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| 1 /* | 1 /* |
| 2 * Copyright (C) 2010, Google Inc. All rights reserved. | 2 * Copyright (C) 2010, Google Inc. All rights reserved. |
| 3 * | 3 * |
| 4 * Redistribution and use in source and binary forms, with or without | 4 * Redistribution and use in source and binary forms, with or without |
| 5 * modification, are permitted provided that the following conditions | 5 * modification, are permitted provided that the following conditions |
| 6 * are met: | 6 * are met: |
| 7 * 1. Redistributions of source code must retain the above copyright | 7 * 1. Redistributions of source code must retain the above copyright |
| 8 * notice, this list of conditions and the following disclaimer. | 8 * notice, this list of conditions and the following disclaimer. |
| 9 * 2. Redistributions in binary form must reproduce the above copyright | 9 * 2. Redistributions in binary form must reproduce the above copyright |
| 10 * notice, this list of conditions and the following disclaimer in the | 10 * notice, this list of conditions and the following disclaimer in the |
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| 178 m_renderingAutomaticPullNodes.resize(m_automaticPullNodes.size()); | 178 m_renderingAutomaticPullNodes.resize(m_automaticPullNodes.size()); |
| 179 ASSERT(!m_renderingAutomaticPullNodes.size()); | 179 ASSERT(!m_renderingAutomaticPullNodes.size()); |
| 180 } | 180 } |
| 181 | 181 |
| 182 void AudioContext::lazyInitialize() | 182 void AudioContext::lazyInitialize() |
| 183 { | 183 { |
| 184 if (!m_isInitialized) { | 184 if (!m_isInitialized) { |
| 185 // Don't allow the context to initialize a second time after it's alread
y been explicitly uninitialized. | 185 // Don't allow the context to initialize a second time after it's alread
y been explicitly uninitialized. |
| 186 ASSERT(!m_isAudioThreadFinished); | 186 ASSERT(!m_isAudioThreadFinished); |
| 187 if (!m_isAudioThreadFinished) { | 187 if (!m_isAudioThreadFinished) { |
| 188 // Creation of a destination node should not start the audio HW. The |
| 189 // creation of any other AudioNode will initialize the audio HW and
start processing |
| 188 if (m_destinationNode.get()) { | 190 if (m_destinationNode.get()) { |
| 189 m_destinationNode->initialize(); | 191 m_destinationNode->initialize(); |
| 190 | 192 |
| 191 if (!isOfflineContext()) { | 193 if (!isOfflineContext()) { |
| 192 // This starts the audio thread. The destination node's prov
ideInput() method will now be called repeatedly to render audio. | 194 // This starts the audio thread. The destination node's prov
ideInput() method will now be called repeatedly to render audio. |
| 193 // Each time provideInput() is called, a portion of the audi
o stream is rendered. Let's call this time period a "render quantum". | 195 // Each time provideInput() is called, a portion of the audi
o stream is rendered. Let's call this time period a "render quantum". |
| 194 // NOTE: for now default AudioContext does not need an expli
cit startRendering() call from JavaScript. | 196 // NOTE: for now default AudioContext does not need an expli
cit startRendering() call from JavaScript. |
| 195 // We may want to consider requiring it for symmetry with Of
flineAudioContext. | 197 // We may want to consider requiring it for symmetry with Of
flineAudioContext. |
| 196 m_destinationNode->startRendering(); | 198 m_destinationNode->startRendering(); |
| 197 ++s_hardwareContextCount; | 199 ++s_hardwareContextCount; |
| 198 } | 200 } |
| 199 | 201 |
| 202 m_isInitialized = true; |
| 200 } | 203 } |
| 201 m_isInitialized = true; | |
| 202 } | 204 } |
| 203 } | 205 } |
| 204 } | 206 } |
| 205 | 207 |
| 206 void AudioContext::clear() | 208 void AudioContext::clear() |
| 207 { | 209 { |
| 208 // We have to release our reference to the destination node before the conte
xt will ever be deleted since the destination node holds a reference to the cont
ext. | 210 // We have to release our reference to the destination node before the conte
xt will ever be deleted since the destination node holds a reference to the cont
ext. |
| 209 if (m_destinationNode) | 211 if (m_destinationNode) |
| 210 m_destinationNode.clear(); | 212 m_destinationNode.clear(); |
| 211 | 213 |
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| 339 SyntaxError, | 341 SyntaxError, |
| 340 "invalid ArrayBuffer for audioData."); | 342 "invalid ArrayBuffer for audioData."); |
| 341 return; | 343 return; |
| 342 } | 344 } |
| 343 m_audioDecoder.decodeAsync(audioData, sampleRate(), successCallback, errorCa
llback); | 345 m_audioDecoder.decodeAsync(audioData, sampleRate(), successCallback, errorCa
llback); |
| 344 } | 346 } |
| 345 | 347 |
| 346 PassRefPtr<AudioBufferSourceNode> AudioContext::createBufferSource() | 348 PassRefPtr<AudioBufferSourceNode> AudioContext::createBufferSource() |
| 347 { | 349 { |
| 348 ASSERT(isMainThread()); | 350 ASSERT(isMainThread()); |
| 349 lazyInitialize(); | |
| 350 RefPtr<AudioBufferSourceNode> node = AudioBufferSourceNode::create(this, m_d
estinationNode->sampleRate()); | 351 RefPtr<AudioBufferSourceNode> node = AudioBufferSourceNode::create(this, m_d
estinationNode->sampleRate()); |
| 351 | 352 |
| 352 // Because this is an AudioScheduledSourceNode, the context keeps a referenc
e until it has finished playing. | 353 // Because this is an AudioScheduledSourceNode, the context keeps a referenc
e until it has finished playing. |
| 353 // When this happens, AudioScheduledSourceNode::finish() calls AudioContext:
:notifyNodeFinishedProcessing(). | 354 // When this happens, AudioScheduledSourceNode::finish() calls AudioContext:
:notifyNodeFinishedProcessing(). |
| 354 refNode(node.get()); | 355 refNode(node.get()); |
| 355 | 356 |
| 356 return node; | 357 return node; |
| 357 } | 358 } |
| 358 | 359 |
| 359 PassRefPtr<MediaElementAudioSourceNode> AudioContext::createMediaElementSource(H
TMLMediaElement* mediaElement, ExceptionState& exceptionState) | 360 PassRefPtr<MediaElementAudioSourceNode> AudioContext::createMediaElementSource(H
TMLMediaElement* mediaElement, ExceptionState& exceptionState) |
| 360 { | 361 { |
| 362 ASSERT(isMainThread()); |
| 361 if (!mediaElement) { | 363 if (!mediaElement) { |
| 362 exceptionState.throwDOMException( | 364 exceptionState.throwDOMException( |
| 363 InvalidStateError, | 365 InvalidStateError, |
| 364 "invalid HTMLMedialElement."); | 366 "invalid HTMLMedialElement."); |
| 365 return nullptr; | 367 return nullptr; |
| 366 } | 368 } |
| 367 | 369 |
| 368 ASSERT(isMainThread()); | |
| 369 lazyInitialize(); | |
| 370 | |
| 371 // First check if this media element already has a source node. | 370 // First check if this media element already has a source node. |
| 372 if (mediaElement->audioSourceNode()) { | 371 if (mediaElement->audioSourceNode()) { |
| 373 exceptionState.throwDOMException( | 372 exceptionState.throwDOMException( |
| 374 InvalidStateError, | 373 InvalidStateError, |
| 375 "invalid HTMLMediaElement."); | 374 "invalid HTMLMediaElement."); |
| 376 return nullptr; | 375 return nullptr; |
| 377 } | 376 } |
| 378 | 377 |
| 379 RefPtr<MediaElementAudioSourceNode> node = MediaElementAudioSourceNode::crea
te(this, mediaElement); | 378 RefPtr<MediaElementAudioSourceNode> node = MediaElementAudioSourceNode::crea
te(this, mediaElement); |
| 380 | 379 |
| 381 mediaElement->setAudioSourceNode(node.get()); | 380 mediaElement->setAudioSourceNode(node.get()); |
| 382 | 381 |
| 383 refNode(node.get()); // context keeps reference until node is disconnected | 382 refNode(node.get()); // context keeps reference until node is disconnected |
| 384 return node; | 383 return node; |
| 385 } | 384 } |
| 386 | 385 |
| 387 PassRefPtr<MediaStreamAudioSourceNode> AudioContext::createMediaStreamSource(Med
iaStream* mediaStream, ExceptionState& exceptionState) | 386 PassRefPtr<MediaStreamAudioSourceNode> AudioContext::createMediaStreamSource(Med
iaStream* mediaStream, ExceptionState& exceptionState) |
| 388 { | 387 { |
| 388 ASSERT(isMainThread()); |
| 389 if (!mediaStream) { | 389 if (!mediaStream) { |
| 390 exceptionState.throwDOMException( | 390 exceptionState.throwDOMException( |
| 391 InvalidStateError, | 391 InvalidStateError, |
| 392 "invalid MediaStream source"); | 392 "invalid MediaStream source"); |
| 393 return nullptr; | 393 return nullptr; |
| 394 } | 394 } |
| 395 | 395 |
| 396 ASSERT(isMainThread()); | |
| 397 lazyInitialize(); | |
| 398 | |
| 399 MediaStreamTrackVector audioTracks = mediaStream->getAudioTracks(); | 396 MediaStreamTrackVector audioTracks = mediaStream->getAudioTracks(); |
| 400 if (audioTracks.isEmpty()) { | 397 if (audioTracks.isEmpty()) { |
| 401 exceptionState.throwDOMException( | 398 exceptionState.throwDOMException( |
| 402 InvalidStateError, | 399 InvalidStateError, |
| 403 "MediaStream has no audio track"); | 400 "MediaStream has no audio track"); |
| 404 return nullptr; | 401 return nullptr; |
| 405 } | 402 } |
| 406 | 403 |
| 407 // Use the first audio track in the media stream. | 404 // Use the first audio track in the media stream. |
| 408 RefPtr<MediaStreamTrack> audioTrack = audioTracks[0]; | 405 RefPtr<MediaStreamTrack> audioTrack = audioTracks[0]; |
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| 436 | 433 |
| 437 PassRefPtr<ScriptProcessorNode> AudioContext::createScriptProcessor(size_t buffe
rSize, size_t numberOfInputChannels, ExceptionState& exceptionState) | 434 PassRefPtr<ScriptProcessorNode> AudioContext::createScriptProcessor(size_t buffe
rSize, size_t numberOfInputChannels, ExceptionState& exceptionState) |
| 438 { | 435 { |
| 439 // Set number of output channels to stereo by default. | 436 // Set number of output channels to stereo by default. |
| 440 return createScriptProcessor(bufferSize, numberOfInputChannels, 2, exception
State); | 437 return createScriptProcessor(bufferSize, numberOfInputChannels, 2, exception
State); |
| 441 } | 438 } |
| 442 | 439 |
| 443 PassRefPtr<ScriptProcessorNode> AudioContext::createScriptProcessor(size_t buffe
rSize, size_t numberOfInputChannels, size_t numberOfOutputChannels, ExceptionSta
te& exceptionState) | 440 PassRefPtr<ScriptProcessorNode> AudioContext::createScriptProcessor(size_t buffe
rSize, size_t numberOfInputChannels, size_t numberOfOutputChannels, ExceptionSta
te& exceptionState) |
| 444 { | 441 { |
| 445 ASSERT(isMainThread()); | 442 ASSERT(isMainThread()); |
| 446 lazyInitialize(); | |
| 447 RefPtr<ScriptProcessorNode> node = ScriptProcessorNode::create(this, m_desti
nationNode->sampleRate(), bufferSize, numberOfInputChannels, numberOfOutputChann
els); | 443 RefPtr<ScriptProcessorNode> node = ScriptProcessorNode::create(this, m_desti
nationNode->sampleRate(), bufferSize, numberOfInputChannels, numberOfOutputChann
els); |
| 448 | 444 |
| 449 if (!node.get()) { | 445 if (!node.get()) { |
| 450 if (!numberOfInputChannels && !numberOfOutputChannels) { | 446 if (!numberOfInputChannels && !numberOfOutputChannels) { |
| 451 exceptionState.throwDOMException( | 447 exceptionState.throwDOMException( |
| 452 IndexSizeError, | 448 IndexSizeError, |
| 453 "number of input channels and output channels cannot both be zer
o."); | 449 "number of input channels and output channels cannot both be zer
o."); |
| 454 } else if (numberOfInputChannels > AudioContext::maxNumberOfChannels())
{ | 450 } else if (numberOfInputChannels > AudioContext::maxNumberOfChannels())
{ |
| 455 exceptionState.throwDOMException( | 451 exceptionState.throwDOMException( |
| 456 IndexSizeError, | 452 IndexSizeError, |
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| 472 return nullptr; | 468 return nullptr; |
| 473 } | 469 } |
| 474 | 470 |
| 475 refNode(node.get()); // context keeps reference until we stop making javascr
ipt rendering callbacks | 471 refNode(node.get()); // context keeps reference until we stop making javascr
ipt rendering callbacks |
| 476 return node; | 472 return node; |
| 477 } | 473 } |
| 478 | 474 |
| 479 PassRefPtr<BiquadFilterNode> AudioContext::createBiquadFilter() | 475 PassRefPtr<BiquadFilterNode> AudioContext::createBiquadFilter() |
| 480 { | 476 { |
| 481 ASSERT(isMainThread()); | 477 ASSERT(isMainThread()); |
| 482 lazyInitialize(); | |
| 483 return BiquadFilterNode::create(this, m_destinationNode->sampleRate()); | 478 return BiquadFilterNode::create(this, m_destinationNode->sampleRate()); |
| 484 } | 479 } |
| 485 | 480 |
| 486 PassRefPtr<WaveShaperNode> AudioContext::createWaveShaper() | 481 PassRefPtr<WaveShaperNode> AudioContext::createWaveShaper() |
| 487 { | 482 { |
| 488 ASSERT(isMainThread()); | 483 ASSERT(isMainThread()); |
| 489 lazyInitialize(); | |
| 490 return WaveShaperNode::create(this); | 484 return WaveShaperNode::create(this); |
| 491 } | 485 } |
| 492 | 486 |
| 493 PassRefPtr<PannerNode> AudioContext::createPanner() | 487 PassRefPtr<PannerNode> AudioContext::createPanner() |
| 494 { | 488 { |
| 495 ASSERT(isMainThread()); | 489 ASSERT(isMainThread()); |
| 496 lazyInitialize(); | |
| 497 return PannerNode::create(this, m_destinationNode->sampleRate()); | 490 return PannerNode::create(this, m_destinationNode->sampleRate()); |
| 498 } | 491 } |
| 499 | 492 |
| 500 PassRefPtr<ConvolverNode> AudioContext::createConvolver() | 493 PassRefPtr<ConvolverNode> AudioContext::createConvolver() |
| 501 { | 494 { |
| 502 ASSERT(isMainThread()); | 495 ASSERT(isMainThread()); |
| 503 lazyInitialize(); | |
| 504 return ConvolverNode::create(this, m_destinationNode->sampleRate()); | 496 return ConvolverNode::create(this, m_destinationNode->sampleRate()); |
| 505 } | 497 } |
| 506 | 498 |
| 507 PassRefPtr<DynamicsCompressorNode> AudioContext::createDynamicsCompressor() | 499 PassRefPtr<DynamicsCompressorNode> AudioContext::createDynamicsCompressor() |
| 508 { | 500 { |
| 509 ASSERT(isMainThread()); | 501 ASSERT(isMainThread()); |
| 510 lazyInitialize(); | |
| 511 return DynamicsCompressorNode::create(this, m_destinationNode->sampleRate())
; | 502 return DynamicsCompressorNode::create(this, m_destinationNode->sampleRate())
; |
| 512 } | 503 } |
| 513 | 504 |
| 514 PassRefPtr<AnalyserNode> AudioContext::createAnalyser() | 505 PassRefPtr<AnalyserNode> AudioContext::createAnalyser() |
| 515 { | 506 { |
| 516 ASSERT(isMainThread()); | 507 ASSERT(isMainThread()); |
| 517 lazyInitialize(); | |
| 518 return AnalyserNode::create(this, m_destinationNode->sampleRate()); | 508 return AnalyserNode::create(this, m_destinationNode->sampleRate()); |
| 519 } | 509 } |
| 520 | 510 |
| 521 PassRefPtr<GainNode> AudioContext::createGain() | 511 PassRefPtr<GainNode> AudioContext::createGain() |
| 522 { | 512 { |
| 523 ASSERT(isMainThread()); | 513 ASSERT(isMainThread()); |
| 524 lazyInitialize(); | |
| 525 return GainNode::create(this, m_destinationNode->sampleRate()); | 514 return GainNode::create(this, m_destinationNode->sampleRate()); |
| 526 } | 515 } |
| 527 | 516 |
| 528 PassRefPtr<DelayNode> AudioContext::createDelay(ExceptionState& exceptionState) | 517 PassRefPtr<DelayNode> AudioContext::createDelay(ExceptionState& exceptionState) |
| 529 { | 518 { |
| 530 const double defaultMaxDelayTime = 1; | 519 const double defaultMaxDelayTime = 1; |
| 531 return createDelay(defaultMaxDelayTime, exceptionState); | 520 return createDelay(defaultMaxDelayTime, exceptionState); |
| 532 } | 521 } |
| 533 | 522 |
| 534 PassRefPtr<DelayNode> AudioContext::createDelay(double maxDelayTime, ExceptionSt
ate& exceptionState) | 523 PassRefPtr<DelayNode> AudioContext::createDelay(double maxDelayTime, ExceptionSt
ate& exceptionState) |
| 535 { | 524 { |
| 536 ASSERT(isMainThread()); | 525 ASSERT(isMainThread()); |
| 537 lazyInitialize(); | |
| 538 RefPtr<DelayNode> node = DelayNode::create(this, m_destinationNode->sampleRa
te(), maxDelayTime, exceptionState); | 526 RefPtr<DelayNode> node = DelayNode::create(this, m_destinationNode->sampleRa
te(), maxDelayTime, exceptionState); |
| 539 if (exceptionState.hadException()) | 527 if (exceptionState.hadException()) |
| 540 return nullptr; | 528 return nullptr; |
| 541 return node; | 529 return node; |
| 542 } | 530 } |
| 543 | 531 |
| 544 PassRefPtr<ChannelSplitterNode> AudioContext::createChannelSplitter(ExceptionSta
te& exceptionState) | 532 PassRefPtr<ChannelSplitterNode> AudioContext::createChannelSplitter(ExceptionSta
te& exceptionState) |
| 545 { | 533 { |
| 546 const unsigned ChannelSplitterDefaultNumberOfOutputs = 6; | 534 const unsigned ChannelSplitterDefaultNumberOfOutputs = 6; |
| 547 return createChannelSplitter(ChannelSplitterDefaultNumberOfOutputs, exceptio
nState); | 535 return createChannelSplitter(ChannelSplitterDefaultNumberOfOutputs, exceptio
nState); |
| 548 } | 536 } |
| 549 | 537 |
| 550 PassRefPtr<ChannelSplitterNode> AudioContext::createChannelSplitter(size_t numbe
rOfOutputs, ExceptionState& exceptionState) | 538 PassRefPtr<ChannelSplitterNode> AudioContext::createChannelSplitter(size_t numbe
rOfOutputs, ExceptionState& exceptionState) |
| 551 { | 539 { |
| 552 ASSERT(isMainThread()); | 540 ASSERT(isMainThread()); |
| 553 lazyInitialize(); | |
| 554 | |
| 555 RefPtr<ChannelSplitterNode> node = ChannelSplitterNode::create(this, m_desti
nationNode->sampleRate(), numberOfOutputs); | 541 RefPtr<ChannelSplitterNode> node = ChannelSplitterNode::create(this, m_desti
nationNode->sampleRate(), numberOfOutputs); |
| 556 | 542 |
| 557 if (!node.get()) { | 543 if (!node.get()) { |
| 558 exceptionState.throwDOMException( | 544 exceptionState.throwDOMException( |
| 559 IndexSizeError, | 545 IndexSizeError, |
| 560 "number of outputs (" + String::number(numberOfOutputs) | 546 "number of outputs (" + String::number(numberOfOutputs) |
| 561 + ") must be between 1 and " | 547 + ") must be between 1 and " |
| 562 + String::number(AudioContext::maxNumberOfChannels()) + "."); | 548 + String::number(AudioContext::maxNumberOfChannels()) + "."); |
| 563 return nullptr; | 549 return nullptr; |
| 564 } | 550 } |
| 565 | 551 |
| 566 return node; | 552 return node; |
| 567 } | 553 } |
| 568 | 554 |
| 569 PassRefPtr<ChannelMergerNode> AudioContext::createChannelMerger(ExceptionState&
exceptionState) | 555 PassRefPtr<ChannelMergerNode> AudioContext::createChannelMerger(ExceptionState&
exceptionState) |
| 570 { | 556 { |
| 571 const unsigned ChannelMergerDefaultNumberOfInputs = 6; | 557 const unsigned ChannelMergerDefaultNumberOfInputs = 6; |
| 572 return createChannelMerger(ChannelMergerDefaultNumberOfInputs, exceptionStat
e); | 558 return createChannelMerger(ChannelMergerDefaultNumberOfInputs, exceptionStat
e); |
| 573 } | 559 } |
| 574 | 560 |
| 575 PassRefPtr<ChannelMergerNode> AudioContext::createChannelMerger(size_t numberOfI
nputs, ExceptionState& exceptionState) | 561 PassRefPtr<ChannelMergerNode> AudioContext::createChannelMerger(size_t numberOfI
nputs, ExceptionState& exceptionState) |
| 576 { | 562 { |
| 577 ASSERT(isMainThread()); | 563 ASSERT(isMainThread()); |
| 578 lazyInitialize(); | |
| 579 | |
| 580 RefPtr<ChannelMergerNode> node = ChannelMergerNode::create(this, m_destinati
onNode->sampleRate(), numberOfInputs); | 564 RefPtr<ChannelMergerNode> node = ChannelMergerNode::create(this, m_destinati
onNode->sampleRate(), numberOfInputs); |
| 581 | 565 |
| 582 if (!node.get()) { | 566 if (!node.get()) { |
| 583 exceptionState.throwDOMException( | 567 exceptionState.throwDOMException( |
| 584 IndexSizeError, | 568 IndexSizeError, |
| 585 "number of inputs (" + String::number(numberOfInputs) | 569 "number of inputs (" + String::number(numberOfInputs) |
| 586 + ") must be between 1 and " | 570 + ") must be between 1 and " |
| 587 + String::number(AudioContext::maxNumberOfChannels()) + "."); | 571 + String::number(AudioContext::maxNumberOfChannels()) + "."); |
| 588 return nullptr; | 572 return nullptr; |
| 589 } | 573 } |
| 590 | 574 |
| 591 return node; | 575 return node; |
| 592 } | 576 } |
| 593 | 577 |
| 594 PassRefPtr<OscillatorNode> AudioContext::createOscillator() | 578 PassRefPtr<OscillatorNode> AudioContext::createOscillator() |
| 595 { | 579 { |
| 596 ASSERT(isMainThread()); | 580 ASSERT(isMainThread()); |
| 597 lazyInitialize(); | |
| 598 | |
| 599 RefPtr<OscillatorNode> node = OscillatorNode::create(this, m_destinationNode
->sampleRate()); | 581 RefPtr<OscillatorNode> node = OscillatorNode::create(this, m_destinationNode
->sampleRate()); |
| 600 | 582 |
| 601 // Because this is an AudioScheduledSourceNode, the context keeps a referenc
e until it has finished playing. | 583 // Because this is an AudioScheduledSourceNode, the context keeps a referenc
e until it has finished playing. |
| 602 // When this happens, AudioScheduledSourceNode::finish() calls AudioContext:
:notifyNodeFinishedProcessing(). | 584 // When this happens, AudioScheduledSourceNode::finish() calls AudioContext:
:notifyNodeFinishedProcessing(). |
| 603 refNode(node.get()); | 585 refNode(node.get()); |
| 604 | 586 |
| 605 return node; | 587 return node; |
| 606 } | 588 } |
| 607 | 589 |
| 608 PassRefPtr<PeriodicWave> AudioContext::createPeriodicWave(Float32Array* real, Fl
oat32Array* imag, ExceptionState& exceptionState) | 590 PassRefPtr<PeriodicWave> AudioContext::createPeriodicWave(Float32Array* real, Fl
oat32Array* imag, ExceptionState& exceptionState) |
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| 641 } | 623 } |
| 642 | 624 |
| 643 if (imag->length() > 4096) { | 625 if (imag->length() > 4096) { |
| 644 exceptionState.throwDOMException( | 626 exceptionState.throwDOMException( |
| 645 IndexSizeError, | 627 IndexSizeError, |
| 646 "length of imaginary array (" + String::number(imag->length()) | 628 "length of imaginary array (" + String::number(imag->length()) |
| 647 + ") exceeds allowed maximum of 4096"); | 629 + ") exceeds allowed maximum of 4096"); |
| 648 return nullptr; | 630 return nullptr; |
| 649 } | 631 } |
| 650 | 632 |
| 651 lazyInitialize(); | |
| 652 return PeriodicWave::create(sampleRate(), real, imag); | 633 return PeriodicWave::create(sampleRate(), real, imag); |
| 653 } | 634 } |
| 654 | 635 |
| 655 void AudioContext::notifyNodeFinishedProcessing(AudioNode* node) | 636 void AudioContext::notifyNodeFinishedProcessing(AudioNode* node) |
| 656 { | 637 { |
| 657 ASSERT(isAudioThread()); | 638 ASSERT(isAudioThread()); |
| 658 m_finishedNodes.append(node); | 639 m_finishedNodes.append(node); |
| 659 } | 640 } |
| 660 | 641 |
| 661 void AudioContext::derefFinishedSourceNodes() | 642 void AudioContext::derefFinishedSourceNodes() |
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| 1038 } | 1019 } |
| 1039 | 1020 |
| 1040 void AudioContext::decrementActiveSourceCount() | 1021 void AudioContext::decrementActiveSourceCount() |
| 1041 { | 1022 { |
| 1042 atomicDecrement(&m_activeSourceCount); | 1023 atomicDecrement(&m_activeSourceCount); |
| 1043 } | 1024 } |
| 1044 | 1025 |
| 1045 } // namespace WebCore | 1026 } // namespace WebCore |
| 1046 | 1027 |
| 1047 #endif // ENABLE(WEB_AUDIO) | 1028 #endif // ENABLE(WEB_AUDIO) |
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