From a32d18fab5a8255ff6624dc33e9d2d03f815f3e4 Mon Sep 17 00:00:00 2001 From: "stefan@webrtc.org" Date: Fri, 5 Jul 2013 14:30:48 +0000 Subject: Breaking out receive-stats, rtp-payload-registry and rtp-receiver from the rtp_rtcp implementation. This refactoring significantly reduces the receive-side RTP parser and receiver complexity, and makes it possible to implement RTX correctly by having two instances of receive-statistics. With this change the dead-or-alive and packet timeout APIs are removed. TEST=trybots, vie_auto_test, voe_auto_test BUG=1811 R=mflodman@webrtc.org, pbos@webrtc.org, xians@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1745004 git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@4301 4adac7df-926f-26a2-2b94-8c16560cd09d --- video_engine/vie_sync_module.h | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) (limited to 'video_engine/vie_sync_module.h') diff --git a/video_engine/vie_sync_module.h b/video_engine/vie_sync_module.h index 51246c1c..cc0d92bd 100644 --- a/video_engine/vie_sync_module.h +++ b/video_engine/vie_sync_module.h @@ -36,7 +36,8 @@ class ViESyncModule : public Module { int ConfigureSync(int voe_channel_id, VoEVideoSync* voe_sync_interface, - RtpRtcp* video_rtcp_module); + RtpRtcp* video_rtcp_module, + RtpReceiver* video_receiver); int VoiceChannel(); @@ -51,6 +52,7 @@ class ViESyncModule : public Module { scoped_ptr data_cs_; VideoCodingModule* vcm_; ViEChannel* vie_channel_; + RtpReceiver* video_receiver_; RtpRtcp* video_rtp_rtcp_; int voe_channel_id_; VoEVideoSync* voe_sync_interface_; -- cgit v1.2.3