charujain 474accebdb Roll chromium_revision d323a482ee..7114a66134 (494468:497367) manually ...

* Enable workaround for building Android with C++14.
* Disable build hooks for WebRTC on Android.

Change log: d323a482ee..7114a66134
Full diff: d323a482ee..7114a66134

Changed dependencies:
* src/base: 66d3c08e82..0b41aca75a
* src/build: 2a5e6515a5..a2a451d3c0
* src/buildtools: ceb050498e..5af0a3a8b8
* src/ios: 2085f316c1..0a0ac63bd3
* src/testing: 3127a16731..8a25f55d8b
* src/third_party: b4122f732f..8c0e65fa05
* src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/f4ecc84644..ae9f0616c5
* src/third_party/catapult: d27175a044..f8acf3b642
* src/third_party/depot_tools: 6d0d04458d..e081cbe5aa
* src/third_party/icu: 98218d1e92..08cb956852
* src/third_party/libvpx/source/libvpx: cbb83ba4aa..6b9c691daf
* src/tools: 5730fb2d3f..3e167a7bd3
DEPS diff: d323a482ee..7114a66134/DEPS

No update to Clang.

TBR=marpan@webrtc.org,
BUG=webrtc:8148
CQ_INCLUDE_TRYBOTS=master.internal.tryserver.corp.webrtc:linux_internal

Review-Url: https://codereview.webrtc.org/3003393002
Cr-Commit-Position: refs/heads/master@{#19517}

2017-08-25 13:21:52 +00:00

..

audio_coding

Reimplement the builtin audio codec factories using the new stuff in api/

2017-08-25 10:10:50 +00:00

audio_conference_mixer

Remove remains of webrtc/base

2017-07-19 17:40:47 +00:00

audio_device

Roll chromium_revision d323a482ee..7114a66134 (494468:497367) manually

2017-08-25 13:21:52 +00:00

audio_mixer

Remove remains of webrtc/base

2017-07-19 17:40:47 +00:00

audio_processing

Utilizing the AEC3 config struct for constants.

2017-08-25 05:36:53 +00:00

bitrate_controller

Final version of BBR, with tweaks made for WebRTC, major changes:

2017-08-20 16:19:58 +00:00

congestion_controller

Use small BWE period when there is a true network degradation.

2017-08-23 17:22:28 +00:00

desktop_capture

Add DesktopFrame::MoveFrameInfoFrom() and DesktopFrame::CopyFrameInfoFrom()

2017-08-24 23:44:22 +00:00

include

Add AudioFrame::ResetWithoutMuting() to address performance regression.

2017-07-08 23:36:26 +00:00

media_file

Replace WEBRTC_TRACE logging in modules/media_file/

2017-08-23 17:30:38 +00:00

pacing

Move PacedSender ownership to RtpTransportControllerSend.

2017-08-22 14:16:49 +00:00

remote_bitrate_estimator

Use small BWE period when there is a true network degradation.

2017-08-23 17:22:28 +00:00

rtp_rtcp

Add audio_level member to RtpSource and set it from RtpReceiverImpl::IncomingRtpPacket.

2017-08-24 21:52:17 +00:00

utility

Tracking mock_process_thread with a GN target

2017-08-03 12:57:11 +00:00

video_capture

Replace old logging mechanism in webrtc/modules/video_capture/linux

2017-08-16 13:51:27 +00:00

video_coding

Change frame bookkeeping in VideoProcessorIntegrationTest.

2017-08-25 11:51:54 +00:00

video_processing

Remove remains of webrtc/base

2017-07-19 17:40:47 +00:00

BUILD.gn

Remove remains of webrtc/base

2017-07-19 17:40:47 +00:00

module_common_types_unittest.cc

Add mute state field to AudioFrame and switch some callers to use it. Also make AudioFrame::data_ private and instead provide:

2017-06-12 19:45:32 +00:00

OWNERS

Remove all references to GYP

2016-11-16 19:11:38 +00:00