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chromium-webrtc-autoroll b629aee669 Roll chromium_revision 17c6abea92..a5d70b42f2 (899632:899910) 
Change log: 17c6abea92..a5d70b42f2
Full diff: 17c6abea92..a5d70b42f2

Changed dependencies
* src/base: 4a5a81e82b..c0c4bfa63c
* src/build: fff5048571..9d1af1fefb
* src/buildtools/linux64: git_revision:31f2bba8aafa8015ca5761100a21f17c2d741062..git_revision:24e2f7df92641de0351a96096fb2c490b2436bb8
* src/buildtools/mac: git_revision:31f2bba8aafa8015ca5761100a21f17c2d741062..git_revision:24e2f7df92641de0351a96096fb2c490b2436bb8
* src/buildtools/win: git_revision:31f2bba8aafa8015ca5761100a21f17c2d741062..git_revision:24e2f7df92641de0351a96096fb2c490b2436bb8
* src/ios: 6d09c985f3..00e6af5206
* src/testing: d8cbec3370..e1152a2ffc
* src/third_party: 89bb511d77..bdfe12f8e0
* src/third_party/androidx: 29574JKqBbhq5FiO3D4ydclUDICPzLTJGfyNc4k4ldYC..7rK3FRn0Lb5wAO4thkxAj_sMaGdwXTOhMCY4YUPpWrIC
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/2fff900ff7..71adf4f171
* src/third_party/icu: a0718d4f12..b9dfc58bf9
* src/third_party/perfetto: 566975367c..cc178a3f17
* src/third_party/r8: gXyBDv_fM87KnLcxvF5AGV5lwnm-JXIALYH8zrzdoaMC..Nu_mvQJe34CotIXadFlA3w732CJ9EvQGuVs4udcZedAC
* src/tools: 3aa2ead994..813923797e
DEPS diff: 17c6abea92..a5d70b42f2/DEPS

No update to Clang.

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

Change-Id: Ic8d0d7970f4730cf9b9eb5fc39f77e41b2b0ba1f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/225585
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@{#34443}
2021-07-09 09:02:25 +00:00
api
Replace assert() with RTC_DCHECK().
2021-07-09 07:49:43 +00:00
audio
Implement nack_count metric for inbound audio rtp streams.
2021-07-01 10:38:44 +00:00
build_overrides
Roll chromium + fix: blacklist -> ignorelist for sanitizers suppressions
2021-05-27 16:16:01 +00:00
call
Update WebRTC code version (2021-07-09T04:05:24).
2021-07-09 05:54:06 +00:00
common_audio
Fix -Wunreachable-code-aggressive.
2021-06-30 11:14:37 +00:00
common_video
Update BitBuffer methods to style guide
2021-05-18 11:10:27 +00:00
data
Remove old data files.
2018-10-05 14:40:21 +00:00
docs
Create a VideoFrameTrackingId RTP header extension.
2021-03-25 17:25:18 +00:00
examples
Replace assert() with RTC_DCHECK().
2021-07-09 07:49:43 +00:00
g3doc
Move supported_platforms.md under g3doc/.
2021-06-28 12:07:45 +00:00
logging
Replace PacketView by vector of pointers in a wrapper class.
2021-07-02 08:17:32 +00:00
media
Unify helpers IsRtpPacket and IsRtcpPacket
2021-07-06 10:39:00 +00:00
modules
Replace assert() with RTC_DCHECK().
2021-07-09 07:49:43 +00:00
net/dcsctp
Replace assert() with RTC_DCHECK().
2021-07-09 07:49:43 +00:00
p2p
Reland "Port: migrate to TaskQueue."
2021-06-21 22:21:04 +00:00
pc
Implement nack_count metric for inbound audio rtp streams.
2021-07-01 10:38:44 +00:00
resources
Disable high-pass filtering of the AEC reference
2021-02-23 07:06:11 +00:00
rtc_base
Replace assert() with RTC_DCHECK().
2021-07-09 07:49:43 +00:00
rtc_tools
Replace assert() with RTC_DCHECK().
2021-07-09 07:49:43 +00:00
sdk
Remove backwards compatible build targets.
2021-07-08 20:26:49 +00:00
stats
Move WebRTC to non deprecated jsoncpp APIs.
2021-06-28 12:09:14 +00:00
system_wrappers
Replace assert() with RTC_DCHECK().
2021-07-09 07:49:43 +00:00
test
Replace assert() with RTC_DCHECK().
2021-07-09 07:49:43 +00:00
tools_webrtc
Add ability to build XCFramework for iOS
2021-07-06 11:23:00 +00:00
video
Replace assert() with RTC_DCHECK().
2021-07-09 07:49:43 +00:00
.clang-format
Add IncludeBlocks to clang-format.
2021-02-03 16:29:07 +00:00
.git-blame-ignore-revs
Let git-hyper-blame ignore new format cleanup.
2019-07-11 16:18:51 +00:00
.gitignore
Add .cache to .gitignore.
2021-01-20 15:01:07 +00:00
.gn
Increase iOS deployment target from 10 to 12.
2021-07-02 17:02:27 +00:00
.vpython
Update six library version
2021-04-26 16:39:07 +00:00
AUTHORS
Added PeerConnectionObserverJni::OnRemoveTrack()
2021-06-03 19:24:55 +00:00
BUILD.gn
Removing RTC_SUPPORTS_METAL compilation flag. This flag is a holdover from before either macOS or the iOS Simulator supported Metal rendering.
2021-06-23 09:56:20 +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 17c6abea92..a5d70b42f2 (899632:899910)
2021-07-09 09:02:25 +00:00
DIR_METADATA
Move metadata in OWNERS files to DIR_METADATA files.
2021-02-08 19:09:33 +00:00
ENG_REVIEW_OWNERS
Remove kwiberg@webrtc.org from OWNERS files
2020-12-04 15:11:26 +00:00
g3doc.lua
Improve webrtc documentation infra. Preview at:
2021-03-30 10:29:30 +00:00
LICENSE
Moving src/webrtc into src/.
2017-09-15 04:25:06 +00:00
license_template.txt
…
native-api.md
Make the remote_bitrate_estimator build target private
2020-11-26 12:21:22 +00:00
OWNERS
Move style guide and abseil-in-webrtc into g3doc subfolder
2021-05-13 14:43:10 +00:00
PATENTS
Moving src/webrtc into src/.
2017-09-15 04:25:06 +00:00
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
Run pylint on presubmit only for modified python files.
2021-07-02 15:06:56 +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
Move style guide and abseil-in-webrtc into g3doc subfolder
2021-05-13 14:43:10 +00:00
WATCHLISTS
Add hta@ to rtc_base/ and api/ WATCHLISTS.
2021-01-06 09:43:34 +00:00
webrtc_lib_link_test.cc
Deprecate PeerConnectionFactory::CreatePeerConnection
2021-05-10 08:47:48 +00:00
webrtc.gni
iOS: Get WebRTC building for Mac Catalyst
2021-07-06 21:07:59 +00:00
whitespace.txt
Revert "Reland "Trigger postsubmit tests.""
2021-06-28 19:44:42 +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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