoder(SdpAudioFormat("isac", 32000, 1)));
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|
#endif
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|
}
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|
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|
TEST(AudioDecoderFactoryTest, CreateL16) {
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|
std::unique_ptr<AudioDecoderFactory> adf = CreateBuiltinAudioDecoderFactory();
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|
ASSERT_TRUE(adf);
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|
// L16 supports any clock rate, any number of channels.
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|
const int clockrates[] = {8000, 16000, 32000, 48000};
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|
const int num_channels[] = {1, 2, 3, 4711};
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|
for (int clockrate : clockrates) {
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|
EXPECT_FALSE(adf->MakeAudioDecoder(SdpAudioFormat("l16", clockrate, 0)));
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|
for (int channels : num_channels) {
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|
EXPECT_TRUE(
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|
adf->MakeAudioDecoder(SdpAudioFormat("l16", clockrate, channels)));
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|
}
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|
}
|
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|
}
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|
|
||||||
|
TEST(AudioDecoderFactoryTest, CreateG722) {
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|
std::unique_ptr<AudioDecoderFactory> adf = CreateBuiltinAudioDecoderFactory();
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|
ASSERT_TRUE(adf);
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|
// g722 supports 8 kHz, 1-2 channels.
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|
EXPECT_FALSE(adf->MakeAudioDecoder(SdpAudioFormat("g722", 8000, 0)));
|
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|
EXPECT_TRUE(adf->MakeAudioDecoder(SdpAudioFormat("g722", 8000, 1)));
|
||||||
|
EXPECT_TRUE(adf->MakeAudioDecoder(SdpAudioFormat("g722", 8000, 2)));
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|
EXPECT_FALSE(adf->MakeAudioDecoder(SdpAudioFormat("g722", 8000, 3)));
|
||||||
|
EXPECT_FALSE(adf->MakeAudioDecoder(SdpAudioFormat("g722", 16000, 1)));
|
||||||
|
EXPECT_FALSE(adf->MakeAudioDecoder(SdpAudioFormat("g722", 32000, 1)));
|
||||||
|
}
|
||||||
|
|
||||||
|
TEST(AudioDecoderFactoryTest, CreateOpus) {
|
||||||
|
std::unique_ptr<AudioDecoderFactory> adf = CreateBuiltinAudioDecoderFactory();
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||||||
|
ASSERT_TRUE(adf);
|
||||||
|
// Opus supports 48 kHz, 1-2 channels.
|
||||||
|
EXPECT_FALSE(adf->MakeAudioDecoder(SdpAudioFormat("opus", 48000, 0)));
|
||||||
|
EXPECT_TRUE(adf->MakeAudioDecoder(SdpAudioFormat("opus", 48000, 1)));
|
||||||
|
EXPECT_TRUE(adf->MakeAudioDecoder(SdpAudioFormat("opus", 48000, 2)));
|
||||||
|
EXPECT_FALSE(adf->MakeAudioDecoder(SdpAudioFormat("opus", 48000, 3)));
|
||||||
|
EXPECT_FALSE(adf->MakeAudioDecoder(SdpAudioFormat("opus", 8000, 1)));
|
||||||
|
EXPECT_FALSE(adf->MakeAudioDecoder(SdpAudioFormat("opus", 16000, 1)));
|
||||||
|
EXPECT_FALSE(adf->MakeAudioDecoder(SdpAudioFormat("opus", 32000, 1)));
|
||||||
|
}
|
||||||
|
|
||||||
|
} // namespace webrtc
|
||||||
50
webrtc/modules/audio_coding/codecs/audio_format.cc
Normal file
50
webrtc/modules/audio_coding/codecs/audio_format.cc
Normal file
@ -0,0 +1,50 @@
|
|||||||
|
/*
|
||||||
|
* 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"
|
||||||
|
|
||||||
|
namespace webrtc {
|
||||||
|
|
||||||
|
SdpAudioFormat::SdpAudioFormat() = default;
|
||||||
|
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() = default;
|
||||||
|
SdpAudioFormat& SdpAudioFormat::operator=(const SdpAudioFormat&) = default;
|
||||||
|
SdpAudioFormat& SdpAudioFormat::operator=(SdpAudioFormat&&) = default;
|
||||||
|
|
||||||
|
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;
|
||||||
|
}
|
||||||
|
|
||||||
|
} // namespace webrtc
|
||||||
48
webrtc/modules/audio_coding/codecs/audio_format.h
Normal file
48
webrtc/modules/audio_coding/codecs/audio_format.h
Normal file
@ -0,0 +1,48 @@
|
|||||||
|
/*
|
||||||
|
* 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 {
|
||||||
|
// TODO(kwiberg): Get rid of the default constructor when rtc::Optional no
|
||||||
|
// longer requires it.
|
||||||
|
SdpAudioFormat();
|
||||||
|
SdpAudioFormat(const SdpAudioFormat&);
|
||||||
|
SdpAudioFormat(SdpAudioFormat&&);
|
||||||
|
SdpAudioFormat(const char* name, int clockrate_hz, int num_channels);
|
||||||
|
~SdpAudioFormat();
|
||||||
|
|
||||||
|
SdpAudioFormat& operator=(const SdpAudioFormat&);
|
||||||
|
SdpAudioFormat& operator=(SdpAudioFormat&&);
|
||||||
|
|
||||||
|
using Parameters = std::map<std::string, std::string>;
|
||||||
|
std::string name;
|
||||||
|
int clockrate_hz;
|
||||||
|
int num_channels;
|
||||||
|
Parameters parameters;
|
||||||
|
// Parameters feedback_parameters; ??
|
||||||
|
};
|
||||||
|
|
||||||
|
void swap(SdpAudioFormat& a, SdpAudioFormat& b);
|
||||||
|
std::ostream& operator<<(std::ostream& os, const SdpAudioFormat& saf);
|
||||||
|
|
||||||
|
} // namespace webrtc
|
||||||
|
|
||||||
|
#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_FORMAT_H_
|
||||||
@ -0,0 +1,140 @@
|
|||||||
|
/*
|
||||||
|
* 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/builtin_audio_decoder_factory.h"
|
||||||
|
|
||||||
|
#include <vector>
|
||||||
|
|
||||||
|
#include "webrtc/base/checks.h"
|
||||||
|
#include "webrtc/common_types.h"
|
||||||
|
#include "webrtc/modules/audio_coding/codecs/cng/webrtc_cng.h"
|
||||||
|
#include "webrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.h"
|
||||||
|
#ifdef WEBRTC_CODEC_G722
|
||||||
|
#include "webrtc/modules/audio_coding/codecs/g722/audio_decoder_g722.h"
|
||||||
|
#endif
|
||||||
|
#ifdef WEBRTC_CODEC_ILBC
|
||||||
|
#include "webrtc/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.h"
|
||||||
|
#endif
|
||||||
|
#ifdef WEBRTC_CODEC_ISACFX
|
||||||
|
#include "webrtc/modules/audio_coding/codecs/isac/fix/include/audio_decoder_isacfix.h"
|
||||||
|
#endif
|
||||||
|
#ifdef WEBRTC_CODEC_ISAC
|
||||||
|
#include "webrtc/modules/audio_coding/codecs/isac/main/include/audio_decoder_isac.h"
|
||||||
|
#endif
|
||||||
|
#ifdef WEBRTC_CODEC_OPUS
|
||||||
|
#include "webrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.h"
|
||||||
|
#endif
|
||||||
|
#include "webrtc/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.h"
|
||||||
|
|
||||||
|
namespace webrtc {
|
||||||
|
|
||||||
|
namespace {
|
||||||
|
|
||||||
|
struct NamedDecoderConstructor {
|
||||||
|
const char* name;
|
||||||
|
std::unique_ptr<AudioDecoder> (*constructor)(int clockrate_hz,
|
||||||
|
int num_channels);
|
||||||
|
};
|
||||||
|
|
||||||
|
std::unique_ptr<AudioDecoder> Unique(AudioDecoder* d) {
|
||||||
|
return std::unique_ptr<AudioDecoder>(d);
|
||||||
|
}
|
||||||
|
|
||||||
|
// TODO(kwiberg): These factory functions should probably be moved to each
|
||||||
|
// decoder.
|
||||||
|
NamedDecoderConstructor decoder_constructors[] = {
|
||||||
|
{"pcmu",
|
||||||
|
[](int clockrate_hz, int num_channels) {
|
||||||
|
return clockrate_hz == 8000 && num_channels >= 1
|
||||||
|
? Unique(new AudioDecoderPcmU(num_channels))
|
||||||
|
: nullptr;
|
||||||
|
}},
|
||||||
|
{"pcma",
|
||||||
|
[](int clockrate_hz, int num_channels) {
|
||||||
|
return clockrate_hz == 8000 && num_channels >= 1
|
||||||
|
? Unique(new AudioDecoderPcmA(num_channels))
|
||||||
|
: nullptr;
|
||||||
|
}},
|
||||||
|
#ifdef WEBRTC_CODEC_ILBC
|
||||||
|
{"ilbc",
|
||||||
|
[](int clockrate_hz, int num_channels) {
|
||||||
|
return clockrate_hz == 8000 && num_channels == 1
|
||||||
|
? Unique(new AudioDecoderIlbc)
|
||||||
|
: nullptr;
|
||||||
|
}},
|
||||||
|
#endif
|
||||||
|
#if defined(WEBRTC_CODEC_ISACFX)
|
||||||
|
{"isac",
|
||||||
|
[](int clockrate_hz, int num_channels) {
|
||||||
|
return clockrate_hz == 16000 && num_channels == 1
|
||||||
|
? Unique(new AudioDecoderIsacFix)
|
||||||
|
: nullptr;
|
||||||
|
}},
|
||||||
|
#elif defined(WEBRTC_CODEC_ISAC)
|
||||||
|
{"isac",
|
||||||
|
[](int clockrate_hz, int num_channels) {
|
||||||
|
return (clockrate_hz == 16000 || clockrate_hz == 32000) &&
|
||||||
|
num_channels == 1
|
||||||
|
? Unique(new AudioDecoderIsac)
|
||||||
|
: nullptr;
|
||||||
|
}},
|
||||||
|
#endif
|
||||||
|
{"l16",
|
||||||
|
[](int clockrate_hz, int num_channels) {
|
||||||
|
return num_channels >= 1 ? Unique(new AudioDecoderPcm16B(num_channels))
|
||||||
|
: nullptr;
|
||||||
|
}},
|
||||||
|
#ifdef WEBRTC_CODEC_G722
|
||||||
|
{"g722",
|
||||||
|
[](int clockrate_hz, int num_channels) {
|
||||||
|
if (clockrate_hz == 8000) {
|
||||||
|
if (num_channels == 1)
|
||||||
|
return Unique(new AudioDecoderG722);
|
||||||
|
if (num_channels == 2)
|
||||||
|
return Unique(new AudioDecoderG722Stereo);
|
||||||
|
}
|
||||||
|
return Unique(nullptr);
|
||||||
|
}},
|
||||||
|
#endif
|
||||||
|
#ifdef WEBRTC_CODEC_OPUS
|
||||||
|
{"opus",
|
||||||
|
[](int clockrate_hz, int num_channels) {
|
||||||
|
return clockrate_hz == 48000 && (num_channels == 1 || num_channels == 2)
|
||||||
|
? Unique(new AudioDecoderOpus(num_channels))
|
||||||
|
: nullptr;
|
||||||
|
}},
|
||||||
|
#endif
|
||||||
|
};
|
||||||
|
|
||||||
|
class BuiltinAudioDecoderFactory : public AudioDecoderFactory {
|
||||||
|
public:
|
||||||
|
std::vector<SdpAudioFormat> GetSupportedFormats() override {
|
||||||
|
FATAL() << "Not implemented yet!";
|
||||||
|
}
|
||||||
|
|
||||||
|
std::unique_ptr<AudioDecoder> MakeAudioDecoder(
|
||||||
|
const SdpAudioFormat& format) override {
|
||||||
|
for (const auto& dc : decoder_constructors) {
|
||||||
|
if (STR_CASE_CMP(format.name.c_str(), dc.name) == 0) {
|
||||||
|
return std::unique_ptr<AudioDecoder>(
|
||||||
|
dc.constructor(format.clockrate_hz, format.num_channels));
|
||||||
|
}
|
||||||
|
}
|
||||||
|
return nullptr;
|
||||||
|
}
|
||||||
|
};
|
||||||
|
|
||||||
|
} // namespace
|
||||||
|
|
||||||
|
std::unique_ptr<AudioDecoderFactory> CreateBuiltinAudioDecoderFactory() {
|
||||||
|
return std::unique_ptr<AudioDecoderFactory>(new BuiltinAudioDecoderFactory);
|
||||||
|
}
|
||||||
|
|
||||||
|
} // namespace webrtc
|
||||||
@ -0,0 +1,26 @@
|
|||||||
|
/*
|
||||||
|
* 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_H_
|
||||||
|
#define WEBRTC_MODULES_AUDIO_CODING_CODECS_BUILTIN_AUDIO_DECODER_FACTORY_H_
|
||||||
|
|
||||||
|
#include <memory>
|
||||||
|
|
||||||
|
#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.
|
||||||
|
std::unique_ptr<AudioDecoderFactory> CreateBuiltinAudioDecoderFactory();
|
||||||
|
|
||||||
|
} // namespace webrtc
|
||||||
|
|
||||||
|
#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_BUILTIN_AUDIO_DECODER_FACTORY_H_
|
||||||
@ -133,6 +133,7 @@
|
|||||||
'audio_processing',
|
'audio_processing',
|
||||||
'audioproc_test_utils',
|
'audioproc_test_utils',
|
||||||
'bitrate_controller',
|
'bitrate_controller',
|
||||||
|
'builtin_audio_decoder_factory',
|
||||||
'bwe_simulator',
|
'bwe_simulator',
|
||||||
'cng',
|
'cng',
|
||||||
'isac_fix',
|
'isac_fix',
|
||||||
@ -167,6 +168,7 @@
|
|||||||
'<(webrtc_root)/tools/tools.gyp:agc_test_utils',
|
'<(webrtc_root)/tools/tools.gyp:agc_test_utils',
|
||||||
],
|
],
|
||||||
'sources': [
|
'sources': [
|
||||||
|
'audio_coding/codecs/audio_decoder_factory_unittest.cc',
|
||||||
'audio_coding/codecs/cng/audio_encoder_cng_unittest.cc',
|
'audio_coding/codecs/cng/audio_encoder_cng_unittest.cc',
|
||||||
'audio_coding/acm2/acm_receiver_unittest_oldapi.cc',
|
'audio_coding/acm2/acm_receiver_unittest_oldapi.cc',
|
||||||
'audio_coding/acm2/audio_coding_module_unittest_oldapi.cc',
|
'audio_coding/acm2/audio_coding_module_unittest_oldapi.cc',
|
||||||
|
|||||||
Reference in New Issue
Block a user