TOPIC:

Setting max volume at -10dB, as instructed by the user manual, will cause the output voltage to drop under 2V.  Even though my passive amp has sufficient power to amplify the volume to enjoyable level, it sounds colder and thinner.  I tried another tube integrated amp and it also sounds less dynamic with SHD at -10dB.

I suffered for months and I think I found a solution now.  Let me share with you.

My solution is to apply -10dB headroom here, not at volume control.  This gives back warmth and punch.



I don't have test instruments but it sounds like Dirac will adjust the output so DAC still gives 2V to the amps.  The music is again dynamic and full.  I didn't hear any clipping either.  (I only listened to a few songs, though.)

What do you guys think?  Do you hear the same?

OK I listened more today and no, Dirac will not adjust the output to 2V.

But adding -10dB before or after Dirac do sound differently, with or without Dirac turned on.  I have no idea why.

I like -10dB before Dirac better.  The end result is not louder but feels fuller.  I'm still happier.

I guess I will have to buy an expensive professional DAC that adds analog gain to the output... 

You're evaluating all of this subjectively???

Dave.

Yeah... I would appreciate you guys' help.  

edmundpan wrote:

My solution is to apply -10dB headroom here, not at volume control.  This gives back warmth and punch.

 

Did you change all of the 0dB values to - 10 dB? If you did I wouldn't expect it to have many any difference whatsoever compared to setting the master volume to -10dB. My suspicion is that there is something else going on here...

The required headroom depends on the music material... "old school" recordings should require less while those that follow the "loudness war" guidelines need more so I'd suggest experimenting with a lower value and checking if it works well with your music genre/material

The OP's issue is unclear and not well understood.
I don't believe this to be a "headroom" issue, nor do I believe subjective evaluation (exclusively) can yield any sort of meaningful conclusion.
This is just a simple gain structure analysis and exercise that needs to evaluated objectively.

Dave.

dreite wrote:

The OP's issue is unclear and not well understood.
I don't believe this to be a "headroom" issue, nor do I believe subjective evaluation (exclusively) can yield any sort of meaningful conclusion.
This is just a simple gain structure analysis and exercise that needs to evaluated objectively.

Dave.

The reason the manual suggests using -10 dB is to provide headroom for Dirac Live filters, although as @flak points out how necessary this will be will depend on the recording and the actual DL filters. It's possible to check the peak boost applied by DL to give a guide if desired.

Yeah, I'm well aware of why the -10db is there for DL.  However, the OP is noting the behavior whether DL is engaged or not.
Regardless, the process of applying a -10db offset on a unit limited to 2VRMS output is problematic, at best.

Dave.

dreite wrote:

Yeah, I'm well aware of why the -10db is there for DL.  However, the OP is noting the behavior whether DL is engaged or not.

Which is exactly what would be expected, since this wouldn't affect the attenuation applied (as you know). 

The point being made was why the -10dB attenuation was being applied and why it may not actually be needed. You know this but the OP probably doesn't.

Thanks for all the people responded to me.  I did more testing today and I'm sure that I'm not illusioning.  Hope you find this interesting.

I created 4 different configurations with attenuations at different stages.
1. input -10, output 0
2. input 0, output -10
3. input 0, output 0
4. input +10, output -10

The purpose is to compare 1 & 2 and 3 & 4 and see whether -10dB at different stages make a difference.  I played music from USB with Dirac off.  To remove the influence of output impedance I added a line preamp. I use RCA outputs cause that's all I have.

And it does make a difference.  Output 0 is more dynamic.  

Since I know little about electronics I will let you guys guess why.  Probably OP, or maybe one of the stages is not true digital.  

I would just use whatever configuration you think sounds best and call it good.

Dave.

OP you are in the land of make-believe. Not trying to bash you but let’s get a few things straight.

There is no such thing as attenuating the input in the sense that you’ve somehow caught it before it arrives to the DSP. You can’t change the input at all. You are misunderstanding the routing matrix.

Familiarize yourself with this signal flow: docs.minidsp.com/product-manuals/shd/signal-flow/index.html

First it’s the input — it comes as it is and cannot be changed — then Dirac, then routing matrix which gives you the chance to adjust gain, and then output which gives you another chance to adjust gain. All of this is happening in the digital domain. The master volume is also in the digital gain.

Lopping off -10db at the master volume is exactly the same as doing it in the routing or output controls. It’s all digital. You’re just moving poker chips around the same table. It’s nonsense that you “solved” your problem with dynamic power that way.

What’s also nonsense is that using Dirac requires running master volume or any other gain control at -10db at all times. miniDSP’s communications about this confuse people and don’t do half the job necessary of explaining what’s going on. This comes up all the time, and frankly I don’t know why I‘m bothering to even help them do their job here.

You need to better understand your input signal strength and your Dirac filter to know whether that’s the case. Watch your input and output meters while listening. If your signal is quiet, e.g. a a low level recording with headroom or your average phono preamp output strength, you can safely go above -10db on the master volume (yes, with everything behind the scenes at 0db). Meaning the input had the room for up to +10db of Dirac anyways so it makes no difference if you run master volume at 0db in that case.

I doubt you’re up for running a loopback through REW and a PC soundcard, but until you do you have no idea what Dirac is doing to the signal. A Dirac off-on microphone measurement can’t possibly catch everything that’s going on. Maybe the most your Dirac does is +3db, in which case you’re worried about -10db for nothing. Since miniDSP and Dirac won’t give you any real information about your filter (more to do with Dirac’s IP than anything), anyone who doesn’t run a loopback measurement is in the dark.

So, watch your input and output meters. See if your input gives you the space you need to run higher volume than you think. -10db is a worst case scenario because that’s the max gain that Dirac will apply in any case. Maybe your initial input strength is quiet enough, and filter gentle enough, that renders all of this -10db stuff moot.

My suggestions

• miniDSP explains this better in the manual. It’s been years of this confusion and they know it, and I asked their US dealer Deer Creek Audio about it one time in 2020 and even they said “we have no idea why miniDSP says it this way, they are totally confusing people about the digital and analog realm here.”
• It’d also be nice if they came up with a device that’s capable of more than 2V RCA / 4V XLR output. Having extra analog output headroom built into the system, even if it’s just a doubling for +3db or quadrupling for +6db would go a long way to mitigating the problems people have with Dirac creating +10db filters in the background with no information transparency. PEQ can apply even more gain.
• They need to implement a simple clipping light into the PC software or ideally on the front panel. Flash something if the output is clipping. This is the kind of thing that is extremely valuable to have on a device like this with enormous DSP power. Watching the meters closely can only take us so far and in any case is a burden, we just need to know if it’s clipping or not. Yes or no.

I don't see any reason to insult the OP here.  But your points are just fine.
Subjective evaluation is, by definition, beyond discussion.  If he says he's hearing something associated with these gain structure changes, I'm inclined to believe him.
Altering nominal levels within the gain structure certainly has the potential to make audible changes......depending upon what other programming is going on.  So, it's possible.

These sort of queries are nearly impossible to answer/address because there's no objective analysis happening.  It's better to just let an audiophile experiment and learn on their own.....in my opinion.  Subjective analysis is, by definition, not objective. 

Dave.

I see this differently, as a perfect example to prove just how far afield the subjective audiophile world lives.

It’s one thing when they’re telling us to trust them, and they’re talking about swapping tubes in crazy load-variant, noisy, distorted power amps, run into speakers with atrocious frequency responses, resonant cabinets, and poor directivity. Not to mention cables with no published LCR specs and other voodoo products scattered around the system. There’s no way to untangle that mess and pin them down.

But here we have someone telling others that they fixed a problem by shaving -10db from a different area of the exact same digital domain. No, sorry, can’t abide that.

-10 + 0 = -10
0 + -10 = -10

It’s that simple. OP fixed nothing this way.

What’s really nuts is you can give an overly subjective audiophile a DSP device like this, and they still manage to make it about their hocus pocus. That’s what blows me away. They have no inkling of “my usual baloney doesn’t remotely apply here” unless we tell them.