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chromium-webrtc-autoroll 7bb3afac79 Roll chromium_revision 708cc13de4..85af621460 (835045:837534) 
Change log: 708cc13de4..85af621460
Full diff: 708cc13de4..85af621460

Changed dependencies
* src/base: 008c159e0b..ddb8bfdd6a
* src/build: 68a15805c2..7665443183
* src/buildtools: 86a7f72ab1..ea9f1f7375
* src/ios: 4c7c2fe022..7cb640c855
* src/testing: a509699f6d..6e6d8d1651
* src/third_party: 418971e2fb..5b5e61a7d6
* src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/3094902fcd..df75139bec
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/4565794f5e..f600415e92
* src/third_party/depot_tools: 0e323e7bff..d5b38571d3
* src/third_party/ffmpeg: 6d9096c9e3..841aa72c9e
* src/third_party/freetype/src: 505943a6a4..f6be92767d
* src/third_party/icu: 8bb622d674..899e18383f
* src/third_party/libaom/source/libaom: https://aomedia.googlesource.com/aom.git/+log/87c414ed32..43927e4611
* src/third_party/perfetto: c6b219431a..54174aefa8
* src/third_party/r8: GXNupZ4wd3_EsDohjcNOss3iuVb5gcv0rAeHUiH05JkC..DR3nwJggFDcmTDz7P8fJQCtRLO1nxDt26czkOqhtZJ8C
* src/third_party/turbine: O_jNDJ4VdwYKBSDbd2BJ3mknaTFoVkvE7Po8XIiKy8sC.._iPtB_ThhxlMOt2TsYqVppwriEEn0mp-NUNRwDwYLUAC
* src/third_party/turbine/src: 0f2a5024fe..3c31e67ae2
* src/third_party/usrsctp/usrsctplib: 2caaff2b0a..995c0b8441
* src/tools: 0825ae7287..833c245d83
DEPS diff: 708cc13de4..85af621460/DEPS

No update to Clang.

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

Change-Id: I4c1f43c760d2f3fd634e6c4a7600c66662a85d40
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/197920
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Andrey Logvin <landrey@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32846}
2020-12-16 13:40:22 +00:00
api
Add a Clone() method to SessionDescriptionInterface
2020-12-16 08:05:10 +00:00
audio
Using absl::optional for round trip time return type handling.
2020-12-15 08:38:12 +00:00
build_overrides
set perfetto flag to default value of false
2020-07-22 10:14:53 +00:00
call
Add WebRTC code freshness version string.
2020-12-14 16:22:35 +00:00
common_audio
Remove kwiberg@webrtc.org from OWNERS files
2020-12-04 15:11:26 +00:00
common_video
Reland "Rename FATAL() into RTC_FATAL()."
2020-11-18 20:49:08 +00:00
data
…
docs
doc: update working-with-native-branches information
2020-12-16 09:02:46 +00:00
examples
Enforcing return type handling on VoIP API.
2020-12-11 20:38:15 +00:00
logging
Cleanup of bwe_defines.h
2020-11-26 12:26:02 +00:00
media
Surface the number of encoded channels
2020-12-15 16:38:04 +00:00
modules
VideoReceiveStream: eliminate task post in decode path.
2020-12-16 11:25:41 +00:00
p2p
Optimize calls to std::string::find() and friends for a single char.
2020-12-16 09:01:44 +00:00
pc
sdp: ignore legacy sctpmap line
2020-12-16 09:05:01 +00:00
resources
Erle metric: avoid the decreasing of the metric when there is no render activity.
2020-12-09 11:48:11 +00:00
rtc_base
Refresh rtc_base/OWNERS.
2020-12-15 13:59:32 +00:00
rtc_tools
Use a task queue for call interaction in video_replay tool
2020-12-14 15:25:43 +00:00
sdk
Add support for toggling builtin voice processing on iOS
2020-12-02 16:07:01 +00:00
stats
stats: add transportId to codec stats
2020-11-17 12:34:39 +00:00
style-guide
Remove kwiberg@webrtc.org from OWNERS files
2020-12-04 15:11:26 +00:00
system_wrappers
Delete use of RWLockWrapper from SimulatedClock
2020-11-04 08:01:08 +00:00
test
Optimize calls to std::string::find() and friends for a single char.
2020-12-16 09:01:44 +00:00
tools_webrtc
Fix typo in auto-updater.
2020-12-15 12:30:53 +00:00
video
VideoReceiveStream: eliminate task post in decode path.
2020-12-16 11:25:41 +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
Reland "Add protobuf-py2_py3 3.13.0 to .vpython."
2020-11-20 07:52:26 +00:00
abseil-in-webrtc.md
Polish the "Using Abseil in WebRTC" docs
2020-10-16 13:42:00 +00:00
AUTHORS
Fix "control reaches end of non-void function" warnings
2020-10-27 10:22:23 +00:00
BUILD.gn
Add WebRTC code freshness version string.
2020-12-14 16:22:35 +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 708cc13de4..85af621460 (835045:837534)
2020-12-16 13:40:22 +00:00
ENG_REVIEW_OWNERS
Remove kwiberg@webrtc.org from OWNERS files
2020-12-04 15:11:26 +00:00
LICENSE
…
license_template.txt
…
native-api.md
Make the remote_bitrate_estimator build target private
2020-11-26 12:21:22 +00:00
OWNERS
Remove kwiberg@webrtc.org from OWNERS files
2020-12-04 15:11:26 +00:00
PATENTS
…
presubmit_test_mocks.py
Reformat python files checked by pylint (part 1/2).
2020-10-30 10:13:11 +00:00
presubmit_test.py
Reformat python files checked by pylint (part 1/2).
2020-10-30 10:13:11 +00:00
PRESUBMIT.py
Fix Authors Check presubmit.
2020-12-15 11:59:03 +00:00
pylintrc
Undo enforcing of PEP-8 pylint changes for method and function names.
2020-11-10 18:26:25 +00:00
README.chromium
Add CPEPrefix.
2020-07-13 11:42:07 +00:00
README.md
doc: move bug reporting instructions to the repository
2020-10-21 14:47:49 +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
Get rid of recently added rtc_openssl_is_boringssl build argument.
2020-12-15 08:34:42 +00:00
whitespace.txt
Trigger bots for FYI waterfall.
2020-11-20 09:35:54 +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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