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chromium-webrtc-autoroll a0e6ded9fc Roll chromium_revision 75cf3925c2..e25e764221 (694706:694813)

Change log: 75cf3925c2..e25e764221
Full diff: 75cf3925c2..e25e764221

Changed dependencies
* src/base: dc03aaff06..2de3a55fc4
* src/build: 6ff11c8756..ccaf07df5c
* src/ios: 2dd8278d55..fe8022e34d
* src/testing: 52393b4916..5b7605a491
* src/third_party: 65cc3d6478..13752878f8
* src/tools: bf23d39327..3a469b7fc1
DEPS diff: 75cf3925c2..e25e764221/DEPS

No update to Clang.

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

Change-Id: I0ac5a6ee823d994163c0b7385ed761ef75b1e7a5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152119
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@{#29118}

2019-09-09 18:31:40 +00:00

api

Add qualityLimitationResolutionChanges stat

2019-09-09 15:22:57 +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

Add option to enable retransmission for all temporal layers in the constructor for rtp_sender_video.

2019-09-09 15:39:23 +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

Adds setAudio[Track/Record]StateCallback interfaces to the Java ADM

2019-09-09 08:10:41 +00:00

logging

Delete deprecated rtc_event_log_factory_interface.h

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

media

Add qualityLimitationResolutionChanges stat

2019-09-09 15:22:57 +00:00

modules

Improve performance of RtpPacketHistory

2019-09-09 16:31:53 +00:00

p2p

Use obfuscated IPs in logging in p2p/ and pc/.

2019-09-06 21:51:56 +00:00

Add qualityLimitationResolutionChanges stat

2019-09-09 15:22:57 +00:00

resources

Use the AEC3 high-pass filter for the whole APM

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

rtc_base

Implement stable rate support in SimulcastRateAllocator

2019-09-06 11:17:30 +00:00

rtc_tools

Revert "Always create output_dir in setup_apprtc.py."

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

sdk

Adds setAudio[Track/Record]StateCallback interfaces to the Java ADM

2019-09-09 08:10:41 +00:00

stats

Add qualityLimitationResolutionChanges stat

2019-09-09 15:22:57 +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

Reland "Refactor FEC code to use COW buffers"

2019-09-09 16:20:33 +00:00

tools_webrtc

Add license for android_ndk

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

video

Add qualityLimitationResolutionChanges stat

2019-09-09 15:22:57 +00:00

.clang-format

…

.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

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

2019-09-06 05:36:23 +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 75cf3925c2..e25e764221 (694706:694813)

2019-09-09 18:31:40 +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++ 88.6%

C 3.3%

Java 3%

Objective-C++ 1.9%

Python 1.9%

Other 1%