This reverts commit e68293553d1bd2504bf3ebf7f6769c3b01e640e3. Reason for revert: Breaks internal projects Original change's description: > Roll chromium_revision f52ff80530..4bafa509ea (501932:502960) > > Change log:f52ff80530..4bafa509ea> Full diff:f52ff80530..4bafa509ea> > Changed dependencies: > * src/base:a3848e11ae..6afcd86a32> * src/build:0119b14e14..67a664871e> * src/buildtools:cbc33b9c0a..26b7e66950> * src/ios:79eaa19c53..2cd231262b> * src/testing:e9068583fe..0c3fb670c3> * src/third_party:3c251d88b3..f56e199ae6> * src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/683ffbbe57..f51f273ee8 > * src/third_party/catapult:dd30f4f383..ab30bb20a8> * src/third_party/gtest-parallel:965cfdccf9..ee20273811> * src/third_party/libyuv:27036e33e8..5b1af9a335> * src/tools:02f1273b3b..4fa8837ea2> DEPS diff:f52ff80530..4bafa509ea/DEPS > > Clang version changed 312679:313222 > Details:f52ff80530..4bafa509ea/tools/clang/scripts/update.py > > TBR=buildbot@webrtc.org, > BUG=None > CQ_INCLUDE_TRYBOTS=master.internal.tryserver.corp.webrtc:linux_internal > > Change-Id: Iade6f94aca4b9d61cf545e80147268bbab5ee3ed > Reviewed-on: https://webrtc-review.googlesource.com/2280 > Commit-Queue: WebRTC Buildbot <buildbot@webrtc.org> > Reviewed-by: WebRTC Buildbot <buildbot@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#19904} TBR=buildbot@webrtc.org Change-Id: I358e1006a0f2708bb106eb8ee81118c3c8702fb4 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: None Cq-Include-Trybots: master.internal.tryserver.corp.webrtc:linux_internal Reviewed-on: https://webrtc-review.googlesource.com/2320 Reviewed-by: Charu Jain <charujain@webrtc.org> Commit-Queue: Charu Jain <charujain@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19905}
Revert of Fix the video buffer size should take rtt into consideration (patchset #3 id:40001 of https://codereview.chromium.org/2980413002/ )
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.
More info
- Official web site: http://www.webrtc.org
- Master source code repo: https://webrtc.googlesource.com/src
- Samples and reference apps: https://github.com/webrtc
- Mailing list: http://groups.google.com/group/discuss-webrtc
- Continuous build: http://build.chromium.org/p/client.webrtc
- Coding style guide
- Code of conduct
Description
The idea is to make CMake build for WebRTC m130 version - for audio processing module
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