platform-external-webrtc

Go to file

chromium-webrtc-autoroll e95fc85cb7 Roll chromium_revision 3c5165bebc..98ef1d6866 (709394:709549)

Change log: 3c5165bebc..98ef1d6866
Full diff: 3c5165bebc..98ef1d6866

Changed dependencies
* src/base: a90d12d9e1..4b2a1bb7bb
* src/build: a193dcc697..72c4ea529a
* src/ios: 596015ccb2..14ebf43f34
* src/testing: 6aa713bd7b..f443db3bd0
* src/third_party: 694eaf9c8e..7234d4b58a
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/a38631cd74..59c3eac206
* src/tools: e4ec472d6e..6ae436b4a1
DEPS diff: 3c5165bebc..98ef1d6866/DEPS

No update to Clang.

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

Change-Id: I94fd70e446ad67c09bdbe8552020ca78e9d4c841
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158329
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@{#29620}

2019-10-25 18:41:28 +00:00

api

Make rtc_json poisonous in WebRTC

2019-10-25 13:46:03 +00:00

audio

Add estimatedPlayoutTimestamp to RTCInboundRTPStreamStats.

2019-10-23 07:46:39 +00:00

build_overrides

Remove crbug.com/904400 workaround.

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

call

Calls OnPacketsAcknowledged on RtpRtcp instead of RTPSender directly.

2019-10-24 12:13:56 +00:00

common_audio

Use source_sets in component builds and static_library in release builds.

2019-10-17 21:17:18 +00:00

common_video

Adding missing RTC_EXPORT for component build.

2019-10-18 09:17:56 +00:00

data

Remove old data files.

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

docs

Fixing some typos.

2019-09-10 10:03:50 +00:00

examples

Fix gtk color-space conversion in peerconnection_client

2019-10-23 11:25:06 +00:00

logging

Move rtc_base/ignore_wundef.h to its own target.

2019-10-19 10:50:36 +00:00

media

Added usrsctp reliablitiy stress test.

2019-10-25 16:51:38 +00:00

modules

Prepares RTPSender for extracting RtpSenderEgress

2019-10-25 14:11:51 +00:00

p2p

Stop pairing local relay candidates with remote mDNS candidates.

2019-10-18 23:36:17 +00:00

Added allow_codec_switching parameter to RTCConfig.

2019-10-25 11:06:31 +00:00

resources

Delete voice_detection() pointer to submodule

2019-10-07 13:06:05 +00:00

rtc_base

Make rtc_json poisonous in WebRTC

2019-10-25 13:46:03 +00:00

rtc_tools

Move rtc_base/ignore_wundef.h to its own target.

2019-10-19 10:50:36 +00:00

sdk

Added allow_codec_switching parameter to RTCConfig.

2019-10-25 11:06:31 +00:00

stats

Add estimatedPlayoutTimestamp to RTCInboundRTPStreamStats.

2019-10-23 07:46:39 +00:00

style-guide

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

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

system_wrappers

Use source_sets in component builds and static_library in release builds.

2019-10-17 21:17:18 +00:00

test

Fix MemoryLogWriter so that it always writes the full data.

2019-10-25 12:07:32 +00:00

tools_webrtc

Use source_sets in component builds and static_library in release builds.

2019-10-17 21:17:18 +00:00

video

change PacketBuffer to return it's result rather that use callback

2019-10-23 16:50:57 +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

Switch to compiling WebRTC -std=c++14 by default

2019-09-09 19:24:16 +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

Update style guide for absl::make_unique.

2019-09-18 06:10:58 +00:00

AUTHORS

Add RTCStatisticsReport.h to WebRTC.framework.

2019-10-21 11:16:10 +00:00

BUILD.gn

Revert "Reland "Define WEBRTC_ENABLE_SYMBOL_EXPORT if is_component_build=true.""

2019-10-25 17:54:32 +00:00

CODE_OF_CONDUCT.md

…

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 3c5165bebc..98ef1d6866 (709394:709549)

2019-10-25 18:41:28 +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 #COMPONENT to WebRTC.

2019-10-08 12:20:39 +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

Use source_sets in component builds and static_library in release builds.

2019-10-17 21:17:18 +00:00

PRESUBMIT.py

Add possibility to skip check_includes presubmit check.

2019-10-22 19:35:31 +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

Update WebRTC's C++ style guide to reflect the switch to C++14.

2019-09-16 11:45:35 +00:00

WATCHLISTS

Add saza to audio watchlists

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

webrtc_lib_link_test.cc

Rewrite the lib link test to just be a binary.

2019-10-18 07:42:20 +00:00

webrtc.gni

Make rtc_json poisonous in WebRTC

2019-10-25 13:46:03 +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%