Move WebRtcVideoRenderFrame from webrtcvideoengine2.cc to webrtcvideoframe.h

The purpose of this CL is to be able to reuse the class WebRtcVideoRenderFrame in webrtcvideoengine.cc.

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32799004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7888 4adac7df-926f-26a2-2b94-8c16560cd09d
Add NEON intrinsics version for min_max_operations_neon.c

WebRtcSpl_MinValueW32Neon, WebRtcSpl_MaxValueW32Neon, WebRtcSpl_MaxValueW16Neon
and WebRtcSpl_MaxAbsValueW32Neon are added. SplTest in common_audio_unittests
is passed on ARM32/ARM64 platforms.

BUG=4002
R=andrew@webrtc.org, jridges@masque.com

Change-Id: Id461d64c3313f56147edadd2231e8845574ead2a

Review URL: https://webrtc-codereview.appspot.com/28259004

Patch from Yang Zhang <yang.zhang@arm.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7889 4adac7df-926f-26a2-2b94-8c16560cd09d
Revert r7886:7887.

Broke build steps in other code that uses securetunnelsessionclient.cc
and others.

TBR=tommi@webrtc.org,pthatcher@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/36439004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7890 4adac7df-926f-26a2-2b94-8c16560cd09d
Add WebRtcIsacfix_FilterMaLoopNeon's intrinsics version.

This intrinsics version gives bit-exact result as the current assembly
neon code. And the performance is 38% better than current assembly
neon version, 5.92 times faster than current C version. The test runs
under Cortex-a53 aarch32 mode, other cpu should give similar performance
result.

BUG=4002
R=andrew@webrtc.org, jridges@masque.com

Change-Id: I257e33ef6d634a519fd71adc4f52b06dd655bd9d

Review URL: https://webrtc-codereview.appspot.com/32749004

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

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7891 4adac7df-926f-26a2-2b94-8c16560cd09d
Revert r7885.

Breaks compile step of other code where network name of
cricket::Candidate is used.

TBR=guoweis@webrtc.org,juberti@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/31229004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7892 4adac7df-926f-26a2-2b94-8c16560cd09d
Merge audio_processing changes.

R=aluebs@webrtc.org, bjornv@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/32769004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7893 4adac7df-926f-26a2-2b94-8c16560cd09d
Add thread asserts to StatsCollector.
Also adding a "ForTest" postfix to a test-only method.

R=perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34489004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7894 4adac7df-926f-26a2-2b94-8c16560cd09d
Remove unneeded ctor and add a more practical one
The default constructor isn't necessary, so I'm removing it.
I'm adding another one so that we can (later) make |type| const.

R=perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36429004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7895 4adac7df-926f-26a2-2b94-8c16560cd09d
(Auto)update libjingle 82121498-> 82126219

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7896 4adac7df-926f-26a2-2b94-8c16560cd09d
Use webrtc_root instead of DEPTH for iSAC.

Un-breaks chromium.webrtc.fyi. Broken as Chromium doesn't have webrtc/
checked out in root.

TBR=bjornv@webrtc.org,tommi@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/28289004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7897 4adac7df-926f-26a2-2b94-8c16560cd09d
Clean up StatsObserver's OnComplete methods (address TODOs).

R=perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29239004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7898 4adac7df-926f-26a2-2b94-8c16560cd09d
Suppressing warnings in GetRTT() in VoE.

GetRTT() was separated from GetRTPStatistics() but the warnings were not updated.

Now GetRTT() is only only used by GetRTPStatistics() and the warning pops up pointlessly and too often.

This CL is to suppress these warnings and maintain a proper warning for GetRTPStatistics().

BUG=
R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36469004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7899 4adac7df-926f-26a2-2b94-8c16560cd09d
resources/audio_processing: Removed unused test files

Two files not used by any tests are removed.

BUG=
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32829004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7900 4adac7df-926f-26a2-2b94-8c16560cd09d
Removes unused test files by audio_processing/transient

BUG=
TBR=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36499004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7901 4adac7df-926f-26a2-2b94-8c16560cd09d
Merge in AGC manager and AGC tools.

R=bjornv@webrtc.org
BUG=4098

Review URL: https://webrtc-codereview.appspot.com/37379004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7902 4adac7df-926f-26a2-2b94-8c16560cd09d
Reenable test case P2PTransportChannelTest.TestIPv6Connections

BUG=3317
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29229004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7903 4adac7df-926f-26a2-2b94-8c16560cd09d
Remove 20ms support in AGC

Today, 10ms is the standard chunk length used in whole AudioProcessing, so this was only adding unnecessary complexity and maintainance.
Removing it doesn't change the bahavior in any use case of today.

R=andrew@webrtc.org, bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28249004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7904 4adac7df-926f-26a2-2b94-8c16560cd09d
Use int64_t for milliseconds more often, primarily for TimeUntilNextProcess.

This fixes a variety of MSVC warnings about value truncations when implicitly
storing the 64-bit values we get back from e.g. TimeTicks in 32-bit objects, and
removes the need for a number of explicit casts.

This also moves a number of constants so they're declared right where they're used, which is easier to read and maintain, and makes some of them of integral type rather than using the "enum hack".

BUG=chromium:81439
TEST=none
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33649004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7905 4adac7df-926f-26a2-2b94-8c16560cd09d
Add adapter_type into Candidate object.

Expose adapter_type from Candidate such that we could add jmidata on top of this.

Created a new type of report just for Ice candidate. The candidate's id is used as part of report identifier. This code change only reports the best connection's local candidate's adapter type. There should be cleaning later to move other candidate's attributes to the new report.

This is migrated from issue 32599004

BUG=
R=juberti@webrtc.org

Committed: https://code.google.com/p/webrtc/source/detail?r=7885

Review URL: https://webrtc-codereview.appspot.com/36379004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7906 4adac7df-926f-26a2-2b94-8c16560cd09d
Fix vp9 setting in vie loopback test.

If vp9 codec was selected then videoCodec.codecSpecific.VP8.numberOfTemporalLayers was being set.

TBR=stefan@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/37389004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7907 4adac7df-926f-26a2-2b94-8c16560cd09d