From a942692725940195ba4fd67a039123dcebaca0cb Mon Sep 17 00:00:00 2001 From: "turaj@webrtc.org" Date: Tue, 23 Apr 2013 16:08:29 +0000 Subject: [PATCH] Buf fix for r3883. Review URL: https://webrtc-codereview.appspot.com/1319012 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3889 4adac7df-926f-26a2-2b94-8c16560cd09d --- .../audio_coding/main/source/audio_coding_module_impl.cc | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/webrtc/modules/audio_coding/main/source/audio_coding_module_impl.cc b/webrtc/modules/audio_coding/main/source/audio_coding_module_impl.cc index 6085181a77..a0cde49116 100644 --- a/webrtc/modules/audio_coding/main/source/audio_coding_module_impl.cc +++ b/webrtc/modules/audio_coding/main/source/audio_coding_module_impl.cc @@ -3004,7 +3004,8 @@ int AudioCodingModuleImpl::PushSyncPacketSafe() { rtp_info.header.markerBit = false; rtp_info.header.sequenceNumber = last_sequence_number_; rtp_info.header.timestamp = last_incoming_send_timestamp_; - rtp_info.type.Audio.channel = stereo_receive_ ? 2 : 1; + rtp_info.type.Audio.channel = stereo_receive_[current_receive_codec_idx_] ? + 2 : 1; last_packet_was_sync_ = true; int payload_len_bytes = neteq_.RecIn(rtp_info, last_receive_timestamp_);