Commit Graph

9 Commits

Author SHA1 Message Date
381da4be9c VCM: Adding API for the size(duration) of the jitter buffer.
Refers to the duration in time of the frames which are ready to be sent to the decoder.

Review URL: https://webrtc-codereview.appspot.com/1319004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3903 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-04-25 21:45:29 +00:00
dc3cd217b2 VCM/JB: FrameForDecoding->IncompleteFrameForDecoding
- Update complete frame for decoding
- Remove FrameForDecodingNack

This CL should only be committed after issue http://webrtc-codereview.appspot.com/1313007/

Review URL: https://webrtc-codereview.appspot.com/1316007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3901 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-04-25 20:27:04 +00:00
2baf5f5fa0 Refactor webrtc specific Event implementation to an EventFactory.
Review URL: https://webrtc-codereview.appspot.com/1187005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3664 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-03-13 08:46:25 +00:00
a64300af50 Refactor NACK list creation to build the NACK list as packets arrive.
Also fixes a timer bug related to NACKing in the RTP module which could cause packets to only be NACKed twice if there's frequent packet losses.

Note that I decided to remove any selective NACKing for now as I don't think the gain of doing it is big enough compared to the added complexity. The same reasoning for empty packets. None of them will be retransmitted by a smart sender since the sender would know that they aren't needed.

BUG=1420
TEST=video_coding_unittests, vie_auto_test, video_coding_integrationtests, trybots

Review URL: https://webrtc-codereview.appspot.com/1115006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3599 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-03-04 15:24:40 +00:00
ef9f76a59d Adding a receive side API for buffering mode.
At the same time, renaming the send side API.

Review URL: https://webrtc-codereview.appspot.com/1104004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3525 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-02-15 23:22:18 +00:00
becf9c897c Fix mismatch between different NACK list lengths and packet buffers.
This is a second version of http://review.webrtc.org/1065006/ which passes the parameters via methods instead of via constructors.

BUG=1289

Review URL: https://webrtc-codereview.appspot.com/1065007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3456 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-02-01 15:09:57 +00:00
a678a3baee Move video_coding to new Clock interface and remove fake clock implementations from RTP module tests.
TEST=video_coding_unittests, video_coding_integrationtests, rtp_rtcp_unittests, trybots

Review URL: https://webrtc-codereview.appspot.com/1044004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3393 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-21 07:42:11 +00:00
1ea4b502ef Refactor receiver.h/.cc.
TEST=video_coding_unittests, vie_auto_test --automated

Review URL: https://webrtc-codereview.appspot.com/994008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3336 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-07 08:49:41 +00:00
14b43beb7c Move src/ -> webrtc/
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