pbos@webrtc.org
2f44673d66
WebRtc_Word32 => int32_t for rtp_rtcp/
...
BUG=314
Review URL: https://webrtc-codereview.appspot.com/1279007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3777 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-04-08 11:08:41 +00:00
phoglund@webrtc.org
244251a9cd
Moved almost all payload-related stuff to the payload registry.
...
The big benefit is we no longer have a circular dependency between the media receiver and the payload registry. The payload registry is starting to take a bit more place on the stage, and now knows how to do different things depending on audio or video.
BUG=
TESTED=rtp_rtcp_unittests, vie_auto_test, voe_auto_test
Review URL: https://webrtc-codereview.appspot.com/1078004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3465 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-02-04 13:23:07 +00:00
phoglund@webrtc.org
5accd370e7
RTP Receiver is now only deals with a receiver strategy. Cleaned up dependencies.
...
BUG=
TESTED=vie/voe_auto_test, rtp_rtcp_unittests
Review URL: https://webrtc-codereview.appspot.com/1058004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3397 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-22 12:31:01 +00:00
phoglund@webrtc.org
a22a9bd9ca
Cleaned up the data path for payload data, made callbacks to rtp_receiver nonoptional.
...
The audio receiver is now completely independent of rtp_receiver: video will hopefully be too in the next patch.
BUG=
TEST=vie & voe_auto_test full runs
Review URL: https://webrtc-codereview.appspot.com/1014006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3372 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-14 10:01:55 +00:00
phoglund@webrtc.org
7659d914bb
Decoupled video rtp receiver from rtp receiver.
...
BUG=
Review URL: https://webrtc-codereview.appspot.com/995005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3292 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-12-14 09:57:37 +00:00
phoglund@webrtc.org
ef90c3227e
Will now correctly identify the first-ever received packet as the first packet in its frame.
...
We used to flag the _second_ packet in the first frame as the first. Subsequent frames worked as intended.
BUG=1103
TEST=vie_auto_test --automated, rtp_rtcp_unittests
Review URL: https://webrtc-codereview.appspot.com/964020
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3164 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-11-26 16:30:40 +00:00
andrew@webrtc.org
14b43beb7c
Move src/ -> webrtc/
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TBR=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/915006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@2963 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-10-22 18:19:23 +00:00
