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23 Commits

Author SHA1 Message Date
stefan@webrtc.org
ef14488d03 Trigger a PLI if the duration of non-decodable frames exceeds a threshold. 
BUG=1663
R=mikhal@webrtc.org, ronghuawu@chromium.org

Review URL: https://webrtc-codereview.appspot.com/1359004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3975 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-07 19:16:33 +00:00
mikhal@webrtc.org
8f86cc8712 VCM/Receiver: Return null when can't extract frame. 
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1435004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3974 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-07 18:05:21 +00:00
mikhal@webrtc.org
759b041019 Relanding r3952: VCM: Updating receiver logic 
BUG=r1734
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1433004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3970 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-07 16:36:00 +00:00
stefan@webrtc.org
4ce19b1664 Revert r3952 "VCM: Updating receiver logic" 
TBR=phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1410005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3963 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-06 13:16:51 +00:00
mikhal@webrtc.org
d3cd565ecf VCM: Updating receiver logic 
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1363005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3952 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-03 17:54:18 +00:00
mikhal@webrtc.org
6faba6edc9 VCM: Setting buffering delay in timing 
Review URL: https://webrtc-codereview.appspot.com/1338004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3921 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-30 15:39:34 +00:00
stefan@webrtc.org
5b7120c81b Fix two issues where we might end up busy looping in decoder_render mode. 
This happens if
- Next frame is far into the future (> 200 ms).
- Next frame is ready for decode/render but incomplete.

BUG=1696
TESTS=trybots

Review URL: https://webrtc-codereview.appspot.com/1354005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3914 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-29 16:41:30 +00:00
mikhal@webrtc.org
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
mikhal@webrtc.org
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
mikhal@webrtc.org
e0e029e8cb Revert 3876 
Review URL: https://webrtc-codereview.appspot.com/1341005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3877 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-19 21:11:41 +00:00
mikhal@webrtc.org
ee184b9520 VCM/Receiver: Only update render time when decoding 
Review URL: https://webrtc-codereview.appspot.com/1336004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3876 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-19 19:15:47 +00:00
mikhal@webrtc.org
dbd6a6d653 Updating delay for first value 
BUG=

Review URL: https://webrtc-codereview.appspot.com/1327005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3865 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-17 16:23:22 +00:00
stefan@webrtc.org
bdb9b971be Change receive statistics bitrate to be provided in bps instead of kbps. 
BUG=1469

Review URL: https://webrtc-codereview.appspot.com/1326004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3854 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-16 09:02:03 +00:00
mikhal@webrtc.org
c2a3aa7926 Partial revert of r3844 
Review URL: https://webrtc-codereview.appspot.com/1320004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3845 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-12 19:53:30 +00:00
mikhal@webrtc.org
d6bd7cd2b1 removing redundant calls to cleanframes 
Review URL: https://webrtc-codereview.appspot.com/1318004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3844 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-12 17:09:51 +00:00
hclam@chromium.org
806dc3b0e6 More trace events 
The goal of this change is to unify tracing events styles
and add trace events for all RTP traffic.

BUG=1555
Review URL: https://webrtc-codereview.appspot.com/1290007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3806 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-09 19:54:10 +00:00
stefan@webrtc.org
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
stefan@webrtc.org
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
mikhal@webrtc.org
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
stefan@webrtc.org
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
stefan@webrtc.org
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
stefan@webrtc.org
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
andrew@webrtc.org
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