static void
CopyChunkToBlock(AudioChunk& aInput, AudioBlock *aBlock,
uint32_t aOffsetInBlock)
{
uint32_t blockChannels = aBlock->ChannelCount();
AutoTArray<const T*,2> channels;
if (aInput.IsNull()) {
channels.SetLength(blockChannels);
PodZero(channels.Elements(), blockChannels);
} else {
const nsTArray<const T*>& inputChannels = aInput.ChannelData<T>();
channels.SetLength(inputChannels.Length());
PodCopy(channels.Elements(), inputChannels.Elements(), channels.Length());
if (channels.Length() != blockChannels) {
// We only need to upmix here because aBlock's channel count has been
// chosen to be a superset of the channel count of every chunk.
AudioChannelsUpMix(&channels, blockChannels, static_cast<T*>(nullptr));
}
}
for (uint32_t c = 0; c < blockChannels; ++c) {
float* outputData = aBlock->ChannelFloatsForWrite(c) + aOffsetInBlock;
if (channels[c]) {
ConvertAudioSamplesWithScale(channels[c], outputData, aInput.GetDuration(), aInput.mVolume);
} else {
PodZero(outputData, aInput.GetDuration());
}
}
}
/**
* Copies the data in aInput to aOffsetInBlock within aBlock.
* aBlock must have been allocated with AllocateInputBlock and have a channel
* count that's a superset of the channels in aInput.
*/
static void
CopyChunkToBlock(const AudioChunk& aInput, AudioChunk *aBlock,
uint32_t aOffsetInBlock)
{
uint32_t blockChannels = aBlock->ChannelCount();
nsAutoTArray<const void*,2> channels;
if (aInput.IsNull()) {
channels.SetLength(blockChannels);
PodZero(channels.Elements(), blockChannels);
} else {
channels.SetLength(aInput.ChannelCount());
PodCopy(channels.Elements(), aInput.mChannelData.Elements(), channels.Length());
if (channels.Length() != blockChannels) {
// We only need to upmix here because aBlock's channel count has been
// chosen to be a superset of the channel count of every chunk.
AudioChannelsUpMix(&channels, blockChannels, nullptr);
}
}
uint32_t duration = aInput.GetDuration();
for (uint32_t c = 0; c < blockChannels; ++c) {
float* outputData =
static_cast<float*>(const_cast<void*>(aBlock->mChannelData[c])) + aOffsetInBlock;
if (channels[c]) {
switch (aInput.mBufferFormat) {
case AUDIO_FORMAT_FLOAT32:
ConvertAudioSamplesWithScale(
static_cast<const float*>(channels[c]), outputData, duration,
aInput.mVolume);
break;
case AUDIO_FORMAT_S16:
ConvertAudioSamplesWithScale(
static_cast<const int16_t*>(channels[c]), outputData, duration,
aInput.mVolume);
break;
default:
NS_ERROR("Unhandled format");
}
} else {
PodZero(outputData, duration);
}
}
}
template<typename SampleFormatType> void
SpeechStreamListener::ConvertAndDispatchAudioChunk(int aDuration, float aVolume,
SampleFormatType* aData, TrackRate aTrackRate )
{
nsRefPtr<SharedBuffer> samples(SharedBuffer::Create(aDuration *
1 * // channel
sizeof(int16_t)));
int16_t* to = static_cast<int16_t*>(samples->Data());
ConvertAudioSamplesWithScale(aData, to, aDuration, aVolume);
mRecognition->FeedAudioData(samples.forget(), aDuration, this,aTrackRate);
}
/**
* Copies the data in aInput to aOffsetInBlock within aBlock. All samples must
* be float. Both chunks must have the same number of channels (or else
* aInput is null). aBlock must have been allocated with AllocateInputBlock.
*/
static void
CopyChunkToBlock(const AudioChunk& aInput, AudioChunk *aBlock, uint32_t aOffsetInBlock)
{
uint32_t d = aInput.GetDuration();
for (uint32_t i = 0; i < aBlock->mChannelData.Length(); ++i) {
float* out = static_cast<float*>(const_cast<void*>(aBlock->mChannelData[i])) +
aOffsetInBlock;
if (aInput.IsNull()) {
PodZero(out, d);
} else {
const float* in = static_cast<const float*>(aInput.mChannelData[i]);
ConvertAudioSamplesWithScale(in, out, d, aInput.mVolume);
}
}
}
long
nsBufferedAudioStream::DataCallback(void* aBuffer, long aFrames)
{
MonitorAutoLock mon(mMonitor);
uint32_t bytesWanted = aFrames * mBytesPerFrame;
// Adjust bytesWanted to fit what is available in mBuffer.
uint32_t available = NS_MIN(bytesWanted, mBuffer.Length());
NS_ABORT_IF_FALSE(available % mBytesPerFrame == 0, "Must copy complete frames");
if (available > 0) {
// Copy each sample from mBuffer to aBuffer, adjusting the volume during the copy.
float scaled_volume = float(GetVolumeScale() * mVolume);
// Fetch input pointers from the ring buffer.
void* input[2];
uint32_t input_size[2];
mBuffer.PopElements(available, &input[0], &input_size[0], &input[1], &input_size[1]);
uint8_t* output = static_cast<uint8_t*>(aBuffer);
for (int i = 0; i < 2; ++i) {
const AudioDataValue* src = static_cast<const AudioDataValue*>(input[i]);
AudioDataValue* dst = reinterpret_cast<AudioDataValue*>(output);
ConvertAudioSamplesWithScale(src, dst, input_size[i]/sizeof(AudioDataValue),
scaled_volume);
output += input_size[i];
}
NS_ABORT_IF_FALSE(mBuffer.Length() % mBytesPerFrame == 0, "Must copy complete frames");
// Notify any blocked Write() call that more space is available in mBuffer.
mon.NotifyAll();
// Calculate remaining bytes requested by caller. If the stream is not
// draining an underrun has occurred, so fill the remaining buffer with
// silence.
bytesWanted -= available;
}
if (mState != DRAINING) {
memset(static_cast<uint8_t*>(aBuffer) + available, 0, bytesWanted);
mLostFrames += bytesWanted / mBytesPerFrame;
bytesWanted = 0;
}
return aFrames - (bytesWanted / mBytesPerFrame);
}
nsresult nsNativeAudioStream::Write(const AudioDataValue* aBuf, uint32_t aFrames)
{
NS_ASSERTION(!mPaused, "Don't write audio when paused, you'll block");
if (mInError)
return NS_ERROR_FAILURE;
uint32_t samples = aFrames * mChannels;
nsAutoArrayPtr<short> s_data(new short[samples]);
float scaled_volume = float(GetVolumeScale() * mVolume);
ConvertAudioSamplesWithScale(aBuf, s_data.get(), samples, scaled_volume);
if (sa_stream_write(static_cast<sa_stream_t*>(mAudioHandle),
s_data.get(),
samples * sizeof(short)) != SA_SUCCESS)
{
PR_LOG(gAudioStreamLog, PR_LOG_ERROR, ("nsNativeAudioStream: sa_stream_write error"));
mInError = true;
return NS_ERROR_FAILURE;
}
return NS_OK;
}
void
AudioNodeExternalInputStream::TrackMapEntry::ResampleChannels(const nsTArray<const void*>& aBuffers,
uint32_t aInputDuration,
AudioSampleFormat aFormat,
float aVolume)
{
NS_ASSERTION(aBuffers.Length() == mResamplerChannelCount,
"Channel count must be correct here");
nsAutoTArray<nsTArray<float>,2> resampledBuffers;
resampledBuffers.SetLength(aBuffers.Length());
nsTArray<float> samplesAdjustedForVolume;
nsAutoTArray<const float*,2> bufferPtrs;
bufferPtrs.SetLength(aBuffers.Length());
for (uint32_t i = 0; i < aBuffers.Length(); ++i) {
AudioSampleFormat format = aFormat;
const void* buffer = aBuffers[i];
if (aVolume != 1.0f) {
format = AUDIO_FORMAT_FLOAT32;
samplesAdjustedForVolume.SetLength(aInputDuration);
switch (aFormat) {
case AUDIO_FORMAT_FLOAT32:
ConvertAudioSamplesWithScale(static_cast<const float*>(buffer),
samplesAdjustedForVolume.Elements(),
aInputDuration, aVolume);
break;
case AUDIO_FORMAT_S16:
ConvertAudioSamplesWithScale(static_cast<const int16_t*>(buffer),
samplesAdjustedForVolume.Elements(),
aInputDuration, aVolume);
break;
default:
MOZ_ASSERT(false);
return;
}
buffer = samplesAdjustedForVolume.Elements();
}
switch (format) {
case AUDIO_FORMAT_FLOAT32:
ResampleChannelBuffer(mResampler, i,
static_cast<const float*>(buffer),
aInputDuration, &resampledBuffers[i]);
break;
case AUDIO_FORMAT_S16:
ResampleChannelBuffer(mResampler, i,
static_cast<const int16_t*>(buffer),
aInputDuration, &resampledBuffers[i]);
break;
default:
MOZ_ASSERT(false);
return;
}
bufferPtrs[i] = resampledBuffers[i].Elements();
NS_ASSERTION(i == 0 ||
resampledBuffers[i].Length() == resampledBuffers[0].Length(),
"Resampler made different decisions for different channels!");
}
uint32_t length = resampledBuffers[0].Length();
nsRefPtr<ThreadSharedObject> buf = new SharedChannelArrayBuffer<float>(&resampledBuffers);
mResampledData.AppendFrames(buf.forget(), bufferPtrs, length);
}
