path: root/dom/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 "AudioMixer.h"
#include "AudioChannelFormat.h"
#include <speex/speex_resampler.h>

namespace mozilla {

const uint8_t
    SilentChannel::gZeroChannel[MAX_AUDIO_SAMPLE_SIZE *
                                SilentChannel::AUDIO_PROCESSING_FRAMES] = {0};

template <>
const float* SilentChannel::ZeroChannel<float>() {
  return reinterpret_cast<const float*>(SilentChannel::gZeroChannel);
}

template <>
const int16_t* SilentChannel::ZeroChannel<int16_t>() {
  return reinterpret_cast<const int16_t*>(SilentChannel::gZeroChannel);
}

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

void AudioSegment::ResampleChunks(nsAutoRef<SpeexResamplerState>& aResampler,
                                  uint32_t* aResamplerChannelCount,
                                  uint32_t aInRate, uint32_t aOutRate) {
  if (mChunks.IsEmpty()) {
    return;
  }

  AudioSampleFormat format = AUDIO_FORMAT_SILENCE;
  for (ChunkIterator ci(*this); !ci.IsEnded(); ci.Next()) {
    if (ci->mBufferFormat != AUDIO_FORMAT_SILENCE) {
      format = ci->mBufferFormat;
    }
  }

  switch (format) {
    // If the format is silence at this point, all the chunks are silent. The
    // actual function we use does not matter, it's just a matter of changing
    // the chunks duration.
    case AUDIO_FORMAT_SILENCE:
    case AUDIO_FORMAT_FLOAT32:
      Resample<float>(aResampler, aResamplerChannelCount, aInRate, aOutRate);
      break;
    case AUDIO_FORMAT_S16:
      Resample<int16_t>(aResampler, aResamplerChannelCount, aInRate, aOutRate);
      break;
    default:
      MOZ_ASSERT(false);
      break;
  }
}

size_t AudioSegment::WriteToInterleavedBuffer(nsTArray<AudioDataValue>& aBuffer,
                                              uint32_t aChannels) const {
  size_t offset = 0;
  if (GetDuration() <= 0) {
    MOZ_ASSERT(GetDuration() == 0);
    return offset;
  }

  // Calculate how many samples in this segment
  size_t frames = static_cast<size_t>(GetDuration());
  CheckedInt<size_t> samples(frames);
  samples *= static_cast<size_t>(aChannels);
  MOZ_ASSERT(samples.isValid());
  if (!samples.isValid()) {
    return offset;
  }

  // Enlarge buffer space if needed
  if (samples.value() > aBuffer.Capacity()) {
    aBuffer.SetCapacity(samples.value());
  }
  aBuffer.SetLengthAndRetainStorage(samples.value());
  aBuffer.ClearAndRetainStorage();

  // Convert the de-interleaved chunks into an interleaved buffer. Note that
  // we may upmix or downmix the audio data if the channel in the chunks
  // mismatch with aChannels
  for (ConstChunkIterator ci(*this); !ci.IsEnded(); ci.Next()) {
    const AudioChunk& c = *ci;
    size_t samplesInChunk = static_cast<size_t>(c.mDuration) * aChannels;
    switch (c.mBufferFormat) {
      case AUDIO_FORMAT_S16:
        WriteChunk<int16_t>(c, aChannels, c.mVolume,
                            aBuffer.Elements() + offset);
        break;
      case AUDIO_FORMAT_FLOAT32:
        WriteChunk<float>(c, aChannels, c.mVolume, aBuffer.Elements() + offset);
        break;
      case AUDIO_FORMAT_SILENCE:
        PodZero(aBuffer.Elements() + offset, samplesInChunk);
        break;
      default:
        MOZ_ASSERT_UNREACHABLE("Unknown format");
        PodZero(aBuffer.Elements() + offset, samplesInChunk);
        break;
    }
    offset += samplesInChunk;
  }
  MOZ_DIAGNOSTIC_ASSERT(samples.value() == offset,
                        "Segment's duration is incorrect");
  aBuffer.SetLengthAndRetainStorage(offset);
  return offset;
}

// This helps to to safely get a pointer to the position we want to start
// writing a planar audio buffer, depending on the channel and the offset in the
// buffer.
static AudioDataValue* PointerForOffsetInChannel(AudioDataValue* aData,
                                                 size_t aLengthSamples,
                                                 uint32_t aChannelCount,
                                                 uint32_t aChannel,
                                                 uint32_t aOffsetSamples) {
  size_t samplesPerChannel = aLengthSamples / aChannelCount;
  size_t beginningOfChannel = samplesPerChannel * aChannel;
  MOZ_ASSERT(aChannel * samplesPerChannel + aOffsetSamples < aLengthSamples,
             "Offset request out of bounds.");
  return aData + beginningOfChannel + aOffsetSamples;
}

void AudioSegment::Mix(AudioMixer& aMixer, uint32_t aOutputChannels,
                       uint32_t aSampleRate) {
  AutoTArray<AudioDataValue,
             SilentChannel::AUDIO_PROCESSING_FRAMES * GUESS_AUDIO_CHANNELS>
      buf;
  AutoTArray<const AudioDataValue*, GUESS_AUDIO_CHANNELS> channelData;
  uint32_t offsetSamples = 0;
  uint32_t duration = GetDuration();

  if (duration <= 0) {
    MOZ_ASSERT(duration == 0);
    return;
  }

  uint32_t outBufferLength = duration * aOutputChannels;
  buf.SetLength(outBufferLength);

  for (ChunkIterator ci(*this); !ci.IsEnded(); ci.Next()) {
    AudioChunk& c = *ci;
    uint32_t frames = c.mDuration;

    // If the chunk is silent, simply write the right number of silence in the
    // buffers.
    if (c.mBufferFormat == AUDIO_FORMAT_SILENCE) {
      for (uint32_t channel = 0; channel < aOutputChannels; channel++) {
        AudioDataValue* ptr =
            PointerForOffsetInChannel(buf.Elements(), outBufferLength,
                                      aOutputChannels, channel, offsetSamples);
        PodZero(ptr, frames);
      }
    } else {
      // Othewise, we need to upmix or downmix appropriately, depending on the
      // desired input and output channels.
      channelData.SetLength(c.mChannelData.Length());
      for (uint32_t i = 0; i < channelData.Length(); ++i) {
        channelData[i] = static_cast<const AudioDataValue*>(c.mChannelData[i]);
      }
      if (channelData.Length() < aOutputChannels) {
        // Up-mix.
        AudioChannelsUpMix(&channelData, aOutputChannels,
                           SilentChannel::ZeroChannel<AudioDataValue>());
        for (uint32_t channel = 0; channel < aOutputChannels; channel++) {
          AudioDataValue* ptr = PointerForOffsetInChannel(
              buf.Elements(), outBufferLength, aOutputChannels, channel,
              offsetSamples);
          PodCopy(ptr,
                  reinterpret_cast<const AudioDataValue*>(channelData[channel]),
                  frames);
        }
        MOZ_ASSERT(channelData.Length() == aOutputChannels);
      } else if (channelData.Length() > aOutputChannels) {
        // Down mix.
        AutoTArray<AudioDataValue*, GUESS_AUDIO_CHANNELS> outChannelPtrs;
        outChannelPtrs.SetLength(aOutputChannels);
        uint32_t offsetSamples = 0;
        for (uint32_t channel = 0; channel < aOutputChannels; channel++) {
          outChannelPtrs[channel] = PointerForOffsetInChannel(
              buf.Elements(), outBufferLength, aOutputChannels, channel,
              offsetSamples);
        }
        AudioChannelsDownMix(channelData, outChannelPtrs.Elements(),
                             aOutputChannels, frames);
      } else {
        // The channel count is already what we want, just copy it over.
        for (uint32_t channel = 0; channel < aOutputChannels; channel++) {
          AudioDataValue* ptr = PointerForOffsetInChannel(
              buf.Elements(), outBufferLength, aOutputChannels, channel,
              offsetSamples);
          PodCopy(ptr,
                  reinterpret_cast<const AudioDataValue*>(channelData[channel]),
                  frames);
        }
      }
    }
    offsetSamples += frames;
  }

  if (offsetSamples) {
    MOZ_ASSERT(offsetSamples == outBufferLength / aOutputChannels,
               "We forgot to write some samples?");
    aMixer.Mix(buf.Elements(), aOutputChannels, offsetSamples, aSampleRate);
  }
}

void AudioSegment::WriteTo(AudioMixer& aMixer, uint32_t aOutputChannels,
                           uint32_t aSampleRate) {
  AutoTArray<AudioDataValue,
             SilentChannel::AUDIO_PROCESSING_FRAMES * GUESS_AUDIO_CHANNELS>
      buf;
  // Offset in the buffer that will be written to the mixer, in samples.
  uint32_t offset = 0;

  if (GetDuration() <= 0) {
    MOZ_ASSERT(GetDuration() == 0);
    return;
  }

  uint32_t outBufferLength = GetDuration() * aOutputChannels;
  buf.SetLength(outBufferLength);

  for (ChunkIterator ci(*this); !ci.IsEnded(); ci.Next()) {
    AudioChunk& c = *ci;

    switch (c.mBufferFormat) {
      case AUDIO_FORMAT_S16:
        WriteChunk<int16_t>(c, aOutputChannels, c.mVolume,
                            buf.Elements() + offset);
        break;
      case AUDIO_FORMAT_FLOAT32:
        WriteChunk<float>(c, aOutputChannels, c.mVolume,
                          buf.Elements() + offset);
        break;
      case AUDIO_FORMAT_SILENCE:
        // The mixer is expecting interleaved data, so this is ok.
        PodZero(buf.Elements() + offset, c.mDuration * aOutputChannels);
        break;
      default:
        MOZ_ASSERT(false, "Not handled");
    }

    offset += c.mDuration * aOutputChannels;
  }

  if (offset) {
    aMixer.Mix(buf.Elements(), aOutputChannels, offset / aOutputChannels,
               aSampleRate);
  }
}

}  // namespace mozilla