| commit | 920abcc9bccb2009b64fb59edeedcf17faf7689b | [log] [tgz] |
|---|---|---|
| author | Danil Chapovalov <danilchap@webrtc.org> | Wed Jul 26 10:33:06 2023 |
| committer | WebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com> | Thu Jul 27 14:35:42 2023 |
| tree | 66685841e85b52bcbd879783af93ab5d83089308 | |
| parent | b90cd919831112f95c5997782f6a88152a98dfc5 [diff] |
In RtpSenderVideo::UpdateConditionalRetransmit use typed time and framerate instead of plain ints Bug: webrtc:13757 Change-Id: If2df5418dacd2b95387fa74a9bc226426b207aee Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/313041 Reviewed-by: Åsa Persson <asapersson@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40483}
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.