commit | e1b685a50a578f18aa9b023506e87a86dbeecf11 | [log] [tgz] |
---|---|---|
author | Florent Castelli <orphis@webrtc.org> | Fri Apr 30 17:11:37 2021 |
committer | WebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com> | Fri Apr 30 18:55:47 2021 |
tree | dd7aaa24156060a5462f967f19cbd492782149a7 | |
parent | c9716e0c71d7aa73bcfca0a0225bd6faddb6e650 [diff] |
simulcast: Limit audio transceivers to single stream We don't support audio simulcast, so we should reject the layers early during an addTransceiver() call. Bug: webrtc:12719 Change-Id: Ieeb92c66de741e9b11943e0173a6f2e052926f13 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/216685 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33886}
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.