Remove former team members from OWNERS and WATCHLISTS

Remove the following (CCed) former team members from all
OWNERS files and the WATCHLISTS file:
* fischman@
* leozwang@
* mikhal@
* pwestin@
* wu@

BUG=
R=henrike@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22509004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6973 4adac7df-926f-26a2-2b94-8c16560cd09d
Remove Android.mk build files.

These files are generally not maintained and break, some contain files
that don't exist anymore and do not build anymore. If we need to add
some of these back we should really set up a bot for them.

R=andrew@webrtc.org, glaznev@webrtc.org, henrike@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/15249004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6974 4adac7df-926f-26a2-2b94-8c16560cd09d
common_audio: Re-enable WebRtcSpl_AddSatW32() and WebRtcSpl_SubSatW32() optimizations on armv7

According to the issue, common_audio_unittests failed on armv7. It currently pass, so we should turn it on again. There is no print out in the issue, so the cause of failure is unknown.

BUG=740
TESTED=locally on N7
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14199004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6975 4adac7df-926f-26a2-2b94-8c16560cd09d
Refactoring common_audio/signal_processing: Remove macro WEBRTC_SPL_UDIV

This macro is a direct use of the division operator without checking for division by zero. Hence, it is dangerous to use.
This CL replaces the macro with '/' at place.

BUG=3348,3353
TESTED=locally on linux and trybots
R=kwiberg@webrtc.org, tina.legrand@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14169004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6976 4adac7df-926f-26a2-2b94-8c16560cd09d
Remove test constructor in WebRtcVideoEngine2.

Removes the need for ::Construct().

BUG=1788
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15279004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6977 4adac7df-926f-26a2-2b94-8c16560cd09d
Updating svn:ignore entries

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6978 4adac7df-926f-26a2-2b94-8c16560cd09d
Fix data races in VideoAdapterTest.

Adressing clear races between the test thread and capturer thread shown
as heap-use-after-free in vpx_codec_destroy in
WebRtcVideoMediaChannelTest.SetSend (way later in the rest run).

When capturing a frame the test copied it to a separate frame that would
then be read by the test without synchronization, if the test didn't
manage to examine the frame in between captures the adapted frame would
be overwritten by the following frame during accesses to it.

The actual races are suppressed by race:webrtc/base/messagequeue.cc and
race:webrtc/base/thread.cc. These fixes reduce the suppression count
locally from around 3000 to 30 for VideoAdapterTest.*.

Also removing tsan suppressions for talk/base as it's been moved to
webrtc/base.

R=tommi@webrtc.org
BUG=3671

Review URL: https://webrtc-codereview.appspot.com/22169004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6979 4adac7df-926f-26a2-2b94-8c16560cd09d
Fix Win64 compile of videoadapter_unittest.cc.

Missed an typecast in videoadapter_unittest.cc in r6979 due to
tryservers being clogged and me waiting for a windows, linux, mac and
tsanv2 bot to finish was not enough. Committing fix straight away to
un-break tree.

TBR=tommi@webrtc.org
BUG=3671

Review URL: https://webrtc-codereview.appspot.com/18279004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6980 4adac7df-926f-26a2-2b94-8c16560cd09d
Disable EndToEndTest.RestartingSendStreamPreservesRtpState in video_engine_tests because it is flaky

BUG=webrtc:3745
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21349004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6981 4adac7df-926f-26a2-2b94-8c16560cd09d
(Auto)update libjingle 74128148-> 74132319

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6982 4adac7df-926f-26a2-2b94-8c16560cd09d
(Auto)update libjingle 74132319-> 74133664

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6983 4adac7df-926f-26a2-2b94-8c16560cd09d
Avoid syncing unnecessary Chromium deps for WebRTC.

This should save several gigabytes of traffic and disk space.

On Linux this is about 2.6 GB:
346M	src/chrome/tools/test/reference_build
340M	src/native_client
170M	src/third_party/ffmpeg
1.5G	src/third_party/WebKit
196M	src/v8

BUG=2863
TESTED=Removed the directories locally, ran a sync and verified they didn't reappear (or fail because of platform-specific ones).
R=iannucci@chromium.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22189004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6984 4adac7df-926f-26a2-2b94-8c16560cd09d
implement handling ALTERNATE-SERVER response from turn protocol as
specified in RFC 5766, also created 2 test cases for both the normal
redirection case as well as when a pingpong situation happens, the
allocation should fail

BUG=1986 TURN ALTERNATE-SERVER support
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21249004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6985 4adac7df-926f-26a2-2b94-8c16560cd09d
Create a copy of talk/sound under webrtc/sound.

BUG=3379
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22379004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6986 4adac7df-926f-26a2-2b94-8c16560cd09d
- Make local constant non-static.
- Remove spammy log line.

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

Review URL: https://webrtc-codereview.appspot.com/21339004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6987 4adac7df-926f-26a2-2b94-8c16560cd09d
Expose setPayloadType on the rtp_sender. Thus allowing other users of this module
to set the payload type to be used without having to call SendOutgoingData.

BUG=3694
R=asapersson@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18289004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6988 4adac7df-926f-26a2-2b94-8c16560cd09d
Move end of namespace inside #ifdef

The code did not compile unless WEBRTC_ANDROID was defined.

R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21319004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6989 4adac7df-926f-26a2-2b94-8c16560cd09d
(Auto)update libjingle 74202294-> 74230205

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6990 4adac7df-926f-26a2-2b94-8c16560cd09d
Fix data races during VideoAdapterTest tear-down.

Explicitly disconnect the VideoCapturer to avoid frames being
delivered during listener destruction. This manifested only on DrMemory
Full on Windows which I was able to repro locally.

BUG=3671
R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15289004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6991 4adac7df-926f-26a2-2b94-8c16560cd09d
Re-enable all VideoAdapterTests on DrMemory.

These bugs should've been resolved as of r6991.

BUG=3655,3671
R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22529004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6992 4adac7df-926f-26a2-2b94-8c16560cd09d