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Building SOTA system with subwoofer optimization

PortalKeeper

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Hello, I’d say I’m somewhat of a noob here.

I have an Audiophonics lpa-s400et power amp and kef q950 towers. I would like to build a system using at minimum two subwoofers with MSO, Dirac, REW, or some other software I currently know nothing about to correct for room modes in my 25’x11’x7.25’ (length by width by ceiling height) basement. I do make music and would like the top of the line ADC that is reviewed here (I think this is the RME ADI-2 pro, although this DAC doesn’t appear to have been reviewed individually by Amir yet, only in comparison). I would like to use a pro dac since I was dealing with speaker hiss (pro dac will let me turn gain to low on my power amp) with my last DAC (Focusrite Scarlett Solo 3rd gen) but they don’t seem to have SOTA SINAD levels which I really value so I thought maybe the Topping pre90 would be a great solution here. I would like to use balanced connections for the entire system.

I would really appreciate any help in setting this system up. Also, I would like to know if there is any way to make this system using a DAC that Amir reviewed with a SINAD>122. The reason for this is I don’t think any minidsp product can achieve this high of a SINAD and it seems they are the main solution for multiple sub optimization.

Thank you for your help.
 

HarmonicTHD

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Hello, I’d say I’m somewhat of a noob here.

I have an Audiophonics lpa-s400et power amp and kef q950 towers. I would like to build a system using at minimum two subwoofers with MSO, Dirac, REW, or some other software I currently know nothing about to correct for room modes in my 25’x11’x7.25’ (length by width by ceiling height) basement. I do make music and would like the top of the line ADC that is reviewed here (I think this is the RME ADI-2 pro, although this DAC doesn’t appear to have been reviewed individually by Amir yet, only in comparison). I would like to use a pro dac since I was dealing with speaker hiss (pro dac will let me turn gain to low on my power amp) with my last DAC (Focusrite Scarlett Solo 3rd gen) but they don’t seem to have SOTA SINAD levels which I really value so I thought maybe the Topping pre90 would be a great solution here. I would like to use balanced connections for the entire system.

I would really appreciate any help in setting this system up. Also, I would like to know if there is any way to make this system using a DAC that Amir reviewed with a SINAD>122. The reason for this is I don’t think any minidsp product can achieve this high of a SINAD and it seems they are the main solution for multiple sub optimization.

Thank you for your help.
Yes to what @staticV3 said.

As to get started with room correction download REW, watch the vids, buy a UMIK1. Then download APO and transfer the correction filters from REW. Do all this before investing into any new hardware and keep your Focusrite for the time being. (Btw it’s 95dB SINAD is not audible).

If you are recording or even listening to yourself when playing. Watch out however as APO inline with the DAW especially when running on a weak PC might introduce more latency then you want to tolerate. This becomes then also the highest priority once you start shopping for new hardware. (I use Neumann KH80,750 and MA1 where all the room correction is done in the speaker DSP and latency is clearly defined by the Neumann spec sheet).
 
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PortalKeeper

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"SINAD>122" is a fool's errand, honestly.
There is no value there, nothing that will even remotely affect hiss or sound quality.
Is this also true for recording, monitoring, and mixing though, as in for the ADC? Did you mean it is useless for playback, I.e. a DAC.
 
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PortalKeeper

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Yes to what @staticV3 said.

As to get started with room correction download REW, watch the vids, buy a UMIK1. Then download APO and transfer the correction filters from REW. Do all this before investing into any new hardware and keep your Focusrite for the time being. (Btw it’s 95dB SINAD is not audible).

If you are recording or even listening to yourself when playing. Watch out however as APO inline with the DAW especially when running on a weak PC might introduce more latency then you want to tolerate. This becomes then also the highest priority once you start shopping for new hardware. (I use Neumann KH80,750 and MA1 where all the room correction is done in the speaker DSP and latency is clearly defined by the Neumann spec sheet).
My PC has a ryzen 7 5800x 8 core 16 thread processor. I’m hoping it will be able to handle everything. And I actually just sold the Focusrite unfortunately lol
 

DVDdoug

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Is this also true for recording, monitoring, and mixing though, as in for the ADC?
Yes, but "S" (signal) can be a problem if you have a quiet or weak signal.

Noise from the actual ADC is normally insignificant.

Assuming you are recording from a microphone, acoustic noise is usually the main problem. Preamp noise is next. The head amp in a condenser mic also generates some noise.
 

DVDdoug

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My PC has a ryzen 7 5800x 8 core 16 thread processor. I’m hoping it will be able to handle everything.
Any computer can handle basic mono or stereo recording.* Muti-tracking requires more-faster data processing and real-time effects can also eat-up resources.

IMO - The best way to deal with latency is to get an interface with direct-hardware monitoring (or set-up a separate mixer for monitoring, etc.). The only issue with direct monitoring is that you can't include effects, except a few interfaces have some built-in DSP effects.



* What usually "causes trouble" is "junk" running in the background. The multitasking operating system is always multitasking and interrupting the audio, even if you're only running one application, and that's why you need buffers, and buffers introduce latency (delay). A faster. more powerful, computer can take-care of that background stuff faster so you are less-likely to get glitches and you can get-away with smaller buffers (so you can get lower latency).
.
 
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PortalKeeper

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Any computer can handle basic mono or stereo recording.* Muti-tracking requires more-faster data processing and real-time effects can also eat-up resources.

IMO - The best way to deal with latency is to get an interface with direct-hardware monitoring (or set-up a separate mixer for monitoring, etc.). The only issue with direct monitoring is that you can't include effects, except a few interfaces have some built-in DSP effects.



* What usually "causes trouble" is "junk" running in the background. The multitasking operating system is always multitasking and interrupting the audio, even if you're only running one application, and that's why you need buffers, and buffers introduce latency (delay). A faster. more powerful, computer can take-care of that background stuff faster so you are less-likely to get glitches and you can get-away with smaller buffers (so you can get lower latency).
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Thanks for that information, I use FL Studio and would like to continue using those plugins for the time being. Does multitracking eat up processing power even when just playing back a project on a DAW, or is it just when recording/monitoring a bunch of people at once with their effects set up in a DAW without any hardware along with miking an entire drum set?
 
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PortalKeeper

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Yes, but "S" (signal) can be a problem if you have a quiet or weak signal.

Noise from the actual ADC is normally insignificant.

Assuming you are recording from a microphone, acoustic noise is usually the main problem. Preamp noise is next. The head amp in a condenser mic also generates some noise.
But then why does Amir measure the SINAD of ADC’s and rank them?
 

bachatero

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But then why does Amir measure the SINAD of ADC’s and rank them?
I think because some of them are super DUPER bad (like the Dell XPS) and therefore testing keeps the manufacturers accountable for their engineering/marketing decisions.
 
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PortalKeeper

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I think because some of them are super DUPER bad (like the Dell XPS) and therefore testing keeps the manufacturers accountable for their engineering/marketing decisions.
But he also says, the higher the better. When making music it’s not uncommon to resample multiple times which I believe increases the level of previously nonexistent (to hearing) noise to now be audible. I’ve read about this somewhere here on ASR but can’t remember where.
 

DVDdoug

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But then why does Amir measure the SINAD of ADC’s and rank them?
I assume he's checking the mic input so it includes the mic preamp. But in any case, a weak input will make the S/N worse. And since you end-up amplifying more, you end-up amplifying the noise more.
Thanks for that information, I use FL Studio and would like to continue using those plugins for the time being.
I don't own a DAW so you know more than me!

Does multitracking eat up processing power even when just playing back a project on a DAW
With multi-track recording more data is flowing in.... 4-tracks requires is twice the data-rate of stereo. Higher resolution audio is also proportionally more data. Real-time mixing and playback is similarly stressful since you are simultaneously reading multiple files from your hard drive and the CPU is summing them.

or is it just when recording/monitoring a bunch of people at once with their effects set up in a DAW without any hardware along with miking an entire drum set?
Of course effects add to the processing requirements. And there is something called "PDC" (plug-in delay compensation) because most plug-ins add latency. (And of course, that's just for mixing... You can't "compensate" for delay in real-time. ;) )
 
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bachatero

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But he also says, the higher the better. When making music it’s not uncommon to resample multiple times which I believe increases the level of previously nonexistent (to hearing) noise to now be audible. I’ve read about this somewhere here on ASR but can’t remember where.
That can matter and it does in a studio setting where you just need "the best" if you're doing a lot of conversions with a lot of headroom. However, if you're just interested in regular audio playback with a DAC, even with a "pretty bad" SINAD rating, it doesn't matter because if that's the only conversion, then you're going to have all the detail left.
 
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PortalKeeper

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That can matter and it does in a studio setting where you just need "the best" if you're doing a lot of conversions with a lot of headroom. However, if you're just interested in regular audio playback with a DAC, even with a "pretty bad" SINAD rating, it doesn't matter because if that's the only conversion, then you're going to have all the detail left.
I’m in a home studio setting is the thing.
 

bachatero

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I’m in a home studio setting is the thing.
I have the same thing, and I ended up with a Motu M2. However, it looks like the M4 performs a lot better for not much more. Consider that or the UltraLite Mk5 as value kings and wait until you need to do something specific before going straight to the ADI.
 

HarmonicTHD

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But he also says, the higher the better. When making music it’s not uncommon to resample multiple times which I believe increases the level of previously nonexistent (to hearing) noise to now be audible. I’ve read about this somewhere here on ASR but can’t remember where.
Latency only becomes important if you are for example playing guitar and run the guitar through the DAW (effects) and then listen to the signal. It is irritating if you hear the sound delayed from you actually picking the string. Same for singing or piano. People differ, but latency somewhere above 10ms usually becomes irritating. And even with the most powerful PCs it is a watchout.

Of course no problem if you only playback or dont run the eg guitar signal through the DAW.
 

kemmler3D

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Is this also true for recording, monitoring, and mixing though, as in for the ADC? Did you mean it is useless for playback, I.e. a DAC.
When it comes to recording, in theory the lower noise the better, period. Because you may want to amplify part of the recording to an extreme extent, you should start with as close to zero noise as you can. Not truly necessary if you don't need to do anything drastic to the recording, though.

But for playback, -60dB is inaudible most of the time, -80dB is inaudible almost all the time, -95dB is really inaudible in any normal playback scenario, and -122dB is just running up the score - hitting the theoretical maximums for the sake of good engineering.
 
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PortalKeeper

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When it comes to recording, in theory the lower noise the better, period. Because you may want to amplify part of the recording to an extreme extent, you should start with as close to zero noise as you can. Not truly necessary if you don't need to do anything drastic to the recording, though.

But for playback, -60dB is inaudible most of the time, -80dB is inaudible almost all the time, -95dB is really inaudible in any normal playback scenario, and -122dB is just running up the score - hitting the theoretical maximums for the sake of good engineering.
Yeah see that’s what I was thinking too.
 
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PortalKeeper

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Anyone know if the RME ADI 2/4 Pro SE has phantom power on mic inputs?
 
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