commit | d82a02c837d33cdfd75121e40dcccd32515e42d6 | [log] [tgz] |
---|---|---|
author | Per Åhgren <peah@webrtc.org> | Thu Mar 12 10:53:30 2020 |
committer | Commit Bot <commit-bot@chromium.org> | Thu Mar 12 12:23:20 2020 |
tree | 65a852402a426bad385ebb592ea95cc803be264b | |
parent | c71be24c82e6788f486ac91866aec1d3124f8efe [diff] |
ACM: Corrected temporary buffer size This CL corrects the temporary buffers size in the pre-processing of the capture audio before encoding. As part of this it removes the ACM-specific hardcoding of the size and instead ensures that the size of the temporary buffer matches that of the AudioFrame. Bug: webrtc:11242 Change-Id: I56dd6cadfd4e140e8e159966c33d1027383ea9fa Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170340 Commit-Queue: Per Åhgren <peah@webrtc.org> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30775}
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.
See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.
Authoritative list of directories that contain the native API header files.