michael@0: /* This Source Code Form is subject to the terms of the Mozilla Public
michael@0:  * License, v. 2.0. If a copy of the MPL was not distributed with this file,
michael@0:  * You can obtain one at http://mozilla.org/MPL/2.0/. */
michael@0: 
michael@0: #include "MediaEngineWebRTC.h"
michael@0: #include <stdio.h>
michael@0: #include <algorithm>
michael@0: #include "mozilla/Assertions.h"
michael@0: #include "MediaTrackConstraints.h"
michael@0: 
michael@0: // scoped_ptr.h uses FF
michael@0: #ifdef FF
michael@0: #undef FF
michael@0: #endif
michael@0: #include "webrtc/modules/audio_device/opensl/single_rw_fifo.h"
michael@0: 
michael@0: #define CHANNELS 1
michael@0: #define ENCODING "L16"
michael@0: #define DEFAULT_PORT 5555
michael@0: 
michael@0: #define SAMPLE_RATE 256000
michael@0: #define SAMPLE_FREQUENCY 16000
michael@0: #define SAMPLE_LENGTH ((SAMPLE_FREQUENCY*10)/1000)
michael@0: 
michael@0: // These are restrictions from the webrtc.org code
michael@0: #define MAX_CHANNELS 2
michael@0: #define MAX_SAMPLING_FREQ 48000 // Hz - multiple of 100
michael@0: 
michael@0: #define MAX_AEC_FIFO_DEPTH 200 // ms - multiple of 10
michael@0: static_assert(!(MAX_AEC_FIFO_DEPTH % 10), "Invalid MAX_AEC_FIFO_DEPTH");
michael@0: 
michael@0: namespace mozilla {
michael@0: 
michael@0: #ifdef LOG
michael@0: #undef LOG
michael@0: #endif
michael@0: 
michael@0: #ifdef PR_LOGGING
michael@0: extern PRLogModuleInfo* GetMediaManagerLog();
michael@0: #define LOG(msg) PR_LOG(GetMediaManagerLog(), PR_LOG_DEBUG, msg)
michael@0: #else
michael@0: #define LOG(msg)
michael@0: #endif
michael@0: 
michael@0: /**
michael@0:  * Webrtc audio source.
michael@0:  */
michael@0: NS_IMPL_ISUPPORTS0(MediaEngineWebRTCAudioSource)
michael@0: 
michael@0: // XXX temp until MSG supports registration
michael@0: StaticAutoPtr<AudioOutputObserver> gFarendObserver;
michael@0: 
michael@0: AudioOutputObserver::AudioOutputObserver()
michael@0:   : mPlayoutFreq(0)
michael@0:   , mPlayoutChannels(0)
michael@0:   , mChunkSize(0)
michael@0:   , mSamplesSaved(0)
michael@0: {
michael@0:   // Buffers of 10ms chunks
michael@0:   mPlayoutFifo = new webrtc::SingleRwFifo(MAX_AEC_FIFO_DEPTH/10);
michael@0: }
michael@0: 
michael@0: AudioOutputObserver::~AudioOutputObserver()
michael@0: {
michael@0: }
michael@0: 
michael@0: void
michael@0: AudioOutputObserver::Clear()
michael@0: {
michael@0:   while (mPlayoutFifo->size() > 0) {
michael@0:     (void) mPlayoutFifo->Pop();
michael@0:   }
michael@0: }
michael@0: 
michael@0: FarEndAudioChunk *
michael@0: AudioOutputObserver::Pop()
michael@0: {
michael@0:   return (FarEndAudioChunk *) mPlayoutFifo->Pop();
michael@0: }
michael@0: 
michael@0: uint32_t
michael@0: AudioOutputObserver::Size()
michael@0: {
michael@0:   return mPlayoutFifo->size();
michael@0: }
michael@0: 
michael@0: // static
michael@0: void
michael@0: AudioOutputObserver::InsertFarEnd(const AudioDataValue *aBuffer, uint32_t aSamples, bool aOverran,
michael@0:                                   int aFreq, int aChannels, AudioSampleFormat aFormat)
michael@0: {
michael@0:   if (mPlayoutChannels != 0) {
michael@0:     if (mPlayoutChannels != static_cast<uint32_t>(aChannels)) {
michael@0:       MOZ_CRASH();
michael@0:     }
michael@0:   } else {
michael@0:     MOZ_ASSERT(aChannels <= MAX_CHANNELS);
michael@0:     mPlayoutChannels = static_cast<uint32_t>(aChannels);
michael@0:   }
michael@0:   if (mPlayoutFreq != 0) {
michael@0:     if (mPlayoutFreq != static_cast<uint32_t>(aFreq)) {
michael@0:       MOZ_CRASH();
michael@0:     }
michael@0:   } else {
michael@0:     MOZ_ASSERT(aFreq <= MAX_SAMPLING_FREQ);
michael@0:     MOZ_ASSERT(!(aFreq % 100), "Sampling rate for far end data should be multiple of 100.");
michael@0:     mPlayoutFreq = aFreq;
michael@0:     mChunkSize = aFreq/100; // 10ms
michael@0:   }
michael@0: 
michael@0: #ifdef LOG_FAREND_INSERTION
michael@0:   static FILE *fp = fopen("insertfarend.pcm","wb");
michael@0: #endif
michael@0: 
michael@0:   if (mSaved) {
michael@0:     // flag overrun as soon as possible, and only once
michael@0:     mSaved->mOverrun = aOverran;
michael@0:     aOverran = false;
michael@0:   }
michael@0:   // Rechunk to 10ms.
michael@0:   // The AnalyzeReverseStream() and WebRtcAec_BufferFarend() functions insist on 10ms
michael@0:   // samples per call.  Annoying...
michael@0:   while (aSamples) {
michael@0:     if (!mSaved) {
michael@0:       mSaved = (FarEndAudioChunk *) moz_xmalloc(sizeof(FarEndAudioChunk) +
michael@0:                                                 (mChunkSize * aChannels - 1)*sizeof(int16_t));
michael@0:       mSaved->mSamples = mChunkSize;
michael@0:       mSaved->mOverrun = aOverran;
michael@0:       aOverran = false;
michael@0:     }
michael@0:     uint32_t to_copy = mChunkSize - mSamplesSaved;
michael@0:     if (to_copy > aSamples) {
michael@0:       to_copy = aSamples;
michael@0:     }
michael@0: 
michael@0:     int16_t *dest = &(mSaved->mData[mSamplesSaved * aChannels]);
michael@0:     ConvertAudioSamples(aBuffer, dest, to_copy * aChannels);
michael@0: 
michael@0: #ifdef LOG_FAREND_INSERTION
michael@0:     if (fp) {
michael@0:       fwrite(&(mSaved->mData[mSamplesSaved * aChannels]), to_copy * aChannels, sizeof(int16_t), fp);
michael@0:     }
michael@0: #endif
michael@0:     aSamples -= to_copy;
michael@0:     mSamplesSaved += to_copy;
michael@0:     aBuffer += to_copy * aChannels;
michael@0: 
michael@0:     if (mSamplesSaved >= mChunkSize) {
michael@0:       int free_slots = mPlayoutFifo->capacity() - mPlayoutFifo->size();
michael@0:       if (free_slots <= 0) {
michael@0:         // XXX We should flag an overrun for the reader.  We can't drop data from it due to
michael@0:         // thread safety issues.
michael@0:         break;
michael@0:       } else {
michael@0:         mPlayoutFifo->Push((int8_t *) mSaved.forget()); // takes ownership
michael@0:         mSamplesSaved = 0;
michael@0:       }
michael@0:     }
michael@0:   }
michael@0: }
michael@0: 
michael@0: void
michael@0: MediaEngineWebRTCAudioSource::GetName(nsAString& aName)
michael@0: {
michael@0:   if (mInitDone) {
michael@0:     aName.Assign(mDeviceName);
michael@0:   }
michael@0: 
michael@0:   return;
michael@0: }
michael@0: 
michael@0: void
michael@0: MediaEngineWebRTCAudioSource::GetUUID(nsAString& aUUID)
michael@0: {
michael@0:   if (mInitDone) {
michael@0:     aUUID.Assign(mDeviceUUID);
michael@0:   }
michael@0: 
michael@0:   return;
michael@0: }
michael@0: 
michael@0: nsresult
michael@0: MediaEngineWebRTCAudioSource::Config(bool aEchoOn, uint32_t aEcho,
michael@0:                                      bool aAgcOn, uint32_t aAGC,
michael@0:                                      bool aNoiseOn, uint32_t aNoise,
michael@0:                                      int32_t aPlayoutDelay)
michael@0: {
michael@0:   LOG(("Audio config: aec: %d, agc: %d, noise: %d",
michael@0:        aEchoOn ? aEcho : -1,
michael@0:        aAgcOn ? aAGC : -1,
michael@0:        aNoiseOn ? aNoise : -1));
michael@0: 
michael@0:   bool update_echo = (mEchoOn != aEchoOn);
michael@0:   bool update_agc = (mAgcOn != aAgcOn);
michael@0:   bool update_noise = (mNoiseOn != aNoiseOn);
michael@0:   mEchoOn = aEchoOn;
michael@0:   mAgcOn = aAgcOn;
michael@0:   mNoiseOn = aNoiseOn;
michael@0: 
michael@0:   if ((webrtc::EcModes) aEcho != webrtc::kEcUnchanged) {
michael@0:     if (mEchoCancel != (webrtc::EcModes) aEcho) {
michael@0:       update_echo = true;
michael@0:       mEchoCancel = (webrtc::EcModes) aEcho;
michael@0:     }
michael@0:   }
michael@0:   if ((webrtc::AgcModes) aAGC != webrtc::kAgcUnchanged) {
michael@0:     if (mAGC != (webrtc::AgcModes) aAGC) {
michael@0:       update_agc = true;
michael@0:       mAGC = (webrtc::AgcModes) aAGC;
michael@0:     }
michael@0:   }
michael@0:   if ((webrtc::NsModes) aNoise != webrtc::kNsUnchanged) {
michael@0:     if (mNoiseSuppress != (webrtc::NsModes) aNoise) {
michael@0:       update_noise = true;
michael@0:       mNoiseSuppress = (webrtc::NsModes) aNoise;
michael@0:     }
michael@0:   }
michael@0:   mPlayoutDelay = aPlayoutDelay;
michael@0: 
michael@0:   if (mInitDone) {
michael@0:     int error;
michael@0: 
michael@0:     if (update_echo &&
michael@0:       0 != (error = mVoEProcessing->SetEcStatus(mEchoOn, (webrtc::EcModes) aEcho))) {
michael@0:       LOG(("%s Error setting Echo Status: %d ",__FUNCTION__, error));
michael@0:       // Overhead of capturing all the time is very low (<0.1% of an audio only call)
michael@0:       if (mEchoOn) {
michael@0:         if (0 != (error = mVoEProcessing->SetEcMetricsStatus(true))) {
michael@0:           LOG(("%s Error setting Echo Metrics: %d ",__FUNCTION__, error));
michael@0:         }
michael@0:       }
michael@0:     }
michael@0:     if (update_agc &&
michael@0:       0 != (error = mVoEProcessing->SetAgcStatus(mAgcOn, (webrtc::AgcModes) aAGC))) {
michael@0:       LOG(("%s Error setting AGC Status: %d ",__FUNCTION__, error));
michael@0:     }
michael@0:     if (update_noise &&
michael@0:       0 != (error = mVoEProcessing->SetNsStatus(mNoiseOn, (webrtc::NsModes) aNoise))) {
michael@0:       LOG(("%s Error setting NoiseSuppression Status: %d ",__FUNCTION__, error));
michael@0:     }
michael@0:   }
michael@0:   return NS_OK;
michael@0: }
michael@0: 
michael@0: nsresult
michael@0: MediaEngineWebRTCAudioSource::Allocate(const AudioTrackConstraintsN &aConstraints,
michael@0:                                        const MediaEnginePrefs &aPrefs)
michael@0: {
michael@0:   if (mState == kReleased) {
michael@0:     if (mInitDone) {
michael@0:       ScopedCustomReleasePtr<webrtc::VoEHardware> ptrVoEHw(webrtc::VoEHardware::GetInterface(mVoiceEngine));
michael@0:       if (!ptrVoEHw || ptrVoEHw->SetRecordingDevice(mCapIndex)) {
michael@0:         return NS_ERROR_FAILURE;
michael@0:       }
michael@0:       mState = kAllocated;
michael@0:       LOG(("Audio device %d allocated", mCapIndex));
michael@0:     } else {
michael@0:       LOG(("Audio device is not initalized"));
michael@0:       return NS_ERROR_FAILURE;
michael@0:     }
michael@0:   } else if (mSources.IsEmpty()) {
michael@0:     LOG(("Audio device %d reallocated", mCapIndex));
michael@0:   } else {
michael@0:     LOG(("Audio device %d allocated shared", mCapIndex));
michael@0:   }
michael@0:   return NS_OK;
michael@0: }
michael@0: 
michael@0: nsresult
michael@0: MediaEngineWebRTCAudioSource::Deallocate()
michael@0: {
michael@0:   if (mSources.IsEmpty()) {
michael@0:     if (mState != kStopped && mState != kAllocated) {
michael@0:       return NS_ERROR_FAILURE;
michael@0:     }
michael@0: 
michael@0:     mState = kReleased;
michael@0:     LOG(("Audio device %d deallocated", mCapIndex));
michael@0:   } else {
michael@0:     LOG(("Audio device %d deallocated but still in use", mCapIndex));
michael@0:   }
michael@0:   return NS_OK;
michael@0: }
michael@0: 
michael@0: nsresult
michael@0: MediaEngineWebRTCAudioSource::Start(SourceMediaStream* aStream, TrackID aID)
michael@0: {
michael@0:   if (!mInitDone || !aStream) {
michael@0:     return NS_ERROR_FAILURE;
michael@0:   }
michael@0: 
michael@0:   {
michael@0:     MonitorAutoLock lock(mMonitor);
michael@0:     mSources.AppendElement(aStream);
michael@0:   }
michael@0: 
michael@0:   AudioSegment* segment = new AudioSegment();
michael@0:   aStream->AddTrack(aID, SAMPLE_FREQUENCY, 0, segment);
michael@0:   aStream->AdvanceKnownTracksTime(STREAM_TIME_MAX);
michael@0:   // XXX Make this based on the pref.
michael@0:   aStream->RegisterForAudioMixing();
michael@0:   LOG(("Start audio for stream %p", aStream));
michael@0: 
michael@0:   if (mState == kStarted) {
michael@0:     MOZ_ASSERT(aID == mTrackID);
michael@0:     return NS_OK;
michael@0:   }
michael@0:   mState = kStarted;
michael@0:   mTrackID = aID;
michael@0: 
michael@0:   // Make sure logger starts before capture
michael@0:   AsyncLatencyLogger::Get(true);
michael@0: 
michael@0:   // Register output observer
michael@0:   // XXX
michael@0:   MOZ_ASSERT(gFarendObserver);
michael@0:   gFarendObserver->Clear();
michael@0: 
michael@0:   // Configure audio processing in webrtc code
michael@0:   Config(mEchoOn, webrtc::kEcUnchanged,
michael@0:          mAgcOn, webrtc::kAgcUnchanged,
michael@0:          mNoiseOn, webrtc::kNsUnchanged,
michael@0:          mPlayoutDelay);
michael@0: 
michael@0:   if (mVoEBase->StartReceive(mChannel)) {
michael@0:     return NS_ERROR_FAILURE;
michael@0:   }
michael@0:   if (mVoEBase->StartSend(mChannel)) {
michael@0:     return NS_ERROR_FAILURE;
michael@0:   }
michael@0: 
michael@0:   // Attach external media processor, so this::Process will be called.
michael@0:   mVoERender->RegisterExternalMediaProcessing(mChannel, webrtc::kRecordingPerChannel, *this);
michael@0: 
michael@0:   return NS_OK;
michael@0: }
michael@0: 
michael@0: nsresult
michael@0: MediaEngineWebRTCAudioSource::Stop(SourceMediaStream *aSource, TrackID aID)
michael@0: {
michael@0:   {
michael@0:     MonitorAutoLock lock(mMonitor);
michael@0: 
michael@0:     if (!mSources.RemoveElement(aSource)) {
michael@0:       // Already stopped - this is allowed
michael@0:       return NS_OK;
michael@0:     }
michael@0:     if (!mSources.IsEmpty()) {
michael@0:       return NS_OK;
michael@0:     }
michael@0:     if (mState != kStarted) {
michael@0:       return NS_ERROR_FAILURE;
michael@0:     }
michael@0:     if (!mVoEBase) {
michael@0:       return NS_ERROR_FAILURE;
michael@0:     }
michael@0: 
michael@0:     mState = kStopped;
michael@0:     aSource->EndTrack(aID);
michael@0:   }
michael@0: 
michael@0:   mVoERender->DeRegisterExternalMediaProcessing(mChannel, webrtc::kRecordingPerChannel);
michael@0: 
michael@0:   if (mVoEBase->StopSend(mChannel)) {
michael@0:     return NS_ERROR_FAILURE;
michael@0:   }
michael@0:   if (mVoEBase->StopReceive(mChannel)) {
michael@0:     return NS_ERROR_FAILURE;
michael@0:   }
michael@0:   return NS_OK;
michael@0: }
michael@0: 
michael@0: void
michael@0: MediaEngineWebRTCAudioSource::NotifyPull(MediaStreamGraph* aGraph,
michael@0:                                          SourceMediaStream *aSource,
michael@0:                                          TrackID aID,
michael@0:                                          StreamTime aDesiredTime,
michael@0:                                          TrackTicks &aLastEndTime)
michael@0: {
michael@0:   // Ignore - we push audio data
michael@0: #ifdef DEBUG
michael@0:   TrackTicks target = TimeToTicksRoundUp(SAMPLE_FREQUENCY, aDesiredTime);
michael@0:   TrackTicks delta = target - aLastEndTime;
michael@0:   LOG(("Audio: NotifyPull: aDesiredTime %ld, target %ld, delta %ld",(int64_t) aDesiredTime, (int64_t) target, (int64_t) delta));
michael@0:   aLastEndTime = target;
michael@0: #endif
michael@0: }
michael@0: 
michael@0: nsresult
michael@0: MediaEngineWebRTCAudioSource::Snapshot(uint32_t aDuration, nsIDOMFile** aFile)
michael@0: {
michael@0:    return NS_ERROR_NOT_IMPLEMENTED;
michael@0: }
michael@0: 
michael@0: void
michael@0: MediaEngineWebRTCAudioSource::Init()
michael@0: {
michael@0:   mVoEBase = webrtc::VoEBase::GetInterface(mVoiceEngine);
michael@0: 
michael@0:   mVoEBase->Init();
michael@0: 
michael@0:   mVoERender = webrtc::VoEExternalMedia::GetInterface(mVoiceEngine);
michael@0:   if (!mVoERender) {
michael@0:     return;
michael@0:   }
michael@0:   mVoENetwork = webrtc::VoENetwork::GetInterface(mVoiceEngine);
michael@0:   if (!mVoENetwork) {
michael@0:     return;
michael@0:   }
michael@0: 
michael@0:   mVoEProcessing = webrtc::VoEAudioProcessing::GetInterface(mVoiceEngine);
michael@0:   if (!mVoEProcessing) {
michael@0:     return;
michael@0:   }
michael@0: 
michael@0:   mVoECallReport = webrtc::VoECallReport::GetInterface(mVoiceEngine);
michael@0:   if (!mVoECallReport) {
michael@0:     return;
michael@0:   }
michael@0: 
michael@0:   mChannel = mVoEBase->CreateChannel();
michael@0:   if (mChannel < 0) {
michael@0:     return;
michael@0:   }
michael@0:   mNullTransport = new NullTransport();
michael@0:   if (mVoENetwork->RegisterExternalTransport(mChannel, *mNullTransport)) {
michael@0:     return;
michael@0:   }
michael@0: 
michael@0:   // Check for availability.
michael@0:   ScopedCustomReleasePtr<webrtc::VoEHardware> ptrVoEHw(webrtc::VoEHardware::GetInterface(mVoiceEngine));
michael@0:   if (!ptrVoEHw || ptrVoEHw->SetRecordingDevice(mCapIndex)) {
michael@0:     return;
michael@0:   }
michael@0: 
michael@0: #ifndef MOZ_B2G
michael@0:   // Because of the permission mechanism of B2G, we need to skip the status
michael@0:   // check here.
michael@0:   bool avail = false;
michael@0:   ptrVoEHw->GetRecordingDeviceStatus(avail);
michael@0:   if (!avail) {
michael@0:     return;
michael@0:   }
michael@0: #endif // MOZ_B2G
michael@0: 
michael@0:   // Set "codec" to PCM, 32kHz on 1 channel
michael@0:   ScopedCustomReleasePtr<webrtc::VoECodec> ptrVoECodec(webrtc::VoECodec::GetInterface(mVoiceEngine));
michael@0:   if (!ptrVoECodec) {
michael@0:     return;
michael@0:   }
michael@0: 
michael@0:   webrtc::CodecInst codec;
michael@0:   strcpy(codec.plname, ENCODING);
michael@0:   codec.channels = CHANNELS;
michael@0:   codec.rate = SAMPLE_RATE;
michael@0:   codec.plfreq = SAMPLE_FREQUENCY;
michael@0:   codec.pacsize = SAMPLE_LENGTH;
michael@0:   codec.pltype = 0; // Default payload type
michael@0: 
michael@0:   if (!ptrVoECodec->SetSendCodec(mChannel, codec)) {
michael@0:     mInitDone = true;
michael@0:   }
michael@0: }
michael@0: 
michael@0: void
michael@0: MediaEngineWebRTCAudioSource::Shutdown()
michael@0: {
michael@0:   if (!mInitDone) {
michael@0:     // duplicate these here in case we failed during Init()
michael@0:     if (mChannel != -1) {
michael@0:       mVoENetwork->DeRegisterExternalTransport(mChannel);
michael@0:     }
michael@0: 
michael@0:     delete mNullTransport;
michael@0:     return;
michael@0:   }
michael@0: 
michael@0:   if (mState == kStarted) {
michael@0:     while (!mSources.IsEmpty()) {
michael@0:       Stop(mSources[0], kAudioTrack); // XXX change to support multiple tracks
michael@0:     }
michael@0:     MOZ_ASSERT(mState == kStopped);
michael@0:   }
michael@0: 
michael@0:   if (mState == kAllocated || mState == kStopped) {
michael@0:     Deallocate();
michael@0:   }
michael@0: 
michael@0:   mVoEBase->Terminate();
michael@0:   if (mChannel != -1) {
michael@0:     mVoENetwork->DeRegisterExternalTransport(mChannel);
michael@0:   }
michael@0: 
michael@0:   delete mNullTransport;
michael@0: 
michael@0:   mVoEProcessing = nullptr;
michael@0:   mVoENetwork = nullptr;
michael@0:   mVoERender = nullptr;
michael@0:   mVoEBase = nullptr;
michael@0: 
michael@0:   mState = kReleased;
michael@0:   mInitDone = false;
michael@0: }
michael@0: 
michael@0: typedef int16_t sample;
michael@0: 
michael@0: void
michael@0: MediaEngineWebRTCAudioSource::Process(int channel,
michael@0:   webrtc::ProcessingTypes type, sample* audio10ms,
michael@0:   int length, int samplingFreq, bool isStereo)
michael@0: {
michael@0:   // On initial capture, throw away all far-end data except the most recent sample
michael@0:   // since it's already irrelevant and we want to keep avoid confusing the AEC far-end
michael@0:   // input code with "old" audio.
michael@0:   if (!mStarted) {
michael@0:     mStarted  = true;
michael@0:     while (gFarendObserver->Size() > 1) {
michael@0:       FarEndAudioChunk *buffer = gFarendObserver->Pop(); // only call if size() > 0
michael@0:       free(buffer);
michael@0:     }
michael@0:   }
michael@0: 
michael@0:   while (gFarendObserver->Size() > 0) {
michael@0:     FarEndAudioChunk *buffer = gFarendObserver->Pop(); // only call if size() > 0
michael@0:     if (buffer) {
michael@0:       int length = buffer->mSamples;
michael@0:       if (mVoERender->ExternalPlayoutData(buffer->mData,
michael@0:                                           gFarendObserver->PlayoutFrequency(),
michael@0:                                           gFarendObserver->PlayoutChannels(),
michael@0:                                           mPlayoutDelay,
michael@0:                                           length) == -1) {
michael@0:         return;
michael@0:       }
michael@0:     }
michael@0:     free(buffer);
michael@0:   }
michael@0: 
michael@0: #ifdef PR_LOGGING
michael@0:   mSamples += length;
michael@0:   if (mSamples > samplingFreq) {
michael@0:     mSamples %= samplingFreq; // just in case mSamples >> samplingFreq
michael@0:     if (PR_LOG_TEST(GetMediaManagerLog(), PR_LOG_DEBUG)) {
michael@0:       webrtc::EchoStatistics echo;
michael@0: 
michael@0:       mVoECallReport->GetEchoMetricSummary(echo);
michael@0: #define DUMP_STATVAL(x) (x).min, (x).max, (x).average
michael@0:       LOG(("Echo: ERL: %d/%d/%d, ERLE: %d/%d/%d, RERL: %d/%d/%d, NLP: %d/%d/%d",
michael@0:            DUMP_STATVAL(echo.erl),
michael@0:            DUMP_STATVAL(echo.erle),
michael@0:            DUMP_STATVAL(echo.rerl),
michael@0:            DUMP_STATVAL(echo.a_nlp)));
michael@0:     }
michael@0:   }
michael@0: #endif
michael@0: 
michael@0:   MonitorAutoLock lock(mMonitor);
michael@0:   if (mState != kStarted)
michael@0:     return;
michael@0: 
michael@0:   uint32_t len = mSources.Length();
michael@0:   for (uint32_t i = 0; i < len; i++) {
michael@0:     nsRefPtr<SharedBuffer> buffer = SharedBuffer::Create(length * sizeof(sample));
michael@0: 
michael@0:     sample* dest = static_cast<sample*>(buffer->Data());
michael@0:     memcpy(dest, audio10ms, length * sizeof(sample));
michael@0: 
michael@0:     AudioSegment segment;
michael@0:     nsAutoTArray<const sample*,1> channels;
michael@0:     channels.AppendElement(dest);
michael@0:     segment.AppendFrames(buffer.forget(), channels, length);
michael@0:     TimeStamp insertTime;
michael@0:     segment.GetStartTime(insertTime);
michael@0: 
michael@0:     SourceMediaStream *source = mSources[i];
michael@0:     if (source) {
michael@0:       // This is safe from any thread, and is safe if the track is Finished
michael@0:       // or Destroyed.
michael@0:       // Make sure we include the stream and the track.
michael@0:       // The 0:1 is a flag to note when we've done the final insert for a given input block.
michael@0:       LogTime(AsyncLatencyLogger::AudioTrackInsertion, LATENCY_STREAM_ID(source, mTrackID),
michael@0:               (i+1 < len) ? 0 : 1, insertTime);
michael@0: 
michael@0:       source->AppendToTrack(mTrackID, &segment);
michael@0:     }
michael@0:   }
michael@0: 
michael@0:   return;
michael@0: }
michael@0: 
michael@0: }