gecko/content/media/webrtc/MediaEngineWebRTCAudio.cpp

/* This Source Code Form is subject to the terms of the Mozilla Public
 * License, v. 2.0. If a copy of the MPL was not distributed with this file,
 * You can obtain one at http://mozilla.org/MPL/2.0/. */

#include "MediaEngineWebRTC.h"
#include <stdio.h>
#include <algorithm>
#include "mozilla/Assertions.h"

// scoped_ptr.h uses FF
#ifdef FF
#undef FF
#endif
#include "webrtc/modules/audio_device/opensl/single_rw_fifo.h"

#define CHANNELS 1
#define ENCODING "L16"
#define DEFAULT_PORT 5555

#define SAMPLE_RATE 256000
#define SAMPLE_FREQUENCY 16000
#define SAMPLE_LENGTH ((SAMPLE_FREQUENCY*10)/1000)

// These are restrictions from the webrtc.org code
#define MAX_CHANNELS 2
#define MAX_SAMPLING_FREQ 48000 // Hz - multiple of 100

#define MAX_AEC_FIFO_DEPTH 200 // ms - multiple of 10
static_assert(!(MAX_AEC_FIFO_DEPTH % 10), "Invalid MAX_AEC_FIFO_DEPTH");

namespace mozilla {

#ifdef LOG
#undef LOG
#endif

#ifdef PR_LOGGING
extern PRLogModuleInfo* GetMediaManagerLog();
#define LOG(msg) PR_LOG(GetMediaManagerLog(), PR_LOG_DEBUG, msg)
#else
#define LOG(msg)
#endif

/**
 * Webrtc audio source.
 */
NS_IMPL_ISUPPORTS0(MediaEngineWebRTCAudioSource)

// XXX temp until MSG supports registration
StaticAutoPtr<AudioOutputObserver> gFarendObserver;

AudioOutputObserver::AudioOutputObserver()
  : mPlayoutFreq(0)
  , mPlayoutChannels(0)
  , mChunkSize(0)
  , mSamplesSaved(0)
{
  // Buffers of 10ms chunks
  mPlayoutFifo = new webrtc::SingleRwFifo(MAX_AEC_FIFO_DEPTH/10);
}

AudioOutputObserver::~AudioOutputObserver()
{
}

void
AudioOutputObserver::Clear()
{
  while (mPlayoutFifo->size() > 0) {
    (void) mPlayoutFifo->Pop();
  }
}

FarEndAudioChunk *
AudioOutputObserver::Pop()
{
  return (FarEndAudioChunk *) mPlayoutFifo->Pop();
}

uint32_t
AudioOutputObserver::Size()
{
  return mPlayoutFifo->size();
}

// static
void
AudioOutputObserver::InsertFarEnd(const AudioDataValue *aBuffer, uint32_t aSamples, bool aOverran,
                                  int aFreq, int aChannels, AudioSampleFormat aFormat)
{
  if (mPlayoutChannels != 0) {
    if (mPlayoutChannels != static_cast<uint32_t>(aChannels)) {
      MOZ_CRASH();
    }
  } else {
    MOZ_ASSERT(aChannels <= MAX_CHANNELS);
    mPlayoutChannels = static_cast<uint32_t>(aChannels);
  }
  if (mPlayoutFreq != 0) {
    if (mPlayoutFreq != static_cast<uint32_t>(aFreq)) {
      MOZ_CRASH();
    }
  } else {
    MOZ_ASSERT(aFreq <= MAX_SAMPLING_FREQ);
    MOZ_ASSERT(!(aFreq % 100), "Sampling rate for far end data should be multiple of 100.");
    mPlayoutFreq = aFreq;
    mChunkSize = aFreq/100; // 10ms
  }

#ifdef LOG_FAREND_INSERTION
  static FILE *fp = fopen("insertfarend.pcm","wb");
#endif

  if (mSaved) {
    // flag overrun as soon as possible, and only once
    mSaved->mOverrun = aOverran;
    aOverran = false;
  }
  // Rechunk to 10ms.
  // The AnalyzeReverseStream() and WebRtcAec_BufferFarend() functions insist on 10ms
  // samples per call.  Annoying...
  while (aSamples) {
    if (!mSaved) {
      mSaved = (FarEndAudioChunk *) moz_xmalloc(sizeof(FarEndAudioChunk) +
                                                (mChunkSize * aChannels - 1)*sizeof(int16_t));
      mSaved->mSamples = mChunkSize;
      mSaved->mOverrun = aOverran;
      aOverran = false;
    }
    uint32_t to_copy = mChunkSize - mSamplesSaved;
    if (to_copy > aSamples) {
      to_copy = aSamples;
    }

    int16_t *dest = &(mSaved->mData[mSamplesSaved * aChannels]);
    ConvertAudioSamples(aBuffer, dest, to_copy * aChannels);

#ifdef LOG_FAREND_INSERTION
    if (fp) {
      fwrite(&(mSaved->mData[mSamplesSaved * aChannels]), to_copy * aChannels, sizeof(int16_t), fp);
    }
#endif
    aSamples -= to_copy;
    mSamplesSaved += to_copy;

    if (mSamplesSaved >= mChunkSize) {
      int free_slots = mPlayoutFifo->capacity() - mPlayoutFifo->size();
      if (free_slots <= 0) {
        // XXX We should flag an overrun for the reader.  We can't drop data from it due to
        // thread safety issues.
        break;
      } else {
        mPlayoutFifo->Push((int8_t *) mSaved.forget()); // takes ownership
        mSamplesSaved = 0;
      }
    }
  }
}

void
MediaEngineWebRTCAudioSource::GetName(nsAString& aName)
{
  if (mInitDone) {
    aName.Assign(mDeviceName);
  }

  return;
}

void
MediaEngineWebRTCAudioSource::GetUUID(nsAString& aUUID)
{
  if (mInitDone) {
    aUUID.Assign(mDeviceUUID);
  }

  return;
}

nsresult
MediaEngineWebRTCAudioSource::Config(bool aEchoOn, uint32_t aEcho,
                                     bool aAgcOn, uint32_t aAGC,
                                     bool aNoiseOn, uint32_t aNoise,
                                     int32_t aPlayoutDelay)
{
  LOG(("Audio config: aec: %d, agc: %d, noise: %d",
       aEchoOn ? aEcho : -1,
       aAgcOn ? aAGC : -1,
       aNoiseOn ? aNoise : -1));

  bool update_agc = (mAgcOn == aAgcOn);
  bool update_noise = (mNoiseOn == aNoiseOn);
  mAgcOn = aAgcOn;
  mNoiseOn = aNoiseOn;

  if ((webrtc::AgcModes) aAGC != webrtc::kAgcUnchanged) {
    if (mAGC != (webrtc::AgcModes) aAGC) {
      update_agc = true;
      mAGC = (webrtc::AgcModes) aAGC;
    }
  }
  if ((webrtc::NsModes) aNoise != webrtc::kNsUnchanged) {
    if (mNoiseSuppress != (webrtc::NsModes) aNoise) {
      update_noise = true;
      mNoiseSuppress = (webrtc::NsModes) aNoise;
    }
  }
  mPlayoutDelay = aPlayoutDelay;

  if (mInitDone) {
    int error;
#if 0
    // Until we can support feeding our full output audio from the browser
    // through the MediaStream, this won't work.  Or we need to move AEC to
    // below audio input and output, perhaps invoked from here.
    mEchoOn = aEchoOn;
    if ((webrtc::EcModes) aEcho != webrtc::kEcUnchanged)
      mEchoCancel = (webrtc::EcModes) aEcho;
    mVoEProcessing->SetEcStatus(mEchoOn, aEcho);
#else
    (void) aEcho; (void) aEchoOn; (void) mEchoCancel; // suppress warnings
#endif

    if (update_agc &&
      0 != (error = mVoEProcessing->SetAgcStatus(mAgcOn, (webrtc::AgcModes) aAGC))) {
      LOG(("%s Error setting AGC Status: %d ",__FUNCTION__, error));
    }
    if (update_noise &&
      0 != (error = mVoEProcessing->SetNsStatus(mNoiseOn, (webrtc::NsModes) aNoise))) {
      LOG(("%s Error setting NoiseSuppression Status: %d ",__FUNCTION__, error));
    }
  }
  return NS_OK;
}

nsresult
MediaEngineWebRTCAudioSource::Allocate(const MediaEnginePrefs &aPrefs)
{
  if (mState == kReleased) {
    if (mInitDone) {
      ScopedCustomReleasePtr<webrtc::VoEHardware> ptrVoEHw(webrtc::VoEHardware::GetInterface(mVoiceEngine));
      if (!ptrVoEHw || ptrVoEHw->SetRecordingDevice(mCapIndex)) {
        return NS_ERROR_FAILURE;
      }
      mState = kAllocated;
      LOG(("Audio device %d allocated", mCapIndex));
    } else {
      LOG(("Audio device is not initalized"));
      return NS_ERROR_FAILURE;
    }
  } else if (mSources.IsEmpty()) {
    LOG(("Audio device %d reallocated", mCapIndex));
  } else {
    LOG(("Audio device %d allocated shared", mCapIndex));
  }
  return NS_OK;
}

nsresult
MediaEngineWebRTCAudioSource::Deallocate()
{
  if (mSources.IsEmpty()) {
    if (mState != kStopped && mState != kAllocated) {
      return NS_ERROR_FAILURE;
    }

    mState = kReleased;
    LOG(("Audio device %d deallocated", mCapIndex));
  } else {
    LOG(("Audio device %d deallocated but still in use", mCapIndex));
  }
  return NS_OK;
}

nsresult
MediaEngineWebRTCAudioSource::Start(SourceMediaStream* aStream, TrackID aID)
{
  if (!mInitDone || !aStream) {
    return NS_ERROR_FAILURE;
  }

  {
    MonitorAutoLock lock(mMonitor);
    mSources.AppendElement(aStream);
  }

  AudioSegment* segment = new AudioSegment();
  aStream->AddTrack(aID, SAMPLE_FREQUENCY, 0, segment);
  aStream->AdvanceKnownTracksTime(STREAM_TIME_MAX);
  // XXX Make this based on the pref.
  aStream->RegisterForAudioMixing();
  LOG(("Start audio for stream %p", aStream));

  if (mState == kStarted) {
    MOZ_ASSERT(aID == mTrackID);
    return NS_OK;
  }
  mState = kStarted;
  mTrackID = aID;

  // Make sure logger starts before capture
  AsyncLatencyLogger::Get(true);

  // Register output observer
  // XXX
  MOZ_ASSERT(gFarendObserver);
  gFarendObserver->Clear();

  // Configure audio processing in webrtc code
  Config(mEchoOn, webrtc::kEcUnchanged,
         mAgcOn, webrtc::kAgcUnchanged,
         mNoiseOn, webrtc::kNsUnchanged,
         mPlayoutDelay);

  if (mVoEBase->StartReceive(mChannel)) {
    return NS_ERROR_FAILURE;
  }
  if (mVoEBase->StartSend(mChannel)) {
    return NS_ERROR_FAILURE;
  }

  // Attach external media processor, so this::Process will be called.
  mVoERender->RegisterExternalMediaProcessing(mChannel, webrtc::kRecordingPerChannel, *this);

  return NS_OK;
}

nsresult
MediaEngineWebRTCAudioSource::Stop(SourceMediaStream *aSource, TrackID aID)
{
  {
    MonitorAutoLock lock(mMonitor);

    if (!mSources.RemoveElement(aSource)) {
      // Already stopped - this is allowed
      return NS_OK;
    }
    if (!mSources.IsEmpty()) {
      return NS_OK;
    }
    if (mState != kStarted) {
      return NS_ERROR_FAILURE;
    }
    if (!mVoEBase) {
      return NS_ERROR_FAILURE;
    }

    mState = kStopped;
    aSource->EndTrack(aID);
  }

  mVoERender->DeRegisterExternalMediaProcessing(mChannel, webrtc::kRecordingPerChannel);

  if (mVoEBase->StopSend(mChannel)) {
    return NS_ERROR_FAILURE;
  }
  if (mVoEBase->StopReceive(mChannel)) {
    return NS_ERROR_FAILURE;
  }
  return NS_OK;
}

void
MediaEngineWebRTCAudioSource::NotifyPull(MediaStreamGraph* aGraph,
                                         SourceMediaStream *aSource,
                                         TrackID aID,
                                         StreamTime aDesiredTime,
                                         TrackTicks &aLastEndTime)
{
  // Ignore - we push audio data
#ifdef DEBUG
  TrackTicks target = TimeToTicksRoundUp(SAMPLE_FREQUENCY, aDesiredTime);
  TrackTicks delta = target - aLastEndTime;
  LOG(("Audio: NotifyPull: aDesiredTime %ld, target %ld, delta %ld",(int64_t) aDesiredTime, (int64_t) target, (int64_t) delta));
  aLastEndTime = target;
#endif
}

nsresult
MediaEngineWebRTCAudioSource::Snapshot(uint32_t aDuration, nsIDOMFile** aFile)
{
   return NS_ERROR_NOT_IMPLEMENTED;
}

void
MediaEngineWebRTCAudioSource::Init()
{
  mVoEBase = webrtc::VoEBase::GetInterface(mVoiceEngine);

  mVoEBase->Init();

  mVoERender = webrtc::VoEExternalMedia::GetInterface(mVoiceEngine);
  if (!mVoERender) {
    return;
  }
  mVoENetwork = webrtc::VoENetwork::GetInterface(mVoiceEngine);
  if (!mVoENetwork) {
    return;
  }

  mVoEProcessing = webrtc::VoEAudioProcessing::GetInterface(mVoiceEngine);
  if (!mVoEProcessing) {
    return;
  }

  mChannel = mVoEBase->CreateChannel();
  if (mChannel < 0) {
    return;
  }
  mNullTransport = new NullTransport();
  if (mVoENetwork->RegisterExternalTransport(mChannel, *mNullTransport)) {
    return;
  }

  // Check for availability.
  ScopedCustomReleasePtr<webrtc::VoEHardware> ptrVoEHw(webrtc::VoEHardware::GetInterface(mVoiceEngine));
  if (!ptrVoEHw || ptrVoEHw->SetRecordingDevice(mCapIndex)) {
    return;
  }

#ifndef MOZ_B2G
  // Because of the permission mechanism of B2G, we need to skip the status
  // check here.
  bool avail = false;
  ptrVoEHw->GetRecordingDeviceStatus(avail);
  if (!avail) {
    return;
  }
#endif // MOZ_B2G

  // Set "codec" to PCM, 32kHz on 1 channel
  ScopedCustomReleasePtr<webrtc::VoECodec> ptrVoECodec(webrtc::VoECodec::GetInterface(mVoiceEngine));
  if (!ptrVoECodec) {
    return;
  }

  webrtc::CodecInst codec;
  strcpy(codec.plname, ENCODING);
  codec.channels = CHANNELS;
  codec.rate = SAMPLE_RATE;
  codec.plfreq = SAMPLE_FREQUENCY;
  codec.pacsize = SAMPLE_LENGTH;
  codec.pltype = 0; // Default payload type

  if (!ptrVoECodec->SetSendCodec(mChannel, codec)) {
    mInitDone = true;
  }
}

void
MediaEngineWebRTCAudioSource::Shutdown()
{
  if (!mInitDone) {
    // duplicate these here in case we failed during Init()
    if (mChannel != -1) {
      mVoENetwork->DeRegisterExternalTransport(mChannel);
    }

    if (mNullTransport) {
      delete mNullTransport;
    }

    return;
  }

  if (mState == kStarted) {
    while (!mSources.IsEmpty()) {
      Stop(mSources[0], kAudioTrack); // XXX change to support multiple tracks
    }
    MOZ_ASSERT(mState == kStopped);
  }

  if (mState == kAllocated || mState == kStopped) {
    Deallocate();
  }

  mVoEBase->Terminate();
  if (mChannel != -1) {
    mVoENetwork->DeRegisterExternalTransport(mChannel);
  }

  if (mNullTransport) {
    delete mNullTransport;
  }

  mVoEProcessing = nullptr;
  mVoENetwork = nullptr;
  mVoERender = nullptr;
  mVoEBase = nullptr;

  mState = kReleased;
  mInitDone = false;
}

typedef int16_t sample;

void
MediaEngineWebRTCAudioSource::Process(int channel,
  webrtc::ProcessingTypes type, sample* audio10ms,
  int length, int samplingFreq, bool isStereo)
{
  // On initial capture, throw away all far-end data except the most recent sample
  // since it's already irrelevant and we want to keep avoid confusing the AEC far-end
  // input code with "old" audio.
  if (!mStarted) {
    mStarted  = true;
    while (gFarendObserver->Size() > 1) {
      FarEndAudioChunk *buffer = gFarendObserver->Pop(); // only call if size() > 0
      free(buffer);
    }
  }

  while (gFarendObserver->Size() > 0) {
    FarEndAudioChunk *buffer = gFarendObserver->Pop(); // only call if size() > 0
    if (buffer) {
      int length = buffer->mSamples;
      if (mVoERender->ExternalPlayoutData(buffer->mData,
                                          gFarendObserver->PlayoutFrequency(),
                                          gFarendObserver->PlayoutChannels(),
                                          mPlayoutDelay,
                                          length) == -1) {
        return;
      }
    }
    free(buffer);
  }

  MonitorAutoLock lock(mMonitor);
  if (mState != kStarted)
    return;

  uint32_t len = mSources.Length();
  for (uint32_t i = 0; i < len; i++) {
    nsRefPtr<SharedBuffer> buffer = SharedBuffer::Create(length * sizeof(sample));

    sample* dest = static_cast<sample*>(buffer->Data());
    memcpy(dest, audio10ms, length * sizeof(sample));

    AudioSegment segment;
    nsAutoTArray<const sample*,1> channels;
    channels.AppendElement(dest);
    segment.AppendFrames(buffer.forget(), channels, length);
    TimeStamp insertTime;
    segment.GetStartTime(insertTime);

    SourceMediaStream *source = mSources[i];
    if (source) {
      // This is safe from any thread, and is safe if the track is Finished
      // or Destroyed.
      // Make sure we include the stream and the track.
      // The 0:1 is a flag to note when we've done the final insert for a given input block.
      LogTime(AsyncLatencyLogger::AudioTrackInsertion, LATENCY_STREAM_ID(source, mTrackID),
              (i+1 < len) ? 0 : 1, insertTime);

      source->AppendToTrack(mTrackID, &segment);
    }
  }

  return;
}

}