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Bug 805254. Part 12: Simplify AudioSegment::WriteTo and related code now that the output format is known statically. r=kinetik

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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.
This commit is contained in:
Robert O'Callahan 2012-10-25 23:09:41 +13:00
parent 3fe2b7b13b
commit 8bd04e7aec
2 changed files with 28 additions and 60 deletions
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65
content/media/AudioSegment.cpp
View File

@ -27,7 +27,7 @@ InterleaveAndConvertBuffer(const SrcT* aSource, int32_t aSourceLength,
}
}
static void
static inline void
InterleaveAndConvertBuffer(const int16_t* aSource, int32_t aSourceLength,
int32_t aLength,
float aVolume,
@ -35,59 +35,51 @@ InterleaveAndConvertBuffer(const int16_t* aSource, int32_t aSourceLength,
int16_t* aOutput)
{
int16_t* output = aOutput;
float v = NS_MAX(NS_MIN(aVolume, 1.0f), -1.0f);
int32_t volume = int32_t((1 << 16) * v);
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) {
int16_t s = aSource[channel*aSourceLength + i];
*output = int16_t((int32_t(s) * volume) >> 16);
float v = AudioSampleToFloat(aSource[channel*aSourceLength + i])*aVolume;
*output = FloatToAudioSample<int16_t>(v);
++output;
}
}
}
template <class SrcT>
static void
InterleaveAndConvertBuffer(const SrcT* aSource, int32_t aSourceLength,
int32_t aLength,
float aVolume,
int32_t aChannels,
void* aOutput, AudioSampleFormat aOutputFormat)
{
switch (aOutputFormat) {
case AUDIO_FORMAT_FLOAT32:
InterleaveAndConvertBuffer(aSource, aSourceLength, aLength, aVolume,
aChannels, static_cast<float*>(aOutput));
break;
case AUDIO_FORMAT_S16:
InterleaveAndConvertBuffer(aSource, aSourceLength, aLength, aVolume,
aChannels, static_cast<int16_t*>(aOutput));
break;
}
}
static void
InterleaveAndConvertBuffer(const void* aSource, AudioSampleFormat aSourceFormat,
int32_t aSourceLength,
int32_t aOffset, int32_t aLength,
float aVolume,
int32_t aChannels,
void* aOutput, AudioSampleFormat aOutputFormat)
AudioDataValue* aOutput)
{
switch (aSourceFormat) {
case AUDIO_FORMAT_FLOAT32:
InterleaveAndConvertBuffer(static_cast<const float*>(aSource) + aOffset, aSourceLength,
InterleaveAndConvertBuffer(static_cast<const float*>(aSource) + aOffset,
aSourceLength,
aLength,
aVolume,
aChannels,
aOutput, aOutputFormat);
aOutput);
break;
case AUDIO_FORMAT_S16:
InterleaveAndConvertBuffer(static_cast<const int16_t*>(aSource) + aOffset, aSourceLength,
InterleaveAndConvertBuffer(static_cast<const int16_t*>(aSource) + aOffset,
aSourceLength,
aLength,
aVolume,
aChannels,
aOutput, aOutputFormat);
aOutput);
break;
}
}
@ -100,30 +92,29 @@ AudioSegment::ApplyVolume(float aVolume)
}
}
static const int STATIC_AUDIO_BUFFER_BYTES = 50000;
static const int STATIC_AUDIO_SAMPLES = 10000;
void
AudioSegment::WriteTo(nsAudioStream* aOutput)
{
NS_ASSERTION(mChannels == aOutput->GetChannels(), "Wrong number of channels");
nsAutoTArray<uint8_t,STATIC_AUDIO_BUFFER_BYTES> buf;
uint32_t frameSize = GetSampleSize(AUDIO_OUTPUT_FORMAT)*mChannels;
nsAutoTArray<AudioDataValue,STATIC_AUDIO_SAMPLES> buf;
for (ChunkIterator ci(*this); !ci.IsEnded(); ci.Next()) {
AudioChunk& c = *ci;
if (frameSize*c.mDuration > UINT32_MAX) {
if (uint64_t(mChannels)*c.mDuration > INT32_MAX) {
NS_ERROR("Buffer overflow");
return;
}
buf.SetLength(int32_t(frameSize*c.mDuration));
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(), AUDIO_OUTPUT_FORMAT);
buf.Elements());
} else {
// Assumes that a bit pattern of zeroes == 0.0f
memset(buf.Elements(), 0, buf.Length());
memset(buf.Elements(), 0, buf.Length()*sizeof(AudioDataValue));
}
aOutput->Write(buf.Elements(), int32_t(c.mDuration));
}

23
content/media/AudioSegment.h
View File

@ -66,16 +66,6 @@ class AudioSegment : public MediaSegmentBase<AudioSegment, AudioChunk> {
public:
typedef mozilla::AudioSampleFormat SampleFormat;
static int GetSampleSize(SampleFormat aFormat)
{
switch (aFormat) {
case AUDIO_FORMAT_S16: return 2;
case AUDIO_FORMAT_FLOAT32: return 4;
}
NS_ERROR("Bad format");
return 0;
}
AudioSegment() : MediaSegmentBase<AudioSegment, AudioChunk>(AUDIO), mChannels(0) {}
bool IsInitialized()
@ -93,19 +83,6 @@ public:
NS_ASSERTION(IsInitialized(), "Not initialized");
return mChannels;
}
/**
* Returns the format of the first audio frame that has data, or
* AUDIO_FORMAT_FLOAT32 if there is none.
*/
SampleFormat GetFirstFrameFormat()
{
for (ChunkIterator ci(*this); !ci.IsEnded(); ci.Next()) {
if (ci->mBuffer) {
return ci->mBufferFormat;
}
}
return AUDIO_FORMAT_FLOAT32;
}
void AppendFrames(already_AddRefed<SharedBuffer> aBuffer, int32_t aBufferLength,
int32_t aStart, int32_t aEnd, SampleFormat aFormat)
{