Propagate base minimum delay from video jitter buffer to webrtc/api.
On api level two methods were added to api/media_stream_interface.cc on VideoSourceInterface,
GetLatency and SetLatency. Latency is measured in seconds, delay in milliseconds but both describes
the same concept.
Bug: webrtc:10287
Change-Id: Ib8dc62a4d73f63fab7e10b82c716096ee6199957
Reviewed-on: https://webrtc-review.googlesource.com/c/123482
Commit-Queue: Ruslan Burakov <kuddai@google.com>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26877}diff --git a/call/video_receive_stream.h b/call/video_receive_stream.h
index ecff63d..0bcb6d3 100644
--- a/call/video_receive_stream.h
+++ b/call/video_receive_stream.h
@@ -252,6 +252,15 @@
virtual std::vector<RtpSource> GetSources() const = 0;
+ // Sets a base minimum for the playout delay. Base minimum delay sets lower
+ // bound on minimum delay value determining lower bound on playout delay.
+ //
+ // Returns true if value was successfully set, false overwise.
+ virtual bool SetBaseMinimumPlayoutDelayMs(int delay_ms) = 0;
+
+ // Returns current value of base minimum delay in milliseconds.
+ virtual int GetBaseMinimumPlayoutDelayMs() const = 0;
+
protected:
virtual ~VideoReceiveStream() {}
};