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Old 28 September 2012, 15:11   #1
h0ffman
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Where should I post this??

Hi Guys

I've just written a little guide on how to convert 44khz 16-bit wav's to IFF. Goes into detail about how to get the best sound quality out of the conversion. Anyone interested? Where should I post it?
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Old 28 September 2012, 15:43   #2
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Hi Guys

I've just written a little guide on how to convert 44khz 16-bit wav's to IFF. Goes into detail about how to get the best sound quality out of the conversion. Anyone interested? Where should I post it?
upload the proggy I wanna check thx
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Old 28 September 2012, 15:51   #3
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It's not a program, it's a tutorial
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Old 28 September 2012, 21:59   #4
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Where should I post this?

Thanks, h0ffman.

Please open a new thread for your tutorial in the support.Other forum, or I could move this thread and give it a new title for you.
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Old 29 September 2012, 00:54   #5
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I'd love to read your findings, mate! I know there are a lot of people wondering how to. Personally I have opted, lately, for resampling in the Amiga, but converting, if done right, would be less time-consuming (and PT's sampling interface is ass).

Been kinda busy too, I owe you an email. I haven't done anything yet this is why I didn't write
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Old 29 September 2012, 04:07   #6
clauddio
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It's not a program, it's a tutorial
there is no need of a tutorial...play 16 do that work very fine it does not need any extra docs
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Old 29 September 2012, 12:15   #7
Akira
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there is no need of a tutorial...play 16 do that work very fine it does not need any extra docs
Read the original post. H0ffman has tips on how to better approach and get best results when converting the audio, coming from his vast experience as a musician and Amiga MOD maker. Of course it is welcome.
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Old 29 September 2012, 12:51   #8
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I`m also interested in your tutorial. How to get best quality by convert/resample 16bit to 8bit is good to know.

ps: Is someone knows how to sample from mc-303 via TST2 sampler in best quality, please let me know. I only get much noise and less dynamic.
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Old 29 September 2012, 15:40   #9
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Dynamics are always a problem. Is your TST2 working OK? I think some times they can have capacitor problems.

I find that my TST2 is working OK, but there's always a lot of noise in the lower end. It's hard for me to sample clear kickdrums for example.
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Old 29 September 2012, 17:58   #10
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Yes please, had a hell of time getting reaper on PC to render a file in 8-bit 11kHZ that didn't sound like complete crap. Spent ages messing around with dithers, limiters and filters, and a reliable procedure would be great to know.
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Old 29 September 2012, 19:18   #11
Jack Burton
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For my audio samples I use Goldwave (on XP). I simply create a new empty 11KHz sound file. It is 16bit by default, so I save it (still empty) as an 8bit wav. I reopen this empty 8bit 11KHz file and record my audio track directly from any source, CD, web etc. then it is recorded in 8bit 11KHz, so I don't need to convert it !
The results aren't that bad and are less noisy than a conversion like 16bit 44KHz -> 8bit 11KHz.
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Old 29 September 2012, 20:17   #12
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Choosing a sampling frequency by default for every sample is wrong. Some things are best kept in different sample rates. Also, supposedly, different Hertz values are related to different notes, so just having them all at 11Khz would be incorrect (and specially since most things are base C and as thus should be multiples of 8363Hz, which 11Khz isn't).

Here's a table from a little book I have that came with CU Amiga and focus for a bit on these matters:


Last edited by Akira; 29 September 2012 at 20:31.
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Old 29 September 2012, 20:44   #13
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Sampling analog out from PC is actually the by far easiest and best way without getting lost in magic frequencies and stuff. But you really need to make the PC sound sound good in mono first, and you must have a sampler that's quality. I bought mine for 100EUR back in the day, but I don't think the 30 EUR "Radio Shack quality" ones back in the day will make you happy. They have dist, noise, and can't sample as fast.

If you get the classic "studio ground potential" hum, you have to check where it comes from, but a good sampler really can be better than conversion.

I wrote a tutorial post myself a while back to guide people which goes through all the options. Convert using sox to 8-bit, and skip the IFF nonsense, and that's as good as it gets digitally.

I started a converter to pseudo 15 bit for OCS 18 months back , but it's only at proof of concept. It needs to be supported in trackers and players, and I'm still looking for the source of my snazzy wave editor... it's somewhere on my old disks...

Wanted to link to (and read) h0ffman's post, but I guess it's not up yet.
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Old 29 September 2012, 20:51   #14
Jack Burton
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Originally Posted by Akira View Post
Choosing a sampling frequency by default for every sample is wrong. Some things are best kept in different sample rates. Also, supposedly, different Hertz values are related to different notes, so just having them all at 11Khz would be incorrect (and specially since most things are base C and as thus should be multiples of 8363Hz, which 11Khz isn't).
Yes indeed. I'm not a musician, so I don't record instruments. Only song parts. And then I cut them to make them correctly loop. It's just for the fun of experimentation. And if I do sample at 11KHz, it's obviously to reduce the file size (and to try to get the better quality for the lower, well, almost, frequency).

As for instruments, of course, you have different harmonics for a single instrument depending on the note/pitch/octave. And some instruments have very particular sounding which sometimes demand different sample rate, I guess. That's why a common "A" (440 Hz), on an electric guitar sample, for example, will sound weird when you change note/pitch/octave on a tracker (and a chord will sound even worse).
Now I didn't know that we should look for multiple of 8363 Hz. Thanks for the info. I will try to record some tune at this frequency to make some more experiments !

Very interesting table, thanks !

Last edited by Jack Burton; 29 September 2012 at 21:00.
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Old 29 September 2012, 21:01   #15
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@Akira:
The TST2 is ok. Records from CD or radio are good. A cable replacement didn`t changed the result. Maybe that is just the normal behavior?! Noise on low frequencies are quite normal. Just love it and you are satisfied.

@Jack Burton:
In reference to the topic are converting and recording different thinks (new tutorial?). However, your solution sounds like a stupid workaround for me. At least on Amiga you just choose frequency, stereo or mono and simple record that.
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Old 29 September 2012, 21:08   #16
Jack Burton
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@Jack Burton:
In reference to the topic are converting and recording different thinks (new tutorial?). However, your solution sounds like a stupid workaround for me. At least on Amiga you just choose frequency, stereo or mono and simple record that.
Okay, sorry for the stupidity, it was only my 2 cts...
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Old 29 September 2012, 21:38   #17
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Now I didn't know that we should look for multiple of 8363 Hz.
As long as you are using C as the base of your scale, yes. But if you aren't, then don't. Look at the table and it should sort out what you need.

In the case of what you do, I would recommend 16726Hz as the sampling frequency (C-3)
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Old 29 September 2012, 22:41   #18
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Akira, I think you're a bit mixed up about sample rate versus musical note frequency. Sample rate only determines the maximum frequency of note you can properly reproduce. (1/2 of the sample frequency). AFAIK, it has no other relation to musical notes. ie, any digital audio system should playback any note below it's maximum with equal accuracy. (antialias and reconstruction filter quality notwithstanding)

11Khz is usually short for 11025 Hz, exactly 1/4 of 44.1kHz, supposed to make sample rate conversion less painful. Again, no relation to musical frequencies, you just can't record or reproduce any notes or harmonics above 5.5kHz.

Maybe you refer to special tricks on amiga to reduce file size, creating multi sample instruments with increased sample rates for higher pitches or. . . ?

EDIT:H0ffman has posted the same table in his tutorial and these numbers relate the musical frequency to the amigas PAL clock frequency. I guess there is something going on here with the amigas unique hardware that I don't understand? Excuse my ignorance but it just looked very odd to me to use different sample rates based on the note you are recording.

Last edited by jimbob; 30 September 2012 at 02:48.
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Old 30 September 2012, 00:12   #19
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Whatever frequency any sound is sampled at, it only requires a fresh table to map it onto keys. A tracker that asks for each instrument what key it was sampled at and keeps a table for each instrument would be required, currently the PT frequency table (1) is stuck in 1988.
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