Implementation of 'AudioContext.getOutputTimestamp' method

third_party/WebKit/Source/modules/webaudio/AudioDestinationNode.h

Index: third_party/WebKit/Source/modules/webaudio/AudioDestinationNode.h
diff --git a/third_party/WebKit/Source/modules/webaudio/AudioDestinationNode.h b/third_party/WebKit/Source/modules/webaudio/AudioDestinationNode.h
index a25e598b53617a89d01aa2b0466ae65ea470a49e..9f1125d0b791091122425ecbea73791adc0f2b07 100644
--- a/third_party/WebKit/Source/modules/webaudio/AudioDestinationNode.h
+++ b/third_party/WebKit/Source/modules/webaudio/AudioDestinationNode.h
@@ -30,11 +30,13 @@
 #include "platform/audio/AudioBus.h"
 #include "platform/audio/AudioIOCallback.h"
 #include "platform/audio/AudioSourceProvider.h"
+#include "wtf/ThreadingPrimitives.h"
 
 namespace blink {
 
 class AudioBus;
 class AbstractAudioContext;
+class AudioTimestamp;
 
 class AudioDestinationHandler : public AudioHandler, public AudioIOCallback {
 public:
@@ -46,7 +48,7 @@ public:
 
     // The audio hardware calls render() to get the next render quantum of audio into destinationBus.
     // It will optionally give us local/live audio input in sourceBus (if it's not 0).
-    void render(AudioBus* sourceBus, AudioBus* destinationBus, size_t numberOfFrames) final;
+    void render(AudioBus* sourceBus, AudioBus* destinationBus, size_t numberOfFrames, const WebAudioTimestamp& outputTimestamp) final;
 
     size_t currentSampleFrame() const { return acquireLoad(&m_currentSampleFrame); }
     double currentTime() const { return currentSampleFrame() / static_cast<double>(sampleRate()); }
@@ -56,6 +58,8 @@ public:
     virtual void startRendering() = 0;
     virtual void stopRendering() = 0;
 
+    WebAudioTimestamp outputTimestamp() const;
+
 protected:
     // LocalAudioInputProvider allows us to expose an AudioSourceProvider for local/live audio input.
     // If there is local/live audio input, we call set() with the audio input data every render quantum.
@@ -88,6 +92,10 @@ protected:
     // Counts the number of sample-frames processed by the destination.
     size_t m_currentSampleFrame;
 
+    // FIXME(Mikhail) : when allowed use std::atomic<WebAudioTimestamp> instead of mutex.

[hongchan, 2016/06/14 17:10:38]

FIXME -> TODO.

[Mikhail, 2016/06/17 09:36:57]

I've changed the approach so that it is using 'lock' and 'tryLock' from context

+    WebAudioTimestamp m_outputTimestamp;
+    mutable Mutex m_mutex;
+
     LocalAudioInputProvider m_localAudioInputProvider;
 };