zqz/platform-external-webrtc
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity
28,287 Commits 1 Branch 0 Tags
38350b1ef26330f3caebe84d4d935671d4e26720
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 38350b1ef2 Roll chromium_revision d74690feb1..248662b1b8 (692875:693000) 
Change log: d74690feb1..248662b1b8
Full diff: d74690feb1..248662b1b8

Changed dependencies
* src/base: 0f64a756c5..b429f3f492
* src/build: b3e93c0482..693faeda4e
* src/ios: 48037d2833..4cea3a8c7d
* src/testing: 7b91b663fc..d2671f94c4
* src/third_party: 8bc6decc0b..07fc4f799d
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/5198ea1a70..e7c719c3e8
* src/third_party/depot_tools: 1db68ea0ba..f38bc17962
* src/tools: 347ed0a547..e4fc4f21f3
DEPS diff: d74690feb1..248662b1b8/DEPS

No update to Clang.

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

Change-Id: Idcae1907eb3a0bc465f1059e7be2b757b3c43956
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151342
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@{#29055}
2019-09-04 04:41:52 +00:00
api
Reland "Reland "Add core multi-channel pipeline in AEC3 This CL adds basic the basic pipeline to support multi-channel processing in AEC3.""
2019-09-03 06:12:32 +00:00
audio
Deprecate SingleThreadedTaskQueueForTesting class.
2019-09-03 10:31:30 +00:00
build_overrides
Remove crbug.com/904400 workaround.
2019-03-15 18:36:23 +00:00
call
Deprecate SingleThreadedTaskQueueForTesting class.
2019-09-03 10:31:30 +00:00
common_audio
Remove rtc_use_lto GN arg.
2019-08-20 14:00:49 +00:00
common_video
Add new FrameRateEstimator utility class for more precis FPS estimation.
2019-08-14 12:15:06 +00:00
data
Remove old data files.
2018-10-05 14:40:21 +00:00
examples
Make the RtpHeaderParserImpl available to tests and tools only.
2019-08-29 15:56:40 +00:00
logging
Delete unneeded dependencies on libjingle_peerconnection_api
2019-08-29 10:52:42 +00:00
media
Buffers non atomic message send with usrsctp lib.
2019-09-03 16:30:21 +00:00
modules
Revert "Refactor FEC code to use COW buffers"
2019-09-03 07:53:05 +00:00
p2p
Use the sanitized pair when surfacing the candidate pair change event.
2019-09-03 17:17:49 +00:00
pc
Split out RtpSource from libjingle_peerconnection_api
2019-09-02 14:04:47 +00:00
resources
Use the AEC3 high-pass filter for the whole APM
2019-08-23 20:04:10 +00:00
rtc_base
Add support for unsigned parameters in FieldTrialParser
2019-08-30 08:23:37 +00:00
rtc_tools
Increased event log visualizer RTP clock estimation tolerance.
2019-08-29 10:08:43 +00:00
sdk
Use the sanitized pair when surfacing the candidate pair change event.
2019-09-03 17:17:49 +00:00
stats
Add support for RTCTransportStats.selectedCandidatePairChanges
2019-08-28 13:22:08 +00:00
style-guide
Add style guide rule about paired .h and .cc files
2018-03-14 13:02:35 +00:00
system_wrappers
Add helper functions to convert between integer milliseconds and fixed-point seconds.
2019-08-16 14:49:46 +00:00
test
Deprecate SingleThreadedTaskQueueForTesting class.
2019-09-03 10:31:30 +00:00
tools_webrtc
Add license for android_ndk
2019-08-28 14:57:28 +00:00
video
Reland "Make the min video bitrate in VideoSendStream configurable."
2019-09-03 15:25:31 +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 webrtc_apprtc_browsertest.cc resource dir to .gitignore.
2019-09-03 09:21:43 +00:00
.gn
Remove last mention of ortc from the codebase.
2019-05-25 07:28:05 +00:00
.vpython
Add vpython dependencies needed to run presubmit tests on LUCI
2018-05-18 08:10:25 +00:00
abseil-in-webrtc.md
Allowing buffering a LNTF (loss notification) feedback message in RTCPSender
2019-06-03 16:28:34 +00:00
AUTHORS
Stun server should return XOR-MAPPED-ADDRESS/MAPPED-ADDRESS correctly
2019-06-28 19:12:14 +00:00
BUILD.gn
Make the RtpHeaderParserImpl available to tests and tools only.
2019-08-29 15:56:40 +00:00
CODE_OF_CONDUCT.md
Add code of conduct to WebRTC repo
2017-05-16 12:09:13 +00:00
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
Format almost everything.
2019-07-08 13:45:15 +00:00
DEPS
Roll chromium_revision d74690feb1..248662b1b8 (692875:693000)
2019-09-04 04:41:52 +00:00
ENG_REVIEW_OWNERS
Enforce LGTM from owners of depends-on paths in DEPS via presubmit.
2018-09-28 12:49:54 +00:00
LICENSE
Moving src/webrtc into src/.
2017-09-15 04:25:06 +00:00
license_template.txt
…
native-api.md
Delete unused I420 "codec"
2018-12-18 12:30:58 +00:00
OWNERS
Add juberti@ to webrtc root owners
2019-05-17 18:11:58 +00:00
PATENTS
Moving src/webrtc into src/.
2017-09-15 04:25:06 +00:00
presubmit_test_mocks.py
Reland: Add presubmit check for changes in 3pp
2018-05-22 13:11:18 +00:00
presubmit_test.py
Fixing py lint errors
2018-07-23 15:28:48 +00:00
PRESUBMIT.py
Disable RunPythonTests on rtc_tools.
2019-08-06 12:48:33 +00:00
pylintrc
Fixing py lint errors
2018-07-23 15:28:48 +00:00
README.chromium
Moving src/webrtc into src/.
2017-09-15 04:25:06 +00:00
README.md
Tell users where they can find the native API headers
2017-11-14 10:36:46 +00:00
style-guide.md
Remove rule that discourages passing optional by const reference
2019-02-05 11:58:05 +00:00
WATCHLISTS
Add saza to audio watchlists
2019-09-03 14:55:43 +00:00
webrtc.gni
Remove rtc_use_lto GN arg.
2019-08-20 14:00:49 +00:00
whitespace.txt
Whitespace change
2018-02-23 10:34:16 +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
No description provided
Readme 255 MiB
Languages
C++ 88.6%
C 3.3%
Java 3%
Objective-C++ 1.9%
Python 1.9%
Other 1%