Tomas Lundqvist b40c559858 Set voice RTCP mode based on the RemoteContent and not based on the LocalContent.
The RTCP mode is a send property for both send and receive channels. Send properties should be configured based on what peers support/prefer, which is described by the remote description (content).


Bug: webrtc:340041654
Change-Id: I18cd59e98aecfbbd8f4919b98381836184c10d77
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368980
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Lundqvist <tomasl@google.com>
Cr-Commit-Position: refs/heads/main@{#43449}
2024-11-25 14:06:39 +00:00
2024-11-12 10:04:10 +00:00
2024-10-31 15:31:38 +00:00
2024-11-14 14:36:58 +00:00
2022-12-02 09:21:47 +00:00
2023-09-25 15:56:09 +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
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