Move isolate path into webrtc/build/android/test_runner.py

This will make it easier to execute tests and allows
for more cleanup in the buildbot recipes.
Now tests can be listed using:
webrtc/build/android/test_runner.py gtest --help
and executed like
webrtc/build/android/test_runner.py gtest -s audio_decoder_unittests

TESTED=
Ran:
webrtc/build/android/test_runner.py gtest --help
and verified the tests were listed.
I wiped /sdcard/resources on my device, executed:
webrtc/build/android/test_runner.py gtest -s audio_decoder_unittests
and verified it passed and that resources/audio_coding/testfile32kHz.pcm
was copied to the device.

BUG=
R=phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33619004
Revert "Revert 7826 "Change Android PeerConnectionUnittest to build usin...""

Original cl description:

Change Android PeerConnectionUnittest to build using Chrome macros.
The purpose is to be able to run the tests using Chromes buildbots. To run:
CHECKOUT_SOURCE_ROOT=`pwd` build/android/test_runner.py instrumentation --test-apk=libjingle_peerconnection_android_unittest

This also add a new build target to build java PeerConnection using Chromes build macros.

BUG=4031
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26349004
Remove old_factory from WebRtcVideoEngine.

Minor pending cleanup.

R=pthatcher@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/28239004
Wire up RTT statistics to webrtc::Call.

R=mflodman@webrtc.org, stefan@webrtc.org
BUG=1667,1788

Review URL: https://webrtc-codereview.appspot.com/32249004
Merge AEC changes.

R=bjornv@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/34459004
Adding DTX to WebRTC Opus wrapper (relanding).

This is relanding of r7846, which failed since the unit test depended on whether Opus is in fixed-point or float-point.

See the review of r7846 here:
https://webrtc-codereview.appspot.com/13219004/

Patch set 1 is the same as r7846. Further fixes are found in patch set 2 and later.

BUG=
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32299004
Move the downmixing out of AudioBuffer

This provides more flexibility if some component in AudioProcessing wants to operate before downmixing.
Now the AudioProcessing does only track the processing rate, but not the processing number of channels. This is tracked by the AudioBuffer itself and can be changed at any time to one smaller or equal the input number of channels. For each chunk it is reset to input number of channels and the end it should be equal to the output number of channels.

R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28169004
Cleanup little things found when refactoring.

R=juberti@google.com

Review URL: https://webrtc-codereview.appspot.com/33519004
Removing manual test pages because they have been moved to github.

BUG=https://github.com/GoogleChrome/webrtc/issues/203
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/35499004
Fix webrtc gn windows build.

R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32779004
Moving encoded_bytes into EncodedInfo

BUG=3926
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/35469004
Switch kStatsValueName* constants to be enums instead of char*.
This is to guard against potentially assigning a value name to an incorrect value, non-static string or otherwise assume they can be treated as strings.

R=perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26359004
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

Review URL: https://webrtc-codereview.appspot.com/36379004
Move the obvious/easy Jingle-specific code into webrtc/libjingle.

R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32669004
Put pseudotcp back because remoting uses it.

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

Review URL: https://webrtc-codereview.appspot.com/36409004
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
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>.
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
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>.
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