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Izotope SRC


levandier

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Well that may be a different beast then, and you may have trouble improving on it with the s/w. How does it sound with just A+, with upsampling turned off?

And does the M51's USB input support integer mode?

 

With upsampling turned off, I find there is more body to the sound but the clarity and refinement are not as good.

 

I connect via an Audiophilleo 2 USB/SPDIF converter which does support integer and direct mode.

 

Question: Is Izotope with up sampling set to NONE the equivalent of CoreAudio with up sampling set to NONE?

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  • 3 weeks later...

I have followed and read this thread over its 90 day life but I did not read through the 200 posts again to see if these questions have previously been asked/answered though I think they have.

 

Early on in the thread I made settings based on several recommendations and have enjoyed listening but now I am curious about going back for comparison. So, is there a "reset" default settings capability or do I just turn the function off? Can you save settings to enable them again when you wish? Can you have more than one set of settings?

"A mind is like a parachute. It doesn't work if it is not open."
Frank Zappa
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Saved settings sets have been requested and I think Damien said he might include such in a future release. To make A+ switch to its defaults, I think you just move the uppermost "quality" slider off of "best" and then back again. Supposedly that resets it to whatever was the "factory" settings for iZotope. (Personally, I recall the default steepness being way too steep for my tastes...)

Hope this info helps you.

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  • 2 weeks later...
The NadM51 supports USB Integer Mode and all the features A+ provides! Since this beast basically upsamples PCM to PWM( maybe

DSD?), I was trying to figure it out what is the best S/W(software) settings would improve from non upsampling with A+ .

 

If anything below 6 steepness does introduce nasties to the audio, the the AYRE settings that Alex(izotope) describe in the begging of this thread is not a good one!

 

I will keep experimenting!

 

Thanks!

 

As has been mentioned in many posts in this thread, how shallow of a filter slope one can "get away with" is highly dependednt on how high a frequency one can output into the DAC. In the case of the Ayre Minimum Phase/Slow Roll Off filter, they are oversampling to 705.6 kHz, so there are way less worries about alias products producing audible problems. If anyone is interested in the measured performance of the Ayre filter, take a look at the Sterophile measurments associated with their review of the QB-9 DAC (I have not looked, but the measurements of the DX-5 Blu Ray player shoudl have the same info as well).

If one can only oversample to 176.4, one will probably prefer a little steeper slope, as the audio band artifacts of a shallower slope will be problematic at that rate.

 

BTW, a little info on the ESS chips: these DACs feature two internal oversampling steps, the first one is 8x with a FIR filter, and the second is 8x with a IIR filter. It is possible to turn off the first 8x oversampling step, and then feed the DAC directly with 352.8 or even 705.6 oversampled/filtered data. If one were to do this, there would be no audible impact of the second filter step, as any artifacts will be way, way, way out of band, and filtered by even the most gentle analog output filter in the DAC's output stage. Note that the DAC conversion stage is ultimately running at DSD speeds, ~2.8 MHz, and can go even higher.

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In the case of the Ayre Minimum Phase/Slow Roll Off filter, they are oversampling to 705.6 kHz, so there are way less worries about alias products producing audible problems. If anyone is interested in the measured performance of the Ayre filter, take a look at the Sterophile measurments associated with their review of the QB-9 DAC (I have not looked, but the measurements of the DX-5 Blu Ray player shoudl have the same info as well).

If one can only oversample to 176.4, one will probably prefer a little steeper slope, as the audio band artifacts of a shallower slope will be problematic at that rate.

 

It is not about the target rate, but about the source rate. Artifacts begin immediately above Nyquist frequency of the source rate.

 

In the Stereophile measurements, you can clearly see that there's really heavy leakage with the "listen" filter by looking at figure 13. The image frequencies of 19 and 20 kHz that should have been filtered away are just couple of dB down. Compared to figure 12 "measure" filter where it's 90 dB down - a lot better figure. You can also see that in figure 13 the audio band spectrum becomes polluted with aliases all over, compared to figure 12.

Signalyst - Developer of HQPlayer

Pulse & Fidelity - Software Defined Amplifiers

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It is not about the target rate, but about the source rate. Artifacts begin immediately above Nyquist frequency of the source rate.

 

In the Stereophile measurements, you can clearly see that there's really heavy leakage with the "listen" filter by looking at figure 13. The image frequencies of 19 and 20 kHz that should have been filtered away are just couple of dB down. Compared to figure 12 "measure" filter where it's 90 dB down - a lot better figure. You can also see that in figure 13 the audio band spectrum becomes polluted with aliases all over, compared to figure 12.

 

Thanks Miska,

 

"In the Stereophile measurements, you can clearly see that there's really heavy leakage with the "listen" filter by looking at figure 13. The image frequencies of 19 and 20 kHz that should have been filtered away are just couple of dB down."

 

Well, this is a subjective judgement. This filter was designed by ear to provide the best performance possible in subjective terms. It is significant to note that John Atkinson mentions that the minimum phase filter used in the dCS was a lot more leaky than the Ayre "listen" option. In any case, if these relatively low level tones bothers one, one can try the more effective alias suppression of the "measure" option, as you note. And I am sure you do not mean that one would be able to hear the primary alias pair above 20 kHz.

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It is not about the target rate, but about the source rate. Artifacts begin immediately above Nyquist frequency of the source rate.

 

In the Stereophile measurements, you can clearly see that there's really heavy leakage with the "listen" filter by looking at figure 13. The image frequencies of 19 and 20 kHz that should have been filtered away are just couple of dB down. Compared to figure 12 "measure" filter where it's 90 dB down - a lot better figure. You can also see that in figure 13 the audio band spectrum becomes polluted with aliases all over, compared to figure 12.

 

Hi, Miska. Since I am interested in filtering, particularly the audible effects of different settings, I am trying to understand the following statement in the Stereophile review (from the Measurements section, in the text between figures 2 and 3):

 

The Listen filter starts to roll off a little earlier, but is down just 6dB at 22.05kHz, meaning that it will not completely eliminate a recording's pre-ringing that had been introduced by the original A/D converter.

 

Can you (and anyone else who cares to contribute) please explain a little about the effects of filter steepness and cutoff on pre-ringing from recordings, as well as how much of a problem you find this (pre-ringing from recordings) to be?

One never knows, do one? - Fats Waller

The fairest thing we can experience is the mysterious. It is the fundamental emotion which stands at the cradle of true art and true science. - Einstein

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Jud, I suspect you will find the info which you are looking for by Googling "Apodising Filters", also take a look at the Stereophile Review of the Meridian 808 CD player. I believe the most common belief is that one wants -100 dB or so at Nyquist to reduce the pre-ring from the anti alias filter in the ADC process, of course, in reality, this will depend on the exact characteristics of the anti alias filter used...

And then, there is the trade off: what do you think is worse, the pre ring from the anti alias filter in the ADC, or the amount of ringing from the filter in OSF (whether in the DAC or SW). Clearly, Ayre has come to the conclusion that reduction of ringing in the DAC OSF is important than reduction of ringing from the ADC filter used in recording, of course, YMMV!

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Jud, I suspect you will find the info which you are looking for by Googling "Apodising Filters", also take a look at the Stereophile Review of the Meridian 808 CD player. I believe the most common belief is that one wants -100 dB or so at Nyquist to reduce the pre-ring from the anti alias filter in the ADC process, of course, in reality, this will depend on the exact characteristics of the anti alias filter used...

And then, there is the trade off: what do you think is worse, the pre ring from the anti alias filter in the ADC, or the amount of ringing from the filter in OSF (whether in the DAC or SW). Clearly, Ayre has come to the conclusion that reduction of ringing in the DAC OSF is important than reduction of ringing from the ADC filter used in recording, of course, YMMV!

 

I suppose what isn't completely clear to me is how a steeper filter would *help* with pre-ringing in the recording - is the pre-ringing from the ADC above Nyquist?

 

From my own steepness setting (currently 3), it's evident I agree with Ayre. What I don't understand regarding Ayre (and would love to know more about) is that apparently their filter would be minimum phase. To me minimum phase settings with iZotope sound bad, and I'm thinking it's because of the dispersive nature of a minimum phase filter. I'm wondering how/if Ayre gets around that (if what I think I've read about Ayre's filtering being minimum phase is correct).

One never knows, do one? - Fats Waller

The fairest thing we can experience is the mysterious. It is the fundamental emotion which stands at the cradle of true art and true science. - Einstein

Computer, Audirvana -> optical to EtherREGEN -> microRendu -> ISO Regen -> iFi NEO iDSD DAC -> Apollon Audio 1ET400A Mini (Purifi based) -> Vandersteen 3A Signature.

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BTW, a little info on the ESS chips: these DACs feature two internal oversampling steps, the first one is 8x with a FIR filter, and the second is 8x with a IIR filter. It is possible to turn off the first 8x oversampling step, and then feed the DAC directly with 352.8 or even 705.6 oversampled/filtered data. If one were to do this, there would be no audible impact of the second filter step, as any artifacts will be way, way, way out of band, and filtered by even the most gentle analog output filter in the DAC's output stage. Note that the DAC conversion stage is ultimately running at DSD speeds, ~2.8 MHz, and can go even higher.

 

 

Would one turn off the filter using something fairly simple, via I2C, or would this be something that the average home DIY guy would be unlikely to be able to accomplish? I've become interested in the idea of upsampling via software and using a NOS DAC, but I also have a Buffalo DAC available. TIA - Pat

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"Would one turn off the filter using something fairly simple, via I2C, or would this be something that the average home DIY guy would be unlikely to be able to accomplish? I've become interested in the idea of upsampling via software and using a NOS DAC, but I also have a Buffalo DAC available. TIA - Pat"

 

For DIY, one can use the Buffalo III or Buffalo IIIse, both have an onboard dip switch which allows one to turn off the OSF, even without having to have a controller (like arduino etc) via I2C. Of course, if you do have a controller OSF can be turned off from their as well.

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"Would one turn off the filter using something fairly simple, via I2C, or would this be something that the average home DIY guy would be unlikely to be able to accomplish? I've become interested in the idea of upsampling via software and using a NOS DAC, but I also have a Buffalo DAC available. TIA - Pat"

 

For DIY, one can use the Buffalo III or Buffalo IIIse, both have an onboard dip switch which allows one to turn off the OSF, even without having to have a controller (like arduino etc) via I2C. Of course, if you do have a controller OSF can be turned off from their as well.

 

Thanks for getting back to me so quickly. I have an "old" 32S, do I'd have to go the Arduino route, but it's nice to see that the TP boys have made allowance for more customizing in the latest Buffs.

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And then, there is the trade off: what do you think is worse, the pre ring from the anti alias filter in the ADC, or the amount of ringing from the filter in OSF (whether in the DAC or SW). Clearly, Ayre has come to the conclusion that reduction of ringing in the DAC OSF is important than reduction of ringing from the ADC filter used in recording, of course, YMMV!

 

From ringing perspective it doesn't matter at what amount of ringing the DAC filter has if it's non-apodizing because it has no impact on the overall ringing... Having a short filter just makes it leak more.

Signalyst - Developer of HQPlayer

Pulse & Fidelity - Software Defined Amplifiers

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Well, this is a subjective judgement.

 

No, it's purely objective. :)

 

 

This filter was designed by ear to provide the best performance possible in subjective terms. It is significant to note that John Atkinson mentions that the minimum phase filter used in the dCS was a lot more leaky than the Ayre "listen" option. In any case, if these relatively low level tones bothers one, one can try the more effective alias suppression of the "measure" option, as you note. And I am sure you do not mean that one would be able to hear the primary alias pair above 20 kHz.

 

You can hear the intermodulation products of those and the low level aliases that pollute the noise floor. Essentially it will largely sound similar to a filter-less NOS-DAC. Some people like it, some people don't. I also offer those kind of filters as an option, but I never use those myself. Instead, in the designs I use, I try to both minimize the ringing and maximize the filter attenuation at the same time. It's not so much about filter design parameters but more about filter design algorithms... Once you have the design algorithm, dialing in nice parameters is easier.

 

dCS doesn't seem to leak much (-125/-135 dB), but it has some more severe aliasing issues instead (see fig 11):

dCS Debussy D/A processor Measurements | Stereophile.com

Signalyst - Developer of HQPlayer

Pulse & Fidelity - Software Defined Amplifiers

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From ringing perspective it doesn't matter at what amount of ringing the DAC filter has if it's non-apodizing because it has no impact on the overall ringing... Having a short filter just makes it leak more.

 

I would like to try to understand better. A less steep non-apodizing filter in the playback PC will not affect the total amount of ringing, even though that specific type of playback filter itself will ring less, because - why? Because it will reduce pre-ringing from the recording by a lesser amount? But wouldn't that vary with the amount of pre-ringing in the recording?

One never knows, do one? - Fats Waller

The fairest thing we can experience is the mysterious. It is the fundamental emotion which stands at the cradle of true art and true science. - Einstein

Computer, Audirvana -> optical to EtherREGEN -> microRendu -> ISO Regen -> iFi NEO iDSD DAC -> Apollon Audio 1ET400A Mini (Purifi based) -> Vandersteen 3A Signature.

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I would like to try to understand better. A less steep non-apodizing filter in the playback PC will not affect the total amount of ringing, even though that specific type of playback filter itself will ring less, because - why? Because it will reduce pre-ringing from the recording by a lesser amount? But wouldn't that vary with the amount of pre-ringing in the recording?

 

Because by definition it will pass the original ringing through as-is...

 

Sorry for the quick-and-dirty example (I don't have time to make anything more pretty now):

 

Let's take a source at 44.1k, converted from 96k dirac pulse using ordinary type of conversion:

source.png

 

If we upsample it to 176.4k through a typical non-ringing (non-apodizing) filter:

non-apodizing.png

 

If we upsample it to 176.4k through a typical apodizing minimum-phase filter:

apodizing.png

 

You can see that the non-apodizing filter that has only single cycle of ringing on both sides passes the original ringing through as-is while the apodizing one replaces the ringing with it's own. It could be as well linear phase, but I used a minimum-phase here just to make the difference more obvious.

 

Now if we inspect the difference in frequency domain...

 

Here's spectrum of the non-ringing upsampled data:

na.png

 

And here' spectrum of the apodizing minimum-phase upsampled data:

a.png

 

You can see that the non-ringing filter leaks strong high frequency images, while the apodizing minimum-phase one has high attenuation of those spurious images.

Signalyst - Developer of HQPlayer

Pulse & Fidelity - Software Defined Amplifiers

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You can see that the non-ringing filter leaks strong high frequency images, while the apodizing minimum-phase one has high attenuation of those spurious images.

 

Miska: It would be helpful to me (and maybe others) if you could take a few minutes to explain:

 

a) What you mean by a "non-ringing filter." Are you just referring to a linear-phase filter?

 

 

b) The distinction between a minimum-phase filter and one that is both minimum-phase and apodizing. As in, how are the two constructed and what do their respective impulse, passband ripple, and response curves look like.

 

Despite the fact that "minimum-phase apodizing filter" is thrown around as one phrase by a number of manufacturers when they are just offering a minimum-phase filter, everything that I have read says the two are different, and I desperately want to get a clear explanation of the difference. And how might one create/simulate a true "apodizing" filter for D/A conversion.

 

The term "apodizing" (literally, removing the foot) was first borrowed from optics and applied in reference to digital filters by Meridian/Peter Craven in his AES paper "Controlled Pre-Response Anti-Alias Filters for Use at 96kHz and 192kHz." I have not purchased the download of that paper to best understand the concept.

Since this discussion is getting around to the concept of trying to mitigate the effects of ringing embedded by the A/D converter's anti-aliasing filters, that is what Meridian first claimed to do with their "apodizing" filter, and I wish to learn more.

 

Go light on the math please. In other words, if you just refer me to the Hann Window function I will remain as ignorant as before.

 

Thanks!,

ALEX

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a) What you mean by a "non-ringing filter." Are you just referring to a linear-phase filter?

 

It is generally used by manufacturers to refer to anything that has only very few cycles in the impulse response, something like fig 3 here:

Ayre Acoustics QB-9 USB DAC Measurements | Stereophile.com

 

In my example it was polynomial interpolation with small number of steering points.

 

Here's another example of FIR with some ringing, but very small amount (< 10 cycles impulse response), it doesn't have much impact on the original ringing either, but cuts off the ultrasonic images a bit more effectively, and doesn't have roll-off in 10-20 kHz region (more aggressive than Ayre's "listen" filter):

non-apodizing-2.png

na-2.png

 

b) The distinction between a minimum-phase filter and one that is both minimum-phase and apodizing. As in, how are the two constructed and what do their respective impulse, passband ripple, and response curves look like.

 

Apodizing filter looks just like any linear or minimum phase filter, but it needs to be fairly efficient. Essentially it needs to be able to filter out the original anti-alias filter.

 

Despite the fact that "minimum-phase apodizing filter" is thrown around as one phrase by a number of manufacturers when they are just offering a minimum-phase filter, everything that I have read says the two are different, and I desperately want to get a clear explanation of the difference. And how might one create/simulate a true "apodizing" filter for D/A conversion.

 

Apodizing filter in itself doesn't have anything to do with minimum-phase, it can be linear or minimum phase or anything between the two. It's defined by other parameters than phase. But commonly people want to use minimum-phase filter as replacement response.

 

I don't want to go to design specifics, and it would anyway involve maths...

 

The term "apodizing" (literally, removing the foot) was first borrowed from optics and applied in reference to digital filters by Meridian/Peter Craven in his AES paper "Controlled Pre-Response Anti-Alias Filters for Use at 96kHz and 192kHz." I have not purchased the download of that paper to best understand the concept.

Since this discussion is getting around to the concept of trying to mitigate the effects of ringing embedded by the A/D converter's anti-aliasing filters, that is what Meridian first claimed to do with their "apodizing" filter, and I wish to learn more.

 

I think it's bad name for a filter that replaces original filter's impulse response. I use it only because it has become common way to describe such behavior. I don't think there's anything special about "apodizing filters" that would be worth AES paper.

 

There are two alternative sources for the ringing, A/D converter in case the recording was made at the final sampling rate (for example 44.1k). Or alternatively software SRC in case recording was made for example in 96k and then converted to 44.1k at mastering stage. iZotope is example of such converter used for mastering.

Signalyst - Developer of HQPlayer

Pulse & Fidelity - Software Defined Amplifiers

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It is generally used by manufacturers to refer to anything that has only very few cycles in the impulse response, something like fig 3 here:

Ayre Acoustics QB-9 USB DAC Measurements | Stereophile.com

 

In my example it was polynomial interpolation with small number of steering points.

 

Here's another example of FIR with some ringing, but very small amount (< 10 cycles impulse response), it doesn't have much impact on the original ringing either, but cuts off the ultrasonic images a bit more effectively, and doesn't have roll-off in 10-20 kHz region (more aggressive than Ayre's "listen" filter):

[ATTACH=CONFIG]6200[/ATTACH]

[ATTACH=CONFIG]6201[/ATTACH]

 

 

 

Apodizing filter looks just like any linear or minimum phase filter, but it needs to be fairly efficient. Essentially it needs to be able to filter out the original anti-alias filter.

 

Apodizing filter in itself doesn't have anything to do with minimum-phase, it can be linear or minimum phase or anything between the two. It's defined by other parameters than phase. But commonly people want to use minimum-phase filter as replacement response.

 

I don't want to go to design specifics, and it would anyway involve maths....

 

I think it's bad name for a filter that replaces original filter's impulse response. I use it only because it has become common way to describe such behavior.

 

Miska, I appreciate very much your help in clarifying this (especially for an amateur like me).

 

There is always the chance that I am completely mistaken, but it looks to me from what you have said, combined with looking at pages 3-4 of the Ayre white paper here - http://www.ayre.com/white_papers/Ayre_MP_White_Paper.pdf - that one thing which should be done to make what is commonly referred to as an "apodizing" filter using is to set the cut-off frequency below "Nyquist."

 

One additional question I have, Miska, is why people commonly want to use a minimum-phase filter, if you have any comment on that. Is it simply to eliminate pre-ringing?

One never knows, do one? - Fats Waller

The fairest thing we can experience is the mysterious. It is the fundamental emotion which stands at the cradle of true art and true science. - Einstein

Computer, Audirvana -> optical to EtherREGEN -> microRendu -> ISO Regen -> iFi NEO iDSD DAC -> Apollon Audio 1ET400A Mini (Purifi based) -> Vandersteen 3A Signature.

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jud, I will try and help some. Hopefully miska will review my comments and add corrections ;-)

 

I am going to do this in the terms as commonly used by manufacturers, which may, or not, be technically correct. Consider first the two sources of ringing: the ringing introduced by the anti-alias filter at the ADC in the recording process, this is encoded on the CD along with the music, and the ringing introduced by the filter in the oversampling engine in the DAC (or per this discussion, in the SW).

 

An apodising filter is used to eliminate/reduce the ringing produced by the anti alias filter at the time of A/D conversion during recording-it has nothing to do with minimum phase

 

A minimum phase filter describes a filter designed to move the ringing energy produced in the oversampling engine, from pre-ringing to post ringing. A minimum phase filter alone does not reduce the overall ringing, it just moves the energy from before the impulse to after the impulse. The trade off with this type of filter is that it is not linear phase, there will be some phase shift with any filter which produces an asymmetric impulse response.

 

A slow roll off filter can be used to reduce the overall amount of ringing energy from the oversampling process. The trade off with slow roll off, is that it allows alias images through.

 

Ayre's "listen" filter is a minimum phase/slow roll off filter (not apodising), as such it is designed to eliminate all pre-ringing, and reduce all post ringing to just one cycle. The trade off is that it has a slight droop in high frequency response, and it allows alias energy through. Ayre's "measure" filter is minimum phase (no pre ring) but with a steeper roll off, allowing for more extended high frequency response, and better suppression of alias products, but lots more (post) ringing energy.

 

Ayre feels that their "listen" filter offers the best compromise in subjective listening tests, that is their decision. All of these filter approaches, and balancing the various response parameters, is a compromise in one area for another, at least when we are considering 44.1 source material. With higher sample rates, the compromises are reduced, as the filters can operate at higher frequencies, where the artifacts are less audible.

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To put it short:

 

1) Non-apodizing over-/up-sampling filter doesn't change overall ringing in significant way. It only defines frequency content (or absence) above source Nyquist. Ringing from the source material dominates ringing behavior.

 

2) Apodizing over-/up-sampling filter replaces the ringing from source material partially or wholly with it's own. Frequency content (or absence) above source Nyquist is also defined by the filter. Filter dominates ringing behavior.

 

Thus, regarding the Ayre QB-9, with "measure" filter the Ayre's filter partially dominates ringing behavior, while the "listen" filter doesn't practically change the ringing behavior. Both have their own amount of ultrasonic leakage from fairly low of the "measure" filter to very high of the "listen" filter. (as you can see, this in line with what JA of Stereophile wrote about the two)

 

(1) as design objective doesn't impose any limitations on the filter design, while (2) imposes certain boundaries for the filter design.

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To put it short:

 

1) Non-apodizing over-/up-sampling filter doesn't change overall ringing in significant way. It only defines frequency content (or absence) above source Nyquist. Ringing from the source material dominates ringing behavior.

 

2) Apodizing over-/up-sampling filter replaces the ringing from source material partially or wholly with it's own. Frequency content (or absence) above source Nyquist is also defined by the filter. Filter dominates ringing behavior.

 

 

So to me from this and some additional reading, it looks like a fairly straightforward tradeoff: The highest frequency extension vs. reduction of ringing (from the original recording) and aliasing.

 

Perhaps one might use the apodizing filter for 44.1 material, and non-apodizing for higher res. Wonder if any 44.1 material might have an apodizing ADC filter used on it, so it might benefit less from an apodizing DAC filter.

One never knows, do one? - Fats Waller

The fairest thing we can experience is the mysterious. It is the fundamental emotion which stands at the cradle of true art and true science. - Einstein

Computer, Audirvana -> optical to EtherREGEN -> microRendu -> ISO Regen -> iFi NEO iDSD DAC -> Apollon Audio 1ET400A Mini (Purifi based) -> Vandersteen 3A Signature.

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The highest frequency extension vs. reduction of ringing (from the original recording) and aliasing.

 

That's where the design algorithms step in. Designing a design algorithm that gives you optimal result, minimal ringing without harming the top octave.

 

Also shape and properties of the transition band has impact on the sound.

 

Perhaps one might use the apodizing filter for 44.1 material, and non-apodizing for higher res.

 

Why? For hires it's much less challenging to nicely squeeze it in because of extra bandwidth...

 

Wonder if any 44.1 material might have an apodizing ADC filter used on it, so it might benefit less from an apodizing DAC filter.

 

That is also possible, but not very common.

 

You can check out ADC datasheets and also the SRC data available for various pieces of software and decide suitable filter parameters form that. You can also just test it with various ADCs by using AWG to generate a suitable test pulse for input. You already cover quite a lot if you check it against ProTools. :)

Signalyst - Developer of HQPlayer

Pulse & Fidelity - Software Defined Amplifiers

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Why? For hires it's much less challenging to nicely squeeze it in because of extra bandwidth.

 

My thought was this might be like the 44.1 apodizing ADC filter case: Perhaps less ringing or chance of aliasing to start with.

One never knows, do one? - Fats Waller

The fairest thing we can experience is the mysterious. It is the fundamental emotion which stands at the cradle of true art and true science. - Einstein

Computer, Audirvana -> optical to EtherREGEN -> microRendu -> ISO Regen -> iFi NEO iDSD DAC -> Apollon Audio 1ET400A Mini (Purifi based) -> Vandersteen 3A Signature.

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