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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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23 */ | 23 */ |
24 | 24 |
25 #include "config.h" | 25 #include "config.h" |
26 | 26 |
27 #if ENABLE(WEB_AUDIO) | 27 #if ENABLE(WEB_AUDIO) |
28 | 28 |
29 #include "modules/webaudio/AudioContext.h" | 29 #include "modules/webaudio/AudioContext.h" |
30 | 30 |
31 #include "bindings/core/v8/ExceptionMessages.h" | 31 #include "bindings/core/v8/ExceptionMessages.h" |
32 #include "bindings/core/v8/ExceptionState.h" | 32 #include "bindings/core/v8/ExceptionState.h" |
33 #include "bindings/core/v8/ScriptState.h" | |
34 #include "core/dom/DOMException.h" | |
33 #include "core/dom/Document.h" | 35 #include "core/dom/Document.h" |
34 #include "core/dom/ExceptionCode.h" | 36 #include "core/dom/ExceptionCode.h" |
35 #include "core/html/HTMLMediaElement.h" | 37 #include "core/html/HTMLMediaElement.h" |
36 #include "core/inspector/ScriptCallStack.h" | 38 #include "core/inspector/ScriptCallStack.h" |
37 #include "platform/audio/FFTFrame.h" | 39 #include "platform/audio/FFTFrame.h" |
38 #include "platform/audio/HRTFPanner.h" | 40 #include "platform/audio/HRTFPanner.h" |
39 #include "modules/mediastream/MediaStream.h" | 41 #include "modules/mediastream/MediaStream.h" |
40 #include "modules/webaudio/AnalyserNode.h" | 42 #include "modules/webaudio/AnalyserNode.h" |
41 #include "modules/webaudio/AudioBuffer.h" | 43 #include "modules/webaudio/AudioBuffer.h" |
42 #include "modules/webaudio/AudioBufferCallback.h" | 44 #include "modules/webaudio/AudioBufferCallback.h" |
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94 return audioContext; | 96 return audioContext; |
95 } | 97 } |
96 | 98 |
97 // Constructor for rendering to the audio hardware. | 99 // Constructor for rendering to the audio hardware. |
98 AudioContext::AudioContext(Document* document) | 100 AudioContext::AudioContext(Document* document) |
99 : ActiveDOMObject(document) | 101 : ActiveDOMObject(document) |
100 , m_isStopScheduled(false) | 102 , m_isStopScheduled(false) |
101 , m_isCleared(false) | 103 , m_isCleared(false) |
102 , m_isInitialized(false) | 104 , m_isInitialized(false) |
103 , m_destinationNode(nullptr) | 105 , m_destinationNode(nullptr) |
106 , m_isResolvingResumePromises(false) | |
104 , m_automaticPullNodesNeedUpdating(false) | 107 , m_automaticPullNodesNeedUpdating(false) |
105 , m_connectionCount(0) | 108 , m_connectionCount(0) |
106 , m_audioThread(0) | 109 , m_audioThread(0) |
107 , m_isOfflineContext(false) | 110 , m_isOfflineContext(false) |
111 , m_contextState(Paused) | |
108 { | 112 { |
109 m_destinationNode = DefaultAudioDestinationNode::create(this); | 113 m_destinationNode = DefaultAudioDestinationNode::create(this); |
110 | 114 |
111 initialize(); | 115 initialize(); |
112 #if DEBUG_AUDIONODE_REFERENCES | 116 #if DEBUG_AUDIONODE_REFERENCES |
113 fprintf(stderr, "%p: AudioContext::AudioContext() #%u\n", this, AudioContext ::s_hardwareContextCount); | 117 fprintf(stderr, "%p: AudioContext::AudioContext() #%u\n", this, AudioContext ::s_hardwareContextCount); |
114 #endif | 118 #endif |
115 } | 119 } |
116 | 120 |
117 // Constructor for offline (non-realtime) rendering. | 121 // Constructor for offline (non-realtime) rendering. |
118 AudioContext::AudioContext(Document* document, unsigned numberOfChannels, size_t numberOfFrames, float sampleRate) | 122 AudioContext::AudioContext(Document* document, unsigned numberOfChannels, size_t numberOfFrames, float sampleRate) |
119 : ActiveDOMObject(document) | 123 : ActiveDOMObject(document) |
120 , m_isStopScheduled(false) | 124 , m_isStopScheduled(false) |
121 , m_isCleared(false) | 125 , m_isCleared(false) |
122 , m_isInitialized(false) | 126 , m_isInitialized(false) |
123 , m_destinationNode(nullptr) | 127 , m_destinationNode(nullptr) |
128 , m_isResolvingResumePromises(false) | |
124 , m_automaticPullNodesNeedUpdating(false) | 129 , m_automaticPullNodesNeedUpdating(false) |
125 , m_connectionCount(0) | 130 , m_connectionCount(0) |
126 , m_audioThread(0) | 131 , m_audioThread(0) |
127 , m_isOfflineContext(true) | 132 , m_isOfflineContext(true) |
133 , m_contextState(Paused) | |
128 { | 134 { |
129 // Create a new destination for offline rendering. | 135 // Create a new destination for offline rendering. |
130 m_renderTarget = AudioBuffer::create(numberOfChannels, numberOfFrames, sampl eRate); | 136 m_renderTarget = AudioBuffer::create(numberOfChannels, numberOfFrames, sampl eRate); |
131 if (m_renderTarget.get()) | 137 if (m_renderTarget.get()) |
132 m_destinationNode = OfflineAudioDestinationNode::create(this, m_renderTa rget.get()); | 138 m_destinationNode = OfflineAudioDestinationNode::create(this, m_renderTa rget.get()); |
133 | 139 |
134 initialize(); | 140 initialize(); |
135 } | 141 } |
136 | 142 |
137 AudioContext::~AudioContext() | 143 AudioContext::~AudioContext() |
haraken
2014/10/16 01:13:43
Can we add ASSERT(!m_resumePromises.size())?
Raymond Toy
2014/10/16 17:56:31
Done.
| |
138 { | 144 { |
139 #if DEBUG_AUDIONODE_REFERENCES | 145 #if DEBUG_AUDIONODE_REFERENCES |
140 fprintf(stderr, "%p: AudioContext::~AudioContext()\n", this); | 146 fprintf(stderr, "%p: AudioContext::~AudioContext()\n", this); |
141 #endif | 147 #endif |
142 // AudioNodes keep a reference to their context, so there should be no way t o be in the destructor if there are still AudioNodes around. | 148 // AudioNodes keep a reference to their context, so there should be no way t o be in the destructor if there are still AudioNodes around. |
143 ASSERT(!m_isInitialized); | 149 ASSERT(!m_isInitialized); |
144 ASSERT(!m_referencedNodes.size()); | 150 ASSERT(!m_referencedNodes.size()); |
145 ASSERT(!m_finishedNodes.size()); | 151 ASSERT(!m_finishedNodes.size()); |
146 ASSERT(!m_automaticPullNodes.size()); | 152 ASSERT(!m_automaticPullNodes.size()); |
147 if (m_automaticPullNodesNeedUpdating) | 153 if (m_automaticPullNodesNeedUpdating) |
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158 m_listener = AudioListener::create(); | 164 m_listener = AudioListener::create(); |
159 | 165 |
160 if (m_destinationNode.get()) { | 166 if (m_destinationNode.get()) { |
161 m_destinationNode->initialize(); | 167 m_destinationNode->initialize(); |
162 | 168 |
163 if (!isOfflineContext()) { | 169 if (!isOfflineContext()) { |
164 // This starts the audio thread. The destination node's provideInput () method will now be called repeatedly to render audio. | 170 // This starts the audio thread. The destination node's provideInput () method will now be called repeatedly to render audio. |
165 // Each time provideInput() is called, a portion of the audio stream is rendered. Let's call this time period a "render quantum". | 171 // Each time provideInput() is called, a portion of the audio stream is rendered. Let's call this time period a "render quantum". |
166 // NOTE: for now default AudioContext does not need an explicit star tRendering() call from JavaScript. | 172 // NOTE: for now default AudioContext does not need an explicit star tRendering() call from JavaScript. |
167 // We may want to consider requiring it for symmetry with OfflineAud ioContext. | 173 // We may want to consider requiring it for symmetry with OfflineAud ioContext. |
168 m_destinationNode->startRendering(); | 174 startRendering(); |
169 ++s_hardwareContextCount; | 175 ++s_hardwareContextCount; |
170 } | 176 } |
171 | 177 |
172 m_isInitialized = true; | 178 m_isInitialized = true; |
173 } | 179 } |
174 } | 180 } |
175 | 181 |
176 void AudioContext::clear() | 182 void AudioContext::clear() |
177 { | 183 { |
178 // We need to run disposers before destructing m_contextGraphMutex. | 184 // We need to run disposers before destructing m_contextGraphMutex. |
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524 exceptionState.throwDOMException( | 530 exceptionState.throwDOMException( |
525 IndexSizeError, | 531 IndexSizeError, |
526 "length of imaginary array (" + String::number(imag->length()) | 532 "length of imaginary array (" + String::number(imag->length()) |
527 + ") exceeds allowed maximum of 4096"); | 533 + ") exceeds allowed maximum of 4096"); |
528 return 0; | 534 return 0; |
529 } | 535 } |
530 | 536 |
531 return PeriodicWave::create(sampleRate(), real, imag); | 537 return PeriodicWave::create(sampleRate(), real, imag); |
532 } | 538 } |
533 | 539 |
540 String AudioContext::state() const | |
541 { | |
542 switch (m_contextState) { | |
543 case Paused: | |
544 return "paused"; | |
545 case Running: | |
546 return "running"; | |
547 case Released: | |
548 return "released"; | |
549 } | |
550 ASSERT_NOT_REACHED(); | |
551 return ""; | |
552 } | |
553 | |
554 void AudioContext::setContextState(AudioContextState newState) | |
555 { | |
556 // Validate the transitions | |
557 switch (newState) { | |
558 case Paused: | |
559 ASSERT(m_contextState == Running); | |
560 break; | |
561 case Running: | |
562 ASSERT(m_contextState == Paused); | |
563 break; | |
564 case Released: | |
565 ASSERT(m_contextState != Released); | |
566 break; | |
567 } | |
568 | |
569 m_contextState = newState; | |
570 } | |
571 | |
572 void AudioContext::suspendContext(ExceptionState& exceptionState) | |
573 { | |
574 ASSERT(isMainThread()); | |
575 AutoLocker locker(this); | |
576 | |
577 if (m_contextState == Released) { | |
578 exceptionState.throwDOMException( | |
579 InvalidStateError, | |
580 "cannot suspend an AudioContext that has been released"); | |
581 return; | |
582 } | |
583 | |
584 if (m_destinationNode && !isOfflineContext()) { | |
585 stopRendering(); | |
586 } | |
587 } | |
588 | |
589 ScriptPromise AudioContext::resumeContext(ScriptState* scriptState) | |
590 { | |
591 ASSERT(isMainThread()); | |
592 AutoLocker locker(this); | |
593 | |
594 RefPtr<ScriptPromiseResolver> resolver = ScriptPromiseResolver::create(scrip tState); | |
595 | |
596 ScriptPromise promise = resolver->promise(); | |
597 | |
598 if (isOfflineContext()) { | |
599 // For offline context, resolve now, but reject if the context has been released. | |
600 if (m_contextState == Released) { | |
601 resolver->reject( | |
602 DOMException::create(InvalidStateError, "Cannot resume a context that has been released")); | |
603 } else { | |
604 resolver->resolve(); | |
605 } | |
606 } else { | |
607 // Restart the destination node to pull on the audio graph. | |
608 if (m_destinationNode) { | |
609 startRendering(); | |
610 } | |
611 | |
612 // Save the promise which will get resolved when the destination node st arts pulling on the | |
613 // graph again. | |
614 m_resumePromises.append(resolver); | |
615 } | |
616 | |
617 return promise; | |
618 } | |
619 | |
534 void AudioContext::notifyNodeFinishedProcessing(AudioNode* node) | 620 void AudioContext::notifyNodeFinishedProcessing(AudioNode* node) |
535 { | 621 { |
536 ASSERT(isAudioThread()); | 622 ASSERT(isAudioThread()); |
537 m_finishedNodes.append(node); | 623 m_finishedNodes.append(node); |
538 } | 624 } |
539 | 625 |
540 void AudioContext::derefFinishedSourceNodes() | 626 void AudioContext::derefFinishedSourceNodes() |
541 { | 627 { |
542 ASSERT(isGraphOwner()); | 628 ASSERT(isGraphOwner()); |
543 ASSERT(isAudioThread()); | 629 ASSERT(isAudioThread()); |
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630 // It's OK if the tryLock() fails, we'll just take slightly longer to pick u p the changes. | 716 // It's OK if the tryLock() fails, we'll just take slightly longer to pick u p the changes. |
631 if (tryLock()) { | 717 if (tryLock()) { |
632 // Update the channel count mode. | 718 // Update the channel count mode. |
633 updateChangedChannelCountMode(); | 719 updateChangedChannelCountMode(); |
634 | 720 |
635 // Fixup the state of any dirty AudioSummingJunctions and AudioNodeOutpu ts. | 721 // Fixup the state of any dirty AudioSummingJunctions and AudioNodeOutpu ts. |
636 handleDirtyAudioSummingJunctions(); | 722 handleDirtyAudioSummingJunctions(); |
637 handleDirtyAudioNodeOutputs(); | 723 handleDirtyAudioNodeOutputs(); |
638 | 724 |
639 updateAutomaticPullNodes(); | 725 updateAutomaticPullNodes(); |
726 resolvePromisesForResume(); | |
727 | |
640 unlock(); | 728 unlock(); |
641 } | 729 } |
642 } | 730 } |
643 | 731 |
644 void AudioContext::handlePostRenderTasks() | 732 void AudioContext::handlePostRenderTasks() |
645 { | 733 { |
646 ASSERT(isAudioThread()); | 734 ASSERT(isAudioThread()); |
647 | 735 |
648 // Must use a tryLock() here too. Don't worry, the lock will very rarely be contended and this method is called frequently. | 736 // Must use a tryLock() here too. Don't worry, the lock will very rarely be contended and this method is called frequently. |
649 // The worst that can happen is that there will be some nodes which will tak e slightly longer than usual to be deleted or removed | 737 // The worst that can happen is that there will be some nodes which will tak e slightly longer than usual to be deleted or removed |
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795 } | 883 } |
796 | 884 |
797 void AudioContext::processAutomaticPullNodes(size_t framesToProcess) | 885 void AudioContext::processAutomaticPullNodes(size_t framesToProcess) |
798 { | 886 { |
799 ASSERT(isAudioThread()); | 887 ASSERT(isAudioThread()); |
800 | 888 |
801 for (unsigned i = 0; i < m_renderingAutomaticPullNodes.size(); ++i) | 889 for (unsigned i = 0; i < m_renderingAutomaticPullNodes.size(); ++i) |
802 m_renderingAutomaticPullNodes[i]->processIfNecessary(framesToProcess); | 890 m_renderingAutomaticPullNodes[i]->processIfNecessary(framesToProcess); |
803 } | 891 } |
804 | 892 |
893 void AudioContext::resolvePromisesForResumeOnMainThread() | |
894 { | |
895 ASSERT(isMainThread()); | |
896 AutoLocker locker(this); | |
897 | |
898 for (unsigned k = 0; k < m_resumePromises.size(); ++k) { | |
899 if (m_contextState == Released) { | |
900 m_resumePromises[k]->reject( | |
901 DOMException::create(InvalidStateError, "Cannot resume a context that has been released")); | |
902 } else { | |
903 m_resumePromises[k]->resolve(); | |
904 } | |
905 } | |
906 | |
907 m_resumePromises.clear(); | |
908 m_isResolvingResumePromises = false; | |
909 } | |
910 | |
911 void AudioContext::resolvePromisesForResume() | |
912 { | |
913 // This runs inside the AudioContext's lock when handling pre-render tasks. | |
914 ASSERT(isAudioThread()); | |
915 ASSERT(isGraphOwner()); | |
916 | |
917 // Resolve any pending promises created by resume(). Only do this we if have n't already started | |
918 // resolving these promises. This gets called very often and it takes some t ime to resolve the | |
919 // promises in the main thread. | |
920 if (!m_isResolvingResumePromises && m_resumePromises.size() > 0) { | |
921 m_isResolvingResumePromises = true; | |
922 callOnMainThread(bind(&AudioContext::resolvePromisesForResumeOnMainThrea d, this)); | |
923 } | |
924 } | |
925 | |
805 const AtomicString& AudioContext::interfaceName() const | 926 const AtomicString& AudioContext::interfaceName() const |
806 { | 927 { |
807 return EventTargetNames::AudioContext; | 928 return EventTargetNames::AudioContext; |
808 } | 929 } |
809 | 930 |
810 ExecutionContext* AudioContext::executionContext() const | 931 ExecutionContext* AudioContext::executionContext() const |
811 { | 932 { |
812 return m_isStopScheduled ? 0 : ActiveDOMObject::executionContext(); | 933 return m_isStopScheduled ? 0 : ActiveDOMObject::executionContext(); |
813 } | 934 } |
814 | 935 |
815 void AudioContext::startRendering() | 936 void AudioContext::startRendering() |
816 { | 937 { |
817 destination()->startRendering(); | 938 ASSERT(isMainThread()); |
haraken
2014/10/16 01:13:43
Shall we add ASSERT(!isOfflineContext())?
haraken
2014/10/16 01:13:43
Shall we add ASSERT(m_destinationNode)?
Raymond Toy
2014/10/16 17:56:31
Done.
Raymond Toy
2014/10/16 17:56:31
Actually no. startRendering is called for both on
| |
939 | |
940 if (m_contextState == Paused) { | |
941 destination()->startRendering(); | |
942 setContextState(Running); | |
943 } | |
944 } | |
945 | |
946 void AudioContext::stopRendering() | |
947 { | |
948 ASSERT(isMainThread()); | |
949 ASSERT(m_destinationNode); | |
950 ASSERT(!isOfflineContext()); | |
951 | |
952 if (m_contextState == Running) { | |
953 destination()->stopRendering(); | |
954 setContextState(Paused); | |
955 } | |
818 } | 956 } |
819 | 957 |
820 void AudioContext::fireCompletionEvent() | 958 void AudioContext::fireCompletionEvent() |
821 { | 959 { |
822 ASSERT(isMainThread()); | 960 ASSERT(isMainThread()); |
823 if (!isMainThread()) | 961 if (!isMainThread()) |
824 return; | 962 return; |
825 | 963 |
826 AudioBuffer* renderedBuffer = m_renderTarget.get(); | 964 AudioBuffer* renderedBuffer = m_renderTarget.get(); |
827 | 965 |
966 setContextState(Released); | |
967 | |
828 ASSERT(renderedBuffer); | 968 ASSERT(renderedBuffer); |
829 if (!renderedBuffer) | 969 if (!renderedBuffer) |
830 return; | 970 return; |
831 | 971 |
832 // Avoid firing the event if the document has already gone away. | 972 // Avoid firing the event if the document has already gone away. |
833 if (executionContext()) { | 973 if (executionContext()) { |
834 // Call the offline rendering completion event listener. | 974 // Call the offline rendering completion event listener. |
835 dispatchEvent(OfflineAudioCompletionEvent::create(renderedBuffer)); | 975 dispatchEvent(OfflineAudioCompletionEvent::create(renderedBuffer)); |
836 } | 976 } |
837 } | 977 } |
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867 | 1007 |
868 for (HashSet<AudioNode*>::iterator k = m_deferredCountModeChange.begin(); k != m_deferredCountModeChange.end(); ++k) | 1008 for (HashSet<AudioNode*>::iterator k = m_deferredCountModeChange.begin(); k != m_deferredCountModeChange.end(); ++k) |
869 (*k)->updateChannelCountMode(); | 1009 (*k)->updateChannelCountMode(); |
870 | 1010 |
871 m_deferredCountModeChange.clear(); | 1011 m_deferredCountModeChange.clear(); |
872 } | 1012 } |
873 | 1013 |
874 } // namespace blink | 1014 } // namespace blink |
875 | 1015 |
876 #endif // ENABLE(WEB_AUDIO) | 1016 #endif // ENABLE(WEB_AUDIO) |
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