Change the way we reference enumerators in H.264 packetization code to be standard C++ compliant.

R=kjellander@webrtc.org, phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21109004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6833 4adac7df-926f-26a2-2b94-8c16560cd09d
GN: Add PRESUBMIT.py check for GN changes + default bots.

Add the GN trybots to the default set and also set them
to be the only bots to run if a CL contains only BUILD.gn
changes.

Update Python exclusions in general and fix a few of the lint
warnings.
The ones in python_charts needs to be disabled since those variables
are actually used when passed via vars() to the template.

BUG=None
TEST=git cl presubmit with the following cases:
A CL with two .gyp changes.
A CL with no changes in .gyp* files.

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

Review URL: https://webrtc-codereview.appspot.com/18719004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6834 4adac7df-926f-26a2-2b94-8c16560cd09d
Add end-to-end H.264 packetization test.

Also correctly wires up H.264 packetization in the new Call api.

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20079004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6835 4adac7df-926f-26a2-2b94-8c16560cd09d
Fixing two bugs in voe_cmd_test.

I am trying to add a new functionality to voe_cmd_test, and I found two bugs:

1. in r5928, a functionality was removed but the item in the menu was not. Functionalities after it are offset.

r5928: https://code.google.com/p/webrtc/source/detail?r=5928&path=/trunk/webrtc/voice_engine/test/cmd_test/voe_cmd_test.cc

2. in r6736, opus are set to output 48 kHz audio. When mixing Opus output with an audio file, channel.cc may go wrong.

r6736: https://code.google.com/p/webrtc/source/detail?r=6736

BUG=
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15099004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6836 4adac7df-926f-26a2-2b94-8c16560cd09d
Fix crashing fake network pipe tests.

These tests are not included in bots, this will be fixed in a follow-up by pbos@webrtc.org.

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13109004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6837 4adac7df-926f-26a2-2b94-8c16560cd09d
RTP video playback tool using Call APIs.

Plays back rtpdump files from Wireshark in realtime as well as save the
resulting raw video to file. Unlike the RTP playback tool it doesn't
support faster-than-realtime playback/rendering, but it instead utilizes
the same path as production code and also contains support for playing
back FEC.

BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16969004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6838 4adac7df-926f-26a2-2b94-8c16560cd09d
Allow root build dependencies to be overridden.

RISK=P2
TESTED=manual
R=andrew@webrtc.org, thorcarpenter@google.com

Review URL: https://webrtc-codereview.appspot.com/19009004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6839 4adac7df-926f-26a2-2b94-8c16560cd09d
Revert 6839 "Allow root build dependencies to be overridden."

> Allow root build dependencies to be overridden.
> 
> RISK=P2
> TESTED=manual
> R=andrew@webrtc.org, thorcarpenter@google.com
> 
> Review URL: https://webrtc-codereview.appspot.com/19009004

TBR=harryjin@google.com

Review URL: https://webrtc-codereview.appspot.com/20099004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6840 4adac7df-926f-26a2-2b94-8c16560cd09d
(Auto)update libjingle 72682155-> 72785180

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6841 4adac7df-926f-26a2-2b94-8c16560cd09d
(Auto)update libjingle 72785180-> 72785516

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6842 4adac7df-926f-26a2-2b94-8c16560cd09d