Merge 7343 "Changed mips_arch_variant variable value correspondi..."

> Changed mips_arch_variant variable value corresponding to Chromium code changes.
> 
> Chromium commit URL: https://crrev.com/c8a5da7455b57b2399e4a69e8100c098d9870052
> 
> R=andrew@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/23809004
> 
> Patch from Ljubomir Papuga <lpapuga@mips.com>.

TBR=ljubomir.papuga@gmail.com
BUG=chromium:438351

Review URL: https://webrtc-codereview.appspot.com/32279004

git-svn-id: http://webrtc.googlecode.com/svn@7785 4adac7df-926f-26a2-2b94-8c16560cd09d
Merge 7729 "Build fix for MIPS Android Webview build."

> Build fix for MIPS Android Webview build.
> 
> Excluding optimizations to support MIPS32R6 platform for Android Webview build (see also https://code.google.com/p/webrtc/source/detail?r=7580).
> 
> R=andrew@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/32459004

TBR=ljubomir.papuga@gmail.com
BUG=chromium:438351

Review URL: https://webrtc-codereview.appspot.com/27329004

git-svn-id: http://webrtc.googlecode.com/svn@7786 4adac7df-926f-26a2-2b94-8c16560cd09d
Change minimum video encoder initialization resolution to
176x144 to ensure HW encoder can be initialized.

R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32269004

git-svn-id: http://webrtc.googlecode.com/svn@7787 4adac7df-926f-26a2-2b94-8c16560cd09d
Don't reset sequence number for a stream on deactivate/reactivate.

BUG=chromium:431908
R=pbos@webrtc.org, sprang@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28119004

git-svn-id: http://webrtc.googlecode.com/svn@7788 4adac7df-926f-26a2-2b94-8c16560cd09d
Re-enable test: VideoProcessorIntegrationTest.ProcessNoLossChangeBitRateVP9.

BUG=4059

TBR=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30319004

git-svn-id: http://webrtc.googlecode.com/svn@7789 4adac7df-926f-26a2-2b94-8c16560cd09d
Simplify audio_buffer APIs

Now there is only one API to get the data or the channels (one const and one no const) merged or by band.
The band is passed in as a parameter, instead of calling different methods.

BUG=webrtc:3146
R=andrew@webrtc.org, bjornv@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27249004

git-svn-id: http://webrtc.googlecode.com/svn@7790 4adac7df-926f-26a2-2b94-8c16560cd09d
Adjust parameter in videoprocessor_integration_test for vp9.

TBR=stefan@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/33459004

git-svn-id: http://webrtc.googlecode.com/svn@7791 4adac7df-926f-26a2-2b94-8c16560cd09d
TurnPort should ignore STUN binding reponses when using shared socket.

BUG=4043
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27289004

git-svn-id: http://webrtc.googlecode.com/svn@7792 4adac7df-926f-26a2-2b94-8c16560cd09d
Convert DEPS to only reference Git repos

Also replace all doublequoted Python strings
with single-quoted ones.

BUG=chromium:412012
R=phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30289004

git-svn-id: http://webrtc.googlecode.com/svn@7793 4adac7df-926f-26a2-2b94-8c16560cd09d
Add Android test runner script for WebRTC.

The Android test execution toolchain scripts in Chromium
has been causing headaches for us several times. Mostly
because they're tailored at running Chrome tests only.

Wrapping their script in our own avoids the pain of
upstreaming new test names to Chromium and rolling them
in to get them running on our bots.

TESTED=Ran a test on a local device using:
webrtc/build/android/test_runner.py gtest -s audio_decoder_unittests  --verbose --isolate-file-path webrtc/modules/audio_coding/neteq/audio_decoder_unittests.isolate --release
TBR=phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25269004

git-svn-id: http://webrtc.googlecode.com/svn@7794 4adac7df-926f-26a2-2b94-8c16560cd09d
Verify that cpu_monitor exists before calling Stop().

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25259004

git-svn-id: http://webrtc.googlecode.com/svn@7795 4adac7df-926f-26a2-2b94-8c16560cd09d
Adding a duration printout to neteq_rtpplay

BUG=2692
TBR=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30339004

git-svn-id: http://webrtc.googlecode.com/svn@7796 4adac7df-926f-26a2-2b94-8c16560cd09d
Check if cpu_monitor_ exists before Stop().

R=asapersson@webrtc.org
BUG=1788

Review URL: https://webrtc-codereview.appspot.com/25279004

git-svn-id: http://webrtc.googlecode.com/svn@7797 4adac7df-926f-26a2-2b94-8c16560cd09d
Move the AudioDecoder interface out of NetEq

It belongs with the codecs, next to the AudioEncoder interface.

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

Review URL: https://webrtc-codereview.appspot.com/27309004

git-svn-id: http://webrtc.googlecode.com/svn@7798 4adac7df-926f-26a2-2b94-8c16560cd09d
Revert r7798 ("Move the AudioDecoder interface out of NetEq")

Apparently, it caused all sorts of problems I don't have time to
straighten out right now.

TBR=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25289004

git-svn-id: http://webrtc.googlecode.com/svn@7799 4adac7df-926f-26a2-2b94-8c16560cd09d
Remove need for assembly offset generation in aecm and ns module.

All *neon.S files in aecm and ns modules have been removed. We need no
assembly offset generation now.

Pass byte to byte conformance test for aecm and ns test in audioproc
between new NEON (written in intrinsics) version and C version on both
ARMv7 and ARM64.

BUG=3580
R=andrew@webrtc.org, jridges@masque.com

Change-Id: I05d43d0c04d00bead65ca8c8fda25f0a42394b2b

Review URL: https://webrtc-codereview.appspot.com/32229004

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

git-svn-id: http://webrtc.googlecode.com/svn@7800 4adac7df-926f-26a2-2b94-8c16560cd09d
Rename internal AudioEncoder::Encode method to EncodeInternal

BUG=3926
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28129004

git-svn-id: http://webrtc.googlecode.com/svn@7801 4adac7df-926f-26a2-2b94-8c16560cd09d
add WebRtcIsacfix_AutocorrNeon's intrinsics version

The modification only uses the unique part of the
 WebRtcIsacfix_AutocorrC function. Pass FiltersTest.AutocorrFixTest test
 on both ARMv7 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-a15 |
| use C as the base on each  |  (1.2Ghz) |   (1.7Ghz) |
| CPU target                 |           |            |
|----------------------------+-----------+------------|
| Neon asm                   |       24% |        23% |
| Neon intrinsics (GCC 4.6)  |       33% |        32% |
| Neon intrinsics (GCC 4.8)  |       27% |        27% |

BUG=3850
R=andrew@webrtc.org, jridges@masque.com

Change-Id: Id6cd0671502fadbebd10b1f5493f5b16c988286f

Review URL: https://webrtc-codereview.appspot.com/27999004

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

git-svn-id: http://webrtc.googlecode.com/svn@7802 4adac7df-926f-26a2-2b94-8c16560cd09d
Adding WebRtcSpl_MaxAbsValueW16 intrinsics version

The modification only uses the unique part of the WebRtcSpl_MaxAbsValue
 function. Pass Spltest.MinMaxOperationTest conformance test on both
 ARMv7 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-a15 |
| use C as the base on each  |  (1.2Ghz) |   (1.7Ghz) |
| CPU target                 |           |            |
|----------------------------+-----------+------------|
| Neon asm                   |       32% |        15% |
| Neon intrinsics (GCC 4.6)  |       36% |        37% |
| Neon intrinsics (GCC 4.8)  |       35% |        18% |

BUG=3580
R=andrew@webrtc.org, jridges@masque.com

Change-Id: Ia2f6822ec58774b401cc440b6751a97e540b5048

Review URL: https://webrtc-codereview.appspot.com/30109004

git-svn-id: http://webrtc.googlecode.com/svn@7803 4adac7df-926f-26a2-2b94-8c16560cd09d
Implement GetState() for channel's connectivity check state.

Previously, IceState is considered completed when there is only one connection (and the rest was trimmed). However, since the trimming logic is only done within the scope of network, when IPv6 and IPv4 both exist, the completion event is never fired.

This change adds the GetState() to each channel and it could decide what Completion means. The transport object then aggregates all channels before determining it's completed.

Each channel's IceState will be aggregrated at Transport level for overall Ice state

BUG=411086
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30949004

git-svn-id: http://webrtc.googlecode.com/svn@7804 4adac7df-926f-26a2-2b94-8c16560cd09d