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Mirko Bonadei fa5ad8c0b5 Revert "AV1: change update freq and disable denoiser explicitly."
This reverts commit abf5701c37.

Reason for revert: Breaks downstream tests.

Original change's description:
> AV1: change update freq and disable denoiser explicitly.
>
> Change speed/thread settings for faster encoding.
>
> Change-Id: I74d93eac26ae8700a48c437fe235643810de1ca0
> Bug: None
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206480
> Reviewed-by: Marco Paniconi <marpan@webrtc.org>
> Reviewed-by: Marco Paniconi <marpan@google.com>
> Commit-Queue: Jerome Jiang <jianj@google.com>
> Cr-Commit-Position: refs/heads/master@{#33208}

TBR=jianj@google.com,marpan@google.com,marpan@webrtc.org

Change-Id: I47b65e1c78ccb055238a44886dac87f8fc2f5330
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: None
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206644
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33213}
2021-02-10 09:30:10 +00:00
api peerconnection: measure bundle policy usage 2021-02-09 17:09:46 +00:00
audio Reland "Fix data race for config_ in AudioSendStream" 2021-02-04 12:33:56 +00:00
build_overrides [build] Remove obsolete gn flag 2021-01-11 17:57:44 +00:00
call Update WebRTC code version (2021-02-09T04:03:22). 2021-02-09 05:57:33 +00:00
common_audio Remove from chromium build targets that are not compatible with it. 2021-02-01 13:46:19 +00:00
common_video Replace rtc::ThreadChecker with webrtc::SequenceChecker 2021-02-02 14:56:27 +00:00
data Remove old data files. 2018-10-05 14:40:21 +00:00
docs doc: update working-with-native-branches information 2020-12-16 09:02:46 +00:00
examples Replace rtc::ThreadChecker with webrtc::SequenceChecker 2021-02-02 14:56:27 +00:00
logging Parse and plot RTCP BYE in RTC event log. 2021-02-04 11:28:46 +00:00
media Reland "Prepare to avoid hops to worker for network events." 2021-02-03 17:44:47 +00:00
modules Revert "AV1: change update freq and disable denoiser explicitly." 2021-02-10 09:30:10 +00:00
p2p Delete unused sigslot variables. 2021-02-09 14:41:14 +00:00
pc peerconnection: measure bundle policy usage 2021-02-09 17:09:46 +00:00
resources Remove from chromium build targets that are not compatible with it. 2021-02-01 13:46:19 +00:00
rtc_base Replace some RecursiveCriticalSection with Mutex, in PhysicalSocketServer. 2021-02-09 15:49:28 +00:00
rtc_tools Add more refined control over dumping of data and the aecdump content 2021-02-06 00:36:10 +00:00
sdk Replace casted uses of [OCMArg anyPointer] with [OCMArg anyObjectRef]. 2021-02-10 08:56:17 +00:00
stats Remove from chromium build targets that are not compatible with it. 2021-02-01 13:46:19 +00:00
style-guide Remove kwiberg@webrtc.org from OWNERS files 2020-12-04 15:11:26 +00:00
system_wrappers Remove from chromium build targets that are not compatible with it. 2021-02-01 13:46:19 +00:00
test Revert "Remove thread hops from events provided by JsepTransportController." 2021-02-09 12:26:26 +00:00
tools_webrtc Roll chromium_revision 4231e93dbb..b86bc4feb4 (850549:852112) 2021-02-09 10:34:08 +00:00
video Reland "Enable Video-QualityScaling experiment by default" 2021-02-05 09:49:13 +00:00
.clang-format Add IncludeBlocks to clang-format. 2021-02-03 16:29:07 +00:00
.git-blame-ignore-revs Let git-hyper-blame ignore new format cleanup. 2019-07-11 16:18:51 +00:00
.gitignore Add .cache to .gitignore. 2021-01-20 15:01:07 +00:00
.gn Rename PlayoutDelay --> VideoPlayoutDelay, move to api/video/video_timing.h 2020-09-07 08:37:14 +00:00
.vpython Reland "Add protobuf-py2_py3 3.13.0 to .vpython." 2020-11-20 07:52:26 +00:00
abseil-in-webrtc.md Polish the "Using Abseil in WebRTC" docs 2020-10-16 13:42:00 +00:00
AUTHORS changed src\modules\audio_device\win\audio_device_core_win.cc , and it is working 2021-02-04 09:31:33 +00:00
BUILD.gn Move SequenceChecker header to API: step 1, move header only 2021-02-08 11:49:58 +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
DEPS Roll chromium_revision b86bc4feb4..1e9271a080 (852112:852360) 2021-02-09 23:40:16 +00:00
DIR_METADATA Move metadata in OWNERS files to DIR_METADATA files. 2021-02-08 19:09:33 +00:00
ENG_REVIEW_OWNERS Remove kwiberg@webrtc.org from OWNERS files 2020-12-04 15:11:26 +00:00
LICENSE Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
license_template.txt
native-api.md Make the remote_bitrate_estimator build target private 2020-11-26 12:21:22 +00:00
OWNERS Move metadata in OWNERS files to DIR_METADATA files. 2021-02-08 19:09:33 +00:00
PATENTS Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
PRESUBMIT.py PRESUBMIT.py: Use input_api.owners_client to check DEPS includes owners. 2021-02-09 21:17:44 +00:00
presubmit_test.py Reformat python files checked by pylint (part 1/2). 2020-10-30 10:13:11 +00:00
presubmit_test_mocks.py Reformat python files checked by pylint (part 1/2). 2020-10-30 10:13:11 +00:00
pylintrc Undo enforcing of PEP-8 pylint changes for method and function names. 2020-11-10 18:26:25 +00:00
README.chromium Add CPEPrefix. 2020-07-13 11:42:07 +00:00
README.md doc: move bug reporting instructions to the repository 2020-10-21 14:47:49 +00:00
style-guide.md C++ style: We don't allow designated initializers 2020-06-03 09:11:09 +00:00
WATCHLISTS Add hta@ to rtc_base/ and api/ WATCHLISTS. 2021-01-06 09:43:34 +00:00
webrtc.gni Make PipeWire 0.3 default version 2021-02-04 11:17:55 +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
whitespace.txt Just adding my message in whitespace. 2021-02-04 12:40:16 +00:00

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