commit 5565981e1722c9aeeab8bd7440eac121417b64b0
author Åsa Persson <asapersson@webrtc.org> Mon Jun 18 17:51:32 2018 +0200
committer Commit Bot <commit-bot@chromium.org> Wed Jun 20 07:26:09 2018 +0000
tree 26100e5850cce1e72f63efd0add093a0ff0ccb9b
parent f88a22cf1150c0aa0c27b9963af250ff928f485a

Add functionality to set min/max bitrate per simulcast layer through RtpEncodingParameters.

Target bitrate is set to 0.75 of the max bitrate.

Bug: webrtc:9341, webrtc:8655
Change-Id: I9a8c8bb95bb1532d45f05578832418464452340e
Reviewed-on: https://webrtc-review.googlesource.com/79821
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23676}

api/rtpsenderinterface.h

diff --git a/api/rtpsenderinterface.h b/api/rtpsenderinterface.h
index e933614..66267c7 100644
--- a/api/rtpsenderinterface.h
+++ b/api/rtpsenderinterface.h
@@ -57,6 +57,7 @@
   virtual RtpParameters GetParameters() = 0;
   // Note that only a subset of the parameters can currently be changed. See
   // rtpparameters.h
+  // The encodings are in increasing quality order for simulcast.
   virtual RTCError SetParameters(const RtpParameters& parameters) = 0;
 
   // Returns null for a video sender.