.gitignore updates

Update after Chromium roll in https://review.webrtc.org/24179004/
and Android project updates in https://review.webrtc.org/25029004/

BUG=
TBR=phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24239004

git-svn-id: http://webrtc.googlecode.com/svn@7665 4adac7df-926f-26a2-2b94-8c16560cd09d
Add unit to dropped frames.

Missing unit causes less dropped frames to be reported as a regression
and not an improvement.

R=stefan@webrtc.org
BUG=chromium:429206

Review URL: https://webrtc-codereview.appspot.com/25139004

git-svn-id: http://webrtc.googlecode.com/svn@7666 4adac7df-926f-26a2-2b94-8c16560cd09d
Refactor SetDefaultEncoderConfig to work on existing codecs.

Addresses issue where SetDefaultEncoderConfig modifies the codec list
rather than just the targeted codec. This was previously done just to
pass more unit tests rather than be done properly. This incidentally
addresses a TODO causing this to work with external codecs as well.

R=stefan@webrtc.org
BUG=1788

Review URL: https://webrtc-codereview.appspot.com/32009004

git-svn-id: http://webrtc.googlecode.com/svn@7667 4adac7df-926f-26a2-2b94-8c16560cd09d
Put send-side bwe probing under finch experiment.

BUG=crbug/425925
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26079004

git-svn-id: http://webrtc.googlecode.com/svn@7668 4adac7df-926f-26a2-2b94-8c16560cd09d
Wire up DSCP support in WebRtcVideoEngine2.

R=stefan@webrtc.org
BUG=1788

Review URL: https://webrtc-codereview.appspot.com/24249004

git-svn-id: http://webrtc.googlecode.com/svn@7669 4adac7df-926f-26a2-2b94-8c16560cd09d
Removes talk/xmllite, talk/xmpp and talk/p2p as they are no longer used by gyp/gn builds.

TBR=niklas.enbom@webrtc.org
BUG=3379

Review URL: https://webrtc-codereview.appspot.com/30959004

git-svn-id: http://webrtc.googlecode.com/svn@7670 4adac7df-926f-26a2-2b94-8c16560cd09d
clear asm code and unused functions in audio processing module

BUG=
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25119004

Patch from Zhongwei Yao <zhongwei.yao@arm.com>.

git-svn-id: http://webrtc.googlecode.com/svn@7671 4adac7df-926f-26a2-2b94-8c16560cd09d
replace inline assembly WebRtcAecm_ResetAdaptiveChannelNeon  by intrinsics.

The modification only uses the unique part of the ResetAdaptiveChannel
 function. Pass byte to byte conformance test both on ARM32 and ARM64,
 and the single function performance is similar with original assembly
 version on different platforms. If not specified, the code is compiled
 by GCC 4.6. The result is the "X version / C version" ratio, and the
 less is better.

| run 100k times             | cortex-a7 | cortex-a9 | cortex-a15 |
| use C as the base on each  |  (1.2Ghz) |  (1.0Ghz) |   (1.7Ghz) |
| CPU target                 |           |           |            |
|----------------------------+-----------+-----------+------------|
| Neon asm                   |       15% |       30% |        12% |
| Neon inline                |       21% |       30% |        12% |
| Neon intrinsics (GCC 4.6)  |       19% |       32% |        12% |
| Neon intrinsics (GCC 4.8)  |       20% |       32% |        12% |
| Neon intrinsics (LLVM 3.4) |       19% |       30% |        12% |

BUG=3580
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29019004

Patch from Zhongwei Yao <zhongwei.yao@arm.com>.

git-svn-id: http://webrtc.googlecode.com/svn@7672 4adac7df-926f-26a2-2b94-8c16560cd09d
Change from talk/p2p (r7572): "Improve the logging when a TCP connection is deleted."

BUG=3379
R=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27109004

git-svn-id: http://webrtc.googlecode.com/svn@7673 4adac7df-926f-26a2-2b94-8c16560cd09d
Change from talk/p2p (r7664) "(Auto)update libjingle 79414100-> 79428003".

BUG=3379
R=tpsiaki@google.com

Review URL: https://webrtc-codereview.appspot.com/27119004

git-svn-id: http://webrtc.googlecode.com/svn@7674 4adac7df-926f-26a2-2b94-8c16560cd09d
Make an AudioEncoder subclass for iSAC

BUG=3926
R=henrik.lundin@webrtc.org, kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25019004

git-svn-id: http://webrtc.googlecode.com/svn@7675 4adac7df-926f-26a2-2b94-8c16560cd09d
Revert 7675 "Make an AudioEncoder subclass for iSAC"

Above CL did not compile on Android. Followings are links to Android builds

http://chromegw.corp.google.com/i/internal.client.webrtc/builders/Android%20Builder%20%28dbg%29/builds/2648

http://chromegw.corp.google.com/i/internal.client.webrtc/builders/Android%20Clang%20%28dbg%29/builds/2369

http://chromegw.corp.google.com/i/internal.client.webrtc/builders/Android%20ARM64%20%28dbg%29/builds/1320

> Make an AudioEncoder subclass for iSAC
> 
> BUG=3926
> R=henrik.lundin@webrtc.org, kjellander@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/25019004

TBR=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32439004

git-svn-id: http://webrtc.googlecode.com/svn@7676 4adac7df-926f-26a2-2b94-8c16560cd09d
Follow-up fixes for G722

This CL addresses post-commit comments on r7662. See
https://webrtc-codereview.appspot.com/27089004/#ps40001.

BUG=3951
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30979004

git-svn-id: http://webrtc.googlecode.com/svn@7677 4adac7df-926f-26a2-2b94-8c16560cd09d