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());
}
}
}
void
AnalyserNode::AppendChunk(const AudioChunk& aChunk)
{
const uint32_t bufferSize = mBuffer.Length();
const uint32_t channelCount = aChunk.mChannelData.Length();
uint32_t chunkDuration = aChunk.mDuration;
MOZ_ASSERT((bufferSize & (bufferSize - 1)) == 0); // Must be a power of two!
MOZ_ASSERT(channelCount > 0);
MOZ_ASSERT(chunkDuration == WEBAUDIO_BLOCK_SIZE);
if (chunkDuration > bufferSize) {
// Copy a maximum bufferSize samples.
chunkDuration = bufferSize;
}
PodCopy(mBuffer.Elements() + mWriteIndex, static_cast<const float*>(aChunk.mChannelData[0]), chunkDuration);
for (uint32_t i = 1; i < channelCount; ++i) {
AudioBlockAddChannelWithScale(static_cast<const float*>(aChunk.mChannelData[i]), 1.0f,
mBuffer.Elements() + mWriteIndex);
}
if (channelCount > 1) {
AudioBlockInPlaceScale(mBuffer.Elements() + mWriteIndex,
1.0f / aChunk.mChannelData.Length());
}
mWriteIndex += chunkDuration;
MOZ_ASSERT(mWriteIndex <= bufferSize);
if (mWriteIndex >= bufferSize) {
mWriteIndex = 0;
}
}
void
CDMProxy::gmp_Decrypted(uint32_t aId,
GMPErr aResult,
const nsTArray<uint8_t>& aDecryptedData)
{
MOZ_ASSERT(IsOnGMPThread());
for (size_t i = 0; i < mDecryptionJobs.Length(); i++) {
DecryptJob* job = mDecryptionJobs[i];
if (job->mId == aId) {
if (aDecryptedData.Length() != job->mSample->size) {
NS_WARNING("CDM returned incorrect number of decrypted bytes");
}
if (GMP_SUCCEEDED(aResult)) {
PodCopy(job->mSample->data,
aDecryptedData.Elements(),
std::min<size_t>(aDecryptedData.Length(), job->mSample->size));
job->mClient->Decrypted(GMPNoErr, job->mSample.forget());
} else if (aResult == GMPNoKeyErr) {
NS_WARNING("CDM returned GMPNoKeyErr");
// We still have the encrypted sample, so we can re-enqueue it to be
// decrypted again once the key is usable again.
job->mClient->Decrypted(GMPNoKeyErr, job->mSample.forget());
} else {
nsAutoCString str("CDM returned decode failure GMPErr=");
str.AppendInt(aResult);
NS_WARNING(str.get());
job->mClient->Decrypted(aResult, nullptr);
}
mDecryptionJobs.RemoveElementAt(i);
return;
}
}
NS_WARNING("GMPDecryptorChild returned incorrect job ID");
}
void InitBrandName()
{
if (sBrandName) {
return;
}
nsXPIDLString brandName;
nsCOMPtr<nsIStringBundleService> stringBundleService =
mozilla::services::GetStringBundleService();
if (stringBundleService) {
nsCOMPtr<nsIStringBundle> brandBundle;
nsresult rv = stringBundleService->CreateBundle(kBrandBundleURL,
getter_AddRefs(brandBundle));
if (NS_SUCCEEDED(rv)) {
rv = brandBundle->GetStringFromName(u"brandShortName",
getter_Copies(brandName));
NS_WARNING_ASSERTION(
NS_SUCCEEDED(rv), "Could not get the program name for a cubeb stream.");
}
}
/* cubeb expects a c-string. */
const char* ascii = NS_LossyConvertUTF16toASCII(brandName).get();
sBrandName = new char[brandName.Length() + 1];
PodCopy(sBrandName.get(), ascii, brandName.Length());
sBrandName[brandName.Length()] = 0;
}
void
AudioBuffer::CopyToChannel(JSContext* aJSContext, const Float32Array& aSource,
uint32_t aChannelNumber, uint32_t aStartInChannel,
ErrorResult& aRv)
{
uint32_t length = aSource.Length();
if (aChannelNumber >= NumberOfChannels() ||
aStartInChannel + length >= mLength) {
aRv.Throw(NS_ERROR_DOM_INDEX_SIZE_ERR);
return;
}
if (!mSharedChannels && JS_GetTypedArrayLength(mJSChannels[aChannelNumber]) != mLength) {
// The array was probably neutered
aRv.Throw(NS_ERROR_DOM_INDEX_SIZE_ERR);
return;
}
if (!RestoreJSChannelData(aJSContext)) {
aRv.Throw(NS_ERROR_OUT_OF_MEMORY);
return;
}
PodCopy(JS_GetFloat32ArrayData(mJSChannels[aChannelNumber]) + aStartInChannel,
aSource.Data(), length);
}
void
AudioBuffer::SetRawChannelContents(uint32_t aChannel, float* aContents)
{
MOZ_ASSERT(!GetWrapperPreserveColor() && !mSharedChannels,
"The AudioBuffer object should not have been handed to JS or have C++ callers neuter its typed array");
PodCopy(JS_GetFloat32ArrayData(mJSChannels[aChannel]), aContents, mLength);
}
void
AudioCaptureStream::MixerCallback(AudioDataValue* aMixedBuffer,
AudioSampleFormat aFormat, uint32_t aChannels,
uint32_t aFrames, uint32_t aSampleRate)
{
AutoTArray<nsTArray<AudioDataValue>, MONO> output;
AutoTArray<const AudioDataValue*, MONO> bufferPtrs;
output.SetLength(MONO);
bufferPtrs.SetLength(MONO);
uint32_t written = 0;
// We need to copy here, because the mixer will reuse the storage, we should
// not hold onto it. Buffers are in planar format.
for (uint32_t channel = 0; channel < aChannels; channel++) {
AudioDataValue* out = output[channel].AppendElements(aFrames);
PodCopy(out, aMixedBuffer + written, aFrames);
bufferPtrs[channel] = out;
written += aFrames;
}
AudioChunk chunk;
chunk.mBuffer = new mozilla::SharedChannelArrayBuffer<AudioDataValue>(&output);
chunk.mDuration = aFrames;
chunk.mBufferFormat = aFormat;
chunk.mVolume = 1.0f;
chunk.mChannelData.SetLength(MONO);
for (uint32_t channel = 0; channel < aChannels; channel++) {
chunk.mChannelData[channel] = bufferPtrs[channel];
}
// Now we have mixed data, simply append it to out track.
EnsureTrack(mTrackId)->Get<AudioSegment>()->AppendAndConsumeChunk(&chunk);
}
/**
* aSrcRect: Rect relative to the aSrc surface
* aDestPoint: Point inside aDest surface
*/
void
CopyRect(DataSourceSurface* aSrc, DataSourceSurface* aDest,
IntRect aSrcRect, IntPoint aDestPoint)
{
if (aSrcRect.Overflows() ||
IntRect(aDestPoint, aSrcRect.Size()).Overflows()) {
MOZ_CRASH("we should never be getting invalid rects at this point");
}
MOZ_RELEASE_ASSERT(aSrc->GetFormat() == aDest->GetFormat(),
"different surface formats");
MOZ_RELEASE_ASSERT(IntRect(IntPoint(), aSrc->GetSize()).Contains(aSrcRect),
"source rect too big for source surface");
MOZ_RELEASE_ASSERT(IntRect(IntPoint(), aDest->GetSize()).Contains(IntRect(aDestPoint, aSrcRect.Size())),
"dest surface too small");
if (aSrcRect.IsEmpty()) {
return;
}
DataSourceSurface::ScopedMap srcMap(aSrc, DataSourceSurface::READ);
DataSourceSurface::ScopedMap destMap(aDest, DataSourceSurface::WRITE);
if (MOZ2D_WARN_IF(!srcMap.IsMapped() || !destMap.IsMapped())) {
return;
}
uint8_t* sourceData = DataAtOffset(aSrc, srcMap.GetMappedSurface(), aSrcRect.TopLeft());
uint32_t sourceStride = srcMap.GetStride();
uint8_t* destData = DataAtOffset(aDest, destMap.GetMappedSurface(), aDestPoint);
uint32_t destStride = destMap.GetStride();
if (BytesPerPixel(aSrc->GetFormat()) == 4) {
for (int32_t y = 0; y < aSrcRect.height; y++) {
PodCopy((int32_t*)destData, (int32_t*)sourceData, aSrcRect.width);
sourceData += sourceStride;
destData += destStride;
}
} else if (BytesPerPixel(aSrc->GetFormat()) == 1) {
for (int32_t y = 0; y < aSrcRect.height; y++) {
PodCopy(destData, sourceData, aSrcRect.width);
sourceData += sourceStride;
destData += destStride;
}
}
}
void
CopyChars(jschar *dest, const JSLinearString &str)
{
AutoCheckCannotGC nogc;
if (str.hasTwoByteChars())
PodCopy(dest, str.twoByteChars(nogc), str.length());
else
CopyAndInflateChars(dest, str.latin1Chars(nogc), str.length());
}
cubeb_log_message(char const str[CUBEB_LOG_MESSAGE_MAX_SIZE])
{
size_t length = strlen(str);
/* paranoia against malformed message */
assert(length < CUBEB_LOG_MESSAGE_MAX_SIZE);
if (length > CUBEB_LOG_MESSAGE_MAX_SIZE - 1) {
return;
}
PodCopy(storage, str, length);
storage[length] = '\0';
}
void AudioBufferCopyWithScale(const float* aInput,
float aScale,
float* aOutput,
uint32_t aSize)
{
if (aScale == 1.0f) {
PodCopy(aOutput, aInput, aSize);
} else {
for (uint32_t i = 0; i < aSize; ++i) {
aOutput[i] = aInput[i]*aScale;
}
}
}
nsresult Output(BufferInfo::Param aInfo, void* aBuffer,
MediaFormat::Param aFormat, const TimeUnit& aDuration)
{
// The output on Android is always 16-bit signed
nsresult rv;
int32_t numChannels;
NS_ENSURE_SUCCESS(rv =
aFormat->GetInteger(NS_LITERAL_STRING("channel-count"), &numChannels), rv);
AudioConfig::ChannelLayout layout(numChannels);
if (!layout.IsValid()) {
return NS_ERROR_FAILURE;
}
int32_t sampleRate;
NS_ENSURE_SUCCESS(rv =
aFormat->GetInteger(NS_LITERAL_STRING("sample-rate"), &sampleRate), rv);
int32_t size;
NS_ENSURE_SUCCESS(rv = aInfo->Size(&size), rv);
int32_t offset;
NS_ENSURE_SUCCESS(rv = aInfo->Offset(&offset), rv);
#ifdef MOZ_SAMPLE_TYPE_S16
const int32_t numSamples = size / 2;
#else
#error We only support 16-bit integer PCM
#endif
const int32_t numFrames = numSamples / numChannels;
AlignedAudioBuffer audio(numSamples);
if (!audio) {
return NS_ERROR_OUT_OF_MEMORY;
}
const uint8_t* bufferStart = static_cast<uint8_t*>(aBuffer) + offset;
PodCopy(audio.get(), reinterpret_cast<const AudioDataValue*>(bufferStart),
numSamples);
int64_t presentationTimeUs;
NS_ENSURE_SUCCESS(rv = aInfo->PresentationTimeUs(&presentationTimeUs), rv);
RefPtr<AudioData> data = new AudioData(0, presentationTimeUs,
aDuration.ToMicroseconds(),
numFrames,
Move(audio),
numChannels,
sampleRate);
INVOKE_CALLBACK(Output, data);
return NS_OK;
}
ReverbConvolverStage::ReverbConvolverStage(const float* impulseResponse, size_t, size_t reverbTotalLatency, size_t stageOffset, size_t stageLength,
size_t fftSize, size_t renderPhase, size_t renderSliceSize, ReverbAccumulationBuffer* accumulationBuffer, bool directMode)
: m_accumulationBuffer(accumulationBuffer)
, m_accumulationReadIndex(0)
, m_inputReadIndex(0)
, m_directMode(directMode)
{
MOZ_ASSERT(impulseResponse);
MOZ_ASSERT(accumulationBuffer);
if (!m_directMode) {
m_fftKernel = new FFTBlock(fftSize);
m_fftKernel->PadAndMakeScaledDFT(impulseResponse + stageOffset, stageLength);
m_fftConvolver = new FFTConvolver(fftSize);
} else {
m_directKernel.SetLength(fftSize / 2);
PodCopy(m_directKernel.Elements(), impulseResponse + stageOffset, fftSize / 2);
m_directConvolver = new DirectConvolver(renderSliceSize);
}
m_temporaryBuffer.SetLength(renderSliceSize);
PodZero(m_temporaryBuffer.Elements(), m_temporaryBuffer.Length());
// The convolution stage at offset stageOffset needs to have a corresponding delay to cancel out the offset.
size_t totalDelay = stageOffset + reverbTotalLatency;
// But, the FFT convolution itself incurs fftSize / 2 latency, so subtract this out...
size_t halfSize = fftSize / 2;
if (!m_directMode) {
MOZ_ASSERT(totalDelay >= halfSize);
if (totalDelay >= halfSize)
totalDelay -= halfSize;
}
// We divide up the total delay, into pre and post delay sections so that we can schedule at exactly the moment when the FFT will happen.
// This is coordinated with the other stages, so they don't all do their FFTs at the same time...
int maxPreDelayLength = std::min(halfSize, totalDelay);
m_preDelayLength = totalDelay > 0 ? renderPhase % maxPreDelayLength : 0;
if (m_preDelayLength > totalDelay)
m_preDelayLength = 0;
m_postDelayLength = totalDelay - m_preDelayLength;
m_preReadWriteIndex = 0;
m_framesProcessed = 0; // total frames processed so far
size_t delayBufferSize = m_preDelayLength < fftSize ? fftSize : m_preDelayLength;
delayBufferSize = delayBufferSize < renderSliceSize ? renderSliceSize : delayBufferSize;
m_preDelayBuffer.SetLength(delayBufferSize);
PodZero(m_preDelayBuffer.Elements(), m_preDelayBuffer.Length());
}
void
MediaEngineWebRTCMicrophoneSource::InsertInGraph(const T* aBuffer,
size_t aFrames,
uint32_t aChannels)
{
if (mState != kStarted) {
return;
}
if (MOZ_LOG_TEST(AudioLogModule(), LogLevel::Debug)) {
mTotalFrames += aFrames;
if (mTotalFrames > mLastLogFrames + mSampleFrequency) { // ~ 1 second
MOZ_LOG(AudioLogModule(), LogLevel::Debug,
("%p: Inserting %" PRIuSIZE " samples into graph, total frames = %" PRIu64,
(void*)this, aFrames, mTotalFrames));
mLastLogFrames = mTotalFrames;
}
}
size_t len = mSources.Length();
for (size_t i = 0; i < len; i++) {
if (!mSources[i]) {
continue;
}
RefPtr<SharedBuffer> buffer =
SharedBuffer::Create(aFrames * aChannels * sizeof(T));
PodCopy(static_cast<T*>(buffer->Data()),
aBuffer, aFrames * aChannels);
TimeStamp insertTime;
// Make sure we include the stream and the track.
// The 0:1 is a flag to note when we've done the final insert for a given input block.
LogTime(AsyncLatencyLogger::AudioTrackInsertion,
LATENCY_STREAM_ID(mSources[i].get(), mTrackID),
(i+1 < len) ? 0 : 1, insertTime);
nsAutoPtr<AudioSegment> segment(new AudioSegment());
AutoTArray<const T*, 1> channels;
// XXX Bug 971528 - Support stereo capture in gUM
MOZ_ASSERT(aChannels == 1,
"GraphDriver only supports us stereo audio for now");
channels.AppendElement(static_cast<T*>(buffer->Data()));
segment->AppendFrames(buffer.forget(), channels, aFrames,
mPrincipalHandles[i]);
segment->GetStartTime(insertTime);
mSources[i]->AppendToTrack(mTrackID, segment);
}
}
nsresult
MediaCodecDataDecoder::QueueSample(const MediaRawData* aSample)
{
MOZ_ASSERT(aSample);
AutoLocalJNIFrame frame(jni::GetEnvForThread(), 1);
// We have a sample, try to feed it to the decoder.
int32_t inputIndex = -1;
nsresult res = mDecoder->DequeueInputBuffer(kDecoderTimeout, &inputIndex);
if (NS_FAILED(res)) {
return res;
}
if (inputIndex < 0) {
// There is no valid input buffer available.
return NS_ERROR_FAILURE;
}
jni::Object::LocalRef buffer(frame.GetEnv());
res = GetInputBuffer(frame.GetEnv(), inputIndex, &buffer);
if (NS_FAILED(res)) {
return res;
}
void* directBuffer = frame.GetEnv()->GetDirectBufferAddress(buffer.Get());
MOZ_ASSERT(frame.GetEnv()->GetDirectBufferCapacity(buffer.Get()) >=
aSample->Size(),
"Decoder buffer is not large enough for sample");
PodCopy(static_cast<uint8_t*>(directBuffer), aSample->Data(), aSample->Size());
CryptoInfo::LocalRef cryptoInfo = GetCryptoInfoFromSample(aSample);
if (cryptoInfo) {
res = mDecoder->QueueSecureInputBuffer(inputIndex, 0, cryptoInfo,
aSample->mTime, 0);
} else {
res = mDecoder->QueueInputBuffer(inputIndex, 0, aSample->Size(),
aSample->mTime, 0);
}
if (NS_FAILED(res)) {
return res;
}
mDurations.push_back(TimeUnit::FromMicroseconds(aSample->mDuration));
return NS_OK;
}
void
ConvolverNode::SetBuffer(JSContext* aCx, AudioBuffer* aBuffer, ErrorResult& aRv)
{
if (aBuffer) {
switch (aBuffer->NumberOfChannels()) {
case 1:
case 2:
case 4:
// Supported number of channels
break;
default:
aRv.Throw(NS_ERROR_DOM_SYNTAX_ERR);
return;
}
}
mBuffer = aBuffer;
// Send the buffer to the stream
AudioNodeStream* ns = static_cast<AudioNodeStream*>(mStream.get());
MOZ_ASSERT(ns, "Why don't we have a stream here?");
if (mBuffer) {
uint32_t length = mBuffer->Length();
nsRefPtr<ThreadSharedFloatArrayBufferList> data =
mBuffer->GetThreadSharedChannelsForRate(aCx);
if (data && length < WEBAUDIO_BLOCK_SIZE) {
// For very small impulse response buffers, we need to pad the
// buffer with 0 to make sure that the Reverb implementation
// has enough data to compute FFTs from.
length = WEBAUDIO_BLOCK_SIZE;
nsRefPtr<ThreadSharedFloatArrayBufferList> paddedBuffer =
new ThreadSharedFloatArrayBufferList(data->GetChannels());
float* channelData = (float*) malloc(sizeof(float) * length * data->GetChannels());
for (uint32_t i = 0; i < data->GetChannels(); ++i) {
PodCopy(channelData + length * i, data->GetData(i), mBuffer->Length());
PodZero(channelData + length * i + mBuffer->Length(), WEBAUDIO_BLOCK_SIZE - mBuffer->Length());
paddedBuffer->SetData(i, (i == 0) ? channelData : nullptr, free, channelData);
}
data = paddedBuffer;
}
SendInt32ParameterToStream(ConvolverNodeEngine::BUFFER_LENGTH, length);
SendDoubleParameterToStream(ConvolverNodeEngine::SAMPLE_RATE,
mBuffer->SampleRate());
ns->SetBuffer(data.forget());
} else {
ns->SetBuffer(nullptr);
}
}
void
MediaEngineWebRTCMicrophoneSource::InsertInGraph(const T* aBuffer,
size_t aFrames,
uint32_t aChannels)
{
if (mState != kStarted) {
return;
}
size_t len = mSources.Length();
for (size_t i = 0; i < len; i++) {
if (!mSources[i]) {
continue;
}
RefPtr<SharedBuffer> buffer =
SharedBuffer::Create(aFrames * aChannels * sizeof(T));
PodCopy(static_cast<T*>(buffer->Data()),
aBuffer, aFrames * aChannels);
TimeStamp insertTime;
// Make sure we include the stream and the track.
// The 0:1 is a flag to note when we've done the final insert for a given input block.
LogTime(AsyncLatencyLogger::AudioTrackInsertion,
LATENCY_STREAM_ID(mSources[i].get(), mTrackID),
(i+1 < len) ? 0 : 1, insertTime);
nsAutoPtr<AudioSegment> segment(new AudioSegment());
AutoTArray<const T*, 1> channels;
// XXX Bug 971528 - Support stereo capture in gUM
MOZ_ASSERT(aChannels == 1,
"GraphDriver only supports us stereo audio for now");
channels.AppendElement(static_cast<T*>(buffer->Data()));
segment->AppendFrames(buffer.forget(), channels, aFrames,
mPrincipalHandles[i]);
segment->GetStartTime(insertTime);
RUN_ON_THREAD(mThread,
WrapRunnable(mSources[i], &SourceMediaStream::AppendToTrack,
mTrackID, segment,
static_cast<AudioSegment*>(nullptr)),
NS_DISPATCH_NORMAL);
}
}
/**
* 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);
}
}
}
DWORD WINAPI
QueryDosDeviceWHook(LPCWSTR lpDeviceName, LPWSTR lpTargetPath, DWORD ucchMax)
{
if (!sDeviceNames) {
return 0;
}
std::wstring name = std::wstring(lpDeviceName);
auto iter = sDeviceNames->find(name);
if (iter == sDeviceNames->end()) {
return 0;
}
const std::wstring& device = iter->second;
if (device.size() + 1 > ucchMax) {
return 0;
}
PodCopy(lpTargetPath, device.c_str(), device.size());
lpTargetPath[device.size()] = 0;
GMP_LOG("QueryDosDeviceWHook %S -> %S", lpDeviceName, lpTargetPath);
return name.size();
}
JSFlatString*
NewStringCopyNDontDeflate(ExclusiveContext* cx, const CharT* s, size_t n)
{
if (JSInlineString::lengthFits<CharT>(n))
return NewInlineString<allowGC>(cx, mozilla::Range<const CharT>(s, n));
ScopedJSFreePtr<CharT> news(cx->pod_malloc<CharT>(n + 1));
if (!news)
return nullptr;
PodCopy(news.get(), s, n);
news[n] = 0;
JSFlatString* str = JSFlatString::new_<allowGC>(cx, news.get(), n);
if (!str)
return nullptr;
news.forget();
return str;
}
void
CDMProxy::DecryptJob::PostResult(GMPErr aResult, const nsTArray<uint8_t>& aDecryptedData)
{
if (aDecryptedData.Length() != mSample->Size()) {
NS_WARNING("CDM returned incorrect number of decrypted bytes");
}
if (GMP_SUCCEEDED(aResult)) {
nsAutoPtr<MediaRawDataWriter> writer(mSample->CreateWriter());
PodCopy(writer->Data(),
aDecryptedData.Elements(),
std::min<size_t>(aDecryptedData.Length(), mSample->Size()));
} else if (aResult == GMPNoKeyErr) {
NS_WARNING("CDM returned GMPNoKeyErr");
// We still have the encrypted sample, so we can re-enqueue it to be
// decrypted again once the key is usable again.
} else {
nsAutoCString str("CDM returned decode failure GMPErr=");
str.AppendInt(aResult);
NS_WARNING(str.get());
}
mPromise.Resolve(DecryptResult(aResult, mSample), __func__);
}
void
CopyChars(Latin1Char* dest, const JSLinearString& str)
{
AutoCheckCannotGC nogc;
if (str.hasLatin1Chars()) {
PodCopy(dest, str.latin1Chars(nogc), str.length());
} else {
/*
* When we flatten a TwoByte rope, we turn child ropes (including Latin1
* ropes) into TwoByte dependent strings. If one of these strings is
* also part of another Latin1 rope tree, we can have a Latin1 rope with
* a TwoByte descendent and we end up here when we flatten it. Although
* the chars are stored as TwoByte, we know they must be in the Latin1
* range, so we can safely deflate here.
*/
size_t len = str.length();
const char16_t* chars = str.twoByteChars(nogc);
for (size_t i = 0; i < len; i++) {
MOZ_ASSERT(chars[i] <= JSString::MAX_LATIN1_CHAR);
dest[i] = chars[i];
}
}
}
nsresult Output(BufferInfo::Param aInfo, void* aBuffer,
MediaFormat::Param aFormat,
const media::TimeUnit& aDuration) {
// The output on Android is always 16-bit signed
nsresult rv;
int32_t numChannels;
NS_ENSURE_SUCCESS(rv =
aFormat->GetInteger(NS_LITERAL_STRING("channel-count"), &numChannels), rv);
int32_t sampleRate;
NS_ENSURE_SUCCESS(rv =
aFormat->GetInteger(NS_LITERAL_STRING("sample-rate"), &sampleRate), rv);
int32_t size;
NS_ENSURE_SUCCESS(rv = aInfo->Size(&size), rv);
const int32_t numFrames = (size / numChannels) / 2;
AudioDataValue* audio = new AudioDataValue[size];
PodCopy(audio, static_cast<AudioDataValue*>(aBuffer), size);
int32_t offset;
NS_ENSURE_SUCCESS(rv = aInfo->Offset(&offset), rv);
int64_t presentationTimeUs;
NS_ENSURE_SUCCESS(rv = aInfo->PresentationTimeUs(&presentationTimeUs), rv);
nsRefPtr<AudioData> data = new AudioData(offset, presentationTimeUs,
aDuration.ToMicroseconds(),
numFrames,
audio,
numChannels,
sampleRate);
ENVOKE_CALLBACK(Output, data);
return NS_OK;
}
void
CDMProxy::gmp_Decrypted(uint32_t aId,
GMPErr aResult,
const nsTArray<uint8_t>& aDecryptedData)
{
MOZ_ASSERT(IsOnGMPThread());
for (size_t i = 0; i < mDecryptionJobs.Length(); i++) {
DecryptJob* job = mDecryptionJobs[i];
if (job->mId == aId) {
if (aDecryptedData.Length() != job->mSample->size) {
NS_WARNING("CDM returned incorrect number of decrypted bytes");
}
PodCopy(job->mSample->data,
aDecryptedData.Elements(),
std::min<size_t>(aDecryptedData.Length(), job->mSample->size));
nsresult rv = GMP_SUCCEEDED(aResult) ? NS_OK : NS_ERROR_FAILURE;
job->mClient->Decrypted(rv, job->mSample.forget());
mDecryptionJobs.RemoveElementAt(i);
return;
} else {
NS_WARNING("GMPDecryptorChild returned incorrect job ID");
}
}
}
void
AudioBuffer::SetRawChannelContents(JSContext* aJSContext, uint32_t aChannel,
float* aContents)
{
PodCopy(JS_GetFloat32ArrayData(mJSChannels[aChannel]), aContents, mLength);
}
void MediaCodecDataDecoder::DecoderLoop()
{
bool outputDone = false;
bool draining = false;
bool waitingEOF = false;
AutoLocalJNIFrame frame(GetJNIForThread(), 1);
nsRefPtr<MediaRawData> sample;
MediaFormat::LocalRef outputFormat(frame.GetEnv());
nsresult res;
for (;;) {
{
MonitorAutoLock lock(mMonitor);
while (!mStopping && !mDraining && !mFlushing && mQueue.empty()) {
if (mQueue.empty()) {
// We could be waiting here forever if we don't signal that we need more input
ENVOKE_CALLBACK(InputExhausted);
}
lock.Wait();
}
if (mStopping) {
// Get out of the loop. This is the only exit point.
break;
}
if (mFlushing) {
mDecoder->Flush();
ClearQueue();
mFlushing = false;
lock.Notify();
continue;
}
if (mDraining && !sample && !waitingEOF) {
draining = true;
}
// We're not stopping or draining, so try to get a sample
if (!mQueue.empty()) {
sample = mQueue.front();
}
}
if (draining && !waitingEOF) {
MOZ_ASSERT(!sample, "Shouldn't have a sample when pushing EOF frame");
int32_t inputIndex;
res = mDecoder->DequeueInputBuffer(DECODER_TIMEOUT, &inputIndex);
HANDLE_DECODER_ERROR();
if (inputIndex >= 0) {
res = mDecoder->QueueInputBuffer(inputIndex, 0, 0, 0, MediaCodec::BUFFER_FLAG_END_OF_STREAM);
HANDLE_DECODER_ERROR();
waitingEOF = true;
}
}
if (sample) {
// We have a sample, try to feed it to the decoder
int inputIndex;
res = mDecoder->DequeueInputBuffer(DECODER_TIMEOUT, &inputIndex);
HANDLE_DECODER_ERROR();
if (inputIndex >= 0) {
jni::Object::LocalRef buffer(frame.GetEnv());
res = GetInputBuffer(frame.GetEnv(), inputIndex, &buffer);
HANDLE_DECODER_ERROR();
void* directBuffer = frame.GetEnv()->GetDirectBufferAddress(buffer.Get());
MOZ_ASSERT(frame.GetEnv()->GetDirectBufferCapacity(buffer.Get()) >= sample->Size(),
"Decoder buffer is not large enough for sample");
{
// We're feeding this to the decoder, so remove it from the queue
MonitorAutoLock lock(mMonitor);
mQueue.pop();
}
PodCopy((uint8_t*)directBuffer, sample->Data(), sample->Size());
res = mDecoder->QueueInputBuffer(inputIndex, 0, sample->Size(),
sample->mTime, 0);
HANDLE_DECODER_ERROR();
mDurations.push(media::TimeUnit::FromMicroseconds(sample->mDuration));
sample = nullptr;
outputDone = false;
}
}
if (!outputDone) {
BufferInfo::LocalRef bufferInfo;
res = BufferInfo::New(&bufferInfo);
HANDLE_DECODER_ERROR();
int32_t outputStatus;
res = mDecoder->DequeueOutputBuffer(bufferInfo, DECODER_TIMEOUT, &outputStatus);
HANDLE_DECODER_ERROR();
if (outputStatus == MediaCodec::INFO_TRY_AGAIN_LATER) {
// We might want to call mCallback->InputExhausted() here, but there seems to be
// some possible bad interactions here with the threading
} else if (outputStatus == MediaCodec::INFO_OUTPUT_BUFFERS_CHANGED) {
res = ResetOutputBuffers();
HANDLE_DECODER_ERROR();
} else if (outputStatus == MediaCodec::INFO_OUTPUT_FORMAT_CHANGED) {
res = mDecoder->GetOutputFormat(ReturnTo(&outputFormat));
HANDLE_DECODER_ERROR();
} else if (outputStatus < 0) {
NS_WARNING("unknown error from decoder!");
ENVOKE_CALLBACK(Error);
// Don't break here just in case it's recoverable. If it's not, others stuff will fail later and
// we'll bail out.
} else {
int32_t flags;
res = bufferInfo->Flags(&flags);
HANDLE_DECODER_ERROR();
// We have a valid buffer index >= 0 here
if (flags & MediaCodec::BUFFER_FLAG_END_OF_STREAM) {
if (draining) {
draining = false;
waitingEOF = false;
mMonitor.Lock();
mDraining = false;
mMonitor.Notify();
mMonitor.Unlock();
ENVOKE_CALLBACK(DrainComplete);
}
mDecoder->ReleaseOutputBuffer(outputStatus, false);
outputDone = true;
// We only queue empty EOF frames, so we're done for now
continue;
}
MOZ_ASSERT(!mDurations.empty(), "Should have had a duration queued");
media::TimeUnit duration;
if (!mDurations.empty()) {
duration = mDurations.front();
mDurations.pop();
}
auto buffer = jni::Object::LocalRef::Adopt(
frame.GetEnv()->GetObjectArrayElement(mOutputBuffers.Get(), outputStatus));
if (buffer) {
// The buffer will be null on Android L if we are decoding to a Surface
void* directBuffer = frame.GetEnv()->GetDirectBufferAddress(buffer.Get());
Output(bufferInfo, directBuffer, outputFormat, duration);
}
// The Surface will be updated at this point (for video)
mDecoder->ReleaseOutputBuffer(outputStatus, true);
PostOutput(bufferInfo, outputFormat, duration);
}
}
}
Cleanup();
// We're done
MonitorAutoLock lock(mMonitor);
mStopping = false;
mMonitor.Notify();
}
void Reverb::process(const AudioBlock* sourceBus, AudioBlock* destinationBus)
{
// Do a fairly comprehensive sanity check.
// If these conditions are satisfied, all of the source and destination pointers will be valid for the various matrixing cases.
bool isSafeToProcess = sourceBus && destinationBus && sourceBus->ChannelCount() > 0 && destinationBus->mChannelData.Length() > 0
&& WEBAUDIO_BLOCK_SIZE <= MaxFrameSize && WEBAUDIO_BLOCK_SIZE <= size_t(sourceBus->GetDuration()) && WEBAUDIO_BLOCK_SIZE <= size_t(destinationBus->GetDuration());
MOZ_ASSERT(isSafeToProcess);
if (!isSafeToProcess)
return;
// For now only handle mono or stereo output
MOZ_ASSERT(destinationBus->ChannelCount() <= 2);
float* destinationChannelL = static_cast<float*>(const_cast<void*>(destinationBus->mChannelData[0]));
const float* sourceBusL = static_cast<const float*>(sourceBus->mChannelData[0]);
// Handle input -> output matrixing...
size_t numInputChannels = sourceBus->ChannelCount();
size_t numOutputChannels = destinationBus->ChannelCount();
size_t numReverbChannels = m_convolvers.Length();
if (numInputChannels == 2 && numReverbChannels == 2 && numOutputChannels == 2) {
// 2 -> 2 -> 2
const float* sourceBusR = static_cast<const float*>(sourceBus->mChannelData[1]);
float* destinationChannelR = static_cast<float*>(const_cast<void*>(destinationBus->mChannelData[1]));
m_convolvers[0]->process(sourceBusL, destinationChannelL);
m_convolvers[1]->process(sourceBusR, destinationChannelR);
} else if (numInputChannels == 1 && numOutputChannels == 2 && numReverbChannels == 2) {
// 1 -> 2 -> 2
for (int i = 0; i < 2; ++i) {
float* destinationChannel = static_cast<float*>(const_cast<void*>(destinationBus->mChannelData[i]));
m_convolvers[i]->process(sourceBusL, destinationChannel);
}
} else if (numInputChannels == 1 && numReverbChannels == 1 && numOutputChannels == 2) {
// 1 -> 1 -> 2
m_convolvers[0]->process(sourceBusL, destinationChannelL);
// simply copy L -> R
float* destinationChannelR = static_cast<float*>(const_cast<void*>(destinationBus->mChannelData[1]));
bool isCopySafe = destinationChannelL && destinationChannelR && size_t(destinationBus->GetDuration()) >= WEBAUDIO_BLOCK_SIZE;
MOZ_ASSERT(isCopySafe);
if (!isCopySafe)
return;
PodCopy(destinationChannelR, destinationChannelL, WEBAUDIO_BLOCK_SIZE);
} else if (numInputChannels == 1 && numReverbChannels == 1 && numOutputChannels == 1) {
// 1 -> 1 -> 1
m_convolvers[0]->process(sourceBusL, destinationChannelL);
} else if (numInputChannels == 2 && numReverbChannels == 4 && numOutputChannels == 2) {
// 2 -> 4 -> 2 ("True" stereo)
const float* sourceBusR = static_cast<const float*>(sourceBus->mChannelData[1]);
float* destinationChannelR = static_cast<float*>(const_cast<void*>(destinationBus->mChannelData[1]));
float* tempChannelL = static_cast<float*>(const_cast<void*>(m_tempBuffer.mChannelData[0]));
float* tempChannelR = static_cast<float*>(const_cast<void*>(m_tempBuffer.mChannelData[1]));
// Process left virtual source
m_convolvers[0]->process(sourceBusL, destinationChannelL);
m_convolvers[1]->process(sourceBusL, destinationChannelR);
// Process right virtual source
m_convolvers[2]->process(sourceBusR, tempChannelL);
m_convolvers[3]->process(sourceBusR, tempChannelR);
AudioBufferAddWithScale(tempChannelL, 1.0f, destinationChannelL, sourceBus->GetDuration());
AudioBufferAddWithScale(tempChannelR, 1.0f, destinationChannelR, sourceBus->GetDuration());
} else if (numInputChannels == 1 && numReverbChannels == 4 && numOutputChannels == 2) {
// 1 -> 4 -> 2 (Processing mono with "True" stereo impulse response)
// This is an inefficient use of a four-channel impulse response, but we should handle the case.
float* destinationChannelR = static_cast<float*>(const_cast<void*>(destinationBus->mChannelData[1]));
float* tempChannelL = static_cast<float*>(const_cast<void*>(m_tempBuffer.mChannelData[0]));
float* tempChannelR = static_cast<float*>(const_cast<void*>(m_tempBuffer.mChannelData[1]));
// Process left virtual source
m_convolvers[0]->process(sourceBusL, destinationChannelL);
m_convolvers[1]->process(sourceBusL, destinationChannelR);
// Process right virtual source
m_convolvers[2]->process(sourceBusL, tempChannelL);
m_convolvers[3]->process(sourceBusL, tempChannelR);
AudioBufferAddWithScale(tempChannelL, 1.0f, destinationChannelL, sourceBus->GetDuration());
AudioBufferAddWithScale(tempChannelR, 1.0f, destinationChannelR, sourceBus->GetDuration());
} else {
// Handle gracefully any unexpected / unsupported matrixing
// FIXME: add code for 5.1 support...
destinationBus->SetNull(destinationBus->GetDuration());
}
}
nsresult
AppleATDecoder::DecodeSample(mp4_demuxer::MP4Sample* aSample)
{
// Array containing the queued decoded audio frames, about to be output.
nsTArray<AudioDataValue> outputData;
UInt32 channels = mOutputFormat.mChannelsPerFrame;
// Pick a multiple of the frame size close to a power of two
// for efficient allocation.
const uint32_t MAX_AUDIO_FRAMES = 128;
const uint32_t maxDecodedSamples = MAX_AUDIO_FRAMES * channels;
// Descriptions for _decompressed_ audio packets. ignored.
nsAutoArrayPtr<AudioStreamPacketDescription>
packets(new AudioStreamPacketDescription[MAX_AUDIO_FRAMES]);
// This API insists on having packets spoon-fed to it from a callback.
// This structure exists only to pass our state.
PassthroughUserData userData =
{ channels, (UInt32)aSample->size, aSample->data };
// Decompressed audio buffer
nsAutoArrayPtr<AudioDataValue> decoded(new AudioDataValue[maxDecodedSamples]);
do {
AudioBufferList decBuffer;
decBuffer.mNumberBuffers = 1;
decBuffer.mBuffers[0].mNumberChannels = channels;
decBuffer.mBuffers[0].mDataByteSize =
maxDecodedSamples * sizeof(AudioDataValue);
decBuffer.mBuffers[0].mData = decoded.get();
// in: the max number of packets we can handle from the decoder.
// out: the number of packets the decoder is actually returning.
UInt32 numFrames = MAX_AUDIO_FRAMES;
OSStatus rv = AudioConverterFillComplexBuffer(mConverter,
_PassthroughInputDataCallback,
&userData,
&numFrames /* in/out */,
&decBuffer,
packets.get());
if (rv && rv != kNoMoreDataErr) {
LOG("Error decoding audio stream: %d\n", rv);
return NS_ERROR_FAILURE;
}
if (numFrames) {
outputData.AppendElements(decoded.get(), numFrames * channels);
}
if (rv == kNoMoreDataErr) {
break;
}
} while (true);
if (outputData.IsEmpty()) {
return NS_OK;
}
size_t numFrames = outputData.Length() / channels;
int rate = mOutputFormat.mSampleRate;
CheckedInt<Microseconds> duration = FramesToUsecs(numFrames, rate);
if (!duration.isValid()) {
NS_WARNING("Invalid count of accumulated audio samples");
return NS_ERROR_FAILURE;
}
#ifdef LOG_SAMPLE_DECODE
LOG("pushed audio at time %lfs; duration %lfs\n",
(double)aSample->composition_timestamp / USECS_PER_S,
(double)duration.value() / USECS_PER_S);
#endif
nsAutoArrayPtr<AudioDataValue> data(new AudioDataValue[outputData.Length()]);
PodCopy(data.get(), &outputData[0], outputData.Length());
nsRefPtr<AudioData> audio = new AudioData(aSample->byte_offset,
aSample->composition_timestamp,
duration.value(),
numFrames,
data.forget(),
channels,
rate);
mCallback->Output(audio);
return NS_OK;
}
void
CopyChars(Latin1Char *dest, const JSLinearString &str)
{
AutoCheckCannotGC nogc;
PodCopy(dest, str.latin1Chars(nogc), str.length());
}
nsresult
OpusTrackEncoder::GetEncodedTrack(EncodedFrameContainer& aData)
{
PROFILER_LABEL("OpusTrackEncoder", "GetEncodedTrack",
js::ProfileEntry::Category::OTHER);
{
ReentrantMonitorAutoEnter mon(mReentrantMonitor);
// Wait until initialized or cancelled.
while (!mCanceled && !mInitialized) {
mReentrantMonitor.Wait();
}
if (mCanceled || mEncodingComplete) {
return NS_ERROR_FAILURE;
}
}
// calculation below depends on the truth that mInitialized is true.
MOZ_ASSERT(mInitialized);
// re-sampled frames left last time which didn't fit into an Opus packet duration.
const int framesLeft = mResampledLeftover.Length() / mChannels;
// When framesLeft is 0, (GetPacketDuration() - framesLeft) is a multiple
// of kOpusSamplingRate. There is not precision loss in the integer division
// in computing framesToFetch. If frameLeft > 0, we need to add 1 to
// framesToFetch to ensure there will be at least n frames after re-sampling.
const int frameRoundUp = framesLeft ? 1 : 0;
MOZ_ASSERT(GetPacketDuration() >= framesLeft);
// Try to fetch m frames such that there will be n frames
// where (n + frameLeft) >= GetPacketDuration() after re-sampling.
const int framesToFetch = !mResampler ? GetPacketDuration()
: (GetPacketDuration() - framesLeft) * mSamplingRate / kOpusSamplingRate
+ frameRoundUp;
{
// Move all the samples from mRawSegment to mSourceSegment. We only hold
// the monitor in this block.
ReentrantMonitorAutoEnter mon(mReentrantMonitor);
// Wait until enough raw data, end of stream or cancelled.
while (!mCanceled && mRawSegment.GetDuration() +
mSourceSegment.GetDuration() < framesToFetch &&
!mEndOfStream) {
mReentrantMonitor.Wait();
}
if (mCanceled || mEncodingComplete) {
return NS_ERROR_FAILURE;
}
mSourceSegment.AppendFrom(&mRawSegment);
// Pad |mLookahead| samples to the end of source stream to prevent lost of
// original data, the pcm duration will be calculated at rate 48K later.
if (mEndOfStream && !mEosSetInEncoder) {
mEosSetInEncoder = true;
mSourceSegment.AppendNullData(mLookahead);
}
}
// Start encoding data.
nsAutoTArray<AudioDataValue, 9600> pcm;
pcm.SetLength(GetPacketDuration() * mChannels);
AudioSegment::ChunkIterator iter(mSourceSegment);
int frameCopied = 0;
while (!iter.IsEnded() && frameCopied < framesToFetch) {
AudioChunk chunk = *iter;
// Chunk to the required frame size.
int frameToCopy = chunk.GetDuration();
if (frameCopied + frameToCopy > framesToFetch) {
frameToCopy = framesToFetch - frameCopied;
}
if (!chunk.IsNull()) {
// Append the interleaved data to the end of pcm buffer.
AudioTrackEncoder::InterleaveTrackData(chunk, frameToCopy, mChannels,
pcm.Elements() + frameCopied * mChannels);
} else {
memset(pcm.Elements() + frameCopied * mChannels, 0,
frameToCopy * mChannels * sizeof(AudioDataValue));
}
frameCopied += frameToCopy;
iter.Next();
}
RefPtr<EncodedFrame> audiodata = new EncodedFrame();
audiodata->SetFrameType(EncodedFrame::OPUS_AUDIO_FRAME);
int framesInPCM = frameCopied;
if (mResampler) {
nsAutoTArray<AudioDataValue, 9600> resamplingDest;
// We want to consume all the input data, so we slightly oversize the
// resampled data buffer so we can fit the output data in. We cannot really
// predict the output frame count at each call.
uint32_t outframes = frameCopied * kOpusSamplingRate / mSamplingRate + 1;
uint32_t inframes = frameCopied;
resamplingDest.SetLength(outframes * mChannels);
#if MOZ_SAMPLE_TYPE_S16
short* in = reinterpret_cast<short*>(pcm.Elements());
short* out = reinterpret_cast<short*>(resamplingDest.Elements());
speex_resampler_process_interleaved_int(mResampler, in, &inframes,
out, &outframes);
#else
float* in = reinterpret_cast<float*>(pcm.Elements());
float* out = reinterpret_cast<float*>(resamplingDest.Elements());
speex_resampler_process_interleaved_float(mResampler, in, &inframes,
out, &outframes);
#endif
MOZ_ASSERT(pcm.Length() >= mResampledLeftover.Length());
PodCopy(pcm.Elements(), mResampledLeftover.Elements(),
mResampledLeftover.Length());
uint32_t outframesToCopy = std::min(outframes,
static_cast<uint32_t>(GetPacketDuration() - framesLeft));
MOZ_ASSERT(pcm.Length() - mResampledLeftover.Length() >=
outframesToCopy * mChannels);
PodCopy(pcm.Elements() + mResampledLeftover.Length(),
resamplingDest.Elements(), outframesToCopy * mChannels);
int frameLeftover = outframes - outframesToCopy;
mResampledLeftover.SetLength(frameLeftover * mChannels);
PodCopy(mResampledLeftover.Elements(),
resamplingDest.Elements() + outframesToCopy * mChannels,
mResampledLeftover.Length());
// This is always at 48000Hz.
framesInPCM = framesLeft + outframesToCopy;
audiodata->SetDuration(framesInPCM);
} else {
// The ogg time stamping and pre-skip is always timed at 48000.
audiodata->SetDuration(frameCopied * (kOpusSamplingRate / mSamplingRate));
}
// Remove the raw data which has been pulled to pcm buffer.
// The value of frameCopied should equal to (or smaller than, if eos)
// GetPacketDuration().
mSourceSegment.RemoveLeading(frameCopied);
// Has reached the end of input stream and all queued data has pulled for
// encoding.
if (mSourceSegment.GetDuration() == 0 && mEndOfStream) {
mEncodingComplete = true;
LOG("[Opus] Done encoding.");
}
MOZ_ASSERT(mEndOfStream || framesInPCM == GetPacketDuration());
// Append null data to pcm buffer if the leftover data is not enough for
// opus encoder.
if (framesInPCM < GetPacketDuration() && mEndOfStream) {
PodZero(pcm.Elements() + framesInPCM * mChannels,
(GetPacketDuration() - framesInPCM) * mChannels);
}
nsTArray<uint8_t> frameData;
// Encode the data with Opus Encoder.
frameData.SetLength(MAX_DATA_BYTES);
// result is returned as opus error code if it is negative.
int result = 0;
#ifdef MOZ_SAMPLE_TYPE_S16
const opus_int16* pcmBuf = static_cast<opus_int16*>(pcm.Elements());
result = opus_encode(mEncoder, pcmBuf, GetPacketDuration(),
frameData.Elements(), MAX_DATA_BYTES);
#else
const float* pcmBuf = static_cast<float*>(pcm.Elements());
result = opus_encode_float(mEncoder, pcmBuf, GetPacketDuration(),
frameData.Elements(), MAX_DATA_BYTES);
#endif
frameData.SetLength(result >= 0 ? result : 0);
if (result < 0) {
LOG("[Opus] Fail to encode data! Result: %s.", opus_strerror(result));
}
if (mEncodingComplete) {
if (mResampler) {
speex_resampler_destroy(mResampler);
mResampler = nullptr;
}
mResampledLeftover.SetLength(0);
}
audiodata->SwapInFrameData(frameData);
aData.AppendEncodedFrame(audiodata);
return result >= 0 ? NS_OK : NS_ERROR_FAILURE;
}
