Jakob Ivarsson 68f4e27794 Add RtpSender OnFirstPacketSent callback.
It works in the same way as the first packet received callback and can be used for latency measurements.

One important detail is that RTCP and probe packets are excluded from triggering the callback.

Bug: b/375148360
Change-Id: I5f99b565f96b622e864669cf227be5534aab0fc7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366644
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43309}
2024-10-25 16:17:04 +00:00
2024-10-18 16:33:17 +00:00
2024-10-22 11:36:58 +00:00
2024-07-11 20:26:16 +00:00
2024-10-15 20:10:24 +00:00
2022-12-02 09:21:47 +00:00
2023-09-25 15:56:09 +00:00
2024-10-14 12:13:31 +00:00
2024-05-27 12:46:57 +00:00

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.

Development

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.

More info

Description
The idea is to make CMake build for WebRTC m130 version - for audio processing module
Readme BSD-3-Clause 446 MiB
Languages
C++ 90.3%
Java 2.9%
C 2.2%
Objective-C++ 2%
Python 1.3%
Other 1%