Diff
Modified: trunk/Source/WebCore/ChangeLog (277552 => 277553)
--- trunk/Source/WebCore/ChangeLog 2021-05-15 23:55:58 UTC (rev 277552)
+++ trunk/Source/WebCore/ChangeLog 2021-05-16 01:05:18 UTC (rev 277553)
@@ -1,3 +1,102 @@
+2021-05-15 Chris Dumez <[email protected]>
+
+ Clean up BaseAudioContext now that legacy/prefixed WebAudio is gone
+ https://bugs.webkit.org/show_bug.cgi?id=225843
+
+ Reviewed by Darin Adler.
+
+ The BaseAudioContext class hierarchy used to be a bit complicated when
+ we supposed legacy WebAudio because OfflineAudioContext would subclass
+ BaseAudioContext directly, while WebKitOfflineAudioContext would
+ subclass BaseAudioContext via AudioContext. The class hierarchy is now
+ a lot simpler, BaseAudioContext is the base class and it has exactly
+ 2 subclasses: OfflineAudioContext and AudioContext (for real-time
+ rendering). Now that the legacy WebAudio code is gone, this patch
+ cleans up BaseAudioContext and moves as much code as possible to its
+ subclasses (OfflineAudioContext & AudioContext).
+
+ * Modules/webaudio/AudioBuffer.cpp:
+ (WebCore::AudioBuffer::create):
+ * Modules/webaudio/AudioBufferSourceNode.cpp:
+ (WebCore::AudioBufferSourceNode::setBuffer):
+ * Modules/webaudio/AudioContext.cpp:
+ (WebCore::AudioContext::create):
+ (WebCore::AudioContext::AudioContext):
+ (WebCore::AudioContext::uninitialize):
+ (WebCore::AudioContext::lazyInitialize):
+ (WebCore::AudioContext::activeDOMObjectName const):
+ * Modules/webaudio/AudioContext.h:
+ (isType):
+ * Modules/webaudio/AudioContextState.h:
+ * Modules/webaudio/AudioNode.cpp:
+ (WebCore::AudioNode::setChannelCount):
+ * Modules/webaudio/AudioNodeOutput.cpp:
+ (WebCore::AudioNodeOutput::AudioNodeOutput):
+ (WebCore::AudioNodeOutput::setNumberOfChannels):
+ * Modules/webaudio/AudioWorkletNode.cpp:
+ (WebCore::AudioWorkletNode::create):
+ * Modules/webaudio/BaseAudioContext.cpp:
+ (WebCore::generateContextID):
+ (WebCore::BaseAudioContext::BaseAudioContext):
+ (WebCore::BaseAudioContext::lazyInitialize):
+ (WebCore::BaseAudioContext::uninitialize):
+ (WebCore::BaseAudioContext::stop):
+ (WebCore::BaseAudioContext::createScriptProcessor):
+ (WebCore::BaseAudioContext::derefFinishedSourceNodes):
+ (WebCore::BaseAudioContext::lockInternal):
+ (WebCore::BaseAudioContext::tryLock):
+ (WebCore::BaseAudioContext::unlock):
+ (WebCore::BaseAudioContext::handlePostRenderTasks):
+ (WebCore::BaseAudioContext::deleteMarkedNodes):
+ (WebCore::BaseAudioContext::updateAutomaticPullNodes):
+ (WebCore::BaseAudioContext::postTask):
+ (WebCore::BaseAudioContext::workletIsReady):
+ * Modules/webaudio/BaseAudioContext.h:
+ (WebCore::BaseAudioContext::isInitialized const):
+ (WebCore::BaseAudioContext::currentSampleFrame const):
+ (WebCore::BaseAudioContext::currentTime const):
+ (WebCore::BaseAudioContext::sampleRate const):
+ (WebCore::BaseAudioContext::listener):
+ (WebCore::BaseAudioContext::incrementConnectionCount):
+ (WebCore::BaseAudioContext::isAudioThread const):
+ (WebCore::BaseAudioContext::isAudioThreadFinished const):
+ (WebCore::BaseAudioContext::isGraphOwner const):
+ * Modules/webaudio/ChannelMergerNode.cpp:
+ (WebCore::ChannelMergerNode::create):
+ * Modules/webaudio/ChannelSplitterNode.cpp:
+ (WebCore::ChannelSplitterNode::create):
+ * Modules/webaudio/DefaultAudioDestinationNode.cpp:
+ (WebCore::DefaultAudioDestinationNode::DefaultAudioDestinationNode):
+ (WebCore::DefaultAudioDestinationNode::context):
+ (WebCore::DefaultAudioDestinationNode::context const):
+ * Modules/webaudio/DefaultAudioDestinationNode.h:
+ * Modules/webaudio/MediaElementAudioSourceNode.cpp:
+ (WebCore::MediaElementAudioSourceNode::setFormat):
+ * Modules/webaudio/MediaStreamAudioSourceNode.cpp:
+ (WebCore::MediaStreamAudioSourceNode::setFormat):
+ * Modules/webaudio/OfflineAudioContext.cpp:
+ (WebCore::OfflineAudioContext::OfflineAudioContext):
+ (WebCore::OfflineAudioContext::create):
+ (WebCore::OfflineAudioContext::uninitialize):
+ (WebCore::OfflineAudioContext::activeDOMObjectName const):
+ (WebCore::OfflineAudioContext::startRendering):
+ (WebCore::OfflineAudioContext::suspendRendering):
+ (WebCore::OfflineAudioContext::resumeRendering):
+ (WebCore::OfflineAudioContext::didSuspendRendering):
+ (WebCore::OfflineAudioContext::finishedRendering):
+ (WebCore::OfflineAudioContext::settleRenderingPromise):
+ (WebCore::OfflineAudioContext::dispatchEvent):
+ * Modules/webaudio/OfflineAudioContext.h:
+ (isType):
+ * Modules/webaudio/OfflineAudioContext.idl:
+ * Modules/webaudio/OfflineAudioDestinationNode.cpp:
+ (WebCore::OfflineAudioDestinationNode::OfflineAudioDestinationNode):
+ (WebCore::OfflineAudioDestinationNode::context):
+ (WebCore::OfflineAudioDestinationNode::context const):
+ * Modules/webaudio/OfflineAudioDestinationNode.h:
+ * Modules/webaudio/ScriptProcessorNode.cpp:
+ (WebCore::ScriptProcessorNode::ScriptProcessorNode):
+
2021-05-15 Alan Bujtas <[email protected]>
[LFC] Move inline formatting geometry to its own class
Modified: trunk/Source/WebCore/Modules/webaudio/AudioBuffer.cpp (277552 => 277553)
--- trunk/Source/WebCore/Modules/webaudio/AudioBuffer.cpp 2021-05-15 23:55:58 UTC (rev 277552)
+++ trunk/Source/WebCore/Modules/webaudio/AudioBuffer.cpp 2021-05-16 01:05:18 UTC (rev 277553)
@@ -42,7 +42,7 @@
RefPtr<AudioBuffer> AudioBuffer::create(unsigned numberOfChannels, size_t numberOfFrames, float sampleRate, LegacyPreventDetaching preventDetaching)
{
- if (!BaseAudioContext::isSupportedSampleRate(sampleRate) || !numberOfChannels || numberOfChannels > AudioContext::maxNumberOfChannels() || !numberOfFrames)
+ if (!BaseAudioContext::isSupportedSampleRate(sampleRate) || !numberOfChannels || numberOfChannels > AudioContext::maxNumberOfChannels || !numberOfFrames)
return nullptr;
auto buffer = adoptRef(*new AudioBuffer(numberOfChannels, numberOfFrames, sampleRate, preventDetaching));
@@ -57,7 +57,7 @@
if (!options.numberOfChannels)
return Exception { NotSupportedError, "Number of channels cannot be 0."_s };
- if (options.numberOfChannels > AudioContext::maxNumberOfChannels())
+ if (options.numberOfChannels > AudioContext::maxNumberOfChannels)
return Exception { NotSupportedError, "Number of channels cannot be more than max supported."_s };
if (!options.length)
Modified: trunk/Source/WebCore/Modules/webaudio/AudioBufferSourceNode.cpp (277552 => 277553)
--- trunk/Source/WebCore/Modules/webaudio/AudioBufferSourceNode.cpp 2021-05-15 23:55:58 UTC (rev 277552)
+++ trunk/Source/WebCore/Modules/webaudio/AudioBufferSourceNode.cpp 2021-05-16 01:05:18 UTC (rev 277553)
@@ -426,7 +426,7 @@
// Do any necesssary re-configuration to the buffer's number of channels.
unsigned numberOfChannels = buffer->numberOfChannels();
- ASSERT(numberOfChannels <= AudioContext::maxNumberOfChannels());
+ ASSERT(numberOfChannels <= AudioContext::maxNumberOfChannels);
output(0)->setNumberOfChannels(numberOfChannels);
Modified: trunk/Source/WebCore/Modules/webaudio/AudioContext.cpp (277552 => 277553)
--- trunk/Source/WebCore/Modules/webaudio/AudioContext.cpp 2021-05-15 23:55:58 UTC (rev 277552)
+++ trunk/Source/WebCore/Modules/webaudio/AudioContext.cpp 2021-05-16 01:05:18 UTC (rev 277553)
@@ -1,6 +1,6 @@
/*
* Copyright (C) 2010 Google Inc. All rights reserved.
- * Copyright (C) 2016 Apple Inc. All rights reserved.
+ * Copyright (C) 2016-2021 Apple Inc. All rights reserved.
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions
@@ -28,6 +28,7 @@
#if ENABLE(WEB_AUDIO)
#include "AudioContext.h"
+#include "AudioContextOptions.h"
#include "AudioTimestamp.h"
#include "DOMWindow.h"
#include "JSDOMPromiseDeferred.h"
@@ -63,6 +64,10 @@
WTF_MAKE_ISO_ALLOCATED_IMPL(AudioContext);
+#if OS(WINDOWS)
+static unsigned hardwareContextCount;
+#endif
+
static Optional<float>& defaultSampleRateForTesting()
{
static Optional<float> sampleRate;
@@ -85,7 +90,7 @@
{
ASSERT(isMainThread());
#if OS(WINDOWS)
- if (s_hardwareContextCount >= maxHardwareContexts)
+ if (hardwareContextCount >= maxHardwareContexts)
return Exception { QuotaExceededError, "Reached maximum number of hardware contexts on this platform"_s };
#endif
@@ -105,11 +110,15 @@
return audioContext;
}
-// Constructor for rendering to the audio hardware.
AudioContext::AudioContext(Document& document, const AudioContextOptions& contextOptions)
- : BaseAudioContext(document, contextOptions)
+ : BaseAudioContext(document)
+ , m_destinationNode(makeUniqueRef<DefaultAudioDestinationNode>(*this, contextOptions.sampleRate))
, m_mediaSession(PlatformMediaSession::create(PlatformMediaSessionManager::sharedManager(), *this))
{
+ // According to spec AudioContext must die only after page navigate.
+ // Lets mark it as ActiveDOMObject with pending activity and unmark it in clear method.
+ setPendingActivity();
+
constructCommon();
// Initialize the destination node's muted state to match the page's current muted state.
@@ -117,6 +126,14 @@
document.addAudioProducer(*this);
document.registerForVisibilityStateChangedCallbacks(*this);
+
+ // Unlike OfflineAudioContext, AudioContext does not require calling resume() to start rendering.
+ // Lazy initialization starts rendering so we schedule a task here to make sure lazy initialization
+ // ends up happening, even if no audio node gets constructed.
+ postTask([this] {
+ if (!isStopped())
+ lazyInitialize();
+ });
}
void AudioContext::constructCommon()
@@ -140,6 +157,21 @@
}
}
+void AudioContext::uninitialize()
+{
+ if (!isInitialized())
+ return;
+
+ BaseAudioContext::uninitialize();
+
+#if OS(WINDOWS)
+ ASSERT(hardwareContextCount);
+ --hardwareContextCount;
+#endif
+
+ setState(State::Closed);
+}
+
double AudioContext::baseLatency()
{
lazyInitialize();
@@ -186,16 +218,6 @@
});
}
-DefaultAudioDestinationNode& AudioContext::destination()
-{
- return static_cast<DefaultAudioDestinationNode&>(BaseAudioContext::destination());
-}
-
-const DefaultAudioDestinationNode& AudioContext::destination() const
-{
- return static_cast<const DefaultAudioDestinationNode&>(BaseAudioContext::destination());
-}
-
void AudioContext::suspendRendering(DOMPromiseDeferred<void>&& promise)
{
if (isOfflineContext()) {
@@ -309,7 +331,9 @@
// NOTE: for now default AudioContext does not need an explicit startRendering() call from _javascript_.
// We may want to consider requiring it for symmetry with OfflineAudioContext.
startRendering();
- ++s_hardwareContextCount;
+#if OS(WINDOWS)
+ ++hardwareContextCount;
+#endif
}
}
}
@@ -435,6 +459,11 @@
document()->updateIsPlayingMedia();
}
+const char* AudioContext::activeDOMObjectName() const
+{
+ return "AudioContext";
+}
+
void AudioContext::suspendPlayback()
{
if (state() == State::Closed || !isInitialized())
Modified: trunk/Source/WebCore/Modules/webaudio/AudioContext.h (277552 => 277553)
--- trunk/Source/WebCore/Modules/webaudio/AudioContext.h 2021-05-15 23:55:58 UTC (rev 277552)
+++ trunk/Source/WebCore/Modules/webaudio/AudioContext.h 2021-05-16 01:05:18 UTC (rev 277553)
@@ -1,6 +1,6 @@
/*
* Copyright (C) 2010 Google Inc. All rights reserved.
- * Copyright (C) 2016-2020 Apple Inc. All rights reserved.
+ * Copyright (C) 2016-2021 Apple Inc. All rights reserved.
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions
@@ -25,7 +25,6 @@
#pragma once
-#include "AudioContextOptions.h"
#include "BaseAudioContext.h"
#include "DefaultAudioDestinationNode.h"
#include "MediaCanStartListener.h"
@@ -32,11 +31,17 @@
#include "MediaProducer.h"
#include "PlatformMediaSession.h"
#include "VisibilityChangeClient.h"
+#include <wtf/UniqueRef.h>
namespace WebCore {
class DOMWindow;
+class HTMLMediaElement;
+class MediaStream;
+class MediaStreamAudioDestinationNode;
+class MediaStreamAudioSourceNode;
+struct AudioContextOptions;
struct AudioTimestamp;
class AudioContext final
@@ -48,7 +53,7 @@
WTF_MAKE_ISO_ALLOCATED(AudioContext);
public:
// Create an AudioContext for rendering to the audio hardware.
- static ExceptionOr<Ref<AudioContext>> create(Document&, AudioContextOptions&& = { });
+ static ExceptionOr<Ref<AudioContext>> create(Document&, AudioContextOptions&&);
~AudioContext();
WEBCORE_EXPORT static void setDefaultSampleRateForTesting(Optional<float>);
@@ -55,8 +60,9 @@
void close(DOMPromiseDeferred<void>&&);
- DefaultAudioDestinationNode& destination();
- const DefaultAudioDestinationNode& destination() const;
+ DefaultAudioDestinationNode& destination() final { return m_destinationNode.get(); }
+ const DefaultAudioDestinationNode& destination() const final { return m_destinationNode.get(); }
+
double baseLatency();
AudioTimestamp getOutputTimestamp(DOMWindow&);
@@ -88,8 +94,8 @@
void addBehaviorRestriction(BehaviorRestrictions restriction) { m_restrictions |= restriction; }
void removeBehaviorRestriction(BehaviorRestrictions restriction) { m_restrictions &= ~restriction; }
-protected:
- explicit AudioContext(Document&, const AudioContextOptions& = { });
+private:
+ AudioContext(Document&, const AudioContextOptions&);
bool willBeginPlayback();
@@ -97,7 +103,6 @@
const Logger& logger() const final;
#endif
-private:
void constructCommon();
bool userGestureRequiredForAudioStart() const { return !isOfflineContext() && m_restrictions & RequireUserGestureForAudioStartRestriction; }
@@ -105,6 +110,9 @@
bool willPausePlayback();
+ void uninitialize() final;
+ bool isOfflineContext() const final { return false; }
+
// MediaProducer
MediaProducer::MediaStateFlags mediaState() const final;
void pageMutedStateDidChange() final;
@@ -129,9 +137,11 @@
void visibilityStateChanged() final;
// ActiveDOMObject
+ const char* activeDOMObjectName() const final;
void suspend(ReasonForSuspension) final;
void resume() final;
+ UniqueRef<DefaultAudioDestinationNode> m_destinationNode;
std::unique_ptr<PlatformMediaSession> m_mediaSession;
BehaviorRestrictions m_restrictions { NoRestrictions };
@@ -142,3 +152,7 @@
};
} // WebCore
+
+SPECIALIZE_TYPE_TRAITS_BEGIN(WebCore::AudioContext)
+ static bool isType(const WebCore::BaseAudioContext& context) { return !context.isOfflineContext(); }
+SPECIALIZE_TYPE_TRAITS_END()
Modified: trunk/Source/WebCore/Modules/webaudio/AudioContextState.h (277552 => 277553)
--- trunk/Source/WebCore/Modules/webaudio/AudioContextState.h 2021-05-15 23:55:58 UTC (rev 277552)
+++ trunk/Source/WebCore/Modules/webaudio/AudioContextState.h 2021-05-16 01:05:18 UTC (rev 277553)
@@ -27,6 +27,6 @@
namespace WebCore {
-enum class AudioContextState { Suspended, Running, Interrupted, Closed };
+enum class AudioContextState : uint8_t { Suspended, Running, Interrupted, Closed };
}
Modified: trunk/Source/WebCore/Modules/webaudio/AudioNode.cpp (277552 => 277553)
--- trunk/Source/WebCore/Modules/webaudio/AudioNode.cpp 2021-05-15 23:55:58 UTC (rev 277552)
+++ trunk/Source/WebCore/Modules/webaudio/AudioNode.cpp 2021-05-16 01:05:18 UTC (rev 277553)
@@ -387,7 +387,7 @@
if (!channelCount)
return Exception { NotSupportedError, "Channel count cannot be 0"_s };
- if (channelCount > AudioContext::maxNumberOfChannels())
+ if (channelCount > AudioContext::maxNumberOfChannels)
return Exception { IndexSizeError, "Channel count exceeds maximum limit"_s };
if (m_channelCount == channelCount)
Modified: trunk/Source/WebCore/Modules/webaudio/AudioNodeOutput.cpp (277552 => 277553)
--- trunk/Source/WebCore/Modules/webaudio/AudioNodeOutput.cpp 2021-05-15 23:55:58 UTC (rev 277552)
+++ trunk/Source/WebCore/Modules/webaudio/AudioNodeOutput.cpp 2021-05-16 01:05:18 UTC (rev 277553)
@@ -42,7 +42,7 @@
, m_numberOfChannels(numberOfChannels)
, m_desiredNumberOfChannels(numberOfChannels)
{
- ASSERT(numberOfChannels <= AudioContext::maxNumberOfChannels());
+ ASSERT(numberOfChannels <= AudioContext::maxNumberOfChannels);
m_internalBus = AudioBus::create(numberOfChannels, AudioUtilities::renderQuantumSize);
}
@@ -49,7 +49,7 @@
void AudioNodeOutput::setNumberOfChannels(unsigned numberOfChannels)
{
- ASSERT(numberOfChannels <= AudioContext::maxNumberOfChannels());
+ ASSERT(numberOfChannels <= AudioContext::maxNumberOfChannels);
ASSERT(context().isGraphOwner());
m_desiredNumberOfChannels = numberOfChannels;
Modified: trunk/Source/WebCore/Modules/webaudio/AudioWorkletNode.cpp (277552 => 277553)
--- trunk/Source/WebCore/Modules/webaudio/AudioWorkletNode.cpp 2021-05-15 23:55:58 UTC (rev 277552)
+++ trunk/Source/WebCore/Modules/webaudio/AudioWorkletNode.cpp 2021-05-16 01:05:18 UTC (rev 277553)
@@ -68,7 +68,7 @@
return Exception { IndexSizeError, "Length of specified outputChannelCount does not match the given number of outputs"_s };
for (auto& channelCount : *options.outputChannelCount) {
- if (channelCount < 1 || channelCount > AudioContext::maxNumberOfChannels())
+ if (channelCount < 1 || channelCount > AudioContext::maxNumberOfChannels)
return Exception { NotSupportedError, "Provided number of channels for output is outside supported range"_s };
}
}
Modified: trunk/Source/WebCore/Modules/webaudio/BaseAudioContext.cpp (277552 => 277553)
--- trunk/Source/WebCore/Modules/webaudio/BaseAudioContext.cpp 2021-05-15 23:55:58 UTC (rev 277552)
+++ trunk/Source/WebCore/Modules/webaudio/BaseAudioContext.cpp 2021-05-16 01:05:18 UTC (rev 277553)
@@ -1,6 +1,6 @@
/*
* Copyright (C) 2010 Google Inc. All rights reserved.
- * Copyright (C) 2016 Apple Inc. All rights reserved.
+ * Copyright (C) 2016-2021 Apple Inc. All rights reserved.
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions
@@ -50,7 +50,6 @@
#include "ConstantSourceNode.h"
#include "ConstantSourceOptions.h"
#include "ConvolverNode.h"
-#include "DefaultAudioDestinationNode.h"
#include "DelayNode.h"
#include "DelayOptions.h"
#include "Document.h"
@@ -68,8 +67,6 @@
#include "JSDOMPromiseDeferred.h"
#include "Logging.h"
#include "NetworkingContext.h"
-#include "OfflineAudioCompletionEvent.h"
-#include "OfflineAudioDestinationNode.h"
#include "OscillatorNode.h"
#include "Page.h"
#include "PannerNode.h"
@@ -80,8 +77,14 @@
#include "StereoPannerNode.h"
#include "StereoPannerOptions.h"
#include "WaveShaperNode.h"
+#include <_javascript_Core/ArrayBuffer.h>
#include <_javascript_Core/ScriptCallStack.h>
+#include <wtf/Atomics.h>
+#include <wtf/IsoMallocInlines.h>
+#include <wtf/MainThread.h>
+#include <wtf/Ref.h>
#include <wtf/Scope.h>
+#include <wtf/text/WTFString.h>
#if DEBUG_AUDIONODE_REFERENCES
#include <stdio.h>
@@ -91,20 +94,6 @@
#include "GStreamerCommon.h"
#endif
-#if PLATFORM(IOS_FAMILY)
-#include "ScriptController.h"
-#include "Settings.h"
-#endif
-
-#include <_javascript_Core/ArrayBuffer.h>
-#include <wtf/Atomics.h>
-#include <wtf/IsoMallocInlines.h>
-#include <wtf/MainThread.h>
-#include <wtf/Ref.h>
-#include <wtf/RefCounted.h>
-#include <wtf/Scope.h>
-#include <wtf/text/WTFString.h>
-
namespace WebCore {
WTF_MAKE_ISO_ALLOCATED_IMPL(BaseAudioContext);
@@ -114,9 +103,7 @@
return sampleRate >= 3000 && sampleRate <= 384000;
}
-unsigned BaseAudioContext::s_hardwareContextCount;
-
-static uint64_t generateAudioContextID()
+static uint64_t generateContextID()
{
ASSERT(isMainThread());
static uint64_t contextIDSeed = 0;
@@ -130,55 +117,21 @@
return contexts;
}
-// Constructor for rendering to the audio hardware.
-BaseAudioContext::BaseAudioContext(Document& document, const AudioContextOptions& contextOptions)
+BaseAudioContext::BaseAudioContext(Document& document)
: ActiveDOMObject(document)
#if !RELEASE_LOG_DISABLED
, m_logger(document.logger())
, m_logIdentifier(uniqueLogIdentifier())
#endif
- , m_contextID(generateAudioContextID())
+ , m_contextID(generateContextID())
, m_worklet(AudioWorklet::create(*this))
- , m_destinationNode(makeUniqueRef<DefaultAudioDestinationNode>(*this, contextOptions.sampleRate))
, m_listener(AudioListener::create(*this))
{
liveAudioContexts().add(m_contextID);
- // According to spec AudioContext must die only after page navigate.
- // Lets mark it as ActiveDOMObject with pending activity and unmark it in clear method.
- setPendingActivity();
-
FFTFrame::initialize();
-
- // Unlike OfflineAudioContext, AudioContext does not require calling resume() to start rendering.
- // Lazy initialization starts rendering so we schedule a task here to make sure lazy initialization
- // ends up happening, even if no audio node gets constructed.
- postTask([this] {
- if (m_isStopScheduled)
- return;
-
- lazyInitialize();
- });
}
-// Constructor for offline (non-realtime) rendering.
-BaseAudioContext::BaseAudioContext(Document& document, unsigned numberOfChannels, float sampleRate, RefPtr<AudioBuffer>&& renderTarget)
- : ActiveDOMObject(document)
-#if !RELEASE_LOG_DISABLED
- , m_logger(document.logger())
- , m_logIdentifier(uniqueLogIdentifier())
-#endif
- , m_contextID(generateAudioContextID())
- , m_worklet(AudioWorklet::create(*this))
- , m_isOfflineContext(true)
- , m_renderTarget(WTFMove(renderTarget))
- , m_destinationNode(makeUniqueRef<OfflineAudioDestinationNode>(*this, numberOfChannels, sampleRate, m_renderTarget.copyRef()))
- , m_listener(AudioListener::create(*this))
-{
- liveAudioContexts().add(m_contextID);
- FFTFrame::initialize();
-}
-
BaseAudioContext::~BaseAudioContext()
{
liveAudioContexts().remove(m_contextID);
@@ -212,7 +165,7 @@
if (m_isAudioThreadFinished)
return;
- m_destinationNode->initialize();
+ destination().initialize();
m_isInitialized = true;
}
@@ -240,19 +193,11 @@
return;
// This stops the audio thread and all audio rendering.
- m_destinationNode->uninitialize();
+ destination().uninitialize();
// Don't allow the context to initialize a second time after it's already been explicitly uninitialized.
m_isAudioThreadFinished = true;
- if (!isOfflineContext()) {
- ASSERT(s_hardwareContextCount);
- --s_hardwareContextCount;
-
- // Offline contexts move to 'Closed' state when dispatching the completion event.
- setState(State::Closed);
- }
-
{
AutoLocker locker(*this);
// This should have been called from handlePostRenderTasks() at the end of rendering.
@@ -268,11 +213,6 @@
m_isInitialized = false;
}
-bool BaseAudioContext::isInitialized() const
-{
- return m_isInitialized;
-}
-
void BaseAudioContext::addReaction(State state, DOMPromiseDeferred<void>&& promise)
{
size_t stateIndex = static_cast<size_t>(state);
@@ -302,14 +242,15 @@
void BaseAudioContext::stop()
{
+ ASSERT(isMainThread());
ALWAYS_LOG(LOGIDENTIFIER);
- ASSERT(isMainThread());
- auto protectedThis = makeRef(*this);
-
// Usually ScriptExecutionContext calls stop twice.
if (m_isStopScheduled)
return;
+
+ auto protectedThis = makeRef(*this);
+
m_isStopScheduled = true;
ASSERT(document());
@@ -319,21 +260,11 @@
clear();
}
-const char* BaseAudioContext::activeDOMObjectName() const
-{
- return "AudioContext";
-}
-
Document* BaseAudioContext::document() const
{
return downcast<Document>(m_scriptExecutionContext);
}
-float BaseAudioContext::sampleRate() const
-{
- return m_destinationNode->sampleRate();
-}
-
bool BaseAudioContext::wouldTaintOrigin(const URL& url) const
{
if (url.protocolIsData())
@@ -378,11 +309,6 @@
});
}
-AudioListener& WebCore::BaseAudioContext::listener()
-{
- return m_listener;
-}
-
ExceptionOr<Ref<AudioBufferSourceNode>> BaseAudioContext::createBufferSource()
{
ALWAYS_LOG(LOGIDENTIFIER);
@@ -435,13 +361,13 @@
// This parameter [numberOfInputChannels] determines the number of channels for this node's input. Values of
// up to 32 must be supported. A NotSupportedError must be thrown if the number of channels is not supported.
- if (numberOfInputChannels > maxNumberOfChannels())
+ if (numberOfInputChannels > maxNumberOfChannels)
return Exception { NotSupportedError, "numberOfInputChannels exceeds maximum number of channels"_s };
// This parameter [numberOfOutputChannels] determines the number of channels for this node's output. Values of
// up to 32 must be supported. A NotSupportedError must be thrown if the number of channels is not supported.
- if (numberOfOutputChannels > maxNumberOfChannels())
+ if (numberOfOutputChannels > maxNumberOfChannels)
return Exception { NotSupportedError, "numberOfOutputChannels exceeds maximum number of channels"_s };
return ScriptProcessorNode::create(*this, bufferSize, numberOfInputChannels, numberOfOutputChannels);
@@ -583,11 +509,6 @@
return IIRFilterNode::create(scriptExecutionContext, *this, WTFMove(options));
}
-static bool isFinishedSourceNode(const AudioConnectionRefPtr<AudioNode>& node)
-{
- return node->isFinishedSourceNode();
-}
-
void BaseAudioContext::derefFinishedSourceNodes()
{
ASSERT(isGraphOwner());
@@ -596,7 +517,7 @@
if (!m_hasFinishedAudioSourceNodes)
return;
- m_referencedSourceNodes.removeAllMatching(isFinishedSourceNode);
+ m_referencedSourceNodes.removeAllMatching([](auto& node) { return node->isFinishedSourceNode(); });
m_hasFinishedAudioSourceNodes = false;
}
@@ -641,7 +562,7 @@
mustReleaseLock = false;
} else {
// Acquire the lock.
- m_contextGraphMutex.lock();
+ m_contextGraphLock.lock();
m_graphOwnerThread = &thisThread;
mustReleaseLock = true;
}
@@ -649,13 +570,10 @@
bool BaseAudioContext::tryLock(bool& mustReleaseLock)
{
- Thread& thisThread = Thread::current();
- bool isAudioThread = &thisThread == audioThread();
-
// Try to catch cases of using try lock on main thread - it should use regular lock.
- ASSERT(isAudioThread || isAudioThreadFinished());
+ ASSERT(isAudioThread() || isAudioThreadFinished());
- if (!isAudioThread) {
+ if (!isAudioThread()) {
// In release build treat tryLock() as lock() (since above ASSERT(isAudioThread) never fires) - this is the best we can do.
lock(mustReleaseLock);
return true;
@@ -663,16 +581,16 @@
bool hasLock;
- if (&thisThread == m_graphOwnerThread) {
+ if (isGraphOwner()) {
// Thread already has the lock.
hasLock = true;
mustReleaseLock = false;
} else {
// Don't already have the lock - try to acquire it.
- hasLock = m_contextGraphMutex.tryLock();
+ hasLock = m_contextGraphLock.tryLock();
if (hasLock)
- m_graphOwnerThread = &thisThread;
+ m_graphOwnerThread = &Thread::current();
mustReleaseLock = hasLock;
}
@@ -685,19 +603,9 @@
ASSERT(m_graphOwnerThread == &Thread::current());
m_graphOwnerThread = nullptr;
- m_contextGraphMutex.unlock();
+ m_contextGraphLock.unlock();
}
-bool BaseAudioContext::isAudioThread() const
-{
- return m_audioThread == &Thread::current();
-}
-
-bool BaseAudioContext::isGraphOwner() const
-{
- return m_graphOwnerThread == &Thread::current();
-}
-
void BaseAudioContext::addDeferredDecrementConnectionCount(AudioNode* node)
{
ASSERT(isAudioThread());
@@ -739,26 +647,27 @@
// The worst that can happen is that there will be some nodes which will take slightly longer than usual to be deleted or removed
// from the render graph (in which case they'll render silence).
bool mustReleaseLock;
- if (tryLock(mustReleaseLock)) {
- // Take care of finishing any derefs where the tryLock() failed previously.
- handleDeferredDecrementConnectionCounts();
+ if (!tryLock(mustReleaseLock))
+ return;
- // Dynamically clean up nodes which are no longer needed.
- derefFinishedSourceNodes();
+ // Take care of finishing any derefs where the tryLock() failed previously.
+ handleDeferredDecrementConnectionCounts();
- // Don't delete in the real-time thread. Let the main thread do it.
- // Ref-counted objects held by certain AudioNodes may not be thread-safe.
- scheduleNodeDeletion();
+ // Dynamically clean up nodes which are no longer needed.
+ derefFinishedSourceNodes();
- // Fixup the state of any dirty AudioSummingJunctions and AudioNodeOutputs.
- handleDirtyAudioSummingJunctions();
- handleDirtyAudioNodeOutputs();
+ // Don't delete in the real-time thread. Let the main thread do it.
+ // Ref-counted objects held by certain AudioNodes may not be thread-safe.
+ scheduleNodeDeletion();
- updateAutomaticPullNodes();
+ // Fixup the state of any dirty AudioSummingJunctions and AudioNodeOutputs.
+ handleDirtyAudioSummingJunctions();
+ handleDirtyAudioNodeOutputs();
- if (mustReleaseLock)
- unlock();
- }
+ updateAutomaticPullNodes();
+
+ if (mustReleaseLock)
+ unlock();
}
void BaseAudioContext::handleDeferredDecrementConnectionCounts()
@@ -820,29 +729,28 @@
// Protect this object from being deleted before we release the mutex locked by AutoLocker.
auto protectedThis = makeRef(*this);
- {
- AutoLocker locker(*this);
- while (m_nodesToDelete.size()) {
- AudioNode* node = m_nodesToDelete.takeLast();
+ AutoLocker locker(*this);
- // Before deleting the node, clear out any AudioNodeInputs from m_dirtySummingJunctions.
- unsigned numberOfInputs = node->numberOfInputs();
- for (unsigned i = 0; i < numberOfInputs; ++i)
- m_dirtySummingJunctions.remove(node->input(i));
+ while (m_nodesToDelete.size()) {
+ AudioNode* node = m_nodesToDelete.takeLast();
- // Before deleting the node, clear out any AudioNodeOutputs from m_dirtyAudioNodeOutputs.
- unsigned numberOfOutputs = node->numberOfOutputs();
- for (unsigned i = 0; i < numberOfOutputs; ++i)
- m_dirtyAudioNodeOutputs.remove(node->output(i));
+ // Before deleting the node, clear out any AudioNodeInputs from m_dirtySummingJunctions.
+ unsigned numberOfInputs = node->numberOfInputs();
+ for (unsigned i = 0; i < numberOfInputs; ++i)
+ m_dirtySummingJunctions.remove(node->input(i));
- ASSERT_WITH_MESSAGE(node->nodeType() != AudioNode::NodeTypeDestination, "Destination node is owned by the BaseAudioContext");
+ // Before deleting the node, clear out any AudioNodeOutputs from m_dirtyAudioNodeOutputs.
+ unsigned numberOfOutputs = node->numberOfOutputs();
+ for (unsigned i = 0; i < numberOfOutputs; ++i)
+ m_dirtyAudioNodeOutputs.remove(node->output(i));
- // Finally, delete it.
- delete node;
- }
- m_isDeletionScheduled = false;
+ ASSERT_WITH_MESSAGE(node->nodeType() != AudioNode::NodeTypeDestination, "Destination node is owned by the BaseAudioContext");
+
+ // Finally, delete it.
+ delete node;
}
+ m_isDeletionScheduled = false;
}
void BaseAudioContext::markSummingJunctionDirty(AudioSummingJunction* summingJunction)
@@ -915,20 +823,21 @@
{
ASSERT(isGraphOwner());
- if (m_automaticPullNodesNeedUpdating) {
- // Heap allocations are forbidden on the audio thread for performance reasons so we need to
- // explicitly allow the following allocation(s).
- DisableMallocRestrictionsForCurrentThreadScope disableMallocRestrictions;
+ if (!m_automaticPullNodesNeedUpdating)
+ return;
- // Copy from m_automaticPullNodes to m_renderingAutomaticPullNodes.
- m_renderingAutomaticPullNodes.resize(m_automaticPullNodes.size());
+ // Heap allocations are forbidden on the audio thread for performance reasons so we need to
+ // explicitly allow the following allocation(s).
+ DisableMallocRestrictionsForCurrentThreadScope disableMallocRestrictions;
- unsigned i = 0;
- for (auto& output : m_automaticPullNodes)
- m_renderingAutomaticPullNodes[i++] = output;
+ // Copy from m_automaticPullNodes to m_renderingAutomaticPullNodes.
+ m_renderingAutomaticPullNodes.resize(m_automaticPullNodes.size());
- m_automaticPullNodesNeedUpdating = false;
- }
+ unsigned i = 0;
+ for (auto& output : m_automaticPullNodes)
+ m_renderingAutomaticPullNodes[i++] = output;
+
+ m_automaticPullNodesNeedUpdating = false;
}
void BaseAudioContext::processAutomaticPullNodes(size_t framesToProcess)
@@ -944,6 +853,7 @@
return ActiveDOMObject::scriptExecutionContext();
}
+// FIXME: This should probably move to AudioContext.
void BaseAudioContext::isPlayingAudioDidChange()
{
// Heap allocations are forbidden on the audio thread for performance reasons so we need to
@@ -958,53 +868,6 @@
});
}
-// FIXME: Move to OfflineAudioContext once WebKitOfflineAudioContext gets removed.
-void BaseAudioContext::finishedRendering(bool didRendering)
-{
- ASSERT(isOfflineContext());
- ASSERT(isMainThread());
- auto finishedRenderingScope = WTF::makeScopeExit([this] {
- didFinishOfflineRendering(Exception { InvalidStateError, "Offline rendering failed"_s });
- });
-
- if (!isMainThread())
- return;
-
- auto clearPendingActivityIfExitEarly = WTF::makeScopeExit([this] {
- clearPendingActivity();
- });
-
-
- ALWAYS_LOG(LOGIDENTIFIER);
-
- if (!didRendering)
- return;
-
- RefPtr<AudioBuffer> renderedBuffer = m_renderTarget.get();
- setState(State::Closed);
-
- ASSERT(renderedBuffer);
- if (!renderedBuffer)
- return;
-
- // Avoid firing the event if the document has already gone away.
- if (m_isStopScheduled)
- return;
-
- clearPendingActivityIfExitEarly.release();
- queueTaskToDispatchEvent(*this, TaskSource::MediaElement, OfflineAudioCompletionEvent::create(*renderedBuffer));
-
- finishedRenderingScope.release();
- didFinishOfflineRendering(renderedBuffer.releaseNonNull());
-}
-
-void BaseAudioContext::dispatchEvent(Event& event)
-{
- EventTarget::dispatchEvent(event);
- if (event.eventInterface() == OfflineAudioCompletionEventInterfaceType)
- clearPendingActivity();
-}
-
void BaseAudioContext::incrementActiveSourceCount()
{
++m_activeSourceCount;
@@ -1015,18 +878,11 @@
--m_activeSourceCount;
}
-void BaseAudioContext::didSuspendRendering(size_t)
+void BaseAudioContext::postTask(Function<void()>&& task)
{
- setState(State::Suspended);
-}
-
-void BaseAudioContext::postTask(WTF::Function<void()>&& task)
-{
ASSERT(isMainThread());
- if (m_isStopScheduled)
- return;
-
- queueTaskKeepingObjectAlive(*this, TaskSource::MediaElement, WTFMove(task));
+ if (!m_isStopScheduled)
+ queueTaskKeepingObjectAlive(*this, TaskSource::MediaElement, WTFMove(task));
}
const SecurityOrigin* BaseAudioContext::origin() const
@@ -1104,7 +960,7 @@
// If we're already rendering when the worklet becomes ready, we need to restart
// rendering in order to switch to the audio worklet thread.
- m_destinationNode->restartRendering();
+ destination().restartRendering();
}
#if !RELEASE_LOG_DISABLED
Modified: trunk/Source/WebCore/Modules/webaudio/BaseAudioContext.h (277552 => 277553)
--- trunk/Source/WebCore/Modules/webaudio/BaseAudioContext.h 2021-05-15 23:55:58 UTC (rev 277552)
+++ trunk/Source/WebCore/Modules/webaudio/BaseAudioContext.h 2021-05-16 01:05:18 UTC (rev 277553)
@@ -1,6 +1,6 @@
/*
* Copyright (C) 2010 Google Inc. All rights reserved.
- * Copyright (C) 2016-2020 Apple Inc. All rights reserved.
+ * Copyright (C) 2016-2021 Apple Inc. All rights reserved.
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions
@@ -27,9 +27,6 @@
#if ENABLE(WEB_AUDIO)
#include "ActiveDOMObject.h"
-#include "AsyncAudioDecoder.h"
-#include "AudioBus.h"
-#include "AudioContextOptions.h"
#include "AudioContextState.h"
#include "AudioDestinationNode.h"
#include "EventTarget.h"
@@ -36,28 +33,22 @@
#include "JSDOMPromiseDeferred.h"
#include "OscillatorType.h"
#include "PeriodicWaveConstraints.h"
-#include "ScriptExecutionContext.h"
-#include <_javascript_Core/ConsoleTypes.h>
-#include <_javascript_Core/Float32Array.h>
#include <atomic>
-#include <wtf/HashSet.h>
+#include <wtf/Forward.h>
#include <wtf/LoggerHelper.h>
#include <wtf/MainThread.h>
-#include <wtf/RefPtr.h>
#include <wtf/ThreadSafeRefCounted.h>
#include <wtf/Threading.h>
-#include <wtf/UniqueRef.h>
-#include <wtf/Vector.h>
-#include <wtf/WeakPtr.h>
-#include <wtf/text/AtomStringHash.h>
namespace WebCore {
class AnalyserNode;
+class AsyncAudioDecoder;
class AudioBuffer;
class AudioBufferCallback;
class AudioBufferSourceNode;
class AudioListener;
+class AudioNodeOutput;
class AudioSummingJunction;
class AudioWorklet;
class BiquadFilterNode;
@@ -69,21 +60,17 @@
class Document;
class DynamicsCompressorNode;
class GainNode;
-class HTMLMediaElement;
class IIRFilterNode;
class MediaElementAudioSourceNode;
-class MediaStream;
-class MediaStreamAudioDestinationNode;
-class MediaStreamAudioSourceNode;
class OscillatorNode;
class PannerNode;
class PeriodicWave;
-class ScriptExecutionContext;
class ScriptProcessorNode;
class SecurityOrigin;
class StereoPannerNode;
class WaveShaperNode;
+struct AudioIOPosition;
struct AudioParamDescriptor;
template<typename IDLType> class DOMPromiseDeferred;
@@ -112,34 +99,25 @@
uint64_t contextID() const { return m_contextID; }
Document* document() const;
- bool isInitialized() const;
+ bool isInitialized() const { return m_isInitialized; }
- bool isOfflineContext() const { return m_isOfflineContext; }
+ virtual bool isOfflineContext() const = 0;
+ virtual AudioDestinationNode& destination() = 0;
+ virtual const AudioDestinationNode& destination() const = 0;
- AudioDestinationNode& destination() { return m_destinationNode.get(); }
- const AudioDestinationNode& destination() const { return m_destinationNode.get(); }
-
- size_t currentSampleFrame() const { return m_destinationNode->currentSampleFrame(); }
- double currentTime() const { return m_destinationNode->currentTime(); }
- float sampleRate() const;
+ size_t currentSampleFrame() const { return destination().currentSampleFrame(); }
+ double currentTime() const { return destination().currentTime(); }
+ float sampleRate() const { return destination().sampleRate(); }
unsigned long activeSourceCount() const { return static_cast<unsigned long>(m_activeSourceCount); }
void incrementActiveSourceCount();
void decrementActiveSourceCount();
- virtual bool shouldSuspend() { return false; }
-
- ExceptionOr<Ref<AudioBuffer>> createBuffer(unsigned numberOfChannels, unsigned length, float sampleRate);
-
// Asynchronous audio file data decoding.
void decodeAudioData(Ref<ArrayBuffer>&&, RefPtr<AudioBufferCallback>&&, RefPtr<AudioBufferCallback>&&, Optional<Ref<DeferredPromise>>&& = WTF::nullopt);
- AudioListener& listener();
+ AudioListener& listener() { return m_listener; }
- virtual void didSuspendRendering(size_t frame);
-
- AudioBuffer* renderTarget() const { return m_renderTarget.get(); }
-
using State = AudioContextState;
State state() const { return m_state; }
bool isClosed() const { return m_state == State::Closed; }
@@ -166,6 +144,7 @@
ExceptionOr<Ref<ConstantSourceNode>> createConstantSource();
ExceptionOr<Ref<StereoPannerNode>> createStereoPanner();
ExceptionOr<Ref<IIRFilterNode>> createIIRFilter(ScriptExecutionContext&, Vector<double>&& feedforward, Vector<double>&& feedback);
+ ExceptionOr<Ref<AudioBuffer>> createBuffer(unsigned numberOfChannels, unsigned length, float sampleRate);
// Called at the start of each render quantum.
void handlePreRenderTasks(const AudioIOPosition& outputPosition);
@@ -175,9 +154,6 @@
// Called at the end of each render quantum.
void handlePostRenderTasks();
- // Called periodically at the end of each render quantum to dereference finished source nodes.
- void derefFinishedSourceNodes();
-
// We schedule deletion of all marked nodes at the end of each realtime render quantum.
void markForDeletion(AudioNode&);
void deleteMarkedNodes();
@@ -194,7 +170,7 @@
void incrementConnectionCount()
{
ASSERT(isMainThread());
- m_connectionCount++;
+ ++m_connectionCount;
}
unsigned connectionCount() const { return m_connectionCount; }
@@ -204,11 +180,10 @@
//
void setAudioThread(Thread& thread) { m_audioThread = &thread; } // FIXME: check either not initialized or the same
- Thread* audioThread() const { return m_audioThread; }
- bool isAudioThread() const;
+ bool isAudioThread() const { return m_audioThread == &Thread::current(); }
// Returns true only after the audio thread has been started and then shutdown.
- bool isAudioThreadFinished() { return m_isAudioThreadFinished; }
+ bool isAudioThreadFinished() const { return m_isAudioThreadFinished; }
// mustReleaseLock is set to true if we acquired the lock in this method call and caller must unlock(), false if it was previously acquired.
void lock(bool& mustReleaseLock);
@@ -220,17 +195,15 @@
void unlock();
// Returns true if this thread owns the context's lock.
- bool isGraphOwner() const;
+ bool isGraphOwner() const { return m_graphOwnerThread == &Thread::current(); }
- // Returns the maximum number of channels we can support.
- static unsigned maxNumberOfChannels() { return MaxNumberOfChannels; }
+ // This is considering 32 is large enough for multiple channels audio.
+ // It is somewhat arbitrary and could be increased if necessary.
+ static constexpr unsigned maxNumberOfChannels = 32;
// In AudioNode::decrementConnectionCount() a tryLock() is used for calling decrementConnectionCountWithLock(), but if it fails keep track here.
void addDeferredDecrementConnectionCount(AudioNode*);
- // In the audio thread at the start of each render cycle, we'll call handleDeferredDecrementConnectionCounts().
- void handleDeferredDecrementConnectionCounts();
-
// Only accessed when the graph lock is held.
void markSummingJunctionDirty(AudioSummingJunction*);
void markAudioNodeOutputDirty(AudioNodeOutput*);
@@ -239,15 +212,8 @@
void removeMarkedSummingJunction(AudioSummingJunction*);
// EventTarget
- EventTargetInterface eventTargetInterface() const final;
ScriptExecutionContext* scriptExecutionContext() const final;
- void refEventTarget() override { ref(); }
- void derefEventTarget() override { deref(); }
- void finishedRendering(bool didRendering);
-
- static unsigned s_hardwareContextCount;
-
void isPlayingAudioDidChange();
virtual void sourceNodeWillBeginPlayback(AudioNode&);
@@ -262,7 +228,7 @@
const void* nextAudioParameterLogIdentifier() { return childLogIdentifier(m_logIdentifier, ++m_nextAudioParameterIdentifier); }
#endif
- void postTask(WTF::Function<void()>&&);
+ void postTask(Function<void()>&&);
bool isStopped() const { return m_isStopScheduled; }
const SecurityOrigin* origin() const;
void addConsoleMessage(MessageSource, MessageLevel, const String& message);
@@ -296,8 +262,7 @@
const HashMap<String, Vector<AudioParamDescriptor>>& parameterDescriptorMap() const { return m_parameterDescriptorMap; }
protected:
- explicit BaseAudioContext(Document&, const AudioContextOptions& = { });
- BaseAudioContext(Document&, unsigned numberOfChannels, float sampleRate, RefPtr<AudioBuffer>&& renderTarget);
+ explicit BaseAudioContext(Document&);
void clearPendingActivity();
void setPendingActivity();
@@ -313,7 +278,6 @@
void addReaction(State, DOMPromiseDeferred<void>&&);
void setState(State);
- virtual void didFinishOfflineRendering(ExceptionOr<Ref<AudioBuffer>>&&) { }
void clear();
private:
@@ -327,12 +291,19 @@
void refSourceNode(AudioNode&);
void derefSourceNode(AudioNode&);
+ // Called periodically at the end of each render quantum to dereference finished source nodes.
+ void derefFinishedSourceNodes();
+
+ // In the audio thread at the start of each render cycle, we'll call handleDeferredDecrementConnectionCounts().
+ void handleDeferredDecrementConnectionCounts();
+
// EventTarget
- void dispatchEvent(Event&) final;
+ EventTargetInterface eventTargetInterface() const final;
+ void refEventTarget() override { ref(); }
+ void derefEventTarget() override { deref(); }
// ActiveDOMObject API.
void stop() override;
- const char* activeDOMObjectName() const override;
// When the context goes away, there might still be some sources which haven't finished playing.
// Make sure to dereference them here.
@@ -366,13 +337,6 @@
// They will be scheduled for deletion (on the main thread) at the end of a render cycle (in realtime thread).
Vector<AudioNode*> m_nodesToDelete;
- bool m_isDeletionScheduled { false };
- bool m_isStopScheduled { false };
- bool m_isInitialized { false };
- bool m_isAudioThreadFinished { false };
- bool m_automaticPullNodesNeedUpdating { false };
- bool m_isOfflineContext { false };
-
// Only accessed when the graph lock is held.
HashSet<AudioSummingJunction*> m_dirtySummingJunctions;
HashSet<AudioNodeOutput*> m_dirtyAudioNodeOutputs;
@@ -385,35 +349,18 @@
Vector<AudioNode*> m_deferredBreakConnectionList;
Vector<Vector<DOMPromiseDeferred<void>>> m_stateReactions;
- RefPtr<AudioBuffer> m_renderTarget;
- UniqueRef<AudioDestinationNode> m_destinationNode;
Ref<AudioListener> m_listener;
- unsigned m_connectionCount { 0 };
+ std::atomic<Thread*> m_audioThread;
+ std::atomic<Thread*> m_graphOwnerThread; // if the lock is held then this is the thread which owns it, otherwise == nullptr.
- // Graph locking.
- Lock m_contextGraphMutex;
- // FIXME: Using volatile seems incorrect.
- // https://bugs.webkit.org/show_bug.cgi?id=180332
- Thread* volatile m_audioThread { nullptr };
- Thread* volatile m_graphOwnerThread { nullptr }; // if the lock is held then this is the thread which owns it, otherwise == nullptr.
-
std::unique_ptr<AsyncAudioDecoder> m_audioDecoder;
- // This is considering 32 is large enough for multiple channels audio.
- // It is somewhat arbitrary and could be increased if necessary.
- enum { MaxNumberOfChannels = 32 };
-
- // Number of AudioBufferSourceNodes that are active (playing).
- std::atomic<int> m_activeSourceCount { 0 };
-
- State m_state { State::Suspended };
RefPtr<PendingActivity<BaseAudioContext>> m_pendingActivity;
AudioIOPosition m_outputPosition;
HashMap<String, Vector<AudioParamDescriptor>> m_parameterDescriptorMap;
- bool m_hasFinishedAudioSourceNodes { false };
// These are cached per audio context for performance reasons. They cannot be
// static because they rely on the sample rate.
@@ -421,6 +368,19 @@
RefPtr<PeriodicWave> m_cachedPeriodicWaveSquare;
RefPtr<PeriodicWave> m_cachedPeriodicWaveSawtooth;
RefPtr<PeriodicWave> m_cachedPeriodicWaveTriangle;
+
+ // Number of AudioBufferSourceNodes that are active (playing).
+ std::atomic<int> m_activeSourceCount;
+
+ unsigned m_connectionCount { 0 };
+ State m_state { State::Suspended };
+ Lock m_contextGraphLock;
+ bool m_isDeletionScheduled { false };
+ bool m_isStopScheduled { false };
+ bool m_isInitialized { false };
+ bool m_isAudioThreadFinished { false };
+ bool m_automaticPullNodesNeedUpdating { false };
+ bool m_hasFinishedAudioSourceNodes { false };
};
} // WebCore
Modified: trunk/Source/WebCore/Modules/webaudio/ChannelMergerNode.cpp (277552 => 277553)
--- trunk/Source/WebCore/Modules/webaudio/ChannelMergerNode.cpp 2021-05-15 23:55:58 UTC (rev 277552)
+++ trunk/Source/WebCore/Modules/webaudio/ChannelMergerNode.cpp 2021-05-16 01:05:18 UTC (rev 277553)
@@ -43,7 +43,7 @@
ExceptionOr<Ref<ChannelMergerNode>> ChannelMergerNode::create(BaseAudioContext& context, const ChannelMergerOptions& options)
{
- if (options.numberOfInputs > AudioContext::maxNumberOfChannels() || !options.numberOfInputs)
+ if (options.numberOfInputs > AudioContext::maxNumberOfChannels || !options.numberOfInputs)
return Exception { IndexSizeError, "Number of inputs is not in the allowed range."_s };
auto merger = adoptRef(*new ChannelMergerNode(context, options.numberOfInputs));
Modified: trunk/Source/WebCore/Modules/webaudio/ChannelSplitterNode.cpp (277552 => 277553)
--- trunk/Source/WebCore/Modules/webaudio/ChannelSplitterNode.cpp 2021-05-15 23:55:58 UTC (rev 277552)
+++ trunk/Source/WebCore/Modules/webaudio/ChannelSplitterNode.cpp 2021-05-16 01:05:18 UTC (rev 277553)
@@ -39,7 +39,7 @@
ExceptionOr<Ref<ChannelSplitterNode>> ChannelSplitterNode::create(BaseAudioContext& context, const ChannelSplitterOptions& options)
{
- if (options.numberOfOutputs > AudioContext::maxNumberOfChannels() || !options.numberOfOutputs)
+ if (options.numberOfOutputs > AudioContext::maxNumberOfChannels || !options.numberOfOutputs)
return Exception { IndexSizeError, "Number of outputs is not in the allowed range"_s };
auto splitter = adoptRef(*new ChannelSplitterNode(context, options.numberOfOutputs));
Modified: trunk/Source/WebCore/Modules/webaudio/DefaultAudioDestinationNode.cpp (277552 => 277553)
--- trunk/Source/WebCore/Modules/webaudio/DefaultAudioDestinationNode.cpp 2021-05-15 23:55:58 UTC (rev 277552)
+++ trunk/Source/WebCore/Modules/webaudio/DefaultAudioDestinationNode.cpp 2021-05-16 01:05:18 UTC (rev 277553)
@@ -46,7 +46,7 @@
WTF_MAKE_ISO_ALLOCATED_IMPL(DefaultAudioDestinationNode);
-DefaultAudioDestinationNode::DefaultAudioDestinationNode(BaseAudioContext& context, Optional<float> sampleRate)
+DefaultAudioDestinationNode::DefaultAudioDestinationNode(AudioContext& context, Optional<float> sampleRate)
: AudioDestinationNode(context, sampleRate.valueOr(AudioDestination::hardwareSampleRate()))
{
ASSERT(BaseAudioContext::isSupportedSampleRate(AudioDestination::hardwareSampleRate()));
@@ -58,6 +58,16 @@
uninitialize();
}
+AudioContext& DefaultAudioDestinationNode::context()
+{
+ return downcast<AudioContext>(AudioDestinationNode::context());
+}
+
+const AudioContext& DefaultAudioDestinationNode::context() const
+{
+ return downcast<AudioContext>(AudioDestinationNode::context());
+}
+
void DefaultAudioDestinationNode::initialize()
{
ASSERT(isMainThread());
Modified: trunk/Source/WebCore/Modules/webaudio/DefaultAudioDestinationNode.h (277552 => 277553)
--- trunk/Source/WebCore/Modules/webaudio/DefaultAudioDestinationNode.h 2021-05-15 23:55:58 UTC (rev 277552)
+++ trunk/Source/WebCore/Modules/webaudio/DefaultAudioDestinationNode.h 2021-05-16 01:05:18 UTC (rev 277553)
@@ -28,12 +28,16 @@
namespace WebCore {
+class AudioContext;
class AudioDestination;
class DefaultAudioDestinationNode final : public AudioDestinationNode {
WTF_MAKE_ISO_ALLOCATED(DefaultAudioDestinationNode);
public:
- explicit DefaultAudioDestinationNode(BaseAudioContext&, Optional<float> = WTF::nullopt);
+ explicit DefaultAudioDestinationNode(AudioContext&, Optional<float> = WTF::nullopt);
+
+ AudioContext& context();
+ const AudioContext& context() const;
virtual ~DefaultAudioDestinationNode();
Modified: trunk/Source/WebCore/Modules/webaudio/MediaElementAudioSourceNode.cpp (277552 => 277553)
--- trunk/Source/WebCore/Modules/webaudio/MediaElementAudioSourceNode.cpp 2021-05-15 23:55:58 UTC (rev 277552)
+++ trunk/Source/WebCore/Modules/webaudio/MediaElementAudioSourceNode.cpp 2021-05-16 01:05:18 UTC (rev 277553)
@@ -86,7 +86,7 @@
m_muted = wouldTaintOrigin();
if (numberOfChannels != m_sourceNumberOfChannels || sourceSampleRate != m_sourceSampleRate) {
- if (!numberOfChannels || numberOfChannels > AudioContext::maxNumberOfChannels() || sourceSampleRate < minSampleRate || sourceSampleRate > maxSampleRate) {
+ if (!numberOfChannels || numberOfChannels > AudioContext::maxNumberOfChannels || sourceSampleRate < minSampleRate || sourceSampleRate > maxSampleRate) {
// process() will generate silence for these uninitialized values.
LOG(Media, "MediaElementAudioSourceNode::setFormat(%u, %f) - unhandled format change", static_cast<unsigned>(numberOfChannels), sourceSampleRate);
m_sourceNumberOfChannels = 0;
Modified: trunk/Source/WebCore/Modules/webaudio/MediaStreamAudioSourceNode.cpp (277552 => 277553)
--- trunk/Source/WebCore/Modules/webaudio/MediaStreamAudioSourceNode.cpp 2021-05-15 23:55:58 UTC (rev 277552)
+++ trunk/Source/WebCore/Modules/webaudio/MediaStreamAudioSourceNode.cpp 2021-05-16 01:05:18 UTC (rev 277553)
@@ -94,7 +94,7 @@
return;
// The sample-rate must be equal to the context's sample-rate.
- if (!numberOfChannels || numberOfChannels > AudioContext::maxNumberOfChannels()) {
+ if (!numberOfChannels || numberOfChannels > AudioContext::maxNumberOfChannels) {
// process() will generate silence for these uninitialized values.
LOG(Media, "MediaStreamAudioSourceNode::setFormat(%u, %f) - unhandled format change", static_cast<unsigned>(numberOfChannels), sourceSampleRate);
m_sourceNumberOfChannels = 0;
Modified: trunk/Source/WebCore/Modules/webaudio/OfflineAudioContext.cpp (277552 => 277553)
--- trunk/Source/WebCore/Modules/webaudio/OfflineAudioContext.cpp 2021-05-15 23:55:58 UTC (rev 277552)
+++ trunk/Source/WebCore/Modules/webaudio/OfflineAudioContext.cpp 2021-05-16 01:05:18 UTC (rev 277553)
@@ -1,5 +1,6 @@
/*
* Copyright (C) 2012, Google Inc. All rights reserved.
+ * Copyright (C) 2020-2021, Apple Inc. All rights reserved.
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions
@@ -32,6 +33,7 @@
#include "AudioUtilities.h"
#include "Document.h"
#include "JSAudioBuffer.h"
+#include "OfflineAudioContextOptions.h"
#include <wtf/IsoMallocInlines.h>
#include <wtf/Scope.h>
@@ -39,53 +41,60 @@
WTF_MAKE_ISO_ALLOCATED_IMPL(OfflineAudioContext);
-inline OfflineAudioContext::OfflineAudioContext(Document& document, unsigned numberOfChannels, unsigned length, float sampleRate, RefPtr<AudioBuffer>&& renderTarget)
- : BaseAudioContext(document, numberOfChannels, sampleRate, WTFMove(renderTarget))
- , m_length(length)
+OfflineAudioContext::OfflineAudioContext(Document& document, const OfflineAudioContextOptions& options)
+ : BaseAudioContext(document)
+ , m_destinationNode(makeUniqueRef<OfflineAudioDestinationNode>(*this, options.numberOfChannels, options.sampleRate, AudioBuffer::create(options.numberOfChannels, options.length, options.sampleRate)))
+ , m_length(options.length)
{
+ if (!renderTarget())
+ document.addConsoleMessage(MessageSource::JS, MessageLevel::Warning, makeString("Failed to construct internal AudioBuffer with ", options.numberOfChannels, " channel(s), a sample rate of ", options.sampleRate, " and a length of ", options.length, "."));
}
-ExceptionOr<Ref<OfflineAudioContext>> OfflineAudioContext::create(ScriptExecutionContext& context, unsigned numberOfChannels, unsigned length, float sampleRate)
+ExceptionOr<Ref<OfflineAudioContext>> OfflineAudioContext::create(ScriptExecutionContext& context, const OfflineAudioContextOptions& options)
{
if (!is<Document>(context))
return Exception { NotSupportedError, "OfflineAudioContext is only supported in Document contexts"_s };
- if (!numberOfChannels || numberOfChannels > maxNumberOfChannels())
+ if (!options.numberOfChannels || options.numberOfChannels > maxNumberOfChannels)
return Exception { SyntaxError, "Number of channels is not in range"_s };
- if (!length)
+ if (!options.length)
return Exception { SyntaxError, "length cannot be 0"_s };
- if (!isSupportedSampleRate(sampleRate))
+ if (!isSupportedSampleRate(options.sampleRate))
return Exception { SyntaxError, "sampleRate is not in range"_s };
- auto renderTarget = AudioBuffer::create(numberOfChannels, length, sampleRate);
- if (!renderTarget)
- context.addConsoleMessage(MessageSource::JS, MessageLevel::Warning, makeString("Failed to construct internal AudioBuffer with ", numberOfChannels, " channel(s), a sample rate of ", sampleRate, " and a length of ", length, "."));
-
- auto audioContext = adoptRef(*new OfflineAudioContext(downcast<Document>(context), numberOfChannels, length, sampleRate, WTFMove(renderTarget)));
+ auto audioContext = adoptRef(*new OfflineAudioContext(downcast<Document>(context), options));
audioContext->suspendIfNeeded();
return audioContext;
}
-ExceptionOr<Ref<OfflineAudioContext>> OfflineAudioContext::create(ScriptExecutionContext& context, const OfflineAudioContextOptions& contextOptions)
+ExceptionOr<Ref<OfflineAudioContext>> OfflineAudioContext::create(ScriptExecutionContext& context, unsigned numberOfChannels, unsigned length, float sampleRate)
{
- return create(context, contextOptions.numberOfChannels, contextOptions.length, contextOptions.sampleRate);
+ return create(context, { numberOfChannels, length, sampleRate });
}
void OfflineAudioContext::uninitialize()
{
+ if (!isInitialized())
+ return;
+
BaseAudioContext::uninitialize();
- if (auto promise = std::exchange(m_pendingOfflineRenderingPromise, nullptr))
+ if (auto promise = std::exchange(m_pendingRenderingPromise, nullptr))
promise->reject(Exception { InvalidStateError, "Context is going away"_s });
}
-void OfflineAudioContext::startOfflineRendering(Ref<DeferredPromise>&& promise)
+const char* OfflineAudioContext::activeDOMObjectName() const
{
+ return "OfflineAudioContext";
+}
+
+void OfflineAudioContext::startRendering(Ref<DeferredPromise>&& promise)
+{
if (isStopped()) {
promise->reject(Exception { InvalidStateError, "Context is stopped"_s });
return;
}
- if (m_didStartOfflineRendering) {
+ if (m_didStartRendering) {
promise->reject(Exception { InvalidStateError, "Rendering was already started"_s });
return;
}
@@ -104,13 +113,13 @@
}
setPendingActivity();
- m_pendingOfflineRenderingPromise = WTFMove(promise);
- m_didStartOfflineRendering = true;
+ m_pendingRenderingPromise = WTFMove(promise);
+ m_didStartRendering = true;
setState(State::Running);
});
}
-void OfflineAudioContext::suspendOfflineRendering(double suspendTime, Ref<DeferredPromise>&& promise)
+void OfflineAudioContext::suspendRendering(double suspendTime, Ref<DeferredPromise>&& promise)
{
if (isStopped()) {
promise->reject(Exception { InvalidStateError, "Context is stopped"_s });
@@ -143,9 +152,9 @@
}
}
-void OfflineAudioContext::resumeOfflineRendering(Ref<DeferredPromise>&& promise)
+void OfflineAudioContext::resumeRendering(Ref<DeferredPromise>&& promise)
{
- if (!m_didStartOfflineRendering) {
+ if (!m_didStartRendering) {
promise->reject(Exception { InvalidStateError, "Cannot resume an offline audio context that has not started"_s });
return;
}
@@ -182,7 +191,7 @@
void OfflineAudioContext::didSuspendRendering(size_t frame)
{
- BaseAudioContext::didSuspendRendering(frame);
+ setState(State::Suspended);
clearPendingActivity();
@@ -196,17 +205,38 @@
promise->resolve();
}
-void OfflineAudioContext::didFinishOfflineRendering(ExceptionOr<Ref<AudioBuffer>>&& result)
+void OfflineAudioContext::finishedRendering(bool didRendering)
{
- auto finishedRenderingScope = WTF::makeScopeExit([this] {
+ ASSERT(isMainThread());
+ ALWAYS_LOG(LOGIDENTIFIER);
+
+ auto uninitializeOnExit = WTF::makeScopeExit([this] {
uninitialize();
clear();
});
- if (!m_pendingOfflineRenderingPromise)
+ setState(State::Closed);
+
+ // Avoid firing the event if the document has already gone away.
+ if (isStopped())
return;
- auto promise = std::exchange(m_pendingOfflineRenderingPromise, nullptr);
+ RefPtr<AudioBuffer> renderedBuffer = renderTarget();
+ ASSERT(renderedBuffer);
+
+ if (didRendering) {
+ queueTaskToDispatchEvent(*this, TaskSource::MediaElement, OfflineAudioCompletionEvent::create(*renderedBuffer));
+ settleRenderingPromise(renderedBuffer.releaseNonNull());
+ } else
+ settleRenderingPromise(Exception { InvalidStateError, "Offline rendering failed"_s });
+}
+
+void OfflineAudioContext::settleRenderingPromise(ExceptionOr<Ref<AudioBuffer>>&& result)
+{
+ auto promise = std::exchange(m_pendingRenderingPromise, nullptr);
+ if (!promise)
+ return;
+
if (result.hasException()) {
promise->reject(result.releaseException());
return;
@@ -220,6 +250,13 @@
lockInternal(mustReleaseLock);
}
+void OfflineAudioContext::dispatchEvent(Event& event)
+{
+ BaseAudioContext::dispatchEvent(event);
+ if (event.eventInterface() == OfflineAudioCompletionEventInterfaceType)
+ clearPendingActivity();
+}
+
} // namespace WebCore
#endif // ENABLE(WEB_AUDIO)
Modified: trunk/Source/WebCore/Modules/webaudio/OfflineAudioContext.h (277552 => 277553)
--- trunk/Source/WebCore/Modules/webaudio/OfflineAudioContext.h 2021-05-15 23:55:58 UTC (rev 277552)
+++ trunk/Source/WebCore/Modules/webaudio/OfflineAudioContext.h 2021-05-16 01:05:18 UTC (rev 277553)
@@ -1,5 +1,6 @@
/*
* Copyright (C) 2012, Google Inc. All rights reserved.
+ * Copyright (C) 2020-2021, Apple Inc. All rights reserved.
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions
@@ -26,7 +27,6 @@
#include "BaseAudioContext.h"
#include "JSDOMPromiseDeferred.h"
-#include "OfflineAudioContextOptions.h"
#include "OfflineAudioDestinationNode.h"
#include <wtf/HashMap.h>
#include <wtf/Lock.h>
@@ -33,23 +33,30 @@
namespace WebCore {
+struct OfflineAudioContextOptions;
+
class OfflineAudioContext final : public BaseAudioContext {
WTF_MAKE_ISO_ALLOCATED(OfflineAudioContext);
public:
+ static ExceptionOr<Ref<OfflineAudioContext>> create(ScriptExecutionContext&, const OfflineAudioContextOptions&);
static ExceptionOr<Ref<OfflineAudioContext>> create(ScriptExecutionContext&, unsigned numberOfChannels, unsigned length, float sampleRate);
-
- static ExceptionOr<Ref<OfflineAudioContext>> create(ScriptExecutionContext&, const OfflineAudioContextOptions&);
+ void startRendering(Ref<DeferredPromise>&&);
+ void suspendRendering(double suspendTime, Ref<DeferredPromise>&&);
+ void resumeRendering(Ref<DeferredPromise>&&);
+ void finishedRendering(bool didRendering);
+ void didSuspendRendering(size_t frame);
- void startOfflineRendering(Ref<DeferredPromise>&&);
- void suspendOfflineRendering(double suspendTime, Ref<DeferredPromise>&&);
- void resumeOfflineRendering(Ref<DeferredPromise>&&);
-
unsigned length() const { return m_length; }
- bool shouldSuspend() final;
+ bool shouldSuspend();
- OfflineAudioDestinationNode& destination() { return static_cast<OfflineAudioDestinationNode&>(BaseAudioContext::destination()); }
- const OfflineAudioDestinationNode& destination() const { return static_cast<const OfflineAudioDestinationNode&>(BaseAudioContext::destination()); }
+ OfflineAudioDestinationNode& destination() final { return m_destinationNode.get(); }
+ const OfflineAudioDestinationNode& destination() const final { return m_destinationNode.get(); }
+private:
+ OfflineAudioContext(Document&, const OfflineAudioContextOptions&);
+
+ AudioBuffer* renderTarget() const { return destination().renderTarget(); }
+
// mustReleaseLock is set to true if we acquired the lock in this method call and caller must unlock(), false if it was previously acquired.
void offlineLock(bool& mustReleaseLock);
@@ -72,17 +79,25 @@
bool m_mustReleaseLock;
};
-private:
- OfflineAudioContext(Document&, unsigned numberOfChannels, unsigned length, float sampleRate, RefPtr<AudioBuffer>&& renderTarget);
+ // ActiveDOMObject
+ const char* activeDOMObjectName() const final;
- void didFinishOfflineRendering(ExceptionOr<Ref<AudioBuffer>>&&) final;
- void didSuspendRendering(size_t frame) final;
+ // EventTarget
+ void dispatchEvent(Event&) final;
+
+ void settleRenderingPromise(ExceptionOr<Ref<AudioBuffer>>&&);
void uninitialize() final;
+ bool isOfflineContext() const final { return true; }
- RefPtr<DeferredPromise> m_pendingOfflineRenderingPromise;
+ UniqueRef<OfflineAudioDestinationNode> m_destinationNode;
+ RefPtr<DeferredPromise> m_pendingRenderingPromise;
HashMap<unsigned /* frame */, RefPtr<DeferredPromise>, WTF::IntHash<unsigned>, WTF::UnsignedWithZeroKeyHashTraits<unsigned>> m_suspendRequests;
unsigned m_length;
- bool m_didStartOfflineRendering { false };
+ bool m_didStartRendering { false };
};
} // namespace WebCore
+
+SPECIALIZE_TYPE_TRAITS_BEGIN(WebCore::OfflineAudioContext)
+ static bool isType(const WebCore::BaseAudioContext& context) { return context.isOfflineContext(); }
+SPECIALIZE_TYPE_TRAITS_END()
Modified: trunk/Source/WebCore/Modules/webaudio/OfflineAudioContext.idl (277552 => 277553)
--- trunk/Source/WebCore/Modules/webaudio/OfflineAudioContext.idl 2021-05-15 23:55:58 UTC (rev 277552)
+++ trunk/Source/WebCore/Modules/webaudio/OfflineAudioContext.idl 2021-05-16 01:05:18 UTC (rev 277553)
@@ -31,10 +31,10 @@
[CallWith=ScriptExecutionContext] constructor(OfflineAudioContextOptions contextOptions);
[CallWith=ScriptExecutionContext] constructor(unsigned long numberOfChannels, unsigned long length, float sampleRate);
- [ImplementedAs=startOfflineRendering] Promise<AudioBuffer> startRendering();
+ Promise<AudioBuffer> startRendering();
- [ImplementedAs=resumeOfflineRendering] Promise<undefined> resume();
- [ImplementedAs=suspendOfflineRendering] Promise<undefined> suspend(double suspendTime);
+ [ImplementedAs=resumeRendering] Promise<undefined> resume();
+ [ImplementedAs=suspendRendering] Promise<undefined> suspend(double suspendTime);
readonly attribute unsigned long length;
Modified: trunk/Source/WebCore/Modules/webaudio/OfflineAudioDestinationNode.cpp (277552 => 277553)
--- trunk/Source/WebCore/Modules/webaudio/OfflineAudioDestinationNode.cpp 2021-05-15 23:55:58 UTC (rev 277552)
+++ trunk/Source/WebCore/Modules/webaudio/OfflineAudioDestinationNode.cpp 2021-05-16 01:05:18 UTC (rev 277553)
@@ -34,6 +34,7 @@
#include "AudioWorklet.h"
#include "AudioWorkletMessagingProxy.h"
#include "HRTFDatabaseLoader.h"
+#include "OfflineAudioContext.h"
#include "WorkerRunLoop.h"
#include <algorithm>
#include <wtf/IsoMallocInlines.h>
@@ -44,7 +45,7 @@
WTF_MAKE_ISO_ALLOCATED_IMPL(OfflineAudioDestinationNode);
-OfflineAudioDestinationNode::OfflineAudioDestinationNode(BaseAudioContext& context, unsigned numberOfChannels, float sampleRate, RefPtr<AudioBuffer>&& renderTarget)
+OfflineAudioDestinationNode::OfflineAudioDestinationNode(OfflineAudioContext& context, unsigned numberOfChannels, float sampleRate, RefPtr<AudioBuffer>&& renderTarget)
: AudioDestinationNode(context, sampleRate)
, m_numberOfChannels(numberOfChannels)
, m_renderTarget(WTFMove(renderTarget))
@@ -59,6 +60,16 @@
uninitialize();
}
+OfflineAudioContext& OfflineAudioDestinationNode::context()
+{
+ return downcast<OfflineAudioContext>(AudioDestinationNode::context());
+}
+
+const OfflineAudioContext& OfflineAudioDestinationNode::context() const
+{
+ return downcast<OfflineAudioContext>(AudioDestinationNode::context());
+}
+
unsigned OfflineAudioDestinationNode::maxChannelCount() const
{
return m_numberOfChannels;
Modified: trunk/Source/WebCore/Modules/webaudio/OfflineAudioDestinationNode.h (277552 => 277553)
--- trunk/Source/WebCore/Modules/webaudio/OfflineAudioDestinationNode.h 2021-05-15 23:55:58 UTC (rev 277552)
+++ trunk/Source/WebCore/Modules/webaudio/OfflineAudioDestinationNode.h 2021-05-16 01:05:18 UTC (rev 277553)
@@ -33,13 +33,19 @@
class AudioBus;
class AudioContext;
+class OfflineAudioContext;
class OfflineAudioDestinationNode final : public AudioDestinationNode {
WTF_MAKE_ISO_ALLOCATED(OfflineAudioDestinationNode);
public:
- OfflineAudioDestinationNode(BaseAudioContext&, unsigned numberOfChannels, float sampleRate, RefPtr<AudioBuffer>&& renderTarget);
+ OfflineAudioDestinationNode(OfflineAudioContext&, unsigned numberOfChannels, float sampleRate, RefPtr<AudioBuffer>&& renderTarget);
virtual ~OfflineAudioDestinationNode();
+
+ OfflineAudioContext& context();
+ const OfflineAudioContext& context() const;
+
+ AudioBuffer* renderTarget() const { return m_renderTarget.get(); }
// AudioNode
void initialize() override;
Modified: trunk/Source/WebCore/Modules/webaudio/ScriptProcessorNode.cpp (277552 => 277553)
--- trunk/Source/WebCore/Modules/webaudio/ScriptProcessorNode.cpp 2021-05-15 23:55:58 UTC (rev 277552)
+++ trunk/Source/WebCore/Modules/webaudio/ScriptProcessorNode.cpp 2021-05-16 01:05:18 UTC (rev 277553)
@@ -62,7 +62,7 @@
if (m_bufferSize < AudioUtilities::renderQuantumSize)
m_bufferSize = AudioUtilities::renderQuantumSize;
- ASSERT(numberOfInputChannels <= AudioContext::maxNumberOfChannels());
+ ASSERT(numberOfInputChannels <= AudioContext::maxNumberOfChannels);
initializeDefaultNodeOptions(numberOfInputChannels, ChannelCountMode::Explicit, ChannelInterpretation::Speakers);
addInput();
