6c75c98964
Propagate capture ntp timestamp from rtp to renderer.
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Mostly the interface changes, the real implementation of ntp timestamp will come in a follow up cl.
TEST=new tests and try bots
BUG=3111
R=niklas.enbom@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11469004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5911 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-15 17:46:33 +00:00
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
f1e807c0e5
Removing FrameForStorage
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R=pwestin@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2142004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4688 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-05 22:34:41 +00:00
3c5a9242fe
Don't force cont' when enabling kWithErrors
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BUG=
R=pbos@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2047004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4669 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-03 20:45:36 +00:00
2b810bf77b
Removing non decodable count from session info: This value isn't used, and therfore can be removed. This is a step towards the refactor of the session info to use maps.
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R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2143004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4667 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-03 19:09:49 +00:00
f31a47abdc
VCM:Accounting for bounds when inserting packets. We currently receive indicators to the first and last packets of the frame, but not have any sanity to verify that all packets are indeed within the bounds (when available). This cl attempts to fix that,
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BUG=
R=marpan@google.com
Review URL: https://webrtc-codereview.appspot.com/2077004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4614 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-26 17:10:11 +00:00
dbf6a81cb5
Follow-up changes to kSelectiveErrors
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Committing cl for agalusza (cl 1992004)
TEST = trybots
R=marpan@google.com
Review URL: https://webrtc-codereview.appspot.com/2085004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4587 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-21 20:40:47 +00:00
d177c10e2d
Added logic for kSelectiveErrors to VCMJitterBuffer and corresponding unit tests.
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R=marpan@google.com , mikhal@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1943004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4503 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-08 01:12:33 +00:00
12dc1a38ca
Switch C++-style C headers with their C equivalents.
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The C++ headers define the C functions within the std:: namespace, but
we mainly don't use the std:: namespace for C functions. Therefore we
should include the C headers.
BUG=1833
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1917004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4486 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-05 16:22:53 +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
d818dcb939
Sets up framework for decoding with errors: collects frame sizes (in number of packets) in JB and passes this information to VCMSessionInfo with rtt_ms as FrameData.
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R=marpan@google.com , mikhal@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1841004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4424 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-29 21:48:11 +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
4cf1a8af69
Removes kStateFree and kStateDecoding, added a free_frames_ list which simplifies finding a free frame.
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The idea is to have all frames not in use be stored in free_frames_, and whenever a packet from a new frame arrives we can just pop a frame from free_frames_. When a frame is grabbed for decoding it will be removed from all lists, and will be added to free_frames_ when it's returned to the jitter buffer.
We should be able to remove the state enum completely later, as their state is defined by the list they are in. But I'll keep it around for now to simplify the cl.
TEST=try bots and vie_auto_test --automated
R=mikhal@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1721004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4273 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-27 15:20:14 +00:00
3417eb49f6
Consolidate GetFrame and InsertPacket and move NACK list processing to after a packet has been successfully inserted.
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TEST=trybots
BUG=1799
R=mikhal@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1509004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4080 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-21 15:25:53 +00:00
885cd13356
Start NACKing as soon as we have the first packet of a key frame.
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BUG=1605
Review URL: https://webrtc-codereview.appspot.com/1307007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3855 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-04-16 09:38:26 +00:00
7b859cc1e9
Webrtc_Word32 => int32_t in video_coding/main/
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BUG=
Review URL: https://webrtc-codereview.appspot.com/1279004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3753 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-04-02 15:54:38 +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
bf535b9b6b
Optimize NACK list creation.
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- No longer looping through all frame buffers.
- Keeping track of the current nack list index when building the list.
- Don't look for changes in the NACK list if the size has increased.
Review URL: https://webrtc-codereview.appspot.com/1076005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3420 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-28 08:48:13 +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
