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chromium-webrtc-autoroll fa068336f9 Roll chromium_revision a0d7df3386..4d555ede52 (754491:754603) 
Change log: a0d7df3386..4d555ede52
Full diff: a0d7df3386..4d555ede52

Changed dependencies
* src/base: 680d009a82..11978dc67d
* src/build: d9d6d0b425..2c249ccb22
* src/ios: fb0239794d..ffc5b22ef1
* src/testing: bcf855500c..7737e73854
* src/third_party: cba3c14889..b7263f3723
* src/third_party/depot_tools: 6a7e234b58..11f4a84bb1
* src/tools: fdde8ed2b0..0389dabc90
DEPS diff: a0d7df3386..4d555ede52/DEPS

No update to Clang.

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

Change-Id: I2e83c7ebdbb471fb2e695c6be6e620e0c383c5cf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172460
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@{#30944}
2020-03-30 21:15:24 +00:00
api
Introduce TransformableFrameInterface.
2020-03-30 13:35:26 +00:00
audio
Remove the histogram flag and all Chart JSON code.
2020-03-28 13:44:43 +00:00
build_overrides
Purge phoglund from most OWNERS files.
2020-03-09 14:08:30 +00:00
call
Allow setting a bandwidth cap for relayed connections.
2020-03-26 20:41:46 +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
…
docs
Rename index.md to README.md to make it automatically show up
2020-03-06 10:43:51 +00:00
examples
Chromium refactor: Replace "resources_dirs" with "sources"
2020-03-25 19:32:15 +00:00
logging
Replace DataSize and DataRate factories with newer versions
2020-02-18 16:09:50 +00:00
media
Reland "Distinguish between send and receive codecs"
2020-03-29 21:03:27 +00:00
modules
Introduce TransformableFrameInterface.
2020-03-30 13:35:26 +00:00
p2p
Enforce "comprehension-required" STUN rules.
2020-03-28 02:07:49 +00:00
pc
Reland "Distinguish between send and receive codecs"
2020-03-29 21:03:27 +00:00
resources
iSAC API wrapper unit test fix
2020-02-27 14:27:23 +00:00
rtc_base
Add test for relay bandwidth capping.
2020-03-30 13:02:46 +00:00
rtc_tools
Add packet rate plots to event_log_visualizer.
2020-03-30 14:46:41 +00:00
sdk
Remove IP address logging from NetworkInformation
2020-03-25 09:52:27 +00:00
stats
Add jitterBufferTargetDelay as RTCNonStandardStatsMember to new GetStats API
2020-03-11 12:08:32 +00:00
style-guide
…
system_wrappers
Add RTC_EXPORT to webrtc::Clock
2020-03-10 23:00:21 +00:00
test
Remove the histogram flag and all Chart JSON code.
2020-03-28 13:44:43 +00:00
tools_webrtc
Revert "Trigger CI bots."
2020-03-28 22:37:03 +00:00
video
Introduce TransformableFrameInterface.
2020-03-30 13:35:26 +00:00
.clang-format
…
.git-blame-ignore-revs
…
.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
Audio egress implementation for initial voip api in api/voip.
2020-03-27 18:45:43 +00:00
CODE_OF_CONDUCT.md
…
codereview.settings
…
common_types.h
Reland "Remove PlayoutDelayOracle and make RtpSenderVideo guarantee delivery"
2020-02-07 08:23:58 +00:00
DEPS
Roll chromium_revision a0d7df3386..4d555ede52 (754491:754603)
2020-03-30 21:15:24 +00:00
ENG_REVIEW_OWNERS
…
LICENSE
…
license_template.txt
…
native-api.md
…
OWNERS
Remove phoglund as root owner.
2020-03-30 12:15:56 +00:00
PATENTS
…
presubmit_test_mocks.py
…
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
…
README.chromium
…
README.md
…
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
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C 2.2%
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Python 1.3%
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