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Adding a receive side API for buffering mode.

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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@3525 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
mikhal@webrtc.org
2013-02-15 23:22:18 +00:00
parent 47fe5736c1
commit ef9f76a59d
28 changed files with 447 additions and 202 deletions
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4
webrtc/video_engine/stream_synchronization.h
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@ -43,11 +43,15 @@ class StreamSynchronization {
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