From c8277db7c825fff1bdebfec43e15af49370ab14c Mon Sep 17 00:00:00 2001
From: "stefan@webrtc.org"
 <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d>
Date: Thu, 12 Jan 2012 15:38:50 +0000
Subject: [PATCH] Fix selective retransmissions after corrupt merge in r1373.

BUG=228
TEST=

Review URL: http://webrtc-codereview.appspot.com/345006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1414 4adac7df-926f-26a2-2b94-8c16560cd09d
---
 src/modules/rtp_rtcp/source/rtp_sender.cc | 9 +++------
 1 file changed, 3 insertions(+), 6 deletions(-)

diff --git a/src/modules/rtp_rtcp/source/rtp_sender.cc b/src/modules/rtp_rtcp/source/rtp_sender.cc
index 6a804a7b61..89470d8b0b 100644
--- a/src/modules/rtp_rtcp/source/rtp_sender.cc
+++ b/src/modules/rtp_rtcp/source/rtp_sender.cc
@@ -968,12 +968,9 @@ WebRtc_Word32 RTPSender::ReSendPacket(WebRtc_UWord16 packetID,
       return -1;
     }
     if (length == 0) {
-      WEBRTC_TRACE(kTraceWarning,
-                   kTraceRtpRtcp,
-                   _id,
-                   "Resend packet length == 0 for seqNum %u",
-                   seqNum);
-      return -1;
+      // This is a valid case since packets which we decide not to retransmit
+      // are stored but with length zero.
+      return 0;
     }
     if (_RTX) {
       CriticalSectionScoped cs(_sendCritsect);