commit | 1142e0bfb2eb27ba04de95e07c163680eb2287cc | [log] [tgz] |
---|---|---|
author | Ivo Creusen <ivoc@webrtc.org> | Thu Feb 04 13:03:44 2021 |
committer | Commit Bot <commit-bot@chromium.org> | Mon Feb 08 16:16:55 2021 |
tree | a16e6b5dd67141eff9e0b5b02370d749bd576be7 | |
parent | 68c225d76d007fc0297a18f8c4881e0bc885279b [diff] |
Avoid crashing on error code 6450 in isac. Isac is not able to effectively compress all types of signals. This should be extremely rare with real audio signals, but mostly happens with artificially created test signals. When this happens, we should avoid crashing and just carry on. Bug: chromium:1170167 Change-Id: I97b551fbbdcccb0186f3e6497991ac52d2301f68 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/205626 Commit-Queue: Ivo Creusen <ivoc@webrtc.org> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33193}
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 here for instructions on how to get started developing with the native code.
Authoritative list of directories that contain the native API header files.