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chromium-webrtc-autoroll 59e3ec2da0 Roll chromium_revision 9b243167aa..89e5d40511 (904119:904223) 
Change log: 9b243167aa..89e5d40511
Full diff: 9b243167aa..89e5d40511

Changed dependencies
* src/base: fa07be9c3c..c3be4f97ad
* src/build: 66e6c2e275..488aac0b88
* src/testing: 0b41bd3fd2..8c3404999d
* src/third_party: 9965543f35..ed132557b6
* src/third_party/androidx: K8c7JN0QlH9y-iVNRC74RIWTTwcUwtM9CIwQ8vPQyFkC..R0benUT8A1oqvs7L0N-hfCvDpMzfzemd6fXXr4tBOrEC
* src/third_party/perfetto: f7ba153308..cb07e44638
* src/tools: a0a9a4bd0c..5ddfcadec9
DEPS diff: 9b243167aa..89e5d40511/DEPS

No update to Clang.

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

Change-Id: I46ff2b5aab4311d8b00d2fd8e1a2371d11ae6c47
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226683
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@{#34524}
2021-07-22 09:01:04 +00:00
api
Move SetVideoCodecs closer to AddVideoConfig.
2021-07-19 13:01:29 +00:00
audio
Adding packetsDiscarded to RTCReceivedRtpStreamStats.
2021-07-13 20:34:45 +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 (2021-07-22T04:03:23).
2021-07-22 05:57:22 +00:00
common_audio
Fix -Wunreachable-code-aggressive.
2021-06-30 11:14:37 +00:00
common_video
Ignore prefix NAL unit for slice parsing.
2021-07-22 06:25:53 +00:00
data
…
docs
Reference "main" branches instead of "master" branches.
2021-07-15 11:07:44 +00:00
examples
Replace assert() with RTC_DCHECK().
2021-07-09 07:49:43 +00:00
g3doc
Reference "main" branches instead of "master" branches.
2021-07-15 11:07:44 +00:00
logging
Stop parsing if current message length is bigger than buffer.
2021-07-19 11:12:23 +00:00
media
Move helpers to parse base rtp packet fields to rtp_rtcp module
2021-07-19 14:27:27 +00:00
modules
Discard over large DataRates in VideoLayersAllocation rtp header extension
2021-07-21 13:35:14 +00:00
net/dcsctp
dcsctp: Abandon chunks consistently
2021-07-12 22:47:51 +00:00
p2p
Revert "Reland "Port: migrate to TaskQueue.""
2021-07-13 09:07:53 +00:00
pc
Move helpers to parse base rtp packet fields to rtp_rtcp module
2021-07-19 14:27:27 +00:00
resources
Disable high-pass filtering of the AEC reference
2021-02-23 07:06:11 +00:00
rtc_base
Reland "Add WebRTC specific dcheck_always_on."
2021-07-21 13:26:14 +00:00
rtc_tools
Replace assert() with RTC_DCHECK().
2021-07-09 07:49:43 +00:00
sdk
Hide AndroidVideoBuffer class and use factory function
2021-07-16 13:54:45 +00:00
stats
Adding packetsDiscarded to RTCReceivedRtpStreamStats.
2021-07-13 20:34:45 +00:00
system_wrappers
Replace assert() with RTC_DCHECK().
2021-07-09 07:49:43 +00:00
test
Delete helper to parse rtcp packet into rtp header
2021-07-20 11:44:49 +00:00
tools_webrtc
Reland "Add WebRTC specific dcheck_always_on."
2021-07-21 13:26:14 +00:00
video
Set video codecs with PeerConfigurer in tests.
2021-07-15 20:44:41 +00:00
.clang-format
Add IncludeBlocks to clang-format.
2021-02-03 16:29:07 +00:00
.git-blame-ignore-revs
…
.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
Reland "Add WebRTC specific dcheck_always_on."
2021-07-21 13:26:14 +00:00
CODE_OF_CONDUCT.md
Reference "main" branches instead of "master" branches.
2021-07-15 11:07:44 +00:00
codereview.settings
…
DEPS
Roll chromium_revision 9b243167aa..89e5d40511 (904119:904223)
2021-07-22 09:01:04 +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
…
license_template.txt
…
native-api.md
Reference "main" branches instead of "master" branches.
2021-07-15 11:07:44 +00:00
OWNERS
Move style guide and abseil-in-webrtc into g3doc subfolder
2021-05-13 14:43:10 +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
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
Reference "main" branches instead of "master" branches.
2021-07-15 11:07:44 +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
Reland "Add WebRTC specific dcheck_always_on."
2021-07-21 13:26:14 +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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