| commit | a305e9612afb9feac6bdbced512beb3a2557c3e2 | [log] [tgz] |
|---|---|---|
| author | turaj@webrtc.org <turaj@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> | Thu Jun 06 19:00:09 2013 +0000 |
| committer | turaj@webrtc.org <turaj@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> | Thu Jun 06 19:00:09 2013 +0000 |
| tree | a339807ff5ba4e577c843fac974a0319fae6e930 | |
| parent | d9c46587567ab2637679b4e7463b082a342875ed [diff] [blame] |
Nack for audio. R=stefan@webrtc.org, tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1507004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4188 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/modules/audio_coding/neteq/mcu.h b/webrtc/modules/audio_coding/neteq/mcu.h index 1be81c2..931e6dc 100644 --- a/webrtc/modules/audio_coding/neteq/mcu.h +++ b/webrtc/modules/audio_coding/neteq/mcu.h
@@ -99,6 +99,9 @@ int usingStereo; #endif + /* The sequence number of the latest decoded RTP payload. */ + int decoded_packet_sequence_number; + uint32_t decoded_packet_timestamp; } MCUInst_t; /****************************************************************************