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chromium-webrtc-autoroll 6e4416cd89 Roll chromium_revision f629a87d19..432e660d6d (787611:787714) 
Change log: f629a87d19..432e660d6d
Full diff: f629a87d19..432e660d6d

Changed dependencies
* src/base: abd5405c28..f0eb30876d
* src/build: 9d828da174..67bfb6e074
* src/ios: 189676fa6d..7b3dd6953a
* src/third_party: 05a05f9845..0cbb97ee80
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/88e5b1f40c..570934c3da
* src/third_party/perfetto: 6ef441ecad..7d22cca0b4
* src/tools: 84642d7d96..d1d0f247b2
DEPS diff: f629a87d19..432e660d6d/DEPS

No update to Clang.

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

Change-Id: Ifa6015f45820588096a09b7031b9a0109b978f3d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179200
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@{#31718}
2020-07-13 14:36:26 +00:00
api
Delete PeerConnectionInterface::BitrateParameters
2020-07-13 10:06:42 +00:00
audio
webrtc::AudioSendStream: Add lock annotation to audio_level_
2020-07-06 17:05:25 +00:00
build_overrides
Rename sanitizers suppression files.
2020-07-03 18:29:58 +00:00
call
[Adaptation] Move AdaptationConstraints to VideoStreamAdapter
2020-07-09 13:06:56 +00:00
common_audio
Replace slave -> helper, master -> reference
2020-06-29 12:18:05 +00:00
common_video
Migrate common_video/ and examples/ to webrtc::Mutex.
2020-07-07 13:33:27 +00:00
data
Remove old data files.
2018-10-05 14:40:21 +00:00
docs
Search and replace gendered terms according to style guide:
2020-06-12 14:12:54 +00:00
examples
Add missing dependencies to java targets.
2020-07-10 09:17:00 +00:00
logging
Search and replace gendered terms according to style guide:
2020-06-12 14:12:54 +00:00
media
Migrate leftovers in media/ and modules/ to webrtc::Mutex.
2020-07-10 08:27:45 +00:00
modules
Reland "Migrate modules/desktop_capture and modules/video_capture to webrtc::Mutex."
2020-07-13 12:22:08 +00:00
p2p
Reland "Implement packets_(sent | received) for RTCTransportStats"
2020-07-10 11:50:59 +00:00
pc
Delete PeerConnectionInterface::BitrateParameters
2020-07-13 10:06:42 +00:00
resources
iSAC API wrapper unit test fix
2020-02-27 14:27:23 +00:00
rtc_base
Migrate stray leftovers from rtc_base/ and test/ to webrtc::Mutex.
2020-07-13 13:14:34 +00:00
rtc_tools
Migrate rtc_tools/, sdk/, and system_wrappers/ to webrtc::Mutex.
2020-07-10 08:02:05 +00:00
sdk
Delete PeerConnectionInterface::BitrateParameters
2020-07-13 10:06:42 +00:00
stats
Reland "Implement packets_(sent | received) for RTCTransportStats"
2020-07-10 11:50:59 +00:00
style-guide
…
system_wrappers
Migrate rtc_tools/, sdk/, and system_wrappers/ to webrtc::Mutex.
2020-07-10 08:02:05 +00:00
test
Migrate stray leftovers from rtc_base/ and test/ to webrtc::Mutex.
2020-07-13 13:14:34 +00:00
tools_webrtc
Auto roller: send trooper notifications only when Commit-Queue+2.
2020-07-13 09:08:48 +00:00
video
Migrate VideoStreamDecoderImpl to webrtc::Mutex.
2020-07-10 11:02:28 +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
Add .clangd to .gitignore
2019-10-28 12:27:50 +00:00
.gn
Reenable libaom decoder by default
2020-03-18 18:04:41 +00:00
.vpython
Add source-side perf upload script for WebRTC.
2019-11-18 14:37:01 +00:00
abseil-in-webrtc.md
Use absl_deps in order to preapre to the Abseil component build release.
2020-06-08 12:59:40 +00:00
AUTHORS
Relanding: Fix data channel message integrity violation
2020-07-07 03:06:24 +00:00
BUILD.gn
Convert GN libs lists to frameworks
2020-07-06 10:08:09 +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
Replaces OverheadObserver with simple getter.
2020-05-07 17:33:45 +00:00
DEPS
Roll chromium_revision f629a87d19..432e660d6d (787611:787714)
2020-07-13 14:36:26 +00:00
ENG_REVIEW_OWNERS
…
LICENSE
…
license_template.txt
…
native-api.md
Make transient suppression optionally excludable via defines
2020-04-02 11:44:07 +00:00
OWNERS
Remove phoglund as root owner.
2020-03-30 12:15:56 +00:00
PATENTS
…
presubmit_test_mocks.py
…
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
Inclusive language in PRESUBMIT.py.
2020-06-30 14:03:10 +00:00
pylintrc
…
README.chromium
Add CPEPrefix.
2020-07-13 11:42:07 +00:00
README.md
Fix link in documentation. (take 2)
2020-04-16 11:08:43 +00:00
style-guide.md
C++ style: We don't allow designated initializers
2020-06-03 09:11:09 +00:00
WATCHLISTS
Remove benwright@webrtc.org from WATCHLISTS
2020-01-31 18:46:52 +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
Add target_sdk_version to rtc_test.
2020-07-11 08:25:04 +00:00
whitespace.txt
Trigger bots.
2020-06-15 08:40:44 +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 here 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

Description
The idea is to make CMake build for WebRTC m130 version - for audio processing module
Readme BSD-3-Clause 446 MiB
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