From 52ce65128347d245820d8992575025139654eb3d Mon Sep 17 00:00:00 2001 From: "braveyao@webrtc.org" Date: Mon, 13 Aug 2012 07:30:08 +0000 Subject: [PATCH] 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 | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/src/modules/rtp_rtcp/source/rtp_sender_audio.cc b/src/modules/rtp_rtcp/source/rtp_sender_audio.cc index 71aa72af8e..0f6f69f353 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 @@ WebRtc_Word32 RTPSenderAudio::SendAudio( fragmentation->fragmentationLength[1]); } else { // silence for too long send only new data - dataBuffer[rtpHeaderLength++] = static_cast(payloadType); + dataBuffer[rtpHeaderLength++] = fragmentation->fragmentationPlType[0]; memcpy(dataBuffer+rtpHeaderLength, payloadData + fragmentation->fragmentationOffset[0], fragmentation->fragmentationLength[0]); @@ -442,6 +442,7 @@ WebRtc_Word32 RTPSenderAudio::SendAudio( } 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]);