admin/webrtc_m130
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity
30,025 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 630917c398 Roll chromium_revision 3874769f74..0c9b27ea8c (750095:750227) 
Change log: 3874769f74..0c9b27ea8c
Full diff: 3874769f74..0c9b27ea8c

Changed dependencies
* src/base: 03e8bf961d..0480c42d1f
* src/build: 1c04ad9904..55d31f7597
* src/ios: fcdb76daab..6f8f463bca
* src/testing: 6fa4be8438..cf3d09eb73
* src/third_party: 6703134d34..565104c5a2
* src/tools: b2129e2f92..89c43f8c62
DEPS diff: 3874769f74..0c9b27ea8c/DEPS

No update to Clang.

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

Change-Id: I7d260e384f9ad5facfe3a1da278072d704dde5f0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170481
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@{#30793}
2020-03-13 18:47:15 +00:00
api
Add jitterBufferTargetDelay as RTCNonStandardStatsMember to new GetStats API
2020-03-11 12:08:32 +00:00
audio
Clamp stable target bitrate to min/max allocated bitrate.
2020-03-12 13:51:10 +00:00
build_overrides
Purge phoglund from most OWNERS files.
2020-03-09 14:08:30 +00:00
call
Pass callbacks for RtcpReceiver at construction
2020-03-12 10:26:17 +00:00
common_audio
Optimizations and refactoring of the APM 3-band split filter
2020-02-24 13:19:14 +00:00
common_video
Remove wildcard ownership for build files.
2020-02-19 14:05:46 +00:00
data
Remove old data files.
2018-10-05 14:40:21 +00:00
docs
Rename index.md to README.md to make it automatically show up
2020-03-06 10:43:51 +00:00
examples
Purge phoglund from most OWNERS files.
2020-03-09 14:08:30 +00:00
logging
Replace DataSize and DataRate factories with newer versions
2020-02-18 16:09:50 +00:00
media
Add jitterBufferTargetDelay as RTCNonStandardStatsMember to new GetStats API
2020-03-11 12:08:32 +00:00
modules
Split out some dependencies from the monolith audio processing target
2020-03-12 21:07:08 +00:00
p2p
Make Connection::id() const
2020-03-05 09:30:18 +00:00
pc
Remove cricket::SessionDescription::Copy()
2020-03-13 07:32:41 +00:00
resources
iSAC API wrapper unit test fix
2020-02-27 14:27:23 +00:00
rtc_base
Revert "Leverage dispatch_queue_create_with_target when possible."
2020-03-13 08:02:34 +00:00
rtc_tools
Write protos as binary.
2020-03-12 09:43:57 +00:00
sdk
Revert "Leverage dispatch_queue_create_with_target when possible."
2020-03-13 08:02:34 +00:00
stats
Add jitterBufferTargetDelay as RTCNonStandardStatsMember to new GetStats API
2020-03-11 12:08:32 +00:00
style-guide
…
system_wrappers
Add RTC_EXPORT to webrtc::Clock
2020-03-10 23:00:21 +00:00
test
Ensures that all simulated TCP packets are at least 4 bytes.
2020-03-12 17:42:13 +00:00
tools_webrtc
Apply some more hacks.
2020-03-13 15:13:17 +00:00
video
Remove the least important WebRTC video tests.
2020-03-13 13:59:41 +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
Add Av1 Decoder wrapper behind a build flag
2020-03-11 11:20:56 +00:00
.vpython
Add source-side perf upload script for WebRTC.
2019-11-18 14:37:01 +00:00
abseil-in-webrtc.md
Fix typo in abseil-in-webrtc.md.
2019-12-18 14:27:34 +00:00
AUTHORS
Reland "Reland "Reland "Reland "Reland "Implemented screen enumeration and selection for desktop capture under X11 using the X Resize and Rotate extension version 1.5."""""
2020-02-26 20:35:54 +00:00
BUILD.gn
Fix 'all' build on non Android platforms.
2020-03-11 08:24:27 +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
common_types.h
Reland "Remove PlayoutDelayOracle and make RtpSenderVideo guarantee delivery"
2020-02-07 08:23:58 +00:00
DEPS
Roll chromium_revision 3874769f74..0c9b27ea8c (750095:750227)
2020-03-13 18:47:15 +00:00
ENG_REVIEW_OWNERS
…
LICENSE
…
license_template.txt
…
native-api.md
Delete unused I420 "codec"
2018-12-18 12:30:58 +00:00
OWNERS
Add #COMPONENT to WebRTC.
2019-10-08 12:20:39 +00:00
PATENTS
…
presubmit_test_mocks.py
…
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 public_deps presubmit and gn format fighting each other.
2020-01-30 11:22:46 +00:00
pylintrc
…
README.chromium
…
README.md
…
style-guide.md
Add guidance to style guide how to reference a bug in a TODO
2019-12-11 11:55:52 +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
Reformat GN files.
2020-01-21 12:13:11 +00:00
whitespace.txt
Whitespace change
2020-01-28 07:53:15 +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 http://www.webrtc.org/native-code/development 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%