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chromium-webrtc-autoroll e323359d6f Roll chromium_revision 25abf646db..4e6fe7f5f7 (667125:667532) 
Change log: 25abf646db..4e6fe7f5f7
Full diff: 25abf646db..4e6fe7f5f7

Changed dependencies
* src/base: e77d1e96aa..01a87f7e42
* src/build: 7d6de605d2..6485681f33
* src/buildtools: 0218c0f9ac..2e18a95798
* src/ios: e475fee991..cb5cc07a7b
* src/testing: 7e2a133ba9..fbe653e44b
* src/third_party: 4dfcbb6775..7cd5dc0f6d
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/abea78f985..93cd912aa6
* src/third_party/depot_tools: 073b8ac2ff..e30736fb09
* src/third_party/libvpx/source/libvpx: 0308a9a132..28cc5f3646
* src/tools: ac4da5bd39..6f560cc4c6
* src/tools/swarming_client: 9b1b0ed1f3..96f125709a
DEPS diff: 25abf646db..4e6fe7f5f7/DEPS

Clang version changed 67510fac36d27b2e22c7cd955fc167136b737b93:80fee25776c2fb61e74c1ecb1a523375c2500b69
Details: 25abf646db..4e6fe7f5f7/tools/clang/scripts/update.py

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

Change-Id: I130e1f8c6aa0dc2c44ca02ea23e90d50d1dab420
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141220
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28203}
2019-06-10 08:33:32 +00:00
api
Remove myself from OWNERS in a few places.
2019-06-10 07:57:46 +00:00
audio
Reland "Avoid encrypting empty audio packet."
2019-05-28 12:30:07 +00:00
build_overrides
Remove crbug.com/904400 workaround.
2019-03-15 18:36:23 +00:00
call
Remove myself from OWNERS in a few places.
2019-06-10 07:57:46 +00:00
common_audio
Add ability to play audio in circle for TestAudioDevice wav file capturer
2019-04-16 15:33:03 +00:00
common_video
Move H.264 SPS VUI rewriting to FrameEncodeMetadataWriter.
2019-05-29 10:37:22 +00:00
crypto
Adding new top-level directory crypto/
2019-03-08 00:35:05 +00:00
data
Remove old data files.
2018-10-05 14:40:21 +00:00
examples
Target SDK level 29 in AppRTCMobile.
2019-06-03 12:40:42 +00:00
logging
Fix for crash in event log when using scenario tests.
2019-05-22 15:22:49 +00:00
media
Remove myself from OWNERS in a few places.
2019-06-10 07:57:46 +00:00
modules
Fix assert check in neteq_quality_test
2019-06-10 08:01:26 +00:00
p2p
Reland: Implement true negotiation for DatagramTransport with fallback to RTP.
2019-06-07 20:14:36 +00:00
pc
Remove obsolete test changes
2019-06-08 00:52:43 +00:00
resources
RNN VAD: clean-up unit tests
2019-04-29 12:55:02 +00:00
rtc_base
Fix error message in rtc_base/system/arch.h
2019-06-07 22:58:48 +00:00
rtc_tools
Fix BWE simulation graph in event log visualization
2019-05-28 10:40:29 +00:00
sdk
Add support for enabling and negotiating raw RTP packetization.
2019-06-04 14:35:54 +00:00
stats
Implement QualityLimitationReasonTracker and expose "reason".
2019-05-28 16:23:55 +00:00
style-guide
Add style guide rule about paired .h and .cc files
2018-03-14 13:02:35 +00:00
system_wrappers
Ensure CpuInfo::DetectNumberOfCores is > 0 and thread safe.
2019-05-24 12:59:14 +00:00
test
Adds TCP fairness test to receive side congestion controller.
2019-06-07 13:49:37 +00:00
tools_webrtc
mb: Implement 'quiet' flag in mb lookup
2019-05-31 11:51:15 +00:00
video
Cleanup video frame metadata copying
2019-06-05 15:05:57 +00:00
.clang-format
…
.git-blame-ignore-revs
Let git-hyper-blame ignore format commit.
2018-06-20 09:26:44 +00:00
.gitignore
Add Visual Studio Code project folder to gitignore file.
2019-01-21 18:42:33 +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
Import proto_library.gni when rtc_enable_protobuf is true
2019-02-27 09:56:42 +00:00
BUILD.gn
Delete rtc_base/unittest_main.cc
2019-05-21 14:44:11 +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
Move kRtpCsrcSize from common_types.h to rtp_headers.h
2019-05-10 09:43:39 +00:00
DEPS
Roll chromium_revision 25abf646db..4e6fe7f5f7 (667125:667532)
2019-06-10 08:33:32 +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
…
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
…
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
Do not run CheckNoStreamUsageIsAdded on tests.
2019-05-07 15:52:35 +00:00
pylintrc
Fixing py lint errors
2018-07-23 15:28:48 +00:00
README.chromium
…
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
Remove myself from OWNERS in a few places.
2019-06-10 07:57:46 +00:00
webrtc.gni
Testing no /DUNICODE assumptions with Win more configs bots.
2019-03-28 08:46:37 +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
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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