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| author | turaj@webrtc.org <turaj@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> | 2013-12-13 21:05:07 +0000 |
|---|---|---|
| committer | turaj@webrtc.org <turaj@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> | 2013-12-13 21:05:07 +0000 |
| commit | f1b92fdb7ebe70333b810ef09e808a8f8505b5cc (patch) | |
| tree | 872865ddd3cb791ecfdc11b399ec40222efdeee2 /voice_engine/channel.h | |
| parent | 79d6dafd3dac32e505bc1a2ccc83ef3ceb6c5ad3 (diff) | |
| download | webrtc-f1b92fdb7ebe70333b810ef09e808a8f8505b5cc.tar.gz | |
Fix jitter buffer delay estimate.
BUG=b/12099925
R=niklas.enbom@webrtc.org, niklase@google.com, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5739004
git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@5289 4adac7df-926f-26a2-2b94-8c16560cd09d
Diffstat (limited to 'voice_engine/channel.h')
| -rw-r--r-- | voice_engine/channel.h | 3 |
1 files changed, 3 insertions, 0 deletions
diff --git a/voice_engine/channel.h b/voice_engine/channel.h index da55e9d7..f8b04fdd 100644 --- a/voice_engine/channel.h +++ b/voice_engine/channel.h @@ -489,6 +489,9 @@ private: uint8_t* _decryptionRTCPBufferPtr; uint32_t _timeStamp; uint8_t _sendTelephoneEventPayloadType; + + // Timestamp of the audio pulled from NetEq. + uint32_t jitter_buffer_playout_timestamp_; uint32_t playout_timestamp_rtp_; uint32_t playout_timestamp_rtcp_; uint32_t playout_delay_ms_; |