wu@webrtc.org
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822fbd8b68
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Update talk to 50918584.
Together with Stefan's http://review.webrtc.org/1960004/.
R=mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2048004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4556 4adac7df-926f-26a2-2b94-8c16560cd09d
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2013-08-15 23:38:54 +00:00 |
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tnakamura@webrtc.org
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aa4d96a134
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Revert r4301
R=mikhal@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1809004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4357 4adac7df-926f-26a2-2b94-8c16560cd09d
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2013-07-16 19:25:04 +00:00 |
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stefan@webrtc.org
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66b2e5c05a
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Breaking out receive-stats, rtp-payload-registry and rtp-receiver from the
rtp_rtcp implementation.
This refactoring significantly reduces the receive-side RTP parser and receiver
complexity, and makes it possible to implement RTX correctly by having two
instances of receive-statistics.
With this change the dead-or-alive and packet timeout APIs are removed.
TEST=trybots, vie_auto_test, voe_auto_test
BUG=1811
R=mflodman@webrtc.org, pbos@webrtc.org, xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1745004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4301 4adac7df-926f-26a2-2b94-8c16560cd09d
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2013-07-05 14:30:48 +00:00 |
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mflodman@webrtc.org
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9f5ebb5251
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Adding a payload type for RTX.
BUG=736
TEST=Modified RTP unittests.
Review URL: https://webrtc-codereview.appspot.com/1278004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3843 4adac7df-926f-26a2-2b94-8c16560cd09d
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2013-04-12 14:55:46 +00:00 |
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pbos@webrtc.org
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2f44673d66
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WebRtc_Word32 => int32_t for rtp_rtcp/
BUG=314
Review URL: https://webrtc-codereview.appspot.com/1279007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3777 4adac7df-926f-26a2-2b94-8c16560cd09d
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2013-04-08 11:08:41 +00:00 |
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mikhal@webrtc.org
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bda7f305c5
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Adding RTX on source
Review URL: https://webrtc-codereview.appspot.com/1190004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3674 4adac7df-926f-26a2-2b94-8c16560cd09d
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2013-03-15 23:21:52 +00:00 |
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stefan@webrtc.org
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becf9c897c
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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
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2013-02-01 15:09:57 +00:00 |
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stefan@webrtc.org
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a678a3baee
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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
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2013-01-21 07:42:11 +00:00 |
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stefan@webrtc.org
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8d0cd07d0c
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Add test to verify that padding only frames are passing through the RTP module.
Review URL: https://webrtc-codereview.appspot.com/934023
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3224 4adac7df-926f-26a2-2b94-8c16560cd09d
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2012-12-03 14:01:46 +00:00 |
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pwestin@webrtc.org
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571a1c035b
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Enable paced sender.
Review URL: https://webrtc-codereview.appspot.com/965016
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3089 4adac7df-926f-26a2-2b94-8c16560cd09d
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2012-11-13 21:12:39 +00:00 |
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andrew@webrtc.org
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14b43beb7c
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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
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2012-10-22 18:19:23 +00:00 |
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