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Replace AudioConferenceMixer with AudioMixer.

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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}
This commit is contained in:
aleloi
2016-11-22 06:42:53 -08:00
committed by Commit bot
parent b426040788
commit 04c0722cf1
8 changed files with 239 additions and 73 deletions
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3
webrtc/audio/BUILD.gn
View File

@ -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",

1
webrtc/audio/DEPS
View File

@ -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",

63
webrtc/audio/audio_receive_stream.cc
View File

@ -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();
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<VoEBase> 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<VoEBase> 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<VoECodec> codec(voice_engine());
@ -216,17 +235,17 @@ webrtc::AudioReceiveStream::Stats AudioReceiveStream::GetStats() const {
}
void AudioReceiveStream::SetSink(std::unique_ptr<AudioSinkInterface> 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<internal::AudioState*>(audio_state_.get());
RTC_DCHECK(audio_state);
return audio_state;
}
VoiceEngine* AudioReceiveStream::voice_engine() const {
internal::AudioState* audio_state =
static_cast<internal::AudioState*>(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<VoEBase> base(voice_engine());
if (playout) {
return base->StartPlayout(config_.voe_channel_id);
} else {
return base->StopPlayout(config_.voe_channel_id);
}
}
} // namespace internal
} // namespace webrtc

5
webrtc/audio/audio_receive_stream.h
View File

@ -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<RtpHeaderParser> rtp_header_parser_;
std::unique_ptr<voe::ChannelProxy> channel_proxy_;
bool playing_ ACCESS_ON(thread_checker_) = false;
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioReceiveStream);
};
} // namespace internal

36
webrtc/audio/audio_receive_stream_unittest.cc
View File

@ -11,11 +11,11 @@
#include <string>
#include <vector>
#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<MockAudioMixer>()) {
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<AudioState> audio_state() { return audio_state_; }
rtc::scoped_refptr<MockAudioMixer> 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<MockVoiceEngine> voice_engine_;
rtc::scoped_refptr<AudioState> audio_state_;
rtc::scoped_refptr<MockAudioMixer> audio_mixer_;
AudioReceiveStream::Config stream_config_;
testing::StrictMock<MockVoEChannelProxy>* 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

135
webrtc/audio/audio_state_unittest.cc
View File

@ -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<MockAudioDeviceModule*>(
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<AudioMixer> mixer() { return audio_mixer; }
MockAudioTransport& original_audio_transport() { return audio_transport; }
AudioTransport* audio_transport_proxy() { return registered_audio_transport; }
private:
testing::StrictMock<MockVoiceEngine> voice_engine_;
AudioState::Config config_;
testing::StrictMock<MockVoiceEngine> mock_voice_engine;
AudioState::Config audio_state_config;
rtc::scoped_refptr<AudioMixer> 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<MockAudioDeviceModule*>(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<AudioState> audio_state =
AudioState::Create(helper.config());
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<internal::AudioState> audio_state(
new internal::AudioState(helper.config()));
// Setup completed. Ensure call of old transport is forwarded to new.
// Setup completed. Ensure call of original 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,
EXPECT_CALL(
helper.original_audio_transport(),
RecordedDataIsAvailable(nullptr, kSampleRate / 100, kBytesPerSample,
kNumberOfChannels, kSampleRate, 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);
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<AudioState> 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;
}));
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

55
webrtc/audio/audio_transport_proxy.cc
View File

@ -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<int16_t>* resampler,
int16_t* destination) {
const int number_of_channels = static_cast<int>(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<uint32_t>(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<int16_t*>(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<size_t>(bits_per_sample), 8 * sizeof(int16_t));
RTC_DCHECK_EQ(static_cast<size_t>(bits_per_sample), 16);
RTC_DCHECK_GE(number_of_channels, 1u);
RTC_DCHECK_LE(number_of_channels, 2u);
RTC_DCHECK_GE(static_cast<int>(sample_rate),
AudioProcessing::NativeRate::kSampleRate8kHz);
// 100 = 1 second / data duration (10 ms).
RTC_DCHECK_EQ(static_cast<int>(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<int16_t*>(audio_data));
RTC_DCHECK_EQ(output_samples, number_of_channels * number_of_frames);
}
} // namespace webrtc

8
webrtc/audio/audio_transport_proxy.h
View File

@ -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<AudioMixer> mixer_;
AudioFrame mixed_frame_;
// Converts mixed audio to the audio device output rate.
PushResampler<int16_t> resampler_;
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioTransportProxy);
};