Simplify pacer interface.
New interface uses two bitrates (max/min). The pace multiplier is also
removed from the interface and instead utilized outside. Min bitrate
will be filled with padding if there's not enough media to transmit.
Also fixes a bug in minimum transmission bitrate that made it ignore
REMBs. A regression test has been added to catch it.
BUG=3014
R=mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/10059004
git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5723 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/video/video_send_stream.cc b/video/video_send_stream.cc
index a480acd..a7eebef 100644
--- a/video/video_send_stream.cc
+++ b/video/video_send_stream.cc
@@ -50,9 +50,9 @@
config_.pacing = true;
rtp_rtcp_->SetTransmissionSmoothingStatus(channel_, config_.pacing);
- assert(config_.rtp.min_transmit_bitrate_kbps >= 0);
+ assert(config_.rtp.min_transmit_bitrate_bps >= 0);
rtp_rtcp_->SetMinTransmitBitrate(channel_,
- config_.rtp.min_transmit_bitrate_kbps);
+ config_.rtp.min_transmit_bitrate_bps / 1000);
for (size_t i = 0; i < config_.rtp.extensions.size(); ++i) {
const std::string& extension = config_.rtp.extensions[i].name;