Commit Graph

7 Commits

Author SHA1 Message Date
fe1ef935e7 Implement I420FrameCallbacks in Call.
BUG=2425
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2393004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5005 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-21 10:34:43 +00:00
e05362916c Make sure the first frame isn't dropped.
If frames were delivered within the same millisecond as VideoCaptureImpl
was created, or the timestamp weren't granular enough then the first
frame would be mistakenly dropped because of having the same timestamp
as a previous one, even though there was no previous one.

BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2599004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5004 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-21 09:02:30 +00:00
c11148b352 Compound/reduced-size RTCP in VideoReceiveStream.
BUG=2424
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2413004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4987 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-17 14:14:42 +00:00
de74b64184 Implement TraceCallbacks in Call.
Uses a global TraceDispatcher in Call. Lazy initialization of it misses
an atomic compare and exchange to be correct. This is expected to work
fine so long as no Calls are created concurrently.

BUG=2421
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2321005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4900 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-02 13:36:09 +00:00
b74b96f487 Test multiple send/receive streams in Call.
Removes renderer in VideoReceiveStream as it wasn't properly
deregistered before. Makes sure that send/receive streams are properly
wired so that receive streams receive the expected stream.

BUG=2423
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2326004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4891 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-01 11:33:24 +00:00
2e246b4e78 Remove test parameters from CallTest.
Since the test parameters weren't used, it made no sense to have a
parameterized test.

BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2316004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4862 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-27 10:54:10 +00:00
6917e19ad4 Rename EngineTest to CallTest.
There's no real notion of VideoEngine left in these classes. They're
end-to-end tests built on Call, so CallTest makes more sense.

This also contains a modification to RtpRtcpObserver moving the
responsibility of creating the event that signals when the observation
is complete to RtpRtcpObserver. New tests are about to be introduced and
this will reduce code duplication.

BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2258005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4793 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-19 14:22:12 +00:00