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Also improves the tests by adding a test case for FEC, and running the a/v sync
tests with NACK and simulated packet loss.
BUG=crbug/374104
R=mflodman@webrtc.org, pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19209004
git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7053 4adac7df-926f-26a2-2b94-8c16560cd09d
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decoder to multiple received streams.
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12949004
git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6651 4adac7df-926f-26a2-2b94-8c16560cd09d
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Stop-gap solution to support VP8 codec settings in the new API until
encoder settings can be passed on to the VideoEncoder without requiring
explicit support for the codec.
BUG=3424
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17929004
git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6650 4adac7df-926f-26a2-2b94-8c16560cd09d
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A restarted VideoSendStream would previously be completely reset,
causing gaps in sequence numbers and potentially RTP timestamps as well.
This broke SRTP which requires fairly sequential sequence numbers.
Presumably, were this sent without SRTP, we'd still have problems on the
receiving end as the corresponding receiver is unaware of this reset.
Also adding annotation to RTPSender and addressing some unlocked
access to ssrc_, ssrc_rtx_ and rtx_.
BUG=
R=mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20819004
git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6612 4adac7df-926f-26a2-2b94-8c16560cd09d
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Defaults for configs are instead placed in the Config constructors.
BUG=
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/18729004
git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6608 4adac7df-926f-26a2-2b94-8c16560cd09d
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Allows setting SSRCs for future simulcast layers even though no set send
codec uses them.
Also re-enabling CanSwitchToUseAllSsrcs as an end-to-end test, required
for bitrate ramp-up, instead of send-side only (resolving issue 3078).
This test was used to verify reserved modules' SSRCs are preserved
correctly.
To enable a multiple-stream end-to-end test test::CallTest was modified
to work on a vector of receive streams instead of just one.
BUG=3078
R=kjellander@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/15859005
git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6565 4adac7df-926f-26a2-2b94-8c16560cd09d
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Greatly reduces duplication of constants and setup code for tests based
on the new webrtc::Call APIs. It also makes it significantly easier to
convert sender-only to end-to-end tests as they share more code.
BUG=3035
R=kjellander@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17789004
git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6551 4adac7df-926f-26a2-2b94-8c16560cd09d
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