I started playing this online game which was designed poorly: Your microphone is always forced on. But I like to talk to my friends through Discord while playing this game, and that creates a problem. Sometimes, I want to talk to only my friends but not the people ingame. This means I need push-to-talk ingame, which doesn't exist.

So my workaround is to create a virtual microphone that acts as a proxy for my real microphone, and I can push-to-talk to "let audio flow through" the virtual microphone. Then, I assign the virtual microphone as the input device for the game, and now I have push-to-talk.


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You might find the setting up and routing of the virtual device complicated when launching the app for the first time. For this, i would advise you check out this video which contains instructions on how to setup the VB-Audio softwares. Simply ignore the part where she moves onto "OVR Advanced Settings" which only applies to her setup. The first two steps involving virtual cable and voicemeeter still apply to you.

The software is not perfect, and users have reported that it is critical you do not quit the app or shutdown the computer when the mic is muted. You should unmute it first, otherwise your system will shut down with the levels of your virtual mic left at 0 and it gets left that way.

So I use Google Meet for transmission and once I use a DSRL camera, I'm using a video capture card, which delays a little bit my video. The idea is that I would enter my audio into OBS, delay it and send it (delayed) for Google Meet, and at the same time I would send my virtual camera, from OBS.

I need to record the lessons for students who do not attend the live class. Google Meet does have a record feature, but that feature will be discontinued at the end of September because my school's G-Suite subscription does not include it in their package. My plan was to use OBS to capture my display as well as webcam (it will actually work out nicely as I can set up different scenes for different situations in class) and output that through VirtualCam and use that as my input for Google Meet, all while recording the class using OBS. However, audio is giving me some issues because if I have my microphone input to Google Meet, it will not be recorded in OBS, and if I have my audio input into OBS, I cannot get my audio into Google Meet.

I was hoping there was an add-on similar to Virtual Cam but for audio, but I have yet to find anything. It would also be nice if I could use my microphone as an input device for two applications, but that doesn't seem to be possible either.

The SD-87 AI is a more modern iteration of this classic microphone. First introduced in 1986, this mic offers an almost entirely flat response with a slight boost in the 10k range. This gives producers an unprecedented level of flexibility, giving you crystal clear, transparent vocals every time.

I found an example that shows how to pass a wave file as microphone input by utilizing "pactl load-module module-pipe-source". The issue with this example is that it relies on an infinite while loop and does not stop when the audio file is success put through the microphone a single time. If someone can a fix to this example that would be great. I heard of:sudo modprobe snd-dummyBut do not know how to use it. documentation is quite lacking.

outside of the while loop. It appears that only a tiny sample of the audio file reaches the microphone, not the entire clip. I have no idea why this is the case. Not sure if like the mic file is regularly purged or something.

Of course, I have done some research. I found out that I can use a software to create a virtual device and do few things to have the result. But my point is if it is possible without installing software but with some kind of python script?

First download: -audio.com/Cable/, this will create a "Virtual Audio Cable" where programs can play music to the input device (What looks like a speaker) and it'll pipe it to the output device (What looks like a microphone).

PPS: I'm still trying to figure out a way to pipe your own mic through there as well since this method will obviously not pipe your real microphone in too, but looking into the source code of pygame is making my head hurt due to it all being written in C.

First, you'll need to download the VB-Cable Virtual Mic Driver for your respective platform and install it. This provides us with a virtual mic that'll allow us to pass audio we play on our machine as a microphone input when using a video calling software (Google Meet, Microsoft Teams, Zoom). After that, it's all handled through the pygame module's audio package.

Then, on the other end, inside the relevant software, just change the microphone input from the default to CABLE Output (VB-Audio Virtual Cable) to have those on the other end hear the audio from the source.

Totally do-able. The idea is you create a virtual speaker device, and have it connected to a virtual microphone device. On Windows, one could use a virtual audio cable. But on Ubuntu, that app doesn't exist. But we can still create a virtual audio cable using something like PulseAudio as described in this Stack Overflow post. Here's the part of the post that answers your question:

My previous post got the video from my smartphoneto show up as a camera device on my desktop, but for a videochat, we probably also want audio. So, now the question is:how to build GStreamer pipelines that will allowminimal-webrtc-gstreamer to use virtualmicrophone and speaker devices that I can point a voice/video chatapplication at, allowing me to use my smartphone's microphone andspeaker for applications on my desktop.

With further research, I found multiple posts talking about makingvirtual microphones in order to combine microphone and game/media outputinto the same stream. This one is the most detailedone I found.

While the null sink automatically includes a "monitor" source, manyprograms know to exclude monitors when listing microphones. To workaround that, the module-remap-source modulelets us clone that source to another one not labeled as being a monitor:

The tutorial I linked above discusses usingmodule-loopback to pipe audio among variousdevices, so, for example, the monitor of your actual speakers could besent to the remote microphone and/or speakers.

Microphone Mist technology subdivides a room into thousands of spatial bubbles. This allows the system to zoom in on individual talkers and optimize both acoustic microphone pickup performance and post-processing of the audio signal.



Traditional microphone technologies, such as beamforming, subdivide the room into large zones (as many zones as there are microphones). If there are, for example, three zones in a beamforming array, you get low spatial resolution information. Although the coverage zones are large, traditional systems are not able to discriminate individual sound sources within a single zone. This means diminished ability to optimize individual sound sources in a zone for the appropriate gain setting, noise filtering and overall audio signal optimization.

Microphone Mist technology fills a room with thousands of virtual microphones, subdividing the room into thousands of spatial bubbles for higher spatial resolution of audio signals. This allows the system to isolate and zoom in on individual talkers, giving it more audio information that in turn allows it to optimize both acoustic microphone pickup performance and post-processing of the audio signal.

Hi. I have installed Background music and other applications that change your sound and then pass it through a virtual device. That worked fine. But now I don't want to use them anymore and I have deleted the apps (via AppCleaner). Unfortunately The devices are still there and I would like to delete them. Its not only for aesthetic reasons since the background music one always gets prioritized in tools like jitsiMeet.

assuming everything is configured correctly; it is likely that the wrong microphone source is being selected for use. normally the last attached audio device would be used, but, possibly something else was attached more recently or for some reason its picking up the wrong things.

I am experimenting using both Mac and Windows clients and getting inconsistent results with regard to camera and microphone functionality. Besides that, the virtual desktops appear to be working as expected (I've tried pooled and personal configurations).

One other note. When using the "Windows 365" application to connect, the cameras works even outside of a Teams call . It's grainy and sub-par, but it's worth noting that it's at least functional. Consider there are at least three ways to connect to the Windows 365 virtual machine, and each offers different behaviors:

* through a browser (arm/webclient)

* Remote Desktop Client

* Windows 365 application

Again. It just shows how this technology is evolving and a person needs to keep checking back for updates and possible fixes. Azure Virtual Desktop will probably evolve, but it's not as polished as Windows 365.....at least for now. Things are changing quickly. Maybe next month or next several months this post will be entirely outdated.

These days, we all spend a lot of time on video, so why not put your best face forward? Using Ecamm's virtual cam and mic, you can have total control over how you look and sound, helping you to stand out from the crowd.

You could probably receive microphone input through an API. However to send it so the other person can here it they would need your software downloaded and have that actively recieveing through wifi and then converting that data into sound. Could be done if you can find the api for the microphone

To hopefully it clear up where necessary:

You want to go between the mic, which turns acustic into digital information, and the program, Skype was your example, and either mix something into that input or replace it so that instead of the actual microphone input that modified/replaced input gets send to Skype? ff782bc1db

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