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path: root/dom/media/AudioConverter.cpp
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/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
/* vim: set ts=8 sts=2 et sw=2 tw=80: */
/* 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 "AudioConverter.h"
#include <string.h>
#include <speex/speex_resampler.h>
#include <cmath>

/*
 *  Parts derived from MythTV AudioConvert Class
 *  Created by Jean-Yves Avenard.
 *
 *  Copyright (C) Bubblestuff Pty Ltd 2013
 *  Copyright (C) foobum@gmail.com 2010
 */

namespace mozilla {

AudioConverter::AudioConverter(const AudioConfig& aIn, const AudioConfig& aOut)
    : mIn(aIn), mOut(aOut), mResampler(nullptr) {
  MOZ_DIAGNOSTIC_ASSERT(
      aIn.Format() == aOut.Format() && aIn.Interleaved() == aOut.Interleaved(),
      "No format or rate conversion is supported at this stage");
  MOZ_DIAGNOSTIC_ASSERT(
      aOut.Channels() <= 2 || aIn.Channels() == aOut.Channels(),
      "Only down/upmixing to mono or stereo is supported at this stage");
  MOZ_DIAGNOSTIC_ASSERT(aOut.Interleaved(),
                        "planar audio format not supported");
  mIn.Layout().MappingTable(mOut.Layout(), mChannelOrderMap);
  if (aIn.Rate() != aOut.Rate()) {
    RecreateResampler();
  }
}

AudioConverter::~AudioConverter() {
  if (mResampler) {
    speex_resampler_destroy(mResampler);
    mResampler = nullptr;
  }
}

bool AudioConverter::CanWorkInPlace() const {
  bool needDownmix = mIn.Channels() > mOut.Channels();
  bool needUpmix = mIn.Channels() < mOut.Channels();
  bool canDownmixInPlace =
      mIn.Channels() * AudioConfig::SampleSize(mIn.Format()) >=
      mOut.Channels() * AudioConfig::SampleSize(mOut.Format());
  bool needResample = mIn.Rate() != mOut.Rate();
  bool canResampleInPlace = mIn.Rate() >= mOut.Rate();
  // We should be able to work in place if 1s of audio input takes less space
  // than 1s of audio output. However, as we downmix before resampling we can't
  // perform any upsampling in place (e.g. if incoming rate >= outgoing rate)
  return !needUpmix && (!needDownmix || canDownmixInPlace) &&
         (!needResample || canResampleInPlace);
}

size_t AudioConverter::ProcessInternal(void* aOut, const void* aIn,
                                       size_t aFrames) {
  if (!aFrames) {
    return 0;
  }
  if (mIn.Channels() > mOut.Channels()) {
    return DownmixAudio(aOut, aIn, aFrames);
  } else if (mIn.Channels() < mOut.Channels()) {
    return UpmixAudio(aOut, aIn, aFrames);
  } else if (mIn.Layout() != mOut.Layout() && CanReorderAudio()) {
    ReOrderInterleavedChannels(aOut, aIn, aFrames);
  } else if (aIn != aOut) {
    memmove(aOut, aIn, FramesOutToBytes(aFrames));
  }
  return aFrames;
}

// Reorder interleaved channels.
// Can work in place (e.g aOut == aIn).
template <class AudioDataType>
void _ReOrderInterleavedChannels(AudioDataType* aOut, const AudioDataType* aIn,
                                 uint32_t aFrames, uint32_t aChannels,
                                 const uint8_t* aChannelOrderMap) {
  MOZ_DIAGNOSTIC_ASSERT(aChannels <= MAX_AUDIO_CHANNELS);
  AudioDataType val[MAX_AUDIO_CHANNELS];
  for (uint32_t i = 0; i < aFrames; i++) {
    for (uint32_t j = 0; j < aChannels; j++) {
      val[j] = aIn[aChannelOrderMap[j]];
    }
    for (uint32_t j = 0; j < aChannels; j++) {
      aOut[j] = val[j];
    }
    aOut += aChannels;
    aIn += aChannels;
  }
}

void AudioConverter::ReOrderInterleavedChannels(void* aOut, const void* aIn,
                                                size_t aFrames) const {
  MOZ_DIAGNOSTIC_ASSERT(mIn.Channels() == mOut.Channels());

  if (mOut.Channels() == 1 || mOut.Layout() == mIn.Layout()) {
    // If channel count is 1, planar and non-planar formats are the same and
    // there's nothing to reorder.
    if (aOut != aIn) {
      memmove(aOut, aIn, FramesOutToBytes(aFrames));
    }
    return;
  }

  uint32_t bits = AudioConfig::FormatToBits(mOut.Format());
  switch (bits) {
    case 8:
      _ReOrderInterleavedChannels((uint8_t*)aOut, (const uint8_t*)aIn, aFrames,
                                  mIn.Channels(), mChannelOrderMap);
      break;
    case 16:
      _ReOrderInterleavedChannels((int16_t*)aOut, (const int16_t*)aIn, aFrames,
                                  mIn.Channels(), mChannelOrderMap);
      break;
    default:
      MOZ_DIAGNOSTIC_ASSERT(AudioConfig::SampleSize(mOut.Format()) == 4);
      _ReOrderInterleavedChannels((int32_t*)aOut, (const int32_t*)aIn, aFrames,
                                  mIn.Channels(), mChannelOrderMap);
      break;
  }
}

static inline int16_t clipTo15(int32_t aX) {
  return aX < -32768 ? -32768 : aX <= 32767 ? aX : 32767;
}

size_t AudioConverter::DownmixAudio(void* aOut, const void* aIn,
                                    size_t aFrames) const {
  MOZ_ASSERT(mIn.Format() == AudioConfig::FORMAT_S16 ||
             mIn.Format() == AudioConfig::FORMAT_FLT);
  MOZ_ASSERT(mIn.Channels() >= mOut.Channels());
  MOZ_ASSERT(mIn.Layout() == AudioConfig::ChannelLayout(mIn.Channels()),
             "Can only downmix input data in SMPTE layout");
  MOZ_ASSERT(mOut.Layout() == AudioConfig::ChannelLayout(2) ||
             mOut.Layout() == AudioConfig::ChannelLayout(1));

  uint32_t channels = mIn.Channels();

  if (channels == 1 && mOut.Channels() == 1) {
    if (aOut != aIn) {
      memmove(aOut, aIn, FramesOutToBytes(aFrames));
    }
    return aFrames;
  }

  if (channels > 2) {
    if (mIn.Format() == AudioConfig::FORMAT_FLT) {
      // Downmix matrix. Per-row normalization 1 for rows 3,4 and 2 for rows
      // 5-8.
      static const float dmatrix[6][8][2] = {
          /*3*/ {{0.5858f, 0}, {0, 0.5858f}, {0.4142f, 0.4142f}},
          /*4*/
          {{0.4226f, 0}, {0, 0.4226f}, {0.366f, 0.2114f}, {0.2114f, 0.366f}},
          /*5*/
          {{0.6510f, 0},
           {0, 0.6510f},
           {0.4600f, 0.4600f},
           {0.5636f, 0.3254f},
           {0.3254f, 0.5636f}},
          /*6*/
          {{0.5290f, 0},
           {0, 0.5290f},
           {0.3741f, 0.3741f},
           {0.3741f, 0.3741f},
           {0.4582f, 0.2645f},
           {0.2645f, 0.4582f}},
          /*7*/
          {{0.4553f, 0},
           {0, 0.4553f},
           {0.3220f, 0.3220f},
           {0.3220f, 0.3220f},
           {0.2788f, 0.2788f},
           {0.3943f, 0.2277f},
           {0.2277f, 0.3943f}},
          /*8*/
          {{0.3886f, 0},
           {0, 0.3886f},
           {0.2748f, 0.2748f},
           {0.2748f, 0.2748f},
           {0.3366f, 0.1943f},
           {0.1943f, 0.3366f},
           {0.3366f, 0.1943f},
           {0.1943f, 0.3366f}},
      };
      // Re-write the buffer with downmixed data
      const float* in = static_cast<const float*>(aIn);
      float* out = static_cast<float*>(aOut);
      for (uint32_t i = 0; i < aFrames; i++) {
        float sampL = 0.0;
        float sampR = 0.0;
        for (uint32_t j = 0; j < channels; j++) {
          sampL +=
              in[i * mIn.Channels() + j] * dmatrix[mIn.Channels() - 3][j][0];
          sampR +=
              in[i * mIn.Channels() + j] * dmatrix[mIn.Channels() - 3][j][1];
        }
        *out++ = sampL;
        *out++ = sampR;
      }
    } else if (mIn.Format() == AudioConfig::FORMAT_S16) {
      // Downmix matrix. Per-row normalization 1 for rows 3,4 and 2 for rows
      // 5-8. Coefficients in Q14.
      static const int16_t dmatrix[6][8][2] = {
          /*3*/ {{9598, 0}, {0, 9598}, {6786, 6786}},
          /*4*/ {{6925, 0}, {0, 6925}, {5997, 3462}, {3462, 5997}},
          /*5*/
          {{10663, 0}, {0, 10663}, {7540, 7540}, {9234, 5331}, {5331, 9234}},
          /*6*/
          {{8668, 0},
           {0, 8668},
           {6129, 6129},
           {6129, 6129},
           {7507, 4335},
           {4335, 7507}},
          /*7*/
          {{7459, 0},
           {0, 7459},
           {5275, 5275},
           {5275, 5275},
           {4568, 4568},
           {6460, 3731},
           {3731, 6460}},
          /*8*/
          {{6368, 0},
           {0, 6368},
           {4502, 4502},
           {4502, 4502},
           {5514, 3184},
           {3184, 5514},
           {5514, 3184},
           {3184, 5514}}};
      // Re-write the buffer with downmixed data
      const int16_t* in = static_cast<const int16_t*>(aIn);
      int16_t* out = static_cast<int16_t*>(aOut);
      for (uint32_t i = 0; i < aFrames; i++) {
        int32_t sampL = 0;
        int32_t sampR = 0;
        for (uint32_t j = 0; j < channels; j++) {
          sampL += in[i * channels + j] * dmatrix[channels - 3][j][0];
          sampR += in[i * channels + j] * dmatrix[channels - 3][j][1];
        }
        *out++ = clipTo15((sampL + 8192) >> 14);
        *out++ = clipTo15((sampR + 8192) >> 14);
      }
    } else {
      MOZ_DIAGNOSTIC_ASSERT(false, "Unsupported data type");
    }

    // If we are to continue downmixing to mono, start working on the output
    // buffer.
    aIn = aOut;
    channels = 2;
  }

  if (mOut.Channels() == 1) {
    if (mIn.Format() == AudioConfig::FORMAT_FLT) {
      const float* in = static_cast<const float*>(aIn);
      float* out = static_cast<float*>(aOut);
      for (size_t fIdx = 0; fIdx < aFrames; ++fIdx) {
        float sample = 0.0;
        // The sample of the buffer would be interleaved.
        sample = (in[fIdx * channels] + in[fIdx * channels + 1]) * 0.5;
        *out++ = sample;
      }
    } else if (mIn.Format() == AudioConfig::FORMAT_S16) {
      const int16_t* in = static_cast<const int16_t*>(aIn);
      int16_t* out = static_cast<int16_t*>(aOut);
      for (size_t fIdx = 0; fIdx < aFrames; ++fIdx) {
        int32_t sample = 0.0;
        // The sample of the buffer would be interleaved.
        sample = (in[fIdx * channels] + in[fIdx * channels + 1]) * 0.5;
        *out++ = sample;
      }
    } else {
      MOZ_DIAGNOSTIC_ASSERT(false, "Unsupported data type");
    }
  }
  return aFrames;
}

size_t AudioConverter::ResampleAudio(void* aOut, const void* aIn,
                                     size_t aFrames) {
  if (!mResampler) {
    return 0;
  }
  uint32_t outframes = ResampleRecipientFrames(aFrames);
  uint32_t inframes = aFrames;

  int error;
  if (mOut.Format() == AudioConfig::FORMAT_FLT) {
    const float* in = reinterpret_cast<const float*>(aIn);
    float* out = reinterpret_cast<float*>(aOut);
    error = speex_resampler_process_interleaved_float(mResampler, in, &inframes,
                                                      out, &outframes);
  } else if (mOut.Format() == AudioConfig::FORMAT_S16) {
    const int16_t* in = reinterpret_cast<const int16_t*>(aIn);
    int16_t* out = reinterpret_cast<int16_t*>(aOut);
    error = speex_resampler_process_interleaved_int(mResampler, in, &inframes,
                                                    out, &outframes);
  } else {
    MOZ_DIAGNOSTIC_ASSERT(false, "Unsupported data type");
    error = RESAMPLER_ERR_ALLOC_FAILED;
  }
  MOZ_ASSERT(error == RESAMPLER_ERR_SUCCESS);
  if (error != RESAMPLER_ERR_SUCCESS) {
    speex_resampler_destroy(mResampler);
    mResampler = nullptr;
    return 0;
  }
  MOZ_ASSERT(inframes == aFrames, "Some frames will be dropped");
  return outframes;
}

void AudioConverter::RecreateResampler() {
  if (mResampler) {
    speex_resampler_destroy(mResampler);
  }
  int error;
  mResampler = speex_resampler_init(mOut.Channels(), mIn.Rate(), mOut.Rate(),
                                    SPEEX_RESAMPLER_QUALITY_DEFAULT, &error);

  if (error == RESAMPLER_ERR_SUCCESS) {
    speex_resampler_skip_zeros(mResampler);
  } else {
    NS_WARNING("Failed to initialize resampler.");
    mResampler = nullptr;
  }
}

size_t AudioConverter::DrainResampler(void* aOut) {
  if (!mResampler) {
    return 0;
  }
  int frames = speex_resampler_get_input_latency(mResampler);
  AlignedByteBuffer buffer(FramesOutToBytes(frames));
  if (!buffer) {
    // OOM
    return 0;
  }
  frames = ResampleAudio(aOut, buffer.Data(), frames);
  // Tore down the resampler as it's easier than handling follow-up.
  RecreateResampler();
  return frames;
}

size_t AudioConverter::UpmixAudio(void* aOut, const void* aIn,
                                  size_t aFrames) const {
  MOZ_ASSERT(mIn.Format() == AudioConfig::FORMAT_S16 ||
             mIn.Format() == AudioConfig::FORMAT_FLT);
  MOZ_ASSERT(mIn.Channels() < mOut.Channels());
  MOZ_ASSERT(mIn.Channels() == 1, "Can only upmix mono for now");
  MOZ_ASSERT(mOut.Channels() == 2, "Can only upmix to stereo for now");

  if (mOut.Channels() != 2) {
    return 0;
  }

  // Upmix mono to stereo.
  // This is a very dumb mono to stereo upmixing, power levels are preserved
  // following the calculation: left = right = -3dB*mono.
  if (mIn.Format() == AudioConfig::FORMAT_FLT) {
    const float m3db = std::sqrt(0.5);  // -3dB = sqrt(1/2)
    const float* in = static_cast<const float*>(aIn);
    float* out = static_cast<float*>(aOut);
    for (size_t fIdx = 0; fIdx < aFrames; ++fIdx) {
      float sample = in[fIdx] * m3db;
      // The samples of the buffer would be interleaved.
      *out++ = sample;
      *out++ = sample;
    }
  } else if (mIn.Format() == AudioConfig::FORMAT_S16) {
    const int16_t* in = static_cast<const int16_t*>(aIn);
    int16_t* out = static_cast<int16_t*>(aOut);
    for (size_t fIdx = 0; fIdx < aFrames; ++fIdx) {
      int16_t sample =
          ((int32_t)in[fIdx] * 11585) >> 14;  // close enough to i*sqrt(0.5)
      // The samples of the buffer would be interleaved.
      *out++ = sample;
      *out++ = sample;
    }
  } else {
    MOZ_DIAGNOSTIC_ASSERT(false, "Unsupported data type");
  }

  return aFrames;
}

size_t AudioConverter::ResampleRecipientFrames(size_t aFrames) const {
  if (!aFrames && mIn.Rate() != mOut.Rate()) {
    if (!mResampler) {
      return 0;
    }
    // We drain by pushing in get_input_latency() samples of 0
    aFrames = speex_resampler_get_input_latency(mResampler);
  }
  return (uint64_t)aFrames * mOut.Rate() / mIn.Rate() + 1;
}

size_t AudioConverter::FramesOutToSamples(size_t aFrames) const {
  return aFrames * mOut.Channels();
}

size_t AudioConverter::SamplesInToFrames(size_t aSamples) const {
  return aSamples / mIn.Channels();
}

size_t AudioConverter::FramesOutToBytes(size_t aFrames) const {
  return FramesOutToSamples(aFrames) * AudioConfig::SampleSize(mOut.Format());
}
}  // namespace mozilla