I want to create a real time voice modulator with the audio output from the modulator going to an app or video game on Windows 10. The concept is the same as MorphVox and VoiceMod. It looks like MorphVox creates a virtual device and it looks like VoiceMod is also using a virtual audio device. I would like to know where to start. I would really like some help finding a starting point to making a virtual audio device. Any language is fine if someone is willing to show code (if it can be achieved in more than one language). But I'd be happy with a good tutorial also! I really appreciate the help!
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I’m looking to essentially use two devices: raspberry pi 3 and Mac 10.15. I am using the pi to capture video from my web cam and I want to use my Mac to kind of extend to the pi so when I use cv2.videocapture I can capture that same video in preferably real-time or something close. I’m programming this using python on bout devices. I thought of putting it on a local server and retrieving it but I have no idea how I could use that with opencv. If someone could provide and explain a useful example, I would greatly appreciate it. Thank you.
To transfer a video stream, you could use instead of a custom solution a RTMP server on the source machine feeding it with the cam source and the target opens the stream and processes it.
A similar approach to mine is widely implemented into IP cameras: They run a RTMP server to make the stream available for phones and PC.
So I'm kinda stuck here.
I have a radio station, but we are mobile. So I have a studio on wheels. The problem is, we have an antenna, but we always have to place that really close to our studio. Now I want to make an device that can stream the audio from the audio mixer to the internet and can be received by another device in another network and send that signal to the antenna (audio output).
to make this clear, I made a schema with raspberry pi's;
I want this to be plug and play So I only have to plug in the device in the modem (or network we have) on both sides and the devices should find each other.
I don't know HOW I can do this, so I need to know a couple of things:
What hardware should I use?
What software should I use?
What is the best configuration to accomplish this?
Can I use 2 raspberry pi's?
How can I let the devices find each other over the internet?
There need to be some features;
The system needs to be able to buffer the audio for 5-10 seconds
It needs to be direct, so it's live and not a file that needs to be played
The system must be failless (beside the fact the internet can die).
Plug and play is a must, I don't want to have a really messy configuration to do. (if possible, without any kind of portforwarding).
I would really appreciate help and a decent explaination.
regards,
Robin
Well, it depends on your capabilities as a programmer.
If you're really fixated on the RPi for it's convenient form factor, there's a ton of community support, so I'd start with something like this project to kick start you in the right direction. If you already know python pretty well, modify away and have fun.
If you have no programming experience, you'll probably want to put a desktop in place of the RPi and launch some instances of VLC. It's not necessarily plug n play, but you can get close enough by getting a command line VLC to launch at startup.
Either way, the more difficult problem here is the "over the internet" part. This would really need to be a server-client model, but who is your server depends on who is more stationary (I'm guessing Location 2?) because the client will need to know the IP address of the server somehow. There are dozens of ways to make this happen, but at the end of the day, you'll want to use sockets accomplish the
It needs to be direct, so it's live and not a file
... which unfortunately gets complicated. See this answer for confirmation. Would love to help with some tips on implementation, but we need more information about your willingness to "dig into the code", the necessity of the RPi, and whether the stationary location has a static web address.
I'm trying to create an interactive voice-tree for an art project. Think of something like a choose-you-own-adventure, but on the phone and with voice commands. I already have a fair amount of experience working with Construct 2 (game-making software), and can easily build a branching, voice controlled interaction loadable through a modern browser with it. For reasons relevant to the overall story, I need players to connect to the interaction through a Google Voice number they will call.
I already have a GV number and have written an AutoHotKey script to auto-answer the Hangouts call, but I'm stuck trying to route the audio from the caller in Hangouts to the browser AND the audio response output of the browser back to the caller.
I know of an extremely primitive way to accomplish this, [which I've illustrated with this diagram:
Unfortunately, this is rather cumbersome and I suspect I can achieve my goal through virtualization or at the VERY least some sort of attenuation cables between two physical machines (I tried running a generic AUX cable between two laptops, but couldn't get speaker audio to go into microphone audio from one to the other).
I've been experimenting on Parallels running Windows 8.1 with Virtual Audio Cable(no luck), JACK(too robust), Chevolume(too limited), and IndieVolume(too limited).
I suspect VAC would be the best bet, but I can't seem to find a way to route Firefox audio output to a microphone input which directs to Chrome and vice versa. If I try accomplishing it all through just one virtual machine I have to use two different browsers for the voice-tree webpage and Hangouts call since Hangouts pushes its audio through Chrome (even the stand-alone application).
Is there any way to route microphone input and speaker output separately between two virtual machines? If not, could I still try and accomplish this with a specific type of cables between two laptops running windows 7/8 that have generic audio jacks?
I am to test voice recognition programs. Some which I have access to the code and others where I don't.
Sadly my (beautiful) voice is not perfect, so when I am reading a text it sounds slightly different each time. Which makes the testing difficult and time consuming. Giving that I can tweak a lot of parameters.
So I was wondering if there was a way to record my own voice (already done). And then play it as normal microphone input so the voice recognition program I am testing will see it as microphone input.
This would also help greatly if it could be done programatically in C#. So I can in my own code specify when to play what.
To play it from speakers and have the voice recognition programs listen to the microphone is not an option, because it is not the same sound on different computers/speakers/microphones.
Thanks.
Edit:
What i have found so far is to use a software sound Card simulator. But I haven't been able to find a suitable one.
Just as there are printer drivers that do not connect to a printer at all but rather write to a PDF file, analogously there are virtual audio drivers available that do not connect to a physical microphone at all but can pipe input from other sources such as files or other programs.
I hope I'm not breaking any rules by recommending free/donation software, but VB-Audio Virtual Cable should let you create a pair of virtual input and output audio devices. Then you could play an MP3 into the virtual output device and then set the virtual input device as your "microphone". In theory I think that should work.
If all else fails, you could always roll your own virtual audio driver. Microsoft provides some sample code but unfortunately it is not applicable to the older Windows XP audio model. There is probably sample code available for XP too.
I want to make (for fun, challenge) a videoconference application, I have some ideas about this:
1) taking the audio/video streams (I don't know what an audio/video stream is)
2) pass this to a server that lets communicate the clients. I can figure out how to write a server(there are a lot of books and documentation about this) but I really don't know how to interact with the webcam and with the audio/video in general.
I want some links, book, suggestions about the basics of digital audio/video expecially on programming. Please help me!!!
I want to make it run on a Linux platform.
Linux makes video grabbing really nice. As long as you have a driver that outputs the video stream to the /dev/video/v* channels. All you have to do is open up a control connection to the device [an exercise for the OP] and then read in the channel like a file [given the parameters set by the control connection. Audio should be the same way, but don't quote me on it.
BTW: Video streaming from a server is a very complex issue. You have to develop or use an existing protocol. You have to be very aware of networking delays, and adjust the information sent (resize or recompress) to the client based on the link size between the client and the server.