Source code

Revision control

Other Tools

/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*-*/
/* This Source Code Form is subject to the terms of the Mozilla Public
* License, v. 2.0. If a copy of the MPL was not distributed with this file,
* You can obtain one at http://mozilla.org/MPL/2.0/. */
#include "AlignedTArray.h"
#include "AlignmentUtils.h"
#include "AudioNodeEngine.h"
#include "AudioNodeExternalInputTrack.h"
#include "AudioChannelFormat.h"
#include "mozilla/dom/MediaStreamAudioSourceNode.h"
using namespace mozilla::dom;
namespace mozilla {
AudioNodeExternalInputTrack::AudioNodeExternalInputTrack(
AudioNodeEngine* aEngine, TrackRate aSampleRate)
: AudioNodeTrack(aEngine, NO_TRACK_FLAGS, aSampleRate) {
MOZ_COUNT_CTOR(AudioNodeExternalInputTrack);
}
AudioNodeExternalInputTrack::~AudioNodeExternalInputTrack() {
MOZ_COUNT_DTOR(AudioNodeExternalInputTrack);
}
/* static */
already_AddRefed<AudioNodeExternalInputTrack>
AudioNodeExternalInputTrack::Create(MediaTrackGraph* aGraph,
AudioNodeEngine* aEngine) {
AudioContext* ctx = aEngine->NodeMainThread()->Context();
MOZ_ASSERT(NS_IsMainThread());
MOZ_ASSERT(aGraph->GraphRate() == ctx->SampleRate());
RefPtr<AudioNodeExternalInputTrack> track =
new AudioNodeExternalInputTrack(aEngine, aGraph->GraphRate());
track->mSuspendedCount += ctx->ShouldSuspendNewTrack();
aGraph->AddTrack(track);
return track.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.
*/
template <typename T>
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());
}
}
}
/**
* Converts the data in aSegment to a single chunk aBlock. aSegment must have
* duration WEBAUDIO_BLOCK_SIZE. aFallbackChannelCount is a superset of the
* channels in every chunk of aSegment. aBlock must be float format or null.
*/
static void ConvertSegmentToAudioBlock(AudioSegment* aSegment,
AudioBlock* aBlock,
int32_t aFallbackChannelCount) {
NS_ASSERTION(aSegment->GetDuration() == WEBAUDIO_BLOCK_SIZE,
"Bad segment duration");
{
AudioSegment::ChunkIterator ci(*aSegment);
NS_ASSERTION(!ci.IsEnded(), "Should be at least one chunk!");
if (ci->GetDuration() == WEBAUDIO_BLOCK_SIZE &&
(ci->IsNull() || ci->mBufferFormat == AUDIO_FORMAT_FLOAT32)) {
bool aligned = true;
for (size_t i = 0; i < ci->mChannelData.Length(); ++i) {
if (!IS_ALIGNED16(ci->mChannelData[i])) {
aligned = false;
break;
}
}
// Return this chunk directly to avoid copying data.
if (aligned) {
*aBlock = *ci;
return;
}
}
}
aBlock->AllocateChannels(aFallbackChannelCount);
uint32_t duration = 0;
for (AudioSegment::ChunkIterator ci(*aSegment); !ci.IsEnded(); ci.Next()) {
switch (ci->mBufferFormat) {
case AUDIO_FORMAT_S16: {
CopyChunkToBlock<int16_t>(*ci, aBlock, duration);
break;
}
case AUDIO_FORMAT_FLOAT32: {
CopyChunkToBlock<float>(*ci, aBlock, duration);
break;
}
case AUDIO_FORMAT_SILENCE: {
// The actual type of the sample does not matter here, but we still need
// to send some audio to the graph.
CopyChunkToBlock<float>(*ci, aBlock, duration);
break;
}
}
duration += ci->GetDuration();
}
}
void AudioNodeExternalInputTrack::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 track being destroyed before this track.
// Handle that.
if (!IsEnabled() || mInputs.IsEmpty() || mPassThrough) {
mLastChunks[0].SetNull(WEBAUDIO_BLOCK_SIZE);
return;
}
MOZ_ASSERT(mInputs.Length() == 1);
MediaTrack* source = mInputs[0]->GetSource();
AutoTArray<AudioSegment, 1> audioSegments;
uint32_t inputChannels = 0;
MOZ_ASSERT(source->GetData()->GetType() == MediaSegment::AUDIO,
"AudioNodeExternalInputTrack shouldn't have a video input");
const AudioSegment& inputSegment =
*mInputs[0]->GetSource()->GetData<AudioSegment>();
if (!inputSegment.IsNull()) {
AudioSegment& segment = *audioSegments.AppendElement();
GraphTime next;
for (GraphTime t = aFrom; t < aTo; t = next) {
MediaInputPort::InputInterval interval =
MediaInputPort::GetNextInputInterval(mInputs[0], t);
interval.mEnd = std::min(interval.mEnd, aTo);
if (interval.mStart >= interval.mEnd) {
break;
}
next = interval.mEnd;
// We know this track does not block during the processing interval ---
// we're not finished, we don't underrun, and we're not suspended.
TrackTime outputStart = GraphTimeToTrackTime(interval.mStart);
TrackTime outputEnd = GraphTimeToTrackTime(interval.mEnd);
TrackTime ticks = outputEnd - outputStart;
if (interval.mInputIsBlocked) {
segment.AppendNullData(ticks);
} else {
// The input track is not blocked in this interval, so no need to call
// GraphTimeToTrackTimeWithBlocking.
TrackTime inputStart =
std::min(inputSegment.GetDuration(),
source->GraphTimeToTrackTime(interval.mStart));
TrackTime inputEnd =
std::min(inputSegment.GetDuration(),
source->GraphTimeToTrackTime(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) {
DownmixBufferType downmixBuffer;
ASSERT_ALIGNED16(downmixBuffer.Elements());
for (auto& audioSegment : audioSegments) {
AudioBlock tmpChunk;
ConvertSegmentToAudioBlock(&audioSegment, &tmpChunk, inputChannels);
if (!tmpChunk.IsNull()) {
if (accumulateIndex == 0) {
mLastChunks[0].AllocateChannels(inputChannels);
}
AccumulateInputChunk(accumulateIndex, tmpChunk, &mLastChunks[0],
&downmixBuffer);
accumulateIndex++;
}
}
}
if (accumulateIndex == 0) {
mLastChunks[0].SetNull(WEBAUDIO_BLOCK_SIZE);
}
}
bool AudioNodeExternalInputTrack::IsEnabled() {
return ((MediaStreamAudioSourceNodeEngine*)Engine())->IsEnabled();
}
} // namespace mozilla