We haven’t tried it but I can’t see any reason why it wouldn’t work

Probably a good point lol

Reaction time is 4ms. You really should only use it for vocal signals, not on the main output. So you can have individual instances on each vocal channel or you can subgroup your vocal channels but not have more than three or four simultaneously singing/talking vocals going through any instance at a time.

@ Thanks! I hope an iOS version is coming too. Or would there be too much latency?

@@MrNicknaymewe haven’t looked into an iOS version yet, although that is certainly possible.

6-10 db depending on system.

I picked the absolutely worst one.

Read the FAQ

What we mean by zero latency is that there’s no difference between running or bypassing the plug-in

Waves lists many of their plug ins as having 0 latency.

Zero processing latency and zero transport latency are two very different things. These don't add latency to the processing chain but you still have to factor in the AD and DA conversions coming in and out of the computer. Stuff like soundgrid, DVS or even the Yamaha dm3s USB interface can bring this down to around 6ms roundtrip in my experience with other plugins

He’s not taking about hardware. He’s taking about the plug in. You choose whatever hardware you want and get the latency you get.

​@@devinlsheets_alphasound Well, there's always _some_ delay when using a plugin. Even something as simple as a gain plugin is not zero latency, it's just really really short. The same goes for the mixers themselves. In the case of analog mixers stem from things like input capacitances and inductances, and a simple capacitor based lowpass filter will also introduce somewhat variable latency in the order of up to a quarter (IIRC?) wavelength of the Nyquist frequency. For digital mixers it's similar, like the quantization time of the ADC and DAC and the delay of whatever fixed function logic/FPGA to process the filters, with the added buffer length of the pipeline. Probably faster than the analog mixer, but still not zero. From my understanding this software uses a real time filter and a non-realtime (but fast) AI algorithm to tune that filter. The filter would then still introduce latency, even if it's short and predictable with fixed function logic. I'm not talking about hardware latency or the inherent latency of Windows subsystems, the actual filter inevitably has some delay no matter how you do it. Like I said the delay can be negligible, but it's not zero.

Using it with Dugan is a perfect solution. We’ve been getting anywhere from 6-10dB of usable gain out of the gate by itself, and Dugan helps even more with overall clarity and balance of sound.

@@devinlsheets_alphasound Compared to using say a parametric or a graphic - how much more gain can the de-feedback machine achieve - maybe post a comparison video?

@@christiansoldier1118 in our experience it’s probably about 6dB more usable gain than the best attempts at hacking away with EQ. Some people have reported as much as 10dB or more, but maybe they’re just not as good at EQ in the first place, and also every situation is different so it’s hard to say.

@@devinlsheets_alphasound Thank you

Well actually, we specified that it’s affordable for “serious” people.

Ideally it would be an ARM based processor, and it should only be running the code and nothing else. And it should be rackmount. Hopefully someone can make that happen!

FAQ

FAQ

​​@@duanesheets5484 I suppose this (below) is the answer, but still seems unclear to me. I saw mention of not running more than 4 on their more expensive system. So I guess 4 is the answer. HOW MANY INSTANCES OF THE PLUGIN CAN I RUN ON MY COMPUTER? As many as your machine CPU will handle, given a certain buffer size.

What whistle problem?

There is a tiny bit of phase shift but not anything like that of feedback-fighting EQ dips

@ would love to test this out. Would be nice to use with Breakouts when you got union clowns as ops

It definitely helps with certain types of bleed, such as room reverb and some amount of background noise, but it isn't specifically designed to remove other forms of bleed from things like drums or whatnot.

@@devinlsheets_alphasound guess that's the next plugin then :)

Yes, but make sure no more than about three people are ever talking at the same time.

yes

@@devinlsheets_alphasound Is it possible to subscribe to a notification / newsletter etc?

It’s complicated, but in some sense the algorithm is buffer independent. It could operate with a buffer of 1 sample if conditions were ideal. Send us an email if you want to get into the weeds, although we can’t talk too much about it without NDAs and such.

Soon we’ll be adding heresy filters for churches, so you can select between denominations like Baptist or Lutheran, etc.

5ms is fine for most people, but for IEMs you may need even less, and that’s where native Dante and other solutions come into play, but they’re much more expensive.

Based on how you set it up, it should be possible on pretty much all digital mixers to not have this inserted after the tap point of the monitor mix, making the latency irrelevent for that. And 5ms is not noticable in most situations.

We don't publish the full specs of our recommended machines because they aren't very meaningful, since most of the magic happens at the OS configuration level. For example, that unit in the video has an N95 processor, but outperforms even some i9 based systems because of how we have set it up.

@@devinlsheets_alphasound a little bit of transparency goes a long way here. we don't believe in magic with computers. how is windows configured for more performance on an n95 than an i9 with stock windows? even if you post a full detailed guide on how to optimize windows for this, many people will still buy the premade kit, and it would be a win-win for you. i'm not interested in buying something shrouded in secrecy and magic sauce, but i could be interested in this plugin if the method of optimizing the computer were available beforehand so i could see if it's something i want to do.

Those sound terrible lol

Honestly I think feedback extermination should be one device per microphone that does not involve computers or plugins, I have used three or four dedicated feedback devices and always they are placed between the microphone preamp stage and the first affect your amplification or mixing stages. I have been a harmonica player in saxophone player for a long time and standing in front of PA systems are amplifiers with a harmonica and a microphone has led me to a lot of experience with electronic feedback extermination. The more I think about it the more I don't even think this product is a good idea in the first place..... The first line of defense for feedback is always the pickup pattern of the microphone and the angle of the speakers. Automatic EQ types of things are just like the fire door in case of emergency. Certainly not something that needs improvement with AI and windows plugins...

@@toehser that’s fine you don’t have to buy it.

Alpha Sound