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[TimoBeil95]

Hello guys,

I am usig the convolution engine of MC in combination with acourate generated filters. In earlier times, when using normal passive crossovers and my old DAC MC showed in the DSP Studio often peak level values between 70 and 100 %.

But Now with this new combination I have maximum 20% max peak and the sound is already very loud.

No problem so far, but could this be and indicator for dynamic loss and a wrong leveling configuration in the Acourate - DAC - Amplifier chain? Or do I missunderstand something?

BTW: Is there any unneccesary processing in the chain? Picture in attachment.

[mattkhan]

100 to 20%  to ~ -14dB so your volume levelling adjustment eats up most of that, your filter will normally be peaking at 0 hence is cutting everywhere else so that would use up a few more dB too.

If you want to dig into this further to be certain then I recommend using some test signals in a controlled test, capture the output and compare to the input to ensure it is exactly as you expect.

[TimoBeil95]

So this is normal and there is no loss in quality and bit depth?

Regards

[RD James]

I don't think the "peak level" display in DSP Studio is accurate at all.
Watching it while playing some music, I never saw it go above 50%, while one of my DSP Plugins reported peaks above -10dB.

[mattkhan]

So this is normal and there is no loss in quality and bit depth?

probably normal

I don't think the "peak level" display in DSP Studio is accurate at all.
Watching it while playing some music, I never saw it go above 50%, while one of my DSP Plugins reported peaks above -10dB.

50% is -6dB

[TimoBeil95]

Thank You,

maybe I should do some measurements with acourate again and adjust the drivers again at a lower volume with the amplifier trimmers. Output volume when recording in Acourate is always between -20db and -30db. Then volume should be about 83dB in Listening position? But I think it was louder. So maybe when setting lower amplifier volumes in the analogue domain, I could give more power in the digital domain and should have more bit depth???

What do You think?

[RD James]

50% is -6dB

Decibel is always really confusing because volume, power, and voltage all scale differently.
-6dB is "88% volume" according to Media Center.
If that's also "50% peak level" then I'd prefer DSP Studio reported peak level in decibels.
 

maybe I should do some measurements with acourate again and adjust the drivers again at a lower volume with the amplifier trimmers. Output volume when recording in Acourate is always between -20db and -30db. Then volume should be about 83dB in Listening position? But I think it was louder.

Unless you have used a sound level meter to calibrate the volume controls in Media Center, you have no idea what the output from your speakers is based on digital values.
 
It is a good idea to try and minimize digital attenuation though, for the highest signal to noise ratio.
Be sure to check that Media Center is also set to the TPDF dither option.

[mattkhan]

Decibel is always really confusing because volume, power, and voltage all scale differently.
-6dB is "88% volume" according to Media Center.
If that's also "50% peak level" then I'd prefer DSP Studio reported peak level in decibels.
 Unless you have a sound level meter to calibrate it, you have no idea what the output from your speakers is based on digital values.

tbh I don't really understand the point of the percentage value listed in internal volume but, for reference, it's described in https://yabb.jriver.com/interact/index.php?topic=69939.msg471528#msg471528 though.

I agree that (voltage) dB relative to full scale is preferable in DSP studio, showing this as a percentage is a bit odd and hard to interpret (for the reasons that give rise to use of dB here in the first place).

[mattkhan]

Thank You,

maybe I should do some measurements with acourate again and adjust the drivers again at a lower volume with the amplifier trimmers. Output volume when recording in Acourate is always between -20db and -30db. Then volume should be about 83dB in Listening position? But I think it was louder. So maybe when setting lower amplifier volumes in the analogue domain, I could give more power in the digital domain and should have more bit depth???

What do You think?

you're basically describing a gain structure problem, first order of business is making sure nothing clips all the way through the signal chain and then second is optimising SNR. You don't need to redo your acourate filters for this, it's just a question of optimising levels at each stage of the signal chain which means you need to know what voltage comes out your DAC and the input sensitivity of your amp(s) (repeat for each step in the chain).

[TimoBeil95]

Okay my dac is a scarlett focusrite 18i20. My amplifiers are l15d pro modules. AFAIK the normal l15d have a gain of 36x. But how to do this now in the pracrical way? Acourate filters could be renewed though, because I have to do time adjustments again, because of moving to a new place

So how can I optimize the SNR?

Regards

[TimoBeil95]

Have readjusted the trimmers and renewed the filters and now I have the wished levels . So when I listen to my "standard" volume (100%) I am between 40% and 90% peak level and I have certainly more dynamic and power in music

Regards

[Hendrik]

So how can I optimize the SNR?

I use a passive analog attenuator to lower the volume some before amplification (I also have an 18i20, which has a relatively high output level being a Pro device), which has the advantage of also lowering the noise-floor - which allows me to have the digital volume higher (ie. better SNR)

[mattkhan]

Have readjusted the trimmers and renewed the filters and now I have the wished levels . So when I listen to my "standard" volume (100%) I am between 40% and 90% peak level and I have certainly more dynamic and power in music

Regards

are you optimising for music or movie content? It seems like movie given what looks like use of an LFE channel in your PEQ setup.

the amp seems to be DIY so not sure what the input sensitivity is, a quick google suggests about 1.5V but hard to be sure. Given it seems to be DIY, I guess you must know this.

the 18i20 appears to put out +16dBu which is 4.9Vrms.

If these numbers are correct this gives you about 10dB more coming out than you need to drive your amps. If you are now peaking at 40-90% then that suggests you are now over driving your amps and the final output will depend on how your amps deal with this. If it really sounds "more dynamic" then that may just mean loudness war style louder as you drive the amps into clipping.

[TimoBeil95]

mhh,

But what I have done, was setting the potis of my L15Ds lower, (analogue attenuation) so the output voltage of the DAC should be properly attenuated or not? I measured again with acourate, but this time not with my previous 90db in the sine swipe, just ~80dB. The idea was, to have the maximum output voltage of the DAC available when needed, what leads to the maximum resolution, and then have set the correct setting of the trim potentiometer of the amps, which divide the voltage in the analogue domain. Was this wrong?

EDIT: I am optimising for movie and audio (also multichannel)

[mattkhan]

Ok then yes, I didn't realise that the trimmers referred to attenuating in the input to the amp. Note that you can use that extra output voltage of the sub output to recover the lfe gain.

[TimoBeil95]

Note that you can use that extra output voltage of the sub output to recover the lfe gain.

Thank You very much, but how to accomplish that? My Mains are crossovered at 150hz to the lfe via convolution filters and how can I then put the 10db gain to the lfe channel? with a attenuating all channels except the sub or with boosting only the sub channel before convolution?

[mattkhan]

Thank You very much, but how to accomplish that? My Mains are crossovered at 150hz to the lfe via convolution filters and how can I then put the 10db gain to the lfe channel? with a attenuating all channels except the sub or with boosting only the sub channel before convolution?

Sorry ignore that last comment, not sure what I was thinking of  ?

[TimoBeil95]

okay,

But that induced me to think further, how to recover the gain. How do you think this could be done without attenuate so much in the digital domain. So instead of setting a -15/-5 on main/sub channels and recovering 15dB it could be done maybe directly in acourate, So I generate two sub filters, one for the .1 lfe channel and one for the low-passed channels. The first is done with a measurement where the sub is lowered on the interface with 10db, the other measurement is done normally. Is there an advantage over the "-15/-5 +15" method or is it a thinking error and results in the same outcome?

[mattkhan]

okay,

But that induced me to think further, how to recover the gain. How do you think this could be done without attenuate so much in the digital domain. So instead of setting a -15/-5 on main/sub channels and recovering 15dB it could be done maybe directly in acourate, So I generate two sub filters, one for the .1 lfe channel and one for the low-passed channels. The first is done with a measurement where the sub is lowered on the interface with 10db, the other measurement is done normally. Is there an advantage over the "-15/-5 +15" method or is it a thinking error and results in the same outcome?

there is no way to avoid that initial -10dB extra attenuation on the main channels, there is no way to avoid that fact that a 1 on the LFE channel counts for more than a 1 on a main channel in SPL terms (if you get what I mean).