Pass the elapsed time from VoE to WebRtcAudioRenderer as the current time for...

Pass the elapsed time from VoE to WebRtcAudioRenderer as the current time for the audio/video element.

BUG=367161
and webrtc 3111

R=xians@chromium.org

Review URL: https://codereview.chromium.org/328493003

git-svn-id: svn://svn.chromium.org/chrome/trunk/src@276673 0039d316-1c4b-4281-b951-d872f2087c98

content/renderer/media/webrtc_audio_device_impl.cc

@@ -126,7 +126,8 @@ void WebRtcAudioDeviceImpl::OnSetFormat(
 
 void WebRtcAudioDeviceImpl::RenderData(media::AudioBus* audio_bus,
                                        int sample_rate,
-                                       int audio_delay_milliseconds) {
+                                       int audio_delay_milliseconds,
+                                       base::TimeDelta* current_time) {
   render_buffer_.resize(audio_bus->frames() * audio_bus->channels());
 
   {
@@ -179,7 +180,9 @@ void WebRtcAudioDeviceImpl::RenderData(media::AudioBus* audio_bus,
                                                   &ntp_time_ms);
       accumulated_audio_frames += num_audio_frames;
     }
-
+    if (elapsed_time_ms >= 0) {
+      *current_time = base::TimeDelta::FromMilliseconds(elapsed_time_ms);
+    }
     audio_data += bytes_per_10_ms;
   }
 

content/renderer/media/webrtc_audio_device_impl.h

@@ -189,7 +189,8 @@ class WebRtcAudioRendererSource {
   // Callback to get the rendered data.
   virtual void RenderData(media::AudioBus* audio_bus,
                           int sample_rate,
-                          int audio_delay_milliseconds) = 0;
+                          int audio_delay_milliseconds,
+                          base::TimeDelta* current_time) = 0;
 
   // Callback to notify the client that the renderer is going away.
   virtual void RemoveAudioRenderer(WebRtcAudioRenderer* renderer) = 0;
@@ -390,7 +391,8 @@ class CONTENT_EXPORT WebRtcAudioDeviceImpl
   // Called on the AudioOutputDevice worker thread.
   virtual void RenderData(media::AudioBus* audio_bus,
                           int sample_rate,
-                          int audio_delay_milliseconds) OVERRIDE;
+                          int audio_delay_milliseconds,
+                          base::TimeDelta* current_time) OVERRIDE;
 
   // Called on the main render thread.
   virtual void RemoveAudioRenderer(WebRtcAudioRenderer* renderer) OVERRIDE;

content/renderer/media/webrtc_audio_renderer.cc

@@ -450,7 +450,9 @@ void WebRtcAudioRenderer::SetVolume(float volume) {
 }
 
 base::TimeDelta WebRtcAudioRenderer::GetCurrentRenderTime() const {
-  return base::TimeDelta();
+  DCHECK(thread_checker_.CalledOnValidThread());
+  base::AutoLock auto_lock(lock_);
+  return current_time_;
 }
 
 bool WebRtcAudioRenderer::IsLocalRenderer() const {
@@ -495,7 +497,8 @@ void WebRtcAudioRenderer::SourceCallback(
   // We need to keep render data for the |source_| regardless of |state_|,
   // otherwise the data will be buffered up inside |source_|.
   source_->RenderData(audio_bus, sink_params_.sample_rate(),
-                      output_delay_milliseconds);
+                      output_delay_milliseconds,
+                      &current_time_);
 
   // Avoid filling up the audio bus if we are not playing; instead
   // return here and ensure that the returned value in Render() is 0.

content/renderer/media/webrtc_audio_renderer.h

@@ -194,8 +194,8 @@ class CONTENT_EXPORT WebRtcAudioRenderer
   // Audio data source from the browser process.
   WebRtcAudioRendererSource* source_;
 
-  // Protects access to |state_|, |source_| and |sink_|.
-  base::Lock lock_;
+  // Protects access to |state_|, |source_|, |sink_| and |current_time_|.
+  mutable base::Lock lock_;
 
   // Ref count for the MediaPlayers which are playing audio.
   int play_ref_count_;
@@ -214,6 +214,8 @@ class CONTENT_EXPORT WebRtcAudioRenderer
   // Delay due to the FIFO in milliseconds.
   int fifo_delay_milliseconds_;
 
+  base::TimeDelta current_time_;
+
   // Saved volume and playing state of the root renderer.
   PlayingState playing_state_;
 

content/renderer/media/webrtc_audio_renderer_unittest.cc

@@ -68,9 +68,10 @@ class MockAudioRendererSource : public WebRtcAudioRendererSource {
  public:
   MockAudioRendererSource() {}
   virtual ~MockAudioRendererSource() {}
-  MOCK_METHOD3(RenderData, void(media::AudioBus* audio_bus,
+  MOCK_METHOD4(RenderData, void(media::AudioBus* audio_bus,
                                 int sample_rate,
-                                int audio_delay_milliseconds));
+                                int audio_delay_milliseconds,
+                                base::TimeDelta* current_time));
   MOCK_METHOD1(RemoveAudioRenderer, void(WebRtcAudioRenderer* renderer));
 };