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Java 2.9%
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chromium-webrtc-autoroll 1ff491b0bb Roll chromium_revision 8907aace7e..e9261a56ad (891631:892013) 
Change log: 8907aace7e..e9261a56ad
Full diff: 8907aace7e..e9261a56ad

Changed dependencies
* src/base: bb0fa97f05..54fedec3aa
* src/build: eb4ebdf32c..b59724c60a
* src/buildtools: c1cef4e23c..be7dcbc361
* src/buildtools/linux64: git_revision:695504d72a30e0b58705b2a1a23964ebf7bca030..git_revision:e0c476ffc83dc10897cb90b45c03ae2539352c5c
* src/buildtools/mac: git_revision:695504d72a30e0b58705b2a1a23964ebf7bca030..git_revision:e0c476ffc83dc10897cb90b45c03ae2539352c5c
* src/buildtools/third_party/libunwind/trunk: 950faeeabc..7e85c7a26b
* src/buildtools/win: git_revision:695504d72a30e0b58705b2a1a23964ebf7bca030..git_revision:e0c476ffc83dc10897cb90b45c03ae2539352c5c
* src/ios: 54a3283c4d..9bf9e6fb50
* src/testing: 3d0002de69..3c85e0a05c
* src/third_party: 00cd360e11..9c4fcf3744
* src/third_party/depot_tools: e5d455cca7..ef579a1192
* src/third_party/googletest/src: aa533abfd4..e2239ee604
* src/third_party/perfetto: adce539a59..9834c0b426
* src/tools: e9eaed516e..24587d48b2
DEPS diff: 8907aace7e..e9261a56ad/DEPS

Clang version changed llvmorg-13-init-12491-g055770d5:llvmorg-13-init-12576-g643b6407
Details: 8907aace7e..e9261a56ad/tools/clang/scripts/update.py

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

Change-Id: Ica6bd348a6de08ea3282b43e306e82cd1a303c57
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222363
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@{#34278}
2021-06-14 11:23:56 +00:00
api count webrtc pranswer usage 2021-06-11 12:59:37 +00:00
audio Report stats from ChannelReceive::GetAudioFrameWithInfo at 1Hz. 2021-06-09 18:41:47 +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 (2021-06-13T04:02:18). 2021-06-13 04:55:38 +00:00
common_audio Avoid undefined behavior in a division operation. 2021-04-23 07:49:24 +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 Added PeerConnectionObserverJni::OnRemoveTrack() 2021-06-03 19:24:55 +00:00
g3doc doc: document rtp payload type mapping behaviour 2021-06-04 06:23:32 +00:00
logging Add documentation for RTC event log 2021-06-03 09:03:18 +00:00
media Add small cooldown to unsignalled ssrc stream creation. 2021-06-10 13:13:21 +00:00
modules AEC3: Unbounded echo spectrum for dominant nearend detection. 2021-06-11 13:30:00 +00:00
net/dcsctp dcsctp: Prevent overflow of missing parameters 2021-06-09 14:12:53 +00:00
p2p Avoid generating a random id for candidate stats. 2021-06-08 08:00:01 +00:00
pc Make JsepTransportCollection self-managing for transports 2021-06-11 16:12:11 +00:00
resources Disable high-pass filtering of the AEC reference 2021-02-23 07:06:11 +00:00
rtc_base Make stopping of the RepeatingTask safer 2021-06-11 21:54:54 +00:00
rtc_tools Update the only 3 remaining kFilterBilinear to kFilterBox. 2021-06-08 13:19:23 +00:00
sdk Don't register invalid encode complete callbacks. 2021-06-14 10:45:46 +00:00
stats Populate qualityLimitationDurations stats for outbound RTP streams 2021-05-31 21:39:37 +00:00
system_wrappers Make Clock::ConvertTimestampToNtpTime pure virtual 2021-05-21 09:55:14 +00:00
test In vp9 encoder fuzzer reduce information stored for older frames 2021-06-11 15:46:00 +00:00
tools_webrtc Add MB configs for M1 bots 2021-06-09 19:02:28 +00:00
video Avoid video stream allocation on configuration change after timeout. 2021-06-14 07:27:45 +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 Switch from check_targets to no_check_targets in .gn 2021-05-20 10:42:21 +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 Switch from check_targets to no_check_targets in .gn 2021-05-20 10:42:21 +00:00
CODE_OF_CONDUCT.md Add code of conduct to WebRTC repo 2017-05-16 12:09:13 +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 8907aace7e..e9261a56ad (891631:892013) 2021-06-14 11:23:56 +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.py sctp: Rename SctpTransport to UsrSctpTransport 2021-04-12 10:40:34 +00:00
presubmit_test.py Reformat python files checked by pylint (part 1/2). 2020-10-30 10:13:11 +00:00
presubmit_test_mocks.py Reformat python files checked by pylint (part 1/2). 2020-10-30 10:13:11 +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.gni Turn on the RTC_ENABLE_WIN_WGC build flag. 2021-05-10 20:16:52 +00:00
webrtc_lib_link_test.cc Deprecate PeerConnectionFactory::CreatePeerConnection 2021-05-10 08:47:48 +00:00
whitespace.txt Reland "Triggering CI." 2021-03-22 11:57:23 +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