34c5da6b5e
Cleaned up logging in video_coding.
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Converted all calls to WEBRTC_TRACE to LOG(). Also removed a large number of less useful logs.
BUG=3153
R=mflodman@webrtc.org , pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11169004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5887 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-11 14:08:35 +00:00
a7e360e89b
Removed lines preventing simultaneous kHardNack and decoding with errors. Also made changes recommended by gcl lint (with the exception of changing non-const references to pointers).
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Propagated orthogonal API for decoding with errors from VideoCodingModule to VCMJitterBuffer.
Modified VCMJitterBuffer to allow three error modes: kNoErrors, kSelectiveErrors, kWithErrors.
R=marpan@google.com , mikhal@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1846004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4463 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-01 03:15:08 +00:00
a4407329d4
Include files from webrtc/.. paths in video_coding/.
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BUG=1662
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1783006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4348 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-16 12:32:05 +00:00
ef14488d03
Trigger a PLI if the duration of non-decodable frames exceeds a threshold.
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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
759b041019
Relanding r3952: VCM: Updating receiver logic
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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
4ce19b1664
Revert r3952 "VCM: Updating receiver logic"
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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
d3cd565ecf
VCM: Updating receiver logic
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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
381da4be9c
VCM: Adding API for the size(duration) of the jitter buffer.
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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
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- 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.
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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.
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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.
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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.
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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.
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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.
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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/
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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
