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chromium-webrtc-autoroll 48f20f849f Roll chromium_revision 31d34f548e..5f083e73a6 (785408:785559) 
Change log: 31d34f548e..5f083e73a6
Full diff: 31d34f548e..5f083e73a6

Changed dependencies
* src/base: f2a563ac13..8aeffef0ee
* src/build: 1a541c1582..729634b87d
* src/buildtools: 6b951673d1..eb3987ec70
* src/buildtools/linux64: git_revision:b6203d186bff6b39ac25af6c1e80e1d3f96c949a..git_revision:d585128cdaf3e6ff7bfd58641965e60c12618eb1
* src/buildtools/mac: git_revision:b6203d186bff6b39ac25af6c1e80e1d3f96c949a..git_revision:d585128cdaf3e6ff7bfd58641965e60c12618eb1
* src/buildtools/win: git_revision:b6203d186bff6b39ac25af6c1e80e1d3f96c949a..git_revision:d585128cdaf3e6ff7bfd58641965e60c12618eb1
* src/ios: 3dfe063f28..d3245aa303
* src/third_party: 38a4b81800..f8c6d72cc0
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/5f3007bf80..9ef36d0c0f
* src/third_party/depot_tools: 49735e227c..81923d6d2c
* src/third_party/freetype/src: a443474755..3bb512bc9f
* src/third_party/gtest-parallel: df0b4e476f..aabba21acd
* src/third_party/perfetto: 1783b3bd55..3039024ba5
* src/third_party/r8: B467c9t23JiW_6XGqhvHvtEKWSkrPS2xG_gho_gbAI4C..vvymFSkKtWKWNmfz0PL_0H8MD8V40P--A9aUfxfpF6QC
* src/tools: ac38d0f35a..ee10cd4775
DEPS diff: 31d34f548e..5f083e73a6/DEPS

No update to Clang.

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

Change-Id: Ic55f8e4c840de1686ef823ddd42f30d19107fde5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178820
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@{#31640}
2020-07-06 22:37:46 +00:00
api
Add simulated time support for PC level test.
2020-07-06 21:18:00 +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
Migrate call/ to webrtc::Mutex.
2020-07-06 15:48:30 +00:00
common_audio
Replace slave -> helper, master -> reference
2020-06-29 12:18:05 +00:00
common_video
In Av1 encoder propagate zero bitrate as inactive decode target
2020-06-25 16:32:28 +00:00
data
…
docs
Search and replace gendered terms according to style guide:
2020-06-12 14:12:54 +00:00
examples
Convert GN libs lists to frameworks
2020-07-06 10:08:09 +00:00
logging
Search and replace gendered terms according to style guide:
2020-06-12 14:12:54 +00:00
media
sdp: parse and serialize non-key=value fmtp lines
2020-07-01 20:06:06 +00:00
modules
Convert GN libs lists to frameworks
2020-07-06 10:08:09 +00:00
p2p
Fix missing dependencies.
2020-06-23 15:46:34 +00:00
pc
Reland "Add rtc::Thread invoke policy."
2020-07-03 11:33:17 +00:00
resources
iSAC API wrapper unit test fix
2020-02-27 14:27:23 +00:00
rtc_base
Inclusive language in //rtc_base.
2020-07-06 10:39:39 +00:00
rtc_tools
Roll chromium_revision 4d95e6c77b..71a0e1904e (776481:782339)
2020-06-26 05:33:14 +00:00
sdk
Convert GN libs lists to frameworks
2020-07-06 10:08:09 +00:00
stats
Correct name of DC.dataChannelIdentifier stats member
2020-06-07 21:57:50 +00:00
style-guide
…
system_wrappers
Use absl_deps in order to preapre to the Abseil component build release.
2020-06-08 12:59:40 +00:00
test
Add simulated time support for PC level test.
2020-07-06 21:18:00 +00:00
tools_webrtc
Rename sanitizers suppression files.
2020-07-03 18:29:58 +00:00
video
[Adaptation] Move TriggerAdaptationDueToFrameDroppedDueToSize
2020-07-06 11:45:30 +00:00
.clang-format
…
.git-blame-ignore-revs
…
.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
Revert "Fix data channel message integrity violation"
2020-07-01 14:45:27 +00:00
BUILD.gn
Convert GN libs lists to frameworks
2020-07-06 10:08:09 +00:00
CODE_OF_CONDUCT.md
…
codereview.settings
…
common_types.h
Replaces OverheadObserver with simple getter.
2020-05-07 17:33:45 +00:00
DEPS
Roll chromium_revision 31d34f548e..5f083e73a6 (785408:785559)
2020-07-06 22:37:46 +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
…
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
Build: Disable the iLBC audio codec by default
2020-07-01 08:52:35 +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
Languages
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