Reason for revert: Re-land the original CL because the reverting it didn't fix the problem. Original issue's description: > Revert of Make AudioSinkInterface::Data hold a const pointer to the audio buffer. (patchset #1 id:1 of https://codereview.webrtc.org/2873803002/ ) > > Reason for revert: > Reverted because it possibly breaks the internal project. > > Original issue's description: > > Make AudioSinkInterface::Data hold a const pointer to the audio buffer. > > > > This is in preparation for https://codereview.webrtc.org/2750783004/, where > > requiring a non-const pointer for AudioSinkInterface would force an unmuting > > and zeroing of every frame. > > > > BUG=webrtc:7343 > > > > Review-Url: https://codereview.webrtc.org/2873803002 > > Cr-Commit-Position: refs/heads/master@{#18107} > > Committed:38605965bd> > TBR=solenberg@webrtc.org,henrik.lundin@webrtc.org,yujo@chromium.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:7343 > > Review-Url: https://codereview.webrtc.org/2877013002 > Cr-Commit-Position: refs/heads/master@{#18112} > Committed:c904634823TBR=solenberg@webrtc.org,henrik.lundin@webrtc.org,yujo@chromium.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:7343 Review-Url: https://codereview.webrtc.org/2880663003 Cr-Commit-Position: refs/heads/master@{#18113}
Revert of CQ: Remove Linux ARM64 Debug trybot from default set. (patchset #1 id:1 of https://codereview.webrtc.org/2790263003/ )
Reland of Make AudioSinkInterface::Data hold a const pointer to the audio buffer. (patchset #1 id:1 of https://codereview.webrtc.org/2877013002/ )
Reland of Enable GN check for webrtc/base (patchset #3 id:230001 of https://codereview.webrtc.org/2838683002/ )
Reland of Moving webrtc.gni up one level from build/ (patchset #1 id:1 of https://codereview.webrtc.org/2657563002/ )
Reland of PyLint fixes for tools-webrtc and webrtc/tools (patchset #1 id:1 of https://codereview.webrtc.org/2737233003/ )
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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://chromium.googlesource.com/external/webrtc
- 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
Description
The idea is to make CMake build for WebRTC m130 version - for audio processing module
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