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chromium-webrtc-autoroll 8577729a01 Roll chromium_revision 52323b9fe0..3a1d849d09 (691589:691713) 
Change log: 52323b9fe0..3a1d849d09
Full diff: 52323b9fe0..3a1d849d09

Changed dependencies
* src/base: c2fc6e22b1..728752022e
* src/build: bfbf6cdf58..a76fcfff9c
* src/ios: d703c43675..7aa9116ecf
* src/testing: c50c6e6fe8..1a68719d73
* src/third_party: 8fc5bda490..0d79d79bf7
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/9b593a3406..43bcbe5155
* src/tools: 2972a9a34a..9a273b7de4
DEPS diff: 52323b9fe0..3a1d849d09/DEPS

No update to Clang.

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

Change-Id: I5d5321448d212f142c45d91345f7d73bd75ebf6c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150862
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@{#29013}
2019-08-29 18:33:49 +00:00
api
Remove backwards compatibility names from api/uma_metrics.h.
2019-08-29 13:35:56 +00:00
audio
Delete unneeded dependencies on libjingle_peerconnection_api
2019-08-29 10:52:42 +00:00
build_overrides
Remove crbug.com/904400 workaround.
2019-03-15 18:36:23 +00:00
call
Remove lock for process thread pointer from PacedSender.
2019-08-29 17:08:24 +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
Make the RtpHeaderParserImpl available to tests and tools only.
2019-08-29 15:56:40 +00:00
modules
Remove lock for process thread pointer from PacedSender.
2019-08-29 17:08:24 +00:00
p2p
Add support for RTCTransportStats.selectedCandidatePairChanges
2019-08-28 13:22:08 +00:00
pc
Refactor SCTP data channels to use DataChannelTransportInterface.
2019-08-29 17:30:27 +00:00
resources
Use the AEC3 high-pass filter for the whole APM
2019-08-23 20:04:10 +00:00
rtc_base
Fix HexEncodeTest.TestZeroLengthNoDelimiter with enable_iterator_debugging=true
2019-08-29 14:16:23 +00:00
rtc_tools
Increased event log visualizer RTP clock estimation tolerance.
2019-08-29 10:08:43 +00:00
sdk
Allows configuration of playout audio buffer
2019-08-29 12:57:14 +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
Refactor SCTP data channels to use DataChannelTransportInterface.
2019-08-29 17:30:27 +00:00
tools_webrtc
Add license for android_ndk
2019-08-28 14:57:28 +00:00
video
Make the RtpHeaderParserImpl available to tests and tools only.
2019-08-29 15:56:40 +00:00
.clang-format
Tune ObjC clang-format configuration
2017-05-11 09:14:18 +00:00
.git-blame-ignore-revs
Let git-hyper-blame ignore new format cleanup.
2019-07-11 16:18:51 +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
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 52323b9fe0..3a1d849d09 (691589:691713)
2019-08-29 18:33:49 +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
Update template to follow chromium copyright style
2013-04-24 01:01:28 +00:00
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
Remove myself from OWNERS in a few places.
2019-06-10 07:57:46 +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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