webrtcdsp: Update code for webrtc-audio-processing-1 (!2942) · Merge requests · GStreamer / gstreamer · GitLab

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Arun Raghavan requested to merge arun/gstreamer:gst-plugins-bad-webrtc-1.0 into main Aug 25, 2022

Copied from https://gitlab.freedesktop.org//gstreamer/gst-plugins-bad/-/merge_requests/2341

Updated API usage appropriately, and now we have a versioned package to track breaking vs. non-breaking updates.

Deprecates a number of properties (and we have to plug in our own values for related enums which are now gone):

echo-suprression-level
experimental-agc
extended-filter
delay-agnostic
voice-detection-frame-size-ms
voice-detection-likelihood

Part-of: gst-plugins-bad!1850 (closed)