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chromium-webrtc-autoroll c9bb4a4a0d Roll chromium_revision c4b575c458..60ef3818ac (883038:883287) 
Change log: c4b575c458..60ef3818ac
Full diff: c4b575c458..60ef3818ac

Changed dependencies
* src/base: 47d56825de..f26f0ec685
* src/build: a51f35cb81..9be9e3818a
* src/ios: b790e9c8a9..340c574ae6
* src/testing: 0e57f3340c..31a90015dd
* src/third_party: 24ea7d20ea..87acabbc64
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/affd272f9e..a532699d20
* src/third_party/depot_tools: 5ab2f3314c..cd152033d0
* src/tools: 786cfa6947..a30c6d0da7
DEPS diff: c4b575c458..60ef3818ac/DEPS

No update to Clang.

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

Change-Id: I1303b1a3e7623727ba858bdfb39461b58d036f13
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/218882
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@{#34011}
2021-05-15 18:50:09 +00:00
api
Move style guide and abseil-in-webrtc into g3doc subfolder
2021-05-13 14:43:10 +00:00
audio
Use Timestamp to represent packet receive timestamps
2021-05-04 13:16:54 +00:00
build_overrides
Roll chromium_revision 34f3c82122..2dffe06711 (867171:871492)
2021-04-12 18:25:58 +00:00
call
Update WebRTC code version (2021-05-14T04:05:03).
2021-05-14 05:24:29 +00:00
common_audio
Avoid undefined behavior in a division operation.
2021-04-23 07:49:24 +00:00
common_video
Reland "Remove Invoke from VideoChannel::FillBitrateInfo."
2021-05-03 15:16:34 +00:00
data
…
docs
Create a VideoFrameTrackingId RTP header extension.
2021-03-25 17:25:18 +00:00
examples
Deprecate PeerConnectionFactory::CreatePeerConnection
2021-05-10 08:47:48 +00:00
g3doc
Move style guide and abseil-in-webrtc into g3doc subfolder
2021-05-13 14:43:10 +00:00
logging
Change rtc event log packet messages implementation to save full rtp packet
2021-05-11 08:06:54 +00:00
media
Revert "Define cricket::MediaType in terms of webrtc::MediaType"
2021-05-12 17:06:58 +00:00
modules
Move some tests out from rtp_sender_unittest.
2021-05-12 14:01:29 +00:00
net/dcsctp
dcsctp: Add socket fuzzer helper
2021-05-12 15:06:17 +00:00
p2p
Define cricket::DtlsTransportState in terms of webrtc::DtlsTransportState.
2021-05-12 07:06:38 +00:00
pc
add dtls docs
2021-05-12 21:16:10 +00:00
resources
Disable high-pass filtering of the AEC reference
2021-02-23 07:06:11 +00:00
rtc_base
Add 64-bit version of BitBuffer::ReadBits
2021-05-12 14:24:46 +00:00
rtc_tools
Remove REMB throttling funcionality from PacketRouter
2021-05-12 11:24:58 +00:00
sdk
Revert "Define cricket::MediaType in terms of webrtc::MediaType"
2021-05-12 17:06:58 +00:00
stats
Simplify reference counting implementation of PendingTaskSafetyFlag.
2021-04-21 07:04:01 +00:00
system_wrappers
Consolidate the different NTP clocks into one.
2021-04-08 13:54:04 +00:00
test
dcsctp: Add socket fuzzer
2021-05-14 06:41:10 +00:00
tools_webrtc
crc32c: Point the licensing script to the LICENSE file
2021-05-03 16:46:30 +00:00
video
Don't store RtpPacketInfo in the PacketBuffer.
2021-05-11 10:37:46 +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
Enable GN check on //net.
2021-05-03 14:23:09 +00:00
.vpython
Update six library version
2021-04-26 16:39:07 +00:00
AUTHORS
Revert "Ensure method which updates UI is called in main thread"
2021-04-30 09:26:03 +00:00
BUILD.gn
Disable range-loop-analysis when use_xcode_clang=true.
2021-05-07 15:24:28 +00:00
CODE_OF_CONDUCT.md
…
codereview.settings
…
DEPS
Roll chromium_revision c4b575c458..60ef3818ac (883038:883287)
2021-05-15 18:50:09 +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
Move style guide and abseil-in-webrtc into g3doc subfolder
2021-05-13 14:43:10 +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
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_lib_link_test.cc
Deprecate PeerConnectionFactory::CreatePeerConnection
2021-05-10 08:47:48 +00:00
webrtc.gni
Turn on the RTC_ENABLE_WIN_WGC build flag.
2021-05-10 20:16:52 +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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