Implement MediaElementAudioSourceNode::setFormat() so numberOfChannels and sampleRate are accounted for

crogers@google.com · crogers@google.com · commit 6dbc720ab63c · 2011-12-22T22:40:00.000Z
https://bugs.webkit.org/show_bug.cgi?id=75057 Reviewed by Eric Carlson. * GNUmakefile.list.am: * WebCore.gypi: * WebCore.xcodeproj/project.pbxproj: Add MultiChannelResampler source files to makefiles. * platform/audio/MultiChannelResampler.cpp: Added. (WebCore::MultiChannelResampler::MultiChannelResampler): (WebCore::MultiChannelResampler::process): * platform/audio/MultiChannelResampler.h: Added. Add MultiChannelResampler implementation which uses one SincResampler per channel. * webaudio/MediaElementAudioSourceNode.cpp: (WebCore::MediaElementAudioSourceNode::create): (WebCore::MediaElementAudioSourceNode::MediaElementAudioSourceNode): (WebCore::MediaElementAudioSourceNode::setFormat): (WebCore::MediaElementAudioSourceNode::process): Implement MediaElementAudioSourceNode::setFormat() so that we can properly setup a sample-rate converter and set the number of channels of the MediaElementAudioSourceNode output. * webaudio/MediaElementAudioSourceNode.h: git-svn-id: svn://svn.chromium.org/blink/trunk@103581 bbb929c8-8fbe-4397-9dbb-9b2b20218538
diff --git a/third_party/WebKit/Source/WebCore/ChangeLog b/third_party/WebKit/Source/WebCore/ChangeLog
@@ -1,3 +1,29 @@
+2011-12-22  Chris Rogers  <crogers@google.com>
+
+        Implement MediaElementAudioSourceNode::setFormat() so numberOfChannels and sampleRate are accounted for
+        https://bugs.webkit.org/show_bug.cgi?id=75057
+
+        Reviewed by Eric Carlson.
+
+        * GNUmakefile.list.am:
+        * WebCore.gypi:
+        * WebCore.xcodeproj/project.pbxproj:
+        Add MultiChannelResampler source files to makefiles.
+        * platform/audio/MultiChannelResampler.cpp: Added.
+        (WebCore::MultiChannelResampler::MultiChannelResampler):
+        (WebCore::MultiChannelResampler::process):
+        * platform/audio/MultiChannelResampler.h: Added.
+        Add MultiChannelResampler implementation which uses one SincResampler per channel.
+        * webaudio/MediaElementAudioSourceNode.cpp:
+        (WebCore::MediaElementAudioSourceNode::create):
+        (WebCore::MediaElementAudioSourceNode::MediaElementAudioSourceNode):
+        (WebCore::MediaElementAudioSourceNode::setFormat):
+        (WebCore::MediaElementAudioSourceNode::process):
+        Implement MediaElementAudioSourceNode::setFormat() so that we can
+        properly setup a sample-rate converter and set the number of channels
+        of the MediaElementAudioSourceNode output.
+        * webaudio/MediaElementAudioSourceNode.h:
+
 2011-12-22  Chris Fleizach  <cfleizach@apple.com>
 
         AX: WebKit should ignore ARIA role=presentation on focusable elements
diff --git a/third_party/WebKit/Source/WebCore/GNUmakefile.list.am b/third_party/WebKit/Source/WebCore/GNUmakefile.list.am
@@ -4829,6 +4829,8 @@ webcore_sources += \
 	Source/WebCore/platform/audio/HRTFKernel.h \
 	Source/WebCore/platform/audio/HRTFPanner.cpp \
 	Source/WebCore/platform/audio/HRTFPanner.h \
+	Source/WebCore/platform/audio/MultiChannelResampler.cpp \
+	Source/WebCore/platform/audio/MultiChannelResampler.h \
 	Source/WebCore/platform/audio/Panner.cpp \
 	Source/WebCore/platform/audio/Panner.h \
 	Source/WebCore/platform/audio/ReverbAccumulationBuffer.cpp \
diff --git a/third_party/WebKit/Source/WebCore/WebCore.gypi b/third_party/WebKit/Source/WebCore/WebCore.gypi
@@ -3217,6 +3217,8 @@
             'platform/audio/HRTFKernel.h',
             'platform/audio/HRTFPanner.cpp',
             'platform/audio/HRTFPanner.h',
+            'platform/audio/MultiChannelResampler.cpp',
+            'platform/audio/MultiChannelResampler.h',
             'platform/audio/Panner.cpp',
             'platform/audio/Panner.h',
             'platform/audio/Reverb.cpp',
diff --git a/third_party/WebKit/Source/WebCore/WebCore.xcodeproj/project.pbxproj b/third_party/WebKit/Source/WebCore/WebCore.xcodeproj/project.pbxproj
@@ -6371,6 +6371,8 @@
 		FDA3E95A134A49EF008D4B5A /* OfflineAudioCompletionEvent.h in Headers */ = {isa = PBXBuildFile; fileRef = FDA3E956134A49EF008D4B5A /* OfflineAudioCompletionEvent.h */; };
 		FDA3E95B134A49EF008D4B5A /* OfflineAudioDestinationNode.cpp in Sources */ = {isa = PBXBuildFile; fileRef = FDA3E957134A49EF008D4B5A /* OfflineAudioDestinationNode.cpp */; };
 		FDA3E95C134A49EF008D4B5A /* OfflineAudioDestinationNode.h in Headers */ = {isa = PBXBuildFile; fileRef = FDA3E958134A49EF008D4B5A /* OfflineAudioDestinationNode.h */; };
+		FDB1700514A2BAB200A2B5D9 /* MultiChannelResampler.cpp in Sources */ = {isa = PBXBuildFile; fileRef = FDB1700314A2BAB200A2B5D9 /* MultiChannelResampler.cpp */; };
+		FDB1700614A2BAB200A2B5D9 /* MultiChannelResampler.h in Headers */ = {isa = PBXBuildFile; fileRef = FDB1700414A2BAB200A2B5D9 /* MultiChannelResampler.h */; };
 		FDC54F041399B0DA008D9117 /* BiquadFilterNode.cpp in Sources */ = {isa = PBXBuildFile; fileRef = FDC54F011399B0DA008D9117 /* BiquadFilterNode.cpp */; };
 		FDC54F051399B0DA008D9117 /* BiquadFilterNode.h in Headers */ = {isa = PBXBuildFile; fileRef = FDC54F021399B0DA008D9117 /* BiquadFilterNode.h */; };
 		FDEAAAF312B02EE400DCF33B /* JSAudioBufferSourceNodeCustom.cpp in Sources */ = {isa = PBXBuildFile; fileRef = FDEAAAEF12B02EE400DCF33B /* JSAudioBufferSourceNodeCustom.cpp */; };
@@ -13843,6 +13845,8 @@
 		FDA3E957134A49EF008D4B5A /* OfflineAudioDestinationNode.cpp */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = OfflineAudioDestinationNode.cpp; sourceTree = "<group>"; };
 		FDA3E958134A49EF008D4B5A /* OfflineAudioDestinationNode.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = OfflineAudioDestinationNode.h; sourceTree = "<group>"; };
 		FDA3E95D134A49FF008D4B5A /* OfflineAudioCompletionEvent.idl */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = text; path = OfflineAudioCompletionEvent.idl; sourceTree = "<group>"; };
+		FDB1700314A2BAB200A2B5D9 /* MultiChannelResampler.cpp */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = MultiChannelResampler.cpp; sourceTree = "<group>"; };
+		FDB1700414A2BAB200A2B5D9 /* MultiChannelResampler.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = MultiChannelResampler.h; sourceTree = "<group>"; };
 		FDC54F011399B0DA008D9117 /* BiquadFilterNode.cpp */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = BiquadFilterNode.cpp; sourceTree = "<group>"; };
 		FDC54F021399B0DA008D9117 /* BiquadFilterNode.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = BiquadFilterNode.h; sourceTree = "<group>"; };
 		FDC54F031399B0DA008D9117 /* BiquadFilterNode.idl */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = text; path = BiquadFilterNode.idl; sourceTree = "<group>"; };
@@ -21757,6 +21761,8 @@
 				FD31606912B026F700C1A359 /* HRTFKernel.h */,
 				FD31606A12B026F700C1A359 /* HRTFPanner.cpp */,
 				FD31606B12B026F700C1A359 /* HRTFPanner.h */,
+				FDB1700314A2BAB200A2B5D9 /* MultiChannelResampler.cpp */,
+				FDB1700414A2BAB200A2B5D9 /* MultiChannelResampler.h */,
 				FD31606C12B026F700C1A359 /* Panner.cpp */,
 				FD31606D12B026F700C1A359 /* Panner.h */,
 				FD31606E12B026F700C1A359 /* Reverb.cpp */,
@@ -24954,6 +24960,7 @@
 				1AA7160B149BC4DB0016EC19 /* TileCache.h in Headers */,
 				C37CDEBD149EF2030042090D /* ColorChooserClient.h in Headers */,
 				1AC6926A14A1253200BD85F1 /* ScrollableAreaClient.h in Headers */,
+				FDB1700614A2BAB200A2B5D9 /* MultiChannelResampler.h in Headers */,
 			);
 			runOnlyForDeploymentPostprocessing = 0;
 		};
@@ -27858,6 +27865,7 @@
 				A80A9423149F225E00989291 /* JSDOMWindowWebAudioCustom.cpp in Sources */,
 				A80A9425149F227100989291 /* JSDOMWindowWebSocketCustom.cpp in Sources */,
 				52F52E1114A0134F00ACC397 /* NSScrollerImpDetails.mm in Sources */,
+				FDB1700514A2BAB200A2B5D9 /* MultiChannelResampler.cpp in Sources */,
 			);
 			runOnlyForDeploymentPostprocessing = 0;
 		};
diff --git a/third_party/WebKit/Source/WebCore/platform/audio/MultiChannelResampler.cpp b/third_party/WebKit/Source/WebCore/platform/audio/MultiChannelResampler.cpp
@@ -0,0 +1,123 @@
+/*
+ * Copyright (C) 2011 Google Inc. All rights reserved.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * 1.  Redistributions of source code must retain the above copyright
+ *     notice, this list of conditions and the following disclaimer.
+ * 2.  Redistributions in binary form must reproduce the above copyright
+ *     notice, this list of conditions and the following disclaimer in the
+ *     documentation and/or other materials provided with the distribution.
+ * 3.  Neither the name of Apple Computer, Inc. ("Apple") nor the names of
+ *     its contributors may be used to endorse or promote products derived
+ *     from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY APPLE AND ITS CONTRIBUTORS "AS IS" AND ANY
+ * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
+ * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
+ * DISCLAIMED. IN NO EVENT SHALL APPLE OR ITS CONTRIBUTORS BE LIABLE FOR ANY
+ * DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
+ * (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
+ * LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND
+ * ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF
+ * THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#include "config.h"
+
+#if ENABLE(WEB_AUDIO)
+
+#include "MultiChannelResampler.h"
+
+#include "AudioBus.h"
+
+namespace WebCore {
+
+namespace {
+
+// ChannelProvider provides a single channel of audio data (one channel at a time) for each channel
+// of data provided to us in a multi-channel provider.
+
+class ChannelProvider : public AudioSourceProvider {
+public:
+    ChannelProvider(AudioSourceProvider* multiChannelProvider, unsigned numberOfChannels)
+        : m_multiChannelProvider(multiChannelProvider)
+        , m_numberOfChannels(numberOfChannels)
+        , m_currentChannel(0)
+        , m_framesToProcess(0)
+    {
+    }
+
+    // provideInput() will be called once for each channel, starting with the first channel.
+    // Each time it's called, it will provide the next channel of data.
+    virtual void provideInput(AudioBus* bus, size_t framesToProcess)
+    {
+        bool isBusGood = bus && bus->numberOfChannels() == 1;
+        ASSERT(isBusGood);
+        if (!isBusGood)
+            return;
+
+        // Get the data from the multi-channel provider when the first channel asks for it.
+        // For subsequent channels, we can just dish out the channel data from that (stored in m_multiChannelBus).
+        if (!m_currentChannel) {
+            m_framesToProcess = framesToProcess;
+            m_multiChannelBus = adoptPtr(new AudioBus(m_numberOfChannels, framesToProcess));
+            m_multiChannelProvider->provideInput(m_multiChannelBus.get(), framesToProcess);
+        }
+
+        // All channels must ask for the same amount. This should always be the case, but let's just make sure.
+        bool isGood = m_multiChannelBus.get() && framesToProcess == m_framesToProcess;
+        ASSERT(isGood);
+        if (!isGood)
+            return;
+
+        // Copy the channel data from what we received from m_multiChannelProvider.
+        ASSERT(m_currentChannel <= m_numberOfChannels);
+        if (m_currentChannel < m_numberOfChannels) {
+            memcpy(bus->channel(0)->data(), m_multiChannelBus->channel(m_currentChannel)->data(), sizeof(float) * framesToProcess);
+            ++m_currentChannel;
+        }
+    }
+
+private:
+    AudioSourceProvider* m_multiChannelProvider;
+    OwnPtr<AudioBus> m_multiChannelBus;
+    unsigned m_numberOfChannels;
+    unsigned m_currentChannel;
+    size_t m_framesToProcess; // Used to verify that all channels ask for the same amount.
+};
+
+} // namespace
+
+MultiChannelResampler::MultiChannelResampler(double scaleFactor, unsigned numberOfChannels)
+    : m_numberOfChannels(numberOfChannels)
+{
+    // Create each channel's resampler.
+    for (unsigned channelIndex = 0; channelIndex < numberOfChannels; ++channelIndex)
+        m_kernels.append(adoptPtr(new SincResampler(scaleFactor)));
+}
+
+void MultiChannelResampler::process(AudioSourceProvider* provider, AudioBus* destination, size_t framesToProcess)
+{
+    // The provider can provide us with multi-channel audio data. But each of our single-channel resamplers (kernels)
+    // below requires a provider which provides a single unique channel of data.
+    // channelProvider wraps the original multi-channel provider and dishes out one channel at a time.
+    ChannelProvider channelProvider(provider, m_numberOfChannels);
+
+    for (unsigned channelIndex = 0; channelIndex < m_numberOfChannels; ++channelIndex) {
+        // Depending on the sample-rate scale factor, and the internal buffering used in a SincResampler
+        // kernel, this call to process() will only sometimes call provideInput() on the channelProvider.
+        // However, if it calls provideInput() for the first channel, then it will call it for the remaining
+        // channels, since they all buffer in the same way and are processing the same number of frames.
+        m_kernels[channelIndex]->process(&channelProvider,
+                                         destination->channel(channelIndex)->data(),
+                                         framesToProcess);
+    }
+}
+
+} // namespace WebCore
+
+#endif // ENABLE(WEB_AUDIO)
diff --git a/third_party/WebKit/Source/WebCore/platform/audio/MultiChannelResampler.h b/third_party/WebKit/Source/WebCore/platform/audio/MultiChannelResampler.h
@@ -0,0 +1,59 @@
+/*
+ * Copyright (C) 2011 Google Inc. All rights reserved.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * 1.  Redistributions of source code must retain the above copyright
+ *     notice, this list of conditions and the following disclaimer.
+ * 2.  Redistributions in binary form must reproduce the above copyright
+ *     notice, this list of conditions and the following disclaimer in the
+ *     documentation and/or other materials provided with the distribution.
+ * 3.  Neither the name of Apple Computer, Inc. ("Apple") nor the names of
+ *     its contributors may be used to endorse or promote products derived
+ *     from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY APPLE AND ITS CONTRIBUTORS "AS IS" AND ANY
+ * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
+ * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
+ * DISCLAIMED. IN NO EVENT SHALL APPLE OR ITS CONTRIBUTORS BE LIABLE FOR ANY
+ * DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
+ * (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
+ * LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND
+ * ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF
+ * THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#ifndef MultiChannelResampler_h
+#define MultiChannelResampler_h
+
+#include "SincResampler.h"
+#include <wtf/OwnPtr.h>
+
+namespace WebCore {
+
+class AudioBus;
+    
+class MultiChannelResampler {
+public:   
+    MultiChannelResampler(double scaleFactor, unsigned numberOfChannels);
+    
+    // Process given AudioSourceProvider for streaming applications.
+    void process(AudioSourceProvider*, AudioBus* destination, size_t framesToProcess);
+
+private:
+    // FIXME: the mac port can have a more highly optimized implementation based on CoreAudio
+    // instead of SincResampler. For now the default implementation will be used on all ports.
+    // https://bugs.webkit.org/show_bug.cgi?id=75118
+    
+    // Each channel will be resampled using a high-quality SincResampler.
+    Vector<OwnPtr<SincResampler> > m_kernels;
+    
+    unsigned m_numberOfChannels;
+};
+
+} // namespace WebCore
+
+#endif // MultiChannelResampler_h
diff --git a/third_party/WebKit/Source/WebCore/webaudio/MediaElementAudioSourceNode.cpp b/third_party/WebKit/Source/WebCore/webaudio/MediaElementAudioSourceNode.cpp
@@ -30,22 +30,30 @@
 
 #include "AudioContext.h"
 #include "AudioNodeOutput.h"
+#include "Locker.h"
+#include "Logging.h"
 #include "MediaPlayer.h"
 
+// These are somewhat arbitrary limits, but we need to do some kind of sanity-checking.
+const unsigned minSampleRate = 8000;
+const unsigned maxSampleRate = 192000;
+
 namespace WebCore {
 
 PassRefPtr<MediaElementAudioSourceNode> MediaElementAudioSourceNode::create(AudioContext* context, HTMLMediaElement* mediaElement)
 {
-    return adoptRef(new MediaElementAudioSourceNode(context, mediaElement));      
+    return adoptRef(new MediaElementAudioSourceNode(context, mediaElement));
 }
 
 MediaElementAudioSourceNode::MediaElementAudioSourceNode(AudioContext* context, HTMLMediaElement* mediaElement)
     : AudioSourceNode(context, context->sampleRate())
     , m_mediaElement(mediaElement)
+    , m_sourceNumberOfChannels(0)
+    , m_sourceSampleRate(0)
 {
     // Default to stereo. This could change depending on what the media element .src is set to.
     addOutput(adoptPtr(new AudioNodeOutput(this, 2)));
-    
+
     setNodeType(NodeTypeMediaElementAudioSource);
 
     initialize();
@@ -57,32 +65,75 @@ MediaElementAudioSourceNode::~MediaElementAudioSourceNode()
     uninitialize();
 }
 
-void MediaElementAudioSourceNode::setFormat(size_t, float)
+void MediaElementAudioSourceNode::setFormat(size_t numberOfChannels, float sourceSampleRate)
 {
-    // FIXME: setup a sample-rate converter if necessary to convert to the AudioContext sample-rate.
-    ASSERT_NOT_REACHED();
+    if (numberOfChannels != m_sourceNumberOfChannels || sourceSampleRate != m_sourceSampleRate) {
+        // FIXME: implement multi-channel greater than stereo.
+        // https://bugs.webkit.org/show_bug.cgi?id=75119
+        if (!numberOfChannels || numberOfChannels > 2 || 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;
+            m_sourceSampleRate = 0;
+            return;
+        }
+
+        m_sourceNumberOfChannels = numberOfChannels;
+        m_sourceSampleRate = sourceSampleRate;
+
+        // Synchronize with process().
+        Locker<MediaElementAudioSourceNode> locker(*this);
+
+        if (sourceSampleRate != sampleRate()) {
+            double scaleFactor = sourceSampleRate / sampleRate();
+            m_multiChannelResampler = adoptPtr(new MultiChannelResampler(scaleFactor, numberOfChannels));
+        } else {
+            // Bypass resampling.
+            m_multiChannelResampler.clear();
+        }
+
+        {
+            // The context must be locked when changing the number of output channels.
+            AudioContext::AutoLocker contextLocker(context());
+
+            // Do any necesssary re-configuration to the output's number of channels.
+            output(0)->setNumberOfChannels(numberOfChannels);
+        }
+    }
 }
 
 void MediaElementAudioSourceNode::process(size_t numberOfFrames)
 {
     AudioBus* outputBus = output(0)->bus();
 
-    if (!mediaElement()) {
+    if (!mediaElement() || !m_sourceNumberOfChannels || !m_sourceSampleRate) {
         outputBus->zero();
         return;
     }
-    
+
     // Use a tryLock() to avoid contention in the real-time audio thread.
     // If we fail to acquire the lock then the HTMLMediaElement must be in the middle of
     // reconfiguring its playback engine, so we output silence in this case.
     if (m_processLock.tryLock()) {
-        if (AudioSourceProvider* provider = mediaElement()->audioSourceProvider())
-            provider->provideInput(outputBus, numberOfFrames);
-        else
+        if (AudioSourceProvider* provider = mediaElement()->audioSourceProvider()) {
+            if (m_multiChannelResampler.get()) {
+                ASSERT(m_sourceSampleRate != sampleRate());
+                m_multiChannelResampler->process(provider, outputBus, numberOfFrames);
+            } else {
+                // Bypass the resampler completely if the source is at the context's sample-rate.
+                ASSERT(m_sourceSampleRate == sampleRate());
+                provider->provideInput(outputBus, numberOfFrames);
+            }
+        } else {
+            // Either this port doesn't yet support HTMLMediaElement audio stream access,
+            // or the stream is not yet available.
             outputBus->zero();
+        }
         m_processLock.unlock();
-    } else
+    } else {
+        // We failed to acquire the lock.
         outputBus->zero();
+    }
 }
 
 void MediaElementAudioSourceNode::reset()
diff --git a/third_party/WebKit/Source/WebCore/webaudio/MediaElementAudioSourceNode.h b/third_party/WebKit/Source/WebCore/webaudio/MediaElementAudioSourceNode.h
