Revert 7184 "Enable ipv6 by default for webrtc under a Finch exp..."

Breaks Chrome build and prevents rolling WebRTC into Chrome DEPS.

> Enable ipv6 by default for webrtc under a Finch experiment.
> 
> BUG=413437 (chromium)
> https://code.google.com/p/chromium/issues/detail?id=413437
> 
> Review URL: https://webrtc-codereview.appspot.com/23529005

TBR=guoweis@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30399004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7190 4adac7df-926f-26a2-2b94-8c16560cd09d
memcheck: suppressions didn't map over directly when moving base from talk to webrtc (part of the suppression that is not related to the signature differed). Fixed suppressions accordingly.

BUG=N/A
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29499004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7191 4adac7df-926f-26a2-2b94-8c16560cd09d
Fix GN for rtc_base_approved target.

In https://webrtc-codereview.appspot.com/22649004
a new target was introduced that duplicated some
source files, breaking the bots in
http://build.chromium.org/p/chromium.webrtc.fyi/waterfall

This updates the GN config to also remove them from
the target where they were moved from in base.gyp.

BUG=3806
TESTED=Trybots + Running GN in a Chromium checkout with
src/third_party/webrtc symlinked to the WebRTC checkout
with this CL applied + passing compile step.

R=perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23669004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7192 4adac7df-926f-26a2-2b94-8c16560cd09d
rtc_unittest: prevent execution of broken tests.

BUG=1976
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29509004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7193 4adac7df-926f-26a2-2b94-8c16560cd09d
(Auto)update libjingle 75610402-> 75610402

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7194 4adac7df-926f-26a2-2b94-8c16560cd09d
Mark all virtual overrides in the hierarchies of UdpTransportData and
UdpSocketWrapper as such.

This will make further changes to these classes safer by ensuring that the
compile breaks if the base class changes and not all overrides are fixed.

This also removes an unused function.

BUG=none
TEST=none
R=henrik.lundin@webrtc.org, henrikg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23579004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7195 4adac7df-926f-26a2-2b94-8c16560cd09d
Add pbos@webrtc.org (myself) to talk/media/webrtc/.

Allows easier reviews of webrtcvideoengine2.cc and landing the new video
API on shorter review cycles.

R=pthatcher@webrtc.org
BUG=1788

Review URL: https://webrtc-codereview.appspot.com/30409004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7196 4adac7df-926f-26a2-2b94-8c16560cd09d
Recreate VideoStreams when setting resolution.

Instead of just changing resolution on the last stream streams are
reallocated to make sure that all streams are updated to match the
new input resolution.

R=pthatcher@webrtc.org
BUG=1788

Review URL: https://webrtc-codereview.appspot.com/29469004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7197 4adac7df-926f-26a2-2b94-8c16560cd09d