c279a5d72c
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
2014-01-24 09:30:53 +00:00
efaeda0c76
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
2014-01-20 08:34:49 +00:00
f777cf2547
Permitting double start/stopping of streams.
...
It doesn't make too much sense to hard enforce that the user keeps track
of which streams are started and which are not.
BUG=
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6899004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5363 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-10 18:47:32 +00:00
bcd124cdba
Reduced execution time for CallTest::ReceivesPliAndRecovers, by dropping only one packet and made it predictable by removing rand().
...
Follow up steps is to support NackConfig.rtp_hostory_ms and/or increase fake encoder bitrate.
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6109005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5316 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-18 09:45:45 +00:00
1fa41be66a
Speeding up CallTest.ReceivesAndRetransmitsNack and removed the random packet loss.
...
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6119004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5315 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-18 09:44:53 +00:00
eb7b7bce3d
Modify video_render/ to allow a single old frame.
...
This stabilizes tests as a single frame reaches end-to-end, as well as
allowing slow or heavily-loaded systems to see any video updates even if
the frame takes more than 500ms in the pipeline.
R=mflodman@webrtc.org , stefan@webrtc.org
BUG=2724
Review URL: https://webrtc-codereview.appspot.com/5949004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5303 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-16 18:24:37 +00:00
919f87fb36
Delete capturers after destroying streams in test.
...
Since the renderers in CallTest.SendsAndReceiveStreams also stopped the
capturers they must be deleted after the VideoReceiveStream is stopped
or an use-after-free may occur.
R=mflodman@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/5969004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5300 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-16 14:55:54 +00:00
5ab756703e
Revert r5294 to re-roll r5293.
...
To fix races in test each stream now owns its own encoder/decoder.
R=mflodman@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/5919004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5297 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-16 12:24:44 +00:00
41e2615e02
Revert 5293 "Auto instantiate RBE depending on whether AST or TO..."
...
> Auto instantiate RBE depending on whether AST or TOF is available in incoming packet stream.
>
> BUG=
> R=mflodman@webrtc.org , stefan@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/5409004
TBR=solenberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5889004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5294 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-15 18:42:32 +00:00
341e91441a
Auto instantiate RBE depending on whether AST or TOF is available in incoming packet stream.
...
BUG=
R=mflodman@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5409004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5293 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-13 23:57:54 +00:00
92c2793154
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
2013-12-13 16:36:28 +00:00
1d096901ac
Move realtime tests to webrtc_perf_tests.
...
New binary not to be run on our VMs as they result in flaky tests. These
will instead be run on baremetal machines.
BUG=2710
R=kjellander@webrtc.org , mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5679004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5283 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-13 12:48:05 +00:00
724947b8ef
Add SwapFrame() to VideoSendStreamInput.
...
Optionally prevents doing a frame copy when putting frames into a
VideoSendStream. PutFrame() is still there, which copies the frame.
Also removes time_since_capture_ms as a parameter, since
I420VideoFrame::render_time_ms() denotes when the frame was captured.
BUG=2657
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5119004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5265 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-11 16:26:16 +00:00
0f3d0bb601
Stop video capturers in multi-stream test.
...
Expected to reduce runtime and flakiness in
CallTest.SendsAndReceivesMultipleStreams on linux_memcheck which is
presumed to be due to contention between the threads.
BUG=
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5249004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5238 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-06 15:48:17 +00:00
5cea89f3e1
Remove CallTest dependency on voice_engine/test/.
...
Loading file out of resources/ instead of data/ which is deprecated.
BUG=
R=holmer@google.com , mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5069004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5226 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-05 14:24:17 +00:00
c49d5b7df8
Move implementation files out of the webrtc/ root.
...
Leaves the root for public headers. Also fixes the issue of requiring
root OWNERS approval for changes in the Call implementation and adding
end-to-end tests.
BUG=
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5049005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5223 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-05 12:11:47 +00:00
