webkitgtk-2.16.5HEADwebkitgtk-2.16.5master

1 files changed, 56 insertions, 39 deletions

diff --git a/Source/WebCore/Modules/webaudio/MediaStreamAudioSourceNode.cpp b/Source/WebCore/Modules/webaudio/MediaStreamAudioSourceNode.cpp
index 867212e51..f640a6826 100644
--- a/Source/WebCore/Modules/webaudio/MediaStreamAudioSourceNode.cpp
+++ b/Source/WebCore/Modules/webaudio/MediaStreamAudioSourceNode.cpp
@@ -35,18 +35,21 @@
 
 namespace WebCore {
 
-PassRefPtr<MediaStreamAudioSourceNode> MediaStreamAudioSourceNode::create(AudioContext* context, MediaStream* mediaStream, MediaStreamTrack* audioTrack, AudioSourceProvider* audioSourceProvider)
+Ref<MediaStreamAudioSourceNode> MediaStreamAudioSourceNode::create(AudioContext& context, MediaStream& mediaStream, MediaStreamTrack& audioTrack)
 {
-    return adoptRef(new MediaStreamAudioSourceNode(context, mediaStream, audioTrack, audioSourceProvider));
+    return adoptRef(*new MediaStreamAudioSourceNode(context, mediaStream, audioTrack));
 }
 
-MediaStreamAudioSourceNode::MediaStreamAudioSourceNode(AudioContext* context, MediaStream* mediaStream, MediaStreamTrack* audioTrack, AudioSourceProvider* audioSourceProvider)
-    : AudioNode(context, context->sampleRate())
+MediaStreamAudioSourceNode::MediaStreamAudioSourceNode(AudioContext& context, MediaStream& mediaStream, MediaStreamTrack& audioTrack)
+    : AudioNode(context, context.sampleRate())
     , m_mediaStream(mediaStream)
     , m_audioTrack(audioTrack)
-    , m_audioSourceProvider(audioSourceProvider)
-    , m_sourceNumberOfChannels(0)
 {
+    AudioSourceProvider* audioSourceProvider = m_audioTrack->audioSourceProvider();
+    ASSERT(audioSourceProvider);
+
+    audioSourceProvider->setClient(this);
+    
     // Default to stereo. This could change depending on the format of the MediaStream's audio track.
     addOutput(std::make_unique<AudioNodeOutput>(this, 2));
 
@@ -57,45 +60,56 @@ MediaStreamAudioSourceNode::MediaStreamAudioSourceNode(AudioContext* context, Me
 
 MediaStreamAudioSourceNode::~MediaStreamAudioSourceNode()
 {
+    AudioSourceProvider* audioSourceProvider = m_audioTrack->audioSourceProvider();
+    ASSERT(audioSourceProvider);
+    audioSourceProvider->setClient(nullptr);
     uninitialize();
 }
 
 void MediaStreamAudioSourceNode::setFormat(size_t numberOfChannels, float sourceSampleRate)
 {
-    if (numberOfChannels != m_sourceNumberOfChannels || sourceSampleRate != sampleRate()) {
-        // The sample-rate must be equal to the context's sample-rate.
-        if (!numberOfChannels || numberOfChannels > AudioContext::maxNumberOfChannels() || sourceSampleRate != sampleRate()) {
-            // 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;
-            return;
-        }
-
-        // Synchronize with process().
-        std::lock_guard<std::mutex> lock(m_processMutex);
-
-        m_sourceNumberOfChannels = numberOfChannels;
-
-        {
-            // 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);
-        }
+    float sampleRate = this->sampleRate();
+    if (numberOfChannels == m_sourceNumberOfChannels && sourceSampleRate == sampleRate)
+        return;
+
+    // The sample-rate must be equal to the context's sample-rate.
+    if (!numberOfChannels || numberOfChannels > AudioContext::maxNumberOfChannels() || sourceSampleRate != sampleRate) {
+        // 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;
+        return;
+    }
+
+    // Synchronize with process().
+    std::lock_guard<Lock> lock(m_processMutex);
+
+    m_sourceNumberOfChannels = numberOfChannels;
+    m_sourceSampleRate = sourceSampleRate;
+
+    if (sourceSampleRate == sampleRate)
+        m_multiChannelResampler = nullptr;
+    else {
+        double scaleFactor = sourceSampleRate / sampleRate;
+        m_multiChannelResampler = std::make_unique<MultiChannelResampler>(scaleFactor, numberOfChannels);
+    }
+
+    m_sourceNumberOfChannels = numberOfChannels;
+
+    {
+        // 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 MediaStreamAudioSourceNode::process(size_t numberOfFrames)
 {
     AudioBus* outputBus = output(0)->bus();
+    AudioSourceProvider* provider = m_audioTrack->audioSourceProvider();
 
-    if (!audioSourceProvider()) {
-        outputBus->zero();
-        return;
-    }
-
-    if (!mediaStream() || m_sourceNumberOfChannels != outputBus->numberOfChannels()) {
+    if (!mediaStream() || !m_sourceNumberOfChannels || !m_sourceSampleRate || !provider) {
         outputBus->zero();
         return;
     }
@@ -103,18 +117,21 @@ void MediaStreamAudioSourceNode::process(size_t numberOfFrames)
     // Use std::try_to_lock to avoid contention in the real-time audio thread.
     // If we fail to acquire the lock then the MediaStream must be in the middle of
     // a format change, so we output silence in this case.
-    std::unique_lock<std::mutex> lock(m_processMutex, std::try_to_lock);
+    std::unique_lock<Lock> lock(m_processMutex, std::try_to_lock);
     if (!lock.owns_lock()) {
         // We failed to acquire the lock.
         outputBus->zero();
         return;
     }
 
-    audioSourceProvider()->provideInput(outputBus, numberOfFrames);
-}
-
-void MediaStreamAudioSourceNode::reset()
-{
+    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);
+    }
 }
 
 } // namespace WebCore