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

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Ideally I would want at least 4 (or better 6) physical audio outputs. Which would allow me to route audio in a studio environment... or could be configured as left and right for standard desktop work, or as multichannel surround sound for entertainment.

Each physical output would require it's own channel... you wouldn't want 8 or 16bit support, 24bit is al that is required.

From a hardware point of view, you would probably really want a small "DSP like" processor and a bunch of SigmaDelta DACs... that would be the simplest way.

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biggun wrote:

warpdesign wrote:

I'm interested in your educated opinion, whether it makes really sense to increase the number of supported channels over 4.

Of course it does... And the more the better...

If you are experienced in creating module music,
please tell us how many channels are in your opinion are needed for a good mod. If you have no experience in creating music, then please do not respond.

In OctaMED Sound studio, I would rarely ever use more than 16 channels at most in mix mode... though I would often use two channels (panned hard left and hard right) for a single sound,  and then mess around with the play back to simulate various effects and delays.

If mixing is not noticable in today computers, it sure will be with a slow 060

What you say is non true at all!

Even with a stock 68000 you can mix 8 channels.
For a 68060 mixing 8 extra channels takes no time.

I totally agree with Piru, have a decent Paula emulation... and a nice modern audio chip with an AHI driver... Audio is a commodity now, technologically it's reached a plateau... you can't really do anything innovative (as is still possible to some degree in the GFX space). Great audio chips can be bought off the shelf for next to nothing.

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Piru wrote:

nowadays that Mac is pretty much obsolete since all that software functionality can be done better on a PC.

Macs use exact the same common audio components as PCs.

I think he's taking about the old Macs (of the 80s)

Also, today's PCs also seem to have a mixer of analog signals as well since the CDROM-audio can be mixed with the wave audio at various volume levels.

I don't know which decade you're living on, cd-rom audio hasn't been analog for years (the analog audio cable isn't even connected).

Very true! I think the last time I connected a CD ROM audio cable was 1999...

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Karlos wrote:
All of my musical equipment has at least 7-bit resolution (often a lot higher) for volume and pan since that's the basic minimum standard required for General MIDI conformitiy.

Just checked all my Audio software and hardware, and everything is 7 bit... -63 to 63... I guess that's the legacy of MIDI...

Going back to a purely musical view, if you could add tuneable low pass filters to each channel, I'd be very happy indeed :-)

12pole resonant (with Adjustable Q) low pass filters please :-) Some distortion effects might be easy to implement too...

While we're at it why not add a filter and amp envelope?

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Karlos wrote:

Well, provided you have registers for the basic controllers I don't think there's much harm in letting the CPU do envelope control :-)

Yeah, I was joking about Envelope control :-)

But that does raise the important issue of Buffers and latency... I don't really care about features sicne the CPU can worry about that... What I do need is a latency under 6ms!

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Karlos wrote:

bloodline wrote:
What I do need is a latency under 6ms!

Hence the reason my hardware synth hasn't been replaced by some jumped up bit of software ;-)

Hahahaha, I still stand by my MBP and an Edirol FA-101 :-)

in Ableton Live 5 and Logic Pro 8; I get about 3ms latency each way (3 in and then 3 out), hence my demand for 6ms latency :-)

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A6000 wrote:
So, from a musicians point of view, would it be better to have MIDI than a superpaula?

NO!!!! :-)

I only have one MIDI device left now (all the rest are USB)... and that is connected to my Mac via a USB-MIDI interface :-)

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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!

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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?

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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!

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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?

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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!!

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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?

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Karlos wrote:

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 ;-)

Well get on and do it... if not let me know, and I'll build a look up table based on my idea...

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ChrisH wrote:

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?

I would imagine the best way to *generate* the Paula response curve would first require you to know what kind of D/A converter Paula used.  I *vaguely* recall it used a series of (9?) resistors to get the voltage drops required for each of the 8 bits, which is not very accurate due to the resistor tolerances, and therefore tends to produce a non-linear response curve.

An R-2R Ladder DAC eh? I guess that would work for something as simple as 8 bits... Ok... are these integrated on the Paula IC or on the Board? I need to find a Paula pinout..

Assuming my recollection is correct, then you need some method to find the relative values of each of those (9?) resistors.  That should be MUCH more accurate & less error prone, especially compared to trying to sample the output of Paula across her entire dynamic range.

It would be easy to work out... but that's not going to give me real word performance.

I imagine that the 14-bit audio calibration tool used some clever method to accurately find out the relative resistor values, so that it could then compensation for them.

IIRC the calibration tool was rather subjective... which required you to decide when the response was linear... it was a long time ago...