Reason for revert: This CL are breaking some of the Android buildbots in Chromium. The CL will need be be revised to exclude the Android platform. Original issue's description: > Added a bitexactness test for the gain controller in the audio processing module. > > BUG=webrtc:5339 > > Committed: https://crrev.com/a49dc36976da44f3d6d75aed2dcab93fe14fc3a0 > Cr-Commit-Position: refs/heads/master@{#12124} TBR=henrik.lundin@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:5339 Review URL: https://codereview.webrtc.org/1829323002 Cr-Commit-Position: refs/heads/master@{#12127}
Revert of Added webrtc/base/safe_conversions.h as a pseudonym (patchset #1 id:20001 of https://codereview.webrtc.org/1774933003/ )
Revert of Revert opus memcheck suppression (patchset #1 id:1 of https://codereview.webrtc.org/1801233002/ )
Revert of Added a bitexactness test for the gain controller in the audio processing module. (patchset #3 id:60001 of https://codereview.webrtc.org/1812433002/ )
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. This page is maintained by the Google Chrome team.
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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