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Buf fix for r3883.

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Review URL: https://webrtc-codereview.appspot.com/1319012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3889 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
turaj@webrtc.org 2013-04-23 16:08:29 +00:00
parent ccd4b2aec8
commit a942692725
1 changed files with 2 additions and 1 deletions
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3
webrtc/modules/audio_coding/main/source/audio_coding_module_impl.cc
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@ -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_);