gecko-dev/content/media/AudioSegment.cpp

193 строки
6.9 KiB
C++

/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
/* 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/. */
#include "AudioSegment.h"
#include "AudioStream.h"
#include "AudioChannelFormat.h"
namespace mozilla {
template <class SrcT, class DestT>
static void
InterleaveAndConvertBuffer(const SrcT** aSourceChannels,
int32_t aLength, float aVolume,
int32_t aChannels,
DestT* aOutput)
{
DestT* output = aOutput;
for (int32_t i = 0; i < aLength; ++i) {
for (int32_t channel = 0; channel < aChannels; ++channel) {
float v = AudioSampleToFloat(aSourceChannels[channel][i])*aVolume;
*output = FloatToAudioSample<DestT>(v);
++output;
}
}
}
static inline void
InterleaveAndConvertBuffer(const int16_t** aSourceChannels,
int32_t aLength, float aVolume,
int32_t aChannels,
int16_t* aOutput)
{
int16_t* output = aOutput;
if (0.0f <= aVolume && aVolume <= 1.0f) {
int32_t scale = int32_t((1 << 16) * aVolume);
for (int32_t i = 0; i < aLength; ++i) {
for (int32_t channel = 0; channel < aChannels; ++channel) {
int16_t s = aSourceChannels[channel][i];
*output = int16_t((int32_t(s) * scale) >> 16);
++output;
}
}
return;
}
for (int32_t i = 0; i < aLength; ++i) {
for (int32_t channel = 0; channel < aChannels; ++channel) {
float v = AudioSampleToFloat(aSourceChannels[channel][i])*aVolume;
*output = FloatToAudioSample<int16_t>(v);
++output;
}
}
}
void
InterleaveAndConvertBuffer(const void** aSourceChannels,
AudioSampleFormat aSourceFormat,
int32_t aLength, float aVolume,
int32_t aChannels,
AudioDataValue* aOutput)
{
switch (aSourceFormat) {
case AUDIO_FORMAT_FLOAT32:
InterleaveAndConvertBuffer(reinterpret_cast<const float**>(aSourceChannels),
aLength,
aVolume,
aChannels,
aOutput);
break;
case AUDIO_FORMAT_S16:
InterleaveAndConvertBuffer(reinterpret_cast<const int16_t**>(aSourceChannels),
aLength,
aVolume,
aChannels,
aOutput);
break;
}
}
void
AudioSegment::ApplyVolume(float aVolume)
{
for (ChunkIterator ci(*this); !ci.IsEnded(); ci.Next()) {
ci->mVolume *= aVolume;
}
}
static const int AUDIO_PROCESSING_FRAMES = 640; /* > 10ms of 48KHz audio */
static const uint8_t gZeroChannel[MAX_AUDIO_SAMPLE_SIZE*AUDIO_PROCESSING_FRAMES] = {0};
void
DownmixAndInterleave(const nsTArray<const void*>& aChannelData,
AudioSampleFormat aSourceFormat, int32_t aDuration,
float aVolume, uint32_t aOutputChannels,
AudioDataValue* aOutput)
{
nsAutoTArray<const void*,GUESS_AUDIO_CHANNELS> channelData;
nsAutoTArray<float,AUDIO_PROCESSING_FRAMES*GUESS_AUDIO_CHANNELS> downmixConversionBuffer;
nsAutoTArray<float,AUDIO_PROCESSING_FRAMES*GUESS_AUDIO_CHANNELS> downmixOutputBuffer;
channelData.SetLength(aChannelData.Length());
if (aSourceFormat != AUDIO_FORMAT_FLOAT32) {
NS_ASSERTION(aSourceFormat == AUDIO_FORMAT_S16, "unknown format");
downmixConversionBuffer.SetLength(aDuration*aChannelData.Length());
for (uint32_t i = 0; i < aChannelData.Length(); ++i) {
float* conversionBuf = downmixConversionBuffer.Elements() + (i*aDuration);
const int16_t* sourceBuf = static_cast<const int16_t*>(aChannelData[i]);
for (uint32_t j = 0; j < (uint32_t)aDuration; ++j) {
conversionBuf[j] = AudioSampleToFloat(sourceBuf[j]);
}
channelData[i] = conversionBuf;
}
} else {
for (uint32_t i = 0; i < aChannelData.Length(); ++i) {
channelData[i] = aChannelData[i];
}
}
downmixOutputBuffer.SetLength(aDuration*aOutputChannels);
nsAutoTArray<float*,GUESS_AUDIO_CHANNELS> outputChannelBuffers;
nsAutoTArray<const void*,GUESS_AUDIO_CHANNELS> outputChannelData;
outputChannelBuffers.SetLength(aOutputChannels);
outputChannelData.SetLength(aOutputChannels);
for (uint32_t i = 0; i < (uint32_t)aOutputChannels; ++i) {
outputChannelData[i] = outputChannelBuffers[i] =
downmixOutputBuffer.Elements() + aDuration*i;
}
if (channelData.Length() > aOutputChannels) {
AudioChannelsDownMix(channelData, outputChannelBuffers.Elements(),
aOutputChannels, aDuration);
}
InterleaveAndConvertBuffer(outputChannelData.Elements(), AUDIO_FORMAT_FLOAT32,
aDuration, aVolume, aOutputChannels, aOutput);
}
void
AudioSegment::WriteTo(AudioStream* aOutput)
{
uint32_t outputChannels = aOutput->GetChannels();
nsAutoTArray<AudioDataValue,AUDIO_PROCESSING_FRAMES*GUESS_AUDIO_CHANNELS> buf;
nsAutoTArray<const void*,GUESS_AUDIO_CHANNELS> channelData;
for (ChunkIterator ci(*this); !ci.IsEnded(); ci.Next()) {
AudioChunk& c = *ci;
TrackTicks offset = 0;
while (offset < c.mDuration) {
TrackTicks durationTicks =
std::min<TrackTicks>(c.mDuration - offset, AUDIO_PROCESSING_FRAMES);
if (uint64_t(outputChannels)*durationTicks > INT32_MAX || offset > INT32_MAX) {
NS_ERROR("Buffer overflow");
return;
}
uint32_t duration = uint32_t(durationTicks);
buf.SetLength(outputChannels*duration);
if (c.mBuffer) {
channelData.SetLength(c.mChannelData.Length());
for (uint32_t i = 0; i < channelData.Length(); ++i) {
channelData[i] =
AddAudioSampleOffset(c.mChannelData[i], c.mBufferFormat, int32_t(offset));
}
if (channelData.Length() < outputChannels) {
// Up-mix. Note that this might actually make channelData have more
// than outputChannels temporarily.
AudioChannelsUpMix(&channelData, outputChannels, gZeroChannel);
}
if (channelData.Length() > outputChannels) {
// Down-mix.
DownmixAndInterleave(channelData, c.mBufferFormat, duration,
c.mVolume, outputChannels, buf.Elements());
} else {
InterleaveAndConvertBuffer(channelData.Elements(), c.mBufferFormat,
duration, c.mVolume,
outputChannels,
buf.Elements());
}
} else {
// Assumes that a bit pattern of zeroes == 0.0f
memset(buf.Elements(), 0, buf.Length()*sizeof(AudioDataValue));
}
aOutput->Write(buf.Elements(), int32_t(duration));
offset += duration;
}
}
aOutput->Start();
}
}