Why do you converse?

[bloodline]

Karlos wrote:
Is it basically a compander-like curve applied to the input then?

Err... I simply shifted the orignal 16bit sample right by 8places (actually it is more complicated than that since MacOSX Core Audio is 32bit floats)... then used a lookup table to see what the 8bit value would map to at 16bits... and built a new sample using that data... :-)

For example:

The original 16bit sample:  27900 (input)
The 8bit Bit crushed sample : 109
The Paula non-linear output output: 27933 (output)

(The Paula perfectly linear output: 27904 - for reference)

I can send you my lookup tables if you like :-)

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Karlos wrote:
I see :-)

The one thing I was never quite sure about was how the non-linearity of paula's audio worked. What I mean is, was it the absolute output voltage that was affected by the curve directly or was it the change in successive sample output voltages that were affected by the curve?

If it's the former, a simple lookup table converting input sample to output sample is fine, but if it's the latter you need to basically keep track of the previous input sample and apply the lookup to the difference between the current sample and the previous sample.

From the Calibration Data from the cybersound package, the former has to be assumed... There is no other way to make it meaningful!

-Edit- Also you would need a set of values for ascending and another for descending!

In both cases, the second thing I wondered about was the effect of the inbuilt amplifier, but that's a wee bit beyond the scope of the discussion, unless realtime FIR filtering etc is your bang :-)

It isn't, but I can solve that by using Logic Pro's audio EQ matching features, by running a test tone through the Amiga and letting it match the pitch/dB relationships or whatever it does :-)

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motorollin wrote:

@bloodline
I really wish I understood what you're on about :lol: Can you explain it in layman's terms?

--
moto

I guess what Karlos and I are demonstrating is the opposite of small talk :-)

The theory is quite  simple actually:

Paula can output 256 discrete voltage levels... which steps from -5 volts to +5 volts... so each level is about 0.039 volts... but due to her design, some levels are a bit larger and some are a bit smaller... this is because the DAC in paula is not perfectly linear (perhaps due to the design or flaws in the manufacturing).

In pure number terms, i.e. the digitized audio sample it looks like this:

-128 to +127 (-5 volts to +5 volts)

Which if the DAC was perfectly linear would look like this:
-2= -0.078 volts
-1= -0.039 volts
  0 =   0 volts
  1 =   0.039 volts
  2 =   0.078 volts

So when we want to pay back the 8bit audio sample on 16bit hardware we need to convert the sample which could be any number from -128 to +127 into any number from (-65536 to +65535)... this is done simply by multiplying the sample by 256... (or we use the logical shift left operation since that is much faster, a good book on Boolean algebra will explain why).

That mean that 1 becomes 256... 2 becomes 512... 3 becomes 768... you get the picture... but that resultant output from the 16bit hardware won't sound like paula, because it is perfectly linear... so instead of multiplying by 256 we can use a lookup table to give the value that paula would have given if her output was sampled at 16bit!

So the value of 1 should have been 256 in a 16bit sample if she was perfectly linear... but Paula would actually give 239 (this isn't the true value since I don't have my tables with me at the moment)!

My plan is to make an Audio Unit, that takes the Audio stream converts it to 8bit, applies this paula derived nonlinear distortion, and than upscales back to the native audio resolution... Eh Viola! Instant Paula Audio from Logic Pro... or Ableton Live... Garage Band... or whatever Audio app you use to make music on your Mac! This is why I love digital music making :-)

Karlos also pointed out that the Amplifiers on the Motherboard colour the audio... But Logic has special EQ software which I should be able to use to match the that... it would be nice to have that as part of the Audio unit... but sadly Fast Fourier Transform isn't my idea of fun.

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Karlos wrote:
Just to complicate it a little further, do we know exactly how the 6-bit channel volume is applied on Paula?

I've noticed that, with proper amplification of the output, you can still playback an 8-bit sample at a channel volume setting of 1 with much more precision than you'd expect of a simple 8-bit DAC.

This hasn't escaped my notice either... though the noise in the signal might be providing some kind of dithering... increasing the quality of what we perceive...

We would need to engineer specific test tones to really see what's going on...

So, does the volume premultiply the sample before sending it to a DAC that has 14-bit actual resolution, or is the analogue output of the DAC fed through a particularly stable volume control? If it's the latter, I don't expect the volume control is particularly linear, either.

THis has puzzled me for quite a while!!! My initial assumption was that the 8bit sample was divided by the 6bit volume value, digitally before the DAC... but Paula certainly doesn't have 4 Dividers in there!!!! Ok... so maybe the 8bit sample is bit shifted... but I can't make that work and I doubt she has 4 barrel shifters in there either... I can only conclude that the Volume is done after the DAC with an opamp. The 8bit Sample value is converted into an audio signal by an 8bit DAC... The 6bit Volume value is converted into a signal by a 6bit DAC and then the Audio signal is attenuated by the volume signal with the OpAmp... does that make sense?

-Edit- Thinking about it, I expect this is where the nonlinearity of Paula comes from!

I got the impression that 14-bit replay tricks play the most significant 8-bit portion of a 16-bit sample word on one channel at full volume then the next 6-bits of the 16-bit sample word on the other channel at minimum volume.

In my own code, I split the 16bit sample into two bytes. The Most significant byte played at full volume (63) in one channel. The least significant byte played at minimum volume (1) on the other channel. It seemed to work... I didn't realise about the volume thing at first... and the noise was horrible :-)

Perhaps I was supposed to shift the least significant byte by two places... but I didn't...

-Edit- Also, it took me quite a to realise that 16bit Wav files were little endian, and my code was fine... :-)

I'll probably have to do some testing of my own now :-D

Please do... this is fun :-)

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Karlos wrote:

-edit-

Unfortunately I'm far too busy trying to persuade my crust old FM synth to reproduce the metallic effects from the first half of Hybrid's remix of "Everything in it's right place"...

Poo to the FM synth... The day I sold my DX9 was a happy one :-) Three months studying a badly translated Manual to make a crappy bell sound... that my Amiga and its 8bit sampler could do far better in a matter of seconds...

I really really don't like FM synth :-D

[bloodline]

Karlos wrote:

bloodline wrote:

I really really don't like FM synth :-D

Trying to recreate natural sounds with FM synthesis is both missing the point and a lost cause. I enjoy it for what it is, a method of producing unnatural sounds that can be realtime controlled in a way no wavetable/sampler can achieve.

You're right of  course! The sonic spectrum of the FM synth is vast! It ranges from crappy bell sound, to unpredictable noise vaguely resembling a bell...
For the really skilled, one could create bell like vibes, resonant metallic bells or hollow bell like noises... I even heard of a guy once, who was able to make a hollow resonant metallic bell like vibe sound... but despite being little bit crappy and sounding like a bell, it was a the stuff of legend... probably because he survived the experience...

Poo to the FM synth...

Ok, I'm being a bit facetious, I did manage to create a really dark spiky bass sound with a DX7 in a studio once... which was so good I sampled it on my W-30 so I could use it (Yamaha's MIDI support sucked).

[bloodline]

Karlos wrote:

bloodline wrote:

In my own code, I split the 16bit sample into two bytes. The Most significant byte played at full volume (63) in one channel. The least significant byte played at minimum volume (1) on the other channel. It seemed to work... I didn't realise about the volume thing at first... and the noise was horrible :-)

Perhaps I was supposed to shift the least significant byte by two places... but I didn't...

I'd have thought that's a prerequisite, since you'd need the LSB to be played at 1/256th the level of the MSB for a perfectly linear channel volume. As the real volume level is only 6-bits, the top 2 significant bits of your LSB will basically overlap the 2 least significant bits of your MSB producing quite noticeable noise :-?

Which goes some way to explaining why I never became a billionaire software engineer... Yeah it was noisy, but I was just happy to get it to work after the not realising about the volume thing, and that all my 16bit samples were little endian... we are taking about weeks of work here! :-)

Of course, if Paula's volume is some freaky logarithmic scale it's possible that 1/256th of "full" volume is achievable with some low volume setting.

I don't think the volume is logarithmic at all. I think I messed up. :-D

[bloodline]

Karlos wrote:

bloodline wrote:
Poo to the FM synth...

Ok, I'm being a bit facetious, I did manage to create a really dark spiky bass sound with a DX7 in a studio once... which was so good I sampled it on my W-30 so I could use it (Yamaha's MIDI support sucked).

The trick is not necessarily trying to understand how it works (really, it's phase modulation rather than frequency modulation but it amounts to the same thing).

I remember being quite perplexed as to how the engineers had managed to get actually quite a decent set of presets in there (compared to what I could do)... I think there big problem was the interface with required a Phd in Condensed Matter Physics , before you had a clue as to what was going on...  

That said, the PPG Wave was a pain to program, but I fell in love with that synth when I got to play with one... The sounds you can make are wonderful... I wish I had one :-)

I've managed to create many interesting pads, effects and bass sounds ranging from basic sinusoidal speaker shakers to deeply resonant and dirty pulses.

Hmmm... I expect you are a bit brighter than me and managed to see into the mind of the engineer who created the concept... I simply couldn't get my head around it... And since I was really into sampling at the time (Yeah, I was listening to industrial music), I didn't bother to waste any more time with the FM idea.

Unfortunately my kit is limited, getting old and noisy. I'm contemplating getting the DX150 (DX7 on a card) plugin for my MU100 to replace the  vocoder I never use :-)

Actually, Moto recently made an interesting suggestion regarding a Vocoder... which has made me play with my vocoders... combining them with ringshifters to add some really harsh tones to my work!!! :-D

[bloodline]

Karlos wrote:

 They just got a sound and thought about what it sounded most like afterwards :lol:

:roflmao:

Many a true word said in jest ;-)

[bloodline]

 :oops: I've just noticed a rather blatant Freudian slip that I made early in my hijacking of this thread :-)

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motorollin wrote:

bloodline wrote:
 :oops: I've just noticed a rather blatant Freudian slip that I made early in my hijacking of this thread :-)

I can't believe I missed it :inquisitive: You going to tell us or make us trawl through you and Karlos rambling on about numbers? :-P

--
moto

I noticed as I was thinking how to clean up my explaination of paula's non linear nature... To be fair it's only a single letter transposition and only you would get it ;-)

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motorollin wrote:
Nope, still can't find it :lol: You'll have to quote it since going through those posts word by word is making my eyes go funny.

--
moto

Never mind... I accidentally corrupted an well known French phrase, while discussing Audio Units... :-)

Anyway, don't you just hate it when a Band ruins an otherwise good song... it's just a travesty:


clickly clickly

Really shouldn't be allowed.. Eh Karlos ;-)

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HenryCase wrote:

motorollin wrote:

HenryCase wrote:
So it's not a lack of confidence in small talk, but I find as I can completely rely on idle chit chat to get me through social situations I hardly ever get to discuss matters which mean more to me, and lack confidence in opening up.

Forgive me for being a little presumptuous, but it sounds to me like you may be using small-talk as a defence mechanism. Small-talk is inconsequential so is easy if you practice it enough, but really opening up is much harder if you lack the confidence to do it.

Abso-f*cking-lutely, that is exactly what I am trying to say, in that small talk is used for different purposes with different people.

I use it to get laid... doesn't work...

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Karlos wrote:

bloodline wrote:

Anyway, don't you just hate it when a Band ruins an otherwise good song... it's just a travesty:


clickly clickly

Really shouldn't be allowed.. Eh Karlos ;-)

That's canny good. I've been responsible for far worse ;-)

We put that song together rather quickly as a special treat for my B'day :-) it is really easy to play and actually rather fun... the crowd, most of whom were around 20... had no idea what was going on :-D

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Karlos wrote:
They probably don't know what the Tan Hauser Gate is either then?

To be fair... no one does... except a select few si-fi geeks ;-) :-D

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Karlos wrote:

bloodline wrote:

To be fair... no one does... except a select few si-fi geeks ;-) :-D

So, do you have any glittering C beams for your live light show?

Actually we do :-D I wish I could show you a DVD of one of our shows... I think the Youtube link in my sig is quite good...