gecko-dev/content/media/AudioSegment.cpp

/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
/* 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 "AudioSegment.h"

#include "AudioStream.h"
#include "AudioChannelFormat.h"
#include "Latency.h"
#include "speex/speex_resampler.h"

namespace mozilla {

template <class SrcT, class DestT>
static void
InterleaveAndConvertBuffer(const SrcT** aSourceChannels,
                           int32_t aLength, float aVolume,
                           int32_t aChannels,
                           DestT* aOutput)
{
  DestT* output = aOutput;
  for (int32_t i = 0; i < aLength; ++i) {
    for (int32_t channel = 0; channel < aChannels; ++channel) {
      float v = AudioSampleToFloat(aSourceChannels[channel][i])*aVolume;
      *output = FloatToAudioSample<DestT>(v);
      ++output;
    }
  }
}

void
InterleaveAndConvertBuffer(const void** aSourceChannels,
                           AudioSampleFormat aSourceFormat,
                           int32_t aLength, float aVolume,
                           int32_t aChannels,
                           AudioDataValue* aOutput)
{
  switch (aSourceFormat) {
  case AUDIO_FORMAT_FLOAT32:
    InterleaveAndConvertBuffer(reinterpret_cast<const float**>(aSourceChannels),
                               aLength,
                               aVolume,
                               aChannels,
                               aOutput);
    break;
  case AUDIO_FORMAT_S16:
    InterleaveAndConvertBuffer(reinterpret_cast<const int16_t**>(aSourceChannels),
                               aLength,
                               aVolume,
                               aChannels,
                               aOutput);
    break;
  }
}

void
AudioSegment::ApplyVolume(float aVolume)
{
  for (ChunkIterator ci(*this); !ci.IsEnded(); ci.Next()) {
    ci->mVolume *= aVolume;
  }
}

static const int AUDIO_PROCESSING_FRAMES = 640; /* > 10ms of 48KHz audio */
static const uint8_t gZeroChannel[MAX_AUDIO_SAMPLE_SIZE*AUDIO_PROCESSING_FRAMES] = {0};

void
DownmixAndInterleave(const nsTArray<const void*>& aChannelData,
                     AudioSampleFormat aSourceFormat, int32_t aDuration,
                     float aVolume, uint32_t aOutputChannels,
                     AudioDataValue* aOutput)
{
  nsAutoTArray<const void*,GUESS_AUDIO_CHANNELS> channelData;
  nsAutoTArray<float,AUDIO_PROCESSING_FRAMES*GUESS_AUDIO_CHANNELS> downmixConversionBuffer;
  nsAutoTArray<float,AUDIO_PROCESSING_FRAMES*GUESS_AUDIO_CHANNELS> downmixOutputBuffer;

  channelData.SetLength(aChannelData.Length());
  if (aSourceFormat != AUDIO_FORMAT_FLOAT32) {
    NS_ASSERTION(aSourceFormat == AUDIO_FORMAT_S16, "unknown format");
    downmixConversionBuffer.SetLength(aDuration*aChannelData.Length());
    for (uint32_t i = 0; i < aChannelData.Length(); ++i) {
      float* conversionBuf = downmixConversionBuffer.Elements() + (i*aDuration);
      const int16_t* sourceBuf = static_cast<const int16_t*>(aChannelData[i]);
      for (uint32_t j = 0; j < (uint32_t)aDuration; ++j) {
        conversionBuf[j] = AudioSampleToFloat(sourceBuf[j]);
      }
      channelData[i] = conversionBuf;
    }
  } else {
    for (uint32_t i = 0; i < aChannelData.Length(); ++i) {
      channelData[i] = aChannelData[i];
    }
  }

  downmixOutputBuffer.SetLength(aDuration*aOutputChannels);
  nsAutoTArray<float*,GUESS_AUDIO_CHANNELS> outputChannelBuffers;
  nsAutoTArray<const void*,GUESS_AUDIO_CHANNELS> outputChannelData;
  outputChannelBuffers.SetLength(aOutputChannels);
  outputChannelData.SetLength(aOutputChannels);
  for (uint32_t i = 0; i < (uint32_t)aOutputChannels; ++i) {
    outputChannelData[i] = outputChannelBuffers[i] =
        downmixOutputBuffer.Elements() + aDuration*i;
  }
  if (channelData.Length() > aOutputChannels) {
    AudioChannelsDownMix(channelData, outputChannelBuffers.Elements(),
                         aOutputChannels, aDuration);
  }
  InterleaveAndConvertBuffer(outputChannelData.Elements(), AUDIO_FORMAT_FLOAT32,
                             aDuration, aVolume, aOutputChannels, aOutput);
}

void AudioSegment::ResampleChunks(SpeexResamplerState* aResampler)
{
  uint32_t inRate, outRate;

  if (mChunks.IsEmpty()) {
    return;
  }

  speex_resampler_get_rate(aResampler, &inRate, &outRate);

  switch (mChunks[0].mBufferFormat) {
    case AUDIO_FORMAT_FLOAT32:
      Resample<float>(aResampler, inRate, outRate);
    break;
    case AUDIO_FORMAT_S16:
      Resample<int16_t>(aResampler, inRate, outRate);
    break;
    default:
      MOZ_ASSERT(false);
    break;
  }
}

void
AudioSegment::WriteTo(uint64_t aID, AudioStream* aOutput)
{
  uint32_t outputChannels = aOutput->GetChannels();
  nsAutoTArray<AudioDataValue,AUDIO_PROCESSING_FRAMES*GUESS_AUDIO_CHANNELS> buf;
  nsAutoTArray<const void*,GUESS_AUDIO_CHANNELS> channelData;

  for (ChunkIterator ci(*this); !ci.IsEnded(); ci.Next()) {
    AudioChunk& c = *ci;
    TrackTicks offset = 0;
    while (offset < c.mDuration) {
      TrackTicks durationTicks =
        std::min<TrackTicks>(c.mDuration - offset, AUDIO_PROCESSING_FRAMES);
      if (uint64_t(outputChannels)*durationTicks > INT32_MAX || offset > INT32_MAX) {
        NS_ERROR("Buffer overflow");
        return;
      }

      uint32_t duration = uint32_t(durationTicks);

      // If we have written data in the past, or we have real (non-silent) data
      // to write, we can proceed. Otherwise, it means we just started the
      // AudioStream, and we don't have real data to write to it (just silence).
      // To avoid overbuffering in the AudioStream, we simply drop the silence,
      // here. The stream will underrun and output silence anyways.
      if (c.mBuffer || aOutput->GetWritten()) {
        buf.SetLength(outputChannels*duration);
        if (c.mBuffer) {
          channelData.SetLength(c.mChannelData.Length());
          for (uint32_t i = 0; i < channelData.Length(); ++i) {
            channelData[i] =
              AddAudioSampleOffset(c.mChannelData[i], c.mBufferFormat, int32_t(offset));
          }

          if (channelData.Length() < outputChannels) {
            // Up-mix. Note that this might actually make channelData have more
            // than outputChannels temporarily.
            AudioChannelsUpMix(&channelData, outputChannels, gZeroChannel);
          }

          if (channelData.Length() > outputChannels) {
            // Down-mix.
            DownmixAndInterleave(channelData, c.mBufferFormat, duration,
                                 c.mVolume, outputChannels, buf.Elements());
          } else {
            InterleaveAndConvertBuffer(channelData.Elements(), c.mBufferFormat,
                                       duration, c.mVolume,
                                       outputChannels,
                                       buf.Elements());
          }
        } else {
          // Assumes that a bit pattern of zeroes == 0.0f
          memset(buf.Elements(), 0, buf.Length()*sizeof(AudioDataValue));
        }
        aOutput->Write(buf.Elements(), int32_t(duration), &(c.mTimeStamp));
      }
      if(!c.mTimeStamp.IsNull()) {
        TimeStamp now = TimeStamp::Now();
        // would be more efficient to c.mTimeStamp to ms on create time then pass here
        LogTime(AsyncLatencyLogger::AudioMediaStreamTrack, aID,
                (now - c.mTimeStamp).ToMilliseconds(), c.mTimeStamp);
      }
      offset += duration;
    }
  }
  aOutput->Start();
}

}