mirror of
https://github.com/mozilla/gecko-dev.git
synced 2024-11-05 00:25:27 +00:00
baab5848f3
Also introduces a SharedBuffer class, representing a blob of binary data with threadsafe refcounting.
194 lines
6.3 KiB
C++
194 lines
6.3 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"
|
|
|
|
namespace mozilla {
|
|
|
|
static PRUint16
|
|
FlipByteOrderIfBigEndian(PRUint16 aValue)
|
|
{
|
|
PRUint16 s = aValue;
|
|
#if defined(IS_BIG_ENDIAN)
|
|
s = (s << 8) | (s >> 8);
|
|
#endif
|
|
return s;
|
|
}
|
|
|
|
/*
|
|
* 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
|
|
*/
|
|
static float
|
|
SampleToFloat(float aValue)
|
|
{
|
|
return aValue;
|
|
}
|
|
static float
|
|
SampleToFloat(PRUint8 aValue)
|
|
{
|
|
return (aValue - 128)/128.0f;
|
|
}
|
|
static float
|
|
SampleToFloat(PRInt16 aValue)
|
|
{
|
|
return PRInt16(FlipByteOrderIfBigEndian(aValue))/32768.0f;
|
|
}
|
|
|
|
static void
|
|
FloatToSample(float aValue, float* aOut)
|
|
{
|
|
*aOut = aValue;
|
|
}
|
|
static void
|
|
FloatToSample(float aValue, PRUint8* aOut)
|
|
{
|
|
float v = aValue*128 + 128;
|
|
float clamped = NS_MAX(0.0f, NS_MIN(255.0f, v));
|
|
*aOut = PRUint8(clamped);
|
|
}
|
|
static void
|
|
FloatToSample(float aValue, PRInt16* aOut)
|
|
{
|
|
float v = aValue*32768.0f;
|
|
float clamped = NS_MAX(-32768.0f, NS_MIN(32767.0f, v));
|
|
*aOut = PRInt16(FlipByteOrderIfBigEndian(PRInt16(clamped)));
|
|
}
|
|
|
|
template <class SrcT, class DestT>
|
|
static void
|
|
InterleaveAndConvertBuffer(const SrcT* aSource, PRInt32 aSourceLength,
|
|
PRInt32 aLength,
|
|
float aVolume,
|
|
PRInt32 aChannels,
|
|
DestT* aOutput)
|
|
{
|
|
DestT* output = aOutput;
|
|
for (PRInt32 i = 0; i < aLength; ++i) {
|
|
for (PRInt32 channel = 0; channel < aChannels; ++channel) {
|
|
float v = SampleToFloat(aSource[channel*aSourceLength + i])*aVolume;
|
|
FloatToSample(v, output);
|
|
++output;
|
|
}
|
|
}
|
|
}
|
|
|
|
static void
|
|
InterleaveAndConvertBuffer(const PRInt16* aSource, PRInt32 aSourceLength,
|
|
PRInt32 aLength,
|
|
float aVolume,
|
|
PRInt32 aChannels,
|
|
PRInt16* aOutput)
|
|
{
|
|
PRInt16* output = aOutput;
|
|
float v = NS_MAX(NS_MIN(aVolume, 1.0f), -1.0f);
|
|
PRInt32 volume = PRInt32((1 << 16) * v);
|
|
for (PRInt32 i = 0; i < aLength; ++i) {
|
|
for (PRInt32 channel = 0; channel < aChannels; ++channel) {
|
|
PRInt16 s = FlipByteOrderIfBigEndian(aSource[channel*aSourceLength + i]);
|
|
*output = FlipByteOrderIfBigEndian(PRInt16((PRInt32(s) * volume) >> 16));
|
|
++output;
|
|
}
|
|
}
|
|
}
|
|
|
|
template <class SrcT>
|
|
static void
|
|
InterleaveAndConvertBuffer(const SrcT* aSource, PRInt32 aSourceLength,
|
|
PRInt32 aLength,
|
|
float aVolume,
|
|
PRInt32 aChannels,
|
|
void* aOutput, nsAudioStream::SampleFormat aOutputFormat)
|
|
{
|
|
switch (aOutputFormat) {
|
|
case nsAudioStream::FORMAT_FLOAT32:
|
|
InterleaveAndConvertBuffer(aSource, aSourceLength, aLength, aVolume,
|
|
aChannels, static_cast<float*>(aOutput));
|
|
break;
|
|
case nsAudioStream::FORMAT_S16_LE:
|
|
InterleaveAndConvertBuffer(aSource, aSourceLength, aLength, aVolume,
|
|
aChannels, static_cast<PRInt16*>(aOutput));
|
|
break;
|
|
case nsAudioStream::FORMAT_U8:
|
|
InterleaveAndConvertBuffer(aSource, aSourceLength, aLength, aVolume,
|
|
aChannels, static_cast<PRUint8*>(aOutput));
|
|
break;
|
|
}
|
|
}
|
|
|
|
static void
|
|
InterleaveAndConvertBuffer(const void* aSource, nsAudioStream::SampleFormat aSourceFormat,
|
|
PRInt32 aSourceLength,
|
|
PRInt32 aOffset, PRInt32 aLength,
|
|
float aVolume,
|
|
PRInt32 aChannels,
|
|
void* aOutput, nsAudioStream::SampleFormat aOutputFormat)
|
|
{
|
|
switch (aSourceFormat) {
|
|
case nsAudioStream::FORMAT_FLOAT32:
|
|
InterleaveAndConvertBuffer(static_cast<const float*>(aSource) + aOffset, aSourceLength,
|
|
aLength,
|
|
aVolume,
|
|
aChannels,
|
|
aOutput, aOutputFormat);
|
|
break;
|
|
case nsAudioStream::FORMAT_S16_LE:
|
|
InterleaveAndConvertBuffer(static_cast<const PRInt16*>(aSource) + aOffset, aSourceLength,
|
|
aLength,
|
|
aVolume,
|
|
aChannels,
|
|
aOutput, aOutputFormat);
|
|
break;
|
|
case nsAudioStream::FORMAT_U8:
|
|
InterleaveAndConvertBuffer(static_cast<const PRUint8*>(aSource) + aOffset, aSourceLength,
|
|
aLength,
|
|
aVolume,
|
|
aChannels,
|
|
aOutput, aOutputFormat);
|
|
break;
|
|
}
|
|
}
|
|
|
|
void
|
|
AudioSegment::ApplyVolume(float aVolume)
|
|
{
|
|
for (ChunkIterator ci(*this); !ci.IsEnded(); ci.Next()) {
|
|
ci->mVolume *= aVolume;
|
|
}
|
|
}
|
|
|
|
static const int STATIC_AUDIO_BUFFER_BYTES = 50000;
|
|
|
|
void
|
|
AudioSegment::WriteTo(nsAudioStream* aOutput)
|
|
{
|
|
NS_ASSERTION(mChannels == aOutput->GetChannels(), "Wrong number of channels");
|
|
nsAutoTArray<PRUint8,STATIC_AUDIO_BUFFER_BYTES> buf;
|
|
PRUint32 frameSize = GetSampleSize(aOutput->GetFormat())*mChannels;
|
|
for (ChunkIterator ci(*this); !ci.IsEnded(); ci.Next()) {
|
|
AudioChunk& c = *ci;
|
|
if (frameSize*c.mDuration > PR_UINT32_MAX) {
|
|
NS_ERROR("Buffer overflow");
|
|
return;
|
|
}
|
|
buf.SetLength(PRInt32(frameSize*c.mDuration));
|
|
if (c.mBuffer) {
|
|
InterleaveAndConvertBuffer(c.mBuffer->Data(), c.mBufferFormat, c.mBufferLength,
|
|
c.mOffset, PRInt32(c.mDuration),
|
|
c.mVolume,
|
|
aOutput->GetChannels(),
|
|
buf.Elements(), aOutput->GetFormat());
|
|
} else {
|
|
// Assumes that a bit pattern of zeroes == 0.0f
|
|
memset(buf.Elements(), 0, buf.Length());
|
|
}
|
|
aOutput->Write(buf.Elements(), PRInt32(c.mDuration));
|
|
}
|
|
}
|
|
|
|
}
|