* Use webrtc-noise-gain instead of webrtcvad package
* Switching to ProcessedAudioChunk
* Refactor VAD and fix tests
* Add vad no chunking test
* Add test that runs audio enhancements
* Make arguments for async_pipeline_from_audio_stream keyword-only after hass
* Use a bytearray ring buffer
* Move generator outside
* Move stt stream generator outside
* Clean up execute
* Refactor VAD to use bytearray
* More tests
* Refactor chunk_samples to be more correct and robust
* Change AudioBuffer to use append instead of setitem
* Cleanup
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Co-authored-by: Paulus Schoutsen <balloob@gmail.com>