gecko/content/media/AudioSegment.cpp
Robert O'Callahan 8d0872c840 Bug 805254. Part 12: Simplify AudioSegment::WriteTo and related code now that the output format is known statically. r=kinetik
Also fixes what I think is a bug in InterleaveAndConvertBuffer converting S16 to S16.
Instead of clamping the volume, we should handle arbitrary volumes by falling back
to the float conversion path.
2012-10-25 23:09:41 +13:00

124 lines
4.1 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 "nsAudioStream.h"
namespace mozilla {
template <class SrcT, class DestT>
static void
InterleaveAndConvertBuffer(const SrcT* aSource, int32_t aSourceLength,
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(aSource[channel*aSourceLength + i])*aVolume;
*output = FloatToAudioSample<DestT>(v);
++output;
}
}
}
static inline void
InterleaveAndConvertBuffer(const int16_t* aSource, int32_t aSourceLength,
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 = aSource[channel*aSourceLength + 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(aSource[channel*aSourceLength + i])*aVolume;
*output = FloatToAudioSample<int16_t>(v);
++output;
}
}
}
static void
InterleaveAndConvertBuffer(const void* aSource, AudioSampleFormat aSourceFormat,
int32_t aSourceLength,
int32_t aOffset, int32_t aLength,
float aVolume,
int32_t aChannels,
AudioDataValue* aOutput)
{
switch (aSourceFormat) {
case AUDIO_FORMAT_FLOAT32:
InterleaveAndConvertBuffer(static_cast<const float*>(aSource) + aOffset,
aSourceLength,
aLength,
aVolume,
aChannels,
aOutput);
break;
case AUDIO_FORMAT_S16:
InterleaveAndConvertBuffer(static_cast<const int16_t*>(aSource) + aOffset,
aSourceLength,
aLength,
aVolume,
aChannels,
aOutput);
break;
}
}
void
AudioSegment::ApplyVolume(float aVolume)
{
for (ChunkIterator ci(*this); !ci.IsEnded(); ci.Next()) {
ci->mVolume *= aVolume;
}
}
static const int STATIC_AUDIO_SAMPLES = 10000;
void
AudioSegment::WriteTo(nsAudioStream* aOutput)
{
NS_ASSERTION(mChannels == aOutput->GetChannels(), "Wrong number of channels");
nsAutoTArray<AudioDataValue,STATIC_AUDIO_SAMPLES> buf;
for (ChunkIterator ci(*this); !ci.IsEnded(); ci.Next()) {
AudioChunk& c = *ci;
if (uint64_t(mChannels)*c.mDuration > INT32_MAX) {
NS_ERROR("Buffer overflow");
return;
}
buf.SetLength(int32_t(mChannels*c.mDuration));
if (c.mBuffer) {
InterleaveAndConvertBuffer(c.mBuffer->Data(), c.mBufferFormat, c.mBufferLength,
c.mOffset, int32_t(c.mDuration),
c.mVolume,
aOutput->GetChannels(),
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(c.mDuration));
}
}
}