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chromium-webrtc-autoroll ca368dd976 Roll chromium_revision 5c2bd4f9ef..3b552b31ee (1353980:1354345) 
Change log: 5c2bd4f9ef..3b552b31ee
Full diff: 5c2bd4f9ef..3b552b31ee

Changed dependencies
* src/base: d8066bf67e..99dc460971
* src/build: 8da4111241..154f06e2b2
* src/ios: ffe22f74f9..774e1938da
* src/testing: ce25820b96..43f298a862
* src/third_party: 46762f4ba9..b434c2e410
* src/third_party/android_build_tools/error_prone/cipd: iksKTcNa8fCfCXLvYa9Og9yhPWH8iTk7xbESPSw243QC..fNCLAzE8NSvOXTryvUGT3NmX8no8lyRHR1yfY0zbv8YC
* src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/40ec347196..e724ef0208
* src/third_party/depot_tools: 1ad5b6c0df..17226d7965
* src/third_party/perfetto: 76c9a3333b..40477ffb51
* src/tools: c523a3ce5f..31966b2a47
Removed dependencies
* src/third_party/android_deps/cipd/libs/com_google_auto_auto_common
* src/third_party/android_deps/cipd/libs/com_google_auto_service_auto_service
DEPS diff: 5c2bd4f9ef..3b552b31ee/DEPS

No update to Clang.

BUG=None

Change-Id: I62090f8ad483bcbd4212f8d09e801d354e942469
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362382
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#43009}
2024-09-12 04:49:35 +00:00
api Ensure <netinet/in.h> is included by using rtc_base/ip_address.h. 2024-09-11 08:11:44 +00:00
audio Remove use of AcmReceiver in ChannelReceive 2024-09-06 12:47:36 +00:00
build_overrides build: add options to configure libsrtp for boringssl or other libraries 2024-08-27 07:17:52 +00:00
call Update WebRTC code version (2024-09-11T04:05:44). 2024-09-11 07:09:55 +00:00
common_audio Migrate absl::optional to std::optional 2024-09-02 12:16:47 +00:00
common_video Add converters for corruption detection structs 2024-09-11 13:44:04 +00:00
data
docs Fix formatting for corruption detection header explainer. 2024-08-27 15:18:27 +00:00
examples Ensure <netinet/in.h> is included by using rtc_base/ip_address.h. 2024-09-11 08:11:44 +00:00
experiments Add WebRTC-MixedCodecSimulcast field trial 2024-09-04 08:45:44 +00:00
g3doc Update ownership of PCLF documentation. 2024-09-11 13:03:08 +00:00
infra Revert "Enable 'iwyu_verifier' bot." 2024-09-02 10:14:35 +00:00
logging Fix lint issues in logging/ 2024-09-04 07:58:47 +00:00
media Update when/how requested_resolution throws for invalid parameters. 2024-09-11 09:45:08 +00:00
modules Ensure <netinet/in.h> is included by using rtc_base/ip_address.h. 2024-09-11 08:11:44 +00:00
net/dcsctp Prepend webrtc ns to StrJoin calls in dcsctp ns 2024-09-09 11:16:56 +00:00
p2p Migrate absl::optional to std::optional 2024-09-02 12:16:47 +00:00
pc Update when/how requested_resolution throws for invalid parameters. 2024-09-11 09:45:08 +00:00
resources Delete unused YUV files 2024-07-11 20:26:16 +00:00
rtc_base Ensure <netinet/in.h> is included by using rtc_base/ip_address.h. 2024-09-11 08:11:44 +00:00
rtc_tools Delete deprecated AudioDecoderFactory::MakeAudioDecoder 2024-09-04 07:17:59 +00:00
sdk Allow sdk/objc owners to approve sdk/BUILD.gn 2024-09-11 10:57:31 +00:00
stats Migrate absl::optional to std::optional 2024-09-02 12:16:47 +00:00
system_wrappers Migrate absl::optional to std::optional 2024-09-02 12:16:47 +00:00
test Update ownership of PCLF documentation. 2024-09-11 13:03:08 +00:00
tools_webrtc Mock call to os.path.isdir in roll_deps_test. 2024-09-09 15:10:07 +00:00
video Pass Environment into RtcpSender 2024-09-09 13:44:21 +00:00
.clang-format Add IncludeBlocks to clang-format. 2021-02-03 16:29:07 +00:00
.git-blame-ignore-revs Add formatting CLs to .git-blame-ignore-revs 2023-05-07 09:27:47 +00:00
.gitignore Add .cache to .gitignore. 2021-01-20 15:01:07 +00:00
.gn Roll chromium_revision ba1ae79f58..6f9b3224db (1319128:1338914) 2024-08-08 09:20:02 +00:00
.mailmap Add .mailmap for git. 2022-02-20 14:22:13 +00:00
.style.yapf Configure YAPF to follow PEP-8 altogether 2023-09-22 10:32:11 +00:00
.vpython3 Update to vpython 3.11 and remove .vpython (v2.x) 2024-01-25 11:12:20 +00:00
AUTHORS Adding ChannelStatistics Logs 2024-09-02 20:50:58 +00:00
BUILD.gn build: add options to configure libsrtp for boringssl or other libraries 2024-08-27 07:17:52 +00:00
CODE_OF_CONDUCT.md Reland "Migrate WebRTC documentation to new renderer" 2023-01-31 09:30:04 +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 5c2bd4f9ef..3b552b31ee (1353980:1354345) 2024-09-12 04:49:35 +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 phoglund from ENG_REVIEW_OWNERS 2021-10-08 08:29:42 +00:00
LICENSE
license_template.txt
native-api.md AudioProcessingImpl: Remove the use of transient suppressor 2024-08-05 12:38:37 +00:00
OWNERS Add infra owners file 2022-12-02 09:21:47 +00:00
OWNERS_INFRA Allow to keep old python style for existing files. 2023-10-17 13:52:56 +00:00
PATENTS
PRESUBMIT.py Ensure <netinet/in.h> is included by using rtc_base/ip_address.h. 2024-09-11 08:11:44 +00:00
presubmit_test.py tools_webrtc dir converted to py3 + top level PRESUBMIT script 2022-02-08 14:42:26 +00:00
presubmit_test_mocks.py tools_webrtc dir converted to py3 + top level PRESUBMIT script 2022-02-08 14:42:26 +00:00
pylintrc Configure Pylint to follow PEP-8 2023-09-25 15:56:09 +00:00
pylintrc_old_style Allow to keep old python style for existing files. 2023-10-17 13:52:56 +00:00
README.chromium Add Security Critical field to README.chromium. 2024-08-27 07:38:26 +00:00
README.md doc: Follow up link rename in I2dbe1ef0c74a0de8c5619b522fab39527e797d9c 2023-05-26 09:20:16 +00:00
WATCHLISTS Remove xooglers from WATCHLISTS and OWNERS 2022-11-30 15:33:25 +00:00
webrtc.gni AudioProcessingImpl: Remove the use of transient suppressor 2024-08-05 12:38:37 +00:00
webrtc_lib_link_test.cc Move webrtc::AudioDeviceModule include to api/ folder 2024-04-22 08:56:31 +00:00
whitespace.txt Test CQ 2024-05-27 12:46:57 +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