tor-browser

AudioSampleFormat.h (10387B)

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/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
/* vim: set ts=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 https://mozilla.org/MPL/2.0/. */
#ifndef MOZILLA_AUDIOSAMPLEFORMAT_H_
#define MOZILLA_AUDIOSAMPLEFORMAT_H_

#include <algorithm>
#include <limits>
#include <type_traits>

#include "mozilla/Assertions.h"
#include "mozilla/PodOperations.h"

namespace mozilla {

/**
* Audio formats supported in MediaTracks 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.
 AUDIO_OUTPUT_FORMAT = AUDIO_FORMAT_FLOAT32
};

enum { MAX_AUDIO_SAMPLE_SIZE = sizeof(float) };

template <AudioSampleFormat Format>
class AudioSampleTraits;

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

using AudioDataValue = AudioSampleTraits<AUDIO_OUTPUT_FORMAT>::Type;

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;
};

template <typename T>
constexpr float MaxAsFloat() {
 return static_cast<float>(std::numeric_limits<T>::max());
}

template <typename T>
constexpr float LowestAsFloat() {
 return static_cast<float>(std::numeric_limits<T>::lowest());
}

// The maximum value for an audio sample. If T is signed, the absolute value of
// this number is smaller (by exactly 1) than ::Min().
template <typename T>
constexpr T Max() {
 return std::numeric_limits<T>::max();
}

// The minimum value for an audio sample. If T is signed, the absolute value of
// this number is greater (by exactly 1) than ::Max()
template <typename T>
constexpr T Min() {
 return std::numeric_limits<T>::lowest();
}

template <>
constexpr float Max<float>() {
 return 1.0f;
}

template <>
constexpr float Min<float>() {
 return -1.0f;
}

// The bias value is the middle of the range. In linear PCM audio, if the
// values are all equal to the bias value, the audio is silent.
template <typename T>
constexpr T Bias() {
 return 0;
}

template <>
constexpr uint8_t Bias<uint8_t>() {
 return 128;
}

// Clip a floating point audio sample to its nominal range. This is
// destructive, and is only used here for avoiding overflow in some edge cases,
// so it's not going to be generally audible.
inline float Clip(float aValue) { return std::clamp(aValue, -1.0f, 1.0f); }

template <typename T>
T FloatToAudioSample(float aValue) {
 if constexpr (std::is_same_v<float, T>) {
   return aValue;
 }
 if constexpr (std::is_same_v<uint8_t, T>) {
   return static_cast<T>(std::clamp((aValue + 1.0f) * 128.f,
                                    LowestAsFloat<T>(), MaxAsFloat<T>()));
 } else if constexpr (std::is_same_v<int16_t, T>) {
   // This produces correct results accross the range.
   return static_cast<T>(std::clamp(aValue * -LowestAsFloat<T>(),
                                    LowestAsFloat<T>(), MaxAsFloat<T>()));
 } else if constexpr (std::is_same_v<int32_t, T>) {
   // We need to handle this differently because of rounding between INT32_MAX
   // and float 32-bits, to maximise precision.
   if (aValue >= 0.) {
     // if the input sample is greater OR EQUAL to 1.0, then clip and return
     // the max value.
     if (aValue >= 1.0) {
       return std::numeric_limits<T>::max();
     }
     // otherwise cast to a double and map to the positive range.
     // float 32-bits cannot represent int32_max (but can represent int32_min)
     constexpr double magnitudePos = std::numeric_limits<int32_t>::max();
     return static_cast<int32_t>(aValue * magnitudePos);
   }
   // Similarly for the negative range.
   if (aValue <= -1.0) {
     return std::numeric_limits<T>::lowest();
   }
   constexpr double magnitudeNegative =
       -1.0 * std::numeric_limits<int32_t>::lowest();
   return static_cast<int32_t>(aValue * magnitudeNegative);
 }
}

template <typename T>
T UInt8bitToAudioSample(uint8_t aValue) {
 if constexpr (std::is_same_v<uint8_t, T>) {
   return aValue;
 } else if constexpr (std::is_same_v<int16_t, T>) {
   return (static_cast<int16_t>(aValue) << 8) - (1 << 15);
 } else if constexpr (std::is_same_v<int32_t, T>) {
   return (static_cast<int32_t>(aValue) << 24) - (1 << 31);
 } else if constexpr (std::is_same_v<float, T>) {
   float biased = static_cast<float>(aValue) - Bias<uint8_t>();
   if (aValue >= Bias<uint8_t>()) {
     return Clip(biased / MaxAsFloat<int8_t>());
   }
   return Clip(biased / -LowestAsFloat<int8_t>());
 }
}

template <typename T>
T Int16ToAudioSample(int16_t aValue) {
 if constexpr (std::is_same_v<uint8_t, T>) {
   return static_cast<uint8_t>(aValue >> 8) + 128;
 } else if constexpr (std::is_same_v<int16_t, T>) {
   return aValue;
 } else if constexpr (std::is_same_v<int32_t, T>) {
   return aValue << 16;
 } else if constexpr (std::is_same_v<float, T>) {
   if (aValue >= 0) {
     return Clip(static_cast<float>(aValue) / MaxAsFloat<int16_t>());
   }
   return Clip(static_cast<float>(aValue) / -LowestAsFloat<int16_t>());
 }
}

// 24-bits audio samples are stored in 32-bits variables.
template <typename T>
T Int24ToAudioSample(int32_t aValue) {
 if constexpr (std::is_same_v<uint8_t, T>) {
   return static_cast<uint8_t>(aValue >> 16) + 128;
 } else if constexpr (std::is_same_v<int16_t, T>) {
   return static_cast<int16_t>(aValue >> 8);
 } else if constexpr (std::is_same_v<int32_t, T>) {
   return aValue << 8;
 } else if constexpr (std::is_same_v<float, T>) {
   const int32_t min = -(2 << 22);
   const int32_t max = (2 << 22) - 1;
   if (aValue >= 0) {
     return Clip(static_cast<float>(aValue) / static_cast<float>(max));
   }
   return Clip(static_cast<float>(aValue) / -static_cast<float>(min));
 }
}

template <typename T>
T Int32ToAudioSample(int32_t aValue) {
 if constexpr (std::is_same_v<uint8_t, T>) {
   return static_cast<uint8_t>(aValue >> 24) + 128;
 } else if constexpr (std::is_same_v<int16_t, T>) {
   return aValue >> 16;
 } else if constexpr (std::is_same_v<int32_t, T>) {
   return aValue;
 } else if constexpr (std::is_same_v<float, T>) {
   if (aValue >= 0) {
     return Clip(static_cast<float>(aValue) / MaxAsFloat<int32_t>());
   }
   return Clip(static_cast<float>(aValue) / -LowestAsFloat<int32_t>());
 }
}

// This does not handle 24-bits audio, call the function explicitly when
// needed.
template <typename D, typename S>
inline D ConvertAudioSample(const S& aSource) {
 if constexpr (std::is_same_v<S, D>) {
   return aSource;
 } else if constexpr (std::is_same_v<S, uint8_t>) {
   return UInt8bitToAudioSample<D>(aSource);
 } else if constexpr (std::is_same_v<S, int16_t>) {
   return Int16ToAudioSample<D>(aSource);
 } else if constexpr (std::is_same_v<S, int32_t>) {
   return Int32ToAudioSample<D>(aSource);
 } else if constexpr (std::is_same_v<S, float>) {
   return FloatToAudioSample<D>(aSource);
 }
}

// Sample buffer conversion
template <typename From, typename To>
inline void ConvertAudioSamples(const From* aFrom, To* aTo, int aCount) {
 if constexpr (std::is_same_v<From, To>) {
   PodCopy(aTo, aFrom, aCount);
   return;
 }
 for (int i = 0; i < aCount; ++i) {
   aTo[i] = ConvertAudioSample<To>(aFrom[i]);
 }
}

// 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] =
       ConvertAudioSample<To>(ConvertAudioSample<float>(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>(ConvertAudioSample<float>(aFrom[i]) *
                                        aScale);
 }
}

template <typename From, typename To>
inline void AddAudioSamplesWithScale(const From* aFrom, To* aTo, int aCount,
                                    float aScale) {
 for (int i = 0; i < aCount; ++i) {
   aTo[i] =
       ConvertAudioSample<To>(ConvertAudioSample<float>(aTo[i]) +
                              ConvertAudioSample<float>(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_ */