philipel 6cbb468b6d In VideoReplayer, use MediaType::ANY when calling DeliverPacket with RTCP packet.
Bug: webrtc:14508
Change-Id: I402b35eb89d6e70122f9ff5bd51db0462c456f5d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/277621
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38269}
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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
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