Multi-Channel support in AudioBufferSourceNode

https://bugs.webkit.org/show_bug.cgi?id=79202 Patch by Wei James <james.wei@intel.com> on 2012-02-27 Reviewed by Chris Rogers. Source/WebCore: Test: webaudio/audiobuffersource-multi-channels.html * webaudio/AudioBufferSourceNode.cpp: (WebCore::AudioBufferSourceNode::renderSilenceAndFinishIfNotLooping): (WebCore::AudioBufferSourceNode::renderFromBuffer): (WebCore::AudioBufferSourceNode::setBuffer): * webaudio/AudioBufferSourceNode.h: (AudioBufferSourceNode): LayoutTests: * webaudio/audiobuffersource-channels-expected.txt: * webaudio/audiobuffersource-channels.html: * webaudio/audiobuffersource-multi-channels-expected.wav: Added. * webaudio/audiobuffersource-multi-channels.html: Added. Canonical link: https://commits.webkit.org/96837@main git-svn-id: https://svn.webkit.org/repository/webkit/trunk@109076 268f45cc-cd09-0410-ab3c-d52691b4dbfc
WebKit · Feb 28, 2012 · 1ee035610dca7d40b6fba0f67203dac4c3d264c4 · 1ee0356
1 parent e0be41c
commit 1ee035610dca7d40b6fba0f67203dac4c3d264c4
diff --git a/LayoutTests/ChangeLog b/LayoutTests/ChangeLog
@@ -1,3 +1,15 @@
+2012-02-27  Wei James  <james.wei@intel.com>
+
+        Multi-Channel support in AudioBufferSourceNode
+        https://bugs.webkit.org/show_bug.cgi?id=79202
+
+        Reviewed by Chris Rogers.
+
+        * webaudio/audiobuffersource-channels-expected.txt:
+        * webaudio/audiobuffersource-channels.html:
+        * webaudio/audiobuffersource-multi-channels-expected.wav: Added.
+        * webaudio/audiobuffersource-multi-channels.html: Added.
+
 2012-02-27  Emil A Eklund  <eae@chromium.org>
 
         Printed font-size should not be dependant on zoom level

diff --git a/LayoutTests/webaudio/audiobuffersource-channels-expected.txt b/LayoutTests/webaudio/audiobuffersource-channels-expected.txt
@@ -5,13 +5,13 @@ On success, you will see a series of "PASS" messages, followed by "TEST COMPLETE
 PASS source.buffer = 57 threw exception TypeError: Value is not of type AudioBuffer.
 PASS Mono buffer can be set.
 PASS Stereo buffer can be set.
-PASS 3 channel buffer is not settable.
-PASS 4 channel buffer is not settable.
-PASS 5 channel buffer is not settable.
-PASS 6 channel buffer is not settable.
-PASS 7 channel buffer is not settable.
-PASS 8 channel buffer is not settable.
-PASS 9 channel buffer is not settable.
+PASS 3 channels buffer can be set.
+PASS 4 channels buffer can be set.
+PASS 5 channels buffer can be set.
+PASS 6 channels buffer can be set.
+PASS 7 channels buffer can be set.
+PASS 8 channels buffer can be set.
+PASS 9 channels buffer can be set.
 PASS successfullyParsed is true
 
 TEST COMPLETE

diff --git a/LayoutTests/webaudio/audiobuffersource-channels.html b/LayoutTests/webaudio/audiobuffersource-channels.html
@@ -48,17 +48,16 @@
         testFailed("Stereo buffer can not be set.");
     }
 
-    // Check a few buffers with more than two channels and check for failure.
-    // (for now the implementation will only work with mono and stereo buffers)
+    // Check buffers with more than two channels.
     for (var i = 3; i < 10; ++i) {
         try {
             var buffer = context.createBuffer(i, 1024, context.sampleRate);
             source.buffer = buffer;
-            var message = i + " channel buffer should not be settable.";
-            testFailed(message);
-        } catch(e) {
-            var message = i + " channel buffer is not settable.";
+            var message = i + " channels buffer can be set.";
             testPassed(message);
+        } catch(e) {
+            var message = i + " channels buffer can not be set.";
+            testFailed(message);
         }
     }
 

diff --git a/LayoutTests/webaudio/audiobuffersource-multi-channels-expected.wav b/LayoutTests/webaudio/audiobuffersource-multi-channels-expected.wav
diff --git a/LayoutTests/webaudio/audiobuffersource-multi-channels.html b/LayoutTests/webaudio/audiobuffersource-multi-channels.html
@@ -0,0 +1,43 @@
+<!DOCTYPE html>
+
+<!--
+Test AudioBufferSourceNode supports 5.1 channel.
+-->
+
+<html>
+<head>
+<script type="text/javascript" src="resources/audio-testing.js"></script>
+<script type="text/javascript" src="resources/mix-testing.js"></script>
+</head>
+<body>
+
+<script>
+
+function runTest() {
+    if (!window.layoutTestController)
+        return;
+
+    layoutTestController.waitUntilDone();
+
+    window.jsTestAsync = true;
+
+    // Create offline audio context
+    var sampleRate = 44100.0;
+    var context = new webkitAudioContext(6, sampleRate * renderLengthSeconds, sampleRate);
+    var toneBuffer = createToneBuffer(context, 440, toneLengthSeconds, 6); 
+
+    var source = context.createBufferSource();
+    source.buffer = toneBuffer;
+
+    source.connect(context.destination);
+    source.noteOn(0);
+
+    context.oncomplete = finishAudioTest;
+    context.startRendering();
+}
+
+runTest();
+</script>
+
+</body>
+</html>
diff --git a/Source/WebCore/ChangeLog b/Source/WebCore/ChangeLog
@@ -1,3 +1,19 @@
+2012-02-27  Wei James  <james.wei@intel.com>
+
+        Multi-Channel support in AudioBufferSourceNode
+        https://bugs.webkit.org/show_bug.cgi?id=79202
+
+        Reviewed by Chris Rogers.
+
+        Test: webaudio/audiobuffersource-multi-channels.html
+
+        * webaudio/AudioBufferSourceNode.cpp:
+        (WebCore::AudioBufferSourceNode::renderSilenceAndFinishIfNotLooping):
+        (WebCore::AudioBufferSourceNode::renderFromBuffer):
+        (WebCore::AudioBufferSourceNode::setBuffer):
+        * webaudio/AudioBufferSourceNode.h:
+        (AudioBufferSourceNode):
+
 2012-02-27  Leo Yang  <leo.yang@torchmobile.com.cn>
 
         [BlackBerry] Upstream the BlackBerry change to platform/graphics/IntRect.h

diff --git a/Source/WebCore/webaudio/AudioBufferSourceNode.cpp b/Source/WebCore/webaudio/AudioBufferSourceNode.cpp
@@ -126,6 +126,9 @@ void AudioBufferSourceNode::process(size_t framesToProcess)
         quantumFrameOffset = min(quantumFrameOffset, framesToProcess); // clamp to valid range
         size_t bufferFramesToProcess = framesToProcess - quantumFrameOffset;
 
+        for (unsigned i = 0; i < outputBus->numberOfChannels(); ++i) 
+            m_destinationChannels[i] = outputBus->channel(i)->mutableData();
+
         // Render by reading directly from the buffer.
         renderFromBuffer(outputBus, quantumFrameOffset, bufferFramesToProcess);
 
@@ -144,7 +147,7 @@ void AudioBufferSourceNode::process(size_t framesToProcess)
 
             if (isSafe) {
                 for (unsigned i = 0; i < outputBus->numberOfChannels(); ++i)
-                    memset(outputBus->channel(i)->mutableData() + zeroStartFrame, 0, sizeof(float) * framesToZero);
+                    memset(m_destinationChannels[i] + zeroStartFrame, 0, sizeof(float) * framesToZero);
             }
 
             m_isPlaying = false;
@@ -160,7 +163,7 @@ void AudioBufferSourceNode::process(size_t framesToProcess)
 }
 
 // Returns true if we're finished.
-bool AudioBufferSourceNode::renderSilenceAndFinishIfNotLooping(float* destinationL, float* destinationR, size_t framesToProcess)
+bool AudioBufferSourceNode::renderSilenceAndFinishIfNotLooping(AudioBus*, unsigned index, size_t framesToProcess)
 {
     if (!loop()) {
         // If we're not looping, then stop playing when we get to the end.
@@ -169,10 +172,8 @@ bool AudioBufferSourceNode::renderSilenceAndFinishIfNotLooping(float* destinatio
         if (framesToProcess > 0) {
             // We're not looping and we've reached the end of the sample data, but we still need to provide more output,
             // so generate silence for the remaining.
-            memset(destinationL, 0, sizeof(float) * framesToProcess);
-
-            if (destinationR)
-                memset(destinationR, 0, sizeof(float) * framesToProcess);
+            for (unsigned i = 0; i < numberOfChannels(); ++i) 
+                memset(m_destinationChannels[i] + index, 0, sizeof(float) * framesToProcess);
         }
 
         finish();
@@ -194,21 +195,11 @@ void AudioBufferSourceNode::renderFromBuffer(AudioBus* bus, unsigned destination
     unsigned numberOfChannels = this->numberOfChannels();
     unsigned busNumberOfChannels = bus->numberOfChannels();
 
-    // FIXME: we can add support for sources with more than two channels, but this is not a common case.
-    bool channelCountGood = numberOfChannels == busNumberOfChannels && (numberOfChannels == 1 || numberOfChannels == 2);
+    bool channelCountGood = numberOfChannels && numberOfChannels == busNumberOfChannels;
     ASSERT(channelCountGood);
     if (!channelCountGood)
         return;
 
-    // Get the destination pointers.
-    float* destinationL = bus->channel(0)->mutableData();
-    ASSERT(destinationL);
-    if (!destinationL)
-        return;
-    float* destinationR = (numberOfChannels < 2) ? 0 : bus->channel(1)->mutableData();
-
-    bool isStereo = destinationR;
-
     // Sanity check destinationFrameOffset, numberOfFrames.
     size_t destinationLength = bus->length();
 
@@ -221,18 +212,15 @@ void AudioBufferSourceNode::renderFromBuffer(AudioBus* bus, unsigned destination
     ASSERT(isOffsetGood);
     if (!isOffsetGood)
         return;
-        
+
     // Potentially zero out initial frames leading up to the offset.
     if (destinationFrameOffset) {
-        memset(destinationL, 0, sizeof(float) * destinationFrameOffset);
-        if (destinationR)
-            memset(destinationR, 0, sizeof(float) * destinationFrameOffset);
+        for (unsigned i = 0; i < numberOfChannels; ++i) 
+            memset(m_destinationChannels[i], 0, sizeof(float) * destinationFrameOffset);
     }
 
     // Offset the pointers to the correct offset frame.
-    destinationL += destinationFrameOffset;
-    if (destinationR)
-        destinationR += destinationFrameOffset;
+    unsigned writeIndex = destinationFrameOffset;
 
     size_t bufferLength = buffer()->length();
     double bufferSampleRate = buffer()->sampleRate();
@@ -264,10 +252,6 @@ void AudioBufferSourceNode::renderFromBuffer(AudioBus* bus, unsigned destination
     if (m_virtualReadIndex >= endFrame)
         m_virtualReadIndex = startFrame; // reset to start
 
-    // Get pointers to the start of the sample buffer.
-    float* sourceL = m_buffer->getChannelData(0)->data();
-    float* sourceR = m_buffer->numberOfChannels() == 2 ? m_buffer->getChannelData(1)->data() : 0;
-
     double pitchRate = totalPitchRate();
 
     // Get local copy.
@@ -276,6 +260,9 @@ void AudioBufferSourceNode::renderFromBuffer(AudioBus* bus, unsigned destination
     // Render loop - reading from the source buffer to the destination using linear interpolation.
     int framesToProcess = numberOfFrames;
 
+    const float** sourceChannels = m_sourceChannels.get();
+    float** destinationChannels = m_destinationChannels.get();
+
     // Optimize for the very common case of playing back with pitchRate == 1.
     // We can avoid the linear interpolation.
     if (pitchRate == 1 && virtualReadIndex == floor(virtualReadIndex)) {
@@ -285,20 +272,17 @@ void AudioBufferSourceNode::renderFromBuffer(AudioBus* bus, unsigned destination
             int framesThisTime = min(framesToProcess, framesToEnd);
             framesThisTime = max(0, framesThisTime);
 
-            memcpy(destinationL, sourceL + readIndex, sizeof(*sourceL) * framesThisTime);
-            destinationL += framesThisTime;
-            if (isStereo) {
-                memcpy(destinationR, sourceR + readIndex, sizeof(*sourceR) * framesThisTime);
-                destinationR += framesThisTime;
-            }
+            for (unsigned i = 0; i < numberOfChannels; ++i) 
+                memcpy(destinationChannels[i] + writeIndex, sourceChannels[i] + readIndex, sizeof(float) * framesThisTime);
 
+            writeIndex += framesThisTime;
             readIndex += framesThisTime;
             framesToProcess -= framesThisTime;
 
             // Wrap-around.
             if (readIndex >= endFrame) {
                 readIndex -= deltaFrames;
-                if (renderSilenceAndFinishIfNotLooping(destinationL, destinationR, framesToProcess))
+                if (renderSilenceAndFinishIfNotLooping(bus, writeIndex, framesToProcess))
                     break;
             }
         }
@@ -324,25 +308,25 @@ void AudioBufferSourceNode::renderFromBuffer(AudioBus* bus, unsigned destination
                 break;
 
             // Linear interpolation.
-            double sampleL1 = sourceL[readIndex];
-            double sampleL2 = sourceL[readIndex2];
-            double sampleL = (1.0 - interpolationFactor) * sampleL1 + interpolationFactor * sampleL2;
-            *destinationL++ = narrowPrecisionToFloat(sampleL);
-
-            if (isStereo) {
-                double sampleR1 = sourceR[readIndex];
-                double sampleR2 = sourceR[readIndex2];
-                double sampleR = (1.0 - interpolationFactor) * sampleR1 + interpolationFactor * sampleR2;
-                *destinationR++ = narrowPrecisionToFloat(sampleR);
+            for (unsigned i = 0; i < numberOfChannels; ++i) {
+                float* destination = destinationChannels[i];
+                const float* source = sourceChannels[i];
+
+                double sample1 = source[readIndex];
+                double sample2 = source[readIndex2];
+                double sample = (1.0 - interpolationFactor) * sample1 + interpolationFactor * sample2;
+
+                destination[writeIndex] = narrowPrecisionToFloat(sample);
             }
+            writeIndex++;
 
             virtualReadIndex += pitchRate;
 
             // Wrap-around, retaining sub-sample position since virtualReadIndex is floating-point.
             if (virtualReadIndex >= endFrame) {
                 virtualReadIndex -= deltaFrames;
 
-                if (renderSilenceAndFinishIfNotLooping(destinationL, destinationR, framesToProcess))
+                if (renderSilenceAndFinishIfNotLooping(bus, writeIndex, framesToProcess))
                     break;
             }
         }
@@ -379,11 +363,13 @@ bool AudioBufferSourceNode::setBuffer(AudioBuffer* buffer)
     if (buffer) {
         // Do any necesssary re-configuration to the buffer's number of channels.
         unsigned numberOfChannels = buffer->numberOfChannels();
-        if (!numberOfChannels || numberOfChannels > 2) {
-            // FIXME: implement multi-channel greater than stereo.
-            return false;
-        }
         output(0)->setNumberOfChannels(numberOfChannels);
+
+        m_sourceChannels = adoptArrayPtr(new const float* [numberOfChannels]);
+        m_destinationChannels = adoptArrayPtr(new float* [numberOfChannels]);
+
+        for (unsigned i = 0; i < numberOfChannels; ++i) 
+            m_sourceChannels[i] = buffer->getChannelData(i)->data();
     }
 
     m_virtualReadIndex = 0;

diff --git a/Source/WebCore/webaudio/AudioBufferSourceNode.h b/Source/WebCore/webaudio/AudioBufferSourceNode.h
@@ -30,6 +30,7 @@
 #include "AudioGain.h"
 #include "AudioPannerNode.h"
 #include "AudioSourceNode.h"
+#include <wtf/OwnArrayPtr.h>
 #include <wtf/PassRefPtr.h>
 #include <wtf/RefPtr.h>
 #include <wtf/Threading.h>
@@ -87,11 +88,16 @@ class AudioBufferSourceNode : public AudioSourceNode {
 
     void renderFromBuffer(AudioBus*, unsigned destinationFrameOffset, size_t numberOfFrames);
 
-    inline bool renderSilenceAndFinishIfNotLooping(float* destinationL, float* destinationR, size_t framesToProcess);
+    // Render silence starting from "index" frame in AudioBus.
+    inline bool renderSilenceAndFinishIfNotLooping(AudioBus*, unsigned index, size_t framesToProcess);
 
     // m_buffer holds the sample data which this node outputs.
     RefPtr<AudioBuffer> m_buffer;
 
+    // Pointers for the buffer and destination.
+    OwnArrayPtr<const float*> m_sourceChannels;
+    OwnArrayPtr<float*> m_destinationChannels;
+
     // Used for the "gain" and "playbackRate" attributes.
     RefPtr<AudioGain> m_gain;
     RefPtr<AudioParam> m_playbackRate;