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chromium-webrtc-autoroll 523070a0ae Roll chromium_revision 9c0fc3b7d7..2f197110b2 (795492:795842) 
Change log: 9c0fc3b7d7..2f197110b2
Full diff: 9c0fc3b7d7..2f197110b2

Changed dependencies
* src/base: 12eee072f9..c97d881cf5
* src/build: 7fbdf1ef42..80633facbe
* src/ios: c652c471d4..402e627cca
* src/testing: ceee9771da..37aa16bc92
* src/third_party: c78ccdcf07..396223229d
* src/third_party/breakpad/breakpad: 3d8daa2c74..e3a62dc550
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/32be452338..c4d3ff474a
* src/third_party/depot_tools: 8351dc1fb7..24289f2e94
* src/third_party/perfetto: 76488e6d8e..e69d5b2de9
* src/tools: 713e3e9aba..fd6ecf7bef
DEPS diff: 9c0fc3b7d7..2f197110b2/DEPS

Clang version changed 1bd7046e4ce0102adef6096a12a289d7f94b8c73:llvmorg-12-init-1771-g1bd7046e
Details: 9c0fc3b7d7..2f197110b2/tools/clang/scripts/update.py

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

Change-Id: Ie911caf2a0106878e61c19eb4b538a874893c221
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181028
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@{#31880}
2020-08-07 11:37:56 +00:00
api
Add logging of decoded video frames.
2020-08-06 17:33:24 +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
red: fix SendCodecSpec == operator
2020-08-06 14:19:34 +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 logging of decoded video frames.
2020-08-06 17:33:24 +00:00
media
Add new fmtp parameter for H.264
2020-08-07 10:32:41 +00:00
modules
Add new fmtp parameter for H.264
2020-08-07 10:32:41 +00:00
p2p
Delete left-over TODO comment
2020-08-07 11:36:51 +00:00
pc
Make Android/iOS local/remote description accessors thread safe.
2020-08-05 22:34:46 +00:00
resources
iSAC API wrapper unit test fix
2020-02-27 14:27:23 +00:00
rtc_base
Add interface for os/firmware to affect ICE selection.
2020-08-07 10:07:43 +00:00
rtc_tools
Reland "Moved VideoReceiveStream::Decoder::decoder_factory to VideoReceiveStream::Config::decoder_factory."
2020-08-06 11:50:08 +00:00
sdk
Add interface for os/firmware to affect ICE selection.
2020-08-07 10:07:43 +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
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
Add new fmtp parameter for H.264
2020-08-07 10:32:41 +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 9c0fc3b7d7..2f197110b2 (795492:795842)
2020-08-07 11:37:56 +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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C 2.2%
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