gecko-dev/content/media/AudioSampleFormat.h

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4.5 KiB
C++

/* -*- 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 "nsAlgorithm.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
{
// 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;
// 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;
}
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);
}
// 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 == 0, "Bad constant");
static_assert(AUDIO_FORMAT_FLOAT32 == 1, "Bad constant");
NS_ASSERTION(aFormat == AUDIO_FORMAT_S16 || aFormat == AUDIO_FORMAT_FLOAT32,
"Unknown format");
return static_cast<const uint8_t*>(aBase) + (aFormat + 1)*2*aOffset;
}
} // namespace mozilla
#endif /* MOZILLA_AUDIOSAMPLEFORMAT_H_ */