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chromium-webrtc-autoroll f6ec68dd62 Roll chromium_revision 7e5c36432b..675968a8c6 (693630:693954)

Change log: 7e5c36432b..675968a8c6
Full diff: 7e5c36432b..675968a8c6

Changed dependencies
* src/base: ebf82dc7c4..cb4ee08593
* src/build: 7f76a96f7e..ecdf3752c2
* src/ios: 0f9030c1a8..0b8db51e2d
* src/testing: 809946d2ac..34b894db4c
* src/third_party: d897faa4f8..51f51a08b9
* src/third_party/android_ndk: https://chromium.googlesource.com/android_ndk.git/+log/62582753e8..89e8db0cdf
* src/third_party/freetype/src: 3fa35aa420..3de1b8d0b0
* src/third_party/libjpeg_turbo: d460d6b1cb..baa5dc2425
* src/tools: 7be347cdad..987faea9da
DEPS diff: 7e5c36432b..675968a8c6/DEPS

Clang version changed 8288453f6aac05080b751b680455349e09d49825:6964027315f70c6817217d8dba0368fd3a274ba3
Details: 7e5c36432b..675968a8c6/tools/clang/scripts/update.py

TBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com,
BUG=None

Change-Id: I35f91ae3b1171f0b7e3840426432048cdf1e3754
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151721
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/master@{#29084}

2019-09-05 22:43:26 +00:00

api

Delete deprecated rtc_event_log_factory_interface.h

2019-09-05 08:57:36 +00:00

audio

Deprecate SingleThreadedTaskQueueForTesting class.

2019-09-03 10:31:30 +00:00

build_overrides

Remove crbug.com/904400 workaround.

2019-03-15 18:36:23 +00:00

call

Deprecate SingleThreadedTaskQueueForTesting class.

2019-09-03 10:31:30 +00:00

common_audio

Remove rtc_use_lto GN arg.

2019-08-20 14:00:49 +00:00

common_video

Add new FrameRateEstimator utility class for more precis FPS estimation.

2019-08-14 12:15:06 +00:00

data

Remove old data files.

2018-10-05 14:40:21 +00:00

examples

Make the RtpHeaderParserImpl available to tests and tools only.

2019-08-29 15:56:40 +00:00

logging

Delete deprecated rtc_event_log_factory_interface.h

2019-09-05 08:57:36 +00:00

media

Buffers non atomic message send with usrsctp lib.

2019-09-03 16:30:21 +00:00

modules

Implemented screen enumeration and selection for desktop capture under X11 using the X Resize and Rotate entension version 1.5.

2019-09-05 21:19:41 +00:00

p2p

Use the sanitized pair when surfacing the candidate pair change event.

2019-09-03 17:17:49 +00:00

Delete old version of PeerConnection::SetConfiguration

2019-09-04 08:23:18 +00:00

resources

Use the AEC3 high-pass filter for the whole APM

2019-08-23 20:04:10 +00:00

rtc_base

Add support for stable bitrate target in SvcRateAllocator

2019-09-04 14:22:43 +00:00

rtc_tools

Revert "Always create output_dir in setup_apprtc.py."

2019-09-04 18:19:12 +00:00

sdk

Add support of AudioRecord.Builder in the ADM for Android

2019-09-05 07:59:30 +00:00

stats

Add support for RTCTransportStats.selectedCandidatePairChanges

2019-08-28 13:22:08 +00:00

style-guide

Add style guide rule about paired .h and .cc files

2018-03-14 13:02:35 +00:00

system_wrappers

Add helper functions to convert between integer milliseconds and fixed-point seconds.

2019-08-16 14:49:46 +00:00

test

Refactor video analyzer injection helper

2019-09-05 14:59:06 +00:00

tools_webrtc

Add license for android_ndk

2019-08-28 14:57:28 +00:00

video

Add support for stable bitrate target in SvcRateAllocator

2019-09-04 14:22:43 +00:00

.clang-format

Tune ObjC clang-format configuration

2017-05-11 09:14:18 +00:00

.git-blame-ignore-revs

Let git-hyper-blame ignore new format cleanup.

2019-07-11 16:18:51 +00:00

.gitignore

Change apprtc_webrtc_browsertest resource dir to avoid MAX_PATH.

2019-09-04 18:49:28 +00:00

.gn

Remove last mention of ortc from the codebase.

2019-05-25 07:28:05 +00:00

.vpython

Add vpython dependencies needed to run presubmit tests on LUCI

2018-05-18 08:10:25 +00:00

abseil-in-webrtc.md

Allowing buffering a LNTF (loss notification) feedback message in RTCPSender

2019-06-03 16:28:34 +00:00

AUTHORS

Implemented screen enumeration and selection for desktop capture under X11 using the X Resize and Rotate entension version 1.5.

2019-09-05 21:19:41 +00:00

BUILD.gn

Make the RtpHeaderParserImpl available to tests and tools only.

2019-08-29 15:56:40 +00:00

CODE_OF_CONDUCT.md

Add code of conduct to WebRTC repo

2017-05-16 12:09:13 +00:00

codereview.settings

Don't add webrtc-reviews@ to CC, it can be added globally on Gerrit

2018-10-25 08:19:53 +00:00

common_types.h

Format almost everything.

2019-07-08 13:45:15 +00:00

DEPS

Roll chromium_revision 7e5c36432b..675968a8c6 (693630:693954)

2019-09-05 22:43:26 +00:00

ENG_REVIEW_OWNERS

Enforce LGTM from owners of depends-on paths in DEPS via presubmit.

2018-09-28 12:49:54 +00:00

LICENSE

Moving src/webrtc into src/.

2017-09-15 04:25:06 +00:00

license_template.txt

…

native-api.md

Delete unused I420 "codec"

2018-12-18 12:30:58 +00:00

OWNERS

Add juberti@ to webrtc root owners

2019-05-17 18:11:58 +00:00

PATENTS

Moving src/webrtc into src/.

2017-09-15 04:25:06 +00:00

presubmit_test_mocks.py

Reland: Add presubmit check for changes in 3pp

2018-05-22 13:11:18 +00:00

presubmit_test.py

Fixing py lint errors

2018-07-23 15:28:48 +00:00

PRESUBMIT.py

Disable RunPythonTests on rtc_tools.

2019-08-06 12:48:33 +00:00

pylintrc

Fixing py lint errors

2018-07-23 15:28:48 +00:00

README.chromium

Moving src/webrtc into src/.

2017-09-15 04:25:06 +00:00

README.md

Tell users where they can find the native API headers

2017-11-14 10:36:46 +00:00

style-guide.md

Remove rule that discourages passing optional by const reference

2019-02-05 11:58:05 +00:00

WATCHLISTS

Add saza to audio watchlists

2019-09-03 14:55:43 +00:00

webrtc.gni

Remove rtc_use_lto GN arg.

2019-08-20 14:00:49 +00:00

whitespace.txt

Whitespace change

2018-02-23 10:34:16 +00:00

README.md

WebRTC is a free, open software project that provides browsers and mobile applications with Real-Time Communications (RTC) capabilities via simple APIs. The WebRTC components have been optimized to best serve this purpose.

Our mission: To enable rich, high-quality RTC applications to be developed for the browser, mobile platforms, and IoT devices, and allow them all to communicate via a common set of protocols.

The WebRTC initiative is a project supported by Google, Mozilla and Opera, amongst others.

Development

See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.

Authoritative list of directories that contain the native API header files.

More info

Official web site: http://www.webrtc.org
Master source code repo: https://webrtc.googlesource.com/src
Samples and reference apps: https://github.com/webrtc
Mailing list: http://groups.google.com/group/discuss-webrtc
Continuous build: http://build.chromium.org/p/client.webrtc
Coding style guide
Code of conduct

Languages

C++ 90.3%

Java 2.9%

C 2.2%

Objective-C++ 2%

Python 1.3%

Other 1%