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11 Commits

Author SHA1 Message Date
pbos@webrtc.org
de1429e9ad Add thread annotations to Call API. 
Also constified a lot of pointers and reordered members to make
protected members more grouped together.

R=kjellander@webrtc.org, stefan@webrtc.org
BUG=2770

Review URL: https://webrtc-codereview.appspot.com/15399004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5998 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-28 13:00:21 +00:00
pbos@webrtc.org
3349ae0cdc Implement minimum transmit bitrate. 
Utilizing minimum transmission bitrate prevents low remote bitrate
estimates (bitrate estimation dips) when encoding non-complex content
such as screenshare of a static image even though there's nothing wrong
with the link.

Requires pacing to be enabled for now, pending issue 3036.

BUG=3014
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/9719004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5694 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-13 12:52:27 +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
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
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
pbos@webrtc.org
b3cc78de28 Add -Wnon-virtual-dtor warning for C++ code. 
BUG=2659
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4119006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5149 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-21 11:42:02 +00:00
pbos@webrtc.org
27326b6a42 Rename newapi::Transport::SendRTP()->SendRtp(). 
Also fit rampup_tests.cc to use internal::TransportAdapter instead of
implementing its own.

BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3419004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5138 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-20 12:17:04 +00:00
stefan@webrtc.org
28bf50f0ec Fix test broken with r5128. 
TBR=pbos@webrtc.org
BUG=2530

Review URL: https://webrtc-codereview.appspot.com/3979004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5129 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-18 11:58:24 +00:00
stefan@webrtc.org
b082ade3db Hook up audio/video sync to Call. 
Adds an end-to-end audio/video sync test.

BUG=2530, 2608
TEST=trybots
R=henrika@webrtc.org, mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3699004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5128 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-18 11:45:11 +00:00
stefan@webrtc.org
69969e2e2f Improve Call tests for RTX. 
Also does some refactoring to reuse RtpRtcpObserver.

BUG=1811
R=mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3809004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5126 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-15 12:32:15 +00:00
pbos@webrtc.org
16e03b7bd8 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
 2013-10-28 16:32:01 +00:00