admin/webrtc_m130
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity
31,549 Commits 1 Branch 0 Tags
Go to file
Clone
Open with VS Code Open with VSCodium Open with Intellij IDEA
Download ZIP Download TAR.GZ Download BUNDLE
chromium-webrtc-autoroll e3aeeb9eb4 Roll chromium_revision ac83b0e7d9..a05846da23 (813657:813761) 
Change log: ac83b0e7d9..a05846da23
Full diff: ac83b0e7d9..a05846da23

Changed dependencies
* src/build: 4317485981..50f81f2452
* src/ios: 90d14b9ef7..5a1c27ee54
* src/testing: 63334ab3ed..2052eba535
* src/third_party: d2796410c9..0d31f91cd7
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/d82d30da43..74f8dbb957
* src/third_party/perfetto: af6c5c6227..7cdc44f903
* src/tools: 14247e8ed5..eb39a750fb
DEPS diff: ac83b0e7d9..a05846da23/DEPS

No update to Clang.

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

Change-Id: I9747688d5968c9615d68429a871439af69b0cc7a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186800
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@{#32317}
2020-10-05 17:09:48 +00:00
api
Add experimental extension RtpVideoLayersAllocation
2020-10-05 13:38:13 +00:00
audio
Async audio processing API
2020-10-02 12:33:34 +00:00
build_overrides
set perfetto flag to default value of false
2020-07-22 10:14:53 +00:00
call
Cleans up code related to video packetization overhead.
2020-10-05 13:22:33 +00:00
common_audio
Reland "Activating AVX2 support by default"
2020-10-03 22:37:13 +00:00
common_video
Remove deprecated i420 buffer pool
2020-09-23 12:22:40 +00:00
data
Remove old data files.
2018-10-05 14:40:21 +00:00
docs
Adding documentation to become a WebRTC committer.
2020-09-02 07:49:04 +00:00
examples
Fix libyuv includes for libyuv roll
2020-10-02 08:03:41 +00:00
logging
Allow RTCP packets longer than 1500 bytes in RTC event log.
2020-10-05 15:26:54 +00:00
media
Take max bitrate into account for target bitrate decision when min bitrate is empty
2020-10-05 11:11:51 +00:00
modules
AGC2 logs
2020-10-05 15:34:52 +00:00
p2p
Give PseudoTcp its own FifoBuffer.
2020-10-05 12:35:13 +00:00
pc
Separate PeerConnection's self-message functions to a new file
2020-10-05 12:41:23 +00:00
resources
Reland "Activating AVX2 support by default"
2020-10-03 22:37:13 +00:00
rtc_base
Revert "[Sheriff] Revert "Remove MessageHandler[AutoCleanup] dependency from StreamInterface.""
2020-10-05 13:46:03 +00:00
rtc_tools
Add method to extract triage alerts from RTC event log analyzer.
2020-10-02 13:41:30 +00:00
sdk
Delete AllowBlockingCalls and android_media_codec_common.h
2020-10-01 16:38:09 +00:00
stats
Reland "Implement packets_(sent | received) for RTCTransportStats"
2020-07-10 11:50:59 +00:00
style-guide
…
system_wrappers
Reland "Activating AVX2 support by default"
2020-10-03 22:37:13 +00:00
test
Add experimental extension RtpVideoLayersAllocation
2020-10-05 13:38:13 +00:00
tools_webrtc
Create isolated output directory when creating the output file.
2020-09-29 15:38:52 +00:00
video
Cleans up code related to video packetization overhead.
2020-10-05 13:22:33 +00:00
.clang-format
…
.git-blame-ignore-revs
Let git-hyper-blame ignore new format cleanup.
2019-07-11 16:18:51 +00:00
.gitignore
Add .clangd to .gitignore
2019-10-28 12:27:50 +00:00
.gn
Rename PlayoutDelay --> VideoPlayoutDelay, move to api/video/video_timing.h
2020-09-07 08:37:14 +00:00
.vpython
Add 'requests' to .vpython.
2020-09-09 14:36:03 +00:00
abseil-in-webrtc.md
Use absl_deps in order to preapre to the Abseil component build release.
2020-06-08 12:59:40 +00:00
AUTHORS
Fix bug with the sps-pps-idr-in-keyframe fmtp parameter.
2020-09-19 09:23:39 +00:00
BUILD.gn
Rename CancerStickCastle to RoboCaller.
2020-09-21 19:50:39 +00:00
CODE_OF_CONDUCT.md
…
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 ac83b0e7d9..a05846da23 (813657:813761)
2020-10-05 17:09:48 +00:00
ENG_REVIEW_OWNERS
…
LICENSE
…
license_template.txt
…
native-api.md
Make transient suppression optionally excludable via defines
2020-04-02 11:44:07 +00:00
OWNERS
Remove phoglund as root owner.
2020-03-30 12:15:56 +00:00
PATENTS
…
presubmit_test_mocks.py
Inclusive language in PRESUBMIT.py.
2020-07-22 10:01:23 +00:00
presubmit_test.py
Use source_sets in component builds and static_library in release builds.
2019-10-17 21:17:18 +00:00
PRESUBMIT.py
Fix PRESUBMIT.py warnings.
2020-09-07 08:34:44 +00:00
pylintrc
…
README.chromium
Add CPEPrefix.
2020-07-13 11:42:07 +00:00
README.md
Fix link in documentation. (take 2)
2020-04-16 11:08:43 +00:00
style-guide.md
C++ style: We don't allow designated initializers
2020-06-03 09:11:09 +00:00
WATCHLISTS
Remove benwright@webrtc.org from WATCHLISTS
2020-01-31 18:46:52 +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
Reland "Activating AVX2 support by default"
2020-10-03 22:37:13 +00:00
whitespace.txt
Trigger CI bots.
2020-09-06 10:13:23 +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
Languages
C++ 90.3%
Java 2.9%
C 2.2%
Objective-C++ 2%
Python 1.3%
Other 1%