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chromium-webrtc-autoroll e75b7f2eb2 Roll chromium_revision 7b152a4cbe..323da96e89 (878194:878307) 
Change log: 7b152a4cbe..323da96e89
Full diff: 7b152a4cbe..323da96e89

Changed dependencies
* src/base: 72f1e2e7f1..ed1d3bf696
* src/build: f292eb8cc4..3309cd8706
* src/ios: 0bf9eaec04..679e9287ee
* src/testing: 81272d4d39..a8c3e0f428
* src/third_party: 02f8a2ccd1..3ade607129
* src/third_party/androidx: u2rB_PZbxW7A06T_fZaRSifQYPLnK7m2hUlbWLjzZd4C..shhcvIQmkadS49s7gxWnbJhKbME6mSxhR188J7CYIyYC
* src/third_party/depot_tools: c64e3902af..f663e54338
* src/tools: 70d816d4cd..330fffe4c6
* src/tools/luci-go: git_revision:7c21dae4ffe132b3bf611dce050d268f1ef4c155..git_revision:1b257aacd4934e5a8b2508b240ffc7d866df9273
* src/tools/luci-go: git_revision:7c21dae4ffe132b3bf611dce050d268f1ef4c155..git_revision:1b257aacd4934e5a8b2508b240ffc7d866df9273
* src/tools/luci-go: git_revision:7c21dae4ffe132b3bf611dce050d268f1ef4c155..git_revision:1b257aacd4934e5a8b2508b240ffc7d866df9273
DEPS diff: 7b152a4cbe..323da96e89/DEPS

No update to Clang.

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

Change-Id: Ie375f29b5c25fcc5b34b5cdf63cf23de32e8b3b2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217060
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@{#33892}
2021-05-03 02:47:39 +00:00
api
Reland "Removed VideoDecoderFactory::LegacyCreateVideoDecoder and VideoReceiveStream::Config::stream_id."
2021-04-30 11:40:38 +00:00
audio
Replace new rtc::RefCountedObject with rtc::make_ref_counted in a few files
2021-04-23 12:04:39 +00:00
build_overrides
Roll chromium_revision 34f3c82122..2dffe06711 (867171:871492)
2021-04-12 18:25:58 +00:00
call
Update WebRTC code version (2021-05-02T04:02:12).
2021-05-02 05:07:13 +00:00
common_audio
Avoid undefined behavior in a division operation.
2021-04-23 07:49:24 +00:00
common_video
Replace new rtc::RefCountedObject with rtc::make_ref_counted in a few files
2021-04-23 12:04:39 +00:00
data
…
docs
Create a VideoFrameTrackingId RTP header extension.
2021-03-25 17:25:18 +00:00
examples
Revert "Ensure method which updates UI is called in main thread"
2021-04-30 09:26:03 +00:00
g3doc
Add conceptual documentation for Audio - Mixer
2021-04-22 08:59:38 +00:00
logging
Remove use of istream in RTC event log parser.
2021-03-31 13:21:58 +00:00
media
Reland "Removed VideoDecoderFactory::LegacyCreateVideoDecoder and VideoReceiveStream::Config::stream_id."
2021-04-30 11:40:38 +00:00
modules
AGC2 config operator= and unit tests update
2021-04-29 16:37:17 +00:00
net/dcsctp
dcsctp: Add Socket
2021-05-01 07:16:21 +00:00
p2p
Reland "Add addr in error msg if stun sock sent with error"
2021-04-28 10:50:21 +00:00
pc
simulcast: Limit audio transceivers to single stream
2021-04-30 18:55:47 +00:00
resources
Disable high-pass filtering of the AEC reference
2021-02-23 07:06:11 +00:00
rtc_base
Reland "Add addr in error msg if stun sock sent with error"
2021-04-28 10:50:21 +00:00
rtc_tools
Replace more instances of rtc::RefCountedObject with make_ref_counted.
2021-04-27 17:01:59 +00:00
sdk
Reland "Expose AV1 encoder&decoder from Android SDK."
2021-04-30 09:12:11 +00:00
stats
Simplify reference counting implementation of PendingTaskSafetyFlag.
2021-04-21 07:04:01 +00:00
style-guide
Remove kwiberg@webrtc.org from OWNERS files
2020-12-04 15:11:26 +00:00
system_wrappers
Consolidate the different NTP clocks into one.
2021-04-08 13:54:04 +00:00
test
Reland "Removed VideoDecoderFactory::LegacyCreateVideoDecoder and VideoReceiveStream::Config::stream_id."
2021-04-30 11:40:38 +00:00
tools_webrtc
Make isac_fix_test an rtc_test
2021-04-28 15:06:11 +00:00
video
Reland "Removed VideoDecoderFactory::LegacyCreateVideoDecoder and VideoReceiveStream::Config::stream_id."
2021-04-30 11:40:38 +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
Configure GN to use python3 to exec_script.
2021-04-14 17:54:11 +00:00
.vpython
Update six library version
2021-04-26 16:39:07 +00:00
abseil-in-webrtc.md
Polish the "Using Abseil in WebRTC" docs
2020-10-16 13:42:00 +00:00
AUTHORS
Revert "Ensure method which updates UI is called in main thread"
2021-04-30 09:26:03 +00:00
BUILD.gn
Move RTC_ENABLE_WIN_WGC define to the top level BUILD.gn
2021-04-08 16:31:49 +00:00
CODE_OF_CONDUCT.md
…
codereview.settings
…
DEPS
Roll chromium_revision 7b152a4cbe..323da96e89 (878194:878307)
2021-05-03 02:47:39 +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
Make the remote_bitrate_estimator build target private
2020-11-26 12:21:22 +00:00
OWNERS
Add mbonadei@ as owner of .pylintrc / .vpython.
2021-04-26 16:34:57 +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
sctp: Rename SctpTransport to UsrSctpTransport
2021-04-12 10:40:34 +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
Fix incorrect link in README.md
2021-04-20 10:58:08 +00:00
style-guide.md
Enhance the readability of the style guide.
2021-04-23 15:35:56 +00:00
WATCHLISTS
Add hta@ to rtc_base/ and api/ WATCHLISTS.
2021-01-06 09:43:34 +00:00
webrtc_lib_link_test.cc
Rewrite the lib link test to just be a binary.
2019-10-18 07:42:20 +00:00
webrtc.gni
Add GetTopLeft to WgcCaptureSource to facilitate cursor capture.
2021-04-23 16:46:49 +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

Description
No description provided
Readme 255 MiB
Languages
C++ 88.6%
C 3.3%
Java 3%
Objective-C++ 1.9%
Python 1.9%
Other 1%