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Remove unused AudioFrameProcessor* parameter from WebRtcVoiceEngine constructor

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Bug: webrtc:15111
Change-Id: Ia55e55f98ffeceeb91fb9b4fc2323a4fd7bc1046
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/326523
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41118}
This commit is contained in:
Danil Chapovalov 2023-11-09 11:48:22 +01:00 committed by WebRTC LUCI CQ
parent fa4d7c92b7
commit e567d8a112
7 changed files with 15 additions and 35 deletions
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3
api/enable_media.cc
View File

@ -54,8 +54,7 @@ class MediaFactoryImpl : public MediaFactory {
deps.task_queue_factory.get(), deps.adm.get(),
std::move(deps.audio_encoder_factory),
std::move(deps.audio_decoder_factory), std::move(deps.audio_mixer),
std::move(deps.audio_processing), /*audio_frame_processor=*/nullptr,
/*owned_audio_frame_processor=*/std::move(deps.audio_frame_processor),
std::move(deps.audio_processing), std::move(deps.audio_frame_processor),
*trials);
auto video_engine = std::make_unique<WebRtcVideoEngine>(
std::move(deps.video_encoder_factory),

2
media/engine/null_webrtc_video_engine_unittest.cc
View File

@ -37,7 +37,7 @@ TEST(NullWebRtcVideoEngineTest, CheckInterface) {
task_queue_factory.get(), adm.get(),
webrtc::MockAudioEncoderFactory::CreateUnusedFactory(),
webrtc::MockAudioDecoderFactory::CreateUnusedFactory(), nullptr,
webrtc::AudioProcessingBuilder().Create(), nullptr, nullptr, trials);
webrtc::AudioProcessingBuilder().Create(), nullptr, trials);
CompositeMediaEngine engine(std::move(audio_engine),
std::make_unique<NullWebRtcVideoEngine>());

1
media/engine/webrtc_media_engine.cc
View File

@ -46,7 +46,6 @@ std::unique_ptr<MediaEngineInterface> CreateMediaEngine(
std::move(dependencies.audio_decoder_factory),
std::move(dependencies.audio_mixer),
std::move(dependencies.audio_processing),
dependencies.audio_frame_processor,
std::move(dependencies.owned_audio_frame_processor), trials);
#ifdef HAVE_WEBRTC_VIDEO
auto video_engine = std::make_unique<WebRtcVideoEngine>(

3
media/engine/webrtc_media_engine.h
View File

@ -49,9 +49,6 @@ struct MediaEngineDependencies {
rtc::scoped_refptr<webrtc::AudioDecoderFactory> audio_decoder_factory;
rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer;
rtc::scoped_refptr<webrtc::AudioProcessing> audio_processing;
// TODO(bugs.webrtc.org/15111):
// Remove the raw AudioFrameProcessor pointer in the follow-up.
webrtc::AudioFrameProcessor* audio_frame_processor = nullptr;
std::unique_ptr<webrtc::AudioFrameProcessor> owned_audio_frame_processor;
std::unique_ptr<webrtc::VideoEncoderFactory> video_encoder_factory;

14
media/engine/webrtc_voice_engine.cc
View File

@ -342,10 +342,7 @@ WebRtcVoiceEngine::WebRtcVoiceEngine(
const rtc::scoped_refptr<webrtc::AudioDecoderFactory>& decoder_factory,
rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer,
rtc::scoped_refptr<webrtc::AudioProcessing> audio_processing,
// TODO(bugs.webrtc.org/15111):
// Remove the raw AudioFrameProcessor pointer in the follow-up.
webrtc::AudioFrameProcessor* audio_frame_processor,
std::unique_ptr<webrtc::AudioFrameProcessor> owned_audio_frame_processor,
std::unique_ptr<webrtc::AudioFrameProcessor> audio_frame_processor,
const webrtc::FieldTrialsView& trials)
: task_queue_factory_(task_queue_factory),
adm_(adm),
@ -353,8 +350,7 @@ WebRtcVoiceEngine::WebRtcVoiceEngine(
decoder_factory_(decoder_factory),
audio_mixer_(audio_mixer),
apm_(audio_processing),
audio_frame_processor_(audio_frame_processor),
owned_audio_frame_processor_(std::move(owned_audio_frame_processor)),
audio_frame_processor_(std::move(audio_frame_processor)),
minimized_remsampling_on_mobile_trial_enabled_(
IsEnabled(trials, "WebRTC-Audio-MinimizeResamplingOnMobile")) {
RTC_LOG(LS_INFO) << "WebRtcVoiceEngine::WebRtcVoiceEngine";
@ -423,11 +419,7 @@ void WebRtcVoiceEngine::Init() {
if (audio_frame_processor_) {
config.async_audio_processing_factory =
rtc::make_ref_counted<webrtc::AsyncAudioProcessing::Factory>(
*audio_frame_processor_, *task_queue_factory_);
} else if (owned_audio_frame_processor_) {
config.async_audio_processing_factory =
rtc::make_ref_counted<webrtc::AsyncAudioProcessing::Factory>(
std::move(owned_audio_frame_processor_), *task_queue_factory_);
std::move(audio_frame_processor_), *task_queue_factory_);
}
audio_state_ = webrtc::AudioState::Create(config);
}

8
media/engine/webrtc_voice_engine.h
View File

@ -90,9 +90,6 @@ class WebRtcVoiceEngine final : public VoiceEngineInterface {
const rtc::scoped_refptr<webrtc::AudioDecoderFactory>& decoder_factory,
rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer,
rtc::scoped_refptr<webrtc::AudioProcessing> audio_processing,
// TODO(bugs.webrtc.org/15111):
// Remove the raw AudioFrameProcessor pointer in the follow-up.
webrtc::AudioFrameProcessor* audio_frame_processor,
std::unique_ptr<webrtc::AudioFrameProcessor> owned_audio_frame_processor,
const webrtc::FieldTrialsView& trials);
@ -166,10 +163,7 @@ class WebRtcVoiceEngine final : public VoiceEngineInterface {
// The audio processing module.
rtc::scoped_refptr<webrtc::AudioProcessing> apm_;
// Asynchronous audio processing.
// TODO(bugs.webrtc.org/15111):
// Remove the raw AudioFrameProcessor pointer in the follow-up.
webrtc::AudioFrameProcessor* const audio_frame_processor_;
std::unique_ptr<webrtc::AudioFrameProcessor> owned_audio_frame_processor_;
std::unique_ptr<webrtc::AudioFrameProcessor> audio_frame_processor_;
// The primary instance of WebRtc VoiceEngine.
rtc::scoped_refptr<webrtc::AudioState> audio_state_;
std::vector<AudioCodec> send_codecs_;

19
media/engine/webrtc_voice_engine_unittest.cc
View File

@ -174,7 +174,7 @@ TEST(WebRtcVoiceEngineTestStubLibrary, StartupShutdown) {
task_queue_factory.get(), adm.get(),
webrtc::MockAudioEncoderFactory::CreateUnusedFactory(),
webrtc::MockAudioDecoderFactory::CreateUnusedFactory(), nullptr, apm,
nullptr, nullptr, trials);
nullptr, trials);
engine.Init();
}
}
@ -220,7 +220,7 @@ class WebRtcVoiceEngineTestFake : public ::testing::TestWithParam<bool> {
auto decoder_factory = webrtc::CreateBuiltinAudioDecoderFactory();
engine_.reset(new cricket::WebRtcVoiceEngine(
task_queue_factory_.get(), adm_.get(), encoder_factory, decoder_factory,
nullptr, apm_, nullptr, nullptr, field_trials_));
nullptr, apm_, nullptr, field_trials_));
engine_->Init();
send_parameters_.codecs.push_back(kPcmuCodec);
recv_parameters_.codecs.push_back(kPcmuCodec);
@ -3689,7 +3689,7 @@ TEST(WebRtcVoiceEngineTest, StartupShutdown) {
task_queue_factory.get(), adm.get(),
webrtc::MockAudioEncoderFactory::CreateUnusedFactory(),
webrtc::MockAudioDecoderFactory::CreateUnusedFactory(), nullptr, apm,
nullptr, nullptr, field_trials);
nullptr, field_trials);
engine.Init();
webrtc::RtcEventLogNull event_log;
CallConfig call_config(&event_log);
@ -3725,7 +3725,7 @@ TEST(WebRtcVoiceEngineTest, StartupShutdownWithExternalADM) {
task_queue_factory.get(), adm.get(),
webrtc::MockAudioEncoderFactory::CreateUnusedFactory(),
webrtc::MockAudioDecoderFactory::CreateUnusedFactory(), nullptr, apm,
nullptr, nullptr, field_trials);
nullptr, field_trials);
engine.Init();
webrtc::RtcEventLogNull event_log;
CallConfig call_config(&event_log);
@ -3765,7 +3765,7 @@ TEST(WebRtcVoiceEngineTest, HasCorrectPayloadTypeMapping) {
task_queue_factory.get(), adm.get(),
webrtc::MockAudioEncoderFactory::CreateUnusedFactory(),
webrtc::MockAudioDecoderFactory::CreateUnusedFactory(), nullptr, apm,
nullptr, nullptr, field_trials);
nullptr, field_trials);
engine.Init();
for (const cricket::AudioCodec& codec : engine.send_codecs()) {
auto is_codec = [&codec](const char* name, int clockrate = 0) {
@ -3815,7 +3815,7 @@ TEST(WebRtcVoiceEngineTest, Has32Channels) {
task_queue_factory.get(), adm.get(),
webrtc::MockAudioEncoderFactory::CreateUnusedFactory(),
webrtc::MockAudioDecoderFactory::CreateUnusedFactory(), nullptr, apm,
nullptr, nullptr, field_trials);
nullptr, field_trials);
engine.Init();
webrtc::RtcEventLogNull event_log;
CallConfig call_config(&event_log);
@ -3861,7 +3861,7 @@ TEST(WebRtcVoiceEngineTest, SetRecvCodecs) {
task_queue_factory.get(), adm.get(),
webrtc::MockAudioEncoderFactory::CreateUnusedFactory(),
webrtc::CreateBuiltinAudioDecoderFactory(), nullptr, apm, nullptr,
nullptr, field_trials);
field_trials);
engine.Init();
webrtc::RtcEventLogNull event_log;
CallConfig call_config(&event_log);
@ -3889,8 +3889,7 @@ TEST(WebRtcVoiceEngineTest, SetRtpSendParametersMaxBitrate) {
cricket::WebRtcVoiceEngine engine(task_queue_factory.get(), adm.get(),
webrtc::CreateBuiltinAudioEncoderFactory(),
webrtc::CreateBuiltinAudioDecoderFactory(),
nullptr, nullptr, nullptr, nullptr,
field_trials);
nullptr, nullptr, nullptr, field_trials);
engine.Init();
webrtc::RtcEventLogNull event_log;
CallConfig call_config(&event_log);
@ -3965,7 +3964,7 @@ TEST(WebRtcVoiceEngineTest, CollectRecvCodecs) {
webrtc::FieldTrialBasedConfig field_trials;
cricket::WebRtcVoiceEngine engine(
task_queue_factory.get(), adm.get(), unused_encoder_factory,
mock_decoder_factory, nullptr, apm, nullptr, nullptr, field_trials);
mock_decoder_factory, nullptr, apm, nullptr, field_trials);
engine.Init();
auto codecs = engine.recv_codecs();
EXPECT_EQ(11u, codecs.size());