Roll chromium_revision 309cf65..24b4c73

Two VP9 tests needed to be disabled (see webrtc:4059) to make all tests pass.

Relevant changes:
* src/third_party/android_tools: ea50ccc..4c47ef6
* src/third_party/icu: dd72764..866ff69
* src/third_party/libvpx: 2e5ced5..429874c
* src/third_party/nss: 258342e..bb4e75a
* src/third_party/yasm/source/patched-yasm: c960eb1..4671120
* src/tools/gyp: 0a381c0..fe00999
* src/tools/swarming_client: 5b827c9..1d4965c
Details: https://chromium.googlesource.com/chromium/src/+/309cf65..24b4c73/DEPS

Clang version was not updated in this roll.

BUG=4059
TBR=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26289004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7778 4adac7df-926f-26a2-2b94-8c16560cd09d
AudioEncoder subclass for G722

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

Review URL: https://webrtc-codereview.appspot.com/30259004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7779 4adac7df-926f-26a2-2b94-8c16560cd09d
Adding a payload type to AudioEncoder objects

The type is set in the Config struct and is provided in the EncodedInfo
output struct from each Encode() call. The audio_decoder_unittest is
updated to verify correct propagation of the payload type.

BUG=3926
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27299004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7780 4adac7df-926f-26a2-2b94-8c16560cd09d
Change back so that Android ApprtcDemo only use one MediaStream containing both audio and video.
The reason is that the desktop apprtcdemo only handle one MediaStream and this doesn't play audio if it receive two streams.

TEST=Test that a call with audio and video can be setup between an Android device and a desktop client using apprtc.appspot.com.
BUG=4051,3786
R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31069004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7781 4adac7df-926f-26a2-2b94-8c16560cd09d
Clear 2 unused functions in audio processing aecm module.

unused functions:
WebRtcAecm_WindowAndFFTNeon
WebRtcAecm_InverseFFTAndWindowNeon

BUG=3580
R=andrew@webrtc.org

Change-Id: I12c50a8706d40f9ea98208b5733c00ede7b1f435

Review URL: https://webrtc-codereview.appspot.com/30269004

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

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7782 4adac7df-926f-26a2-2b94-8c16560cd09d
Remove -flax-vector-conversions flag for ARM NEON building.

Pass compilation on both ARMv7 and ARM64. The generated
binary (audioproc) is byte to byte (with symbol striped) same as
before. The output of audioproc -aecm is also byte to byte same between
C and NEON version on ARMv7 and ARM64.

Change-Id: Ibdf40fe085f6bad1311f59bf9318bbcf37dd7ce5

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

Review URL: https://webrtc-codereview.appspot.com/30239004

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

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7783 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/trunk@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/trunk@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/trunk@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/trunk@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/trunk@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/trunk@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/trunk@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/trunk@7794 4adac7df-926f-26a2-2b94-8c16560cd09d