admin/webrtc_m130
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity
31,092 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 0031b27671 Roll chromium_revision ada5164b5b..ae72e95193 (794977:795090) 
Change log: ada5164b5b..ae72e95193
Full diff: ada5164b5b..ae72e95193

Changed dependencies
* src/base: e6c4077b24..3f1aa91003
* src/build: 725c95091a..4d6f87b564
* src/ios: 47c35b1e27..53b3915d26
* src/testing: 36e4d4e9a5..1766d1e17d
* src/third_party: c36638882d..cd36d44fcf
* src/third_party/freetype/src: 8b3601324f..f9f6adb625
* src/third_party/perfetto: 23c5b63d1c..9a43f48ed4
* src/tools: 16f8e073d7..5baca2e05d
DEPS diff: ada5164b5b..ae72e95193/DEPS

No update to Clang.

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

Change-Id: I1e8910027e685f82ed4fa1d7053ea7065114d78c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181021
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@{#31860}
2020-08-05 18:32:51 +00:00
api
Reland "Add EncodedImageCallback::OnEncodedImage without RTPFragmentationHeader"
2020-08-04 12:17:06 +00:00
audio
Remove unused critical section includes.
2020-07-16 13:52:28 +00:00
build_overrides
set perfetto flag to default value of false
2020-07-22 10:14:53 +00:00
call
Revert "Moved VideoReceiveStream::Decoder::decoder_factory to VideoReceiveStream::Config::decoder_factory."
2020-08-03 15:45:41 +00:00
common_audio
Revert "Support AVX2/FMA intrinsics in Audio Resampler module"
2020-07-30 17:35:30 +00:00
common_video
Ignore RTPFragmentationHeader when rewriting H264 SPS
2020-07-27 09:42:25 +00:00
data
…
docs
Search and replace gendered terms according to style guide:
2020-06-12 14:12:54 +00:00
examples
Implemented Android Demo Application for VoIP API
2020-07-21 16:34:22 +00:00
logging
Search and replace gendered terms according to style guide:
2020-06-12 14:12:54 +00:00
media
Reducing threshold for usrsctp "buffer low" callback.
2020-08-04 20:08:06 +00:00
modules
Delete deprecated RTPSenderVideo::SendVideo function
2020-08-05 13:10:36 +00:00
p2p
Reland "Implement packets_(sent | received) for RTCTransportStats"
2020-07-10 11:50:59 +00:00
pc
sdp: reject sdp with malformed b= lines
2020-08-03 18:21:14 +00:00
resources
iSAC API wrapper unit test fix
2020-02-27 14:27:23 +00:00
rtc_base
Delete old name ProcessAllMessageQueues
2020-08-03 11:24:17 +00:00
rtc_tools
Revert "Moved VideoReceiveStream::Decoder::decoder_factory to VideoReceiveStream::Config::decoder_factory."
2020-08-03 15:45:41 +00:00
sdk
Changed AndroidVideoDecoder to also handle IllegalArgumentException and IllegalStateException during the init of the decoder and fallback to a software decoder
2020-08-05 09:41:49 +00:00
stats
Reland "Implement packets_(sent | received) for RTCTransportStats"
2020-07-10 11:50:59 +00:00
style-guide
…
system_wrappers
Revert "Support AVX2/FMA intrinsics in Audio Resampler module"
2020-07-30 17:35:30 +00:00
test
Add networks stats collector from PeerConnection GetStats API
2020-08-04 09:05:14 +00:00
tools_webrtc
Fix "Assignment had no effect." error during chromium roll.
2020-07-30 12:57:20 +00:00
video
Reland "Only enable conference mode simulcast allocations with flag enabled"
2020-08-04 10:30:08 +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
Reenable libaom decoder by default
2020-03-18 18:04:41 +00:00
.vpython
Add source-side perf upload script for WebRTC.
2019-11-18 14:37:01 +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
Changed AndroidVideoDecoder to also handle IllegalArgumentException and IllegalStateException during the init of the decoder and fallback to a software decoder
2020-08-05 09:41:49 +00:00
BUILD.gn
Revert "Support AVX2/FMA intrinsics in Audio Resampler module"
2020-07-30 17:35:30 +00:00
CODE_OF_CONDUCT.md
…
codereview.settings
…
common_types.h
Replaces OverheadObserver with simple getter.
2020-05-07 17:33:45 +00:00
DEPS
Roll chromium_revision ada5164b5b..ae72e95193 (794977:795090)
2020-08-05 18:32:51 +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
Inclusive language in PRESUBMIT.py.
2020-07-22 10:01:23 +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
Revert "Support AVX2/FMA intrinsics in Audio Resampler module"
2020-07-30 17:35:30 +00:00
whitespace.txt
Trigger CI bots.
2020-07-15 17:50:55 +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%