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chromium-webrtc-autoroll 43d8e93fa7 Roll chromium_revision 8d9e658d11..2d9b6439f0 (746798:746902) 
Change log: 8d9e658d11..2d9b6439f0
Full diff: 8d9e658d11..2d9b6439f0

Changed dependencies
* src/build: fa4450f206..b04917d42d
* src/ios: 1de797c11d..0e97c075f4
* src/testing: 5a0d4442c4..b36dfa5cdf
* src/third_party: 1937f3afa3..26c37119b0
* src/tools: e710efd3fa..39818018be
DEPS diff: 8d9e658d11..2d9b6439f0/DEPS

No update to Clang.

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

Change-Id: Ie33cc7f9fa47b7ddfa2b72dde07fabf831b8e86a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169760
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@{#30683}
2020-03-04 20:36:42 +00:00
api
VoIP interfaces API enhancement (continuation of 169000)
2020-03-03 18:19:54 +00:00
audio
Only update the current time of a played out frame if a new frame is played out.
2020-02-25 12:48:32 +00:00
build_overrides
Remove crbug.com/904400 workaround.
2019-03-15 18:36:23 +00:00
call
Transform encoded frame in RTPSenderVideo.
2020-03-03 08:17:49 +00:00
common_audio
Optimizations and refactoring of the APM 3-band split filter
2020-02-24 13:19:14 +00:00
common_video
Remove wildcard ownership for build files.
2020-02-19 14:05:46 +00:00
data
Remove old data files.
2018-10-05 14:40:21 +00:00
docs
Change faq.md titles.
2020-02-20 14:24:29 +00:00
examples
Remove wildcard ownership for build files.
2020-02-19 14:05:46 +00:00
logging
Replace DataSize and DataRate factories with newer versions
2020-02-18 16:09:50 +00:00
media
Enable quality scaler for simulcast and SVC if only one stream is active
2020-03-04 15:22:00 +00:00
modules
Enable quality scaler for simulcast and SVC if only one stream is active
2020-03-04 15:22:00 +00:00
p2p
Add jonaso@ to p2p/OWNERS
2020-03-03 14:10:18 +00:00
pc
Unbreak ICE renomination
2020-03-03 13:26:27 +00:00
resources
iSAC API wrapper unit test fix
2020-02-27 14:27:23 +00:00
rtc_base
Makes Thread::Send execute sent messages after pending posted messages.
2020-03-03 12:15:55 +00:00
rtc_tools
Add YUV to IVF video converter util
2020-02-19 14:44:21 +00:00
sdk
Change ownership of encoded data buffer in H264 encoder.
2020-03-04 13:26:26 +00:00
stats
Remove wildcard ownership for build files.
2020-02-19 14:05:46 +00:00
style-guide
…
system_wrappers
Remove wildcard ownership for build files.
2020-02-19 14:05:46 +00:00
test
Simplify initialization of test FrameGeneratorCapturerConfig.
2020-03-03 16:15:08 +00:00
tools_webrtc
Whitespace change to kick bots.
2020-02-28 06:44:59 +00:00
video
Fix video_loopback crash when selecting all streams
2020-03-04 10:25:06 +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
Switch to compiling WebRTC -std=c++14 by default
2019-09-09 19:24:16 +00:00
.vpython
Add source-side perf upload script for WebRTC.
2019-11-18 14:37:01 +00:00
abseil-in-webrtc.md
Fix typo in abseil-in-webrtc.md.
2019-12-18 14:27:34 +00:00
AUTHORS
Reland "Reland "Reland "Reland "Reland "Implemented screen enumeration and selection for desktop capture under X11 using the X Resize and Rotate extension version 1.5."""""
2020-02-26 20:35:54 +00:00
BUILD.gn
Move video_replay under rtc_tools/.
2020-02-07 17:57:30 +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
Reland "Remove PlayoutDelayOracle and make RtpSenderVideo guarantee delivery"
2020-02-07 08:23:58 +00:00
DEPS
Roll chromium_revision 8d9e658d11..2d9b6439f0 (746798:746902)
2020-03-04 20:36:42 +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
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
…
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
Fix public_deps presubmit and gn format fighting each other.
2020-01-30 11:22:46 +00:00
pylintrc
Fixing py lint errors
2018-07-23 15:28:48 +00:00
README.chromium
…
README.md
…
style-guide.md
Add guidance to style guide how to reference a bug in a TODO
2019-12-11 11:55:52 +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
Reformat GN files.
2020-01-21 12:13:11 +00:00
whitespace.txt
Whitespace change
2020-01-28 07:53:15 +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

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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