mirror of
https://gitlab.winehq.org/wine/wine-gecko.git
synced 2024-09-13 09:24:08 -07:00
b33bfca306
to remove any implication that the function might be for calculating the number of output channels for an AudioNode. --HG-- extra : transplant_source : %D9%12%BB%23n%92%EB%F7%1Df%11%3F%04%B4z%606%ADT%3A
554 lines
18 KiB
C++
554 lines
18 KiB
C++
/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*-*/
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/* This Source Code Form is subject to the terms of the Mozilla Public
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* License, v. 2.0. If a copy of the MPL was not distributed with this file,
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* You can obtain one at http://mozilla.org/MPL/2.0/. */
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#include "AudioNodeStream.h"
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#include "MediaStreamGraphImpl.h"
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#include "AudioNodeEngine.h"
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#include "ThreeDPoint.h"
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#include "AudioChannelFormat.h"
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#include "AudioParamTimeline.h"
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#include "AudioContext.h"
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using namespace mozilla::dom;
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namespace mozilla {
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/**
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* An AudioNodeStream produces a single audio track with ID
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* AUDIO_TRACK. This track has rate AudioContext::sIdealAudioRate
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* for regular audio contexts, and the rate requested by the web content
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* for offline audio contexts.
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* Each chunk in the track is a single block of WEBAUDIO_BLOCK_SIZE samples.
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* Note: This must be a different value than MEDIA_STREAM_DEST_TRACK_ID
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*/
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AudioNodeStream::~AudioNodeStream()
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{
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MOZ_COUNT_DTOR(AudioNodeStream);
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}
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void
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AudioNodeStream::SetStreamTimeParameter(uint32_t aIndex, AudioContext* aContext,
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double aStreamTime)
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{
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class Message : public ControlMessage {
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public:
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Message(AudioNodeStream* aStream, uint32_t aIndex, MediaStream* aRelativeToStream,
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double aStreamTime)
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: ControlMessage(aStream), mStreamTime(aStreamTime),
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mRelativeToStream(aRelativeToStream), mIndex(aIndex) {}
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virtual void Run()
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{
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static_cast<AudioNodeStream*>(mStream)->
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SetStreamTimeParameterImpl(mIndex, mRelativeToStream, mStreamTime);
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}
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double mStreamTime;
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MediaStream* mRelativeToStream;
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uint32_t mIndex;
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};
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MOZ_ASSERT(this);
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GraphImpl()->AppendMessage(new Message(this, aIndex,
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aContext->DestinationStream(),
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aContext->DOMTimeToStreamTime(aStreamTime)));
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}
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void
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AudioNodeStream::SetStreamTimeParameterImpl(uint32_t aIndex, MediaStream* aRelativeToStream,
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double aStreamTime)
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{
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TrackTicks ticks = TicksFromDestinationTime(aRelativeToStream, aStreamTime);
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mEngine->SetStreamTimeParameter(aIndex, ticks);
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}
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void
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AudioNodeStream::SetDoubleParameter(uint32_t aIndex, double aValue)
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{
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class Message : public ControlMessage {
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public:
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Message(AudioNodeStream* aStream, uint32_t aIndex, double aValue)
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: ControlMessage(aStream), mValue(aValue), mIndex(aIndex) {}
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virtual void Run()
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{
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static_cast<AudioNodeStream*>(mStream)->Engine()->
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SetDoubleParameter(mIndex, mValue);
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}
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double mValue;
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uint32_t mIndex;
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};
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MOZ_ASSERT(this);
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GraphImpl()->AppendMessage(new Message(this, aIndex, aValue));
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}
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void
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AudioNodeStream::SetInt32Parameter(uint32_t aIndex, int32_t aValue)
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{
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class Message : public ControlMessage {
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public:
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Message(AudioNodeStream* aStream, uint32_t aIndex, int32_t aValue)
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: ControlMessage(aStream), mValue(aValue), mIndex(aIndex) {}
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virtual void Run()
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{
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static_cast<AudioNodeStream*>(mStream)->Engine()->
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SetInt32Parameter(mIndex, mValue);
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}
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int32_t mValue;
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uint32_t mIndex;
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};
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MOZ_ASSERT(this);
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GraphImpl()->AppendMessage(new Message(this, aIndex, aValue));
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}
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void
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AudioNodeStream::SetTimelineParameter(uint32_t aIndex,
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const AudioParamTimeline& aValue)
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{
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class Message : public ControlMessage {
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public:
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Message(AudioNodeStream* aStream, uint32_t aIndex,
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const AudioParamTimeline& aValue)
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: ControlMessage(aStream),
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mValue(aValue),
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mSampleRate(aStream->SampleRate()),
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mIndex(aIndex) {}
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virtual void Run()
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{
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static_cast<AudioNodeStream*>(mStream)->Engine()->
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SetTimelineParameter(mIndex, mValue, mSampleRate);
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}
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AudioParamTimeline mValue;
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TrackRate mSampleRate;
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uint32_t mIndex;
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};
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GraphImpl()->AppendMessage(new Message(this, aIndex, aValue));
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}
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void
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AudioNodeStream::SetThreeDPointParameter(uint32_t aIndex, const ThreeDPoint& aValue)
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{
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class Message : public ControlMessage {
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public:
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Message(AudioNodeStream* aStream, uint32_t aIndex, const ThreeDPoint& aValue)
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: ControlMessage(aStream), mValue(aValue), mIndex(aIndex) {}
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virtual void Run()
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{
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static_cast<AudioNodeStream*>(mStream)->Engine()->
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SetThreeDPointParameter(mIndex, mValue);
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}
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ThreeDPoint mValue;
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uint32_t mIndex;
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};
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MOZ_ASSERT(this);
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GraphImpl()->AppendMessage(new Message(this, aIndex, aValue));
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}
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void
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AudioNodeStream::SetBuffer(already_AddRefed<ThreadSharedFloatArrayBufferList>&& aBuffer)
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{
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class Message : public ControlMessage {
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public:
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Message(AudioNodeStream* aStream,
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already_AddRefed<ThreadSharedFloatArrayBufferList>& aBuffer)
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: ControlMessage(aStream), mBuffer(aBuffer) {}
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virtual void Run()
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{
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static_cast<AudioNodeStream*>(mStream)->Engine()->
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SetBuffer(mBuffer.forget());
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}
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nsRefPtr<ThreadSharedFloatArrayBufferList> mBuffer;
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};
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MOZ_ASSERT(this);
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GraphImpl()->AppendMessage(new Message(this, aBuffer));
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}
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void
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AudioNodeStream::SetRawArrayData(nsTArray<float>& aData)
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{
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class Message : public ControlMessage {
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public:
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Message(AudioNodeStream* aStream,
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nsTArray<float>& aData)
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: ControlMessage(aStream)
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{
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mData.SwapElements(aData);
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}
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virtual void Run()
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{
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static_cast<AudioNodeStream*>(mStream)->Engine()->SetRawArrayData(mData);
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}
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nsTArray<float> mData;
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};
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MOZ_ASSERT(this);
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GraphImpl()->AppendMessage(new Message(this, aData));
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}
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void
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AudioNodeStream::SetChannelMixingParameters(uint32_t aNumberOfChannels,
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ChannelCountMode aChannelCountMode,
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ChannelInterpretation aChannelInterpretation)
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{
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class Message : public ControlMessage {
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public:
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Message(AudioNodeStream* aStream,
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uint32_t aNumberOfChannels,
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ChannelCountMode aChannelCountMode,
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ChannelInterpretation aChannelInterpretation)
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: ControlMessage(aStream),
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mNumberOfChannels(aNumberOfChannels),
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mChannelCountMode(aChannelCountMode),
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mChannelInterpretation(aChannelInterpretation)
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{}
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virtual void Run()
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{
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static_cast<AudioNodeStream*>(mStream)->
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SetChannelMixingParametersImpl(mNumberOfChannels, mChannelCountMode,
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mChannelInterpretation);
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}
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uint32_t mNumberOfChannels;
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ChannelCountMode mChannelCountMode;
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ChannelInterpretation mChannelInterpretation;
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};
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MOZ_ASSERT(this);
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GraphImpl()->AppendMessage(new Message(this, aNumberOfChannels,
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aChannelCountMode,
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aChannelInterpretation));
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}
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void
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AudioNodeStream::SetChannelMixingParametersImpl(uint32_t aNumberOfChannels,
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ChannelCountMode aChannelCountMode,
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ChannelInterpretation aChannelInterpretation)
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{
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// Make sure that we're not clobbering any significant bits by fitting these
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// values in 16 bits.
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MOZ_ASSERT(int(aChannelCountMode) < INT16_MAX);
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MOZ_ASSERT(int(aChannelInterpretation) < INT16_MAX);
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mNumberOfInputChannels = aNumberOfChannels;
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mChannelCountMode = aChannelCountMode;
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mChannelInterpretation = aChannelInterpretation;
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}
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uint32_t
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AudioNodeStream::ComputedNumberOfChannels(uint32_t aInputChannelCount)
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{
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switch (mChannelCountMode) {
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case ChannelCountMode::Explicit:
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// Disregard the channel count we've calculated from inputs, and just use
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// mNumberOfInputChannels.
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return mNumberOfInputChannels;
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case ChannelCountMode::Clamped_max:
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// Clamp the computed output channel count to mNumberOfInputChannels.
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return std::min(aInputChannelCount, mNumberOfInputChannels);
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default:
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case ChannelCountMode::Max:
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// Nothing to do here, just shut up the compiler warning.
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return aInputChannelCount;
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}
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}
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void
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AudioNodeStream::ObtainInputBlock(AudioChunk& aTmpChunk, uint32_t aPortIndex)
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{
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uint32_t inputCount = mInputs.Length();
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uint32_t outputChannelCount = 1;
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nsAutoTArray<AudioChunk*,250> inputChunks;
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for (uint32_t i = 0; i < inputCount; ++i) {
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if (aPortIndex != mInputs[i]->InputNumber()) {
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// This input is connected to a different port
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continue;
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}
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MediaStream* s = mInputs[i]->GetSource();
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AudioNodeStream* a = static_cast<AudioNodeStream*>(s);
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MOZ_ASSERT(a == s->AsAudioNodeStream());
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if (a->IsAudioParamStream()) {
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continue;
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}
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// It is possible for mLastChunks to be empty here, because `a` might be a
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// AudioNodeStream that has not been scheduled yet, because it is further
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// down the graph _but_ as a connection to this node. Because we enforce the
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// presence of at least one DelayNode, with at least one block of delay, and
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// because the output of a DelayNode when it has been fed less that
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// `delayTime` amount of audio is silence, we can simply continue here,
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// because this input would not influence the output of this node. Next
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// iteration, a->mLastChunks.IsEmpty() will be false, and everthing will
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// work as usual.
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if (a->mLastChunks.IsEmpty()) {
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continue;
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}
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AudioChunk* chunk = &a->mLastChunks[mInputs[i]->OutputNumber()];
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MOZ_ASSERT(chunk);
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if (chunk->IsNull() || chunk->mChannelData.IsEmpty()) {
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continue;
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}
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inputChunks.AppendElement(chunk);
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outputChannelCount =
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GetAudioChannelsSuperset(outputChannelCount, chunk->mChannelData.Length());
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}
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outputChannelCount = ComputedNumberOfChannels(outputChannelCount);
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uint32_t inputChunkCount = inputChunks.Length();
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if (inputChunkCount == 0 ||
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(inputChunkCount == 1 && inputChunks[0]->mChannelData.Length() == 0)) {
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aTmpChunk.SetNull(WEBAUDIO_BLOCK_SIZE);
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return;
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}
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if (inputChunkCount == 1 &&
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inputChunks[0]->mChannelData.Length() == outputChannelCount) {
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aTmpChunk = *inputChunks[0];
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return;
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}
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if (outputChannelCount == 0) {
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aTmpChunk.SetNull(WEBAUDIO_BLOCK_SIZE);
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return;
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}
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AllocateAudioBlock(outputChannelCount, &aTmpChunk);
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// The static storage here should be 1KB, so it's fine
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nsAutoTArray<float, GUESS_AUDIO_CHANNELS*WEBAUDIO_BLOCK_SIZE> downmixBuffer;
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for (uint32_t i = 0; i < inputChunkCount; ++i) {
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AccumulateInputChunk(i, *inputChunks[i], &aTmpChunk, &downmixBuffer);
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}
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}
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void
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AudioNodeStream::AccumulateInputChunk(uint32_t aInputIndex, const AudioChunk& aChunk,
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AudioChunk* aBlock,
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nsTArray<float>* aDownmixBuffer)
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{
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nsAutoTArray<const void*,GUESS_AUDIO_CHANNELS> channels;
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UpMixDownMixChunk(&aChunk, aBlock->mChannelData.Length(), channels, *aDownmixBuffer);
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for (uint32_t c = 0; c < channels.Length(); ++c) {
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const float* inputData = static_cast<const float*>(channels[c]);
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float* outputData = static_cast<float*>(const_cast<void*>(aBlock->mChannelData[c]));
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if (inputData) {
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if (aInputIndex == 0) {
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AudioBlockCopyChannelWithScale(inputData, aChunk.mVolume, outputData);
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} else {
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AudioBlockAddChannelWithScale(inputData, aChunk.mVolume, outputData);
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}
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} else {
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if (aInputIndex == 0) {
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PodZero(outputData, WEBAUDIO_BLOCK_SIZE);
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}
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}
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}
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}
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void
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AudioNodeStream::UpMixDownMixChunk(const AudioChunk* aChunk,
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uint32_t aOutputChannelCount,
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nsTArray<const void*>& aOutputChannels,
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nsTArray<float>& aDownmixBuffer)
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{
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static const float silenceChannel[WEBAUDIO_BLOCK_SIZE] = {0.f};
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aOutputChannels.AppendElements(aChunk->mChannelData);
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if (aOutputChannels.Length() < aOutputChannelCount) {
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if (mChannelInterpretation == ChannelInterpretation::Speakers) {
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AudioChannelsUpMix(&aOutputChannels, aOutputChannelCount, nullptr);
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NS_ASSERTION(aOutputChannelCount == aOutputChannels.Length(),
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"We called GetAudioChannelsSuperset to avoid this");
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} else {
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// Fill up the remaining aOutputChannels by zeros
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for (uint32_t j = aOutputChannels.Length(); j < aOutputChannelCount; ++j) {
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aOutputChannels.AppendElement(silenceChannel);
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}
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}
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} else if (aOutputChannels.Length() > aOutputChannelCount) {
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if (mChannelInterpretation == ChannelInterpretation::Speakers) {
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nsAutoTArray<float*,GUESS_AUDIO_CHANNELS> outputChannels;
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outputChannels.SetLength(aOutputChannelCount);
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aDownmixBuffer.SetLength(aOutputChannelCount * WEBAUDIO_BLOCK_SIZE);
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for (uint32_t j = 0; j < aOutputChannelCount; ++j) {
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outputChannels[j] = &aDownmixBuffer[j * WEBAUDIO_BLOCK_SIZE];
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}
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AudioChannelsDownMix(aOutputChannels, outputChannels.Elements(),
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aOutputChannelCount, WEBAUDIO_BLOCK_SIZE);
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aOutputChannels.SetLength(aOutputChannelCount);
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for (uint32_t j = 0; j < aOutputChannels.Length(); ++j) {
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aOutputChannels[j] = outputChannels[j];
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}
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} else {
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// Drop the remaining aOutputChannels
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aOutputChannels.RemoveElementsAt(aOutputChannelCount,
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aOutputChannels.Length() - aOutputChannelCount);
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}
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}
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}
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// The MediaStreamGraph guarantees that this is actually one block, for
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// AudioNodeStreams.
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void
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AudioNodeStream::ProcessInput(GraphTime aFrom, GraphTime aTo, uint32_t aFlags)
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{
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EnsureTrack(AUDIO_TRACK, mSampleRate);
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// No more tracks will be coming
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mBuffer.AdvanceKnownTracksTime(STREAM_TIME_MAX);
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uint16_t outputCount = std::max(uint16_t(1), mEngine->OutputCount());
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mLastChunks.SetLength(outputCount);
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// Consider this stream blocked if it has already finished output. Normally
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// mBlocked would reflect this, but due to rounding errors our audio track may
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// appear to extend slightly beyond aFrom, so we might not be blocked yet.
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bool blocked = mFinished || mBlocked.GetAt(aFrom);
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// If the stream has finished at this time, it will be blocked.
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if (mMuted || blocked) {
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for (uint16_t i = 0; i < outputCount; ++i) {
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mLastChunks[i].SetNull(WEBAUDIO_BLOCK_SIZE);
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}
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} else {
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// We need to generate at least one input
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uint16_t maxInputs = std::max(uint16_t(1), mEngine->InputCount());
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OutputChunks inputChunks;
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inputChunks.SetLength(maxInputs);
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for (uint16_t i = 0; i < maxInputs; ++i) {
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ObtainInputBlock(inputChunks[i], i);
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}
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bool finished = false;
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if (maxInputs <= 1 && mEngine->OutputCount() <= 1) {
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mEngine->ProcessBlock(this, inputChunks[0], &mLastChunks[0], &finished);
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} else {
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mEngine->ProcessBlocksOnPorts(this, inputChunks, mLastChunks, &finished);
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}
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for (uint16_t i = 0; i < outputCount; ++i) {
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NS_ASSERTION(mLastChunks[i].GetDuration() == WEBAUDIO_BLOCK_SIZE,
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"Invalid WebAudio chunk size");
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}
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if (finished) {
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mMarkAsFinishedAfterThisBlock = true;
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}
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if (mDisabledTrackIDs.Contains(static_cast<TrackID>(AUDIO_TRACK))) {
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for (uint32_t i = 0; i < outputCount; ++i) {
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mLastChunks[i].SetNull(WEBAUDIO_BLOCK_SIZE);
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}
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}
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}
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if (!blocked) {
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// Don't output anything while blocked
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AdvanceOutputSegment();
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if (mMarkAsFinishedAfterThisBlock && (aFlags & ALLOW_FINISH)) {
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// This stream was finished the last time that we looked at it, and all
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// of the depending streams have finished their output as well, so now
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// it's time to mark this stream as finished.
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FinishOutput();
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}
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}
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}
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void
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AudioNodeStream::AdvanceOutputSegment()
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{
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StreamBuffer::Track* track = EnsureTrack(AUDIO_TRACK, mSampleRate);
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AudioSegment* segment = track->Get<AudioSegment>();
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if (mKind == MediaStreamGraph::EXTERNAL_STREAM) {
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segment->AppendAndConsumeChunk(&mLastChunks[0]);
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} else {
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segment->AppendNullData(mLastChunks[0].GetDuration());
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}
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for (uint32_t j = 0; j < mListeners.Length(); ++j) {
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MediaStreamListener* l = mListeners[j];
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AudioChunk copyChunk = mLastChunks[0];
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AudioSegment tmpSegment;
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tmpSegment.AppendAndConsumeChunk(©Chunk);
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l->NotifyQueuedTrackChanges(Graph(), AUDIO_TRACK,
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mSampleRate, segment->GetDuration(), 0,
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tmpSegment);
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}
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}
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TrackTicks
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AudioNodeStream::GetCurrentPosition()
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{
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return EnsureTrack(AUDIO_TRACK, mSampleRate)->Get<AudioSegment>()->GetDuration();
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}
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void
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AudioNodeStream::FinishOutput()
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{
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if (IsFinishedOnGraphThread()) {
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return;
|
|
}
|
|
|
|
StreamBuffer::Track* track = EnsureTrack(AUDIO_TRACK, mSampleRate);
|
|
track->SetEnded();
|
|
FinishOnGraphThread();
|
|
|
|
for (uint32_t j = 0; j < mListeners.Length(); ++j) {
|
|
MediaStreamListener* l = mListeners[j];
|
|
AudioSegment emptySegment;
|
|
l->NotifyQueuedTrackChanges(Graph(), AUDIO_TRACK,
|
|
mSampleRate,
|
|
track->GetSegment()->GetDuration(),
|
|
MediaStreamListener::TRACK_EVENT_ENDED, emptySegment);
|
|
}
|
|
}
|
|
|
|
double
|
|
AudioNodeStream::TimeFromDestinationTime(AudioNodeStream* aDestination,
|
|
double aSeconds)
|
|
{
|
|
MOZ_ASSERT(aDestination->SampleRate() == SampleRate());
|
|
|
|
double destinationSeconds = std::max(0.0, aSeconds);
|
|
StreamTime streamTime = SecondsToMediaTime(destinationSeconds);
|
|
// MediaTime does not have the resolution of double
|
|
double offset = destinationSeconds - MediaTimeToSeconds(streamTime);
|
|
|
|
GraphTime graphTime = aDestination->StreamTimeToGraphTime(streamTime);
|
|
StreamTime thisStreamTime = GraphTimeToStreamTimeOptimistic(graphTime);
|
|
double thisSeconds = MediaTimeToSeconds(thisStreamTime) + offset;
|
|
MOZ_ASSERT(thisSeconds >= 0.0);
|
|
return thisSeconds;
|
|
}
|
|
|
|
TrackTicks
|
|
AudioNodeStream::TicksFromDestinationTime(MediaStream* aDestination,
|
|
double aSeconds)
|
|
{
|
|
AudioNodeStream* destination = aDestination->AsAudioNodeStream();
|
|
MOZ_ASSERT(destination);
|
|
|
|
double thisSeconds = TimeFromDestinationTime(destination, aSeconds);
|
|
// Round to nearest
|
|
TrackTicks ticks = thisSeconds * SampleRate() + 0.5;
|
|
return ticks;
|
|
}
|
|
|
|
double
|
|
AudioNodeStream::DestinationTimeFromTicks(AudioNodeStream* aDestination,
|
|
TrackTicks aPosition)
|
|
{
|
|
MOZ_ASSERT(SampleRate() == aDestination->SampleRate());
|
|
StreamTime sourceTime = TicksToTimeRoundDown(SampleRate(), aPosition);
|
|
GraphTime graphTime = StreamTimeToGraphTime(sourceTime);
|
|
StreamTime destinationTime = aDestination->GraphTimeToStreamTimeOptimistic(graphTime);
|
|
return MediaTimeToSeconds(destinationTime);
|
|
}
|
|
|
|
}
|