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chromium-webrtc-autoroll 34f5ab82fe Roll chromium_revision b0398ace4b..2c970cbe2f (806731:806885) 
Change log: b0398ace4b..2c970cbe2f
Full diff: b0398ace4b..2c970cbe2f

Changed dependencies
* src/base: 2bc64ef15e..ea9d15525e
* src/build: 73be302438..9f51c92f5c
* src/ios: 7ecdca2225..ac90fd0782
* src/testing: 2444860bdd..49c4639b9a
* src/third_party: 533fbd1f56..fb6ac2eca4
* src/third_party/depot_tools: a1e15498cd..0547fb2917
* src/third_party/perfetto: bef393a5a3..cfda0b2aaf
* src/third_party/usrsctp/usrsctplib: a8c51df76c..995c0b8441
* src/tools: 81ed796261..084749d9cb
* src/tools/luci-go: git_revision:18502e6a01811b0c2dd2a0250e81bd82e47d2c2d..git_revision:b022173f8069cf8001d4cf2a87ce7c5f0eae220f
* src/tools/luci-go: git_revision:18502e6a01811b0c2dd2a0250e81bd82e47d2c2d..git_revision:b022173f8069cf8001d4cf2a87ce7c5f0eae220f
* src/tools/luci-go: git_revision:18502e6a01811b0c2dd2a0250e81bd82e47d2c2d..git_revision:b022173f8069cf8001d4cf2a87ce7c5f0eae220f
* src/tools/swarming_client: 4c095d0417..44c13d7315
Removed dependency
* src/third_party/android_deps/libs/androidx_legacy_legacy_support_v13
DEPS diff: b0398ace4b..2c970cbe2f/DEPS

No update to Clang.

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

Change-Id: I96ae6acdf60dd2fb802721ba0e4ee6e4cca94562
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/184221
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@{#32102}
2020-09-15 05:11:45 +00:00
api
Removed old OnDecodedFrame callback.
2020-09-14 09:58:21 +00:00
audio
Fix standard GetStats to not modify NetEq state.
2020-09-14 09:51:21 +00:00
build_overrides
set perfetto flag to default value of false
2020-07-22 10:14:53 +00:00
call
Adds ability to delay pacer start until media is added.
2020-09-14 21:42:55 +00:00
common_audio
Refactor cpu_features_wrapper.h functions from C to C++.
2020-09-07 08:44:24 +00:00
common_video
Move FrameCounts and FrameCountObserver to common_video/frame_counts.h
2020-08-27 09:53:18 +00:00
data
Remove old data files.
2018-10-05 14:40:21 +00:00
docs
Adding documentation to become a WebRTC committer.
2020-09-02 07:49:04 +00:00
examples
rename "sdp" to description in a few places
2020-09-11 12:36:54 +00:00
logging
Ensure RtcEventLogEncoderNewFormat::EncodeRemoteEstimate handles infite
2020-08-25 09:22:49 +00:00
media
Fix standard GetStats to not modify NetEq state.
2020-09-14 09:51:21 +00:00
modules
Fix standard GetStats to not modify NetEq state.
2020-09-14 09:51:21 +00:00
p2p
Make StunRequest not use auto cleanup.
2020-09-14 14:46:49 +00:00
pc
Avoid deallocating the async invoker when clearing the transport.
2020-09-14 12:51:12 +00:00
resources
iSAC API wrapper unit test fix
2020-02-27 14:27:23 +00:00
rtc_base
Use PostTask instead of MessageHandler for cert generation.
2020-09-14 14:48:39 +00:00
rtc_tools
Migrate android_resources targets to not create R.java files
2020-09-01 20:29:36 +00:00
sdk
Synchronize modifications to frame_extra_info_ in jni::VideoEncoderWrapper.
2020-09-14 11:15:32 +00:00
stats
Reland "Implement packets_(sent | received) for RTCTransportStats"
2020-07-10 11:50:59 +00:00
style-guide
…
system_wrappers
Expand is_linux to is_linux || is_chromeos.
2020-09-10 17:01:16 +00:00
test
Fix standard GetStats to not modify NetEq state.
2020-09-14 09:51:21 +00:00
tools_webrtc
Reland "Switch from "rtc_ios_xctest_test" to "rtc_test"."
2020-09-10 20:03:34 +00:00
video
Removed old OnDecodedFrame callback.
2020-09-14 09:58:21 +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
Rename PlayoutDelay --> VideoPlayoutDelay, move to api/video/video_timing.h
2020-09-07 08:37:14 +00:00
.vpython
Add 'requests' to .vpython.
2020-09-09 14:36:03 +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
authors: add comments about individual / org sections
2020-09-07 18:33:21 +00:00
BUILD.gn
Expand is_linux to is_linux || is_chromeos.
2020-09-10 17:01:16 +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
DEPS
Roll chromium_revision b0398ace4b..2c970cbe2f (806731:806885)
2020-09-15 05:11:45 +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
…
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
Inclusive language in PRESUBMIT.py.
2020-07-22 10:01:23 +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
Fix PRESUBMIT.py warnings.
2020-09-07 08:34:44 +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
build: move rtc_prefer_fixed_point
2020-09-11 12:38:24 +00:00
whitespace.txt
Trigger CI bots.
2020-09-06 10:13:23 +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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Java 2.9%
C 2.2%
Objective-C++ 2%
Python 1.3%
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