Tomas Gunnarsson ac72dda645 Code consolidation in BaseChannel and derived classes.
This is a bit of refactoring to clear the way for some more upcoming
changes and fix little oddities here and there that are basically
artifacts of many small incremental changes throughout the years.

* Remove the CryptoOptions member variable and instead only keep around
  the filter for rtp header extensions.
* Remove several member methods that only forwarded calls to
  media_channel() and effectively reduced readability.
* Consolidated quite a bit of code related to UpdateRemoteStreams_w
  and the copy/pasted code in the Video/Voice classes around calling it.
* UpdateRemoteStreams_w now returns an error when it's encountered.
  Before, an error would still be returned in those cases but all
  operations were unnecessarily performed.

Bug: none
Change-Id: I85a37b9e8f00584aa794abef11abfe89dec5d0a6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/244098
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35637}
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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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