From 04c0722cf1fc38e6888ef44e413603e0cb461ea4 Mon Sep 17 00:00:00 2001 From: aleloi Date: Tue, 22 Nov 2016 06:42:53 -0800 Subject: [PATCH] Replace AudioConferenceMixer with AudioMixer. This CL re-routes audio through AudioMixer instead of AudioConferenceMixer. This is done without any modifications to VoiceEngine. Previously, output audio was polled by an AudioDevice through an AudioTransport pointer, which was an instance of VoEBaseImpl. VoiceEngineImpl sent the request for data on to OutputMixer and further to AudioConferenceMixer. This CL changes the audio flow to an AudioDevice. We reconfigure the AudioDevice to have another AudioTransport pointer, which points to an AudioTransportProxy. The AudioTransportProxy is responsible for feeding mixed data to the AudioProcessing component for echo cancellation, and to resample the audio data after AudioProcessing and before it is sent to the AudioDevice. The set up of the audio path was previously done during VoiceEngine initialization. Now it is changed in the AudioState constructor. This list shows where audio-path-related VoiceEngine functionality has been moved: OutputMixer --> AudioTransportProxy VoiceEngineImpl --> AudioState, AudioTransportProxy SharedData --> AudioState Channel --> AudioReceiveStream, ChannelProxy, Channel AudioState owns the new mixer and connects it to AudioTransport and AudioDevice on initialization. The audio input source is AudioReceiveStream, which registers itself with the mixer (which it gets from AudioState) on Start and Stop. # Since the AudioTransport interface contains non-const references. NOPRESUBMIT=True BUG=webrtc:6346 Review-Url: https://codereview.webrtc.org/2436033002 Cr-Commit-Position: refs/heads/master@{#15193} --- webrtc/audio/BUILD.gn | 3 + webrtc/audio/DEPS | 1 + webrtc/audio/audio_receive_stream.cc | 65 ++++++-- webrtc/audio/audio_receive_stream.h | 5 + webrtc/audio/audio_receive_stream_unittest.cc | 36 ++++- webrtc/audio/audio_state_unittest.cc | 139 ++++++++++++------ webrtc/audio/audio_transport_proxy.cc | 55 +++++-- webrtc/audio/audio_transport_proxy.h | 8 +- 8 files changed, 239 insertions(+), 73 deletions(-) diff --git a/webrtc/audio/BUILD.gn b/webrtc/audio/BUILD.gn index e89350e89a..8200448401 100644 --- a/webrtc/audio/BUILD.gn +++ b/webrtc/audio/BUILD.gn @@ -32,6 +32,7 @@ rtc_static_library("audio") { "../api:audio_mixer_api", "../api:call_api", "../base:rtc_base_approved", + "../common_audio", "../modules/audio_device", "../modules/audio_processing", "../system_wrappers", @@ -48,8 +49,10 @@ if (rtc_include_tests) { ] deps = [ ":audio", + "../api:mock_audio_mixer", "../base:rtc_base_approved", "../modules/audio_device:mock_audio_device", + "../modules/audio_mixer:audio_mixer_impl", "../test:test_common", "//testing/gmock", "//testing/gtest", diff --git a/webrtc/audio/DEPS b/webrtc/audio/DEPS index 6e6259b9e5..f383c1456b 100644 --- a/webrtc/audio/DEPS +++ b/webrtc/audio/DEPS @@ -1,6 +1,7 @@ include_rules = [ "+webrtc/base", "+webrtc/call", + "+webrtc/common_audio/resampler", "+webrtc/logging/rtc_event_log", "+webrtc/modules/audio_coding/codecs/mock", "+webrtc/modules/audio_device", diff --git a/webrtc/audio/audio_receive_stream.cc b/webrtc/audio/audio_receive_stream.cc index 5a8efc223e..81ffdade5c 100644 --- a/webrtc/audio/audio_receive_stream.cc +++ b/webrtc/audio/audio_receive_stream.cc @@ -138,9 +138,11 @@ AudioReceiveStream::AudioReceiveStream( } AudioReceiveStream::~AudioReceiveStream() { - RTC_DCHECK(thread_checker_.CalledOnValidThread()); + RTC_DCHECK_RUN_ON(&thread_checker_); LOG(LS_INFO) << "~AudioReceiveStream: " << config_.ToString(); - Stop(); + if (playing_) { + Stop(); + } channel_proxy_->DisassociateSendChannel(); channel_proxy_->DeRegisterExternalTransport(); channel_proxy_->ResetCongestionControlObjects(); @@ -151,22 +153,39 @@ AudioReceiveStream::~AudioReceiveStream() { } void AudioReceiveStream::Start() { - RTC_DCHECK(thread_checker_.CalledOnValidThread()); - ScopedVoEInterface base(voice_engine()); - int error = base->StartPlayout(config_.voe_channel_id); + RTC_DCHECK_RUN_ON(&thread_checker_); + if (playing_) { + return; + } + + int error = SetVoiceEnginePlayout(true); if (error != 0) { LOG(LS_ERROR) << "AudioReceiveStream::Start failed with error: " << error; + return; } + + if (!audio_state()->mixer()->AddSource(this)) { + LOG(LS_ERROR) << "Failed to add source to mixer."; + SetVoiceEnginePlayout(false); + return; + } + + playing_ = true; } void AudioReceiveStream::Stop() { - RTC_DCHECK(thread_checker_.CalledOnValidThread()); - ScopedVoEInterface base(voice_engine()); - base->StopPlayout(config_.voe_channel_id); + RTC_DCHECK_RUN_ON(&thread_checker_); + if (!playing_) { + return; + } + playing_ = false; + + audio_state()->mixer()->RemoveSource(this); + SetVoiceEnginePlayout(false); } webrtc::AudioReceiveStream::Stats AudioReceiveStream::GetStats() const { - RTC_DCHECK(thread_checker_.CalledOnValidThread()); + RTC_DCHECK_RUN_ON(&thread_checker_); webrtc::AudioReceiveStream::Stats stats; stats.remote_ssrc = config_.rtp.remote_ssrc; ScopedVoEInterface codec(voice_engine()); @@ -216,17 +235,17 @@ webrtc::AudioReceiveStream::Stats AudioReceiveStream::GetStats() const { } void AudioReceiveStream::SetSink(std::unique_ptr sink) { - RTC_DCHECK(thread_checker_.CalledOnValidThread()); + RTC_DCHECK_RUN_ON(&thread_checker_); channel_proxy_->SetSink(std::move(sink)); } void AudioReceiveStream::SetGain(float gain) { - RTC_DCHECK(thread_checker_.CalledOnValidThread()); + RTC_DCHECK_RUN_ON(&thread_checker_); channel_proxy_->SetChannelOutputVolumeScaling(gain); } const webrtc::AudioReceiveStream::Config& AudioReceiveStream::config() const { - RTC_DCHECK(thread_checker_.CalledOnValidThread()); + RTC_DCHECK_RUN_ON(&thread_checker_); return config_; } @@ -243,7 +262,7 @@ void AudioReceiveStream::AssociateSendStream(AudioSendStream* send_stream) { } void AudioReceiveStream::SignalNetworkState(NetworkState state) { - RTC_DCHECK(thread_checker_.CalledOnValidThread()); + RTC_DCHECK_RUN_ON(&thread_checker_); } bool AudioReceiveStream::DeliverRtcp(const uint8_t* packet, size_t length) { @@ -296,12 +315,26 @@ int AudioReceiveStream::Ssrc() const { return config_.rtp.local_ssrc; } +internal::AudioState* AudioReceiveStream::audio_state() const { + auto* audio_state = static_cast(audio_state_.get()); + RTC_DCHECK(audio_state); + return audio_state; +} + VoiceEngine* AudioReceiveStream::voice_engine() const { - internal::AudioState* audio_state = - static_cast(audio_state_.get()); - VoiceEngine* voice_engine = audio_state->voice_engine(); + auto* voice_engine = audio_state()->voice_engine(); RTC_DCHECK(voice_engine); return voice_engine; } + +int AudioReceiveStream::SetVoiceEnginePlayout(bool playout) { + ScopedVoEInterface base(voice_engine()); + if (playout) { + return base->StartPlayout(config_.voe_channel_id); + } else { + return base->StopPlayout(config_.voe_channel_id); + } +} + } // namespace internal } // namespace webrtc diff --git a/webrtc/audio/audio_receive_stream.h b/webrtc/audio/audio_receive_stream.h index aeaadc6d68..6e7da09d1d 100644 --- a/webrtc/audio/audio_receive_stream.h +++ b/webrtc/audio/audio_receive_stream.h @@ -16,6 +16,7 @@ #include "webrtc/api/audio/audio_mixer.h" #include "webrtc/api/call/audio_receive_stream.h" #include "webrtc/api/call/audio_state.h" +#include "webrtc/audio/audio_state.h" #include "webrtc/base/constructormagic.h" #include "webrtc/base/thread_checker.h" #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h" @@ -64,6 +65,8 @@ class AudioReceiveStream final : public webrtc::AudioReceiveStream, private: VoiceEngine* voice_engine() const; + AudioState* audio_state() const; + int SetVoiceEnginePlayout(bool playout); rtc::ThreadChecker thread_checker_; RemoteBitrateEstimator* remote_bitrate_estimator_ = nullptr; @@ -72,6 +75,8 @@ class AudioReceiveStream final : public webrtc::AudioReceiveStream, std::unique_ptr rtp_header_parser_; std::unique_ptr channel_proxy_; + bool playing_ ACCESS_ON(thread_checker_) = false; + RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioReceiveStream); }; } // namespace internal diff --git a/webrtc/audio/audio_receive_stream_unittest.cc b/webrtc/audio/audio_receive_stream_unittest.cc index f263948fc4..bc071970e1 100644 --- a/webrtc/audio/audio_receive_stream_unittest.cc +++ b/webrtc/audio/audio_receive_stream_unittest.cc @@ -11,11 +11,11 @@ #include #include +#include "webrtc/api/test/mock_audio_mixer.h" #include "webrtc/audio/audio_receive_stream.h" #include "webrtc/audio/conversion.h" #include "webrtc/logging/rtc_event_log/mock/mock_rtc_event_log.h" #include "webrtc/modules/audio_coding/codecs/mock/mock_audio_decoder_factory.h" -#include "webrtc/modules/audio_mixer/audio_mixer_impl.h" #include "webrtc/modules/bitrate_controller/include/mock/mock_bitrate_controller.h" #include "webrtc/modules/congestion_controller/include/mock/mock_congestion_controller.h" #include "webrtc/modules/pacing/packet_router.h" @@ -72,7 +72,8 @@ struct ConfigHelper { congestion_controller_(&simulated_clock_, &bitrate_observer_, &remote_bitrate_observer_, - &event_log_) { + &event_log_), + audio_mixer_(new rtc::RefCountedObject()) { using testing::Invoke; EXPECT_CALL(voice_engine_, @@ -85,7 +86,7 @@ struct ConfigHelper { AudioState::Config config; config.voice_engine = &voice_engine_; - config.audio_mixer = AudioMixerImpl::Create(); + config.audio_mixer = audio_mixer_; audio_state_ = AudioState::Create(config); EXPECT_CALL(voice_engine_, ChannelProxyFactory(kChannelId)) @@ -122,7 +123,6 @@ struct ConfigHelper { EXPECT_CALL(*channel_proxy_, DisassociateSendChannel()).Times(1); return channel_proxy_; })); - EXPECT_CALL(voice_engine_, StopPlayout(kChannelId)).WillOnce(Return(0)); stream_config_.voe_channel_id = kChannelId; stream_config_.rtp.local_ssrc = kLocalSsrc; stream_config_.rtp.remote_ssrc = kRemoteSsrc; @@ -143,6 +143,7 @@ struct ConfigHelper { MockRtcEventLog* event_log() { return &event_log_; } AudioReceiveStream::Config& config() { return stream_config_; } rtc::scoped_refptr audio_state() { return audio_state_; } + rtc::scoped_refptr audio_mixer() { return audio_mixer_; } MockVoiceEngine& voice_engine() { return voice_engine_; } MockVoEChannelProxy* channel_proxy() { return channel_proxy_; } @@ -185,6 +186,7 @@ struct ConfigHelper { MockRtcEventLog event_log_; testing::StrictMock voice_engine_; rtc::scoped_refptr audio_state_; + rtc::scoped_refptr audio_mixer_; AudioReceiveStream::Config stream_config_; testing::StrictMock* channel_proxy_ = nullptr; }; @@ -368,5 +370,31 @@ TEST(AudioReceiveStreamTest, SetGain) { SetChannelOutputVolumeScaling(FloatEq(0.765f))); recv_stream.SetGain(0.765f); } + +TEST(AudioReceiveStreamTest, StreamShouldNotBeAddedToMixerWhenVoEReturnsError) { + ConfigHelper helper; + internal::AudioReceiveStream recv_stream( + helper.congestion_controller(), helper.config(), helper.audio_state(), + helper.event_log()); + + EXPECT_CALL(helper.voice_engine(), StartPlayout(_)).WillOnce(Return(-1)); + EXPECT_CALL(*helper.audio_mixer(), AddSource(_)).Times(0); + + recv_stream.Start(); +} + +TEST(AudioReceiveStreamTest, StreamShouldBeAddedToMixerOnStart) { + ConfigHelper helper; + internal::AudioReceiveStream recv_stream( + helper.congestion_controller(), helper.config(), helper.audio_state(), + helper.event_log()); + + EXPECT_CALL(helper.voice_engine(), StartPlayout(_)).WillOnce(Return(0)); + EXPECT_CALL(helper.voice_engine(), StopPlayout(_)); + EXPECT_CALL(*helper.audio_mixer(), AddSource(&recv_stream)) + .WillOnce(Return(true)); + + recv_stream.Start(); +} } // namespace test } // namespace webrtc diff --git a/webrtc/audio/audio_state_unittest.cc b/webrtc/audio/audio_state_unittest.cc index 38485a88bf..05c5003459 100644 --- a/webrtc/audio/audio_state_unittest.cc +++ b/webrtc/audio/audio_state_unittest.cc @@ -19,26 +19,65 @@ namespace webrtc { namespace test { namespace { -struct ConfigHelper { - ConfigHelper() { - EXPECT_CALL(voice_engine_, - RegisterVoiceEngineObserver(testing::_)).WillOnce(testing::Return(0)); - EXPECT_CALL(voice_engine_, - DeRegisterVoiceEngineObserver()).WillOnce(testing::Return(0)); - EXPECT_CALL(voice_engine_, audio_device_module()); - EXPECT_CALL(voice_engine_, audio_processing()); - EXPECT_CALL(voice_engine_, audio_transport()); +const int kSampleRate = 8000; +const int kNumberOfChannels = 1; +const int kBytesPerSample = 2; - config_.voice_engine = &voice_engine_; - config_.audio_mixer = AudioMixerImpl::Create(); +struct ConfigHelper { + ConfigHelper() : audio_mixer(AudioMixerImpl::Create()) { + EXPECT_CALL(mock_voice_engine, RegisterVoiceEngineObserver(testing::_)) + .WillOnce(testing::Return(0)); + EXPECT_CALL(mock_voice_engine, DeRegisterVoiceEngineObserver()) + .WillOnce(testing::Return(0)); + EXPECT_CALL(mock_voice_engine, audio_device_module()) + .Times(testing::AtLeast(1)); + EXPECT_CALL(mock_voice_engine, audio_processing()) + .Times(testing::AtLeast(1)); + EXPECT_CALL(mock_voice_engine, audio_transport()) + .WillRepeatedly(testing::Return(&audio_transport)); + + auto device = static_cast( + voice_engine().audio_device_module()); + + // Populate the audio transport proxy pointer to the most recent + // transport connected to the Audio Device. + ON_CALL(*device, RegisterAudioCallback(testing::_)) + .WillByDefault(testing::Invoke([this](AudioTransport* transport) { + registered_audio_transport = transport; + return 0; + })); + + audio_state_config.voice_engine = &mock_voice_engine; + audio_state_config.audio_mixer = audio_mixer; } - AudioState::Config& config() { return config_; } - MockVoiceEngine& voice_engine() { return voice_engine_; } + AudioState::Config& config() { return audio_state_config; } + MockVoiceEngine& voice_engine() { return mock_voice_engine; } + rtc::scoped_refptr mixer() { return audio_mixer; } + MockAudioTransport& original_audio_transport() { return audio_transport; } + AudioTransport* audio_transport_proxy() { return registered_audio_transport; } private: - testing::StrictMock voice_engine_; - AudioState::Config config_; + testing::StrictMock mock_voice_engine; + AudioState::Config audio_state_config; + rtc::scoped_refptr audio_mixer; + MockAudioTransport audio_transport; + AudioTransport* registered_audio_transport = nullptr; }; + +class FakeAudioSource : public AudioMixer::Source { + public: + // TODO(aleloi): Valid overrides commented out, because the gmock + // methods don't use any override declarations, and we want to avoid + // warnings from -Winconsistent-missing-override. See + // http://crbug.com/428099. + int Ssrc() const /*override*/ { return 0; } + + int PreferredSampleRate() const /*override*/ { return kSampleRate; } + + MOCK_METHOD2(GetAudioFrameWithInfo, + AudioFrameInfo(int sample_rate_hz, AudioFrame* audio_frame)); +}; + } // namespace TEST(AudioStateTest, Create) { @@ -84,38 +123,52 @@ TEST(AudioStateTest, TypingNoiseDetected) { } // Test that RecordedDataIsAvailable calls get to the original transport. -TEST(AudioStateTest, RecordedAudioArrivesAtOriginalTransport) { - using testing::_; - +TEST(AudioStateAudioPathTest, RecordedAudioArrivesAtOriginalTransport) { ConfigHelper helper; - auto& voice_engine = helper.voice_engine(); - auto device = - static_cast(voice_engine.audio_device_module()); - AudioTransport* audio_transport_proxy = nullptr; - ON_CALL(*device, RegisterAudioCallback(_)) - .WillByDefault( - testing::Invoke([&audio_transport_proxy](AudioTransport* transport) { - audio_transport_proxy = transport; - return 0; + rtc::scoped_refptr audio_state = + AudioState::Create(helper.config()); + + // Setup completed. Ensure call of original transport is forwarded to new. + uint32_t new_mic_level; + EXPECT_CALL( + helper.original_audio_transport(), + RecordedDataIsAvailable(nullptr, kSampleRate / 100, kBytesPerSample, + kNumberOfChannels, kSampleRate, 0, 0, 0, false, + testing::Ref(new_mic_level))); + + helper.audio_transport_proxy()->RecordedDataIsAvailable( + nullptr, kSampleRate / 100, kBytesPerSample, kNumberOfChannels, + kSampleRate, 0, 0, 0, false, new_mic_level); +} + +TEST(AudioStateAudioPathTest, + QueryingProxyForAudioShouldResultInGetAudioCallOnMixerSource) { + ConfigHelper helper; + + rtc::scoped_refptr audio_state = + AudioState::Create(helper.config()); + + FakeAudioSource fake_source; + + helper.mixer()->AddSource(&fake_source); + + EXPECT_CALL(fake_source, GetAudioFrameWithInfo(testing::_, testing::_)) + .WillOnce( + testing::Invoke([](int sample_rate_hz, AudioFrame* audio_frame) { + audio_frame->sample_rate_hz_ = sample_rate_hz; + audio_frame->samples_per_channel_ = sample_rate_hz / 100; + audio_frame->num_channels_ = kNumberOfChannels; + return AudioMixer::Source::AudioFrameInfo::kNormal; })); - MockAudioTransport original_audio_transport; - ON_CALL(voice_engine, audio_transport()) - .WillByDefault(testing::Return(&original_audio_transport)); - - EXPECT_CALL(voice_engine, audio_device_module()); - std::unique_ptr audio_state( - new internal::AudioState(helper.config())); - - // Setup completed. Ensure call of old transport is forwarded to new. - uint32_t new_mic_level; - EXPECT_CALL(original_audio_transport, - RecordedDataIsAvailable(nullptr, 80, 2, 1, 8000, 0, 0, 0, false, - testing::Ref(new_mic_level))); - - audio_transport_proxy->RecordedDataIsAvailable(nullptr, 80, 2, 1, 8000, 0, 0, - 0, false, new_mic_level); + int16_t audio_buffer[kSampleRate / 100 * kNumberOfChannels]; + size_t n_samples_out; + int64_t elapsed_time_ms; + int64_t ntp_time_ms; + helper.audio_transport_proxy()->NeedMorePlayData( + kSampleRate / 100, kBytesPerSample, kNumberOfChannels, kSampleRate, + audio_buffer, n_samples_out, &elapsed_time_ms, &ntp_time_ms); } } // namespace test } // namespace webrtc diff --git a/webrtc/audio/audio_transport_proxy.cc b/webrtc/audio/audio_transport_proxy.cc index ed72200379..163cc11c77 100644 --- a/webrtc/audio/audio_transport_proxy.cc +++ b/webrtc/audio/audio_transport_proxy.cc @@ -12,12 +12,32 @@ namespace webrtc { +namespace { +// Resample audio in |frame| to given sample rate preserving the +// channel count and place the result in |destination|. +int Resample(const AudioFrame& frame, + const int destination_sample_rate, + PushResampler* resampler, + int16_t* destination) { + const int number_of_channels = static_cast(frame.num_channels_); + const int target_number_of_samples_per_channel = + destination_sample_rate / 100; + resampler->InitializeIfNeeded(frame.sample_rate_hz_, destination_sample_rate, + number_of_channels); + + return resampler->Resample( + frame.data_, frame.samples_per_channel_ * number_of_channels, destination, + number_of_channels * target_number_of_samples_per_channel); +} +} // namespace + AudioTransportProxy::AudioTransportProxy(AudioTransport* voe_audio_transport, AudioProcessing* apm, AudioMixer* mixer) - : voe_audio_transport_(voe_audio_transport) { + : voe_audio_transport_(voe_audio_transport), apm_(apm), mixer_(mixer) { RTC_DCHECK(voe_audio_transport); RTC_DCHECK(apm); + RTC_DCHECK(mixer); } AudioTransportProxy::~AudioTransportProxy() {} @@ -53,14 +73,23 @@ int32_t AudioTransportProxy::NeedMorePlayData(const size_t nSamples, RTC_DCHECK_GE( samplesPerSec, static_cast(AudioProcessing::NativeRate::kSampleRate8kHz)); + + // 100 = 1 second / data duration (10 ms). RTC_DCHECK_EQ(nSamples * 100, samplesPerSec); RTC_DCHECK_LE(nBytesPerSample * nSamples * nChannels, sizeof(AudioFrame::data_)); - // Pass call through to original audio transport instance. - return voe_audio_transport_->NeedMorePlayData( - nSamples, nBytesPerSample, nChannels, samplesPerSec, audioSamples, - nSamplesOut, elapsed_time_ms, ntp_time_ms); + mixer_->Mix(nChannels, &mixed_frame_); + *elapsed_time_ms = mixed_frame_.elapsed_time_ms_; + *ntp_time_ms = mixed_frame_.ntp_time_ms_; + + const auto error = apm_->ProcessReverseStream(&mixed_frame_); + RTC_DCHECK_EQ(error, AudioProcessing::kNoError); + + nSamplesOut = Resample(mixed_frame_, samplesPerSec, &resampler_, + static_cast(audioSamples)); + RTC_DCHECK_EQ(nSamplesOut, nChannels * nSamples); + return 0; } void AudioTransportProxy::PushCaptureData(int voe_channel, @@ -81,17 +110,25 @@ void AudioTransportProxy::PullRenderData(int bits_per_sample, void* audio_data, int64_t* elapsed_time_ms, int64_t* ntp_time_ms) { - RTC_DCHECK_EQ(static_cast(bits_per_sample), 8 * sizeof(int16_t)); + RTC_DCHECK_EQ(static_cast(bits_per_sample), 16); RTC_DCHECK_GE(number_of_channels, 1u); RTC_DCHECK_LE(number_of_channels, 2u); RTC_DCHECK_GE(static_cast(sample_rate), AudioProcessing::NativeRate::kSampleRate8kHz); + + // 100 = 1 second / data duration (10 ms). RTC_DCHECK_EQ(static_cast(number_of_frames * 100), sample_rate); + + // 8 = bits per byte. RTC_DCHECK_LE(bits_per_sample / 8 * number_of_frames * number_of_channels, sizeof(AudioFrame::data_)); - voe_audio_transport_->PullRenderData( - bits_per_sample, sample_rate, number_of_channels, number_of_frames, - audio_data, elapsed_time_ms, ntp_time_ms); + mixer_->Mix(number_of_channels, &mixed_frame_); + *elapsed_time_ms = mixed_frame_.elapsed_time_ms_; + *ntp_time_ms = mixed_frame_.ntp_time_ms_; + + const auto output_samples = Resample(mixed_frame_, sample_rate, &resampler_, + static_cast(audio_data)); + RTC_DCHECK_EQ(output_samples, number_of_channels * number_of_frames); } } // namespace webrtc diff --git a/webrtc/audio/audio_transport_proxy.h b/webrtc/audio/audio_transport_proxy.h index 05e52fce85..1d03378402 100644 --- a/webrtc/audio/audio_transport_proxy.h +++ b/webrtc/audio/audio_transport_proxy.h @@ -13,6 +13,8 @@ #include "webrtc/api/audio/audio_mixer.h" #include "webrtc/base/constructormagic.h" +#include "webrtc/base/scoped_ref_ptr.h" +#include "webrtc/common_audio/resampler/include/push_resampler.h" #include "webrtc/modules/audio_device/include/audio_device_defines.h" #include "webrtc/modules/audio_processing/include/audio_processing.h" @@ -63,7 +65,11 @@ class AudioTransportProxy : public AudioTransport { private: AudioTransport* voe_audio_transport_; - AudioFrame frame_for_mixing_; + AudioProcessing* apm_; + rtc::scoped_refptr mixer_; + AudioFrame mixed_frame_; + // Converts mixed audio to the audio device output rate. + PushResampler resampler_; RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioTransportProxy); };