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chromium-webrtc-autoroll 358df5fb4d Roll chromium_revision adcdc4730c..5963c7d69a (794031:794142) 
Change log: adcdc4730c..5963c7d69a
Full diff: adcdc4730c..5963c7d69a

Changed dependencies
* src/base: d531067a81..bdf691693c
* src/build: ae943e3e61..edf75c7295
* src/buildtools: 42e9461958..6139217878
* src/ios: 88d7ebbdb0..aadb08b0e6
* src/testing: 596969c21e..e7d0644589
* src/third_party: a0c94917b3..aa574c831b
* src/third_party/perfetto: fce06b2000..4b656a010b
* src/tools: 61751efb0a..3469aa4447
DEPS diff: adcdc4730c..5963c7d69a/DEPS

No update to Clang.

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

Change-Id: Idbd586672de96bfea43b61b6ecc56b123f7a1aef
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180771
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@{#31837}
2020-08-03 16:48:13 +00:00
api
Add EncodedImageCallback::OnEncodedImage without RTPFragmentationHeader
2020-08-03 13:58:17 +00:00
audio
Remove unused critical section includes.
2020-07-16 13:52:28 +00:00
build_overrides
set perfetto flag to default value of false
2020-07-22 10:14:53 +00:00
call
Revert "Moved VideoReceiveStream::Decoder::decoder_factory to VideoReceiveStream::Config::decoder_factory."
2020-08-03 15:45:41 +00:00
common_audio
Revert "Support AVX2/FMA intrinsics in Audio Resampler module"
2020-07-30 17:35:30 +00:00
common_video
Ignore RTPFragmentationHeader when rewriting H264 SPS
2020-07-27 09:42:25 +00:00
data
…
docs
Search and replace gendered terms according to style guide:
2020-06-12 14:12:54 +00:00
examples
Implemented Android Demo Application for VoIP API
2020-07-21 16:34:22 +00:00
logging
Search and replace gendered terms according to style guide:
2020-06-12 14:12:54 +00:00
media
Revert "Moved VideoReceiveStream::Decoder::decoder_factory to VideoReceiveStream::Config::decoder_factory."
2020-08-03 15:45:41 +00:00
modules
AudioDeviceMac: fix mutex re-entry.
2020-08-03 10:54:56 +00:00
p2p
Reland "Implement packets_(sent | received) for RTCTransportStats"
2020-07-10 11:50:59 +00:00
pc
Reland "Pass NetworkMonitorFactory through PeerConnectionFactory."
2020-08-01 00:36:27 +00:00
resources
iSAC API wrapper unit test fix
2020-02-27 14:27:23 +00:00
rtc_base
Delete old name ProcessAllMessageQueues
2020-08-03 11:24:17 +00:00
rtc_tools
Revert "Moved VideoReceiveStream::Decoder::decoder_factory to VideoReceiveStream::Config::decoder_factory."
2020-08-03 15:45:41 +00:00
sdk
Stop using RTPFragmentationHeader in sdk/android as unneeded
2020-08-03 15:57:43 +00:00
stats
Reland "Implement packets_(sent | received) for RTCTransportStats"
2020-07-10 11:50:59 +00:00
style-guide
…
system_wrappers
Revert "Support AVX2/FMA intrinsics in Audio Resampler module"
2020-07-30 17:35:30 +00:00
test
Revert "Moved VideoReceiveStream::Decoder::decoder_factory to VideoReceiveStream::Config::decoder_factory."
2020-08-03 15:45:41 +00:00
tools_webrtc
Fix "Assignment had no effect." error during chromium roll.
2020-07-30 12:57:20 +00:00
video
Revert "Moved VideoReceiveStream::Decoder::decoder_factory to VideoReceiveStream::Config::decoder_factory."
2020-08-03 15:45:41 +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
Use absl_deps in order to preapre to the Abseil component build release.
2020-06-08 12:59:40 +00:00
AUTHORS
Explicitly set presentation style to full screen, because,
2020-07-17 07:15:36 +00:00
BUILD.gn
Revert "Support AVX2/FMA intrinsics in Audio Resampler module"
2020-07-30 17:35:30 +00:00
CODE_OF_CONDUCT.md
…
codereview.settings
…
common_types.h
Replaces OverheadObserver with simple getter.
2020-05-07 17:33:45 +00:00
DEPS
Roll chromium_revision adcdc4730c..5963c7d69a (794031:794142)
2020-08-03 16:48:13 +00:00
ENG_REVIEW_OWNERS
…
LICENSE
…
license_template.txt
…
native-api.md
Make transient suppression optionally excludable via defines
2020-04-02 11:44:07 +00:00
OWNERS
Remove phoglund as root owner.
2020-03-30 12:15:56 +00:00
PATENTS
…
presubmit_test_mocks.py
Inclusive language in PRESUBMIT.py.
2020-07-22 10:01:23 +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
Inclusive language in PRESUBMIT.py.
2020-07-22 10:01:23 +00:00
pylintrc
…
README.chromium
Add CPEPrefix.
2020-07-13 11:42:07 +00:00
README.md
Fix link in documentation. (take 2)
2020-04-16 11:08:43 +00:00
style-guide.md
C++ style: We don't allow designated initializers
2020-06-03 09:11:09 +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
Revert "Support AVX2/FMA intrinsics in Audio Resampler module"
2020-07-30 17:35:30 +00:00
whitespace.txt
Trigger CI bots.
2020-07-15 17:50:55 +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 here 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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