Does convolution involve sample rate conversion?

Iansr

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Let me first explain why I am asking this question. In the past I have experimented with a MiniDSP 4x10HD unit to try and determine whether it was completely transparent. I did this by inserting it into a passive system between the digital source and my DAC i.e. I used the digital inputs and outputs. No filters were applied so in theory the signal was just passing straight through the unit. I compared the sound with it in the chain and without it. I did not take any measurements, but I didn’t need to; there was a significant degradation in sound quality with the unit in the chain. As I understand it, the only thing the 4x10HD was doing in this set up was a sample rate conversion on the incoming digital signal. My conclusion was that it must be the SR converter in the unit that was somehow causing the degradation - most likely hardware related.

So this prompts potentially 2 questions:

1. Does convolution involve SR conversion when applying AL generated filters?
2. If it does, how is it handled and is it likely to be more “transparent” than it evidently is in MiniDSP kit? (Convolution takes places entirely in the digital domain so I can see there might be a difference.)

Can anybody provide any insight on this?
 

Ofer

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Well I can't help you with this. I find what you are saying interesting. In my setup the signal chain is as you have experimented with. The MiniDSP HDx10HD is used for volume control, is some cases for switching between sources. The volume control using a remote control is very user friendly. How did you find the signal degraded? Can you elaborate? What solution do you use today?
As a side note I can attest that the sonic improvement achieved with AL is dramatic.
 

Iansr

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Well I can't help you with this. I find what you are saying interesting. In my setup the signal chain is as you have experimented with. The MiniDSP HDx10HD is used for volume control, is some cases for switching between sources. The volume control using a remote control is very user friendly. How did you find the signal degraded? Can you elaborate? What solution do you use today?
As a side note I can attest that the sonic improvement achieved with AL is dramatic.
The best way I can answer is to use a cliche I’m afraid, but it does describe the effect; it was as though a heavy curtain had been drawn in front of the speakers.
 

juicehifi

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Let me first explain why I am asking this question. In the past I have experimented with a MiniDSP 4x10HD unit to try and determine whether it was completely transparent. I did this by inserting it into a passive system between the digital source and my DAC i.e. I used the digital inputs and outputs. No filters were applied so in theory the signal was just passing straight through the unit. I compared the sound with it in the chain and without it. I did not take any measurements, but I didn’t need to; there was a significant degradation in sound quality with the unit in the chain. As I understand it, the only thing the 4x10HD was doing in this set up was a sample rate conversion on the incoming digital signal. My conclusion was that it must be the SR converter in the unit that was somehow causing the degradation - most likely hardware related.

So this prompts potentially 2 questions:

1. Does convolution involve SR conversion when applying AL generated filters?
2. If it does, how is it handled and is it likely to be more “transparent” than it evidently is in MiniDSP kit? (Convolution takes places entirely in the digital domain so I can see there might be a difference.)

Can anybody provide any insight on this?
A sample rate conversion or not is not up to the correction filters created by Audiolense, but how they are used.

If you e.g. use jriver as a player, it will resample the filters, which will always be 100 times better than sample rate converting a stream of music.

If you use the Audiolense convolver you will get a sample rate conversion of the correction filter that is as perfect as it gets ( I assume that goes for Jriver as well) ... and I expect that to be auditory vice perfect.

But sample rate conversions of the audio stream can be anything between seriously degrading and virtually pretty good.

Bottom line: You can have a virtually perfect correction filtert that is converted, but conversion of an audio stream is a lesser solution. OTOH, back in the days, aftwer modifying a Tact Room correction unit, I was unable to hear its asynchronous sample rate conversion. Nevertheless I recommend an unconverted bitstream from source to dac.
 

Iansr

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Bernt
That’s very informative, thank you. An unconverted bitstream from source to DAC must certainly be preferable - that is what I was hoping for but I didn’t know if it was possible in practice.

So, just to confirm I’ve understood correctly: with JRiver and the AL convolver, the original digital signal remains unconverted and instead the convolver converts the filters to the same sample rate / bit depth as the source signal, thereby facilitating the application of the filters to the signal - otherwise known as the convolution.

Have I got that right?
 

whoareyou

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Bernt
An unconverted bitstream from source to DAC must certainly be preferable - that is what I was hoping for but I didn’t know if it was possible in practice.
I think the exception to this is DSD (unless someone knows of app that can natively process this format), which must be converted for processing.

For me, the conversion + processing via convolution filter provides a better result than the native DSD. Of course that is with my room and with as much treatment as allowed by law :)

As far as JRiver's resampling of the filters, does anyone know if they adjust the filter's sample size? If they do not, couldn't you run into resolution issues?
 

Mitchco

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Hi @whoareyou re: JRiver resampling filters, I would expect so as double the sample rate you need to double the filter length to keep the same resolution. I know I don't see any visual indication that this is happening in JRiver, so the only way to know for sure is either post a note on JRiver's forum or run your own tests.

Some convolver resamplers seem to have issues: https://community.roonlabs.com/t/roon-adjusting-convolution-filter-tap-length/161188

As a side note, HLC has a new resampler that will be released shortly.
 

Ofer

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The best way I can answer is to use a cliche I’m afraid, but it does describe the effect; it was as though a heavy curtain had been drawn in front of the speakers.
OK. so playing files from the computer (not lossless streaming) in original bit rate does sound somewhat better yet it is much more cumbersome. Did you find a digital preamp that doesn't resample the stream like the miniDSP (it looks like all miniDSPs do this)? it seems that there aren't many digital preamps without DAC.
 

Iansr

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OK. so playing files from the computer (not lossless streaming) in original bit rate does sound somewhat better yet it is much more cumbersome. Did you find a digital preamp that doesn't resample the stream like the miniDSP (it looks like all miniDSPs do this)? it seems that there aren't many digital preamps without DAC.
I’m not sure what you mean by digital preamp?
 

Ofer

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Marantz original 5E CD
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In my signal chain it goes as follows:
digital source (TV/CD) > 4x10HD used for source swiching and volume control>computer+RME ff400 used for DSP (audiolese filters via HLC).
So the miniDSP serves as a digital pre amplifier. Is this how you used it?
 

Iansr

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No I was considering using it as a crossover, but I wanted to know if it was transparent. Hence my experiment.
 

Ofer

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Marantz original 5E CD
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I see. I actually did use it as a xover before going full computer DSP.
 

juicehifi

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Bernt
That’s very informative, thank you. An unconverted bitstream from source to DAC must certainly be preferable - that is what I was hoping for but I didn’t know if it was possible in practice.

So, just to confirm I’ve understood correctly: with JRiver and the AL convolver, the original digital signal remains unconverted and instead the convolver converts the filters to the same sample rate / bit depth as the source signal, thereby facilitating the application of the filters to the signal - otherwise known as the convolution.

Have I got that right?
Yes
 
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