Jakob Ivarsson dcb09ff218 Reset encoder when audio send stream is stopped.
This is to clear any remaining buffers and other state such as the next packet timestamp.

Bug: webrtc:12397
Change-Id: I2ef9a6f7254d82a69a2896ec8aa619ced2694fb8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291327
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39206}
2023-01-26 15:20:02 +00:00
2023-01-23 14:00:21 +00:00
2021-01-20 15:01:07 +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
2022-12-02 09:21:47 +00:00
2022-12-02 09:21:47 +00:00
2022-05-13 09:01:34 +00:00
2020-07-13 11:42:07 +00:00
2022-11-17 21:29:53 +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
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