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);
}
}
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;
}
// 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
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);
}
}
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));
}
}
}
}
}
