platform-external-webrtc

Go to file

chromium-webrtc-autoroll ebdf9f8163 Roll chromium_revision 097ffaa18d..6f0434662d (673789:673926)

Change log: 097ffaa18d..6f0434662d
Full diff: 097ffaa18d..6f0434662d

Changed dependencies
* src/base: 5413464fb7..3d13a7aa98
* src/build: b1f7a47cfa..3fa1f27670
* src/ios: 07244c207c..dfb6071255
* src/testing: 971f346545..45f0246c12
* src/third_party: ad26420900..13250e00d5
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/0384e1a460..15ad0b0b62
* src/tools: 3e04aeaa49..a27a29d288
DEPS diff: 097ffaa18d..6f0434662d/DEPS

No update to Clang.

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

Change-Id: I05b6a5cae838906bd6e7fdcd6792cd77af96fc13
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144265
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@{#28438}

2019-07-02 02:55:51 +00:00

api

Add overload to CreateIceTransport that takes additional dependencies

2019-07-01 21:37:54 +00:00

audio

Add plumbing of RtpPacketInfos to each AudioFrame as input for SourceTracker.

2019-07-01 15:56:40 +00:00

build_overrides

Remove crbug.com/904400 workaround.

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

call

Implement RtpVideoSender::SetFecAllowed()

2019-06-28 18:24:16 +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

Delete RtcEventLogFactory factory as now unused

2019-06-27 10:03:22 +00:00

logging

Delete RtcEventLogFactory factory as now unused

2019-06-27 10:03:22 +00:00

media

Forced vp8 sw encoder fallback: only use min bitrate config if codec type is vp8.

2019-07-01 13:32:35 +00:00

modules

Add plumbing of RtpPacketInfos to each AudioFrame as input for SourceTracker.

2019-07-01 15:56:40 +00:00

p2p

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

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

Add overload to CreateIceTransport that takes additional dependencies

2019-07-01 21:37:54 +00:00

resources

Cleanup of resources from removed remote bitrate estimate test framework.

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

rtc_base

RateControlSettings: add option to set min pixels per frame for libvpx vp8.

2019-06-28 10:54:23 +00:00

rtc_tools

Remove Simple Command Line Parser.

2019-06-27 10:22:49 +00:00

sdk

Use Default instead of GlobalTaskQueueFactory to create AudioDeviceBuffer for ios

2019-07-01 11:20:27 +00:00

stats

Add RTCStats for keyFramesEncoded, keyFramesDecoded.

2019-06-27 14:59:47 +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

Report freeze_time_ms from PC test framework

2019-07-01 12:41:20 +00:00

tools_webrtc

Roll chromium_revision 6ae0f0cd4c..bf62d746a4 (669703:669828) + fix AndroidManifest

2019-06-18 17:10:06 +00:00

video

Forced vp8 sw encoder fallback: only use min bitrate config if codec type is vp8.

2019-07-01 13:32:35 +00:00

.clang-format

…

.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

Delete RtcEventLogFactory factory as now unused

2019-06-27 10:03:22 +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 097ffaa18d..6f0434662d (673789:673926)

2019-07-02 02:55:51 +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++ 88.6%

C 3.3%

Java 3%

Objective-C++ 1.9%

Python 1.9%

Other 1%