content/media/webaudio/AudioBufferSourceNode.cpp@6474c204b198
content/media/webaudio/AudioBufferSourceNode.cpp
Wed, 31 Dec 2014 06:09:35 +0100
- author
- Michael Schloh von Bennewitz <michael@schloh.com>
- date
- Wed, 31 Dec 2014 06:09:35 +0100
- changeset 0
- 6474c204b198
- permissions
- -rw-r--r--
Cloned upstream origin tor-browser at tor-browser-31.3.0esr-4.5-1-build1
revision ID fc1c9ff7c1b2defdbc039f12214767608f46423f for hacking purpose.
1 /* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
2 /* vim:set ts=2 sw=2 sts=2 et cindent: */
3 /* This Source Code Form is subject to the terms of the Mozilla Public
4 * License, v. 2.0. If a copy of the MPL was not distributed with this
5 * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
7 #include "AudioBufferSourceNode.h"
8 #include "mozilla/dom/AudioBufferSourceNodeBinding.h"
9 #include "mozilla/dom/AudioParam.h"
10 #include "nsMathUtils.h"
11 #include "AudioNodeEngine.h"
12 #include "AudioNodeStream.h"
13 #include "AudioDestinationNode.h"
14 #include "AudioParamTimeline.h"
15 #include "speex/speex_resampler.h"
16 #include <limits>
18 namespace mozilla {
19 namespace dom {
21 NS_IMPL_CYCLE_COLLECTION_CLASS(AudioBufferSourceNode)
23 NS_IMPL_CYCLE_COLLECTION_UNLINK_BEGIN(AudioBufferSourceNode)
24 NS_IMPL_CYCLE_COLLECTION_UNLINK(mBuffer)
25 NS_IMPL_CYCLE_COLLECTION_UNLINK(mPlaybackRate)
26 if (tmp->Context()) {
27 // AudioNode's Unlink implementation disconnects us from the graph
28 // too, but we need to do this right here to make sure that
29 // UnregisterAudioBufferSourceNode can properly untangle us from
30 // the possibly connected PannerNodes.
31 tmp->DisconnectFromGraph();
32 tmp->Context()->UnregisterAudioBufferSourceNode(tmp);
33 }
34 NS_IMPL_CYCLE_COLLECTION_UNLINK_END_INHERITED(AudioNode)
36 NS_IMPL_CYCLE_COLLECTION_TRAVERSE_BEGIN_INHERITED(AudioBufferSourceNode, AudioNode)
37 NS_IMPL_CYCLE_COLLECTION_TRAVERSE(mBuffer)
38 NS_IMPL_CYCLE_COLLECTION_TRAVERSE(mPlaybackRate)
39 NS_IMPL_CYCLE_COLLECTION_TRAVERSE_END
41 NS_INTERFACE_MAP_BEGIN_CYCLE_COLLECTION_INHERITED(AudioBufferSourceNode)
42 NS_INTERFACE_MAP_END_INHERITING(AudioNode)
44 NS_IMPL_ADDREF_INHERITED(AudioBufferSourceNode, AudioNode)
45 NS_IMPL_RELEASE_INHERITED(AudioBufferSourceNode, AudioNode)
47 /**
48 * Media-thread playback engine for AudioBufferSourceNode.
49 * Nothing is played until a non-null buffer has been set (via
50 * AudioNodeStream::SetBuffer) and a non-zero mBufferEnd has been set (via
51 * AudioNodeStream::SetInt32Parameter).
52 */
53 class AudioBufferSourceNodeEngine : public AudioNodeEngine
54 {
55 public:
56 explicit AudioBufferSourceNodeEngine(AudioNode* aNode,
57 AudioDestinationNode* aDestination) :
58 AudioNodeEngine(aNode),
59 mStart(0.0), mBeginProcessing(0),
60 mStop(TRACK_TICKS_MAX),
61 mResampler(nullptr), mRemainingResamplerTail(0),
62 mBufferEnd(0),
63 mLoopStart(0), mLoopEnd(0),
64 mBufferSampleRate(0), mBufferPosition(0), mChannels(0),
65 mDopplerShift(1.0f),
66 mDestination(static_cast<AudioNodeStream*>(aDestination->Stream())),
67 mPlaybackRateTimeline(1.0f), mLoop(false)
68 {}
70 ~AudioBufferSourceNodeEngine()
71 {
72 if (mResampler) {
73 speex_resampler_destroy(mResampler);
74 }
75 }
77 void SetSourceStream(AudioNodeStream* aSource)
78 {
79 mSource = aSource;
80 }
82 virtual void SetTimelineParameter(uint32_t aIndex,
83 const dom::AudioParamTimeline& aValue,
84 TrackRate aSampleRate) MOZ_OVERRIDE
85 {
86 switch (aIndex) {
87 case AudioBufferSourceNode::PLAYBACKRATE:
88 mPlaybackRateTimeline = aValue;
89 WebAudioUtils::ConvertAudioParamToTicks(mPlaybackRateTimeline, mSource, mDestination);
90 break;
91 default:
92 NS_ERROR("Bad AudioBufferSourceNodeEngine TimelineParameter");
93 }
94 }
95 virtual void SetStreamTimeParameter(uint32_t aIndex, TrackTicks aParam)
96 {
97 switch (aIndex) {
98 case AudioBufferSourceNode::STOP: mStop = aParam; break;
99 default:
100 NS_ERROR("Bad AudioBufferSourceNodeEngine StreamTimeParameter");
101 }
102 }
103 virtual void SetDoubleParameter(uint32_t aIndex, double aParam)
104 {
105 switch (aIndex) {
106 case AudioBufferSourceNode::START:
107 MOZ_ASSERT(!mStart, "Another START?");
108 mStart = mSource->TimeFromDestinationTime(mDestination, aParam) *
109 mSource->SampleRate();
110 // Round to nearest
111 mBeginProcessing = mStart + 0.5;
112 break;
113 case AudioBufferSourceNode::DOPPLERSHIFT:
114 mDopplerShift = aParam > 0 && aParam == aParam ? aParam : 1.0;
115 break;
116 default:
117 NS_ERROR("Bad AudioBufferSourceNodeEngine double parameter.");
118 };
119 }
120 virtual void SetInt32Parameter(uint32_t aIndex, int32_t aParam)
121 {
122 switch (aIndex) {
123 case AudioBufferSourceNode::SAMPLE_RATE: mBufferSampleRate = aParam; break;
124 case AudioBufferSourceNode::BUFFERSTART:
125 if (mBufferPosition == 0) {
126 mBufferPosition = aParam;
127 }
128 break;
129 case AudioBufferSourceNode::BUFFEREND: mBufferEnd = aParam; break;
130 case AudioBufferSourceNode::LOOP: mLoop = !!aParam; break;
131 case AudioBufferSourceNode::LOOPSTART: mLoopStart = aParam; break;
132 case AudioBufferSourceNode::LOOPEND: mLoopEnd = aParam; break;
133 default:
134 NS_ERROR("Bad AudioBufferSourceNodeEngine Int32Parameter");
135 }
136 }
137 virtual void SetBuffer(already_AddRefed<ThreadSharedFloatArrayBufferList> aBuffer)
138 {
139 mBuffer = aBuffer;
140 }
142 bool BegunResampling()
143 {
144 return mBeginProcessing == -TRACK_TICKS_MAX;
145 }
147 void UpdateResampler(int32_t aOutRate, uint32_t aChannels)
148 {
149 if (mResampler &&
150 (aChannels != mChannels ||
151 // If the resampler has begun, then it will have moved
152 // mBufferPosition to after the samples it has read, but it hasn't
153 // output its buffered samples. Keep using the resampler, even if
154 // the rates now match, so that this latent segment is output.
155 (aOutRate == mBufferSampleRate && !BegunResampling()))) {
156 speex_resampler_destroy(mResampler);
157 mResampler = nullptr;
158 mRemainingResamplerTail = 0;
159 mBeginProcessing = mStart + 0.5;
160 }
162 if (aOutRate == mBufferSampleRate && !mResampler) {
163 return;
164 }
166 if (!mResampler) {
167 mChannels = aChannels;
168 mResampler = speex_resampler_init(mChannels, mBufferSampleRate, aOutRate,
169 SPEEX_RESAMPLER_QUALITY_DEFAULT,
170 nullptr);
171 } else {
172 uint32_t currentOutSampleRate, currentInSampleRate;
173 speex_resampler_get_rate(mResampler, ¤tInSampleRate,
174 ¤tOutSampleRate);
175 if (currentOutSampleRate == static_cast<uint32_t>(aOutRate)) {
176 return;
177 }
178 speex_resampler_set_rate(mResampler, currentInSampleRate, aOutRate);
179 }
181 if (!BegunResampling()) {
182 // Low pass filter effects from the resampler mean that samples before
183 // the start time are influenced by resampling the buffer. The input
184 // latency indicates half the filter width.
185 int64_t inputLatency = speex_resampler_get_input_latency(mResampler);
186 uint32_t ratioNum, ratioDen;
187 speex_resampler_get_ratio(mResampler, &ratioNum, &ratioDen);
188 // The output subsample resolution supported in aligning the resampler
189 // is ratioNum. First round the start time to the nearest subsample.
190 int64_t subsample = mStart * ratioNum + 0.5;
191 // Now include the leading effects of the filter, and round *up* to the
192 // next whole tick, because there is no effect on samples outside the
193 // filter width.
194 mBeginProcessing =
195 (subsample - inputLatency * ratioDen + ratioNum - 1) / ratioNum;
196 }
197 }
199 // Borrow a full buffer of size WEBAUDIO_BLOCK_SIZE from the source buffer
200 // at offset aSourceOffset. This avoids copying memory.
201 void BorrowFromInputBuffer(AudioChunk* aOutput,
202 uint32_t aChannels)
203 {
204 aOutput->mDuration = WEBAUDIO_BLOCK_SIZE;
205 aOutput->mBuffer = mBuffer;
206 aOutput->mChannelData.SetLength(aChannels);
207 for (uint32_t i = 0; i < aChannels; ++i) {
208 aOutput->mChannelData[i] = mBuffer->GetData(i) + mBufferPosition;
209 }
210 aOutput->mVolume = 1.0f;
211 aOutput->mBufferFormat = AUDIO_FORMAT_FLOAT32;
212 }
214 // Copy aNumberOfFrames frames from the source buffer at offset aSourceOffset
215 // and put it at offset aBufferOffset in the destination buffer.
216 void CopyFromInputBuffer(AudioChunk* aOutput,
217 uint32_t aChannels,
218 uintptr_t aOffsetWithinBlock,
219 uint32_t aNumberOfFrames) {
220 for (uint32_t i = 0; i < aChannels; ++i) {
221 float* baseChannelData = static_cast<float*>(const_cast<void*>(aOutput->mChannelData[i]));
222 memcpy(baseChannelData + aOffsetWithinBlock,
223 mBuffer->GetData(i) + mBufferPosition,
224 aNumberOfFrames * sizeof(float));
225 }
226 }
228 // Resamples input data to an output buffer, according to |mBufferSampleRate| and
229 // the playbackRate.
230 // The number of frames consumed/produced depends on the amount of space
231 // remaining in both the input and output buffer, and the playback rate (that
232 // is, the ratio between the output samplerate and the input samplerate).
233 void CopyFromInputBufferWithResampling(AudioNodeStream* aStream,
234 AudioChunk* aOutput,
235 uint32_t aChannels,
236 uint32_t* aOffsetWithinBlock,
237 TrackTicks* aCurrentPosition,
238 int32_t aBufferMax) {
239 // TODO: adjust for mStop (see bug 913854 comment 9).
240 uint32_t availableInOutputBuffer =
241 WEBAUDIO_BLOCK_SIZE - *aOffsetWithinBlock;
242 SpeexResamplerState* resampler = mResampler;
243 MOZ_ASSERT(aChannels > 0);
245 if (mBufferPosition < aBufferMax) {
246 uint32_t availableInInputBuffer = aBufferMax - mBufferPosition;
247 uint32_t ratioNum, ratioDen;
248 speex_resampler_get_ratio(resampler, &ratioNum, &ratioDen);
249 // Limit the number of input samples copied and possibly
250 // format-converted for resampling by estimating how many will be used.
251 // This may be a little small if still filling the resampler with
252 // initial data, but we'll get called again and it will work out.
253 uint32_t inputLimit = availableInOutputBuffer * ratioNum / ratioDen + 10;
254 if (!BegunResampling()) {
255 // First time the resampler is used.
256 uint32_t inputLatency = speex_resampler_get_input_latency(resampler);
257 inputLimit += inputLatency;
258 // If starting after mStart, then play from the beginning of the
259 // buffer, but correct for input latency. If starting before mStart,
260 // then align the resampler so that the time corresponding to the
261 // first input sample is mStart.
262 uint32_t skipFracNum = inputLatency * ratioDen;
263 double leadTicks = mStart - *aCurrentPosition;
264 if (leadTicks > 0.0) {
265 // Round to nearest output subsample supported by the resampler at
266 // these rates.
267 skipFracNum -= leadTicks * ratioNum + 0.5;
268 MOZ_ASSERT(skipFracNum < INT32_MAX, "mBeginProcessing is wrong?");
269 }
270 speex_resampler_set_skip_frac_num(resampler, skipFracNum);
272 mBeginProcessing = -TRACK_TICKS_MAX;
273 }
274 inputLimit = std::min(inputLimit, availableInInputBuffer);
276 for (uint32_t i = 0; true; ) {
277 uint32_t inSamples = inputLimit;
278 const float* inputData = mBuffer->GetData(i) + mBufferPosition;
280 uint32_t outSamples = availableInOutputBuffer;
281 float* outputData =
282 static_cast<float*>(const_cast<void*>(aOutput->mChannelData[i])) +
283 *aOffsetWithinBlock;
285 WebAudioUtils::SpeexResamplerProcess(resampler, i,
286 inputData, &inSamples,
287 outputData, &outSamples);
288 if (++i == aChannels) {
289 mBufferPosition += inSamples;
290 MOZ_ASSERT(mBufferPosition <= mBufferEnd || mLoop);
291 *aOffsetWithinBlock += outSamples;
292 *aCurrentPosition += outSamples;
293 if (inSamples == availableInInputBuffer && !mLoop) {
294 // We'll feed in enough zeros to empty out the resampler's memory.
295 // This handles the output latency as well as capturing the low
296 // pass effects of the resample filter.
297 mRemainingResamplerTail =
298 2 * speex_resampler_get_input_latency(resampler) - 1;
299 }
300 return;
301 }
302 }
303 } else {
304 for (uint32_t i = 0; true; ) {
305 uint32_t inSamples = mRemainingResamplerTail;
306 uint32_t outSamples = availableInOutputBuffer;
307 float* outputData =
308 static_cast<float*>(const_cast<void*>(aOutput->mChannelData[i])) +
309 *aOffsetWithinBlock;
311 // AudioDataValue* for aIn selects the function that does not try to
312 // copy and format-convert input data.
313 WebAudioUtils::SpeexResamplerProcess(resampler, i,
314 static_cast<AudioDataValue*>(nullptr), &inSamples,
315 outputData, &outSamples);
316 if (++i == aChannels) {
317 mRemainingResamplerTail -= inSamples;
318 MOZ_ASSERT(mRemainingResamplerTail >= 0);
319 *aOffsetWithinBlock += outSamples;
320 *aCurrentPosition += outSamples;
321 break;
322 }
323 }
324 }
325 }
327 /**
328 * Fill aOutput with as many zero frames as we can, and advance
329 * aOffsetWithinBlock and aCurrentPosition based on how many frames we write.
330 * This will never advance aOffsetWithinBlock past WEBAUDIO_BLOCK_SIZE or
331 * aCurrentPosition past aMaxPos. This function knows when it needs to
332 * allocate the output buffer, and also optimizes the case where it can avoid
333 * memory allocations.
334 */
335 void FillWithZeroes(AudioChunk* aOutput,
336 uint32_t aChannels,
337 uint32_t* aOffsetWithinBlock,
338 TrackTicks* aCurrentPosition,
339 TrackTicks aMaxPos)
340 {
341 MOZ_ASSERT(*aCurrentPosition < aMaxPos);
342 uint32_t numFrames =
343 std::min<TrackTicks>(WEBAUDIO_BLOCK_SIZE - *aOffsetWithinBlock,
344 aMaxPos - *aCurrentPosition);
345 if (numFrames == WEBAUDIO_BLOCK_SIZE) {
346 aOutput->SetNull(numFrames);
347 } else {
348 if (*aOffsetWithinBlock == 0) {
349 AllocateAudioBlock(aChannels, aOutput);
350 }
351 WriteZeroesToAudioBlock(aOutput, *aOffsetWithinBlock, numFrames);
352 }
353 *aOffsetWithinBlock += numFrames;
354 *aCurrentPosition += numFrames;
355 }
357 /**
358 * Copy as many frames as possible from the source buffer to aOutput, and
359 * advance aOffsetWithinBlock and aCurrentPosition based on how many frames
360 * we write. This will never advance aOffsetWithinBlock past
361 * WEBAUDIO_BLOCK_SIZE, or aCurrentPosition past mStop. It takes data from
362 * the buffer at aBufferOffset, and never takes more data than aBufferMax.
363 * This function knows when it needs to allocate the output buffer, and also
364 * optimizes the case where it can avoid memory allocations.
365 */
366 void CopyFromBuffer(AudioNodeStream* aStream,
367 AudioChunk* aOutput,
368 uint32_t aChannels,
369 uint32_t* aOffsetWithinBlock,
370 TrackTicks* aCurrentPosition,
371 int32_t aBufferMax)
372 {
373 MOZ_ASSERT(*aCurrentPosition < mStop);
374 uint32_t numFrames =
375 std::min(std::min<TrackTicks>(WEBAUDIO_BLOCK_SIZE - *aOffsetWithinBlock,
376 aBufferMax - mBufferPosition),
377 mStop - *aCurrentPosition);
378 if (numFrames == WEBAUDIO_BLOCK_SIZE && !mResampler) {
379 MOZ_ASSERT(mBufferPosition < aBufferMax);
380 BorrowFromInputBuffer(aOutput, aChannels);
381 *aOffsetWithinBlock += numFrames;
382 *aCurrentPosition += numFrames;
383 mBufferPosition += numFrames;
384 } else {
385 if (*aOffsetWithinBlock == 0) {
386 AllocateAudioBlock(aChannels, aOutput);
387 }
388 if (!mResampler) {
389 MOZ_ASSERT(mBufferPosition < aBufferMax);
390 CopyFromInputBuffer(aOutput, aChannels, *aOffsetWithinBlock, numFrames);
391 *aOffsetWithinBlock += numFrames;
392 *aCurrentPosition += numFrames;
393 mBufferPosition += numFrames;
394 } else {
395 CopyFromInputBufferWithResampling(aStream, aOutput, aChannels, aOffsetWithinBlock, aCurrentPosition, aBufferMax);
396 }
397 }
398 }
400 int32_t ComputeFinalOutSampleRate(float aPlaybackRate)
401 {
402 // Make sure the playback rate and the doppler shift are something
403 // our resampler can work with.
404 int32_t rate = WebAudioUtils::
405 TruncateFloatToInt<int32_t>(mSource->SampleRate() /
406 (aPlaybackRate * mDopplerShift));
407 return rate ? rate : mBufferSampleRate;
408 }
410 void UpdateSampleRateIfNeeded(uint32_t aChannels)
411 {
412 float playbackRate;
414 if (mPlaybackRateTimeline.HasSimpleValue()) {
415 playbackRate = mPlaybackRateTimeline.GetValue();
416 } else {
417 playbackRate = mPlaybackRateTimeline.GetValueAtTime(mSource->GetCurrentPosition());
418 }
419 if (playbackRate <= 0 || playbackRate != playbackRate) {
420 playbackRate = 1.0f;
421 }
423 int32_t outRate = ComputeFinalOutSampleRate(playbackRate);
424 UpdateResampler(outRate, aChannels);
425 }
427 virtual void ProcessBlock(AudioNodeStream* aStream,
428 const AudioChunk& aInput,
429 AudioChunk* aOutput,
430 bool* aFinished)
431 {
432 if (!mBuffer || !mBufferEnd) {
433 aOutput->SetNull(WEBAUDIO_BLOCK_SIZE);
434 return;
435 }
437 uint32_t channels = mBuffer->GetChannels();
438 if (!channels) {
439 aOutput->SetNull(WEBAUDIO_BLOCK_SIZE);
440 return;
441 }
443 // WebKit treats the playbackRate as a k-rate parameter in their code,
444 // despite the spec saying that it should be an a-rate parameter. We treat
445 // it as k-rate. Spec bug: https://www.w3.org/Bugs/Public/show_bug.cgi?id=21592
446 UpdateSampleRateIfNeeded(channels);
448 uint32_t written = 0;
449 TrackTicks streamPosition = aStream->GetCurrentPosition();
450 while (written < WEBAUDIO_BLOCK_SIZE) {
451 if (mStop != TRACK_TICKS_MAX &&
452 streamPosition >= mStop) {
453 FillWithZeroes(aOutput, channels, &written, &streamPosition, TRACK_TICKS_MAX);
454 continue;
455 }
456 if (streamPosition < mBeginProcessing) {
457 FillWithZeroes(aOutput, channels, &written, &streamPosition,
458 mBeginProcessing);
459 continue;
460 }
461 if (mLoop) {
462 // mLoopEnd can become less than mBufferPosition when a LOOPEND engine
463 // parameter is received after "loopend" is changed on the node or a
464 // new buffer with lower samplerate is set.
465 if (mBufferPosition >= mLoopEnd) {
466 mBufferPosition = mLoopStart;
467 }
468 CopyFromBuffer(aStream, aOutput, channels, &written, &streamPosition, mLoopEnd);
469 } else {
470 if (mBufferPosition < mBufferEnd || mRemainingResamplerTail) {
471 CopyFromBuffer(aStream, aOutput, channels, &written, &streamPosition, mBufferEnd);
472 } else {
473 FillWithZeroes(aOutput, channels, &written, &streamPosition, TRACK_TICKS_MAX);
474 }
475 }
476 }
478 // We've finished if we've gone past mStop, or if we're past mDuration when
479 // looping is disabled.
480 if (streamPosition >= mStop ||
481 (!mLoop && mBufferPosition >= mBufferEnd && !mRemainingResamplerTail)) {
482 *aFinished = true;
483 }
484 }
486 virtual size_t SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const MOZ_OVERRIDE
487 {
488 // Not owned:
489 // - mBuffer - shared w/ AudioNode
490 // - mPlaybackRateTimeline - shared w/ AudioNode
492 size_t amount = AudioNodeEngine::SizeOfExcludingThis(aMallocSizeOf);
494 // NB: We need to modify speex if we want the full memory picture, internal
495 // fields that need measuring noted below.
496 // - mResampler->mem
497 // - mResampler->sinc_table
498 // - mResampler->last_sample
499 // - mResampler->magic_samples
500 // - mResampler->samp_frac_num
501 amount += aMallocSizeOf(mResampler);
503 return amount;
504 }
506 virtual size_t SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const MOZ_OVERRIDE
507 {
508 return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
509 }
511 double mStart; // including the fractional position between ticks
512 // Low pass filter effects from the resampler mean that samples before the
513 // start time are influenced by resampling the buffer. mBeginProcessing
514 // includes the extent of this filter. The special value of -TRACK_TICKS_MAX
515 // indicates that the resampler has begun processing.
516 TrackTicks mBeginProcessing;
517 TrackTicks mStop;
518 nsRefPtr<ThreadSharedFloatArrayBufferList> mBuffer;
519 SpeexResamplerState* mResampler;
520 // mRemainingResamplerTail, like mBufferPosition, and
521 // mBufferEnd, is measured in input buffer samples.
522 int mRemainingResamplerTail;
523 int32_t mBufferEnd;
524 int32_t mLoopStart;
525 int32_t mLoopEnd;
526 int32_t mBufferSampleRate;
527 int32_t mBufferPosition;
528 uint32_t mChannels;
529 float mDopplerShift;
530 AudioNodeStream* mDestination;
531 AudioNodeStream* mSource;
532 AudioParamTimeline mPlaybackRateTimeline;
533 bool mLoop;
534 };
536 AudioBufferSourceNode::AudioBufferSourceNode(AudioContext* aContext)
537 : AudioNode(aContext,
538 2,
539 ChannelCountMode::Max,
540 ChannelInterpretation::Speakers)
541 , mLoopStart(0.0)
542 , mLoopEnd(0.0)
543 // mOffset and mDuration are initialized in Start().
544 , mPlaybackRate(new AudioParam(MOZ_THIS_IN_INITIALIZER_LIST(),
545 SendPlaybackRateToStream, 1.0f))
546 , mLoop(false)
547 , mStartCalled(false)
548 , mStopped(false)
549 {
550 AudioBufferSourceNodeEngine* engine = new AudioBufferSourceNodeEngine(this, aContext->Destination());
551 mStream = aContext->Graph()->CreateAudioNodeStream(engine, MediaStreamGraph::SOURCE_STREAM);
552 engine->SetSourceStream(static_cast<AudioNodeStream*>(mStream.get()));
553 mStream->AddMainThreadListener(this);
554 }
556 AudioBufferSourceNode::~AudioBufferSourceNode()
557 {
558 if (Context()) {
559 Context()->UnregisterAudioBufferSourceNode(this);
560 }
561 }
563 size_t
564 AudioBufferSourceNode::SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const
565 {
566 size_t amount = AudioNode::SizeOfExcludingThis(aMallocSizeOf);
567 if (mBuffer) {
568 amount += mBuffer->SizeOfIncludingThis(aMallocSizeOf);
569 }
571 amount += mPlaybackRate->SizeOfIncludingThis(aMallocSizeOf);
572 return amount;
573 }
575 size_t
576 AudioBufferSourceNode::SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const
577 {
578 return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
579 }
581 JSObject*
582 AudioBufferSourceNode::WrapObject(JSContext* aCx)
583 {
584 return AudioBufferSourceNodeBinding::Wrap(aCx, this);
585 }
587 void
588 AudioBufferSourceNode::Start(double aWhen, double aOffset,
589 const Optional<double>& aDuration, ErrorResult& aRv)
590 {
591 if (!WebAudioUtils::IsTimeValid(aWhen) ||
592 (aDuration.WasPassed() && !WebAudioUtils::IsTimeValid(aDuration.Value()))) {
593 aRv.Throw(NS_ERROR_DOM_NOT_SUPPORTED_ERR);
594 return;
595 }
597 if (mStartCalled) {
598 aRv.Throw(NS_ERROR_DOM_INVALID_STATE_ERR);
599 return;
600 }
601 mStartCalled = true;
603 AudioNodeStream* ns = static_cast<AudioNodeStream*>(mStream.get());
604 if (!ns) {
605 // Nothing to play, or we're already dead for some reason
606 return;
607 }
609 // Remember our arguments so that we can use them when we get a new buffer.
610 mOffset = aOffset;
611 mDuration = aDuration.WasPassed() ? aDuration.Value()
612 : std::numeric_limits<double>::min();
613 // We can't send these parameters without a buffer because we don't know the
614 // buffer's sample rate or length.
615 if (mBuffer) {
616 SendOffsetAndDurationParametersToStream(ns);
617 }
619 // Don't set parameter unnecessarily
620 if (aWhen > 0.0) {
621 ns->SetDoubleParameter(START, mContext->DOMTimeToStreamTime(aWhen));
622 }
623 }
625 void
626 AudioBufferSourceNode::SendBufferParameterToStream(JSContext* aCx)
627 {
628 AudioNodeStream* ns = static_cast<AudioNodeStream*>(mStream.get());
629 MOZ_ASSERT(ns, "Why don't we have a stream here?");
631 if (mBuffer) {
632 float rate = mBuffer->SampleRate();
633 nsRefPtr<ThreadSharedFloatArrayBufferList> data =
634 mBuffer->GetThreadSharedChannelsForRate(aCx);
635 ns->SetBuffer(data.forget());
636 ns->SetInt32Parameter(SAMPLE_RATE, rate);
638 if (mStartCalled) {
639 SendOffsetAndDurationParametersToStream(ns);
640 }
641 } else {
642 ns->SetBuffer(nullptr);
644 MarkInactive();
645 }
646 }
648 void
649 AudioBufferSourceNode::SendOffsetAndDurationParametersToStream(AudioNodeStream* aStream)
650 {
651 NS_ASSERTION(mBuffer && mStartCalled,
652 "Only call this when we have a buffer and start() has been called");
654 float rate = mBuffer->SampleRate();
655 int32_t bufferEnd = mBuffer->Length();
656 int32_t offsetSamples = std::max(0, NS_lround(mOffset * rate));
658 // Don't set parameter unnecessarily
659 if (offsetSamples > 0) {
660 aStream->SetInt32Parameter(BUFFERSTART, offsetSamples);
661 }
663 if (mDuration != std::numeric_limits<double>::min()) {
664 bufferEnd = std::min(bufferEnd,
665 offsetSamples + NS_lround(mDuration * rate));
666 }
667 aStream->SetInt32Parameter(BUFFEREND, bufferEnd);
669 MarkActive();
670 }
672 void
673 AudioBufferSourceNode::Stop(double aWhen, ErrorResult& aRv)
674 {
675 if (!WebAudioUtils::IsTimeValid(aWhen)) {
676 aRv.Throw(NS_ERROR_DOM_NOT_SUPPORTED_ERR);
677 return;
678 }
680 if (!mStartCalled) {
681 aRv.Throw(NS_ERROR_DOM_INVALID_STATE_ERR);
682 return;
683 }
685 AudioNodeStream* ns = static_cast<AudioNodeStream*>(mStream.get());
686 if (!ns || !Context()) {
687 // We've already stopped and had our stream shut down
688 return;
689 }
691 ns->SetStreamTimeParameter(STOP, Context(), std::max(0.0, aWhen));
692 }
694 void
695 AudioBufferSourceNode::NotifyMainThreadStateChanged()
696 {
697 if (mStream->IsFinished()) {
698 class EndedEventDispatcher : public nsRunnable
699 {
700 public:
701 explicit EndedEventDispatcher(AudioBufferSourceNode* aNode)
702 : mNode(aNode) {}
703 NS_IMETHODIMP Run()
704 {
705 // If it's not safe to run scripts right now, schedule this to run later
706 if (!nsContentUtils::IsSafeToRunScript()) {
707 nsContentUtils::AddScriptRunner(this);
708 return NS_OK;
709 }
711 mNode->DispatchTrustedEvent(NS_LITERAL_STRING("ended"));
712 return NS_OK;
713 }
714 private:
715 nsRefPtr<AudioBufferSourceNode> mNode;
716 };
717 if (!mStopped) {
718 // Only dispatch the ended event once
719 NS_DispatchToMainThread(new EndedEventDispatcher(this));
720 mStopped = true;
721 }
723 // Drop the playing reference
724 // Warning: The below line might delete this.
725 MarkInactive();
726 }
727 }
729 void
730 AudioBufferSourceNode::SendPlaybackRateToStream(AudioNode* aNode)
731 {
732 AudioBufferSourceNode* This = static_cast<AudioBufferSourceNode*>(aNode);
733 SendTimelineParameterToStream(This, PLAYBACKRATE, *This->mPlaybackRate);
734 }
736 void
737 AudioBufferSourceNode::SendDopplerShiftToStream(double aDopplerShift)
738 {
739 SendDoubleParameterToStream(DOPPLERSHIFT, aDopplerShift);
740 }
742 void
743 AudioBufferSourceNode::SendLoopParametersToStream()
744 {
745 // Don't compute and set the loop parameters unnecessarily
746 if (mLoop && mBuffer) {
747 float rate = mBuffer->SampleRate();
748 double length = (double(mBuffer->Length()) / mBuffer->SampleRate());
749 double actualLoopStart, actualLoopEnd;
750 if (mLoopStart >= 0.0 && mLoopEnd > 0.0 &&
751 mLoopStart < mLoopEnd) {
752 MOZ_ASSERT(mLoopStart != 0.0 || mLoopEnd != 0.0);
753 actualLoopStart = (mLoopStart > length) ? 0.0 : mLoopStart;
754 actualLoopEnd = std::min(mLoopEnd, length);
755 } else {
756 actualLoopStart = 0.0;
757 actualLoopEnd = length;
758 }
759 int32_t loopStartTicks = NS_lround(actualLoopStart * rate);
760 int32_t loopEndTicks = NS_lround(actualLoopEnd * rate);
761 if (loopStartTicks < loopEndTicks) {
762 SendInt32ParameterToStream(LOOPSTART, loopStartTicks);
763 SendInt32ParameterToStream(LOOPEND, loopEndTicks);
764 SendInt32ParameterToStream(LOOP, 1);
765 } else {
766 // Be explicit about looping not happening if the offsets make
767 // looping impossible.
768 SendInt32ParameterToStream(LOOP, 0);
769 }
770 } else if (!mLoop) {
771 SendInt32ParameterToStream(LOOP, 0);
772 }
773 }
775 }
776 }
