Philipp Hancke af512281b1 audio: make packets lost a signed integer
as it is defined in RFC 3550. This avoids implicit casts
between signed and unsigned definitions.

BUG=webrtc:8626

Change-Id: I919b7c38ede1aa8d32f8e31b55660f540e5f5a6b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/279240
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#38522}
2022-11-01 11:46:49 +00:00
2022-10-31 15:43:16 +00:00
2022-08-29 21:04:32 +00:00
2022-10-31 14:51:25 +00:00
2022-10-10 15:51:33 +00:00
2022-10-10 10:18:37 +00:00
.gn
2022-09-14 08:49:56 +00:00
2022-02-20 14:22:13 +00:00
2021-12-08 08:53:00 +00:00
2022-05-13 09:01:34 +00:00
2022-09-30 13:50:49 +00:00
2022-10-14 08:53:38 +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%