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chromium-webrtc-autoroll c93bfcfd2f Roll chromium_revision 619b1dc518..b624ecb939 (675093:675206)

Change log: 619b1dc518..b624ecb939
Full diff: 619b1dc518..b624ecb939

Changed dependencies
* src/base: 4c0433c9cd..5c757011f1
* src/build: ebd7d61b05..fda79d2c10
* src/ios: b61081dc76..34b5a9388e
* src/testing: 0fa22669f3..acf5b006e3
* src/third_party: 93cfdd52ab..783a66851b
* src/tools: af10189827..59a1366a63
DEPS diff: 619b1dc518..b624ecb939/DEPS

No update to Clang.

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

Change-Id: I6cc0def7d7a23ede485ca7817f2be6793acf4f9c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144920
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@{#28504}

2019-07-08 12:50:15 +00:00

api

Remove deprecation notice for VideoFrame::ntp_time_ms

2019-07-05 18:37:41 +00:00

audio

Introduce PC level audio quality test.

2019-07-05 11:41:38 +00:00

build_overrides

Remove crbug.com/904400 workaround.

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

call

[getStats] Implement "media-source" audio levels, fixing Chrome bug.

2019-07-04 08:13:45 +00:00

common_audio

Initialize signal processing function pointers statically

2019-06-28 14:20:03 +00:00

common_video

Extend structures to store updated version of the dependency descriptor

2019-06-27 12:10:38 +00:00

crypto

Adding new top-level directory crypto/

2019-03-08 00:35:05 +00:00

data

Remove old data files.

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

examples

Fix platform-specific header dependencies to be more precise

2019-07-08 11:21:30 +00:00

logging

Switch rtc_event_log2rtp_dump to ABSL_FLAG.

2019-07-08 11:40:00 +00:00

media

Remove trial WebRTC-SimulcastMaxLayers and make its behavior default

2019-07-05 14:55:46 +00:00

modules

Remove bwe_rtp_play and add rtp_to_text to the build.

2019-07-08 11:45:20 +00:00

p2p

Stun server should return XOR-MAPPED-ADDRESS/MAPPED-ADDRESS correctly

2019-06-28 19:12:14 +00:00

Reduce kMaxSimulcastStreams to 3

2019-07-04 15:54:36 +00:00

resources

Cleanup of resources from removed remote bitrate estimate test framework.

2019-06-18 10:22:01 +00:00

rtc_base

BalancedDegradationSettings: Add option to configure fps based on codec type.

2019-07-08 11:46:25 +00:00

rtc_tools

Switch unpack_aecdump to ABSL_FLAG.

2019-07-04 14:19:40 +00:00

sdk

Android: Add error callback for GL_OUT_OF_MEMORY in EglRenderer

2019-07-05 13:04:17 +00:00

stats

[getStats] Implement "media-source" audio levels, fixing Chrome bug.

2019-07-04 08:13:45 +00:00

style-guide

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

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

system_wrappers

Sanity-check field trial string at initialization.

2019-06-11 14:11:06 +00:00

test

Fix platform-specific header dependencies to be more precise

2019-07-08 11:21:30 +00:00

tools_webrtc

Remove android_tools deps

2019-07-02 06:48:30 +00:00

video

BalancedDegradationSettings: Add option to configure fps based on codec type.

2019-07-08 11:46:25 +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 format commit.

2018-06-20 09:26:44 +00:00

.gitignore

Add Visual Studio Code project folder to gitignore file.

2019-01-21 18:42:33 +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

Stun server should return XOR-MAPPED-ADDRESS/MAPPED-ADDRESS correctly

2019-06-28 19:12:14 +00:00

BUILD.gn

Remove bwe_rtp_play and add rtp_to_text to the build.

2019-07-08 11:45:20 +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

Move kRtpCsrcSize from common_types.h to rtp_headers.h

2019-05-10 09:43:39 +00:00

DEPS

Roll chromium_revision 619b1dc518..b624ecb939 (675093:675206)

2019-07-08 12:50:15 +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

Convert file objects to strings, before passing to PresubmitNotifyResult

2019-06-26 11:31:21 +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

Remove myself from OWNERS in a few places.

2019-06-10 07:57:46 +00:00

webrtc.gni

Specify min_sdk_version for unittest apks also in GN configs

2019-06-26 07:18:15 +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%