commit | a5ba250c262704441668fe89e439e89f3f41de5c | [log] [tgz] |
---|---|---|
author | Lennart Grahl <lennart.grahl@gmail.com> | Thu Oct 06 12:34:39 2022 |
committer | WebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com> | Mon Oct 10 12:24:52 2022 |
tree | d0eb04cc91a1483474f757a3c7a0a17ec6548334 | |
parent | 828ef91817f617c341ca3cae9680f29c8379f870 [diff] |
Fix apply frame transformer to MID demuxed audio streams Manually tested with libwebrtc built for Android and a solution running into the same problem as the linked repro in chromium:1348132. Bug: chromium:1348132 Change-Id: I88260b9ac72c67f1a6ad927e075d1490ac06ce91 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278241 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38335}
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.