Just load into Audacity (freeware) and re-save with the correct file type, raw/no header etc.
No, the standard WAV header ist just 44 bytes (as i mentioned above).
The thing is: WAV is only a container format and the first block in the header describes whats in it.
WAV files support different sample rates, multiple audio channels and different bits per sample … and even different data encodings (like compression). All this is encoded in the header so that the software later knows how to load (and play) the WAV file.
The .ws6 file has and needs no header, because it has no sample rate, only one channel (=mono) and allways 16 bits per sample. It’s pure data in a predefined format.
If your WAV file with 4096 samples is 16.xxxkB, it might have 32bits per sample or two channels.
And even if it had the right amount of data, a text editor ist not the rigth tool für cutting of the header of any binary data file.
But to be honest: THIS is really not the challenge for preparing a correctly sounding waveform!
If it’s just that one waveform you need, send the WAV to me via PN, i’ll take it as test data for my WAV2ws6 Converter and send you back the result.
But it would be better if you sent me a longer WAV (more than 10 000 samples) that contains your desired waveform within a countinuos tone, so that i could make sure to get the right start/end points for one single cycle waveform.
2 Likes
MOOGelPackung
Thanks for offering to convert my sample material, but I will decline your kindness.
This is the most detailed explanation I have seen on this topic. It is clearly way too complicated for a non computer literate person to do anything meaningful. My secondhand Applemac is OS10.5 and I will not get a new computer just to dabble in this. I have other music gear for making weird noises, and maybe in time more waveform banks will be downloadable. UDO Super6 is a tiny player community, so it’s never going to be a big commercial market.
You can only put 16X .ws6 samples in at one time, and it is, let’s face it, a right pain to do the job.
It’s not a question of computing power, just a question of background knowledge in using the right tools in the right order.
This is only quarter of the truth!
You can have 16 waveforms PLUS 16 alternate waveforms, so there are at least 32 predefined waveforms available for random selection (and in fact, because of the lack of wavescanning or wavemorphing or granular synthesis functions is see no real need for having much more).
On top of this, each patch stores it’s own set of DDS1 & LFO1 waveforms, that can differ from the predefined waveforms … and each patch can be used as a starting point for futher sound manipulations, so there is more than enough space for user waveforms, if you want to use exactly this single feature excessively.
1 Like
MOOGelPackung
Many thanks. I did know patches retain their waveform, when the waveform set is replaced. I did not know about the second bank flashing LED thing also applies to waveforms, and will give it a try. Don’t get me wrong, I love my Super6, because the hi-rez audio is so crystaline clear and free of aliasing and the sub-bass is incredible.
@MOOGelPackung
Hi man!
What’s the best way to convert a 1024 sample wav file to a 4096 one?
- option 1: I copy 4 times the 1024 sample wav after each other to get the 4096 one.
- option 2: I copy the value of each sample 4 times after each other
- option 3: ?
There are many AKWF at 1024 samples…
Thx in advance for your ideas!
Someone told me to use a different sample rate. That’s actually a good Idea. In that way, you are thinking about a cycle. If you copy the waveform which is long 1024 sample (i.e. x 4 times) you’d have a higher pitched waveform, and UDO does not aloud you to assign a key note. This would be great to be implemented honestly (would facilitate a lot the management of the waveforms) @udo-audio
What I did in my waveform converter is linear interpolation.
So if you have a source array s[0…1023] and a target array t[0…4095]
than
t[0] = s[0]
t[1] = (3 * s[0] + s[1]) / 4
t[2] = (s[0] + s[1]) / 2
t[3] = (s[0] + 3 * s[1]) / 4
t[4] = s[1]
t[5] = (3 * s[1] + s[2]) / 4
… and so on
This gets a little more complex if the source sample length is any given size, but it stays simple linear arithmetic.
After all you should apply the required lowpass filter anyway.
1 Like
Thx for your answer and confirmation!
This is what I already did (can handle any number of samples and done with a loop of course) 
What do you mean by “you should apply the required lowpass filter anyway”?
Is there some code/algorithm somewhere?
1 Like
How do you get the waves from the Prophet VS Arturia plugin? I have it, but I can only find a file called waverom.bin that I can’t open.
EDIT: I was able to get the waveforms, but I truly don’t understand this. I wish there was an editor that I could use to convert to the correct format. I read through this entire thread and don’t understand how to accomplish any of it.
1 Like
Ive tried all the instructions to get a waveform from sheets of sound into the super 6 via audacity and just having no luck. Would someone be able to post a fool proof set of instructions to get it to work by any chance please?
2 Likes
Here is what worked for me:
I just recorded a bunch of Buchla waveforms pretty low in register. Maybe A1 or lower, I can’t remember. This isn’t super important, they just needs to be at the bottom of the register. Most of the stock waveforms are one cycle, sometimes they are two cycles. Two cycles raises the pitch an octave. Not a big deal since DDS1 goes to 64’. I like to combine two different waves on each cycle. That can lead to some interesting results.
DAW
Record your waves in your DAW.
Bring one of the stock waveforms into your DAW and look at the size of the sample and cycle length.
Enable loop recording and tune your source so a duty cycle matches the stock waveform duty cycle.
Zoom in and crop recordings to two cycles at the zero points.
Use Fabfilter low pass filter 48db linear phase max @ 512hz
Render to a separate track below.
Fabfilter reports latency so you should be able to cut along the borders of the wavs pre-lpf. Use the other track and clip edges as a guide. Ableton will snap to the borders.
Crop the post-lpf waves to match the length of the pre-lpf waves.
OTHER WAVEFORMS
If you are using Adventure Kid waveforms or other single cycle waveforms that are cleaned up(zero crossings are perfect), I recommend importing them into sheets of sound, changing the sample size to 2048, and exporting. In Audacity, you will duplicate the wave to make two cycles, thus 4096 samples. No need to low pass filter or normalize AWFK waveforms.
SHEETS OF SOUND
Load the cropped waves into sheets of sound website and change sample size to 4096
Normalize
Export
AUDACITY
-Locate your post Sheets of Sounds samples/recordings and import them into Audacity as mono wavs.
-Export all as raw and headerless 16-bit signed.
The file size is 8kb (actually 8,192b)
FILE NAMES
-Now rename the extension .raw as .ws6
My mac was showing the .raw as panasonic raw under right-click get info/kind tab. I changed the opens with association to textedit and restarted the computer. When I renamed it or removed the .raw a pop-up window says something about keeping it .raw or changing it to .ws6. If this doesn’t happen right click on the file and unclick hide extensions.
Change it and now it works. Hopefully, these clues will help you.
5 Likes
thats awesome, thanks for taking the time to write that. I’ll try it out right away 
The instructions are updated. I ran the process again and greatly simplified it.
Oh man, I’m failing over here in making my own user waves.
To my mind what we need is one program that enables us to make 4096 sample user waves from longer samples BUT also offers the ability to remove the periodicity (which I will call false harmonics) which may arise from trying to crop longer samples into only 4096 points.
I have Wavelab and Halion over here and they’re needlessly complicated. I was never a fan of Audacity as great a free program as that is.
So what I’m realising is that we need some sort of very involved resynthesis program. If for argument’s sake we say we have a sample which is longer than 4096 samples, if we limit it to 4096 samples, any periodicity that does not fit within 4096 manifests itself as a harmonic of 4096. So what we need is some program that will either allow us to remove/lower said harmonics or more preferably take an average of the harmonic makeup of a longer but still periodic waveform.
The latter seems the remit of some very bright and diligent people that might (understandably) not want to offer their software for free. I can appreciate how much resources UDO are willing/unwilling to put into a separate user wave editor but I feel I’m totally underutilising my S6 if I don’t exploit the user wave part of it.
TL;DR I want to extract 4096 point samples from my ROMplers and edit them so they are as pure as 4096 points will allow, I wish one program would allow me to do this neatly and relatively painlessly.
I don’t know if this has already been mentionned here:
3 Likes
I had not seen that!
Thanks, will check it out now 
1 Like
Shouldn’t the rule-of-thumb cutoff frequency stated in the manual actually be 5,512Hz (as “sampling frequency / 8” assuming a common “rule-of-thumb” sampling frequency of 44,100Hz for source material), and not 512Hz ? My reason for this is reinforced by looking at the spectral plots of the UDO-Audio factory waveforms themselves (imported into Audacity using File > Import > Raw Data): most of those appear to contain frequency content well in excess of this 512Hz figure e.g. w1_.ws6, with a maximum frequency below 8kHz, usually around 5-6kHz
Applying a 512Hz 48dB/oct filter to the factory waveform w1_.ws6 alters its character significantly.
Before:
After:
Whereas applying similar low pass filter at 5,512 Hz makes very little difference to the original:
