SuperPAULA - if you have experinece in amiga music please give feedback

[bloodline]

AmiDelf wrote:
@BigGun

I`ve been an DelfinaDSP DelfMpeg betatester.  Its one of the best soundcards out there. I am not a hardware person, but for me it seems that it is the Motorola DSP cpu which is inside DelfinaDSP.

Either its 68030 or 68060.

it's the Motorola 56002...

Its really nice to have a DSP that handles the sound.

As opposed to?

Ive never in my whole life heard such great musicquality as from my sold Amiga 4000 with DelfinaDSPz2 that Ive overclocked from 40MHz to 66MHz.

What did you overclock?

The sound quality of that board is one of the best I`ve ever heard! Ive listened to several Soundblaster cards and other more expensive PCI cards. But they havent beaten my old DelfinaDSP card at all.

Sound blaster is hardly the gold standard of audio quaility... but the Delgina uses the same CS4215 codec... as best I could suggest that the SB had poor gound sheilding or badly written drivers... But the Delfina is hardly top quality audio.

Even with my bad Creative speakers. The sound quality of Delfina showed it strength. And my ears are made for music and sound.

You use Creative Speakers... yet you say your ears are "made for Sound and Music"... Get some Mordaunt-Short's or Celestion's if you want to hear good quality at a resonable price...

I am not saying this because I love Amiga,

Yes you are.

but because it is how I have captured the sound quality of DelfinaDSP which I have tested from A to Z practically.

Um... good?

With delfinampeg.device you can listen to the music thru AmigaAMP etc.

How nice!

[Karlos]

Going back to point 4 (I think it was) on my suggestion list, the ability to turn off interpolation all together should definately be an option.

As a musician, one of the things I use paula for is her dirty, heavily-aliased sound when playing back samples. Additionally, the non-linear sample value->amplitude is rather nice.

I suggest, in addition to the ability to turn off interpolation, perhaps that 8-bit sample playback should have an optional 8->16-bit lookup table that emulates the response curve of the original Paula.

[polardark]

Karlos wrote:
As a musician, one of the things I use paula for is her dirty, heavily-aliased sound when playing back samples. Additionally, the non-linear sample value->amplitude is rather nice.

The Amiga doesn't alias. It does have frequency mirroring though, which is a much more pleasant and harmonic and can still sound very raw. A lot of Amiga musicians took this into account and sampled their sounds at a note frequency relative to the sampling frequency to minimize non-harmonic distortion.

With 192 KHz sampling frequency (which, i guess, could be thought of as 4x oversampling) you won't have much audible aliasing even without interpolation.

I agree that being able to select between an interpolated and a non-interpolated sample playback mode for each channel would be very useful.

[bloodline]

Karlos wrote:

I suggest, in addition to the ability to turn off interpolation, perhaps that 8-bit sample playback should have an optional 8->16-bit lookup table that emulates the response curve of the original Paula.

Can anyone here publish the original Paula response curve? I'd love to write a paula effect processor! :-)

Actually... I probably have all the equipment needed to find it out myself... does anyone know a decent menthod?

[Karlos]

Remember the cybersound 14-bit driver calibration tool?

One could probably create something similar to estimate the response curve.

[bloodline]

My plan was to build an 8bit audio sample at 44100Hz which gradually steps down through the dynamic range (with some stupidly low period)... play that back on one of my Amiga's and sample it at 24bit 192Khz on my Mac... I should be able to examine the sampled sound and compare it with the original to find out the actual repsonse...

This is my first thought, If I think about it a little more... I should be able to optimise the procedure a little!

[Karlos]

The problem I have with that approach is that a very low frequency output of that nature is not likely to make it through the analogue amplification stage.

The calibration file produced by cybersounds 14-bit calibration tool probably already contains the information you need, albeit in the inverse.

[bloodline]

Karlos wrote:
The problem I have with that approach is that a very low frequency output of that nature is not likely to make it through the analogue amplification stage.

Hmmm... I was hoping that since the frequency would be fractional (I would hope to hold the sample at each voltage level for 10s of thousands of samples) it would pass through the electronics totally unaffected... I would run it all at line level, so no preamps filtering out sub 10Hz...

I am open to suggestions!! If someone has an Oscope then perhaps they could do this for us?

[Karlos]

I was thinking the onboard amplifiers on the motherboard would basically filter out your very low frequencies as they look too much like DC when coming up against a capacitor.

You'd have the same problem on your sampler input too, I'd expect.

[bloodline]

Karlos wrote:
I was thinking the onboard amplifiers on the motherboard would basically filter out your very low frequencies as they look too much like DC when coming up against a capacitor.

You'd have the same problem on your sampler input too, I'd expect.

Yes... I was relying on it looking like DC... ok... Then perhaps I can do the same with a 1Khz tone playing on the Amiga... Yes I think I can...

I'm loathed to use the 14bit calibration, since I know nothing about how that works!!

[Karlos]

bloodline wrote:

I'm loathed to use the 14bit calibration, since I know nothing about how that works!!

Me neither. I planned to simply compare a fresh uncalibrated file with the calibrated and see if I could intuit the relationship from the differences ;-)

[bloodline]

Karlos wrote:

bloodline wrote:

I'm loathed to use the 14bit calibration, since I know nothing about how that works!!

Me neither. I planned to simply compare a fresh uncalibrated file with the calibrated and see if I could intuit the relationship from the differences ;-)

That's rather clever and almost certain to take a million years...

I should be able to do it with a 1Khz tone, sampled at 192Khz... as that will hold the voltage, for 192 samples... hopefully long enough for me to identify it...

What type of Audio DAC does the Amiga use anyway? Perhaps a simpler solution would be to simulate it?

[TheGoose]

SuperPaula!? What! Crazy talk. I need to find some different street drugs. Anyway, mix modes sucked, period. Even though lots a people seemed to use them in the mod world. Sound was terrible.

Keep your octamed a quadraped if you know what i mean.

[Karlos]

bloodline wrote:

That's rather clever and almost certain to take a million years...

It's seemingly just a lookup table. I wasn't going to do it by manual inspection. I was going to graph the differences ;-)

[Krusher]

Karlos wrote:
...

It's seemingly just a lookup table. I wasn't going to do it by manual inspection. I was going to graph the differences ;-)

Wavelab has a nice tool for that  :-D

[SamOS39]

8 channel mode on octamed is VERY unstable  :-P

ummm .. doesnt octamed support 16bit samples..? im sure ive imported 16bit samples intooctamed v6 ..?