gecko/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 "nsAudioStream.h"

namespace mozilla {

template <class SrcT, class DestT>
static void
InterleaveAndConvertBuffer(const SrcT* aSource, int32_t aSourceLength,
                           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(aSource[channel*aSourceLength + i])*aVolume;
      *output = FloatToAudioSample<DestT>(v);
      ++output;
    }
  }
}

static void
InterleaveAndConvertBuffer(const int16_t* aSource, int32_t aSourceLength,
                           int32_t aLength,
                           float aVolume,
                           int32_t aChannels,
                           int16_t* aOutput)
{
  int16_t* output = aOutput;
  float v = NS_MAX(NS_MIN(aVolume, 1.0f), -1.0f);
  int32_t volume = int32_t((1 << 16) * v);
  for (int32_t i = 0; i < aLength; ++i) {
    for (int32_t channel = 0; channel < aChannels; ++channel) {
      int16_t s = aSource[channel*aSourceLength + i];
      *output = int16_t((int32_t(s) * volume) >> 16);
      ++output;
    }
  }
}

template <class SrcT>
static void
InterleaveAndConvertBuffer(const SrcT* aSource, int32_t aSourceLength,
                           int32_t aLength,
                           float aVolume,
                           int32_t aChannels,
                           void* aOutput, AudioSampleFormat aOutputFormat)
{
  switch (aOutputFormat) {
  case AUDIO_FORMAT_FLOAT32:
    InterleaveAndConvertBuffer(aSource, aSourceLength, aLength, aVolume,
                               aChannels, static_cast<float*>(aOutput));
    break;
  case AUDIO_FORMAT_S16:
    InterleaveAndConvertBuffer(aSource, aSourceLength, aLength, aVolume,
                               aChannels, static_cast<int16_t*>(aOutput));
    break;
  }
}

static void
InterleaveAndConvertBuffer(const void* aSource, AudioSampleFormat aSourceFormat,
                           int32_t aSourceLength,
                           int32_t aOffset, int32_t aLength,
                           float aVolume,
                           int32_t aChannels,
                           void* aOutput, AudioSampleFormat aOutputFormat)
{
  switch (aSourceFormat) {
  case AUDIO_FORMAT_FLOAT32:
    InterleaveAndConvertBuffer(static_cast<const float*>(aSource) + aOffset, aSourceLength,
                               aLength,
                               aVolume,
                               aChannels,
                               aOutput, aOutputFormat);
    break;
  case AUDIO_FORMAT_S16:
    InterleaveAndConvertBuffer(static_cast<const int16_t*>(aSource) + aOffset, aSourceLength,
                               aLength,
                               aVolume,
                               aChannels,
                               aOutput, aOutputFormat);
    break;
  }
}

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

static const int STATIC_AUDIO_BUFFER_BYTES = 50000;

void
AudioSegment::WriteTo(nsAudioStream* aOutput)
{
  NS_ASSERTION(mChannels == aOutput->GetChannels(), "Wrong number of channels");
  nsAutoTArray<uint8_t,STATIC_AUDIO_BUFFER_BYTES> buf;
  uint32_t frameSize = GetSampleSize(AUDIO_OUTPUT_FORMAT)*mChannels;
  for (ChunkIterator ci(*this); !ci.IsEnded(); ci.Next()) {
    AudioChunk& c = *ci;
    if (frameSize*c.mDuration > UINT32_MAX) {
      NS_ERROR("Buffer overflow");
      return;
    }
    buf.SetLength(int32_t(frameSize*c.mDuration));
    if (c.mBuffer) {
      InterleaveAndConvertBuffer(c.mBuffer->Data(), c.mBufferFormat, c.mBufferLength,
                                 c.mOffset, int32_t(c.mDuration),
                                 c.mVolume,
                                 aOutput->GetChannels(),
                                 buf.Elements(), AUDIO_OUTPUT_FORMAT);
    } else {
      // Assumes that a bit pattern of zeroes == 0.0f
      memset(buf.Elements(), 0, buf.Length());
    }
    aOutput->Write(buf.Elements(), int32_t(c.mDuration));
  }
}

}