Yes, I've read it in the manual, but I mean that if it possible to control this potentiometer digital with AVR or Attiny etc (and to control it with a IR-remote)
I understand that the volume is controlled with a voltage from 0-5V.
The question is, what is the step (in dB) per voltage?
my nerves plugin not keeps config
I bought 2 minidsp boards I open the package and I love the Rca that is induced.
Then I connect the board to my pc and upgrade/ synchronize everything works ok , I close the plug in (not find any save at board ) all still work fine , then I unplug the usb and I connect again to check if it works … anything is mute (thanks god not bypassed filters) I open the plug in press the synchronize button everything worked again but when I unplug and plug again always the same
What I can do ?
Thanks in advance
I bought 2 minidsp boards I open the package and I love the Rca that is induced.
Then I connect the board to my pc and upgrade/ synchronize everything works ok , I close the plug in (not find any save at board ) all still work fine , then I unplug the usb and I connect again to check if it works … anything is mute (thanks god not bypassed filters) I open the plug in press the synchronize button everything worked again but when I unplug and plug again always the same
An externally hosted image should be here but it was not working when we last tested it.
What I can do ?
Thanks in advance
Following a similar comment from one user, We've noticed on last Friday a potential memory leak which could cause this symptom with the latest plug-in, for some specific set of configuration. After working over the week end to find a solution, we have a fix which we're starting to implement across all plug-ins as a safe measure. New plug-ins will be released tomorrow or the day after if all goes well. It will fix this issue. Please stay tuned.
FYI to all miniDSP users: Note that we randomly check DIYaudio forum, since this is a 3rd party forum and not our primary forum. Please post all tech support inquiries (similar to the one you had) on minidsp forum or better by email for faster processing. Much easier to manage them this way. Please do not double post though since it becomes a nightmare for us.. Thanks for your understanding.
Last edited:
No problem i also post it here so maybe a user tell me what am i doing wrong not know that was an isue .
you know when i post you was sleeping (time diference ) so i say lets see if a user knows anything .
Now that i know is pc configuration problem i will use it on another pc. maybe it worked before the plugin upgrade
I will stay tuned
you know when i post you was sleeping (time diference ) so i say lets see if a user knows anything .
Now that i know is pc configuration problem i will use it on another pc. maybe it worked before the plugin upgrade
I will stay tuned
Can you be a bit more specific about this?
Maybe a schematic example and more information about the output
How is the logaritmic aspect of volume control organised? Should you provide a linear input voltage, or logaritmic.
The same question but then when using a pot: Should a linear potmeter be used, or a logaritmic?
you are right about adding it to the documentation
Unfortunately, if it indeed needs to be a logaritmic pot, this makes the control range rather limited. With an 8 bit resolution of the ADC this yield only 24dB of range...
I've read some datasheets of the ADAU1701 and also the manual and datasheets of the Evaluation Board.
In those documents they are talking about using a linear potentiometer from 0-3,3V.
To quote some text:
and
I don't know if miniDSP had changed some things with the "implementation" according to the volume control?
In those documents they are talking about using a linear potentiometer from 0-3,3V.
To quote some text:
and
I don't know if miniDSP had changed some things with the "implementation" according to the volume control?
Thank you !
Another question (maybe it is discussed before?)
Will it be possible (in the future) to make filters with a custom Q and Fs (2nd, 3th, 4th, 8th order)?
The difficult part is that you can't use tables with only some data points, but you have to calculate the transfer function. I've done it for 2nd order systems and I'm working on 4th order systems. Don't know how it can be implemented in your source code.
Another question (maybe it is discussed before?)
Will it be possible (in the future) to make filters with a custom Q and Fs (2nd, 3th, 4th, 8th order)?
The difficult part is that you can't use tables with only some data points, but you have to calculate the transfer function. I've done it for 2nd order systems and I'm working on 4th order systems. Don't know how it can be implemented in your source code.
Great, thanks!
I am very satisfied!
Sorry for my first two unfriendly mail to your support I sent today!
Everything is working perfectly now. The volume poti, the linear 10k version, is working perfectly with no remarkable delay.
Just great.
And also the problem we had before (the plugin was not stored permanently on the board) is solved.
I was never in doubt about the hardware and support quality you are providing, but during the last week I had my doubts about the software and documentations quality.
Completely gone, my doubts, now!
Thanks!
Franz
Last edited:
Thank you !
Another question (maybe it is discussed before?)
Will it be possible (in the future) to make filters with a custom Q and Fs (2nd, 3th, 4th, 8th order)?
I think that our future advance plug-in will give you something close to what you're looking for... More to follow when it's close to release.
I'm not a MiniDSP user. But I am a DSP engineer. To me all those questions about custom Q, custom frequency, ... look more and more ridiculous. What a silly game ! And this is MiniDSP software architects fault. MiniDSP software architects, WHY DON'T YOU ALLOW PEOPLE TO UPLOAD THEIR OWN 2nd ORDER IIR FILTER COEFFICIENTS ? AND PUT, SAY, FOUR SUCH 2nd ORDER IIRS IN SERIES ON EACH CHANNEL ? DSP ALLOWS FAR MORE THAN LOW PASS OR HIGH PASS IF YOU KNOW WHAT YOU ARE DOING WITH THE NUMERATOR COEFFICIENTS. DSP ALLOWS YOU TO IMPLEMENT IN ONE SINGLE 2nd ORDER IIR, A STATE VARIABLE CELL CONTAINING a% LOWPASS + b% BANDPASS + c% HIGHPASS, ON A COMMON DENOMINATOR, with a, b and c in the range of (-1,+1). If you don't know yourself how to use the DSP56K platform, please be so kind to allow other people to use it like it should. Have you noticed that the ALLPASS (pure phase shifter) is inside this, also ? Are you listening to people, MiniDSP software architects ? At the moment you are wasting people time and energy by not taking the right approach. You generate user frustration. You will crash if you keep this attitude. In a few weeks or months, somebody else will release the exact same hardware with a completely free IIR access like described above. For less money. Instead of trying to get people captive on a ill-designed software, please make your software as simple as possible, using IIR coefficients tables and IIR chaining tables. Your market doesn't need more. Your market DOESN'T WANT MORE. Audiophiles may allow tightly controlled IIRs because that's something simple to understand. But if you bring on the market something that is executing "some kind of IIRs" in an opaque way, your customers will vanish in the nature. Because your market is computer litterate people able to learn themselves about IIRs, able to download and use free PC software aiming at designing IIRs. Now, if you want to simplify our life, please make a PC application enabling people to design their own IIRs, put them in series, and see what's coming out. Before uploading the tables on MiniDSP. This is a two-weeks job, and will open lots of doors that you have maintained closed. Have you realized, that if you keep your ill-designed approach, the day you will need to open those doors (because somebody else, a competitor, doing it), it will cost you time and money in software development, you'll end up in a version + revision hell, and all your customer base will vanish ? Please, apply a lot of pedagogy in teaching people exactly what's happening while executing a 2nd order IIR, because, I hope you are aware of the fact that doing 24-bit DSP on a 56-bit accumulator is perceived as nearly obsolescent in 2010. And you and me we know this is a misconception, as 48-bit accumulation plus 8 bit headroom, in linear coding, is actually better than a bruteforce floating point 32 bits. WHY DON'T YOU EXPLAIN THIS to people ? Do you need to hide something ? Is there a dog in your DSP engine ? Is your system exhibiting a nasty behaviour regarding DC, with a drift caused by asymetric rounding, introducing the need for some dirty VLF high-pass in the chain ? The more time is passing, the more I think there is something like this, kind of dog, in your system. This, of course, would explain why you don't want to provide transparency regarding what's happening inside the DSP and the way the IIR filters are organized.
- Status
- This old topic is closed. If you want to reopen this topic, contact a moderator using the "Report Post" button.