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chromium-webrtc-autoroll 28330b5793 Roll chromium_revision 8cce5f8121..9868e92510 (796325:796451) 
Change log: 8cce5f8121..9868e92510
Full diff: 8cce5f8121..9868e92510

Changed dependencies
* src/base: f8948f2a5d..6e594ed76f
* src/build: 4e325d6559..abf3d46936
* src/ios: 44bc7ad8bf..1277bf7de8
* src/testing: 3fbb9c3e6e..05caa751d7
* src/third_party: 9a5350bb4d..4823540433
* src/third_party/perfetto: a4cba6578f..01e0ccdc9b
* src/tools: 41cabe8a5f..f23a0d4447
DEPS diff: 8cce5f8121..9868e92510/DEPS

No update to Clang.

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

Change-Id: I9111f610f6afca5128b4ccdd9c25e56a7e72aad1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181320
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@{#31899}
2020-08-10 18:40:20 +00:00
api
Revert "Implement transceiver.stop()"
2020-08-10 18:06:30 +00:00
audio
Added Error Checking in Ingress/Egress and Extra Unit Tests
2020-08-06 20:48:13 +00:00
build_overrides
set perfetto flag to default value of false
2020-07-22 10:14:53 +00:00
call
[Adaptation] Delete AdaptationListener
2020-08-10 17:43:56 +00:00
common_audio
Revert "Support AVX2/FMA intrinsics in Audio Resampler module"
2020-07-30 17:35:30 +00:00
common_video
Move function PrintVideoFrame to the test file where it is used.
2020-08-06 15:20:19 +00:00
data
Remove old data files.
2018-10-05 14:40:21 +00:00
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
Add missing tests for DTLS state logging in RTC event log.
2020-08-07 11:49:43 +00:00
media
Add new fmtp parameter for H.264
2020-08-07 10:32:41 +00:00
modules
[Adaptation] Move Balanced MinFpsDiff logic to VideoStreamAdapter
2020-08-10 15:56:07 +00:00
p2p
Delete left-over TODO comment
2020-08-07 11:36:51 +00:00
pc
Revert "Implement transceiver.stop()"
2020-08-10 18:06:30 +00:00
resources
iSAC API wrapper unit test fix
2020-02-27 14:27:23 +00:00
rtc_base
Update settings for balanced degradation.
2020-08-10 10:59:17 +00:00
rtc_tools
Reland "Moved VideoReceiveStream::Decoder::decoder_factory to VideoReceiveStream::Config::decoder_factory."
2020-08-06 11:50:08 +00:00
sdk
Revert "Implement transceiver.stop()"
2020-08-10 18:06:30 +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
Remove stat API from EmulatedEndpoint.
2020-08-06 12:53:33 +00:00
tools_webrtc
Fix clang revision regexp in chromium autoroller to match new format
2020-08-07 09:22:03 +00:00
video
[Adaptation] Remove ResourceAdaptation queue from VideoStreamEncoderResources
2020-08-10 18:08:56 +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
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
Changed AndroidVideoDecoder to also handle IllegalArgumentException and IllegalStateException during the init of the decoder and fallback to a software decoder
2020-08-05 09:41:49 +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
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
Replaces OverheadObserver with simple getter.
2020-05-07 17:33:45 +00:00
DEPS
Roll chromium_revision 8cce5f8121..9868e92510 (796325:796451)
2020-08-10 18:40:20 +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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