Remove GetDefaultConfigs() from Call.

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@6608 4adac7df-926f-26a2-2b94-8c16560cd09d
Fix data races related with traces in bitrate estimator test.

BUG=3549
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16019004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6609 4adac7df-926f-26a2-2b94-8c16560cd09d
Fix pacer to accept duplicate sequence numbers on different SSRCs.

BUG=3550
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17889004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6610 4adac7df-926f-26a2-2b94-8c16560cd09d
Add initial gn build files for video_coding and video_processing.

BUG=
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19819004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6611 4adac7df-926f-26a2-2b94-8c16560cd09d
Preserve RTP states for restarted VideoSendStreams.

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@6612 4adac7df-926f-26a2-2b94-8c16560cd09d
Add test for VideoEncoder setup/teardown.

Verifies that InitEncode and RegisterEncodeCompleteCallback gets
called before Encode is called. Also verifies that teardown is correctly
done during DestroyVideoSendStream().

BUG=2339
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20859004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6613 4adac7df-926f-26a2-2b94-8c16560cd09d
drmemory flaky: EndToEndTest.RestartingSendStreamPreservesRtpState[WithRtx] suppressed on drMemory.

BUG=3552
TBR=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17899004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6614 4adac7df-926f-26a2-2b94-8c16560cd09d
webrtc/base: Fixes miss in base.gyp for windows. See https://code.google.com/p/webrtc/source/browse/trunk/talk/libjingle.gyp?r=6503#764 for the corresponding condition.

BUG=3379
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19849004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6615 4adac7df-926f-26a2-2b94-8c16560cd09d
Fix potential deadlock in webrtc/system_wrappers/source/logging_unittest.cc.

crit_ should not be held while calling Trace.

BUG=3003
R=henrike@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17859005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6616 4adac7df-926f-26a2-2b94-8c16560cd09d
Clean data races from system_wrappers_unittests.

 - Remove unittest_utilities that are not used.
 - Remove SetLevelFilter that does not seems necessary and anyhow was racy.

BUG=3549
R=henrike@webrtc.org, henrike

Review URL: https://webrtc-codereview.appspot.com/16819004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6617 4adac7df-926f-26a2-2b94-8c16560cd09d
Disabling shared socket mode for TURN ports. This is done as currently when
TURN server also used as STUN server, binding responses will be handed over
to TURN port, which simply discard these messages, as requests are originated
from StunPort.

Until we find the right solution for this problem, it's better we disable this
feature.

BUG=https://code.google.com/p/webrtc/issues/detail?id=3537
R=jiayl@webrtc.org, juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19889004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6618 4adac7df-926f-26a2-2b94-8c16560cd09d
Extract RTP-header SSRC inline in Call.

Prevents unknown-RTP-header-extension warnings to be flooding from the
RTP-header parsing as there's no way to register RTP extensions for the
parser in Call as they're allowed to differ between RTP streams.

RTP-header parsing should instead be done separately in every
VideoReceiveStream.

BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16829004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6619 4adac7df-926f-26a2-2b94-8c16560cd09d
Tight data race suppressions around thread_posix.

BUG=3372,3549
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19879004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6620 4adac7df-926f-26a2-2b94-8c16560cd09d
Adding explicit check for using dummy file devices.

Calling into the file device factory without being compiled with file
devices makes no sense and would cause hard-to-debug errors. Therefore
I'm adding an explicit check so this isn't allowed.

R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13849004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6621 4adac7df-926f-26a2-2b94-8c16560cd09d
Fixing compile error.

Made a mistake in https://webrtc-codereview.appspot.com/13849004/,
fixing that here.

TBR=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13859004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6622 4adac7df-926f-26a2-2b94-8c16560cd09d
Some refactoring inside rtp_rtcp/.

Renaming ModuleRTPUtility -> RtpUtility.
Renaming RTPHeaderParser -> RtpHeaderParser.
Making RtpHeaderParser accept size_t instead of int for packet length.
Making RtpUtility::RtpHeaderParser accept size_t for packet length.

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

Review URL: https://webrtc-codereview.appspot.com/19899004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6623 4adac7df-926f-26a2-2b94-8c16560cd09d