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webrtc-version-updater cae23d9cc3 Update WebRTC code version (2021-07-24T04:02:31). 
TBR=webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com,mbonadei@webrtc.org

Bug: None
Change-Id: I544fc7e2de2f7828fa5c173117f7b9c389cdb26f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226920
Reviewed-by: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/master@{#34546}
2021-07-24 05:22:12 +00:00
api
Use the underlying type of Java Video Buffer on Java -> C++ Frame Buffer
2021-07-24 01:04:40 +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 (2021-07-24T04:02:31).
2021-07-24 05:22:12 +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
Remove old data files.
2018-10-05 14:40:21 +00:00
docs
Update links to point at main branch
2021-07-22 16:41:26 +00:00
examples
Rename release_aar.py and modify it for validation of aar file.
2021-07-23 09:06:34 +00:00
g3doc
Update links to point at main branch
2021-07-22 16:41:26 +00:00
logging
Update links to point at main branch
2021-07-22 16:41:26 +00:00
media
Move helpers to parse base rtp packet fields to rtp_rtcp module
2021-07-19 14:27:27 +00:00
modules
Use **** code for codec of unknown type
2021-07-23 21:04:59 +00:00
net/dcsctp
Update links to point at main branch
2021-07-22 16:41:26 +00:00
p2p
[sigslot] - Remove signal from StunPort::AddressResolver.
2021-07-22 19:35:56 +00:00
pc
Fix bug where we assume new m= sections will always be bundled.
2021-07-23 22:08:00 +00:00
resources
Disable high-pass filtering of the AEC reference
2021-02-23 07:06:11 +00:00
rtc_base
RepeatingTask: equip with DTrace probes.
2021-07-23 10:36:04 +00:00
rtc_tools
Replace assert() with RTC_DCHECK().
2021-07-09 07:49:43 +00:00
sdk
Use the underlying type of Java Video Buffer on Java -> C++ Frame Buffer
2021-07-24 01:04:40 +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
Update links to point at main branch
2021-07-22 16:41:26 +00:00
tools_webrtc
Ensure release bots run with dchecks off.
2021-07-23 14:57:16 +00:00
video
NackModule2: Rename to NackRequester.
2021-07-23 08:30:33 +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 links to point at main branch
2021-07-22 16:41:26 +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
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 89e5d40511..03a29cf406 (904223:904346)
2021-07-22 17:25:16 +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
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
Moving src/webrtc into src/.
2017-09-15 04:25:06 +00:00
presubmit_test_mocks.py
Add presubmit check to guard against assert() usage.
2021-07-22 17:08:26 +00:00
presubmit_test.py
Add presubmit check to guard against assert() usage.
2021-07-22 17:08:26 +00:00
PRESUBMIT.py
Fix PRESUBMIT.py to not run pylint on deleted files.
2021-07-23 06:45:57 +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: add g3doc sitemap to toplevel readme
2021-07-23 07:55:17 +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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