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Bug 842243 - Part 0: Modify MediaSegment and AudioSegment for use by MediaEncoder. r=roc

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This commit is contained in:
Shelly Lin 2013-06-03 17:59:50 +08:00
parent 0adbf805bd
commit 5bc51793ee
3 changed files with 63 additions and 31 deletions
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76
content/media/AudioSegment.cpp
View File

@ -55,7 +55,7 @@ InterleaveAndConvertBuffer(const int16_t** aSourceChannels,
}
}
static void
void
InterleaveAndConvertBuffer(const void** aSourceChannels,
AudioSampleFormat aSourceFormat,
int32_t aLength, float aVolume,
@ -91,14 +91,54 @@ AudioSegment::ApplyVolume(float 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, int32_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;
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;
}
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;
nsAutoTArray<float,AUDIO_PROCESSING_FRAMES*GUESS_AUDIO_CHANNELS> downmixConversionBuffer;
nsAutoTArray<float,AUDIO_PROCESSING_FRAMES*GUESS_AUDIO_CHANNELS> downmixOutputBuffer;
for (ChunkIterator ci(*this); !ci.IsEnded(); ci.Next()) {
AudioChunk& c = *ci;
@ -127,34 +167,8 @@ AudioSegment::WriteTo(AudioStream* aOutput)
if (channelData.Length() > outputChannels) {
// Down-mix.
if (c.mBufferFormat != AUDIO_FORMAT_FLOAT32) {
NS_ASSERTION(c.mBufferFormat == AUDIO_FORMAT_S16, "unknown format");
downmixConversionBuffer.SetLength(duration*channelData.Length());
for (uint32_t i = 0; i < channelData.Length(); ++i) {
float* conversionBuf = downmixConversionBuffer.Elements() + (i*duration);
const int16_t* sourceBuf = static_cast<const int16_t*>(channelData[i]);
for (uint32_t j = 0; j < duration; ++j) {
conversionBuf[j] = AudioSampleToFloat(sourceBuf[j]);
}
channelData[i] = conversionBuf;
}
}
downmixOutputBuffer.SetLength(duration*outputChannels);
nsAutoTArray<float*,GUESS_AUDIO_CHANNELS> outputChannelBuffers;
nsAutoTArray<const void*,GUESS_AUDIO_CHANNELS> outputChannelData;
outputChannelBuffers.SetLength(outputChannels);
outputChannelData.SetLength(outputChannels);
for (uint32_t i = 0; i < outputChannels; ++i) {
outputChannelData[i] = outputChannelBuffers[i] =
downmixOutputBuffer.Elements() + duration*i;
}
AudioChannelsDownMix(channelData, outputChannelBuffers.Elements(),
outputChannels, duration);
InterleaveAndConvertBuffer(outputChannelData.Elements(), AUDIO_FORMAT_FLOAT32,
duration, c.mVolume,
outputChannels,
buf.Elements());
DownmixAndInterleave(channelData, c.mBufferFormat, duration,
c.mVolume, channelData.Length(), buf.Elements());
} else {
InterleaveAndConvertBuffer(channelData.Elements(), c.mBufferFormat,
duration, c.mVolume,

14
content/media/AudioSegment.h
View File

@ -25,6 +25,20 @@ const int GUESS_AUDIO_CHANNELS = 2;
const uint32_t WEBAUDIO_BLOCK_SIZE_BITS = 7;
const uint32_t WEBAUDIO_BLOCK_SIZE = 1 << WEBAUDIO_BLOCK_SIZE_BITS;
void InterleaveAndConvertBuffer(const void** aSourceChannels,
AudioSampleFormat aSourceFormat,
int32_t aLength, float aVolume,
int32_t aChannels,
AudioDataValue* aOutput);
/**
* Down-mix audio channels, and interleave the channel data. A total of
* aOutputChannels*aDuration interleaved samples will be stored into aOutput.
*/
void DownmixAndInterleave(const nsTArray<const void*>& aChannelData,
AudioSampleFormat aSourceFormat, int32_t aDuration,
float aVolume, int32_t aOutputChannels,
AudioDataValue* aOutput);
/**
* An AudioChunk represents a multi-channel buffer of audio samples.
* It references an underlying ThreadSharedObject which manages the lifetime

4
content/media/MediaSegment.h
View File

@ -209,6 +209,10 @@ public:
uint32_t mIndex;
};
void RemoveLeading(TrackTicks aDuration)
{
RemoveLeading(aDuration, 0);
}
protected:
MediaSegmentBase(Type aType) : MediaSegment(aType) {}