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chromium-webrtc-autoroll f5bbd1f068 Roll chromium_revision c145d4805f..73b2f9fef8 (752799:753151) 
Change log: c145d4805f..73b2f9fef8
Full diff: c145d4805f..73b2f9fef8

Changed dependencies
* src/base: 32ccf425fa..0795c973af
* src/build: 8d56e63759..88a5c9e25f
* src/ios: ac693a241c..9e0b58d90f
* src/testing: 7dd30f951a..ff5b7da935
* src/third_party: 8f0649bced..fb585927dc
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/341e229c3d..054d61d117
* src/third_party/depot_tools: 8f41598abf..7c5efb2a7e
* src/tools: 58560b651b..516912e509
DEPS diff: c145d4805f..73b2f9fef8/DEPS

Clang version changed 5d881dd8a8b8ea6f80bf4ef5b900ca006dacd9bf:a1762f9ceb9549b781b7418c7dbe23fe620648f6
Details: c145d4805f..73b2f9fef8/tools/clang/scripts/update.py

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

Change-Id: Ia5c3f6ceb7c7b4c326d5704f727ebc76c34b5339
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171700
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@{#30875}
2020-03-25 08:46:37 +00:00
api
Add optional output audio file to NetEq simulation API
2020-03-24 16:31:08 +00:00
audio
Split RMS level measurement utility from APM
2020-03-23 16:54:22 +00:00
build_overrides
Purge phoglund from most OWNERS files.
2020-03-09 14:08:30 +00:00
call
[Stats] Explicit RTP-RTX and RTP-FEC mappings. Unblocks simulcast stats.
2020-03-24 13:31:54 +00:00
common_audio
Changed fft4g to be built as C++
2020-03-20 12:10:16 +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
[Stats] Explicit RTP-RTX and RTP-FEC mappings. Unblocks simulcast stats.
2020-03-24 13:31:54 +00:00
modules
Hide Av1 specfic logic from RtpVideoReceiver into depacketizer interface.
2020-03-24 15:55:00 +00:00
p2p
remove deprecated fields in rtc::NetworkRoute
2020-03-25 06:44:06 +00:00
pc
Refactor ssl_stream_adapter API to show object ownership
2020-03-21 18:53:46 +00:00
resources
iSAC API wrapper unit test fix
2020-02-27 14:27:23 +00:00
rtc_base
remove deprecated fields in rtc::NetworkRoute
2020-03-25 06:44:06 +00:00
rtc_tools
Remove RtpPacketSendInfo.has_rtp_sequence_number
2020-03-23 18:34:43 +00:00
sdk
Revert "Remove bitratePriority from the Obj-C RTCRtpEncodingParameters wrapper."
2020-03-24 10:48:26 +00:00
stats
Add jitterBufferTargetDelay as RTCNonStandardStatsMember to new GetStats API
2020-03-11 12:08:32 +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 RTC_EXPORT to webrtc::Clock
2020-03-10 23:00:21 +00:00
test
Hide Av1 specfic logic from RtpVideoReceiver into depacketizer interface.
2020-03-24 15:55:00 +00:00
tools_webrtc
Fix mb.py when using isolate archive
2020-03-19 08:49:57 +00:00
video
Hide Av1 specfic logic from RtpVideoReceiver into depacketizer interface.
2020-03-24 15:55:00 +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 .clangd to .gitignore
2019-10-28 12:27:50 +00:00
.gn
Reenable libaom decoder by default
2020-03-18 18:04:41 +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
Fix bad frees in error paths of WebRtcAecm_Create
2020-03-21 23:01:47 +00:00
BUILD.gn
Fix 'all' build on non Android platforms.
2020-03-11 08:24:27 +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
Reland "Remove PlayoutDelayOracle and make RtpSenderVideo guarantee delivery"
2020-02-07 08:23:58 +00:00
DEPS
Roll chromium_revision c145d4805f..73b2f9fef8 (752799:753151)
2020-03-25 08:46:37 +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 #COMPONENT to WebRTC.
2019-10-08 12:20:39 +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
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
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
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%