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
8392cd9edd
VCM/JB: Using last decoded state for waiting for key
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relanding 1323006
BUG=
Review URL: https://webrtc-codereview.appspot.com/1354004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3902 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-04-25 21:30:50 +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
df9c0e5ec9
Revert 3892 "VCM/JB: Using last decoded state for waiting for key"
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> VCM/JB: Using last decoded state for waiting for key
>
> Review URL: https://webrtc-codereview.appspot.com/1323006
Although I have no idea why, it appears this might be causing failures in ViEStandardIntegrationTest.RunsFileTestWithoutErrors. I was unable to reproduce locally. This is a trial revert to verify. If the errors continue to happen, I will restore this change.
TBR=mikhal@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1321010
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3896 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-04-24 02:13:18 +00:00
1248d4effc
VCM/JB: Using last decoded state for waiting for key
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Review URL: https://webrtc-codereview.appspot.com/1323006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3892 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-04-23 20:57:06 +00:00
d6bd7cd2b1
removing redundant calls to cleanframes
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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
9da751715f
VCM/JB:Removing hybrid and setting a decodable state.
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Review URL: https://webrtc-codereview.appspot.com/1283004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3834 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-04-11 18:49:13 +00:00
7bc465bd21
Fix issues with incorrect wrap checks when having big buffers and high bitrate.
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Introduces shared functions for timestamp and sequence number wrap checks.
BUG=1607
TESTS=trybots
Review URL: https://webrtc-codereview.appspot.com/1291005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3833 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-04-11 17:48:02 +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
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
119c67df36
Adding a max jitter filter to the JB estimate - allowing two modes, one will return the last estimate (current setting), and another will return the max value seen, and allow setting an initial value.
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This cl also includes tests and some clean up.
Review URL: https://webrtc-codereview.appspot.com/1019007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3445 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-31 17:18:02 +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
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
