Audirvana + How to select Izotope settings?

[One and a half]

With all the bewildering array of different numbers in the Audio Filter settings in Audirvana +, how does one set them?

 

Does it matter if the DAC is NOS, 8 x oversampled, converted to DSD & existing filters, how to tune and match:

 

- What's the main criteria

- What if a a setting is too high/low

- Will settings cause instability

- Rule of thumbs

- Safe values

- Don't go there settings

- For specific music, such as chamber, high EQ'd pop, trio jazz, solo piano

- Do certain filter settings need to be set for a different range of upsampling settings, eg 2 x , 4x

[Musicophile]

I'm far from being an expert, but short answer is: there is no short answer.

 

It all depends very much on your system and DAC.

 

And on the Izotope parameters, good luck with the 25 plus pages of this thread:

 

http://www.computeraudiophile.com/f11-software/izotope-sample-rate-convertor-15352/

 

Where you can try other peoples settings. Don't expect a turnkey solution though.

[copy_of_a]

some - personal! - notes regarding "rules of thumb"...

 

This is from the manual of HQPlayer and I agree:

linear phase = better space

minimum phase (low pre-ringing respectively) = better transients ("attack")

 

"steepness" in iZotope SRC:

- higher steepness = better filtering above the nyqiust limit

- higher steepness = more ringing

- lower steepness = better instrumental seperation (especially in complex, 'full' musical arrangements)

- lower steepness = smooth roll off of high frequencies (unless you increase "cutoff scaling")

 

From these few notes alone you can see that we are always dealing with trade-offs/ compromises.

 

If you prefer low steepness (at the expense of filtering quality) you can also filter out aliasing/artefacts above the nyquist limit with an appropriate EQ in the AU Plugin section of A+.

If so, you would consequently need different EQ/Filter-settings for different source sample rates... so you would have to manually change the plugin settings whenever the sample rate of the source changes (which is pretty cumbersome).

 

If you want to effectively remove aliasing/artefacts above the nyquist limit while upsampling on the fly and without using a plugin I would say "steepness" should at least be set to '31' or higher (and "Anti-Aliasing" to '200').

 

-----

 

Meanwhile, me personally I prefer offline upsampling (of 44.1kHz an 48kHz source files) in iZotope RX4. Actually I was really happy with steepness: 16, cutoff: 1, pre-ringing: 0.72... and afterwards removed all the aliasing/artefacts above the nyquist limit with Fabfilter Pro Q2. But before I started to convert my entire library I've further fine-tuned my SRC settings and am now at steepness: 3, cutoff: 1.3 (to preserve all high frequencies), pre-ringing: 0.36. I then remove all the aliasing/artefacts above the nyquist limit with said Fabfilter plugin in linear phase mode at the highest quality setting and save the files as 24bit TPDF dithered files.

To further upsample these files to DSD in A+ I set the iZotope settings to almost the same values: steepness: 3, cutoff: 1, pre-ringing: 0.36, Anti-Aliasing: 50 (so the lowest value). Since the files are now at 176.4kHz (192kHz respectively) the nyquist limit is so high that there is no need to shift the cutoff scaling or to apply any Anti-Aliasing filtering...

[copy_of_a]

- Safe values

 

a follow up of my previous post re "safe" settings...

 

 

 

 

"Safe" settings if you want to remove aliasing/artefacts above the nyquist limit:

 

 

linear:

• this is one of iZotopes presets

- steepness: 31

- cutoff: 1

- Anti-Aliasing: 200

- pre-ringing: 1

 

 

moderate pre-ringing:

• good compromise between decent filter quality & moderate ringing & moderate pre-ringing

- steepness: 31

- cutoff: 1

- Anti-Aliasing: 200

- pre-ringing: 0.72 (0.72 - 0.78)

 

 

 

 

"proven" settings if you don't care about aliasing/artefacts above the nyquist limit (for whatever reason):

 

 

linear:

• great instrumental seperation

• soft rolloff of high frequencies (starting at 16kHz or so... IIRC)

• very weak filter quality

- steepness: 5

- cutoff: 1

- Anti-Aliasing: 200*

- pre-ringing: 1

 

 

moderate pre-ringing:

• great instrumental seperation

• pretty good transients

• very weak filter quality

- steepness: 7

- cutoff: 1.02

- Anti-Aliasing: 200*

- pre-ringing: 0.86

 

 

low pre-ringing:

• very good instrumental seperation

• very good transients

• mediocre filter quality

• for me overall a really great compromise between filter quality & low ringing & low pre-ringing

- steepness: 16

- cutoff: 1

- Anti-Aliasing: 100*

- pre-ringing: 0.72

 

 

my settings from above just to be listed again here...

 

 

low pre-ringing "extreme" setting:

• phantastic instrumental seperation

• great transients

• super weak (actualy 'no') filter quality

steepness: 3

cutoff: 1.3

Anti-Aliasing: 50

pre-ringing: 0.36

- cutoff 1.3 requires to reduce the gain by 1db to avoid too high intersample peaks. So if you set the gain by default for instance to -3db you should set it to -4db with this setting.

- Too, as outlined above, IMHO settings with such a low filter steepness are better used in conjunction with an appropriate high quality lowpass (highcut) filter in the AU plugin section of A+ if you want to upsample on the fly (and if you do so better set Anti-Aliasing to 50). But this goes for all the settings listed in this second section...

 

 

--------

 

 

personally I don't think that the filterlength really affects the sound (if so it's mostly likely related to computer-power...). I always set the max filter length 200.000. Furthermore me personally I prefer upsampling by power of 2 (with all the settings noted above).

 

 

--------

 

 

* Anti-Aliasing

The Anti-Aliasing filter in the iZotope SRC section of A+ does NOT help to remove aliasing/artefacts ABOVE the nyquist limit. It only suppresses freuquencies right AT the nyquist limit (see here for steepness: 5 / cutoff: 1 / AA: 200 / pre-ringing: 1 : http://www.computeraudiophile.com/attachments/f11-software/29056d1474067957-izotope-sample-rate-convertor-jud_5_1_200_1.jpg ). This is why I think it's pretty useless at low steepness settings and I would rather set the AA-filter to 100 or 50.

[Jud]

Depends what you're upsampling to. I'm using DSD256 most of the time, so it doesn't take a very steep filter to get rid of most aliasing at that rate.

[copy_of_a]

Depends what you're upsampling to. I'm using DSD256 most of the time, so it doesn't take a very steep filter to get rid of most aliasing at that rate.

the filter acts at the nyquist limit of the _source_ file... so low steepness filters produce artefacts ("fake details" if you want so...) right above the nyquist limit of the original file. My screenshot above shows this pretty clear (source is 44.1kHz / target is 176.4kHz).

[Rezseeker]

In my system, the lower the max filer length (ranging between a quarter to two thirds of 200.000 max has been the best depending on the gear I was using over the last two years) , the better the grip on the lower frequencies. The higher end of that range brought a little more refinement in the very high frequencies.

 

 

Sent from my iPhone using Computer Audiophile

[HardrockInMiniMac]

a follow up of my previous post re "safe" settings...

 

 

 

 

"Safe" settings if you want to remove aliasing/artefacts above the nyquist limit:

 

 

linear:

• this is one of iZotopes presets

- steepness: 31

- cutoff: 1

- Anti-Aliasing: 200

- pre-ringing: 1

 

 

moderate pre-ringing:

• good compromise between decent filter quality & moderate ringing & moderate pre-ringing

- steepness: 31

- cutoff: 1

- Anti-Aliasing: 200

- pre-ringing: 0.72 (0.72 - 0.78)

 

 

 

 

"proven" settings if you don't care about aliasing/artefacts above the nyquist limit (for whatever reason):

 

 

linear:

• great instrumental seperation

• soft rolloff of high frequencies (starting at 16kHz or so... IIRC)

• very weak filter quality

- steepness: 5

- cutoff: 1

- Anti-Aliasing: 200*

- pre-ringing: 1

 

 

moderate pre-ringing:

• great instrumental seperation

• pretty good transients

• very weak filter quality

- steepness: 7

- cutoff: 1.02

- Anti-Aliasing: 200*

- pre-ringing: 0.86

 

 

low pre-ringing:

• very good instrumental seperation

• very good transients

• mediocre filter quality

• for me overall a really great compromise between filter quality & low ringing & low pre-ringing

- steepness: 16

- cutoff: 1

- Anti-Aliasing: 100*

- pre-ringing: 0.72

 

 

my settings from above just to be listed again here...

 

 

low pre-ringing "extreme" setting:

• phantastic instrumental seperation

• great transients

• super weak (actualy 'no') filter quality

steepness: 3

cutoff: 1.3

Anti-Aliasing: 50

pre-ringing: 0.36

- cutoff 1.3 requires to reduce the gain by 1db to avoid too high intersample peaks. So if you set the gain by default for instance to -3db you should set it to -4db with this setting.

- Too, as outlined above, IMHO settings with such a low filter steepness are better used in conjunction with an appropriate high quality lowpass (highcut) filter in the AU plugin section of A+ if you want to upsample on the fly (and if you do so better set Anti-Aliasing to 50). But this goes for all the settings listed in this second section...

 

 

--------

 

 

personally I don't think that the filterlength really affects the sound (if so it's mostly likely related to computer-power...). I always set the max filter length 200.000. Furthermore me personally I prefer upsampling by power of 2 (with all the settings noted above).

 

 

--------

 

 

* Anti-Aliasing

The Anti-Aliasing filter in the iZotope SRC section of A+ does NOT help to remove aliasing/artefacts ABOVE the nyquist limit. It only suppresses freuquencies right AT the nyquist limit (see here for steepness: 5 / cutoff: 1 / AA: 200 / pre-ringing: 1 : http://www.computeraudiophile.com/attachments/f11-software/29056d1474067957-izotope-sample-rate-convertor-jud_5_1_200_1.jpg ). This is why I think it's pretty useless at low steepness settings and I would rather set the AA-filter to 100 or 50.

 

Pretty neat. What about filter max length? Is it all the same for all the settings, what do you prefer?

 

Thanks,

Chris

[Jud]

the filter acts at the nyquist limit of the _source_ file... so low steepness filters produce artefacts ("fake details" if you want so...) right above the nyquist limit of the original file. My screenshot above shows this pretty clear (source is 44.1kHz / target is 176.4kHz).

You know more about this than I do. But remember these responses from mansr with regard to noise at various DSD rates:

 

http://www.computeraudiophile.com/f11-software/audirvana-plus-2-multi-channel-sound-5-surround-channels-and-1-lfe-low-frequency-effects-channel-pulse-code-modulation-direct-stream-digital-conversion-28428/index30.html#post594906

 

http://www.computeraudiophile.com/f11-software/audirvana-plus-2-multi-channel-sound-5-surround-channels-and-1-lfe-low-frequency-effects-channel-pulse-code-modulation-direct-stream-digital-conversion-28428/index30.html#post594932

[Jud]

Pretty neat. What about filter max length? Is it all the same for all the settings, what do you prefer?

Thanks,

Chris

 

Filter length is analogous to number of taps, which is how many times the filter acts on the signal. All else being equal, Miska has said the more the filter acts on the signal the steeper the attenuation, which would result in more ringing. I personally like less steep filters, so I keep the filter length to a minimum. But other people may be more concerned about aliasing and want a stronger filter, so that would mean a greater filter length.

[mansr]

You know more about this than I do. But remember these responses from mansr with regard to noise at various DSD rates:

 

http://www.computeraudiophile.com/f11-software/audirvana-plus-2-multi-channel-sound-5-surround-channels-and-1-lfe-low-frequency-effects-channel-pulse-code-modulation-direct-stream-digital-conversion-28428/index30.html#post594906

 

http://www.computeraudiophile.com/f11-software/audirvana-plus-2-multi-channel-sound-5-surround-channels-and-1-lfe-low-frequency-effects-channel-pulse-code-modulation-direct-stream-digital-conversion-28428/index30.html#post594932

Those graphs and comments don't concern the upsampling stage, which is where images have to be filtered out.

[Jud]

Those graphs and comments don't concern the upsampling stage, which is where images have to be filtered out.

 

Thanks to you and copy_of_a.

I did have a look at my filter settings in RX3 way back when, and from what little I knew (and know) they seemed to produce a reasonable graph. But obviously I could easily be wrong about that.

[copy_of_a]

Thanks to you and copy_of_a.

I did have a look at my filter settings in RX3 way back when, and from what little I knew (and know) they seemed to produce a reasonable graph. But obviously I could easily be wrong about that.

Here are 2 screenshots that make things a bit clearer ... I hope...

I've upsampled a track with iZotope RX4 to 96kHz with your settings (5/1/1).

The screenshots show the spectrometer while playing the file and the spectrum (in black & white; sensitivity = content from -120db and above) of the upsampled file.

The first screenshot shows a lot of "fake-details" above 22.050Hz (up to at least -50db in this section of this particular track - the horizontal white line in the spectrum is 22050kHz).

The second screenshot shows the same file filtered through "Fabfilter Pro Q2" to remove anything above the nyquist limit (the nyquist limit of the original file's resolution that is).

 

These screenshots also show the underlying intend of my previous posts: the best of 2 worlds - very high instrumental seperation facilitated through super low filter steepness in the resampler module to gain very, very low ringing (and literally no pre-ringing) but at the same time high quality filtering of aliasing/artefacts/fake-details above the nyquist limit (in my case with the Fabilter EQ plugin in an offline upsampling workflow...).

 

[copy_of_a]

I've upsampled a track ...

a 44.1kHz track ... (sorry!)

[Bob Stern]

Filter length is analogous to number of taps, which is how many times the filter acts on the signal. All else being equal, Miska has said the more the filter acts on the signal the steeper the attenuation, which would result in more ringing. I personally like less steep filters, so I keep the filter length to a minimum. But other people may be more concerned about aliasing and want a stronger filter, so that would mean a greater filter length.

Jud, you misunderstood Miska. The amount of ringing increases with filter steepness, but *neither* is affected by the filter length.

 

The filter length determines how long the "tail" of the impulse response is allowed to continue as the ringing magnitude progressively declines. There is no benefit in extending the filter length beyond the time at which the magnitude of the impulse response (the ringing) falls below the resolution of the DAC chip.

 

In other words, a filter length shorter than this amount truncates the impulse response so that the ringing stops abruptly instead of declining until it is buried in the inherent noise of the DAC. In the frequency domain, such truncation causes a tiny irregularity in the frequency response curve.

 

The steeper the filter slope, the longer the impulse response continues before it falls below the resolution of the DAC. So a longer filter does NOT cause ringing, but an impulse response with more ringing will be reproduced more accurately if its filter length is correspondingly longer.

[Bob Stern]

You can measure the above phenomena by playing in Audirvana a music file consisting of a pure digital impulse, i.e., a single non-zero sample followed by silence (all zeroes). Capture the output of Audirvana. Then:

 

1. Observe the output in the time domain to see when it drops to zero.

 

2. Do a Fourier transform to test how different iZotope settings affect the frequency response.

[copy_of_a]

Jud, you misunderstood Miska. The amount of ringing increases with filter steepness, but *neither* is affected by the filter length.

The filter length determines how long the "tail" of the impulse response is allowed to continue as the ringing magnitude progressively declines. There is no benefit in extending the filter length beyond the time at which the magnitude of the impulse response (the ringing) falls below the resolution of the DAC chip.

In other words, a filter length shorter than this amount truncates the impulse response so that the ringing stops abruptly instead of declining until it is buried in the inherent noise of the DAC. In the frequency domain, such truncation causes a tiny irregularity in the frequency response curve.

The steeper the filter slope, the longer the impulse response continues before it falls below the resolution of the DAC. So a longer filter does NOT cause ringing, but an impulse response with more ringing will be reproduced more accurately if its filter length is correspondingly longer.

Since the "filter length" refers to "samples" me I also thought the filter length defines how many samples are loaded into the SRC module (RAM resp.) to be upsampled at once as a "package" (simplified speaking in my bad English...). What you are saying kind of confirms this.

Thanks for your post!

[Jud]

Great thread. It's helped to clear up several incorrect ideas I've had.

[copy_of_a]

The filter length determines how long the "tail" of the impulse response is allowed to continue as the ringing magnitude progressively declines. There is no benefit in extending the filter length beyond the time at which the magnitude of the impulse response (the ringing) falls below the resolution of the DAC chip.

(...) The steeper the filter slope, the longer the impulse response continues before it falls below the resolution of the DAC. So a longer filter does NOT cause ringing, but an impulse response with more ringing will be reproduced more accurately if its filter length is correspondingly longer.

Bob,

I've checked the number of samples with my SRC settings (steepness: 3 / pre-ringing: 0.36).

With regard to the screenshot below my setting requires a filter length of ~100 samples when upsamling 44.1kHz to 176.4kHz (16bit, Mono in the pulse file shown...).

Just to be sure...: with regard to your post I assume I can easily set the filter length to the lowest value (10.000 samples in A+) when I am only upsampling (no downsampling from DSD to PCM or so).

Is that correct?

Many thanks in advance!

 

[HardrockInMiniMac]

Bob,

I've checked the number of samples with my SRC settings (steepness: 3 / pre-ringing: 0.36).

With regard to the screenshot below my setting requires a filter length of ~100 samples when upsamling 44.1kHz to 176.4kHz (16bit, Mono in the pulse file shown...).

Just to be sure...: with regard to your post I assume I can easily set the filter length to the lowest value (10.000 samples in A+) when I am only upsampling (no downsampling from DSD to PCM or so).

Is that correct?

Many thanks in advance!

 

[ATTACH=CONFIG]30414[/ATTACH]

 

Well, if that is true, a lot of people could set it to 10.000 and many pages in the 35page Isotope settings thread would look pointless regarding filter length - or any of "copy of a" settings above would work well so to speak.

[copy_of_a]

Well, if that is true, a lot of people could set it to 10.000 and many pages in the 35page Isotope settings thread would look pointless regarding filter length - or any of "copy of a" settings above would work well so to speak.

My question has been only a technichal question to make sure that I did understand Bob correctly.

It was absolutely not a question whether or not a low filter length is "better" or sounds "better".

[Bob Stern]

Bob,

I've checked the number of samples with my SRC settings (steepness: 3 / pre-ringing: 0.36).

With regard to the screenshot below my setting requires a filter length of ~100 samples when upsamling 44.1kHz to 176.4kHz (16bit, Mono in the pulse file shown...).

Just to be sure...: with regard to your post I assume I can easily set the filter length to the lowest value (10.000 samples in A+) when I am only upsampling (no downsampling from DSD to PCM or so).

Is that correct?

Yes, that is a correct interpretation of my post.

 

However, bear in mind that 16-bit resolution equals a dynamic range of 96 dB, and your graph is not sufficiently detailed to show when the ringing amplitude declines to –96 dB.

[Jud]

Yes, that is a correct interpretation of my post.

 

However, bear in mind that 16-bit resolution equals a dynamic range of 96 dB, and your graph is not sufficiently detailed to show when the ringing amplitude declines to –96 dB.

 

Hmm - OK, this (as always ) leads to another question.

 

This is the bit depth of the *original* file?

[copy_of_a]

Yes, that is a correct interpretation of my post.

cool - many thanks!

However, bear in mind that 16-bit resolution equals a dynamic range of 96 dB, and your graph is not sufficiently detailed to show when the ringing amplitude declines to –96 dB.

correct! When I zoom in vertically to see the full -96db the flat lines left and right of the visible waveform show more ringing (all in all around 100 samples... not only the ~50 samples the waveform shows at this magnification). However, I've only prepared the graph for a better understanding. 100 samples would be a bit low :-) ... and can't be set in A+ anyway.

Thanks again!

 

edit: for reference here's the same graph zoomed in vertically to show the scale down to -120db.

[RunHomeSlow]

On 10/28/2016 at 1:17 PM, copy_of_a said:

low pre-ringing:

• very good instrumental seperation

• very good transients

• mediocre filter quality

• for me overall a really great compromise between filter quality & low ringing & low pre-ringing

- steepness: 16

- cutoff: 1

- Anti-Aliasing: 100*

- pre-ringing: 0.72

 

 

my settings from above just to be listed again here...

low pre-ringing "extreme" setting:

• phantastic instrumental seperation

• great transients

• super weak (actualy 'no') filter quality

steepness: 3

cutoff: 1.3

Anti-Aliasing: 50

pre-ringing: 0.36

- cutoff 1.3 requires to reduce the gain by 1db to avoid too high intersample peaks. So if you set the gain by default for instance to -3db you should set it to -4db with this setting.

- Too, as outlined above, IMHO settings with such a low filter steepness are better used in conjunction with an appropriate high quality lowpass (highcut) filter in the AU plugin section of A+ if you want to upsample on the fly (and if you do so better set Anti-Aliasing to 50). But this goes for all the settings listed in this second section...

 

 

personally I don't think that the filterlength really affects the sound (if so it's mostly likely related to computer-power...). I always set the max filter length 200.000. Furthermore me personally I prefer upsampling by power of 2 (with all the settings noted above).

 

Posting today to keep this thread alive...

 

Thank you greatly copy_of_a for your knowledge... i'm upsampling everything to DSD128 with your low pre-rigning settings up there... sound really good now with my system. 16 - 1 - 200,000 - 100 - 0.72

 

But i would prefer the 'Fantastic Separation' over the 'Very good'

and also the 'Great transients over the 'Very Good' 

 

You lost me with the -3db to -4db change in settings...is it in A+ DSD filter safe volume that i change that, like in my picture? and also can you put a picture how to set your second point about the AU plugin in A+ if i would like to use your low pre-ringning 'extreme' settings.

 

Thank you again.