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09315705b9
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Wire up statistics in video receive stream of new API
This CL includes Call tests that test both send and receive sides.
BUG=2235
R=mflodman@webrtc.org, pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8049004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5499 4adac7df-926f-26a2-2b94-8c16560cd09d
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2014-02-07 12:06:29 +00:00 |
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c279a5d72c
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Wire up RTX in VideoReceiveStream.
Also adds a test to make sure that a retransmitted frame is actually
received and decoded on the remote side. The previous NACK test checked
retransmission, but not that the receiver actually takes care of the
retransmitted packet.
BUG=2399
R=mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7469004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5422 4adac7df-926f-26a2-2b94-8c16560cd09d
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2014-01-24 09:30:53 +00:00 |
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e02d47515f
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Set up receiver RTX config using a std::map.
This change removes video_payload_type from RtxConfig as it can be
inferred from the map key or config otherwise. Wiring up this config is
part of issue 2399.
BUG=
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7349004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5402 4adac7df-926f-26a2-2b94-8c16560cd09d
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2014-01-20 14:43:55 +00:00 |
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efaeda0c76
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Add configuration and test for extended RTCP reference time reports to new video api.
R=mflodman@webrtc.org, pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6989004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5401 4adac7df-926f-26a2-2b94-8c16560cd09d
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2014-01-20 08:34:49 +00:00 |
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b429e516a9
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cpplint cleaning new API and its implementation files.
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6089005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5317 4adac7df-926f-26a2-2b94-8c16560cd09d
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2013-12-18 09:46:22 +00:00 |
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92c2793154
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Adding REMB to receive stream configuration, the send side will always
react to incoming REMB for now.
Adding a test to verify the receive side is generating RTCP REMB and
will follow up with a send side test as soon as the bitrate stats are
wired up for the new API.
TEST=See above.
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5779004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5286 4adac7df-926f-26a2-2b94-8c16560cd09d
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2013-12-13 16:36:28 +00:00 |
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b613b5ab2b
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Set local SSRC for VideoReceiveStream.
As a bonus, also removes GenerateRandomSsrc, which only worked on sender
configs. There's no point to generate random SSRCs in tests.
BUG=2691
R=mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4689004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5201 4adac7df-926f-26a2-2b94-8c16560cd09d
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2013-12-03 10:13:04 +00:00 |
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53c8573525
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Rename video streams' start/stop methods.
{Start,Stop}{Send,Receive}() -> {Start,Stop}{Sending,Receiving}().
BUG=
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3609005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5136 4adac7df-926f-26a2-2b94-8c16560cd09d
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2013-11-20 11:36:47 +00:00 |
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16e03b7bd8
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Separate Call API/build files from video_engine/.
BUG=2535
R=andrew@webrtc.org, mflodman@webrtc.org, niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2659004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5042 4adac7df-926f-26a2-2b94-8c16560cd09d
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2013-10-28 16:32:01 +00:00 |
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