commit | fea41f540c72d15c4f499fead611a0c5e65db8ec | [log] [tgz] |
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author | Victor Boivie <boivie@webrtc.org> | Mon Mar 11 15:43:31 2024 |
committer | WebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com> | Wed Mar 13 15:44:17 2024 |
tree | a861272abf9e6edeeacc61c32f8e0f32a39039bf | |
parent | 2f3b75d30d1664cd9b0c2c7285c1c31974147c3d [diff] |
pc: Include SCTP queued bytes in buffered_amount Before this change, calling buffered_amount only included what was buffered on top of what was already buffered in the SCTP socket. With the defaults, the SCTP socket can buffer up to 2MB of data (that is not put on the wire) before the additional external bufferering in SctpDataChannel will be used. The buffering that I am working on removing completely. Until it's removed completely, to avoid the issue reported in crbug.com/41221056, include the bytes buffered in the SCTP socket to what is returned when calling RTCDataChannel::buffered_amount. This means that when this value is zero, it can be safe to know that all bytes have been sent, but not necessarily acknowledged. And calling close will not discard any messages. This is a stopgap solution, but as functional as the proper solution that removes all additional buffering. Follow-up CLs will merely improve this solution. Bug: chromium:41221056 Change-Id: I06edd52188d3bf13a17827381a15a4730722685a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/342520 Commit-Queue: Victor Boivie <boivie@webrtc.org> Reviewed-by: Florent Castelli <orphis@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41898}
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.