NS_IMETHOD Run() override
{
auto engine =
static_cast<ScriptProcessorNodeEngine*>(mStream->Engine());
AudioChunk output;
output.SetNull(engine->mBufferSize);
{
auto node = static_cast<ScriptProcessorNode*>
(engine->NodeMainThread());
if (!node) {
return NS_OK;
}
if (node->HasListenersFor(nsGkAtoms::onaudioprocess)) {
DispatchAudioProcessEvent(node, &output);
}
// The node may have been destroyed during event dispatch.
}
// Append it to our output buffer queue
engine->GetSharedBuffers()->FinishProducingOutputBuffer(output);
return NS_OK;
}
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 PAudioBuffer::sendToNetwork()
{
int toReach;
int i;
AudioChunk *chunk;
if (ABS(fRdIn - fWrIn) >= SOUNDBUFF_SIZE)
{
toReach = fRdIn + SOUNDBUFF_SIZE;
if (toReach >= fMaxIn)
toReach -= fMaxIn;
i = 0;
while (fRdIn != toReach)
{
_frameBuff[i++] = input[fRdIn++];
if (fRdIn >= fMaxIn)
fRdIn = 0;
}
chunk = _bridge.popUnused();
if (chunk == NULL)
return ;
chunk->clean();
//compressed = _codec->encode(_frameBuff, FRAME_PACKET_SIZE, encodedSize);
//chunk->assign(_frameBuff, (FRAME_PACKET_SIZE * sizeof(float)));
// Raw Mode
chunk->assign(_frameBuff, SOUNDBUFF_SIZE);
_bridge.inputPush(chunk);
}
}
void PAudioBuffer::feed()
{
SAMPLE *frames;
AudioChunk *chunk;
unsigned int i = 0;
chunk = _bridge.outputPop();
if (chunk == NULL)
return ;
if (chunk->size() == 0)
return ;
frames = chunk->getContent();
if (frames == NULL)
return ;
i = 0;
while (i < chunk->size())
{
if (fWrOut >= fMaxOut)
fWrOut = 0;
output[fWrOut++] = frames[i++];
// if (fWrOut == fRdOut)
// return ;
}
_bridge.pushUnused(chunk);
}
virtual void pull(AudioChunk &chunk)
{
if (!chunk.length()) return;
if (!valid || finished) {chunk.silence(); return;}
int samples, have = 0, need = chunk.length();
//Create pointers to 16-bit data
short *d16[PG_MAX_CHANNELS];
for (Uint32 i = 0; i < chunk.format().channels; ++i)
d16[i] = (short*) chunk.start(i);
while (true)
{
samples = stb_vorbis_get_samples_short(ogg,
chunk.format().channels, d16, (need-have));
if (samples < 0)
{
finished = true;
//cout << " VORBIS ERROR" << endl;
break;
}
if (samples == 0)
{
//File's end
if (loop)
{
stb_vorbis_seek_start(ogg);
continue;
}
else
{
finished = true;
break;
}
}
for (Uint32 i=0; i < chunk.format().channels; ++i)
d16[i] += samples;
have += samples;
//if (have > need) cout << "VORBIS OVERDRAW" << endl;
//std::cout << "OGG pull: " << have << "/" << need << std::endl;
if (have >= need) break;
}
//Cutoff marker if necessary
if (have < need) chunk.cutoff(have);
//Upsample data to 24-bit Sint32s
for (Uint32 i=0; i < chunk.format().channels; ++i)
{
Sint32 *start = chunk.start(i), *op = start + have;
short *ip = d16[i];
while (op!=start) {*(--op) = 256 * Sint32(*(--ip));}
}
}
void
AudioNodeStream::ObtainInputBlock(AudioChunk& aTmpChunk, uint32_t aPortIndex)
{
uint32_t inputCount = mInputs.Length();
uint32_t outputChannelCount = 1;
nsAutoTArray<AudioChunk*,250> inputChunks;
for (uint32_t i = 0; i < inputCount; ++i) {
if (aPortIndex != mInputs[i]->InputNumber()) {
// This input is connected to a different port
continue;
}
MediaStream* s = mInputs[i]->GetSource();
AudioNodeStream* a = static_cast<AudioNodeStream*>(s);
MOZ_ASSERT(a == s->AsAudioNodeStream());
if (a->IsAudioParamStream()) {
continue;
}
AudioChunk* chunk = &a->mLastChunks[mInputs[i]->OutputNumber()];
MOZ_ASSERT(chunk);
if (chunk->IsNull() || chunk->mChannelData.IsEmpty()) {
continue;
}
inputChunks.AppendElement(chunk);
outputChannelCount =
GetAudioChannelsSuperset(outputChannelCount, chunk->mChannelData.Length());
}
outputChannelCount = ComputedNumberOfChannels(outputChannelCount);
uint32_t inputChunkCount = inputChunks.Length();
if (inputChunkCount == 0 ||
(inputChunkCount == 1 && inputChunks[0]->mChannelData.Length() == 0)) {
aTmpChunk.SetNull(WEBAUDIO_BLOCK_SIZE);
return;
}
if (inputChunkCount == 1 &&
inputChunks[0]->mChannelData.Length() == outputChannelCount) {
aTmpChunk = *inputChunks[0];
return;
}
if (outputChannelCount == 0) {
aTmpChunk.SetNull(WEBAUDIO_BLOCK_SIZE);
return;
}
AllocateAudioBlock(outputChannelCount, &aTmpChunk);
// The static storage here should be 1KB, so it's fine
nsAutoTArray<float, GUESS_AUDIO_CHANNELS*WEBAUDIO_BLOCK_SIZE> downmixBuffer;
for (uint32_t i = 0; i < inputChunkCount; ++i) {
AccumulateInputChunk(i, *inputChunks[i], &aTmpChunk, &downmixBuffer);
}
}
// Read audio data in aChunk, resample them if needed,
// and then send the result to OMX input buffer (or buffers if one buffer is not enough).
// aSamplesRead will be the number of samples that have been read from aChunk.
BufferState ReadChunk(AudioChunk& aChunk, size_t* aSamplesRead)
{
size_t chunkSamples = aChunk.GetDuration();
size_t bytesToCopy = chunkSamples * mOMXAEncoder.mResamplingRatio
* mOMXAEncoder.mChannels * sizeof(AudioDataValue);
size_t bytesCopied = 0;
if (bytesToCopy <= AvailableSize()) {
if (aChunk.IsNull()) {
bytesCopied = SendSilenceToBuffer(chunkSamples);
} else {
bytesCopied = SendChunkToBuffer(aChunk, chunkSamples);
}
UpdateAfterSendChunk(chunkSamples, bytesCopied, aSamplesRead);
} else {
// Interleave data to a temporary buffer.
nsAutoTArray<AudioDataValue, 9600> pcm;
pcm.SetLength(bytesToCopy);
AudioDataValue* interleavedSource = pcm.Elements();
AudioTrackEncoder::InterleaveTrackData(aChunk, chunkSamples,
mOMXAEncoder.mChannels,
interleavedSource);
// When the data size of chunk is larger than the buffer capacity,
// we split it into sub-chunks to fill up buffers.
size_t subChunkSamples = 0;
while(GetNextSubChunk(bytesToCopy, subChunkSamples)) {
// To avoid enqueueing an empty buffer, we follow the order that
// clear up buffer first, then create one, send data to it in the end.
if (!IsEmpty()) {
// Submit the filled-up buffer and request a new buffer.
status_t result = Enqueue(mOMXAEncoder.mTimestamp,
mInputFlags & ~OMXCodecWrapper::BUFFER_EOS);
if (result != OK) {
return BUFFER_FAIL;
}
result = Dequeue();
if (result == -EAGAIN) {
return WAIT_FOR_NEW_BUFFER;
}
if (result != OK) {
return BUFFER_FAIL;
}
}
if (aChunk.IsNull()) {
bytesCopied = SendSilenceToBuffer(subChunkSamples);
} else {
bytesCopied = SendInterleavedSubChunkToBuffer(interleavedSource, subChunkSamples);
}
UpdateAfterSendChunk(subChunkSamples, bytesCopied, aSamplesRead);
// Move to the position where samples are not yet send to the buffer.
interleavedSource += subChunkSamples * mOMXAEncoder.mChannels;
}
}
return BUFFER_OK;
}
void
AudioTrackEncoder::NotifyQueuedTrackChanges(MediaStreamGraph* aGraph,
TrackID aID,
StreamTime aTrackOffset,
uint32_t aTrackEvents,
const MediaSegment& aQueuedMedia)
{
if (mCanceled) {
return;
}
const AudioSegment& audio = static_cast<const AudioSegment&>(aQueuedMedia);
// Check and initialize parameters for codec encoder.
if (!mInitialized) {
mInitCounter++;
TRACK_LOG(LogLevel::Debug, ("Init the audio encoder %d times", mInitCounter));
AudioSegment::ChunkIterator iter(const_cast<AudioSegment&>(audio));
while (!iter.IsEnded()) {
AudioChunk chunk = *iter;
// The number of channels is determined by the first non-null chunk, and
// thus the audio encoder is initialized at this time.
if (!chunk.IsNull()) {
nsresult rv = Init(chunk.mChannelData.Length(), aGraph->GraphRate());
if (NS_FAILED(rv)) {
LOG("[AudioTrackEncoder]: Fail to initialize the encoder!");
NotifyCancel();
}
break;
}
iter.Next();
}
mNotInitDuration += aQueuedMedia.GetDuration();
if (!mInitialized &&
(mNotInitDuration / aGraph->GraphRate() > INIT_FAILED_DURATION) &&
mInitCounter > 1) {
LOG("[AudioTrackEncoder]: Initialize failed for 30s.");
NotifyEndOfStream();
return;
}
}
// Append and consume this raw segment.
AppendAudioSegment(audio);
// The stream has stopped and reached the end of track.
if (aTrackEvents == MediaStreamListener::TRACK_EVENT_ENDED) {
LOG("[AudioTrackEncoder]: Receive TRACK_EVENT_ENDED .");
NotifyEndOfStream();
}
}
void Splicer::pull(AudioChunk &chunk)
{
Uint32 left = chunk.length(), chans = chunk.format().channels;
Sint32 *data[PG_MAX_CHANNELS];
for (Uint32 i = 0; i < chans; ++i) data[i] = chunk.start(i);
//Query exhausted each loop to refresh the value of "current".
while (!exhausted())
{
//Pull data from next stream
AudioChunk sub(chunk.audio, output, data, left,
chunk.frame(), chunk.a(), chunk.b());
current->pull(sub);
//Partial advance
if (current->exhausted())
{
Uint32 cut = sub.cutoff();
for (Uint32 i = 0; i < chans; ++i) data[i] += cut;
left -= cut;
current = NULL;
if (left) continue;
}
return;
}
//The Splicer is exhausted!
chunk.cutoff(data[0] - chunk.start(0));
}
void
AudioTrackEncoder::NotifyQueuedTrackChanges(MediaStreamGraph* aGraph,
TrackID aID,
TrackRate aTrackRate,
TrackTicks aTrackOffset,
uint32_t aTrackEvents,
const MediaSegment& aQueuedMedia)
{
if (mCanceled) {
return;
}
const AudioSegment& audio = static_cast<const AudioSegment&>(aQueuedMedia);
// Check and initialize parameters for codec encoder.
if (!mInitialized) {
#ifdef PR_LOGGING
mAudioInitCounter++;
TRACK_LOG(PR_LOG_DEBUG, ("Init the audio encoder %d times", mAudioInitCounter));
#endif
AudioSegment::ChunkIterator iter(const_cast<AudioSegment&>(audio));
while (!iter.IsEnded()) {
AudioChunk chunk = *iter;
// The number of channels is determined by the first non-null chunk, and
// thus the audio encoder is initialized at this time.
if (!chunk.IsNull()) {
nsresult rv = Init(chunk.mChannelData.Length(), aTrackRate);
if (NS_FAILED(rv)) {
LOG("[AudioTrackEncoder]: Fail to initialize the encoder!");
NotifyCancel();
}
break;
}
iter.Next();
}
}
// Append and consume this raw segment.
AppendAudioSegment(audio);
// The stream has stopped and reached the end of track.
if (aTrackEvents == MediaStreamListener::TRACK_EVENT_ENDED) {
LOG("[AudioTrackEncoder]: Receive TRACK_EVENT_ENDED .");
NotifyEndOfStream();
}
}
/**
* 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);
}
}
}
void Bandpass::pull(AudioChunk &chunk,
const Bandpass_Node &a, const Bandpass_Node &b)
{
//Pull source data
source.pull(chunk);
//Calculate RC multipliers
float
al = RCCONV / ((a.low<=0.0f)?40000.0f:a.low),
ah = RCCONV / ((a.high<=0.0f)?10.0f:a.high),
bl = RCCONV / ((b.low<=0.0f)?40000.0f:b.low),
bh = RCCONV / ((b.high<=0.0f)?10.0f:b.high);
float lpRC = al, hpRC = ah,
lpM = pow(bl/al, 1.0f / float(chunk.length())),
hpM = pow(bh/ah, 1.0f / float(chunk.length())),
lpA, hpA, samp,
dt = 1.0f / float(chunk.format().rate);
//Apply effect!
Uint32 chan = source.format().channels;
for (Uint32 i = 0; i < chan; ++i)
{
Sint32 *pos = chunk.start(i), *end = chunk.end(i);
float &lpPc = lpP[i], &hpDc = hpD[i];
while (pos < end)
{
//Interpolate settings
lpA = dt / (lpRC + dt); lpRC *= lpM;
hpA = hpRC / (hpRC + dt); hpRC *= hpM;
//Get samples
samp = float(*pos);
//Lowpass
samp = lpPc + lpA * (samp-lpPc);
lpPc = samp;
//Highpass (confusing but correct)
samp = hpA * (samp+hpDc);
hpDc = samp - lpPc;
//Set samples
*pos = Sint32(samp);
++pos;
}
}
}
void Signal::pull(AudioChunk &chunk) const
{
if (!data) {chunk.silence(); return;}
if (data->trans) data->trans->pull(chunk);
else data->stream->pull(chunk);
}
/* static */ already_AddRefed<AudioBuffer>
AudioBuffer::Create(nsPIDOMWindowInner* aWindow, float aSampleRate,
AudioChunk&& aInitialContents)
{
AudioChunk initialContents = aInitialContents;
ErrorResult rv;
RefPtr<AudioBuffer> buffer =
new AudioBuffer(aWindow, initialContents.ChannelCount(),
initialContents.mDuration, aSampleRate, rv);
if (rv.Failed()) {
return nullptr;
}
buffer->mSharedChannels = Move(aInitialContents);
return buffer.forget();
}
/**
* 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);
}
}
}
virtual void pull(AudioChunk &chunk)
{
//This is possible at startup; race conditions are bad.
if (!back) chunk.silence();
//Shorthand.
Uint32 frame = back->mikeFrame, length = back->mikeLength;
const Sint32 *data = back->mikeData;
//How much information is available?
if (readFrame != frame) {readFrame = frame; prog = 0;}
Uint32 want = chunk.length(), get = std::min(length-prog, want);
//Read what we can from the buffer
std::memcpy((void*)chunk.start(0), (const void*)(data+prog), 4*get);
prog += get;
//Fill your cup too full and it will spill...
if (get < want)
std::memset((void*)(chunk.start(0)+get), 0, 4*(want-get));
}
// The MediaStreamGraph guarantees that this is actually one block, for
// AudioNodeStreams.
void
AudioNodeStream::ProduceOutput(GraphTime aFrom, GraphTime aTo)
{
StreamBuffer::Track* track = EnsureTrack();
AudioChunk outputChunk;
AudioSegment* segment = track->Get<AudioSegment>();
outputChunk.SetNull(0);
if (mInCycle) {
// XXX DelayNode not supported yet so just produce silence
outputChunk.SetNull(WEBAUDIO_BLOCK_SIZE);
} else {
AudioChunk tmpChunk;
AudioChunk* inputChunk = ObtainInputBlock(&tmpChunk);
bool finished = false;
mEngine->ProduceAudioBlock(this, *inputChunk, &outputChunk, &finished);
if (finished) {
FinishOutput();
}
}
mLastChunk = outputChunk;
if (mKind == MediaStreamGraph::EXTERNAL_STREAM) {
segment->AppendAndConsumeChunk(&outputChunk);
} else {
segment->AppendNullData(outputChunk.GetDuration());
}
for (uint32_t j = 0; j < mListeners.Length(); ++j) {
MediaStreamListener* l = mListeners[j];
AudioChunk copyChunk = outputChunk;
AudioSegment tmpSegment;
tmpSegment.AppendAndConsumeChunk(©Chunk);
l->NotifyQueuedTrackChanges(Graph(), AUDIO_NODE_STREAM_TRACK_ID,
IdealAudioRate(), segment->GetDuration(), 0,
tmpSegment);
}
}
// graph thread
AudioChunk GetOutputBuffer()
{
MOZ_ASSERT(!NS_IsMainThread());
AudioChunk buffer;
{
MutexAutoLock lock(mOutputQueue.Lock());
if (mOutputQueue.ReadyToConsume() > 0) {
if (mDelaySoFar == STREAM_TIME_MAX) {
mDelaySoFar = 0;
}
buffer = mOutputQueue.Consume();
} else {
// If we're out of buffers to consume, just output silence
buffer.SetNull(WEBAUDIO_BLOCK_SIZE);
if (mDelaySoFar != STREAM_TIME_MAX) {
// Remember the delay that we just hit
mDelaySoFar += WEBAUDIO_BLOCK_SIZE;
}
}
}
return buffer;
}
void
DelayBuffer::Write(const AudioChunk& aInputChunk)
{
// We must have a reference to the buffer if there are channels
MOZ_ASSERT(aInputChunk.IsNull() == !aInputChunk.mChannelData.Length());
#ifdef DEBUG
MOZ_ASSERT(!mHaveWrittenBlock);
mHaveWrittenBlock = true;
#endif
if (!EnsureBuffer()) {
return;
}
if (mCurrentChunk == mLastReadChunk) {
mLastReadChunk = -1; // invalidate cache
}
mChunks[mCurrentChunk] = aInputChunk;
}
nsresult
VorbisTrackEncoder::GetEncodedTrack(EncodedFrameContainer& aData)
{
if (mEosSetInEncoder) {
return NS_OK;
}
PROFILER_LABEL("VorbisTrackEncoder", "GetEncodedTrack",
js::ProfileEntry::Category::OTHER);
nsAutoPtr<AudioSegment> sourceSegment;
sourceSegment = new AudioSegment();
{
// Move all the samples from mRawSegment to sourceSegment. We only hold
// the monitor in this block.
ReentrantMonitorAutoEnter mon(mReentrantMonitor);
// Wait if mEncoder is not initialized, or when not enough raw data, but is
// not the end of stream nor is being canceled.
while (!mCanceled && mRawSegment.GetDuration() < GetPacketDuration() &&
!mEndOfStream) {
mon.Wait();
}
VORBISLOG("GetEncodedTrack passes wait, duration is %lld\n",
mRawSegment.GetDuration());
if (mCanceled || mEncodingComplete) {
return NS_ERROR_FAILURE;
}
sourceSegment->AppendFrom(&mRawSegment);
}
if (mEndOfStream && (sourceSegment->GetDuration() == 0)
&& !mEosSetInEncoder) {
mEncodingComplete = true;
mEosSetInEncoder = true;
VORBISLOG("[Vorbis] Done encoding.");
vorbis_analysis_wrote(&mVorbisDsp, 0);
GetEncodedFrames(aData);
return NS_OK;
}
// Start encoding data.
AudioSegment::ChunkIterator iter(*sourceSegment);
AudioDataValue **vorbisBuffer =
vorbis_analysis_buffer(&mVorbisDsp, (int)sourceSegment->GetDuration());
int framesCopied = 0;
AutoTArray<AudioDataValue, 9600> interleavedPcm;
AutoTArray<AudioDataValue, 9600> nonInterleavedPcm;
interleavedPcm.SetLength(sourceSegment->GetDuration() * mChannels);
nonInterleavedPcm.SetLength(sourceSegment->GetDuration() * mChannels);
while (!iter.IsEnded()) {
AudioChunk chunk = *iter;
int frameToCopy = chunk.GetDuration();
if (!chunk.IsNull()) {
InterleaveTrackData(chunk, frameToCopy, mChannels,
interleavedPcm.Elements() + framesCopied * mChannels);
} else { // empty data
memset(interleavedPcm.Elements() + framesCopied * mChannels, 0,
frameToCopy * mChannels * sizeof(AudioDataValue));
}
framesCopied += frameToCopy;
iter.Next();
}
// De-interleave the interleavedPcm.
DeInterleaveTrackData(interleavedPcm.Elements(), framesCopied, mChannels,
nonInterleavedPcm.Elements());
// Copy the nonInterleavedPcm to vorbis buffer.
for(uint8_t i = 0; i < mChannels; ++i) {
memcpy(vorbisBuffer[i], nonInterleavedPcm.Elements() + framesCopied * i,
framesCopied * sizeof(AudioDataValue));
}
// Now the vorbisBuffer contain the all data in non-interleaved.
// Tell the library how much we actually submitted.
vorbis_analysis_wrote(&mVorbisDsp, framesCopied);
VORBISLOG("vorbis_analysis_wrote framesCopied %d\n", framesCopied);
GetEncodedFrames(aData);
return NS_OK;
}
void
PannerNodeEngine::EqualPowerPanningFunction(const AudioChunk& aInput,
AudioChunk* aOutput)
{
if (aInput.IsNull()) {
*aOutput = aInput;
return;
}
float azimuth, elevation, gainL, gainR, normalizedAzimuth, distanceGain, coneGain;
int inputChannels = aInput.mChannelData.Length();
// If both the listener are in the same spot, and no cone gain is specified,
// this node is noop.
if (mListenerPosition == mPosition &&
mConeInnerAngle == 360 &&
mConeOuterAngle == 360) {
*aOutput = aInput;
return;
}
// The output of this node is always stereo, no matter what the inputs are.
AllocateAudioBlock(2, aOutput);
ComputeAzimuthAndElevation(azimuth, elevation);
coneGain = ComputeConeGain();
// The following algorithm is described in the spec.
// Clamp azimuth in the [-90, 90] range.
azimuth = min(180.f, max(-180.f, azimuth));
// Wrap around
if (azimuth < -90.f) {
azimuth = -180.f - azimuth;
} else if (azimuth > 90) {
azimuth = 180.f - azimuth;
}
// Normalize the value in the [0, 1] range.
if (inputChannels == 1) {
normalizedAzimuth = (azimuth + 90.f) / 180.f;
} else {
if (azimuth <= 0) {
normalizedAzimuth = (azimuth + 90.f) / 90.f;
} else {
normalizedAzimuth = azimuth / 90.f;
}
}
distanceGain = ComputeDistanceGain();
// Actually compute the left and right gain.
gainL = cos(0.5 * M_PI * normalizedAzimuth) * aInput.mVolume;
gainR = sin(0.5 * M_PI * normalizedAzimuth) * aInput.mVolume;
// Compute the output.
if (inputChannels == 1) {
GainMonoToStereo(aInput, aOutput, gainL, gainR);
} else {
GainStereoToStereo(aInput, aOutput, gainL, gainR, azimuth);
}
DistanceAndConeGain(aOutput, distanceGain * coneGain);
}
void
AudioNodeStream::ObtainInputBlock(AudioChunk& aTmpChunk, uint32_t aPortIndex)
{
uint32_t inputCount = mInputs.Length();
uint32_t outputChannelCount = 1;
nsAutoTArray<AudioChunk*,250> inputChunks;
for (uint32_t i = 0; i < inputCount; ++i) {
if (aPortIndex != mInputs[i]->InputNumber()) {
// This input is connected to a different port
continue;
}
MediaStream* s = mInputs[i]->GetSource();
AudioNodeStream* a = static_cast<AudioNodeStream*>(s);
MOZ_ASSERT(a == s->AsAudioNodeStream());
if (a->IsAudioParamStream()) {
continue;
}
// It is possible for mLastChunks to be empty here, because `a` might be a
// AudioNodeStream that has not been scheduled yet, because it is further
// down the graph _but_ as a connection to this node. Because we enforce the
// presence of at least one DelayNode, with at least one block of delay, and
// because the output of a DelayNode when it has been fed less that
// `delayTime` amount of audio is silence, we can simply continue here,
// because this input would not influence the output of this node. Next
// iteration, a->mLastChunks.IsEmpty() will be false, and everthing will
// work as usual.
if (a->mLastChunks.IsEmpty()) {
continue;
}
AudioChunk* chunk = &a->mLastChunks[mInputs[i]->OutputNumber()];
MOZ_ASSERT(chunk);
if (chunk->IsNull() || chunk->mChannelData.IsEmpty()) {
continue;
}
inputChunks.AppendElement(chunk);
outputChannelCount =
GetAudioChannelsSuperset(outputChannelCount, chunk->mChannelData.Length());
}
outputChannelCount = ComputedNumberOfChannels(outputChannelCount);
uint32_t inputChunkCount = inputChunks.Length();
if (inputChunkCount == 0 ||
(inputChunkCount == 1 && inputChunks[0]->mChannelData.Length() == 0)) {
aTmpChunk.SetNull(WEBAUDIO_BLOCK_SIZE);
return;
}
if (inputChunkCount == 1 &&
inputChunks[0]->mChannelData.Length() == outputChannelCount) {
aTmpChunk = *inputChunks[0];
return;
}
if (outputChannelCount == 0) {
aTmpChunk.SetNull(WEBAUDIO_BLOCK_SIZE);
return;
}
AllocateAudioBlock(outputChannelCount, &aTmpChunk);
// The static storage here should be 1KB, so it's fine
nsAutoTArray<float, GUESS_AUDIO_CHANNELS*WEBAUDIO_BLOCK_SIZE> downmixBuffer;
for (uint32_t i = 0; i < inputChunkCount; ++i) {
AccumulateInputChunk(i, *inputChunks[i], &aTmpChunk, &downmixBuffer);
}
}
AudioChunk*
AudioNodeStream::ObtainInputBlock(AudioChunk* aTmpChunk)
{
uint32_t inputCount = mInputs.Length();
uint32_t outputChannelCount = 0;
nsAutoTArray<AudioChunk*,250> inputChunks;
for (uint32_t i = 0; i < inputCount; ++i) {
MediaStream* s = mInputs[i]->GetSource();
AudioNodeStream* a = static_cast<AudioNodeStream*>(s);
MOZ_ASSERT(a == s->AsAudioNodeStream());
if (a->IsFinishedOnGraphThread()) {
continue;
}
AudioChunk* chunk = &a->mLastChunk;
// XXX when we implement DelayNode, this will no longer be true and we'll
// need to treat a null chunk (when the DelayNode hasn't had a chance
// to produce data yet) as silence here.
MOZ_ASSERT(chunk);
if (chunk->IsNull()) {
continue;
}
inputChunks.AppendElement(chunk);
outputChannelCount =
GetAudioChannelsSuperset(outputChannelCount, chunk->mChannelData.Length());
}
uint32_t inputChunkCount = inputChunks.Length();
if (inputChunkCount == 0) {
aTmpChunk->SetNull(WEBAUDIO_BLOCK_SIZE);
return aTmpChunk;
}
if (inputChunkCount == 1) {
return inputChunks[0];
}
AllocateAudioBlock(outputChannelCount, aTmpChunk);
for (uint32_t i = 0; i < inputChunkCount; ++i) {
AudioChunk* chunk = inputChunks[i];
nsAutoTArray<const void*,GUESS_AUDIO_CHANNELS> channels;
channels.AppendElements(chunk->mChannelData);
if (channels.Length() < outputChannelCount) {
AudioChannelsUpMix(&channels, outputChannelCount, nullptr);
NS_ASSERTION(outputChannelCount == channels.Length(),
"We called GetAudioChannelsSuperset to avoid this");
}
for (uint32_t c = 0; c < channels.Length(); ++c) {
const float* inputData = static_cast<const float*>(channels[c]);
float* outputData = static_cast<float*>(const_cast<void*>(aTmpChunk->mChannelData[c]));
if (inputData) {
if (i == 0) {
AudioBlockCopyChannelWithScale(inputData, chunk->mVolume, outputData);
} else {
AudioBlockAddChannelWithScale(inputData, chunk->mVolume, outputData);
}
} else {
if (i == 0) {
memset(outputData, 0, WEBAUDIO_BLOCK_SIZE*sizeof(float));
}
}
}
}
return aTmpChunk;
}
void
AudioNodeExternalInputStream::ProcessInput(GraphTime aFrom, GraphTime aTo,
uint32_t aFlags)
{
// According to spec, number of outputs is always 1.
MOZ_ASSERT(mLastChunks.Length() == 1);
// GC stuff can result in our input stream being destroyed before this stream.
// Handle that.
if (!IsEnabled() || mInputs.IsEmpty() || mPassThrough) {
mLastChunks[0].SetNull(WEBAUDIO_BLOCK_SIZE);
AdvanceOutputSegment();
return;
}
MOZ_ASSERT(mInputs.Length() == 1);
MediaStream* source = mInputs[0]->GetSource();
nsAutoTArray<AudioSegment,1> audioSegments;
uint32_t inputChannels = 0;
for (StreamBuffer::TrackIter tracks(source->mBuffer, MediaSegment::AUDIO);
!tracks.IsEnded(); tracks.Next()) {
const StreamBuffer::Track& inputTrack = *tracks;
const AudioSegment& inputSegment =
*static_cast<AudioSegment*>(inputTrack.GetSegment());
if (inputSegment.IsNull()) {
continue;
}
AudioSegment& segment = *audioSegments.AppendElement();
GraphTime next;
for (GraphTime t = aFrom; t < aTo; t = next) {
MediaInputPort::InputInterval interval = mInputs[0]->GetNextInputInterval(t);
interval.mEnd = std::min(interval.mEnd, aTo);
if (interval.mStart >= interval.mEnd)
break;
next = interval.mEnd;
StreamTime outputStart = GraphTimeToStreamTime(interval.mStart);
StreamTime outputEnd = GraphTimeToStreamTime(interval.mEnd);
StreamTime ticks = outputEnd - outputStart;
if (interval.mInputIsBlocked) {
segment.AppendNullData(ticks);
} else {
StreamTime inputStart =
std::min(inputSegment.GetDuration(),
source->GraphTimeToStreamTime(interval.mStart));
StreamTime inputEnd =
std::min(inputSegment.GetDuration(),
source->GraphTimeToStreamTime(interval.mEnd));
segment.AppendSlice(inputSegment, inputStart, inputEnd);
// Pad if we're looking past the end of the track
segment.AppendNullData(ticks - (inputEnd - inputStart));
}
}
for (AudioSegment::ChunkIterator iter(segment); !iter.IsEnded(); iter.Next()) {
inputChannels = GetAudioChannelsSuperset(inputChannels, iter->ChannelCount());
}
}
uint32_t accumulateIndex = 0;
if (inputChannels) {
nsAutoTArray<float,GUESS_AUDIO_CHANNELS*WEBAUDIO_BLOCK_SIZE> downmixBuffer;
for (uint32_t i = 0; i < audioSegments.Length(); ++i) {
AudioChunk tmpChunk;
ConvertSegmentToAudioBlock(&audioSegments[i], &tmpChunk, inputChannels);
if (!tmpChunk.IsNull()) {
if (accumulateIndex == 0) {
AllocateAudioBlock(inputChannels, &mLastChunks[0]);
}
AccumulateInputChunk(accumulateIndex, tmpChunk, &mLastChunks[0], &downmixBuffer);
accumulateIndex++;
}
}
}
if (accumulateIndex == 0) {
mLastChunks[0].SetNull(WEBAUDIO_BLOCK_SIZE);
}
// Using AudioNodeStream's AdvanceOutputSegment to push the media stream graph along with null data.
AdvanceOutputSegment();
}
void
AudioNodeExternalInputStream::ProcessInput(GraphTime aFrom, GraphTime aTo,
uint32_t aFlags)
{
// According to spec, number of outputs is always 1.
mLastChunks.SetLength(1);
// GC stuff can result in our input stream being destroyed before this stream.
// Handle that.
if (mInputs.IsEmpty()) {
mLastChunks[0].SetNull(WEBAUDIO_BLOCK_SIZE);
AdvanceOutputSegment();
return;
}
MOZ_ASSERT(mInputs.Length() == 1);
MediaStream* source = mInputs[0]->GetSource();
nsAutoTArray<AudioSegment,1> audioSegments;
nsAutoTArray<bool,1> trackMapEntriesUsed;
uint32_t inputChannels = 0;
for (StreamBuffer::TrackIter tracks(source->mBuffer, MediaSegment::AUDIO);
!tracks.IsEnded(); tracks.Next()) {
const StreamBuffer::Track& inputTrack = *tracks;
// Create a TrackMapEntry if necessary.
size_t trackMapIndex = GetTrackMapEntry(inputTrack, aFrom);
// Maybe there's nothing in this track yet. If so, ignore it. (While the
// track is only playing silence, we may not be able to determine the
// correct number of channels to start resampling.)
if (trackMapIndex == nsTArray<TrackMapEntry>::NoIndex) {
continue;
}
while (trackMapEntriesUsed.Length() <= trackMapIndex) {
trackMapEntriesUsed.AppendElement(false);
}
trackMapEntriesUsed[trackMapIndex] = true;
TrackMapEntry* trackMap = &mTrackMap[trackMapIndex];
AudioSegment segment;
GraphTime next;
TrackRate inputTrackRate = inputTrack.GetRate();
for (GraphTime t = aFrom; t < aTo; t = next) {
MediaInputPort::InputInterval interval = mInputs[0]->GetNextInputInterval(t);
interval.mEnd = std::min(interval.mEnd, aTo);
if (interval.mStart >= interval.mEnd)
break;
next = interval.mEnd;
// Ticks >= startTicks and < endTicks are in the interval
StreamTime outputEnd = GraphTimeToStreamTime(interval.mEnd);
TrackTicks startTicks = trackMap->mSamplesPassedToResampler + segment.GetDuration();
StreamTime outputStart = GraphTimeToStreamTime(interval.mStart);
NS_ASSERTION(startTicks == TimeToTicksRoundUp(inputTrackRate, outputStart),
"Samples missing");
TrackTicks endTicks = TimeToTicksRoundUp(inputTrackRate, outputEnd);
TrackTicks ticks = endTicks - startTicks;
if (interval.mInputIsBlocked) {
segment.AppendNullData(ticks);
} else {
// See comments in TrackUnionStream::CopyTrackData
StreamTime inputStart = source->GraphTimeToStreamTime(interval.mStart);
StreamTime inputEnd = source->GraphTimeToStreamTime(interval.mEnd);
TrackTicks inputTrackEndPoint =
inputTrack.IsEnded() ? inputTrack.GetEnd() : TRACK_TICKS_MAX;
if (trackMap->mEndOfLastInputIntervalInInputStream != inputStart ||
trackMap->mEndOfLastInputIntervalInOutputStream != outputStart) {
// Start of a new series of intervals where neither stream is blocked.
trackMap->mEndOfConsumedInputTicks = TimeToTicksRoundDown(inputTrackRate, inputStart) - 1;
}
TrackTicks inputStartTicks = trackMap->mEndOfConsumedInputTicks;
TrackTicks inputEndTicks = inputStartTicks + ticks;
trackMap->mEndOfConsumedInputTicks = inputEndTicks;
trackMap->mEndOfLastInputIntervalInInputStream = inputEnd;
trackMap->mEndOfLastInputIntervalInOutputStream = outputEnd;
if (inputStartTicks < 0) {
// Data before the start of the track is just null.
segment.AppendNullData(-inputStartTicks);
inputStartTicks = 0;
}
if (inputEndTicks > inputStartTicks) {
segment.AppendSlice(*inputTrack.GetSegment(),
std::min(inputTrackEndPoint, inputStartTicks),
std::min(inputTrackEndPoint, inputEndTicks));
}
// Pad if we're looking past the end of the track
segment.AppendNullData(ticks - segment.GetDuration());
}
}
trackMap->mSamplesPassedToResampler += segment.GetDuration();
trackMap->ResampleInputData(&segment);
if (trackMap->mResampledData.GetDuration() < mCurrentOutputPosition + WEBAUDIO_BLOCK_SIZE) {
// We don't have enough data. Delay it.
trackMap->mResampledData.InsertNullDataAtStart(
mCurrentOutputPosition + WEBAUDIO_BLOCK_SIZE - trackMap->mResampledData.GetDuration());
}
audioSegments.AppendElement()->AppendSlice(trackMap->mResampledData,
mCurrentOutputPosition, mCurrentOutputPosition + WEBAUDIO_BLOCK_SIZE);
trackMap->mResampledData.ForgetUpTo(mCurrentOutputPosition + WEBAUDIO_BLOCK_SIZE);
inputChannels = GetAudioChannelsSuperset(inputChannels, trackMap->mResamplerChannelCount);
}
for (int32_t i = mTrackMap.Length() - 1; i >= 0; --i) {
if (i >= int32_t(trackMapEntriesUsed.Length()) || !trackMapEntriesUsed[i]) {
mTrackMap.RemoveElementAt(i);
}
}
uint32_t accumulateIndex = 0;
if (inputChannels) {
nsAutoTArray<float,GUESS_AUDIO_CHANNELS*WEBAUDIO_BLOCK_SIZE> downmixBuffer;
for (uint32_t i = 0; i < audioSegments.Length(); ++i) {
AudioChunk tmpChunk;
ConvertSegmentToAudioBlock(&audioSegments[i], &tmpChunk);
if (!tmpChunk.IsNull()) {
if (accumulateIndex == 0) {
AllocateAudioBlock(inputChannels, &mLastChunks[0]);
}
AccumulateInputChunk(accumulateIndex, tmpChunk, &mLastChunks[0], &downmixBuffer);
accumulateIndex++;
}
}
}
if (accumulateIndex == 0) {
mLastChunks[0].SetNull(WEBAUDIO_BLOCK_SIZE);
}
mCurrentOutputPosition += WEBAUDIO_BLOCK_SIZE;
// Using AudioNodeStream's AdvanceOutputSegment to push the media stream graph along with null data.
AdvanceOutputSegment();
}
void
AudioNodeStream::ObtainInputBlock(AudioChunk& aTmpChunk, uint32_t aPortIndex)
{
uint32_t inputCount = mInputs.Length();
uint32_t outputChannelCount = 1;
nsAutoTArray<AudioChunk*,250> inputChunks;
for (uint32_t i = 0; i < inputCount; ++i) {
if (aPortIndex != mInputs[i]->InputNumber()) {
// This input is connected to a different port
continue;
}
MediaStream* s = mInputs[i]->GetSource();
AudioNodeStream* a = static_cast<AudioNodeStream*>(s);
MOZ_ASSERT(a == s->AsAudioNodeStream());
if (a->IsFinishedOnGraphThread() ||
a->IsAudioParamStream()) {
continue;
}
AudioChunk* chunk = &a->mLastChunks[mInputs[i]->OutputNumber()];
MOZ_ASSERT(chunk);
if (chunk->IsNull()) {
continue;
}
inputChunks.AppendElement(chunk);
outputChannelCount =
GetAudioChannelsSuperset(outputChannelCount, chunk->mChannelData.Length());
}
switch (mChannelCountMode) {
case ChannelCountMode::Explicit:
// Disregard the output channel count that we've calculated, and just use
// mNumberOfInputChannels.
outputChannelCount = mNumberOfInputChannels;
break;
case ChannelCountMode::Clamped_max:
// Clamp the computed output channel count to mNumberOfInputChannels.
outputChannelCount = std::min(outputChannelCount, mNumberOfInputChannels);
break;
case ChannelCountMode::Max:
// Nothing to do here, just shut up the compiler warning.
break;
}
uint32_t inputChunkCount = inputChunks.Length();
if (inputChunkCount == 0 ||
(inputChunkCount == 1 && inputChunks[0]->mChannelData.Length() == 0)) {
aTmpChunk.SetNull(WEBAUDIO_BLOCK_SIZE);
return;
}
if (inputChunkCount == 1 &&
inputChunks[0]->mChannelData.Length() == outputChannelCount) {
aTmpChunk = *inputChunks[0];
return;
}
AllocateAudioBlock(outputChannelCount, &aTmpChunk);
float silenceChannel[WEBAUDIO_BLOCK_SIZE] = {0.f};
// The static storage here should be 1KB, so it's fine
nsAutoTArray<float, GUESS_AUDIO_CHANNELS*WEBAUDIO_BLOCK_SIZE> downmixBuffer;
for (uint32_t i = 0; i < inputChunkCount; ++i) {
AudioChunk* chunk = inputChunks[i];
nsAutoTArray<const void*,GUESS_AUDIO_CHANNELS> channels;
channels.AppendElements(chunk->mChannelData);
if (channels.Length() < outputChannelCount) {
if (mChannelInterpretation == ChannelInterpretation::Speakers) {
AudioChannelsUpMix(&channels, outputChannelCount, nullptr);
NS_ASSERTION(outputChannelCount == channels.Length(),
"We called GetAudioChannelsSuperset to avoid this");
} else {
// Fill up the remaining channels by zeros
for (uint32_t j = channels.Length(); j < outputChannelCount; ++j) {
channels.AppendElement(silenceChannel);
}
}
} else if (channels.Length() > outputChannelCount) {
if (mChannelInterpretation == ChannelInterpretation::Speakers) {
nsAutoTArray<float*,GUESS_AUDIO_CHANNELS> outputChannels;
outputChannels.SetLength(outputChannelCount);
downmixBuffer.SetLength(outputChannelCount * WEBAUDIO_BLOCK_SIZE);
for (uint32_t j = 0; j < outputChannelCount; ++j) {
outputChannels[j] = &downmixBuffer[j * WEBAUDIO_BLOCK_SIZE];
}
AudioChannelsDownMix(channels, outputChannels.Elements(),
outputChannelCount, WEBAUDIO_BLOCK_SIZE);
channels.SetLength(outputChannelCount);
for (uint32_t j = 0; j < channels.Length(); ++j) {
channels[j] = outputChannels[j];
}
} else {
// Drop the remaining channels
channels.RemoveElementsAt(outputChannelCount,
channels.Length() - outputChannelCount);
}
}
for (uint32_t c = 0; c < channels.Length(); ++c) {
const float* inputData = static_cast<const float*>(channels[c]);
float* outputData = static_cast<float*>(const_cast<void*>(aTmpChunk.mChannelData[c]));
if (inputData) {
if (i == 0) {
AudioBlockCopyChannelWithScale(inputData, chunk->mVolume, outputData);
} else {
AudioBlockAddChannelWithScale(inputData, chunk->mVolume, outputData);
}
} else {
if (i == 0) {
memset(outputData, 0, WEBAUDIO_BLOCK_SIZE*sizeof(float));
}
}
}
}
}
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;
}
void Oscillator::pull(AudioChunk &chunk, const State &a, const State &b)
{
//Phase goes from -1 to 1. Wierd? Maybe.
while (phase >= 1.0f) phase -= 2.0f;
while (phase < -1.0f) phase += 2.0f;
Sint32 samp;
float v, x, sq;
float step = (b.freq / output.rate) * 2.0f,
amp = b.amp * 16777215.0f,
ph = phase;
Sint32 *i = chunk.start(0), *e = chunk.end(0);
switch (type)
{
case SINE:
while (i != e)
{
//Simple sine wave
samp = amp*sin(PI * ph);
ph += step;
*i = samp; ++i;
ph -= 2.0f*int(ph*.5f); //Restrict to [-1, 1]
}
break;
case SQUARE:
while (i != e)
{
samp = amp * ((ph>=0.0f) ? 1.0f : -1.0f);
ph += step;
*i = samp; ++i;
ph -= 2.0f*int(ph*.5f); //Restrict to [-1, 1]
}
break;
case TRIANGLE:
while (i != e)
{
samp = amp * (2.0f*std::abs(ph) - 1.0f);
ph += step;
*i = samp; ++i;
ph -= 2.0f*int(ph*.5f); //Restrict to [-1, 1]
}
break;
case SAWTOOTH:
while (i != e)
{
samp = amp * ph;
ph += step;
*i = samp; ++i;
ph -= 2.0f*int(ph*.5f); //Restrict to [-1, 1]
}
break;
case KLAXON:
default:
{
const float PISQ = PI*PI;
while (i != e)
{
//Sine wave approximation gone wrong
sq = PISQ*ph*ph; x = PI*ph*amp;
v = x * 1.02394347; //Makes the endpoints line up
x *= sq; v -= x/6.0f;
x *= sq; v += x/120.0f;
x *= sq; v -= x/5040.0f;
samp = x;
ph += step;
*i = samp; ++i;
ph -= 2.0f*int(ph*.5f); //Restrict to [-1, 1]
}
}
break;
}
phase = ph;
//Copy signal into all other channels
for (Uint32 c = chunk.channels()-1; c > 0; --c)
{
std::memcpy(chunk.start(c), chunk.start(0), 4*chunk.length());
}
}
