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chromium-webrtc-autoroll fab9256fb6 Roll chromium_revision a0592593ad..19bcae5aaa (883988:884108) 
Change log: a0592593ad..19bcae5aaa
Full diff: a0592593ad..19bcae5aaa

Changed dependencies
* src/base: 65528ba02a..db30053b7a
* src/build: d2ba2f2766..8ef5618407
* src/ios: c42253cc6c..5b15656973
* src/testing: fbeaf7e18b..d5026c75fa
* src/third_party: f977052770..575566d758
* src/third_party/depot_tools: fe7d709f05..8c9a5b84cb
* src/third_party/perfetto: 9109f50a74..f96c36dc3c
* src/tools: 018f0f1344..05f166d97e
DEPS diff: a0592593ad..19bcae5aaa/DEPS

No update to Clang.

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

Change-Id: Ica23bc3172edfc3b3c4ca85f644f1bba7e8b14e3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219247
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@{#34043}
2021-05-18 21:43:19 +00:00
api
VideoRtpReceiver & AudioRtpReceiver threading fixes.
2021-05-17 14:37:55 +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
Remove redundant VideoSendStream::rtcp_stats field
2021-05-18 13:37:51 +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
…
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
Update BitBuffer methods to style guide
2021-05-18 11:10:27 +00:00
media
Remove redundant VideoSendStream::rtcp_stats field
2021-05-18 13:37:51 +00:00
modules
Improve test coverage for padding packet generation.
2021-05-18 16:47:16 +00:00
net/dcsctp
dcsctp: Restrict fuzzing input length
2021-05-18 09:22:37 +00:00
p2p
Define cricket::DtlsTransportState in terms of webrtc::DtlsTransportState.
2021-05-12 07:06:38 +00:00
pc
peerconnection: add test for createOffer({voiceActivityDetection: true})
2021-05-18 09:10:27 +00:00
resources
Disable high-pass filtering of the AEC reference
2021-02-23 07:06:11 +00:00
rtc_base
Update BitBuffer methods to style guide
2021-05-18 11:10:27 +00:00
rtc_tools
Remove REMB throttling funcionality from PacketRouter
2021-05-12 11:24:58 +00:00
sdk
Set nativeObserver to 0 to avoid double release.
2021-05-17 19:53:43 +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
Remove redundant VideoSendStream::rtcp_stats field
2021-05-18 13:37:51 +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 a0592593ad..19bcae5aaa (883988:884108)
2021-05-18 21:43:19 +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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