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