commit 52ce65128347d245820d8992575025139654eb3d
author braveyao@webrtc.org <braveyao@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> Mon Aug 13 07:30:08 2012
committer braveyao@webrtc.org <braveyao@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> Mon Aug 13 07:30:08 2012
tree 63b7a2510f1ea82808b60651fecd6e6d26522940
parent 1d9521eeed265d34be17002d1d51017300889a4f

Fix the auido noise issue with FEC enabled 
BUG = issue 652
TEST=manual test
Review URL: https://webrtc-codereview.appspot.com/720006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2601 4adac7df-926f-26a2-2b94-8c16560cd09d

src/modules/rtp_rtcp/source/rtp_sender_audio.cc

diff --git a/src/modules/rtp_rtcp/source/rtp_sender_audio.cc b/src/modules/rtp_rtcp/source/rtp_sender_audio.cc
index 71aa72a..0f6f69f 100644
--- a/src/modules/rtp_rtcp/source/rtp_sender_audio.cc
+++ b/src/modules/rtp_rtcp/source/rtp_sender_audio.cc
@@ -431,7 +431,7 @@
             fragmentation->fragmentationLength[1]);
       } else {
         // silence for too long send only new data
-        dataBuffer[rtpHeaderLength++] = static_cast<WebRtc_UWord8>(payloadType);
+        dataBuffer[rtpHeaderLength++] = fragmentation->fragmentationPlType[0];
         memcpy(dataBuffer+rtpHeaderLength,
                payloadData + fragmentation->fragmentationOffset[0],
                fragmentation->fragmentationLength[0]);
@@ -442,6 +442,7 @@
     } else {
       if (fragmentation && fragmentation->fragmentationVectorSize > 0) {
         // use the fragment info if we have one
+        dataBuffer[rtpHeaderLength++] = fragmentation->fragmentationPlType[0];
         memcpy( dataBuffer+rtpHeaderLength,
                 payloadData + fragmentation->fragmentationOffset[0],
                 fragmentation->fragmentationLength[0]);