/* -*- 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_ */