Fire OnRenegotiationNeeded only for the first SCTP DataChannel.
Subsequent DataChannels do not need renegotiation since SCTP data streams are not negotiated through SDP.

BUG=2431
R=pthatcher@google.com, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12629004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6268 4adac7df-926f-26a2-2b94-8c16560cd09d
Closes the DataChannel when the send buffer is full or on transport errors.
As stated in the spec.

BUG=2645
R=pthatcher@google.com, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12619004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6270 4adac7df-926f-26a2-2b94-8c16560cd09d
Rebase webrtc/base with r6250:
cd webrtc/base
svn diff -r 6249:6250 http://webrtc.googlecode.com/svn/trunk/talk/base >
6250.diff
patch -p0 -i 6250.diff

BUG=3379
TBR=phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16589004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6271 4adac7df-926f-26a2-2b94-8c16560cd09d
Update generated asm offsets scripts.

Libvpx updated the unpack scripts to fix building dependencies.

Roll libvpx 269083:273304
See https://codereview.chromium.org/295313002/
https://codereview.chromium.org/298063002/
https://codereview.chromium.org/305533008/
https://codereview.chromium.org/305703002/
https://codereview.chromium.org/298383003/
https://codereview.chromium.org/302863004/
for the libvpx changes.

BUG=377062
R=andrew@webrtc.org, michaelbai@chromium.org

Review URL: https://webrtc-codereview.appspot.com/12579008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6272 4adac7df-926f-26a2-2b94-8c16560cd09d
* Revert clock.cc changes made in 6178, but keep the changes to the test.
* Use the new appoach proposed by jib in https://review.webrtc.org/10439004/ to fix the windows clock issue.

BUg=3325

R=niklas.enbom@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15569005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6273 4adac7df-926f-26a2-2b94-8c16560cd09d
Increase VPMVideoDecimator's initial max_frame_rate_ to 60, which allow us potentially do 60fps.

BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21499006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6274 4adac7df-926f-26a2-2b94-8c16560cd09d
AppRTCDemo(android): run in full-screen & immersive mode.

Also:
- Only show stats HUD on demand
- Only collect stats when HUD is showing
- Don't render solid green frame when video is not present in either direction

R=glaznev@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12639004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6275 4adac7df-926f-26a2-2b94-8c16560cd09d
Make MediaSessionDescriptionFactory accept offers with UDP/TLS/RTP/SAVPF.
This is the first step toward switching completely to UDP/TLS/RTP/SAVPF.

BUG=2796
R=juberti@webrtc.org, pthatcher@google.com

Review URL: https://webrtc-codereview.appspot.com/13439004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6276 4adac7df-926f-26a2-2b94-8c16560cd09d
(Auto)update libjingle 68203780-> 68206793

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6277 4adac7df-926f-26a2-2b94-8c16560cd09d