| commit | 4c556219e59f2ff3ea62da44e1ff81c6cd1f8021 | [log] [tgz] |
|---|---|---|
| author | Danil Chapovalov <danilchap@webrtc.org> | Mon Sep 04 10:27:59 2023 |
| committer | WebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com> | Mon Sep 04 11:27:42 2023 |
| tree | 175a224db42777c0c17b5b95fa768f5af578d2da | |
| parent | 8219cc3dc9b3b17730f551d303e86929503a4416 [diff] |
Cleanup RTPSenderAudio::SendAudio Combine all parameters into single struct so that it is easier to add and remove optional parameters Use Timestamp type instad of plain int to represent capture time Use rtc::ArrayView instead of pointer+size to represent payload Merge passing audio level into send function. Bug: webrtc:13757, webrtc:14870 Change-Id: I0386b710eb99b864334d61235add9abcde9bc69d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/317442 Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40688}
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.