4637b6afca
(Or, in less flattering terms, fixing a performance issue introduced a few months ago by me). In GN release mode (is_debug = false), the version of the mixer code before this CL generated code that multiplied each sample (tens of thousands/second for each input stream) with a floating point number. This number is almost always exactly 1.0f. The only situation when it's not 1 is when an audio steam is added or removed. For one input stream early return leads to a 30% improvement of audio mixing time profiled on x86-64 under a release build (is_debug = false, enable_profiling, enable_full_stack_frames_for_profiling) with 16kHz and no APM limiter. There can be up to 3 streams. BUG=chromium:687502 Review-Url: https://codereview.webrtc.org/2659423002 Cr-Commit-Position: refs/heads/master@{#16396}
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