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chromium-webrtc-autoroll cc49311969 Roll chromium_revision 42f2ed2fa5..5954b85f34 (877422:877544) 
Change log: 42f2ed2fa5..5954b85f34
Full diff: 42f2ed2fa5..5954b85f34

Changed dependencies
* src/base: e6728e4cbb..2d7025bff0
* src/build: 1c369f1e0d..5d7a2fb65e
* src/ios: 56aab7bebd..07d4b83f84
* src/testing: a660a95e6f..bb0cb73182
* src/third_party: 536f2d8a8e..392c32b18c
* src/third_party/perfetto: 424d889847..a6c4cea7f7
* src/tools: c4ab2cb887..2d8757a32a
DEPS diff: 42f2ed2fa5..5954b85f34/DEPS

No update to Clang.

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

Change-Id: I336a78487d604e6f4fd4f2cb748c69fe574b9e12
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/216660
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@{#33877}
2021-04-29 18:30:49 +00:00
api
Add QueryCodecSupport() to VideoEncoderFactory and VideoDecoderFactory
2021-04-27 10:23:05 +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
Provide FrameDependecyStructure for VP9 when encoder doesn't fill it
2021-04-29 12:28:46 +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
Fix iOS chromium roll issue
2021-04-21 08:06:11 +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
Remove Mutex from BaseChannel.
2021-04-27 12:46:10 +00:00
modules
AGC2 config operator= and unit tests update
2021-04-29 16:37:17 +00:00
net/dcsctp
dcsctp: Add Transmission Control Block
2021-04-28 22:45:03 +00:00
p2p
Reland "Add addr in error msg if stun sock sent with error"
2021-04-28 10:50:21 +00:00
pc
datachannel: Check we transition the state properly on remote close
2021-04-29 14:14:42 +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
Replace more instances of rtc::RefCountedObject with make_ref_counted.
2021-04-27 17:01:59 +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
Fix typo in metric.
2021-04-29 08:46:40 +00:00
tools_webrtc
Make isac_fix_test an rtc_test
2021-04-28 15:06:11 +00:00
video
Replace new rtc::RefCountedObject with rtc::make_ref_counted in a few files
2021-04-23 12:04:39 +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
Adds missing header to fix compilation error when compiling with use_custom_libcxx set to false.
2021-03-25 09:57:00 +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 42f2ed2fa5..5954b85f34 (877422:877544)
2021-04-29 18:30:49 +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
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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