vosk-api VS NAudio

So I don't know if my issue comes from my lack of knowledge of discord.js/voice or VOSK. so I guess the most important thing I need to see is if I am creating a proper stream for the Vosk API to capture the audio. if I can figure out how to capture an audio stream I can probably import that in to vosk and figure out how to use vosk myself. but right now I can't even get close! Thank you in advance...Sorry if this isn't the right place for this

I remember a while ago checking out the issues with Vosk speech recognition (written in C). A handful of it's issues are related to segfaults and null pointers.

first, good initiative! thanks for sharing. i think you gotta be more diligent and careful with the problem statement.

checking the weather in Sofia, Bulgaria requires cloud, current information. it's not "random speech". ESP SR capability issues don't mean that you cannot process it locally.

the comment was on "voice processing" i.e. sending speech to the cloud, not sending a call request to get the weather information.

besides, local intent detection, beyond 400 commands, there are great local STT options, working better than most cloud STTs for "random speech"

https://github.com/alphacep/vosk-api

I did a one-off text to speech tool for someone last year and had pretty good results with VOSK. One upside is that it works offline, although I imagine if you use TTS a lot you'll notice issues I didn't.

You can use vosk-api (https://github.com/alphacep/vosk-api) to listen to your audio, transform it to text, and then post the text to GPT-3, then using the vector sdk, have your responses said by vector.

The set up script wants to download https://github.com/alphacep/vosk-api/releases/download/v0.3.45/vosk-model-en-v0.3.45.zip, but this resource is not found. AFAICT all releases never contained a model file. Remedy: hardcode one model from https://alphacephei.com/vosk/models. I guessed and picked the one with the closest name, vosk-model-en-us-0.22.zip, just so I could continue.

Thanks for the suggestion. I have to rewrite most of the audio signal processing code anyway as I am using NAudio currently. Have heard of PiSound before, and will definitely check it out as it is pretty popular. I just need a way to output an array of numbers, representing the processed sound wave, to turn into an analog signal. Writing this from scratch is unnecessarily complicated when these libraries already exist.

https://github.com/naudio/NAudio/blob/master/NAudio.Core/Dsp/BiQuadFilter.cs Simple formulas for all major types of filters

Not tried it but there is also NAudio (https://github.com/naudio/NAudio)

I’ve used NAudio for this. You get Windows support for free, and it’s not too difficult to implement for other platforms. I have gists published for Android and iOS that should get you going.

This might help: NAudio, they have a doc file for working with pitch here

Hello. I'm working on a school assignment where I'm creating an application where the user can play .wav files slowed down to 0.5 speed. I'm using Naudio and Varispeed to accomplish this. There is a varispeed demo that shows how to do it and from what I can tell, I'm doing more or less the exact same in my code as the demo. But the wave file is played in the normal speed anyways. Does anyone have a clue on what I'm doing wrong?