Digitally Remastered 8 bit released

[meynaf]

Originally Posted by Pat the Cat

I would love to have uploaded some WAV equivalents for Amiga users to play back... but github limits to a 25mb file size. I guess I could put a zipped wav up, or lha maybe... nope, didn't work. Zipped version is even bigger than the original wav. Sorry.

Uploading flac files should be enough (at least for me it is).

[Pat the Cat]

Originally Posted by hazydave

W

I have absolutely no idea why there's a compressor even being mentioned. The very last thing I want is to take my 8-bit or 9-bit sample and reduce its resolution any more. Maybe I'd play around with an expander, but no, never a compressor. And absolutely not one that's clipping ... that's just bad work.

I have zillions of various audio plug-ins designed for various purposes. It's possible that you might find something that would make the sound more pleasing. That's of course changing the original sound -- all you can ever do is reduce the amount of information from the original sample. But again, it's the art... perhaps a DSP algorithm meant for
"clarity" or some other generic function would be useful.

Mr Dave, a compressor doesn't just compress - it also expands, to make quieter parts of a music piece sound louder.

I always used the term "Compander", ie compression/expander, but musos and sound engineers don't like the word, they're too used to the idea that the equipment just compresses. It doesn't. It uses the available bandwidth as best it can.

"HeyMrMusic" is the best example of that, on the demonstrator. Double bass is not easy for a microphone to pick up in the first place, if youre recording on Jamaica in the early 60s. Recorded and mixed on a four track, that's not the real ringer...

[Thorham]

Originally Posted by Pat the Cat

I would love to have uploaded some WAV equivalents for Amiga users to play back... but github limits to a 25mb file size. I guess I could put a zipped wav up, or lha maybe... nope, didn't work. Zipped version is even bigger than the original wav. Sorry.

25mb of stereo 28khz 16 bit audio is still 223 seconds. should be plenty.

Originally Posted by Akira

Let's please stop with this and stick to topic, what's left to do here, talk about the Earth being flat and Pizzagate?

Can't handle a little off-topic-ness ? Lol

[Amiga1992]

Originally Posted by Thorham

Can't handle a little off-topic-ness ? Lol

Well, there is an off-topic forum where off-topic should go, and to be honest, I can't handle the original topic either :P

[Pat the Cat]

OK Akira, I'm off to get that scanning sorted. The HDCaddy?

Laser connections keep cutting out, I must have a bad connection... somewhere inside the thing.

Won't be posting here until those scans are online. Including calibration scan so people can check for distortion and adjust accordingly. Better safe than sorry.

[demolition]

Originally Posted by Pat the Cat

Mr Dave, a compressor doesn't just compress - it also expands, to make quieter parts of a music piece sound louder.

No, a compressor only compresses. Making quieter parts sound louder is the definition of dynamic compression. Expanders are also common and so are companders which can both compress and expand. All audio engineers know well how they all work.

[paul1981]

So what did Prodigy do?

[demolition]

They made music?

[Anakirob]

Originally Posted by paul1981

So what did Prodigy do?

I imagine in this context they are talking about making heavy use of overdrive/distortion (which amplify the waveform, and when the peaks exceed the bandwidth it produces 'clipping') in the production of their tracks.

Now I have a question about recording Amiga HW, so if the Paula has four channels all able to play at a sample rate independent of the other three channels, and usually at an unusual rate of Hz. Would there be an optimal frequency to record at which would do as little approximation as possible on the largest amount of notes. Like is there some magic frequency which is a multiple of at least half a dozen of the Amiga's notes.

I have a table of Amiga sample rates

	PAL 	NTSC
C-2 	8287 	8363
C-1 	4143 	4181
C#1 	4389 	4430
D-1 	4654 	4697
D#1 	4926 	4971
E-1 	5231 	5279
F-1 	5542 	5593
F#1 	5872 	5926
G-1 	6222 	6279
G#1 	6592 	6653
A-1 	6982 	7046
A#1 	7389 	7457
B-1 	7829 	7901
C-2 	8287 	8363
C#2 	8779 	8860
D-2 	9309 	9395
D#2 	9852 	9943
E-2 	10462 	10559
F-2 	11084 	11186
F#2 	11744 	11852
G-2 	12445 	12559
G#2 	13185 	13306
A-2 	13964 	14092
A#2 	14778 	14914
B-2 	15694 	15838
C-3 	16574 	16726
C#3 	17558 	17720
D-3 	18667 	18839
D#3 	19704 	19886
E-3 	20864 	21056
F-3 	22168 	22372
F#3 	23489 	23705
G-3 	24803 	25031
G#3 	26273 	26515
A-3 	27928 	28185
A#3 	29557 	29829
B-3 	31388 	31677

But I feel it's missing a column, PERIOD, the value which is fed to the Paula to play at a specific pitch. That sure would come in handy when programming audio routines. If someone could please complete this table for me that would be very much appreciated.

[meynaf]

Originally Posted by Anakirob

Would there be an optimal frequency to record at which would do as little approximation as possible on the largest amount of notes. Like is there some magic frequency which is a multiple of at least half a dozen of the Amiga's notes.

Yes there is a magic frequency. It is : 3546895 Hz
(Think about things such as finetune and vibrato/portamento effects if you still believe it can be lower than this.)

[demolition]

Originally Posted by meynaf

Yes there is a magic frequency. It is : 3546895 Hz
(Think about things such as finetune and vibrato/portamento effects if you still believe it can be lower than this.)

It can easily be sampled with 48 kHz. Any errors sampling at that rate will only be present above 20kHz, thus inaudible. Doesn't matter how the sound was generated.

[meynaf]

Originally Posted by demolition

It can easily be sampled with 48 kHz. Any errors sampling at that rate will only be present above 20kHz, thus inaudible. Doesn't matter how the sound was generated.

I'm not sure of this. You can't hear the frequency itself but you can eventually hear its harmonics.

[demolition]

Any harmonics from frequencies above 20 kHz will be located from 40 kHz and up, so even less audible. In practice it is almost impossible to tell the difference between 48 kHz 16 bit and 192 kHz 24 bit as long as the 48 kHz signal is sampled using the best available methods/filters and not a simple non-oversampling comb-filter like what was used in the early days of the CD. That is one of the reasons why CDs got a bad rep in some circles, but this has been massively improved since then.

[paul1981]

Where did you get 48 KHz from? From your logic regarding inaudiability, wouldn't 44.1 KHz suffice?

[demolition]

Sure, but 48 kHz is the most common sampling rate used today.

[paul1981]

In practice, 24-bit 96 KHz sounds a heck of a lot better than 16-bit 44 KHz... I'd say around 4 times better. You get denser sound, more constant, more alive, more real - more accurate.

[Anakirob]

Originally Posted by meynaf

I'm not sure of this. You can't hear the frequency itself but you can eventually hear its harmonics.

I'm with meynaf on this one, there is a notable 'feeling of warmth' which digital technology never quite seems to match. Perhaps this is just the effect of mild analogue distortion, but nonetheless there are many audio purists who swear by analog.

[paul1981]

Originally Posted by Anakirob

I'm with meynaf on this one, there is a notable 'feeling of warmth' which digital technology never quite seems to match. Perhaps this is just the effect of mild analogue distortion, but nonetheless there are many audio purists who swear by analog.

The sample rates need upping to a couple of MHz or more then it'll be as good as analog. It's all coming now, you can buy recorders that can do this. They're not doing it for a joke, they're doing it because people have been telling them for over 30 years that CD's sound crap, and finally they have listened.

[meynaf]

Originally Posted by demolition

Any harmonics from frequencies above 20 kHz will be located from 40 kHz and up, so even less audible.

Where did you fetch that ? Harmonics are divisors of original frequencies, not multiples !
So 20Khz can give 10Khz, 5Khz, and many others.
They ARE audible. This is why 44.1Khz, in spite being already kinda oversampled, can be differentated from 48Khz by someone with good musician's ear.

[demolition]

It can be proven mathematically that 44.1kHz can provide a perfect reproduction of all frequencies from DC to 20 kHz. Since the human hearing does not go beyond this range, 44.1 kHz should be enough for perfect reproduction. The only issue here is that it is impossible to make infinitely steep low-pass filters. In the early days, the filters were very poorly made and allowed high frequencies to fold down into the audible range, known as aliasing. This sounds very bad. Today modern DSP and oversampling techniques means that these filters can be made a lot better and in practice can push aliasing products down below -100dBFS in the audible range so they are no longer a problem. This is a bit harder with 44.1kHz than with 48kHz.

Formats like SACD which samples at MHz is a different process altogether since they only sample with 1 bit but with heavy oversampling. Most 48 kHz samplers also sample in the MHz range but will then decimate the signal to reduce the sampling rate but gaining bit depth.

The reason some people prefer analog is due to the imperfections which can add 'warmth' and 'personality' to the sound. This is also why some people prefer valve amps over transistor-based ones.

Regarding harmonics, they are per definition multiples of the fundamental frequency.
From https://en.wikipedia.org/wiki/Harmonic:

A harmonic of such a wave is a wave with a frequency that is a positive integer multiple of the frequency of the original wave, known as the fundamental frequency. The original wave is also called 1st harmonic, the following harmonics are known as higher harmonics.

You may be thinking of subharmonics but this doesn't really apply with sampled signals.