mirror of
https://github.com/classilla/tenfourfox.git
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841 lines
29 KiB
C++
841 lines
29 KiB
C++
/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
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/* vim:set ts=2 sw=2 sts=2 et cindent: */
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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
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* file, You can obtain one at http://mozilla.org/MPL/2.0/. */
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#include "AudioBufferSourceNode.h"
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#include "mozilla/dom/AudioBufferSourceNodeBinding.h"
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#include "mozilla/dom/AudioParam.h"
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#include "mozilla/FloatingPoint.h"
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#include "nsContentUtils.h"
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#include "nsMathUtils.h"
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#include "AudioNodeEngine.h"
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#include "AudioNodeStream.h"
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#include "AudioDestinationNode.h"
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#include "AudioParamTimeline.h"
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#include <limits>
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#include <algorithm>
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namespace mozilla {
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namespace dom {
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NS_IMPL_CYCLE_COLLECTION_INHERITED(AudioBufferSourceNode, AudioNode, mBuffer, mPlaybackRate, mDetune)
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NS_INTERFACE_MAP_BEGIN_CYCLE_COLLECTION_INHERITED(AudioBufferSourceNode)
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NS_INTERFACE_MAP_END_INHERITING(AudioNode)
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NS_IMPL_ADDREF_INHERITED(AudioBufferSourceNode, AudioNode)
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NS_IMPL_RELEASE_INHERITED(AudioBufferSourceNode, AudioNode)
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/**
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* Media-thread playback engine for AudioBufferSourceNode.
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* Nothing is played until a non-null buffer has been set (via
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* AudioNodeStream::SetBuffer) and a non-zero mBufferEnd has been set (via
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* AudioNodeStream::SetInt32Parameter).
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*/
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class AudioBufferSourceNodeEngine final : public AudioNodeEngine
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{
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public:
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AudioBufferSourceNodeEngine(AudioNode* aNode,
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AudioDestinationNode* aDestination) :
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AudioNodeEngine(aNode),
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mStart(0.0), mBeginProcessing(0),
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mStop(STREAM_TIME_MAX),
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mResampler(nullptr), mRemainingResamplerTail(0),
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mBufferEnd(0),
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mLoopStart(0), mLoopEnd(0),
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mBufferPosition(0), mBufferSampleRate(0),
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// mResamplerOutRate is initialized in UpdateResampler().
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mChannels(0),
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mDopplerShift(1.0f),
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mDestination(aDestination->Stream()),
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mPlaybackRateTimeline(1.0f),
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mDetuneTimeline(0.0f),
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mLoop(false)
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{}
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~AudioBufferSourceNodeEngine()
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{
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if (mResampler) {
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speex_resampler_destroy(mResampler);
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}
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}
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void SetSourceStream(AudioNodeStream* aSource)
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{
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mSource = aSource;
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}
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virtual void RecvTimelineEvent(uint32_t aIndex,
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dom::AudioTimelineEvent& aEvent) override
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{
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MOZ_ASSERT(mDestination);
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WebAudioUtils::ConvertAudioTimelineEventToTicks(aEvent,
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mDestination);
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switch (aIndex) {
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case AudioBufferSourceNode::PLAYBACKRATE:
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mPlaybackRateTimeline.InsertEvent<int64_t>(aEvent);
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break;
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case AudioBufferSourceNode::DETUNE:
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mDetuneTimeline.InsertEvent<int64_t>(aEvent);
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break;
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default:
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NS_ERROR("Bad AudioBufferSourceNodeEngine TimelineParameter");
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}
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}
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virtual void SetStreamTimeParameter(uint32_t aIndex, StreamTime aParam) override
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{
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switch (aIndex) {
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case AudioBufferSourceNode::STOP: mStop = aParam; break;
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default:
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NS_ERROR("Bad AudioBufferSourceNodeEngine StreamTimeParameter");
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}
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}
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virtual void SetDoubleParameter(uint32_t aIndex, double aParam) override
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{
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switch (aIndex) {
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case AudioBufferSourceNode::START:
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MOZ_ASSERT(!mStart, "Another START?");
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mStart = aParam * mDestination->SampleRate();
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// Round to nearest
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mBeginProcessing = mStart + 0.5;
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break;
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case AudioBufferSourceNode::DOPPLERSHIFT:
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mDopplerShift = (aParam <= 0 || mozilla::IsNaN(aParam)) ? 1.0 : aParam;
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break;
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default:
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NS_ERROR("Bad AudioBufferSourceNodeEngine double parameter.");
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};
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}
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virtual void SetInt32Parameter(uint32_t aIndex, int32_t aParam) override
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{
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switch (aIndex) {
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case AudioBufferSourceNode::SAMPLE_RATE:
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MOZ_ASSERT(aParam > 0);
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mBufferSampleRate = aParam;
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mSource->SetActive();
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break;
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case AudioBufferSourceNode::BUFFERSTART:
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MOZ_ASSERT(aParam >= 0);
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if (mBufferPosition == 0) {
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mBufferPosition = aParam;
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}
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break;
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case AudioBufferSourceNode::BUFFEREND:
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MOZ_ASSERT(aParam >= 0);
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mBufferEnd = aParam;
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break;
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case AudioBufferSourceNode::LOOP: mLoop = !!aParam; break;
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case AudioBufferSourceNode::LOOPSTART:
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MOZ_ASSERT(aParam >= 0);
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mLoopStart = aParam;
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break;
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case AudioBufferSourceNode::LOOPEND:
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MOZ_ASSERT(aParam >= 0);
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mLoopEnd = aParam;
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break;
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default:
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NS_ERROR("Bad AudioBufferSourceNodeEngine Int32Parameter");
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}
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}
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virtual void SetBuffer(already_AddRefed<ThreadSharedFloatArrayBufferList> aBuffer) override
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{
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mBuffer = aBuffer;
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}
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bool BegunResampling()
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{
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return mBeginProcessing == -STREAM_TIME_MAX;
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}
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void UpdateResampler(int32_t aOutRate, uint32_t aChannels)
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{
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if (mResampler &&
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(aChannels != mChannels ||
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// If the resampler has begun, then it will have moved
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// mBufferPosition to after the samples it has read, but it hasn't
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// output its buffered samples. Keep using the resampler, even if
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// the rates now match, so that this latent segment is output.
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(aOutRate == mBufferSampleRate && !BegunResampling()))) {
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speex_resampler_destroy(mResampler);
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mResampler = nullptr;
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mRemainingResamplerTail = 0;
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mBeginProcessing = mStart + 0.5;
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}
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if (aChannels == 0 ||
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(aOutRate == mBufferSampleRate && !mResampler)) {
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mResamplerOutRate = aOutRate;
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return;
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}
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if (!mResampler) {
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mChannels = aChannels;
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mResampler = speex_resampler_init(mChannels, mBufferSampleRate, aOutRate,
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SPEEX_RESAMPLER_QUALITY_MIN,
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nullptr);
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} else {
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if (mResamplerOutRate == aOutRate) {
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return;
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}
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speex_resampler_set_rate(mResampler, mBufferSampleRate, aOutRate);
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}
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mResamplerOutRate = aOutRate;
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if (!BegunResampling()) {
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// Low pass filter effects from the resampler mean that samples before
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// the start time are influenced by resampling the buffer. The input
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// latency indicates half the filter width.
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int64_t inputLatency = speex_resampler_get_input_latency(mResampler);
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uint32_t ratioNum, ratioDen;
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speex_resampler_get_ratio(mResampler, &ratioNum, &ratioDen);
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// The output subsample resolution supported in aligning the resampler
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// is ratioNum. First round the start time to the nearest subsample.
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int64_t subsample = mStart * ratioNum + 0.5;
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// Now include the leading effects of the filter, and round *up* to the
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// next whole tick, because there is no effect on samples outside the
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// filter width.
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mBeginProcessing =
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(subsample - inputLatency * ratioDen + ratioNum - 1) / ratioNum;
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}
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}
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// Borrow a full buffer of size WEBAUDIO_BLOCK_SIZE from the source buffer
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// at offset aSourceOffset. This avoids copying memory.
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void BorrowFromInputBuffer(AudioBlock* aOutput,
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uint32_t aChannels)
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{
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aOutput->SetBuffer(mBuffer);
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aOutput->mChannelData.SetLength(aChannels);
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for (uint32_t i = 0; i < aChannels; ++i) {
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aOutput->mChannelData[i] = mBuffer->GetData(i) + mBufferPosition;
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}
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aOutput->mVolume = 1.0f;
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aOutput->mBufferFormat = AUDIO_FORMAT_FLOAT32;
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}
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// Copy aNumberOfFrames frames from the source buffer at offset aSourceOffset
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// and put it at offset aBufferOffset in the destination buffer.
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void CopyFromInputBuffer(AudioBlock* aOutput,
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uint32_t aChannels,
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uintptr_t aOffsetWithinBlock,
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uint32_t aNumberOfFrames) {
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for (uint32_t i = 0; i < aChannels; ++i) {
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float* baseChannelData = aOutput->ChannelFloatsForWrite(i);
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memcpy(baseChannelData + aOffsetWithinBlock,
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mBuffer->GetData(i) + mBufferPosition,
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aNumberOfFrames * sizeof(float));
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}
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}
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// Resamples input data to an output buffer, according to |mBufferSampleRate| and
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// the playbackRate/detune.
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// The number of frames consumed/produced depends on the amount of space
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// remaining in both the input and output buffer, and the playback rate (that
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// is, the ratio between the output samplerate and the input samplerate).
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void CopyFromInputBufferWithResampling(AudioBlock* aOutput,
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uint32_t aChannels,
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uint32_t* aOffsetWithinBlock,
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uint32_t aAvailableInOutput,
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StreamTime* aCurrentPosition,
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uint32_t aBufferMax)
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{
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if (*aOffsetWithinBlock == 0) {
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aOutput->AllocateChannels(aChannels);
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}
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SpeexResamplerState* resampler = mResampler;
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MOZ_ASSERT(aChannels > 0);
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if (mBufferPosition < aBufferMax) {
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uint32_t availableInInputBuffer = aBufferMax - mBufferPosition;
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uint32_t ratioNum, ratioDen;
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speex_resampler_get_ratio(resampler, &ratioNum, &ratioDen);
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// Limit the number of input samples copied and possibly
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// format-converted for resampling by estimating how many will be used.
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// This may be a little small if still filling the resampler with
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// initial data, but we'll get called again and it will work out.
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uint32_t inputLimit = aAvailableInOutput * ratioNum / ratioDen + 10;
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if (!BegunResampling()) {
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// First time the resampler is used.
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uint32_t inputLatency = speex_resampler_get_input_latency(resampler);
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inputLimit += inputLatency;
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// If starting after mStart, then play from the beginning of the
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// buffer, but correct for input latency. If starting before mStart,
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// then align the resampler so that the time corresponding to the
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// first input sample is mStart.
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int64_t skipFracNum = static_cast<int64_t>(inputLatency) * ratioDen;
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double leadTicks = mStart - *aCurrentPosition;
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if (leadTicks > 0.0) {
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// Round to nearest output subsample supported by the resampler at
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// these rates.
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int64_t leadSubsamples = leadTicks * ratioNum + 0.5;
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MOZ_ASSERT(leadSubsamples <= skipFracNum,
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"mBeginProcessing is wrong?");
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skipFracNum -= leadSubsamples;
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}
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speex_resampler_set_skip_frac_num(resampler,
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std::min<int64_t>(skipFracNum, UINT32_MAX));
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mBeginProcessing = -STREAM_TIME_MAX;
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}
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inputLimit = std::min(inputLimit, availableInInputBuffer);
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for (uint32_t i = 0; true; ) {
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uint32_t inSamples = inputLimit;
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const float* inputData = mBuffer->GetData(i) + mBufferPosition;
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uint32_t outSamples = aAvailableInOutput;
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float* outputData =
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aOutput->ChannelFloatsForWrite(i) + *aOffsetWithinBlock;
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WebAudioUtils::SpeexResamplerProcess(resampler, i,
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inputData, &inSamples,
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outputData, &outSamples);
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if (++i == aChannels) {
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mBufferPosition += inSamples;
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MOZ_ASSERT(mBufferPosition <= mBufferEnd || mLoop);
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*aOffsetWithinBlock += outSamples;
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*aCurrentPosition += outSamples;
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if (inSamples == availableInInputBuffer && !mLoop) {
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// We'll feed in enough zeros to empty out the resampler's memory.
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// This handles the output latency as well as capturing the low
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// pass effects of the resample filter.
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mRemainingResamplerTail =
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2 * speex_resampler_get_input_latency(resampler) - 1;
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}
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return;
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}
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}
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} else {
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for (uint32_t i = 0; true; ) {
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uint32_t inSamples = mRemainingResamplerTail;
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uint32_t outSamples = aAvailableInOutput;
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float* outputData =
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aOutput->ChannelFloatsForWrite(i) + *aOffsetWithinBlock;
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// AudioDataValue* for aIn selects the function that does not try to
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// copy and format-convert input data.
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WebAudioUtils::SpeexResamplerProcess(resampler, i,
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static_cast<AudioDataValue*>(nullptr), &inSamples,
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outputData, &outSamples);
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if (++i == aChannels) {
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MOZ_ASSERT(inSamples <= mRemainingResamplerTail);
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mRemainingResamplerTail -= inSamples;
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*aOffsetWithinBlock += outSamples;
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*aCurrentPosition += outSamples;
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break;
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}
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}
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}
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}
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/**
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* Fill aOutput with as many zero frames as we can, and advance
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* aOffsetWithinBlock and aCurrentPosition based on how many frames we write.
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* This will never advance aOffsetWithinBlock past WEBAUDIO_BLOCK_SIZE or
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* aCurrentPosition past aMaxPos. This function knows when it needs to
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* allocate the output buffer, and also optimizes the case where it can avoid
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* memory allocations.
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*/
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void FillWithZeroes(AudioBlock* aOutput,
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uint32_t aChannels,
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uint32_t* aOffsetWithinBlock,
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StreamTime* aCurrentPosition,
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StreamTime aMaxPos)
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{
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MOZ_ASSERT(*aCurrentPosition < aMaxPos);
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uint32_t numFrames =
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std::min<StreamTime>(WEBAUDIO_BLOCK_SIZE - *aOffsetWithinBlock,
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aMaxPos - *aCurrentPosition);
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if (numFrames == WEBAUDIO_BLOCK_SIZE || !aChannels) {
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aOutput->SetNull(numFrames);
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} else {
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if (*aOffsetWithinBlock == 0) {
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aOutput->AllocateChannels(aChannels);
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}
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WriteZeroesToAudioBlock(aOutput, *aOffsetWithinBlock, numFrames);
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}
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*aOffsetWithinBlock += numFrames;
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*aCurrentPosition += numFrames;
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}
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/**
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* Copy as many frames as possible from the source buffer to aOutput, and
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* advance aOffsetWithinBlock and aCurrentPosition based on how many frames
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* we write. This will never advance aOffsetWithinBlock past
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* WEBAUDIO_BLOCK_SIZE, or aCurrentPosition past mStop. It takes data from
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* the buffer at aBufferOffset, and never takes more data than aBufferMax.
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* This function knows when it needs to allocate the output buffer, and also
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* optimizes the case where it can avoid memory allocations.
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*/
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void CopyFromBuffer(AudioBlock* aOutput,
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uint32_t aChannels,
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uint32_t* aOffsetWithinBlock,
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StreamTime* aCurrentPosition,
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uint32_t aBufferMax)
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{
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MOZ_ASSERT(*aCurrentPosition < mStop);
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uint32_t availableInOutput =
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std::min<StreamTime>(WEBAUDIO_BLOCK_SIZE - *aOffsetWithinBlock,
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mStop - *aCurrentPosition);
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if (mResampler) {
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CopyFromInputBufferWithResampling(aOutput, aChannels,
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aOffsetWithinBlock, availableInOutput,
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aCurrentPosition, aBufferMax);
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return;
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}
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if (aChannels == 0) {
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aOutput->SetNull(WEBAUDIO_BLOCK_SIZE);
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// There is no attempt here to limit advance so that mBufferPosition is
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// limited to aBufferMax. The only observable affect of skipping the
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// check would be in the precise timing of the ended event if the loop
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// attribute is reset after playback has looped.
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*aOffsetWithinBlock += availableInOutput;
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*aCurrentPosition += availableInOutput;
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// Rounding at the start and end of the period means that fractional
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// increments essentially accumulate if outRate remains constant. If
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// outRate is varying, then accumulation happens on average but not
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// precisely.
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TrackTicks start = *aCurrentPosition *
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mBufferSampleRate / mResamplerOutRate;
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TrackTicks end = (*aCurrentPosition + availableInOutput) *
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mBufferSampleRate / mResamplerOutRate;
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mBufferPosition += end - start;
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return;
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}
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uint32_t numFrames = std::min(aBufferMax - mBufferPosition,
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availableInOutput);
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if (numFrames == WEBAUDIO_BLOCK_SIZE) {
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MOZ_ASSERT(mBufferPosition < aBufferMax);
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BorrowFromInputBuffer(aOutput, aChannels);
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} else {
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if (*aOffsetWithinBlock == 0) {
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aOutput->AllocateChannels(aChannels);
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}
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MOZ_ASSERT(mBufferPosition < aBufferMax);
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CopyFromInputBuffer(aOutput, aChannels, *aOffsetWithinBlock, numFrames);
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}
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*aOffsetWithinBlock += numFrames;
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*aCurrentPosition += numFrames;
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mBufferPosition += numFrames;
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}
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int32_t ComputeFinalOutSampleRate(float aPlaybackRate, float aDetune)
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{
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float computedPlaybackRate = aPlaybackRate * pow(2, aDetune / 1200.f);
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// Make sure the playback rate and the doppler shift are something
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// our resampler can work with.
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int32_t rate = WebAudioUtils::
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TruncateFloatToInt<int32_t>(mSource->SampleRate() /
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(computedPlaybackRate * mDopplerShift));
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return rate ? rate : mBufferSampleRate;
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}
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void UpdateSampleRateIfNeeded(uint32_t aChannels, StreamTime aStreamPosition)
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{
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float playbackRate;
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float detune;
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if (mPlaybackRateTimeline.HasSimpleValue()) {
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playbackRate = mPlaybackRateTimeline.GetValue();
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} else {
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playbackRate = mPlaybackRateTimeline.GetValueAtTime(aStreamPosition);
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}
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if (mDetuneTimeline.HasSimpleValue()) {
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detune = mDetuneTimeline.GetValue();
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} else {
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detune = mDetuneTimeline.GetValueAtTime(aStreamPosition);
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}
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if (playbackRate <= 0 || mozilla::IsNaN(playbackRate)) {
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playbackRate = 1.0f;
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}
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detune = std::min(std::max(-1200.f, detune), 1200.f);
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int32_t outRate = ComputeFinalOutSampleRate(playbackRate, detune);
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UpdateResampler(outRate, aChannels);
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}
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virtual void ProcessBlock(AudioNodeStream* aStream,
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GraphTime aFrom,
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const AudioBlock& aInput,
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AudioBlock* aOutput,
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bool* aFinished) override
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{
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if (mBufferSampleRate == 0) {
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// start() has not yet been called or no buffer has yet been set
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|
aOutput->SetNull(WEBAUDIO_BLOCK_SIZE);
|
|
return;
|
|
}
|
|
|
|
StreamTime streamPosition = mDestination->GraphTimeToStreamTime(aFrom);
|
|
uint32_t channels = mBuffer ? mBuffer->GetChannels() : 0;
|
|
|
|
UpdateSampleRateIfNeeded(channels, streamPosition);
|
|
|
|
uint32_t written = 0;
|
|
while (written < WEBAUDIO_BLOCK_SIZE) {
|
|
if (mStop != STREAM_TIME_MAX &&
|
|
streamPosition >= mStop) {
|
|
FillWithZeroes(aOutput, channels, &written, &streamPosition, STREAM_TIME_MAX);
|
|
continue;
|
|
}
|
|
if (streamPosition < mBeginProcessing) {
|
|
FillWithZeroes(aOutput, channels, &written, &streamPosition,
|
|
mBeginProcessing);
|
|
continue;
|
|
}
|
|
if (mLoop) {
|
|
// mLoopEnd can become less than mBufferPosition when a LOOPEND engine
|
|
// parameter is received after "loopend" is changed on the node or a
|
|
// new buffer with lower samplerate is set.
|
|
if (mBufferPosition >= mLoopEnd) {
|
|
mBufferPosition = mLoopStart;
|
|
}
|
|
CopyFromBuffer(aOutput, channels, &written, &streamPosition, mLoopEnd);
|
|
} else {
|
|
if (mBufferPosition < mBufferEnd || mRemainingResamplerTail) {
|
|
CopyFromBuffer(aOutput, channels, &written, &streamPosition, mBufferEnd);
|
|
} else {
|
|
FillWithZeroes(aOutput, channels, &written, &streamPosition, STREAM_TIME_MAX);
|
|
}
|
|
}
|
|
}
|
|
|
|
// We've finished if we've gone past mStop, or if we're past mDuration when
|
|
// looping is disabled.
|
|
if (streamPosition >= mStop ||
|
|
(!mLoop && mBufferPosition >= mBufferEnd && !mRemainingResamplerTail)) {
|
|
*aFinished = true;
|
|
}
|
|
}
|
|
|
|
virtual bool IsActive() const override
|
|
{
|
|
// Whether buffer has been set and start() has been called.
|
|
return mBufferSampleRate != 0;
|
|
}
|
|
|
|
virtual size_t SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const override
|
|
{
|
|
// Not owned:
|
|
// - mBuffer - shared w/ AudioNode
|
|
// - mPlaybackRateTimeline - shared w/ AudioNode
|
|
// - mDetuneTimeline - shared w/ AudioNode
|
|
|
|
size_t amount = AudioNodeEngine::SizeOfExcludingThis(aMallocSizeOf);
|
|
|
|
// NB: We need to modify speex if we want the full memory picture, internal
|
|
// fields that need measuring noted below.
|
|
// - mResampler->mem
|
|
// - mResampler->sinc_table
|
|
// - mResampler->last_sample
|
|
// - mResampler->magic_samples
|
|
// - mResampler->samp_frac_num
|
|
amount += aMallocSizeOf(mResampler);
|
|
|
|
return amount;
|
|
}
|
|
|
|
virtual size_t SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const override
|
|
{
|
|
return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
|
|
}
|
|
|
|
double mStart; // including the fractional position between ticks
|
|
// Low pass filter effects from the resampler mean that samples before the
|
|
// start time are influenced by resampling the buffer. mBeginProcessing
|
|
// includes the extent of this filter. The special value of -STREAM_TIME_MAX
|
|
// indicates that the resampler has begun processing.
|
|
StreamTime mBeginProcessing;
|
|
StreamTime mStop;
|
|
RefPtr<ThreadSharedFloatArrayBufferList> mBuffer;
|
|
SpeexResamplerState* mResampler;
|
|
// mRemainingResamplerTail, like mBufferPosition, and
|
|
// mBufferEnd, is measured in input buffer samples.
|
|
uint32_t mRemainingResamplerTail;
|
|
uint32_t mBufferEnd;
|
|
uint32_t mLoopStart;
|
|
uint32_t mLoopEnd;
|
|
uint32_t mBufferPosition;
|
|
int32_t mBufferSampleRate;
|
|
int32_t mResamplerOutRate;
|
|
uint32_t mChannels;
|
|
float mDopplerShift;
|
|
AudioNodeStream* mDestination;
|
|
AudioNodeStream* mSource;
|
|
AudioParamTimeline mPlaybackRateTimeline;
|
|
AudioParamTimeline mDetuneTimeline;
|
|
bool mLoop;
|
|
};
|
|
|
|
AudioBufferSourceNode::AudioBufferSourceNode(AudioContext* aContext)
|
|
: AudioNode(aContext,
|
|
2,
|
|
ChannelCountMode::Max,
|
|
ChannelInterpretation::Speakers)
|
|
, mLoopStart(0.0)
|
|
, mLoopEnd(0.0)
|
|
// mOffset and mDuration are initialized in Start().
|
|
, mPlaybackRate(new AudioParam(this, PLAYBACKRATE, 1.0f, "playbackRate"))
|
|
, mDetune(new AudioParam(this, DETUNE, 0.0f, "detune"))
|
|
, mLoop(false)
|
|
, mStartCalled(false)
|
|
{
|
|
AudioBufferSourceNodeEngine* engine = new AudioBufferSourceNodeEngine(this, aContext->Destination());
|
|
mStream = AudioNodeStream::Create(aContext, engine,
|
|
AudioNodeStream::NEED_MAIN_THREAD_FINISHED);
|
|
engine->SetSourceStream(mStream);
|
|
mStream->AddMainThreadListener(this);
|
|
}
|
|
|
|
AudioBufferSourceNode::~AudioBufferSourceNode()
|
|
{
|
|
}
|
|
|
|
void
|
|
AudioBufferSourceNode::DestroyMediaStream()
|
|
{
|
|
bool hadStream = mStream;
|
|
if (hadStream) {
|
|
mStream->RemoveMainThreadListener(this);
|
|
}
|
|
AudioNode::DestroyMediaStream();
|
|
if (hadStream && Context()) {
|
|
Context()->UnregisterAudioBufferSourceNode(this);
|
|
}
|
|
}
|
|
|
|
size_t
|
|
AudioBufferSourceNode::SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const
|
|
{
|
|
size_t amount = AudioNode::SizeOfExcludingThis(aMallocSizeOf);
|
|
if (mBuffer) {
|
|
amount += mBuffer->SizeOfIncludingThis(aMallocSizeOf);
|
|
}
|
|
|
|
amount += mPlaybackRate->SizeOfIncludingThis(aMallocSizeOf);
|
|
amount += mDetune->SizeOfIncludingThis(aMallocSizeOf);
|
|
return amount;
|
|
}
|
|
|
|
size_t
|
|
AudioBufferSourceNode::SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const
|
|
{
|
|
return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
|
|
}
|
|
|
|
JSObject*
|
|
AudioBufferSourceNode::WrapObject(JSContext* aCx, JS::Handle<JSObject*> aGivenProto)
|
|
{
|
|
return AudioBufferSourceNodeBinding::Wrap(aCx, this, aGivenProto);
|
|
}
|
|
|
|
void
|
|
AudioBufferSourceNode::Start(double aWhen, double aOffset,
|
|
const Optional<double>& aDuration, ErrorResult& aRv)
|
|
{
|
|
if (!WebAudioUtils::IsTimeValid(aWhen) ||
|
|
(aDuration.WasPassed() && !WebAudioUtils::IsTimeValid(aDuration.Value()))) {
|
|
aRv.Throw(NS_ERROR_DOM_NOT_SUPPORTED_ERR);
|
|
return;
|
|
}
|
|
|
|
if (mStartCalled) {
|
|
aRv.Throw(NS_ERROR_DOM_INVALID_STATE_ERR);
|
|
return;
|
|
}
|
|
mStartCalled = true;
|
|
|
|
AudioNodeStream* ns = mStream;
|
|
if (!ns) {
|
|
// Nothing to play, or we're already dead for some reason
|
|
return;
|
|
}
|
|
|
|
// Remember our arguments so that we can use them when we get a new buffer.
|
|
mOffset = aOffset;
|
|
mDuration = aDuration.WasPassed() ? aDuration.Value()
|
|
: std::numeric_limits<double>::min();
|
|
|
|
WEB_AUDIO_API_LOG("%f: %s %u Start(%f, %g, %g)", Context()->CurrentTime(),
|
|
NodeType(), Id(), aWhen, aOffset, mDuration);
|
|
|
|
// We can't send these parameters without a buffer because we don't know the
|
|
// buffer's sample rate or length.
|
|
if (mBuffer) {
|
|
SendOffsetAndDurationParametersToStream(ns);
|
|
}
|
|
|
|
// Don't set parameter unnecessarily
|
|
if (aWhen > 0.0) {
|
|
ns->SetDoubleParameter(START, aWhen);
|
|
}
|
|
}
|
|
|
|
void
|
|
AudioBufferSourceNode::SendBufferParameterToStream(JSContext* aCx)
|
|
{
|
|
AudioNodeStream* ns = mStream;
|
|
if (!ns) {
|
|
return;
|
|
}
|
|
|
|
if (mBuffer) {
|
|
RefPtr<ThreadSharedFloatArrayBufferList> data =
|
|
mBuffer->GetThreadSharedChannelsForRate(aCx);
|
|
ns->SetBuffer(data.forget());
|
|
|
|
if (mStartCalled) {
|
|
SendOffsetAndDurationParametersToStream(ns);
|
|
}
|
|
} else {
|
|
ns->SetInt32Parameter(BUFFEREND, 0);
|
|
ns->SetBuffer(nullptr);
|
|
|
|
MarkInactive();
|
|
}
|
|
}
|
|
|
|
void
|
|
AudioBufferSourceNode::SendOffsetAndDurationParametersToStream(AudioNodeStream* aStream)
|
|
{
|
|
NS_ASSERTION(mBuffer && mStartCalled,
|
|
"Only call this when we have a buffer and start() has been called");
|
|
|
|
float rate = mBuffer->SampleRate();
|
|
aStream->SetInt32Parameter(SAMPLE_RATE, rate);
|
|
|
|
int32_t bufferEnd = mBuffer->Length();
|
|
int32_t offsetSamples = std::max(0, NS_lround(mOffset * rate));
|
|
|
|
// Don't set parameter unnecessarily
|
|
if (offsetSamples > 0) {
|
|
aStream->SetInt32Parameter(BUFFERSTART, offsetSamples);
|
|
}
|
|
|
|
if (mDuration != std::numeric_limits<double>::min()) {
|
|
MOZ_ASSERT(mDuration >= 0.0); // provided by Start()
|
|
MOZ_ASSERT(rate >= 0.0f); // provided by AudioBuffer::Create()
|
|
static_assert(std::numeric_limits<double>::digits >=
|
|
std::numeric_limits<decltype(bufferEnd)>::digits,
|
|
"bufferEnd should be represented exactly by double");
|
|
// + 0.5 rounds mDuration to nearest sample when assigned to bufferEnd.
|
|
bufferEnd = std::min<double>(bufferEnd,
|
|
offsetSamples + mDuration * rate + 0.5);
|
|
}
|
|
aStream->SetInt32Parameter(BUFFEREND, bufferEnd);
|
|
|
|
MarkActive();
|
|
}
|
|
|
|
void
|
|
AudioBufferSourceNode::Stop(double aWhen, ErrorResult& aRv)
|
|
{
|
|
if (!WebAudioUtils::IsTimeValid(aWhen)) {
|
|
aRv.Throw(NS_ERROR_DOM_NOT_SUPPORTED_ERR);
|
|
return;
|
|
}
|
|
|
|
if (!mStartCalled) {
|
|
aRv.Throw(NS_ERROR_DOM_INVALID_STATE_ERR);
|
|
return;
|
|
}
|
|
|
|
WEB_AUDIO_API_LOG("%f: %s %u Stop(%f)", Context()->CurrentTime(),
|
|
NodeType(), Id(), aWhen);
|
|
|
|
AudioNodeStream* ns = mStream;
|
|
if (!ns || !Context()) {
|
|
// We've already stopped and had our stream shut down
|
|
return;
|
|
}
|
|
|
|
ns->SetStreamTimeParameter(STOP, Context(), std::max(0.0, aWhen));
|
|
}
|
|
|
|
void
|
|
AudioBufferSourceNode::NotifyMainThreadStreamFinished()
|
|
{
|
|
MOZ_ASSERT(mStream->IsFinished());
|
|
|
|
class EndedEventDispatcher final : public nsRunnable
|
|
{
|
|
public:
|
|
explicit EndedEventDispatcher(AudioBufferSourceNode* aNode)
|
|
: mNode(aNode) {}
|
|
NS_IMETHODIMP Run() override
|
|
{
|
|
// If it's not safe to run scripts right now, schedule this to run later
|
|
if (!nsContentUtils::IsSafeToRunScript()) {
|
|
nsContentUtils::AddScriptRunner(this);
|
|
return NS_OK;
|
|
}
|
|
|
|
mNode->DispatchTrustedEvent(NS_LITERAL_STRING("ended"));
|
|
// Release stream resources.
|
|
mNode->DestroyMediaStream();
|
|
return NS_OK;
|
|
}
|
|
private:
|
|
RefPtr<AudioBufferSourceNode> mNode;
|
|
};
|
|
|
|
NS_DispatchToMainThread(new EndedEventDispatcher(this));
|
|
|
|
// Drop the playing reference
|
|
// Warning: The below line might delete this.
|
|
MarkInactive();
|
|
}
|
|
|
|
void
|
|
AudioBufferSourceNode::SendDopplerShiftToStream(double aDopplerShift)
|
|
{
|
|
MOZ_ASSERT(mStream, "Should have disconnected panner if no stream");
|
|
SendDoubleParameterToStream(DOPPLERSHIFT, aDopplerShift);
|
|
}
|
|
|
|
void
|
|
AudioBufferSourceNode::SendLoopParametersToStream()
|
|
{
|
|
if (!mStream) {
|
|
return;
|
|
}
|
|
// Don't compute and set the loop parameters unnecessarily
|
|
if (mLoop && mBuffer) {
|
|
float rate = mBuffer->SampleRate();
|
|
double length = (double(mBuffer->Length()) / mBuffer->SampleRate());
|
|
double actualLoopStart, actualLoopEnd;
|
|
if (mLoopStart >= 0.0 && mLoopEnd > 0.0 &&
|
|
mLoopStart < mLoopEnd) {
|
|
MOZ_ASSERT(mLoopStart != 0.0 || mLoopEnd != 0.0);
|
|
actualLoopStart = (mLoopStart > length) ? 0.0 : mLoopStart;
|
|
actualLoopEnd = std::min(mLoopEnd, length);
|
|
} else {
|
|
actualLoopStart = 0.0;
|
|
actualLoopEnd = length;
|
|
}
|
|
int32_t loopStartTicks = NS_lround(actualLoopStart * rate);
|
|
int32_t loopEndTicks = NS_lround(actualLoopEnd * rate);
|
|
if (loopStartTicks < loopEndTicks) {
|
|
SendInt32ParameterToStream(LOOPSTART, loopStartTicks);
|
|
SendInt32ParameterToStream(LOOPEND, loopEndTicks);
|
|
SendInt32ParameterToStream(LOOP, 1);
|
|
} else {
|
|
// Be explicit about looping not happening if the offsets make
|
|
// looping impossible.
|
|
SendInt32ParameterToStream(LOOP, 0);
|
|
}
|
|
} else {
|
|
SendInt32ParameterToStream(LOOP, 0);
|
|
}
|
|
}
|
|
|
|
} // namespace dom
|
|
} // namespace mozilla
|