示例#1
0
long
AudioStream::GetUnprocessed(void* aBuffer, long aFrames, int64_t &aTimeMs)
{
  mMonitor.AssertCurrentThreadOwns();
  uint8_t* wpos = reinterpret_cast<uint8_t*>(aBuffer);

  // Flush the timestretcher pipeline, if we were playing using a playback rate
  // other than 1.0.
  uint32_t flushedFrames = 0;
  if (mTimeStretcher && mTimeStretcher->numSamples()) {
    flushedFrames = mTimeStretcher->receiveSamples(reinterpret_cast<AudioDataValue*>(wpos), aFrames);
    wpos += FramesToBytes(flushedFrames);
  }
  uint32_t toPopBytes = FramesToBytes(aFrames - flushedFrames);
  uint32_t available = std::min(toPopBytes, mBuffer.Length());

  void* input[2];
  uint32_t input_size[2];
  mBuffer.PopElements(available, &input[0], &input_size[0], &input[1], &input_size[1]);
  memcpy(wpos, input[0], input_size[0]);
  wpos += input_size[0];
  memcpy(wpos, input[1], input_size[1]);

  // First time block now has our first returned sample
  mReadPoint += BytesToFrames(available);
  GetBufferInsertTime(aTimeMs);

  return BytesToFrames(available) + flushedFrames;
}
示例#2
0
long
AudioStream::GetTimeStretched(void* aBuffer, long aFrames, int64_t &aTimeMs)
{
  mMonitor.AssertCurrentThreadOwns();
  long processedFrames = 0;

  // We need to call the non-locking version, because we already have the lock.
  if (EnsureTimeStretcherInitializedUnlocked() != NS_OK) {
    return 0;
  }

  uint8_t* wpos = reinterpret_cast<uint8_t*>(aBuffer);
  double playbackRate = static_cast<double>(mInRate) / mOutRate;
  uint32_t toPopBytes = FramesToBytes(ceil(aFrames * playbackRate));
  uint32_t available = 0;
  bool lowOnBufferedData = false;
  do {
    // Check if we already have enough data in the time stretcher pipeline.
    if (mTimeStretcher->numSamples() <= static_cast<uint32_t>(aFrames)) {
      void* input[2];
      uint32_t input_size[2];
      available = std::min(mBuffer.Length(), toPopBytes);
      if (available != toPopBytes) {
        lowOnBufferedData = true;
      }
      mBuffer.PopElements(available, &input[0], &input_size[0],
                                     &input[1], &input_size[1]);
      mReadPoint += BytesToFrames(available);
      for(uint32_t i = 0; i < 2; i++) {
        mTimeStretcher->putSamples(reinterpret_cast<AudioDataValue*>(input[i]), BytesToFrames(input_size[i]));
      }
    }
    uint32_t receivedFrames = mTimeStretcher->receiveSamples(reinterpret_cast<AudioDataValue*>(wpos), aFrames - processedFrames);
    wpos += FramesToBytes(receivedFrames);
    processedFrames += receivedFrames;
  } while (processedFrames < aFrames && !lowOnBufferedData);

  GetBufferInsertTime(aTimeMs);

  return processedFrames;
}
示例#3
0
// Get unprocessed samples, and pad the beginning of the buffer with silence if
// there is not enough data.
long
AudioStream::GetUnprocessedWithSilencePadding(void* aBuffer, long aFrames, int64_t& aTimeMs)
{
  uint32_t toPopBytes = FramesToBytes(aFrames);
  uint32_t available = std::min(toPopBytes, mBuffer.Length());
  uint32_t silenceOffset = toPopBytes - available;

  uint8_t* wpos = reinterpret_cast<uint8_t*>(aBuffer);

  memset(wpos, 0, silenceOffset);
  wpos += silenceOffset;

  void* input[2];
  uint32_t input_size[2];
  mBuffer.PopElements(available, &input[0], &input_size[0], &input[1], &input_size[1]);
  memcpy(wpos, input[0], input_size[0]);
  wpos += input_size[0];
  memcpy(wpos, input[1], input_size[1]);

  GetBufferInsertTime(aTimeMs);

  return aFrames;
}
示例#4
0
long
AudioStream::DataCallback(void* aBuffer, long aFrames)
{
  MonitorAutoLock mon(mMonitor);
  MOZ_ASSERT(mState != SHUTDOWN, "No data callback after shutdown");
  uint32_t available = std::min(static_cast<uint32_t>(FramesToBytes(aFrames)), mBuffer.Length());
  NS_ABORT_IF_FALSE(available % mBytesPerFrame == 0, "Must copy complete frames");
  AudioDataValue* output = reinterpret_cast<AudioDataValue*>(aBuffer);
  uint32_t underrunFrames = 0;
  uint32_t servicedFrames = 0;
  int64_t insertTime;

  // NOTE: wasapi (others?) can call us back *after* stop()/Shutdown() (mState == SHUTDOWN)
  // Bug 996162

  // callback tells us cubeb succeeded initializing
  if (mState == STARTED) {
    // For low-latency streams, we want to minimize any built-up data when
    // we start getting callbacks.
    // Simple version - contract on first callback only.
    if (mLatencyRequest == LowLatency) {
#ifdef PR_LOGGING
      uint32_t old_len = mBuffer.Length();
#endif
      available = mBuffer.ContractTo(FramesToBytes(aFrames));
#ifdef PR_LOGGING
      TimeStamp now = TimeStamp::Now();
      if (!mStartTime.IsNull()) {
        int64_t timeMs = (now - mStartTime).ToMilliseconds();
        PR_LOG(gAudioStreamLog, PR_LOG_WARNING,
               ("Stream took %lldms to start after first Write() @ %u", timeMs, mOutRate));
      } else {
        PR_LOG(gAudioStreamLog, PR_LOG_WARNING,
          ("Stream started before Write() @ %u", mOutRate));
      }

      if (old_len != available) {
        // Note that we may have dropped samples in Write() as well!
        PR_LOG(gAudioStreamLog, PR_LOG_WARNING,
               ("AudioStream %p dropped %u + %u initial frames @ %u", this,
                 mReadPoint, BytesToFrames(old_len - available), mOutRate));
        mReadPoint += BytesToFrames(old_len - available);
      }
#endif
    }
    mState = RUNNING;
  }

  if (available) {
    // When we are playing a low latency stream, and it is the first time we are
    // getting data from the buffer, we prefer to add the silence for an
    // underrun at the beginning of the buffer, so the first buffer is not cut
    // in half by the silence inserted to compensate for the underrun.
    if (mInRate == mOutRate) {
      if (mLatencyRequest == LowLatency && !mWritten) {
        servicedFrames = GetUnprocessedWithSilencePadding(output, aFrames, insertTime);
      } else {
        servicedFrames = GetUnprocessed(output, aFrames, insertTime);
      }
    } else {
      servicedFrames = GetTimeStretched(output, aFrames, insertTime);
    }
    float scaled_volume = float(GetVolumeScale() * mVolume);

    ScaleAudioSamples(output, aFrames * mOutChannels, scaled_volume);

    NS_ABORT_IF_FALSE(mBuffer.Length() % mBytesPerFrame == 0, "Must copy complete frames");

    // Notify any blocked Write() call that more space is available in mBuffer.
    mon.NotifyAll();
  } else {
    GetBufferInsertTime(insertTime);
  }

  underrunFrames = aFrames - servicedFrames;

  // Always send audible frames first, and silent frames later.
  // Otherwise it will break the assumption of FrameHistory.
  if (mState != DRAINING) {
    mAudioClock.UpdateFrameHistory(servicedFrames, underrunFrames);
    uint8_t* rpos = static_cast<uint8_t*>(aBuffer) + FramesToBytes(aFrames - underrunFrames);
    memset(rpos, 0, FramesToBytes(underrunFrames));
    if (underrunFrames) {
      PR_LOG(gAudioStreamLog, PR_LOG_WARNING,
             ("AudioStream %p lost %d frames", this, underrunFrames));
    }
    servicedFrames += underrunFrames;
  } else {
    mAudioClock.UpdateFrameHistory(servicedFrames, 0);
  }

  WriteDumpFile(mDumpFile, this, aFrames, aBuffer);
  // Don't log if we're not interested or if the stream is inactive
  if (PR_LOG_TEST(GetLatencyLog(), PR_LOG_DEBUG) &&
      mState != SHUTDOWN &&
      insertTime != INT64_MAX && servicedFrames > underrunFrames) {
    uint32_t latency = UINT32_MAX;
    if (cubeb_stream_get_latency(mCubebStream, &latency)) {
      NS_WARNING("Could not get latency from cubeb.");
    }
    TimeStamp now = TimeStamp::Now();

    mLatencyLog->Log(AsyncLatencyLogger::AudioStream, reinterpret_cast<uint64_t>(this),
                     insertTime, now);
    mLatencyLog->Log(AsyncLatencyLogger::Cubeb, reinterpret_cast<uint64_t>(mCubebStream.get()),
                     (latency * 1000) / mOutRate, now);
  }

  return servicedFrames;
}
示例#5
0
long
AudioStream::DataCallback(void* aBuffer, long aFrames)
{
  MonitorAutoLock mon(mMonitor);
  uint32_t available = std::min(static_cast<uint32_t>(FramesToBytes(aFrames)), mBuffer.Length());
  NS_ABORT_IF_FALSE(available % mBytesPerFrame == 0, "Must copy complete frames");
  AudioDataValue* output = reinterpret_cast<AudioDataValue*>(aBuffer);
  uint32_t underrunFrames = 0;
  uint32_t servicedFrames = 0;
  int64_t insertTime;

  if (available) {
    // When we are playing a low latency stream, and it is the first time we are
    // getting data from the buffer, we prefer to add the silence for an
    // underrun at the beginning of the buffer, so the first buffer is not cut
    // in half by the silence inserted to compensate for the underrun.
    if (mInRate == mOutRate) {
      if (mLatencyRequest == LowLatency && !mWritten) {
        servicedFrames = GetUnprocessedWithSilencePadding(output, aFrames, insertTime);
      } else {
        servicedFrames = GetUnprocessed(output, aFrames, insertTime);
      }
    } else {
      servicedFrames = GetTimeStretched(output, aFrames, insertTime);
    }
    float scaled_volume = float(GetVolumeScale() * mVolume);

    ScaleAudioSamples(output, aFrames * mOutChannels, scaled_volume);

    NS_ABORT_IF_FALSE(mBuffer.Length() % mBytesPerFrame == 0, "Must copy complete frames");

    // Notify any blocked Write() call that more space is available in mBuffer.
    mon.NotifyAll();
  } else {
    GetBufferInsertTime(insertTime);
  }

  underrunFrames = aFrames - servicedFrames;

  if (mState != DRAINING) {
    uint8_t* rpos = static_cast<uint8_t*>(aBuffer) + FramesToBytes(aFrames - underrunFrames);
    memset(rpos, 0, FramesToBytes(underrunFrames));
    if (underrunFrames) {
      PR_LOG(gAudioStreamLog, PR_LOG_WARNING,
             ("AudioStream %p lost %d frames", this, underrunFrames));
    }
    mLostFrames += underrunFrames;
    servicedFrames += underrunFrames;
  }

  WriteDumpFile(mDumpFile, this, aFrames, aBuffer);
  // Don't log if we're not interested or if the stream is inactive
  if (PR_LOG_TEST(GetLatencyLog(), PR_LOG_DEBUG) &&
      insertTime != INT64_MAX && servicedFrames > underrunFrames) {
    uint32_t latency = UINT32_MAX;
    if (cubeb_stream_get_latency(mCubebStream, &latency)) {
      NS_WARNING("Could not get latency from cubeb.");
    }
    TimeStamp now = TimeStamp::Now();

    mLatencyLog->Log(AsyncLatencyLogger::AudioStream, reinterpret_cast<uint64_t>(this),
                     insertTime, now);
    mLatencyLog->Log(AsyncLatencyLogger::Cubeb, reinterpret_cast<uint64_t>(mCubebStream.get()),
                     (latency * 1000) / mOutRate, now);
  }

  mAudioClock.UpdateWritePosition(servicedFrames);
  return servicedFrames;
}