Prevent clipping

Is there a way in Audirvana to preventing distortion or clipping when upsampling or Plugins?

Yes… You are able to reduce the gain before upsampling in the Upsampling Module… :wink: :+1:

:notes: :eye: :headphones: :eye: :notes:

I can’t find anything either in upsampling or processing

What version of Audirvana are you using?

Audirvana Studio 2.8.2 (20802)

Mac OS

Okay… sorry… This is available when upsampling to DSD… I upsample/modulate all PCM files to DSD128…

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You’ll just need to manage the overall gain structure in the plug-in(s) or in the Volume setting… You can try limiting the bit-depth to 24bit…

Thank you very much. How I choose the values of two parameters of dsd upsampling?

Read the the online Help Guide information pointed-to in this screen-shot to get a better understanding as to what the settings/parameters will affect…

I’m not sure which parameters you are speaking about, however, you will need to study a bit about Sigma-Delta Modulator Filter Types in order to change this from the default DSD Sigma-Delta Modulator Filter Type settings… Otherwise just experiment with what sounds best with your system… I utilize r8Brain in concert with the FIR filter settings of my DSD-centric DAC…

:notes: :eye: :headphones: :eye: :notes:

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The bandwidth should be set at maximum. This allows the full audible range to pass through the low-pass filter.

Stop band attenuation should also be set at maximum. Have a look at the 3 graphs here:

See the orange graph? That’s maximum attenuation. That is closest to the behavior of an ideal filter, allowing maximum possible bandwidth up to the Nyquist setting, then as quickly as possible blocking all frequencies above the Nyquist setting. As you can see from the green and blue graphs, setting the stop band attenuation to lower settings creates a more gradual response slope, which means frequencies below the Nyquist setting are being artificially limited and thus the output is less faithful to the source.

Safe volume reduction before upsampling should be in the range of -3 to -6db. Try -3db and increase if you hear any distortion.

What else would you like to know?

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Thank you very much,
I’m speaking about

  1. r8brain bandwidth (% Nyquist)
    and
  2. r8brain stop band attenuation (dB)

Thank you very much.
I’ll tray then will come back with questions?
Very kind of you.

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Here I am

So, just setting maximum bandwidth and band attenuation in all situations?

Filter type?

Yes.

You can keep it at 7th order or try 8th order. Sigma-delta modulation creates lower frequency signal and higher frequency noise, and these higher order modulators push the noise to the highest possible frequencies, making it easier for the DAC to filter them out.

One other thing, what is the maximum input rate for your DAC? If your DAC will accept it and the computer will produce it, I would suggest trying DSD256. Otherwise you can try DSD128 and compare it to the highest rate PCM your DAC will accept and see what you like better.

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I’ve got two little DAC but with BurrBrown and AKM. Both support DSD512 Native

Filter A, B, C ?

From the online User Guide information that I pointed you to in my screen-shot:

For the R8Brain algorithm, you have access to the following settings:
○ Bandwidth: this indicates the limit of the low-pass filter as a % of the Nyquist frequency (half the sampling frequency). The slope of the filter is low at 74% and very steep at 99.5%. Make sure that this does not induce too much brightness (i.e. highs are too aggressive).
○ Cutoff Band Attenuation: this is the setting of the slope of the low pass filter expressed in dB per octave. With the maximum slope (218 dB), there may be too much brightness (i.e. the highs are too aggressive)
○ Phase: All low pass filters have some level of overshoot. The steeper the slope of the filter, the greater the overshoots. There are two types of overshoot: pre-oscillations and post- oscillations. The former are audible as a “pre-echo” arriving before the signal itself, and are the least natural to hear. You can choose between a filter setting with linear phase, but with an equal level of pre- and post-oscillation or a minimum phase filter, with no pre-oscillation, but non-linear phase distortion.

. Very small incremental changes have a big impact on the sound output…

Thank you very much. So is not necessary the max value of all this parameters. If I understood well an high value is often associated with an aggressive high frequencies

And the modulator filter type? How can I choose what’s using?

Hi Jud,
would this also apply if upsampling to other than DSD?
I assume it does… As this appears to be the default setting.

Try the maximum values first. Unless you’re a young woman you may not even be capable of hearing those “aggressive high frequencies” that are actually a part of the true source sound. So don’t start modifying the sound from what is actually correct unless you need to in order to enjoy it.

The “A” modulators are there for DACs that become unstable with the B modulators. If your DAC works fine with the B modulators, use them. C is identical to B7 so you can use them interchangeably. (I don’t know why C is shown separately; I was told the two were identical by the person who developed Audirvāna’s modulators.)