gecko-dev/dom/media/AudioSampleFormat.h

/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
/* vim:set ts=2 sw=2 sts=2 et cindent: */
/* 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/. */
#ifndef MOZILLA_AUDIOSAMPLEFORMAT_H_
#define MOZILLA_AUDIOSAMPLEFORMAT_H_

#include "mozilla/Assertions.h"
#include <algorithm>

namespace mozilla {

/**
 * Audio formats supported in MediaStreams and media elements.
 *
 * Only one of these is supported by AudioStream, and that is determined
 * at compile time (roughly, FLOAT32 on desktops, S16 on mobile). Media decoders
 * produce that format only; queued AudioData always uses that format.
 */
enum AudioSampleFormat {
  // Silence: format will be chosen later
  AUDIO_FORMAT_SILENCE,
  // Native-endian signed 16-bit audio samples
  AUDIO_FORMAT_S16,
  // Signed 32-bit float samples
  AUDIO_FORMAT_FLOAT32,
// The format used for output by AudioStream.
#ifdef MOZ_SAMPLE_TYPE_S16
  AUDIO_OUTPUT_FORMAT = AUDIO_FORMAT_S16
#else
  AUDIO_OUTPUT_FORMAT = AUDIO_FORMAT_FLOAT32
#endif
};

enum { MAX_AUDIO_SAMPLE_SIZE = sizeof(float) };

template <AudioSampleFormat Format>
class AudioSampleTraits;

template <>
class AudioSampleTraits<AUDIO_FORMAT_FLOAT32> {
 public:
  typedef float Type;
};
template <>
class AudioSampleTraits<AUDIO_FORMAT_S16> {
 public:
  typedef int16_t Type;
};

typedef AudioSampleTraits<AUDIO_OUTPUT_FORMAT>::Type AudioDataValue;

template <typename T>
class AudioSampleTypeToFormat;

template <>
class AudioSampleTypeToFormat<float> {
 public:
  static const AudioSampleFormat Format = AUDIO_FORMAT_FLOAT32;
};

template <>
class AudioSampleTypeToFormat<short> {
 public:
  static const AudioSampleFormat Format = AUDIO_FORMAT_S16;
};

// Single-sample conversion
/*
 * Use "2^N" conversion since it's simple, fast, "bit transparent", used by
 * many other libraries and apparently behaves reasonably.
 * http://blog.bjornroche.com/2009/12/int-float-int-its-jungle-out-there.html
 * http://blog.bjornroche.com/2009/12/linearity-and-dynamic-range-in-int.html
 */
inline float AudioSampleToFloat(float aValue) { return aValue; }
inline float AudioSampleToFloat(int16_t aValue) { return aValue / 32768.0f; }
inline float AudioSampleToFloat(int32_t aValue) {
  return aValue / (float)(1U << 31);
}

template <typename T>
T FloatToAudioSample(float aValue);

template <>
inline float FloatToAudioSample<float>(float aValue) {
  return aValue;
}
template <>
inline int16_t FloatToAudioSample<int16_t>(float aValue) {
  float v = aValue * 32768.0f;
  float clamped = std::max(-32768.0f, std::min(32767.0f, v));
  return int16_t(clamped);
}

template <typename T>
T UInt8bitToAudioSample(uint8_t aValue);

template <>
inline float UInt8bitToAudioSample<float>(uint8_t aValue) {
  return aValue * (static_cast<float>(2) / UINT8_MAX) - static_cast<float>(1);
}
template <>
inline int16_t UInt8bitToAudioSample<int16_t>(uint8_t aValue) {
  return (int16_t(aValue) << 8) + aValue + INT16_MIN;
}

template <typename T>
T IntegerToAudioSample(int16_t aValue);

template <>
inline float IntegerToAudioSample<float>(int16_t aValue) {
  return aValue / 32768.0f;
}
template <>
inline int16_t IntegerToAudioSample<int16_t>(int16_t aValue) {
  return aValue;
}

template <typename T>
T Int24bitToAudioSample(int32_t aValue);

template <>
inline float Int24bitToAudioSample<float>(int32_t aValue) {
  return aValue / static_cast<float>(1 << 23);
}
template <>
inline int16_t Int24bitToAudioSample<int16_t>(int32_t aValue) {
  return aValue / 256;
}

template <typename SrcT, typename DstT>
inline void ConvertAudioSample(SrcT aIn, DstT& aOut);

template <>
inline void ConvertAudioSample(int16_t aIn, int16_t& aOut) {
  aOut = aIn;
}

template <>
inline void ConvertAudioSample(int16_t aIn, float& aOut) {
  aOut = AudioSampleToFloat(aIn);
}

template <>
inline void ConvertAudioSample(float aIn, float& aOut) {
  aOut = aIn;
}

template <>
inline void ConvertAudioSample(float aIn, int16_t& aOut) {
  aOut = FloatToAudioSample<int16_t>(aIn);
}

// Sample buffer conversion

template <typename From, typename To>
inline void ConvertAudioSamples(const From* aFrom, To* aTo, int aCount) {
  for (int i = 0; i < aCount; ++i) {
    aTo[i] = FloatToAudioSample<To>(AudioSampleToFloat(aFrom[i]));
  }
}
inline void ConvertAudioSamples(const int16_t* aFrom, int16_t* aTo,
                                int aCount) {
  memcpy(aTo, aFrom, sizeof(*aTo) * aCount);
}
inline void ConvertAudioSamples(const float* aFrom, float* aTo, int aCount) {
  memcpy(aTo, aFrom, sizeof(*aTo) * aCount);
}

// Sample buffer conversion with scale

template <typename From, typename To>
inline void ConvertAudioSamplesWithScale(const From* aFrom, To* aTo, int aCount,
                                         float aScale) {
  if (aScale == 1.0f) {
    ConvertAudioSamples(aFrom, aTo, aCount);
    return;
  }
  for (int i = 0; i < aCount; ++i) {
    aTo[i] = FloatToAudioSample<To>(AudioSampleToFloat(aFrom[i]) * aScale);
  }
}
inline void ConvertAudioSamplesWithScale(const int16_t* aFrom, int16_t* aTo,
                                         int aCount, float aScale) {
  if (aScale == 1.0f) {
    ConvertAudioSamples(aFrom, aTo, aCount);
    return;
  }
  if (0.0f <= aScale && aScale < 1.0f) {
    int32_t scale = int32_t((1 << 16) * aScale);
    for (int i = 0; i < aCount; ++i) {
      aTo[i] = int16_t((int32_t(aFrom[i]) * scale) >> 16);
    }
    return;
  }
  for (int i = 0; i < aCount; ++i) {
    aTo[i] = FloatToAudioSample<int16_t>(AudioSampleToFloat(aFrom[i]) * aScale);
  }
}

// In place audio sample scaling.
inline void ScaleAudioSamples(float* aBuffer, int aCount, float aScale) {
  for (int32_t i = 0; i < aCount; ++i) {
    aBuffer[i] *= aScale;
  }
}

inline void ScaleAudioSamples(short* aBuffer, int aCount, float aScale) {
  int32_t volume = int32_t((1 << 16) * aScale);
  for (int32_t i = 0; i < aCount; ++i) {
    aBuffer[i] = short((int32_t(aBuffer[i]) * volume) >> 16);
  }
}

inline const void* AddAudioSampleOffset(const void* aBase,
                                        AudioSampleFormat aFormat,
                                        int32_t aOffset) {
  static_assert(AUDIO_FORMAT_S16 == 1, "Bad constant");
  static_assert(AUDIO_FORMAT_FLOAT32 == 2, "Bad constant");
  MOZ_ASSERT(aFormat == AUDIO_FORMAT_S16 || aFormat == AUDIO_FORMAT_FLOAT32);

  return static_cast<const uint8_t*>(aBase) + aFormat * 2 * aOffset;
}

}  // namespace mozilla

#endif /* MOZILLA_AUDIOSAMPLEFORMAT_H_ */