MiniDSP -> Passive preamp/potentiometer. Impedance matching?

Status
This old topic is closed. If you want to reopen this topic, contact a moderator using the "Report Post" button.
Hi!

I'm currently planning my first real DIY audio project.
The first stage to building this is to design a MiniDSP based "speaker processor" wich will basically play the role as DAC with s/pdif input, active crossover and preamplifier with analog passive attenuation between the MiniDSP analogue outputs and the 4 channels of amplification for the pair of 2-way speakers.

Like such, from start to finish in the signal chain:

1. Digital signal (s/pdif)
2. MiniDIGI
3. MiniDSP 2x4 (4 analogue outputs)
4. Vishay P11S logarithmic potentiometers (passive preamp) for 4 channel attenuation (i.e., a total of four stacked P11S on a single shaft connected to a single volume knob)
5. 4 poweramps for 4 channels of amplification for a pair of 2-way speakers

Now, my concerns are with impedance matching, as I want each dedicated amplifier to have full control over the driver it's connected to, in terms of impedance and damping factor.

What do you think would be the safest impedance on the actual potentiometer? I believe the output impedance of the MiniDSP is 0.5k Ohms (560 Ohms to be specific).

Perhaps a 5k Ohms? Are more precisely, a total of four in a stacked configuration (the Vishay P11 series can be easily modified to be stacked), like this one (in english): https://www.elfaelektronikk.no/elfa3~no_en/elfa/init.do?item=64-349-85&toc=20164

Open the PDF datasheet to see the stacked configuration

Thanks! :)
 
I am new to the MiniDSP and am thinking of doing something similar....my understanding of this...and probably something you may benefit from,,,is the use of the PEQ active XO plugin will handle all of the XO operations including output to each driver. If you use the PEQ plugin I don't think you would need an analogue control to each driver....just the one main volume control as mentions in the last post. Why use an analogue "pot" in this signal path when the PEQ will do this digitally? Or am I incorrect in my understanding of the way the system works. The PEQ handles the XO function as well as the attenuation of the signal to each driver in the 2way arrangement.....or did I misunderstand the way it works?

My idea was to use my own DIY DAC End 2 and my Parasound Jfet preamp driving into the MiniDSP/PEQ......output to each of the four drivers going into a Parasound HCA800II for the tweeters and a Parasound HCA1000 for the mid woofers. Of course the Parasound amps have a nice arrangement of pots, one for each channel, that allows you to set the volume output of each channel, or they can be easily bypassed which is what my intentions were.

Please let me know if my plans are flawed....

Jeff
 
Thanks for your replies
The reason I want to avoid volume control in the digital domain is because it's a lossy process, how overrated the problem may be, I want to maintain the dynamic range from the DAC of the MiniDSP. And by feeding an un-attenuated, loud signal to the analogue volume control, maximum SNR is maintained. Even if it has "more headroom" by doing this in 24 bits, the SNR ratio still narrows when going down.

This might become noticable if I decide to go the high-efficiency pro-driver route, not sure.
Granted I'll probably start with digital volume and then "upgrade"
 
The reason I want to avoid volume control in the digital domain is because it's a lossy process, ... I want to maintain the dynamic range from the DAC of the MiniDSP. And by feeding an un-attenuated, loud signal to the analogue volume control, maximum SNR is maintained. Even if it has "more headroom" by doing this in 24 bits, the SNR ratio still narrows when going down.

No, you lose headroom just as surely with an analog volume control.
 
no..the SNR is NOT maintained, you are already acting in the digital domain to attenuate signals for your XO, you are not even adding a further process, doing it in the digital domain with the one pot is a far better solution and cheaper too, with perfect channel matching. by adding noise (and you will surely be adding more noise with your 4 pots), you ARE losing bits and unnecessarily adding a FURTHER lossy process. this feeling and folklore that seems to ignore analogue attenuation as a source of information loss has got to wake up sometime, its by far the lossier of the 2 given enough bit depth in the digital process and certainly less convenient. no need for worry about impedance matching that way, the mini DSP will surely drive your power amps inputs.
 
Last edited:
Thanks for your replies
The reason I want to avoid volume control in the digital domain is because it's a lossy process, how overrated the problem may be, I want to maintain the dynamic range from the DAC of the MiniDSP. And by feeding an un-attenuated, loud signal to the analogue volume control, maximum SNR is maintained. Even if it has "more headroom" by doing this in 24 bits, the SNR ratio still narrows when going down.

This might become noticable if I decide to go the high-efficiency pro-driver route, not sure.
Granted I'll probably start with digital volume and then "upgrade"

You don't loose bits by using the volume control built into the MiniDSP AFAIK. I have used it and would not take other approach.

vac.
 
Member
Joined 2010
Paid Member
Interesting discussion!

You will definitely loose resolution (bits) when attenuating in digital domain, however as qusb mentioned you will also loose resolution attenuating in the analog domain.

For me the question is, is analog attenuation (if done right!) really by far lossier as qusb mentioned?

As a first approach I will take a look to the datasheets (e.g. PGA, Codec,...): e.g. THD+N at -60dB.

Next step will be doing some measurements. I am about to set up a "preamplifier" with miniDSP 2x8 and "optional" analog attenuation (will try different things like PGA, Pot, stepped attenuator, w or w/o buffer) and plan to do some measurements (Though might take some weeks or months, 2 kids are so time consuming ;-)). Lets see if my equippment can find out some obvious differences for low output signal levels. One thing is clear: analog attenuation if (far) more effort! Build in attenuation of miniDSP is really easy.
 
Thanks for all your replies, yeah as I said "perhaps i'm just paranoid", what I meant by that statement is that I'm not "religious" about my opinions at all :)
But as I said, my biggest concern was the effect (if at all noticable) when using high efficiency drivers, like the Beyma TPL150 with it's 99dB/1w/1m efficiency, considering it's a 10dB "head start" over many typical non-pro drivers, and 10dB is often mentioned in discussions around digital attenuation.

If I understand correctly you loose 1 bit for every 6dB of attenuation in the digital domain, but with 24bits processing, you can attenuate -48dB and still have 16bits left, so yes, I was being paranoid :) And then there's dithering methods, not sure if that applies to minidsp or not.
Double-blind study would probably confirm what you're all saying, that it's irrelevant and unnoticable and that loss of dynamic range with analogue attenuation is equally lossy as digital.

curryman : That would be a interesting experiment! :) I seem to recall Meridian using some kind of hybrid at one point, that is hybrid digital/analog attenuation. There's probably lots of ways of doing this, perhaps even mechanically (i.e. linking the "digital" pot (the one connected to the MINIDSP board) mechanically with an "analogue" pot would perhaps be one way of doing it. Question is if it's worth the hassle.

For my own project I'll be sure to go the digital-domain route since I'm a newbie at DIY audio and I don't want to become overwhelmed by complexity :D
 
curryman, for sure its the lesser of the 2 evils, ive been using 40 and 64bit attenuation on my sabre dac and usb interface for years now and theres no turning back. good luck even getting close with a 200 dollar pot. with these methods an ADC is used to read a reference voltage applied through a resistor, so you can use a cheapie carbon pot, a relay based attenuator, or as ive done as well you can make a simple stepped attenuator, single layer with smd resistors and this works really well, the quality of the resistive element matters nothing to the sound, but the action and feel is nicer. it takes a little getting used to the perfect channel matching all the way down, a hard ask for an analogue pot of any value
 
Member
Joined 2010
Paid Member
Hi all,
I totally agree that digital attenuation is the far easier solution (only one simple pot) and automatically gives good channel matching which is not easy to achieve with pots or stepped attenuators (PGA might be better) and I also will definately go this way first!

However looking specifically at the miniDSP solution of digital attenuation (by the way: done in DSP or in AD/DA Codec? I'll ask miniDSP devteam) it is not working with 64bits like sabre DAC and the DAC is not 32bit resolution. Don't get me wrong: I don't claim that analog is better! I am not sure if digital is better either. I just want to get a feeling and understanding by doing measurements and learning the theory behind. In the end it's definitely some kind of paranoia

In general gain structure is very important doing signal processing in the digital domain. Just thinking about the tweeter channel: put a highpass filter, say 2,5 kHz to a typical music signal not much signal energy is left. If you attenuate this signal by 40dB you might get into trouble even with 24bit resolution. Thus it makes sense to think about it a little bit ;-)

@tnargs: sorry, missed it
 
Last edited:
oh sure, its not as advanced in the minidsp, but there was thorough testing done at a recent audio meet here in australia with minidsp specifically and the guys there ended up not being able to discern a difference between high end preamp with analogue attenuation and the internal digital pot. so i think its wise you try that first.

btw the sabre uses 40bit and i've been using that the longest, my multichannel usb ->i2s interface (titan) however uses 64bits, this is only a recent addition to my system, I cant discern the difference between them as you would expect, but as titan is the source i find it more convenient to use that, its also multichannel and allows me to control all dacs simultaneously.

oh and yeah i was just tnargs's backup :D
 
Status
This old topic is closed. If you want to reopen this topic, contact a moderator using the "Report Post" button.