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chromium-webrtc-autoroll e333505d7f Roll chromium_revision b931c7fd8b..c27b8dde0e (693252:693394) 
Change log: b931c7fd8b..c27b8dde0e
Full diff: b931c7fd8b..c27b8dde0e

Changed dependencies
* src/base: a5a8025280..cc58e00ad3
* src/build: 340bc5e290..c02a50e0df
* src/ios: a34b43d924..024fbc8d42
* src/testing: 60f36f6977..2609e6c29c
* src/third_party: 39638c69d9..89733cb4dc
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/030358ed22..53a464d7b4
* src/third_party/depot_tools: f38bc17962..183971ca2a
* src/tools: 321808d348..d0b6ce04e9
DEPS diff: b931c7fd8b..c27b8dde0e/DEPS

No update to Clang.

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

Change-Id: If1c31595a9edf5b71adfc5caf0ed0a9da2be6457
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151500
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@{#29070}
2019-09-04 22:35:09 +00:00
api
Delete old version of PeerConnection::SetConfiguration
2019-09-04 08:23:18 +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
Add support for stable bitrate target in SvcRateAllocator
2019-09-04 14:22:43 +00:00
p2p
Use the sanitized pair when surfacing the candidate pair change event.
2019-09-03 17:17:49 +00:00
pc
Delete old version of PeerConnection::SetConfiguration
2019-09-04 08:23:18 +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 stable bitrate target in SvcRateAllocator
2019-09-04 14:22:43 +00:00
rtc_tools
Revert "Always create output_dir in setup_apprtc.py."
2019-09-04 18:19:12 +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
Add support for stable bitrate target in SvcRateAllocator
2019-09-04 14:22:43 +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
Change apprtc_webrtc_browsertest resource dir to avoid MAX_PATH.
2019-09-04 18:49:28 +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
…
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 b931c7fd8b..c27b8dde0e (693252:693394)
2019-09-04 22:35:09 +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
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
…
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
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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