Izotope SRC

[Jud]

From the Audirvana 1.6 wish list thread:

 

I mean, when I read phrases like "Anti-aliasing: Alias suppression in the stop-band." -- well, that doesn't help me understand what that parameter does. What's the "stop-band," for example? And why would I want to suppress the alias?

 

Hey, perhaps understanding Izotope parameters is only for music-tech types. But I bet there might be even further ways to explain it in plain English.

 

 

My attempted explanation (including the usual call for corrections where I've messed up): An alias is an "image" or "ghost," if you like, that is "mirrored" around the Nyquist frequency. Let's say you have a Redbook sample rate and in the music you're sampling there's signal at 10kHz, that generates second and third harmonics of the 10kHz signal at 20 and 30kHz. If the 30kHz signal isn't suppressed, it's above half the 44.1kHz sample rate. That will be "folded" or "mirrored" around the Nyquist frequency of 22.05kHz. 30-22.05=7.95; 22.05-7.95=14.1kHz. Allowing that 30kHz signal, therefore, will cause a false "mirror" signal - i.e., distortion - at 14.1kHz, in the audible range.

 

You don't want this type of thing to happen, so you want to eliminate or at least substantially lower signals between the top of the audible range and the sample rate. You'd want the anti-aliasing filter, all else being equal, to have a stop band (where the filter has maximum effect) covering as much of the frequency range between audibility and sample rate as possible.

 

For iZotope settings purposes: I haven't personally heard any advice other than to leave the setting at default.

[Jud]

Apologies for monopolizing the thread, but I think the following is interesting and may be helpful: http://www.wolfsonmicro.com/documents/uploads/misc/en/Ultra_High_Performance_DAC_whitepaper.pdf

 

Figure 7 on page 6 and the accompanying text at the top left of page 7 are particularly noteworthy to me, with my interest in filters that have low "dispersion" or "group delay."

[Miska]

Apologies for monopolizing the thread, but I think the following is interesting and may be helpful: http://www.wolfsonmicro.com/documents/uploads/misc/en/Ultra_High_Performance_DAC_whitepaper.pdf

 

The stuff they explain there is what you find on WM8741/8742 DAC chips.

[Blake]

[wdw]

+1

Wdw

[JohanS]

Jud, thank you. I've been reading your posts and each time it becomes a bit more clear to me! I've some new music to listen to, so I'll leave the settings untouched for a couple of days to enjoy the music. :-)

 

Johan

[Jud]

The stuff they explain there is what you find on WM8741/8742 DAC chips.

 

Thanks Miska. For me it was helpful because, like most people new to a subject, I tend to over-generalize. For example, having read in a couple of places that minimum phase filters are dispersive, I was puzzled why Ayre or you would want to use one. The Wolfson paper shows a minimum phase filter that appears to have reasonable group delay characteristics, so at least I now understand such a thing is possible. It (or was it a paper about the Meridian filter?) also shows a relatively steep filter eliminating ringing from source material, while I thought a steep filter was not the way to go if you wanted to eliminate ringing.

 

So I'm slowly learning from all sources.

[carsten s chr]

Hi There !..... :-)

 

I've been playing arround with Audirvana for quite some time - always leaving iZotope settings at the default settings !....

 

Yesterday I thought it was time to play a bit with above mentioned settings !.....

 

In general, I believe that all kind of filters has to be evaluated very carefully - often they do more harm, than good !.....

 

Playing arround with minor settings - why not give it all there is !..... :-)

 

My settings is as follows:

 

Steepness: 0 dB

Filter max length: 200.000 Samples

Cutoff freq: 2.0 X Nyquist

Anti-aliasing: 0

Pre-ringing: 0

 

"Power of 2 oversampling only (x2, x4, ....) - in my case to either 176 or 192 kHz.

 

I can't tell what's best for everyone, but I believe above settings are pretty close to what I want !..... :-)

[Superdad]

Hi There !..... :-)

 

I've been playing arround with Audirvana for quite some time - always leaving iZotope settings at the default settings !....

 

Yesterday I thought it was time to play a bit with above mentioned settings !.....

 

In general, I believe that all kind of filters has to be evaluated very carefully - often they do more harm, than good !.....

 

Playing arround with minor settings - why not give it all there is !..... :-)

 

My settings is as follows:

 

Steepness: 0 dB

Filter max length: 200.000 Samples

Cutoff freq: 2.0 X Nyquist

Anti-aliasing: 0

Pre-ringing: 0

 

"Power of 2 oversampling only (x2, x4, ....) - in my case to either 176 or 192 kHz.

 

I can't tell what's best for everyone, but I believe above settings are pretty close to what I want !..... :-)

 

With a steepness setting of zero (and zero for anti-aliasing, meaning final attenuation of filter), you have effectively turned off the filter entirely! So you are back to listening to your DAC's internal filters.

[carsten s chr]

Superdad !......

 

I know - and think it's better !...... :-)

[Jud]

With a steepness setting of zero (and zero for anti-aliasing, meaning final attenuation of filter), you have effectively turned off the filter entirely! So you are back to listening to your DAC's internal filters.

 

But only one pass of the internal filter, no matter what resolution he feeds A+.

[Superdad]

Superdad !......

 

I know - and think it's better !...... :-)

 

So what DAC do you have?

[Marino Marco]

Is there any settings posted here that everyone agree to have a positive effect? or does everyone have a different opinion?

[carsten s chr]

Superdad:

 

My DAC is a Antelope Zodiac Gold with a Paul Hynes SR7 PSU !.....

 

But before the signal comes to the DAC, it's converted by a Sonicweld Diverter HR !.......

[JohanS]

Is there any settings posted here that everyone agree to have a positive effect? or does everyone have a different opinion?

Hi Marino,

 

I guess everyone prefers other settings; we've different equipment, room acoustics and ears (personal 'taste'). That said, I think several forum members prefer a lower steepness setting (lower than default 150), as it gives music (and certainly piano) more clarity. However, the lower settings used range from 2 or 3 (or even 0, but that's turning the filter off) to 50.

 

The pre-ringing is mostly set lower than 1,00 as well. Note that the ringing becomes less with a lower steepness, so the ringing value might have less impact on sound.

 

For Nyquist cut-off I suggest you try a range around 1,00. For example between 0,90 and 1,10. Do you hear any difference? If no, change to the default (1,00), if yes play around with it. You may notice a broader / deeper sound stage with a setting > 1,00 and a smaller, but maybe more focused (the instruments have a better defined position) with a setting lower than 1,00.

 

Anti-aliasing is left at it's standard value. The filter max length? I don't really know. Damien said a higher value is better. So, if your system has no problems (hic-ups) with for example 1.000.000 than use that as your setting.

 

To summarize, on my equipment the steepness setting and Nyquist cut-off have the most noticeble impact on sound. I'd try these first before changing all other values.

 

Good Luck!

Johan

 

PS Please tell us about your findings!

[Marino Marco]

Who's it going, JohanS? your post is really appreciated. thank you very much.

[mkrzych]

Hello,

I've just found this thread and it interested me very much. I am using Beresford Caiman Gatorized DAC connected over the optical to my Macbook with A+.

 

Before I post my experience with playing with these setting, I'd like to know if those parameters have any impact when A+ is feed by the 44.1kHz stream which is sent to my DAC without any up/over-sampling? I have "custom" setting used in A+, when redbook is not touched and any other are upsampled to 96kHz (my DAC max). If not at all, I may fooled myself little bit, but if yes, I have found a cure for my so called "boomy" bass, especially for contrabass playing with steepness/cut-off/pre-ringing.

 

I have WM8716 chip inside which is a 64X oversampling DAC.

 

What do you think, is the filter used only when I upsample or always when the bits going through the A+ engine?

[Jud]

Hello,

I've just found this thread and it interested me very much. I am using Beresford Caiman Gatorized DAC connected over the optical to my Macbook with A+.

 

Before I post my experience with playing with these setting, I'd like to know if those parameters have any impact when A+ is feed by the 44.1kHz stream which is sent to my DAC without any up/over-sampling? I have "custom" setting used in A+, when redbook is not touched and any other are upsampled to 96kHz (my DAC max). If not at all, I may fooled myself little bit, but if yes, I have found a cure for my so called "boomy" bass, especially for contrabass playing with steepness/cut-off/pre-ringing.

 

I have WM8716 chip inside which is a 64X oversampling DAC.

 

What do you think, is the filter used only when I upsample or always when the bits going through the A+ engine?

 

Only when upsampling. You've fooled yourself. But if it works, keep doing it! ;-)

[mkrzych]

Only when upsampling. You've fooled yourself. But if it works, keep doing it! ;-)

 

Hm, as I thought, but something has changed. Now I need to find what exactly. In my case when playing redbook not upsampled by A+, it will be anyway upsampled 64x internally by my DAC before going to the analogue output right? So any filters are involved in the middle of this process?

 

Is it better to upsample redbook by A+ or leave it as it is sending to the DAC?

[Jud]

Hm, as I thought, but something has changed. Now I need to find what exactly. In my case when playing redbook not upsampled by A+, it will be anyway upsampled 64x internally by my DAC before going to the analogue output right? So any filters are involved in the middle of this process?

 

Yes, the filtering built into your DAC chip.

 

Is it better to upsample redbook by A+ or leave it as it is sending to the DAC?

 

Listen and see what you like better, or if you hear any difference. Really, that's the best way to do it. There are so many variations between systems that it is difficult for someone else to tell you what you will hear and like.

[mkrzych]

Yes, the filtering built into your DAC chip.

 

 

 

Listen and see what you like better, or if you hear any difference. Really, that's the best way to do it. There are so many variations between systems that it is difficult for someone else to tell you what you will hear and like.

 

Sure, will do, but in theory is there any benefit to upsample redbook to my DAC maximum available input support? It is 96kHz, but internally it's 64x DAC, so the signal will be upsample again I guess to 64x44.1kHz which gives the frequency around DSD before going to the low pass filter and analogue section If I am right.

 

And the last question, previously mentioned options in A+ will work also for the signals with 192kHz which are not natively supported by my DAC, so I guess in this case those will be downsample to 96kHz?

[Jud]

Sure, will do, but in theory is there any benefit to upsample redbook to my DAC maximum available input support? It is 96kHz, but internally it's 64x DAC, so the signal will be upsample again I guess to 64x44.1kHz which gives the frequency around DSD before going to the low pass filter and analogue section If I am right.

 

Nearly all DACs do similar conversion to DSD frequencies in a sigma-delta modulation step. See for more detail: http://www.computeraudiophile.com/f11-software/audirvana-plus-optimizations-16766/index2.html#post240424

 

The answer depends on the interaction between your DAC's internal filter and A+, which would require knowing the detailed characteristics of your DAC's internal filtering and someone with the knowledge to understand how this internal filtering would work with particular iZotope settings. I have neither. :-) So this is the reason for saying the best (and certainly the easiest) way to find out is to listen.

 

And the last question, previously mentioned options in A+ will work also for the signals with 192kHz which are not natively supported by my DAC, so I guess in this case those will be downsample to 96kHz?

 

Yes.

[mkrzych]

Nearly all DACs do similar conversion to DSD frequencies in a sigma-delta modulation step. See for more detail: http://www.computeraudiophile.com/f11-software/audirvana-plus-optimizations-16766/index2.html#post240424

 

The answer depends on the interaction between your DAC's internal filter and A+, which would require knowing the detailed characteristics of your DAC's internal filtering and someone with the knowledge to understand how this internal filtering would work with particular iZotope settings. I have neither. :-) So this is the reason for saying the best (and certainly the easiest) way to find out is to listen.

 

 

 

Yes.

 

Thanks a lot Jud. It is indeed very interesting subject, at least to me ;-)

[Axiom05]

Thanks a lot Jud. It is indeed very interesting subject, at least to me ;-)

 

Actually, it is a mind-boggling subject. :-) There are very few DAC's that either do no upsampling of their own (NOS-type DAC's) or let you turn the upsampling off. Therefore, as Jud said, what you hear will always be the result of an interaction of the two upsampling/filtering processes. The only way to know what this effect is, is to listen. There are people who claim that this filter "stacking" is not a good thing. I have a Resonessence Concero which allows you to switch off all internal upsampling/ filtering so I am able to evaluate upsampling within my music player without additional filter interactions in the DAC. As you said, an interesting subject.

[Neil]

Playing with the settings after reading this far trying 20db 1,500,000. , 1.04, 200 and 0.70 with power of 2. I played one of the HD tracks downloaded , Be Still by Dave Douglas, and there was a super high pitched tone, almost painful!!! Any one had this? I run a new Mac mini, USB into an Emotiva XDA2, then XPA2. The DAC has a SRC bypass I thought this was asynchronous mode, or does this bypass the SRC of A+? Sorry if these are noob questions but still trying to cram my head around this crazy new hobby....

 

neil