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So ive been converting a bunch of files for use with my amiga.
I made them all into 22.05khz mono files and transferred them across through compact flash.
When i played around with them before i did the conversion i had just noise, and after conversion it was the same.
BUT turns out when i upsample a bunch i eventually reach the sound.
Is there any reason why my files are playing at what seems deathly slow speed and not what they should be?
You mean sample files? File conversion to IFF or RAW for Amiga is a fucking bitch. We all ran into this issue at one point or the other.
My simple solution (albeit maybe not the best): Open Milkytracker, load your WAVs there, then save in MOD format. Open the module in Protracker and you can have access to the samples. you will have to fix any existing loop points, but they will sound OK.
The standard rate for the Amiga (to be in C) is 16726Hz or half that if you want to be an octave lower. Throw all the modern day "standard" 44.1khz and 22.05khz convention out of the window because there were no standards back in the 80's!! You'll want to export using a program that can write raw signed files - like early version of Audition (or Cool Edit) and I think Goldwave still supports signed / unsigned conversion too.
As Akira says tho, the easiest way to get your samples across is to use MODPlug or Flakeytracker and "save as compatible". The take the samples across in a MOD. Again - 16726hz is the tuning point for C ... and you can do up to about 29000hz if you want clear / clean audio for high hats or something else critical but beyond about 29k it'll be out of range. I'd let Milky or MODPlug do their own 16bit>8bit conversions rather than saving 8 bit out of your DAW as they will both do a rough quantize which usually sounds better than a "dithered" nuskool conversion.
(ps. Max sample length is 32k - which will be 64k as a 16bit file)
Last edited by cTrix (Jul 12, 2011 4:52 pm)
I do something like "sox -D in.wav -r 16726 -b 8 -u out.raw" using SoX to convert my samples, but most often, as cTrix says, a rough conversion sounds better for most sounds. I haven't figured out how to do that in SoX yet.
EDIT:
Maybe I did! I think -D is for disabling dithering.
Last edited by boomlinde (Jul 12, 2011 6:01 pm)
Ok I'm using this program called switch to convert everything, converting to .raw 16000hz 8bit unsigned.
I'll see how this goes.
you can't choose a custom samplerate? It's also SIGNED that you want ;-)
ah, fair enough time to do a new batch convert then
I think you can set PT 2.3 to import either signed or unsigned. In the former case, remove the "-u" switch from my example.
Use goldwave batch converter. I managed to convert thousands of files and they all work perfectly. Keep in mind that nothing sounds quite as good (crunchy) as sampling direct.
But if your sample is NOT on C, defaulting to a certain Hz range would introduce MANY problems.
Sample and listen is best policy on rates. You usually find a sample rate which trades fidelity for file-size. Although don't go on Nyquest as something like a kick or sine bass will ring annoyingly at say 8khz... and will sound a lot more rounded at say 26 - 28khz. Because of the way an Amiga aliases the sound, sometimes hats and the like sound crunchy and awesome at lower sample rates. Don't rely on your PC to correctly preview what a low sample rate will sound like :-P (it fills in the "missing" samples on an upconversion to 44.1khz/48khz with a curved waveform instead. The amiga will leave the sample values as DC with square amplitude changes unless you have the filter turned on)
Sometimes a sample rate is chosen so that the "highest" note for that instrument hits the top A# (highest non-tempormental safe note). I try to do this for leads and rounded bass instruments. I've lost count of the amount of times I've gone out of range by a note or two and had to go back and knock a few Hz off the samplerate then transpose an instrument a few notes down. Else you end up octave-resampling for the sake of one or two notes!
Last edited by cTrix (Jul 13, 2011 9:15 pm)
Curved/linear interpolation on upsampling depends entirely on the player, and none that I use do it unless I explicitly tell them to. Try switching interpolation on/off in xmplay to hear the unmistakable difference. The Amiga DACs are not very linear though, so that could account for some difference in sound quality.
ms dos batch file
it's based on the example batch file that came with sox
cd %~dp0
mkdir converted
FOR %%A IN (%*) DO sox %%A --type raw --rate 16726 --encoding signed-integer --channels 1 --bits 8 "converted/%%~nxA" rate -v 16726
pauseit seems to work ok
i agree SoX is the way to go. if you don't want to bother with the command line, creating a batch file like above and dragging your samples onto it is convenient.
heres a table of frequencies and corresponding pitches for the amiga:
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 31677I didn't think anyone actually bothered with notes.
Pssh. FINE.
Yeah most of mine is crunchy as hell drums to go with my bass. The less tuning involved the better.