Diff
Modified: trunk/LayoutTests/ChangeLog (113768 => 113769)
--- trunk/LayoutTests/ChangeLog 2012-04-10 21:25:50 UTC (rev 113768)
+++ trunk/LayoutTests/ChangeLog 2012-04-10 21:28:57 UTC (rev 113769)
@@ -1,3 +1,16 @@
+2012-04-10 Chris Rogers <[email protected]>
+
+ AudioParam must support connections from audio-rate signals
+ https://bugs.webkit.org/show_bug.cgi?id=83524
+
+ Reviewed by Eric Carlson.
+
+ * webaudio/audioparam-connect-audioratesignal-expected.txt: Added.
+ * webaudio/audioparam-connect-audioratesignal.html: Added.
+ * webaudio/resources/audio-testing.js:
+ (createLinearRampBuffer):
+ (createConstantBuffer):
+
2012-04-10 Abhishek Arya <[email protected]>
Crash due to captions list not updated after section recalc.
Added: trunk/LayoutTests/webaudio/audioparam-connect-audioratesignal-expected.txt (0 => 113769)
--- trunk/LayoutTests/webaudio/audioparam-connect-audioratesignal-expected.txt (rev 0)
+++ trunk/LayoutTests/webaudio/audioparam-connect-audioratesignal-expected.txt 2012-04-10 21:28:57 UTC (rev 113769)
@@ -0,0 +1,6 @@
+PASS Rendered signal is of correct length.
+PASS Rendered signal exactly matches the audio-rate gain changing signal.
+PASS successfullyParsed is true
+
+TEST COMPLETE
+
Added: trunk/LayoutTests/webaudio/audioparam-connect-audioratesignal.html (0 => 113769)
--- trunk/LayoutTests/webaudio/audioparam-connect-audioratesignal.html (rev 0)
+++ trunk/LayoutTests/webaudio/audioparam-connect-audioratesignal.html 2012-04-10 21:28:57 UTC (rev 113769)
@@ -0,0 +1,112 @@
+<!DOCTYPE html>
+
+<!--
+Tests that an audio-rate signal (AudioNode output) can be connected to an AudioParam.
+Specifically, this tests that an audio-rate signal coming from an AudioBufferSourceNode
+playing an AudioBuffer containing a specific curve can be connected to an AudioGainNode's
+.gain attribute (an AudioParam). Another AudioBufferSourceNode will be the audio source
+having its gain changed. We load this one with an AudioBuffer containing a constant value of 1.
+Thus it's easy to check that the resultant signal should be equal to the gain-scaling curve.
+-->
+
+<html>
+<head>
+<link rel="stylesheet" href=""
+<script src=""
+<script src=""
+
+</head>
+<body>
+
+<script>
+
+var sampleRate = 44100.0;
+var lengthInSeconds = 1;
+
+var context = 0;
+var constantOneBuffer = 0;
+var linearRampBuffer = 0;
+
+function checkResult(event) {
+ var renderedBuffer = event.renderedBuffer;
+ var renderedData = renderedBuffer.getChannelData(0);
+ var expectedData = linearRampBuffer.getChannelData(0);
+ var n = renderedBuffer.length;
+
+ if (n == linearRampBuffer.length) {
+ testPassed("Rendered signal is of correct length.");
+ } else {
+ testFailed("Rendered signal is not of correct length.");
+ }
+
+ // Check that the rendered result exactly matches the buffer used to control gain.
+ // This is because we're changing the gain of a signal having constant value 1.
+ var success = true;
+ for (var i = 0; i < n; ++i) {
+ if (renderedData[i] != expectedData[i]) {
+ success = false;
+ break;
+ }
+ }
+
+ if (success) {
+ testPassed("Rendered signal exactly matches the audio-rate gain changing signal.");
+ } else {
+ testFailed("Rendered signal differs from the audio-rate gain changing signal.");
+ }
+
+ finishJSTest();
+}
+
+function runTest() {
+ if (window.layoutTestController) {
+ layoutTestController.dumpAsText();
+ layoutTestController.waitUntilDone();
+ }
+
+ window.jsTestIsAsync = true;
+
+ var sampleFrameLength = sampleRate * lengthInSeconds;
+
+ // Create offline audio context.
+ context = new webkitAudioContext(1, sampleFrameLength, sampleRate);
+
+ // Create buffer used by the source which will have its gain controlled.
+ constantOneBuffer = createConstantBuffer(context, sampleFrameLength, 1);
+
+ // Create buffer used to control gain.
+ linearRampBuffer = createLinearRampBuffer(context, sampleFrameLength);
+
+ // Create the two sources.
+
+ var constantSource = context.createBufferSource();
+ constantSource.buffer = constantOneBuffer;
+
+ var gainChangingSource = context.createBufferSource();
+ gainChangingSource.buffer = linearRampBuffer;
+
+ // Create a gain node controlling the gain of constantSource and make the connections.
+ var gainNode = context.createGainNode();
+ constantSource.connect(gainNode);
+ gainNode.connect(context.destination);
+
+ // Connect an audio-rate signal to control the .gain AudioParam.
+ // This is the heart of what is being tested.
+ gainChangingSource.connect(gainNode.gain);
+
+ // Start both sources at time 0.
+ constantSource.noteOn(0);
+ gainChangingSource.noteOn(0);
+
+ context._oncomplete_ = checkResult;
+ context.startRendering();
+}
+
+runTest();
+successfullyParsed = true;
+
+</script>
+<script src=""
+
+</body>
+</html>
Modified: trunk/LayoutTests/webaudio/resources/audio-testing.js (113768 => 113769)
--- trunk/LayoutTests/webaudio/resources/audio-testing.js 2012-04-10 21:25:50 UTC (rev 113768)
+++ trunk/LayoutTests/webaudio/resources/audio-testing.js 2012-04-10 21:28:57 UTC (rev 113769)
@@ -115,6 +115,30 @@
return audioBuffer;
}
+// Create a buffer of the given length with a linear ramp having values 0 <= x < 1.
+function createLinearRampBuffer(context, sampleFrameLength) {
+ var audioBuffer = context.createBuffer(1, sampleFrameLength, context.sampleRate);
+ var n = audioBuffer.length;
+ var dataL = audioBuffer.getChannelData(0);
+
+ for (var i = 0; i < n; ++i)
+ dataL[i] = i / n;
+
+ return audioBuffer;
+}
+
+// Create a buffer of the given length having a constant value.
+function createConstantBuffer(context, sampleFrameLength, constantValue) {
+ var audioBuffer = context.createBuffer(1, sampleFrameLength, context.sampleRate);
+ var n = audioBuffer.length;
+ var dataL = audioBuffer.getChannelData(0);
+
+ for (var i = 0; i < n; ++i)
+ dataL[i] = constantValue;
+
+ return audioBuffer;
+}
+
// Convert time (in seconds) to sample frames.
function timeToSampleFrame(time, sampleRate) {
return Math.floor(0.5 + time * sampleRate);
Modified: trunk/Source/WebCore/ChangeLog (113768 => 113769)
--- trunk/Source/WebCore/ChangeLog 2012-04-10 21:25:50 UTC (rev 113768)
+++ trunk/Source/WebCore/ChangeLog 2012-04-10 21:28:57 UTC (rev 113769)
@@ -1,3 +1,69 @@
+2012-04-10 Chris Rogers <[email protected]>
+
+ AudioParam must support connections from audio-rate signals
+ https://bugs.webkit.org/show_bug.cgi?id=83524
+
+ Reviewed by Eric Carlson.
+
+ In the Web Audio API, it's possible to connect one AudioNode to another AudioNode.
+ Similary we should allow an AudioNode to connect to an AudioParam, thus controlling
+ a parameter with an audio-rate signal. This is important in many audio processing
+ applications.
+
+ Test: webaudio/audioparam-connect-audioratesignal.html
+
+ Simple method name change of AudioParam::hasTimelineValues() to AudioParam::hasSampleAccurateValues().
+ * Modules/webaudio/AudioGainNode.cpp:
+ (WebCore::AudioGainNode::process):
+
+ * Modules/webaudio/AudioNode.cpp:
+ (WebCore::AudioNode::connect): Add connect() method from AudioNode -> AudioParam.
+ (WebCore):
+ (WebCore::AudioNode::disconnect):
+ (WebCore::AudioNode::finishDeref):
+ Use AudioNodeOutput::disconnectAll() instead of AudioNodeOutput::disconnectAllInputs().
+ * Modules/webaudio/AudioNode.h: Add connect() method from AudioNode -> AudioParam.
+ (WebCore):
+ (AudioNode):
+ * Modules/webaudio/AudioNode.idl: Add connect() method from AudioNode -> AudioParam.
+
+ Implement support for an AudioNodeOutput to fanout to multiple AudioParams.
+ * Modules/webaudio/AudioNodeOutput.cpp:
+ (WebCore::AudioNodeOutput::AudioNodeOutput):
+ (WebCore::AudioNodeOutput::updateRenderingState): Update rendering state related to AudioParams.
+ (WebCore::AudioNodeOutput::pull): pull() must now take into account fanout to AudioParams for in-place processing.
+ (WebCore::AudioNodeOutput::fanOutCount):
+ (WebCore):
+ (WebCore::AudioNodeOutput::paramFanOutCount): New method keeping track of number of connections to AudioParams.
+ (WebCore::AudioNodeOutput::renderingParamFanOutCount): New method keeping track of number of connections to AudioParams for rendering.
+ (WebCore::AudioNodeOutput::addParam): Add a connection to an AudioParam.
+ (WebCore::AudioNodeOutput::removeParam): Remove a connection to an AudioParam.
+ (WebCore::AudioNodeOutput::disconnectAllParams): Remove all connections to AudioParams.
+ (WebCore::AudioNodeOutput::disconnectAll): New method to disconnect all AudioNodeInputs and AudioParams.
+ * Modules/webaudio/AudioNodeOutput.h:
+ (AudioNodeOutput):
+
+ Allow an AudioParam to accept a connection from an AudioNodeOutput, thus being controlled
+ by an audio-rate signal.
+ * Modules/webaudio/AudioParam.cpp:
+ (WebCore::AudioParam::calculateSampleAccurateValues): Calculates sample-accurate values from timeline or an AudioNode.
+ (WebCore):
+ (WebCore::AudioParam::calculateAudioRateSignalValues): Calculates sample-accurate values from an AudioNode.
+ (WebCore::AudioParam::calculateTimelineValues): Calculates sample-accurate values scheduled on the timeline.
+ (WebCore::AudioParam::connect): Connect from an AudioNodeOutput for control from an audio-rate signal.
+ (WebCore::AudioParam::disconnect): Disconnect from an AudioNodeOutput.
+ * Modules/webaudio/AudioParam.h:
+ (WebCore):
+ (WebCore::AudioParam::AudioParam):
+ (WebCore::AudioParam::hasSampleAccurateValues): Change name from hasTimelineValues() and return true
+ either if we have timeline values or if we've been connected from an AudioNode.
+ (AudioParam):
+
+ Simple method name change of AudioParam::hasTimelineValues() to AudioParam::hasSampleAccurateValues().
+ * Modules/webaudio/Oscillator.cpp:
+ (WebCore::Oscillator::calculateSampleAccuratePhaseIncrements):
+ (WebCore::Oscillator::process):
+
2012-04-10 Patrick Gansterer <[email protected]>
[CMake] Enable USE_FOLDERS property
Modified: trunk/Source/WebCore/Modules/webaudio/AudioGainNode.cpp (113768 => 113769)
--- trunk/Source/WebCore/Modules/webaudio/AudioGainNode.cpp 2012-04-10 21:25:50 UTC (rev 113768)
+++ trunk/Source/WebCore/Modules/webaudio/AudioGainNode.cpp 2012-04-10 21:28:57 UTC (rev 113769)
@@ -64,7 +64,7 @@
else {
AudioBus* inputBus = input(0)->bus();
- if (gain()->hasTimelineValues()) {
+ if (gain()->hasSampleAccurateValues()) {
// Apply sample-accurate gain scaling for precise envelopes, grain windows, etc.
ASSERT(framesToProcess <= m_sampleAccurateGainValues.size());
if (framesToProcess <= m_sampleAccurateGainValues.size()) {
Modified: trunk/Source/WebCore/Modules/webaudio/AudioNode.cpp (113768 => 113769)
--- trunk/Source/WebCore/Modules/webaudio/AudioNode.cpp 2012-04-10 21:25:50 UTC (rev 113768)
+++ trunk/Source/WebCore/Modules/webaudio/AudioNode.cpp 2012-04-10 21:28:57 UTC (rev 113769)
@@ -31,6 +31,7 @@
#include "AudioContext.h"
#include "AudioNodeInput.h"
#include "AudioNodeOutput.h"
+#include "AudioParam.h"
#include "ExceptionCode.h"
#include <wtf/Atomics.h>
#include <wtf/MainThread.h>
@@ -149,6 +150,30 @@
context()->incrementConnectionCount();
}
+void AudioNode::connect(AudioParam* param, unsigned outputIndex, ExceptionCode& ec)
+{
+ ASSERT(isMainThread());
+ AudioContext::AutoLocker locker(context());
+
+ if (!param) {
+ ec = SYNTAX_ERR;
+ return;
+ }
+
+ if (outputIndex >= numberOfOutputs()) {
+ ec = INDEX_SIZE_ERR;
+ return;
+ }
+
+ if (context() != param->context()) {
+ ec = SYNTAX_ERR;
+ return;
+ }
+
+ AudioNodeOutput* output = this->output(outputIndex);
+ param->connect(output);
+}
+
void AudioNode::disconnect(unsigned outputIndex, ExceptionCode& ec)
{
ASSERT(isMainThread());
@@ -161,7 +186,7 @@
}
AudioNodeOutput* output = this->output(outputIndex);
- output->disconnectAllInputs();
+ output->disconnectAll();
}
void AudioNode::processIfNecessary(size_t framesToProcess)
@@ -337,7 +362,7 @@
if (!m_isMarkedForDeletion) {
// All references are gone - we need to go away.
for (unsigned i = 0; i < m_outputs.size(); ++i)
- output(i)->disconnectAllInputs(); // this will deref() nodes we're connected to...
+ output(i)->disconnectAll(); // This will deref() nodes we're connected to.
// Mark for deletion at end of each render quantum or when context shuts down.
context()->markForDeletion(this);
Modified: trunk/Source/WebCore/Modules/webaudio/AudioNode.h (113768 => 113769)
--- trunk/Source/WebCore/Modules/webaudio/AudioNode.h 2012-04-10 21:25:50 UTC (rev 113768)
+++ trunk/Source/WebCore/Modules/webaudio/AudioNode.h 2012-04-10 21:28:57 UTC (rev 113769)
@@ -37,6 +37,7 @@
class AudioContext;
class AudioNodeInput;
class AudioNodeOutput;
+class AudioParam;
typedef int ExceptionCode;
@@ -116,6 +117,7 @@
// Called from main thread by corresponding _javascript_ methods.
void connect(AudioNode*, unsigned outputIndex, unsigned inputIndex, ExceptionCode&);
+ void connect(AudioParam*, unsigned outputIndex, ExceptionCode&);
void disconnect(unsigned outputIndex, ExceptionCode&);
virtual float sampleRate() const { return m_sampleRate; }
Modified: trunk/Source/WebCore/Modules/webaudio/AudioNode.idl (113768 => 113769)
--- trunk/Source/WebCore/Modules/webaudio/AudioNode.idl 2012-04-10 21:25:50 UTC (rev 113768)
+++ trunk/Source/WebCore/Modules/webaudio/AudioNode.idl 2012-04-10 21:28:57 UTC (rev 113769)
@@ -33,6 +33,9 @@
void connect(in AudioNode destination, in [Optional=DefaultIsUndefined] unsigned long output, in [Optional=DefaultIsUndefined] unsigned long input)
raises(DOMException);
+ void connect(in AudioParam destination, in [Optional=DefaultIsUndefined] unsigned long output)
+ raises(DOMException);
+
void disconnect(in [Optional=DefaultIsUndefined] unsigned long output)
raises(DOMException);
};
Modified: trunk/Source/WebCore/Modules/webaudio/AudioNodeOutput.cpp (113768 => 113769)
--- trunk/Source/WebCore/Modules/webaudio/AudioNodeOutput.cpp 2012-04-10 21:25:50 UTC (rev 113768)
+++ trunk/Source/WebCore/Modules/webaudio/AudioNodeOutput.cpp 2012-04-10 21:28:57 UTC (rev 113769)
@@ -31,6 +31,7 @@
#include "AudioBus.h"
#include "AudioContext.h"
#include "AudioNodeInput.h"
+#include "AudioParam.h"
#include <wtf/Threading.h>
namespace WebCore {
@@ -42,6 +43,7 @@
, m_actualDestinationBus(0)
, m_isEnabled(true)
, m_renderingFanOutCount(0)
+ , m_renderingParamFanOutCount(0)
{
ASSERT(numberOfChannels <= AudioContext::maxNumberOfChannels());
@@ -80,6 +82,7 @@
{
updateNumberOfChannels();
m_renderingFanOutCount = fanOutCount();
+ m_renderingParamFanOutCount = paramFanOutCount();
}
void AudioNodeOutput::updateNumberOfChannels()
@@ -110,7 +113,7 @@
AudioBus* AudioNodeOutput::pull(AudioBus* inPlaceBus, size_t framesToProcess)
{
ASSERT(context()->isAudioThread());
- ASSERT(m_renderingFanOutCount > 0);
+ ASSERT(m_renderingFanOutCount > 0 || m_renderingParamFanOutCount > 0);
// Causes our AudioNode to process if it hasn't already for this render quantum.
// We try to do in-place processing (using inPlaceBus) if at all possible,
@@ -118,7 +121,7 @@
// In this case pull() is called multiple times per rendering quantum, and the processIfNecessary() call below will
// cause our node to process() only the first time, caching the output in m_internalOutputBus for subsequent calls.
- bool isInPlace = inPlaceBus && inPlaceBus->numberOfChannels() == numberOfChannels() && m_renderingFanOutCount == 1;
+ bool isInPlace = inPlaceBus && inPlaceBus->numberOfChannels() == numberOfChannels() && (m_renderingFanOutCount + m_renderingParamFanOutCount) == 1;
// Setup the actual destination bus for processing when our node's process() method gets called in processIfNecessary() below.
m_actualDestinationBus = isInPlace ? inPlaceBus : m_internalBus.get();
@@ -134,15 +137,26 @@
return m_actualDestinationBus;
}
+unsigned AudioNodeOutput::fanOutCount()
+{
+ ASSERT(context()->isGraphOwner());
+ return m_inputs.size();
+}
+
+unsigned AudioNodeOutput::paramFanOutCount()
+{
+ ASSERT(context()->isGraphOwner());
+ return m_params.size();
+}
+
unsigned AudioNodeOutput::renderingFanOutCount() const
{
return m_renderingFanOutCount;
}
-unsigned AudioNodeOutput::fanOutCount()
+unsigned AudioNodeOutput::renderingParamFanOutCount() const
{
- ASSERT(context()->isGraphOwner());
- return m_inputs.size();
+ return m_renderingParamFanOutCount;
}
void AudioNodeOutput::addInput(AudioNodeInput* input)
@@ -178,6 +192,46 @@
}
}
+void AudioNodeOutput::addParam(AudioParam* param)
+{
+ ASSERT(context()->isGraphOwner());
+
+ ASSERT(param);
+ if (!param)
+ return;
+
+ m_params.add(param);
+}
+
+void AudioNodeOutput::removeParam(AudioParam* param)
+{
+ ASSERT(context()->isGraphOwner());
+
+ ASSERT(param);
+ if (!param)
+ return;
+
+ m_params.remove(param);
+}
+
+void AudioNodeOutput::disconnectAllParams()
+{
+ ASSERT(context()->isGraphOwner());
+
+ for (ParamsIterator i = m_params.begin(); i != m_params.end(); ++i) {
+ AudioParam* param = *i;
+ param->disconnect(this);
+ }
+
+ m_params.clear();
+}
+
+void AudioNodeOutput::disconnectAll()
+{
+ disconnectAllInputs();
+ disconnectAllParams();
+}
+
void AudioNodeOutput::disable()
{
ASSERT(context()->isGraphOwner());
Modified: trunk/Source/WebCore/Modules/webaudio/AudioNodeOutput.h (113768 => 113769)
--- trunk/Source/WebCore/Modules/webaudio/AudioNodeOutput.h 2012-04-10 21:25:50 UTC (rev 113768)
+++ trunk/Source/WebCore/Modules/webaudio/AudioNodeOutput.h 2012-04-10 21:28:57 UTC (rev 113769)
@@ -57,18 +57,17 @@
// Called from context's audio thread.
AudioBus* bus() const;
- // fanOutCount() is the number of AudioNodeInputs that we're connected to.
- // This function should not be called in audio thread rendering code, instead renderingFanOutCount() should be used.
- // It must be called with the context's graph lock.
- unsigned fanOutCount();
-
// renderingFanOutCount() is the number of AudioNodeInputs that we're connected to during rendering.
// Unlike fanOutCount() it will not change during the course of a render quantum.
unsigned renderingFanOutCount() const;
- // It must be called with the context's graph lock.
- void disconnectAllInputs();
+ // renderingParamFanOutCount() is the number of AudioParams that we're connected to during rendering.
+ // Unlike paramFanOutCount() it will not change during the course of a render quantum.
+ unsigned renderingParamFanOutCount() const;
+ // Must be called with the context's graph lock.
+ void disconnectAll();
+
void setNumberOfChannels(unsigned);
unsigned numberOfChannels() const { return m_numberOfChannels; }
bool isChannelCountKnown() const { return numberOfChannels() > 0; }
@@ -87,12 +86,29 @@
AudioNode* m_node;
friend class AudioNodeInput;
+ friend class AudioParam;
// These are called from AudioNodeInput.
// They must be called with the context's graph lock.
void addInput(AudioNodeInput*);
void removeInput(AudioNodeInput*);
+ void addParam(AudioParam*);
+ void removeParam(AudioParam*);
+ // fanOutCount() is the number of AudioNodeInputs that we're connected to.
+ // This method should not be called in audio thread rendering code, instead renderingFanOutCount() should be used.
+ // It must be called with the context's graph lock.
+ unsigned fanOutCount();
+
+ // Similar to fanOutCount(), paramFanOutCount() is the number of AudioParams that we're connected to.
+ // This method should not be called in audio thread rendering code, instead renderingParamFanOutCount() should be used.
+ // It must be called with the context's graph lock.
+ unsigned paramFanOutCount();
+
+ // Must be called with the context's graph lock.
+ void disconnectAllInputs();
+ void disconnectAllParams();
+
// updateInternalBus() updates m_internalBus appropriately for the number of channels.
// It is called in the constructor or in the audio thread with the context's graph lock.
void updateInternalBus();
@@ -120,10 +136,14 @@
HashSet<AudioNodeInput*> m_inputs;
typedef HashSet<AudioNodeInput*>::iterator InputsIterator;
bool m_isEnabled;
-
- // For the purposes of rendering, keeps track of the number of inputs we're connected to.
- // This value should only be changed at the very start or end of the rendering quantum.
+
+ // For the purposes of rendering, keeps track of the number of inputs and AudioParams we're connected to.
+ // These value should only be changed at the very start or end of the rendering quantum.
unsigned m_renderingFanOutCount;
+ unsigned m_renderingParamFanOutCount;
+
+ HashSet<AudioParam*> m_params;
+ typedef HashSet<AudioParam*>::iterator ParamsIterator;
};
} // namespace WebCore
Modified: trunk/Source/WebCore/Modules/webaudio/AudioParam.cpp (113768 => 113769)
--- trunk/Source/WebCore/Modules/webaudio/AudioParam.cpp 2012-04-10 21:25:50 UTC (rev 113768)
+++ trunk/Source/WebCore/Modules/webaudio/AudioParam.cpp 2012-04-10 21:28:57 UTC (rev 113769)
@@ -30,6 +30,7 @@
#include "AudioParam.h"
#include "AudioNode.h"
+#include "AudioNodeOutput.h"
#include "AudioUtilities.h"
#include "FloatConversion.h"
#include <wtf/MathExtras.h>
@@ -95,11 +96,43 @@
void AudioParam::calculateSampleAccurateValues(float* values, unsigned numberOfValues)
{
- bool isSafe = context() && context()->isAudioThread() && values;
+ bool isSafe = context() && context()->isAudioThread() && values && numberOfValues;
ASSERT(isSafe);
if (!isSafe)
return;
+ if (m_audioRateSignal)
+ calculateAudioRateSignalValues(values, numberOfValues);
+ else
+ calculateTimelineValues(values, numberOfValues);
+}
+
+void AudioParam::calculateAudioRateSignalValues(float* values, unsigned numberOfValues)
+{
+ // FIXME: support fan-in (multiple audio connections to this parameter with unity-gain summing).
+ // https://bugs.webkit.org/show_bug.cgi?id=83610
+ ASSERT(m_audioRateSignal);
+
+ AudioBus* bus = m_audioRateSignal->pull(0, numberOfValues);
+ bool isBusGood = bus && bus->numberOfChannels() && bus->length() >= numberOfValues;
+ ASSERT(isBusGood);
+ if (!isBusGood)
+ return;
+
+ if (bus->numberOfChannels() == 1) {
+ // The normal case is to deal with a mono audio-rate signal.
+ memcpy(values, bus->channel(0)->data(), sizeof(float) * numberOfValues);
+ } else {
+ // Do a standard mixdown to one channel if necessary.
+ AudioBus wrapperBus(1, numberOfValues, false);
+ wrapperBus.setChannelMemory(0, values, numberOfValues);
+ wrapperBus.copyFrom(*bus); // Mixdown.
+ }
+ m_value = values[0]; // Update to first value.
+}
+
+void AudioParam::calculateTimelineValues(float* values, unsigned numberOfValues)
+{
// Calculate values for this render quantum.
// Normally numberOfValues will equal AudioNode::ProcessingSizeInFrames (the render quantum size).
float sampleRate = context()->sampleRate();
@@ -111,6 +144,35 @@
m_value = m_timeline.valuesForTimeRange(startTime, endTime, narrowPrecisionToFloat(m_value), values, numberOfValues, sampleRate, sampleRate);
}
+void AudioParam::connect(AudioNodeOutput* audioRateSignal)
+{
+ ASSERT(context()->isGraphOwner());
+ ASSERT(audioRateSignal);
+ if (!audioRateSignal)
+ return;
+
+ if (m_audioRateSignal && m_audioRateSignal != audioRateSignal) {
+ // Because we don't currently support fan-in we must explicitly disconnect from an old output.
+ m_audioRateSignal->removeParam(this);
+ }
+
+ audioRateSignal->addParam(this);
+ m_audioRateSignal = audioRateSignal;
+}
+
+void AudioParam::disconnect(AudioNodeOutput* audioRateSignal)
+{
+ ASSERT(context()->isGraphOwner());
+ ASSERT(audioRateSignal);
+ if (!audioRateSignal)
+ return;
+
+ // FIXME: support fan-in (multiple audio connections to this parameter with unity-gain summing).
+ // https://bugs.webkit.org/show_bug.cgi?id=83610
+ if (m_audioRateSignal == audioRateSignal)
+ m_audioRateSignal = 0;
+}
+
} // namespace WebCore
#endif // ENABLE(WEB_AUDIO)
Modified: trunk/Source/WebCore/Modules/webaudio/AudioParam.h (113768 => 113769)
--- trunk/Source/WebCore/Modules/webaudio/AudioParam.h 2012-04-10 21:25:50 UTC (rev 113768)
+++ trunk/Source/WebCore/Modules/webaudio/AudioParam.h 2012-04-10 21:28:57 UTC (rev 113769)
@@ -39,6 +39,8 @@
namespace WebCore {
+class AudioNodeOutput;
+
class AudioParam : public RefCounted<AudioParam> {
public:
static const double DefaultSmoothingConstant;
@@ -58,6 +60,7 @@
, m_units(units)
, m_smoothedValue(defaultValue)
, m_smoothingConstant(DefaultSmoothingConstant)
+ , m_audioRateSignal(0)
{
}
@@ -96,13 +99,20 @@
void setValueCurveAtTime(Float32Array* curve, float time, float duration) { m_timeline.setValueCurveAtTime(curve, time, duration); }
void cancelScheduledValues(float startTime) { m_timeline.cancelScheduledValues(startTime); }
- bool hasTimelineValues() { return m_timeline.hasValues(); }
+ bool hasSampleAccurateValues() { return m_timeline.hasValues() || m_audioRateSignal; }
// Calculates numberOfValues parameter values starting at the context's current time.
// Must be called in the context's render thread.
void calculateSampleAccurateValues(float* values, unsigned numberOfValues);
+ // Connect an audio-rate signal to control this parameter.
+ void connect(AudioNodeOutput*);
+ void disconnect(AudioNodeOutput*);
+
private:
+ void calculateAudioRateSignalValues(float* values, unsigned numberOfValues);
+ void calculateTimelineValues(float* values, unsigned numberOfValues);
+
RefPtr<AudioContext> m_context;
String m_name;
double m_value;
@@ -116,6 +126,11 @@
double m_smoothingConstant;
AudioParamTimeline m_timeline;
+
+ // An audio-rate signal directly providing parameter values.
+ // FIXME: support fan-in (multiple audio connections to this parameter with unity-gain summing).
+ // https://bugs.webkit.org/show_bug.cgi?id=83610
+ AudioNodeOutput* m_audioRateSignal;
};
} // namespace WebCore
Modified: trunk/Source/WebCore/Modules/webaudio/Oscillator.cpp (113768 => 113769)
--- trunk/Source/WebCore/Modules/webaudio/Oscillator.cpp 2012-04-10 21:25:50 UTC (rev 113768)
+++ trunk/Source/WebCore/Modules/webaudio/Oscillator.cpp 2012-04-10 21:28:57 UTC (rev 113769)
@@ -112,14 +112,14 @@
if (!isGood)
return false;
- bool hasTimelineValues = false;
+ bool hasSampleAccurateValues = false;
bool hasFrequencyChanges = false;
float* phaseIncrements = m_phaseIncrements.data();
float finalScale = m_waveTable->rateScale();
- if (m_frequency->hasTimelineValues()) {
- hasTimelineValues = true;
+ if (m_frequency->hasSampleAccurateValues()) {
+ hasSampleAccurateValues = true;
hasFrequencyChanges = true;
// Get the sample-accurate frequency values and convert to phase increments.
@@ -132,8 +132,8 @@
finalScale *= frequency;
}
- if (m_detune->hasTimelineValues()) {
- hasTimelineValues = true;
+ if (m_detune->hasSampleAccurateValues()) {
+ hasSampleAccurateValues = true;
// Get the sample-accurate detune values.
float* detuneValues = hasFrequencyChanges ? m_detuneValues.data() : phaseIncrements;
@@ -157,20 +157,18 @@
finalScale *= detuneScale;
}
- if (hasTimelineValues) {
+ if (hasSampleAccurateValues) {
// Convert from frequency to wavetable increment.
vsmul(phaseIncrements, 1, &finalScale, phaseIncrements, 1, framesToProcess);
}
- return hasTimelineValues;
+ return hasSampleAccurateValues;
}
void Oscillator::process(size_t framesToProcess)
{
AudioBus* outputBus = output(0)->bus();
- outputBus->zero();
-
if (!isInitialized() || !outputBus->numberOfChannels()) {
outputBus->zero();
return;
@@ -206,14 +204,14 @@
float rateScale = m_waveTable->rateScale();
float invRateScale = 1 / rateScale;
- bool hasTimelineValues = calculateSampleAccuratePhaseIncrements(framesToProcess);
+ bool hasSampleAccurateValues = calculateSampleAccuratePhaseIncrements(framesToProcess);
float frequency = 0;
float* higherWaveData = 0;
float* lowerWaveData = 0;
float tableInterpolationFactor;
- if (!hasTimelineValues) {
+ if (!hasSampleAccurateValues) {
frequency = m_frequency->smoothedValue();
float detune = m_detune->smoothedValue();
float detuneScale = powf(2, detune / 1200);
@@ -234,7 +232,7 @@
readIndex = readIndex & readIndexMask;
readIndex2 = readIndex2 & readIndexMask;
- if (hasTimelineValues) {
+ if (hasSampleAccurateValues) {
incr = *phaseIncrements++;
frequency = invRateScale * incr;
