admin/webrtc_m130
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity
5,681 Commits 1 Branch 0 Tags
Commit Graph

116 Commits

Author SHA1 Message Date
henrik.lundin@webrtc.org
1598b80f52 Routing SuspendChange to VideoSendStream::Stats 
Also checking that the statistics are properly updated in
VideoSendStreamTest.SuspendBelowMinBitrate.

Adding a test to SendStatisticsProxyTest.

Checking callback status in rampup test, too.

BUG=2457
R=mflodman@webrtc.org, pbos@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/9439004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5678 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-11 14:57:35 +00:00
henrik.lundin@webrtc.org
36b6221cd4 Adding a link to issue 
BUG=3010
TBR=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/9639004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5668 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-10 10:24:35 +00:00
henrik.lundin@webrtc.org
ed865b5d46 NetEq4: Changing the behavior of playout_timestamp_ update 
The variable playout_timestamp_ was not updated to the latest decoded
timestamp while comfort noise was played. Instead, it was upadted using
dead reckoning, which caused it to drift away from the timestamps of the
incoming CNG packets. Now it is updated also during comfort noise
playout.

Since the change is only in NetEq4, this change also makes the test
PlaysOutAudioAndVideoInSync use both ACM1/NetEq3 and ACM2/NetEq4.

Re-enabling one NetEq unit test that is no longer failing thanks to this CL.

BUG=2932
R=stefan@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8799004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5649 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-06 10:28:07 +00:00
sprang@webrtc.org
60ad5fdadf Potential deadlock in VideoSendStreamTest::ProducesStats 
VideoSendStream::GetStats() should not be called by
RtpRtcpObserver::OnSendRtcp(), as at this stage that thread will still
hold internal send locks.

Use an event and signal the test thread to call GetStats() instead.

BUG=
R=mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/9359004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5648 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-06 10:03:36 +00:00
henrik.lundin@webrtc.org
998cb8fcd0 Use DISABLE_ instead of commenting out tests 
BUG=2636
TBR=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/9449004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5647 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-06 09:12:00 +00:00
henrik.lundin@webrtc.org
845862f279 Adding a new ramp-up-down-up test 
The new test is based upon the exisiting rampup test, but also adds
a low-rate period. The main purpose of the test is to verify the
SuspendBelowMinBitrate functionality, which must be enabled for the
test to pass.

The CL also adds a change to the min bitrate in the send-side bandwidth
estimator when SuspendBelowMinBitrate is enabled.

An anonymous namespace is added around the StreamObserver classes
in the test to avoid silent linker conflicts that could happen
otherwise.

Note: this CL depends on https://webrtc-codereview.appspot.com/9049004/

BUG=2636
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/9059004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5646 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-06 07:19:28 +00:00
pbos@webrtc.org
0117d1c48c Fix compilation errors under clang 3.5. 
Enables building tip-of-tree clang which introduces new warnings that
cause compilation errors in our code base (-Werror).

BUG=
R=andrew@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/9319004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5630 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-03 16:47:03 +00:00
sprang@webrtc.org
346094cb01 Incorrect overhead calculation when using FEC + RTP extension headers. 
When frames are fragmented inte multiple RTP packets in order to not
exceed a maximum packet size, the header overhead calculation must
take into account that FEC redundancy packets may use more than the
12 bytes of the basic RTP header. For example, a csrc list or extension
headers may be present.

BUG=2899
R=phoglund@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8769005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5562 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-18 08:40:33 +00:00
sprang@webrtc.org
9510e53cc0 Make VideoReceiveStream::GetStats() const. 
BUG=
R=mflodman@webrtc.org, pbos@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8169004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5501 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-07 15:32:45 +00:00
sprang@webrtc.org
09315705b9 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
 2014-02-07 12:06:29 +00:00
asapersson@webrtc.org
bdc5ed2e7d Add configuration for cpu overuse detection to video send stream. 
BUG=2422
R=mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7129004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5468 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-31 10:05:07 +00:00
solenberg@webrtc.org
094ac39b5a Fix race when deleting video receive streams in Call. 
BUG=
R=mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6889004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5457 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-29 11:21:58 +00:00
sprang@webrtc.org
d9b9560ee5 Drop early packets when not sending in TransportAdapter. 
Particularly, suppress periodic RTCP packets before
VideoSendStream.StartSending() or VideoReceiveStream.StartReceiving() have been called, respectively.

RTCP packets are sent periodically, by the Process thread, for every ViE channel even those not sending.

BUG=
R=pbos@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7569004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5438 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-27 13:03:02 +00:00
pbos@webrtc.org
c98882dcd3 Always initialize Trace in Call TraceDispatcher. 
Prevents violation of lock order occuring previously when
RegisterCallback called SetTraceCallback while holding its lock, which
called Print back (which acquires the lock).

BUG=
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7559004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5433 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-27 09:11:10 +00:00
pbos@webrtc.org
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
pbos@webrtc.org
e7223e7795 Set NACKed packet to -1 in TestNackRetransmission. 
Zero is a valid sequence number which may occur even if there are no
retransmissions, this caused the test to flake as an incoming packet
would be mistaken for a retransmission.

BUG=2830
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7509005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5417 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-23 16:14:34 +00:00
asapersson@webrtc.org
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
pbos@webrtc.org
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
solenberg@webrtc.org
ab2405164a Add another test case for AST/TOF switching. 
BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/5899005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5352 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-08 08:59:44 +00:00
sprang@webrtc.org
ccd42840bc Wire up statistics in video send stream of new video engine api 
Note, this CL does not contain any tests. Those are implemeted as call
tests and will be submitted when the receive stream is wired up as well.

BUG=2235
R=mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/5559006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5344 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-07 09:54:34 +00:00
sprang@webrtc.org
54ae4ffb9e Add callbacks for receive channel RTCP statistics. 
This allows a listener to receive new statistics as it is generated - avoiding the need to poll. This also makes handling stats from multiple RTP streams more tractable.
The change is primarily targeted at the new video engine API.

TEST=Unit test in ReceiveStatisticsTest. Integration tests to follow as call tests when fully wired up.

BUG=2235
R=henrika@webrtc.org, pbos@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/5089004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5323 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-19 13:26:02 +00:00
stefan@webrtc.org
faada6e604 Integrate fake_network_pipe into direct_transport. 
TEST=trybots
R=mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/5529004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5321 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-18 20:28:25 +00:00
mflodman@webrtc.org
b429e516a9 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
 2013-12-18 09:46:22 +00:00
mflodman@webrtc.org
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
mflodman@webrtc.org
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
pbos@webrtc.org
052fa6243a Stop transport in test SuspendBelowMinBitrate. 
Avoids race when packets are still left in the network while the Call is
being destroyed.

R=mflodman@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/6009004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5307 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-17 11:19:58 +00:00
pbos@webrtc.org
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
pbos@webrtc.org
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
pbos@webrtc.org
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
turaj@webrtc.org
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
solenberg@webrtc.org
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
mflodman@webrtc.org
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
pbos@webrtc.org
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
mflodman@webrtc.org
f3973e81d5 Make sure channels in the same call are in the same channel group. 
Tested manually. I'll make a follow CL with a proper test once review.webrtc.org/5619004 has been committed.

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/5509004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5280 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-13 09:40:45 +00:00
henrik.lundin@webrtc.org
e9abd591d7 Making RemoteRateControl::min_configured_bit_rate_ configurable 
The minimum bitrate can now be configured from WrappingBitrateEstimator.

BUG=2698
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/5699004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5279 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-13 08:42:42 +00:00
wu@webrtc.org
a9890800e0 Update talk to 58127566 together with 
https://webrtc-codereview.appspot.com/5309005/.

R=mallinath@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/5719004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5277 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-13 00:21:03 +00:00
wu@webrtc.org
2018269dc3 Revert 5274 "Update talk to 58113193 together with https://webrt..." 
> Update talk to 58113193 together with https://webrtc-codereview.appspot.com/5309005/.
> 
> R=mallinath@webrtc.org, niklas.enbom@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/5719004

TBR=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/5729004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5275 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-12 22:54:25 +00:00
wu@webrtc.org
a129b6cd13 Update talk to 58113193 together with https://webrtc-codereview.appspot.com/5309005/. 
R=mallinath@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/5719004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5274 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-12 22:40:39 +00:00
pbos@webrtc.org
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
sprang@webrtc.org
8b8819262f Improve VideoSendStreamTest::MaxPacketSize 
This CL was submitted as issue https://webrtc-codereview.appspot.com/4849004/, but was reverted because of flakiness. This new issue will correct that.

Patch Set 1 contains the code that was submitted in 4849004.

BUG=2428
R=pbos@webrtc.org, phoglund@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/5399004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5251 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-10 10:05:17 +00:00
andrew@webrtc.org
797522f9f2 Revert 5229 "Make VideoSendStreamTest::MaxPacketSize test a whol..." 
> Make VideoSendStreamTest::MaxPacketSize test a whole range of frame sizes, to make sure all corner cases are covered.
> 
> BUG=2428
> R=pbos@webrtc.org, phoglund@webrtc.org, stefan@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/4849004

It caused a failure in video_engine_tests on the Linux Tsan bot.

TBR=sprang@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/5269004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5240 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-06 17:42:32 +00:00
pbos@webrtc.org
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
sprang@webrtc.org
7104fc1906 Make VideoSendStreamTest::MaxPacketSize test a whole range of frame sizes, to make sure all corner cases are covered. 
BUG=2428
R=pbos@webrtc.org, phoglund@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4849004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5229 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-05 16:15:11 +00:00
pbos@webrtc.org
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
pbos@webrtc.org
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
stefan@webrtc.org
7e9315b42e Adds support for sending redundant payloads over RTX. 
TEST=trybots
BUG=1812
R=mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4169004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5209 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-04 10:24:26 +00:00
pbos@webrtc.org
b613b5ab2b 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
 2013-12-03 10:13:04 +00:00
pbos@webrtc.org
e1fc3f22ea Disable check for all sent SSRCs being valid. 
Since the code for setting these up will set the codec before setting
SSRCs for the streams, any frames sent in between will be sent on
random-generated SSRCs.

This part should be added back during work on issue 1695.

BUG=1695
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4629004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5192 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-28 15:40:12 +00:00
pbos@webrtc.org
13d38a13e3 Set up SSRCs correctly after switching codec. 
Before SSRCs were not set up correctly, as the old VideoEngine API
doesn't support setting additional SSRCs before a codec with as many
streams are set.

No test was in place to catch this, so two tests are added to make sure
that we send the SSRCs that are set, and also that we can switch from
using one to using all SSRCs, even though initially not all of them are
set up.

BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4539004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5188 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-28 11:59:31 +00:00
sprang@webrtc.org
4070935f4f Implement and test EncodedImageCallback in new ViE API. 
R=mflodman@webrtc.org, pbos@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4059004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5179 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-26 11:41:59 +00:00