commit | 2ace3f94064e4d1980483d9ea3b0da8537aee540 | [log] [tgz] |
---|---|---|
author | peah <peah@webrtc.org> | Sat Sep 10 11:42:27 2016 |
committer | Commit bot <commit-bot@chromium.org> | Sat Sep 10 11:42:36 2016 |
tree | b0b6839185b610a99909855c6634d50f0383a22a | |
parent | 1d02d3e5e68e5025ddf8ab3af11e416778cd0201 [diff] |
The audio processing module (APM) relies on two for functionalities doing sample-rate conversions: -The implicit resampling done in the AudioBuffer CopyTo, CopyFrom, InterleaveTo and DeinterleaveFrom methods. -The multi-band splitting scheme. The selection of rates in these have been difficult and complicated, partly due to that the APM API which allows for activating the APM submodules without notifying the APM. This CL adds functionality that for each capture frame polls all submodules for whether they are active or not and compares this against a cached result. Furthermore, new functionality is added that based on the results of the comparison do a reinitialization of the APM. This has several advantages -The code deciding on whether to analysis and synthesis is needed for the bandsplitting can be much simplified and centralized. -The selection of the processing rate can be done such as to avoid the implicit resampling that was in some cases unnecessarily done. -The optimization for whether an output copy is needed that was done to improve performance due to the implicit resampling is no longer needed, which simplifies the code and makes it less error-prone in the sense that is no longer neccessary to keep track of whether any module has changed the signal. Finally, it should be noted that the polling of the state for all the submodules was done previously as well, but in a less obvious and distributed manner. BUG=webrtc:6181, webrtc:6220, webrtc:5298, webrtc:6296, webrtc:6298, webrtc:6297 Review-Url: https://codereview.webrtc.org/2304123002 Cr-Commit-Position: refs/heads/master@{#14175}
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.
See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.