DXR is a code search and navigation tool aimed at making sense of large projects. It supports full-text and regex searches as well as structural queries.

Mercurial (b6d82b1a6b02)

VCS Links

Line Code
1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20 21 22 23 24 25 26 27 28 29 30 31 32 33 34 35 36 37 38 39 40 41 42 43 44 45 46 47 48 49 50 51 52 53 54 55 56 57 58 59 60 61 62 63 64 65 66 67 68 69 70 71 72 73 74 75 76 77 78 79 80 81 82 83 84 85 86 87 88 89 90 91 92 93 94 95 96 97 98 99 100 101 102 103 104 105 106 107 108 109 110 111 112 113 114 115 116 117 118 119 120 121 122 123 124 125 126 127 128 129 130 131 132 133 134 135 136 137 138 139 140 141 142 143 144 145 146 147 148 149 150 151 152 153 154 155 156 157 158 159 160 161 162 163 164 165 166 167 168 169 170 171 172 173 174 175 176 177 178 179 180 181 182 183 184 185 186 187 188 189 190 191 192 193 194 195 196 197 198 199 200 201 202 203 204 205 206 207 208 209 210 211 212 213 214 215 216 217 218 219 220 221 222 223 224 225 226 227 228
/* -*- 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 static_cast<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 static_cast<int16_t>(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_ */