Exporting raw audio on a PC for Amiga

[pulselooper]

Hi all,

I want to put on some samples on my a600 and use them with OctaMED, but I'm having some issues. These samples are in wav or aiff, and I'm using Audacity on the PC for exporting them in raw data. I tried to export 8-bit and 16-bit raw, both signed and unsigned. None of them worked. When I import the samples on OctaMED (or Protracker), the sounds are awfully distorted and with their pitch all messed up. Also tried Soundforge to export the raw files, but doesn't work as well.

I really don't know if this is the right way. Anyway, some tips on how to do it properly on the PC? Or is this a proper tracker setting on the Amiga?

Thanks for your time.

[Vegas]

Hi, i have the same problem. Apparently Audacity can export Amiga IFF Samples in 8bit but my results were the same as yours. Of course some quality loss is inavoidable. I also tried to convert the samples on the Amiga side, using various Aminet tools. But somehow the most effective way until now seemed to be resampling. If someone has some hints, i would be thankfull too!

By the way, Octamed SS can playback 16bit wave samples, if i remember correctly.

[Thorham]

Use Sox for the peecee. It's a command line program, which isn't too handy, but it should do what you want, and it can down sample quite well (which is why I use it). Get it here: Sox. Just one thing, the documentation is a little annoying

[pulselooper]

Originally Posted by Thorham

Use Sox for the peecee. It's a command line program, which isn't too handy, but it should do what you want, and it can down sample quite well (which is why I use it). Get it here: Sox. Just one thing, the documentation is a little annoying

Thanks a lot Thorham! I'm at work right now, when I get home I'll try Sox and post the results here. Cheers

[Vegas]

Thank you as well! I will give it a try soon.

[CMDR]

/wave pulselooper

i can sell you a sampler for £8, bargain!

problems solved??

[beefheartvliet]

Damn, Just paid £15.0 for one :-) Still Sox sounds useful also.

[Photon]

Note that samplers usually introduce a bias (shift of the zero-point crossing), and depending on how well you have grounded both computers, hum. Normal cables act as antennas, picking up interference from electronic equipment to varying degree. Make sure you connect sound only to line-out/line-in jacks, not amplified outputs or mic inputs, or you will get distortion. USB sound cards for musicians are usually the best (least likely to pick up electronic interference on recording/playback) for the PC, and AMAS is a good sampler for Amiga (bit cumbersome software though, but it samples stereo if you need it.) Combine stereo to mono by software/settings in the PC before sampling, ie. don't plug a stereo 3.5mm plug in a sampler's mono 3.5mm hole.

GoldWave can load/save Amiga IFF and reads raw, just input frequency, 8-bit signed PCM mono and it loads correctly. Raw can be had by saving as IFF and re-save the sample in Protracker or Audiomaster, or by saving as raw and specifying PCM 8-bit signed mono.

When converting from 16-bit to 8-bit, you *will* get noise and distortion on low volumes (such as faded out ends of instruments), simply due to aliasing. Going from 11kHz to 28.8kHz improves this, but only slightly. I've planned and am about to code an improved converter, but I'll give sox a go - if it's better than GoldWave/Soundforge/Wavelab, I might save myself some time

[Coagulus]

I'm sure I got samples onto my Miggy from PC by using Modplug tracker or similar on the PC and saving as .mod - might work!

[Photon]

Sox output sounds identical to the output from GoldWave (which also has batch processing), BUT: Goldwave has a strange distortion of the fadeout end. So it was a nice find, sox is better - but disregarding the endfade flaw in Goldwave, the sounds were identical.

Here's the commandline I used:

sox pcfile.wav -b 8 -e signed-integer -t raw amigafile.raw channels 1 rate -v 28800

Mine was already mono, you could add mixer -l, -r, or -1 if you want to select left, select right, or mix channels ONLY if the sample is stereo (sox will throw up if it's mono). Put it after the output filename.

Tried several dither options, they did not effect the sound *at all* audibly (I have good equipment). Tried some treble cutoff filter settings that could improve things, but concluded that the above was as good as it would get.

I'll describe how do I do it. You can use SoundForge if you want. First, mix sample to mono and normalize. Or better, use compressor/limiter to use most of all 8 bits. Otherwise your sample can be extremely noisy. Resample to 22 050 Hz and save sample asi 22kHz, mono, 8bit standard wav. You can open it with protracker 3.61.

[Photon]

Originally Posted by nooly

use compressor/limiter to use most of all 8 bits. Otherwise your sample can be extremely noisy.

That will change the sound envelope, and (if used in a song) will change the sound mix levels etc.

That might be quite okay, if done right - ie. if the musician had already compressed it and knows what he's doing/what he wants.

Also, it doesn't so much remove aliasing noise as drown it out. And it won't fix a fadeout. An mp3 converts fine to mono 8-bit with only a slight background noise which stay at the same level all the way until the end of the song. When the song fades out the same noise level is suddenly quite noticable.

Normalizing is important, though! sox supports normalizing but is bitchy about it, it may throw up at the gain level for any random sample and not convert it.