From fd3a0efd1518a1f135ebc6cb2cb393f58b64f848 Mon Sep 17 00:00:00 2001 From: "mflodman@webrtc.org" <mflodman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> Date: Fri, 11 Nov 2011 10:55:26 +0000 Subject: [PATCH] RTP bw estimate fix. Review URL: http://webrtc-codereview.appspot.com/279004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@932 4adac7df-926f-26a2-2b94-8c16560cd09d --- src/modules/rtp_rtcp/source/rtp_rtcp_impl.cc | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/src/modules/rtp_rtcp/source/rtp_rtcp_impl.cc b/src/modules/rtp_rtcp/source/rtp_rtcp_impl.cc index f420ceb4f..ffc31b322 100644 --- a/src/modules/rtp_rtcp/source/rtp_rtcp_impl.cc +++ b/src/modules/rtp_rtcp/source/rtp_rtcp_impl.cc @@ -2827,6 +2827,11 @@ void ModuleRtpRtcpImpl::ProcessDefaultModuleBandwidth() { it++; } } // end critsect + + if (count == 0) { + // No sending modules and no bitrate estimate. + return; + } _bandwidthManagement.SetSendBitrate(minBitrateBps, 0, 0); // Update default module bitrate. Don't care about min max. // Check if we should trigger OnNetworkChanged via video callback