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}
Name: WebRTC URL: http://www.webrtc.org Version: 90 License: BSD License File: LICENSE Description: WebRTC provides real time voice and video processing functionality to enable the implementation of PeerConnection/MediaStream. Third party code used in this project is described in the file LICENSE_THIRD_PARTY.