6 Commits

Author SHA1 Message Date
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
efae5d5901 Extracted rtp receiver payload management to its own class, made video receiver depend on that instead.
Eliminated need for video receiver to talk to its parent. Also we will now determine if the packet is the first one already in the rtp general receiver. The possible downside would be that recovered video packets no longer can be flagged as the first packet, but I don't think that can happen. Even if it can happen, maybe the bit was set anyway at an earlier stage. The tests run fine.

BUG=
TEST=rtp_rtcp_unittests, vie_auto_test, voe_auto_test

Review URL: https://webrtc-codereview.appspot.com/1022011

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3382 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-17 16:10:45 +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
07bf43c673 Replaced the _audio parameter with a strategy.
The purpose is to make _rtpReceiver mostly agnostic to if it processes audio or video, and make its delegates responsible for that. This patch makes the actual interfaces and interactions between the classes a lot clearer which will probably help straighten out the rather convoluted business logic in here. There are a number of rough edges I hope to address in coming patches.

In particular, I think there are a lot of audio-specific hacks, especially when it comes to telephone event handling. I think we will see a lot of benefit once that stuff moves out of rtp_receiver altogether. The new strategy I introduced doesn't quite pull its own weight yet, but I think I will be able to remove a lot of that interface later once the responsibilities of the classes becomes move cohesive (e.g. that audio specific stuff actually lives in the audio class, and so on). Also I think it should be possible to extract payload type management to a helper class later on.

BUG=
TEST=vie/voe_auto_test, trybots

Review URL: https://webrtc-codereview.appspot.com/1001006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3306 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-12-18 15:40:53 +00:00