Of course, functions called WebRtcSpl_AddSatW32 and WebRtcSpl_SubSatW32 are supposed to handle overflow gracefully, and they probably did. But since the overflow handling depended on undefined behavior, a sufficiently smart optimizing compiler would have realized that it could just ignore the possibility of overflow and omit all the overflow handling code, leaving only the unadorned addition or subtraction. Also, the new implementations, unlike the old ones, result in branch-free code (tested with clang 3.9 with -O2). BUG=chromium:601728 Review-Url: https://codereview.webrtc.org/2002523002 Cr-Commit-Position: refs/heads/master@{#12846}
Revert of Remove Android x86 compilation trybot from CQ. (patchset #1 id:1 of https://codereview.webrtc.org/1959923002/ )
Revert of Added webrtc/base/safe_conversions.h as a pseudonym (patchset #1 id:20001 of https://codereview.webrtc.org/1774933003/ )
Revert "Revert of FrameBuffer for the new jitter buffer. (patchset #9 id:160001 of https://codereview.webrtc.org/1969403007/ )"
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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. 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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