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Have changes to REMB trigger RTCP to be sent immediately.

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R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10339004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5763 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
stefan@webrtc.org
2014-03-24 17:13:00 +00:00
parent 1e6cb2c5d2
commit 9d4762e8b6
2 changed files with 10 additions and 12 deletions
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10
webrtc/modules/rtp_rtcp/source/rtcp_sender.cc
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@ -348,6 +348,9 @@ RTCPSender::SetREMBData(const uint32_t bitrate,
_rembSSRC[i] = SSRC[i];
}
_sendREMB = true;
// Send a REMB immediately if we have a new REMB. The frequency of REMBs is
// throttled by the caller.
_nextTimeToSendRTCP = _clock->TimeInMilliseconds();
return 0;
}
@ -483,14 +486,15 @@ RTCPSender::TimeToSendRTCPReport(const bool sendKeyframeBeforeRTP) const
For audio we use a fix 5 sec interval
For video we use 1 sec interval fo a BW smaller than 360 kbit/s,
technicaly we break the max 5% RTCP BW for video below 10 kbit/s but that should be extreamly rare
technicaly we break the max 5% RTCP BW for video below 10 kbit/s but
that should be extremely rare
From RFC 3550
MAX RTCP BW is 5% if the session BW
A send report is approximately 65 bytes inc CNAME
A report report is approximately 28 bytes
A receiver report is approximately 28 bytes
The RECOMMENDED value for the reduced minimum in seconds is 360
divided by the session bandwidth in kilobits/second. This minimum
@ -552,7 +556,7 @@ From RFC 3550
now += RTCP_SEND_BEFORE_KEY_FRAME_MS;
}
if(now > _nextTimeToSendRTCP)
if(now >= _nextTimeToSendRTCP)
{
return true;