Implement minimum transmit bitrate.
Utilizing minimum transmission bitrate prevents low remote bitrate
estimates (bitrate estimation dips) when encoding non-complex content
such as screenshare of a static image even though there's nothing wrong
with the link.
Requires pacing to be enabled for now, pending issue 3036.
BUG=3014
R=mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/9719004
git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5694 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/video/video_send_stream.cc b/video/video_send_stream.cc
index 8ec8575..5cacb56 100644
--- a/video/video_send_stream.cc
+++ b/video/video_send_stream.cc
@@ -49,6 +49,10 @@
config_.pacing = true;
rtp_rtcp_->SetTransmissionSmoothingStatus(channel_, config_.pacing);
+ assert(config_.rtp.min_transmit_bitrate_kbps >= 0);
+ rtp_rtcp_->SetMinTransmitBitrate(channel_,
+ config_.rtp.min_transmit_bitrate_kbps);
+
for (size_t i = 0; i < config_.rtp.extensions.size(); ++i) {
const std::string& extension = config_.rtp.extensions[i].name;
int id = config_.rtp.extensions[i].id;