| commit | f009e38fe0cf7a6d25e0b5964296490cdb72224f | [log] [tgz] |
|---|---|---|
| author | Guy Hershenbaum <hershi@fb.com> | Mon Aug 19 16:06:12 2024 |
| committer | WebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com> | Tue Aug 20 16:22:04 2024 |
| tree | 8664a299fe3e5f6d6610e42439057f93ef2251e2 | |
| parent | f2d31361d9975cdf291ce41c11d87613a0ff1596 [diff] |
Fix AudioSendStream reconfigure - stop processing during unconfigured state When Reconfiguring there's a call to ResetSenderCongestionControlObjects followed by a later call to RegisterSenderCongestionControlObjects which happen on the worker thread, while enqueuing packets is happening on a different thread. If packets are enqueued in between these calls we get a null dereference of the `rtp_packet_pacer_` This change fixes it by pausing processing of incoming audio in the interim state Bug: webrtc:358290775 Change-Id: I77cebfb131545ce2a6fdb26105dd999da3f7c443 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/359080 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42815}
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.