pbos@webrtc.org
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96edd56170
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Sorted headers under rtp_rtcp/.
BUG=
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1781005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4325 4adac7df-926f-26a2-2b94-8c16560cd09d
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2013-07-10 15:40:42 +00:00 |
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stefan@webrtc.org
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717d147ebb
|
Support sending multiple report blocks and keeping track of statistics on several SSRCs.
BUG=1811
TEST=vie_auto_test --automated, voe_auto_test --automated, trybots
R=andresp@webrtc.org, tommi@webrtc.org, xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1768004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4322 4adac7df-926f-26a2-2b94-8c16560cd09d
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2013-07-10 13:39:27 +00:00 |
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stefan@webrtc.org
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452d853c43
|
Fix three uninitialized members in rtp_receiver_impl.cc.
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1781004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4320 4adac7df-926f-26a2-2b94-8c16560cd09d
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2013-07-10 10:54:56 +00:00 |
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pbos@webrtc.org
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08933a5dfb
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Initialize payload-type frequency in channel.cc.
Uninitialized values triggered divide-by-zero crashes in voe_auto_test.
BUG=
R=stefan@webrtc.org, xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1780004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4319 4adac7df-926f-26a2-2b94-8c16560cd09d
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2013-07-10 10:06:29 +00:00 |
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hclam@chromium.org
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1a7b9b94be
|
Cleanup WebRTC tracing
The goal of this change is to:
1. Remove unused tracing events.
2. Organize tracing events to facilitate measurement of end to end latency.
The major change in this CL is to use ASYNC_STEP such that operation
flow can be traced for the same frame.
R=marpan@webrtc.org, pwestin@webrtc.org, turaj@webrtc.org, wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1761004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4308 4adac7df-926f-26a2-2b94-8c16560cd09d
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2013-07-08 21:31:18 +00:00 |
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stefan@webrtc.org
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66b2e5c05a
|
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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