Adding a receive side API for buffering mode.
At the same time, renaming the send side API.
Review URL: https://webrtc-codereview.appspot.com/1104004
git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@3525 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/video_engine/stream_synchronization.h b/video_engine/stream_synchronization.h
index 25a370c..9b7780c 100644
--- a/video_engine/stream_synchronization.h
+++ b/video_engine/stream_synchronization.h
@@ -43,11 +43,15 @@
static bool ComputeRelativeDelay(const Measurements& audio_measurement,
const Measurements& video_measurement,
int* relative_delay_ms);
+ // Set target buffering delay - All audio and video will be delayed by at
+ // least target_delay_ms.
+ void SetTargetBufferingDelay(int target_delay_ms);
private:
ViESyncDelay* channel_delay_;
int audio_channel_id_;
int video_channel_id_;
+ int base_target_delay_ms_;
};
} // namespace webrtc