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  • GStreamer
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  • !2341

webrtcdsp: Update code for webrtc-audio-processing-1

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Open Arun Raghavan requested to merge arun/gst-plugins-bad:webrtc-1.0 into master Jun 18, 2021
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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: !1850 (closed)

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Source branch: webrtc-1.0