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nw1 Wavetable Module for Eurorack
nw 1 frontnw1 is our debut product for the popular Eurorack format. It includes an advanced wavetable engine with independent control of spectral envelope and noisiness – just like Nave.

Wavetable scanning is cyclic with optional modulation of travel speed, position, spectrum and more. By providing the worshipped Waldorf wavetable set from the classic Microwave and Wave synthesizers, nw1 will enrich your Eurorack modular system as a potently powerful sound source.
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link

http://www.waldorf-music.info/nw1-overview

I always liked the grainy wavetable sound of Scope - others prefered the smoother result when using a higher sample rate
(96k is a very obvious change with Waldorf OSCs)
the Nave (iPad App) mentioned on Waldorf's page is similiar to the quasi oversampled sound
very smooth tone, nice filters - one of the best iDevice synths btw
so I'd expect another more 'modern' implementation in the module - price is cool

cheers, Tom

That's such a good point about S/R. I love the grit too...my Korg DSS-1 has a very gritty sampler (12b @ various slower rates), but combined with analog filters ... Becomes a great combo of grit and cream. Makes me wish scope modular could run each module at different rates!

Decimator?

forget it...
I haven't come across any decimator that didn't sound like crap
it's NOT the same (as all those side effects in analog devices - a converter is an analog part after all)

cheers, Tom

I replied to jk wish "Makes me wish scope modular could run each module at different rates"

By the way a decimator job is to make something sound like crap. Do you know how to use it?

In the digital domain it is the same effect more or less

no need to explain - I wrote 'crap' because you answered to jk
he referred to 12bit hardware - and no decimator will achieve that sound
I have a couple of 8bit Casio Samplers - and some decimator fx - nothing in common
not all crap is created equal

cheers, Tom

Can you post a sample of your unique crappy output?

why a sample if you can have the whole thing in HD ?
https://www.youtube.com/watch?v=s0_pz6cTUU4
watch out around 3.10: the lead sound from Air's Kelly watch the Stars ...

it's a PCM Drum/Synth, a Sampler and an Additive Synth with portamento and some crude signal processing/looping
the user Interface is brilliant - you never miss the display
an 8bit machine, but still has some nice 'round' tone to it - well shaped noises
(pure decimation will never get you there)

cheers, Tom

decimator doesn't always match the older digital hardware, but using shapers such as the flexor ones, something good can be done.

I can't listen to that right now but come on guys, if you 'd like post a sample of what you are referring to...

What means a decimator "can't match old digital hardware".
In that sentence there are 5 contradictions.
I'm using "old" "digital" "hardware", and post me a sample of what you are referring to
and i will try to create something similar if not i will admit what you are talking about..

Most likely you don't know how to use the decimator...

I watched the video, i can't understand what you are talking about...

I don't have any problem if you like that synth, but don't say "forget it" and similar nonsense
The old cheesiness can be replicated any time....

For me, running a filter oversampled at 96khz would be great...more control in the lower end of the spectrum, and greater resolution in the feedback paths. This taxes the DSP some more, and switching globally means all I/O run at that rate too..including my DAW.

Having the sample player use a 24/32khz rate, instead of 44.1khz then using decimation to down sample, produces the same band limited signal (or very close), but doesn't achieve the same harmonics.

the sampler can use any sample rate i use a lot of 24khz samples in 8bits with linear 16bits this is 8 bit information with the last 8 bits padded with zeros

they are gritty and play well

in the sts oscillator and because i use multisamples i have a sample per octave starting from b0 - fo depending on playback rate

the sts osc will still render the samples in the sample output rate but because mine are that low they keep the transient the problem with the vectron and the digital wavetable osc is they use 128 samples only when they are transposed down the sample is played slower the more slower the transient thump lags

mine dont suffer that as there root key is always on the transient and so have that clang

I wait for any sample/bit rate limited sample and i will mimic it with my scope running at 96kHz

Jhulk, I hadn't played with the new STS modules, and that's great they can play any rate. I'd guess they get converted to whatever sample rate scope is set for, but I'm also happy to hear they retain their grit and character.

Anyone know if the mod IV ladder filter is oversampled internally? It seems to respond differently at 96kHz vs 44.1kHz (Freq controls)...but I'm just getting 32bit control over it now, so maybe that will help over MIDI control.

The impossible guys is to go from a lower sample/bit rate to a higher one. The opposite is simple.

Not impossible...the up-sampler will insert 0s or extra copies of the previous sample, and then filter.

The bandwidth of the input signal won't expand, but the harmonics resulting from the conversion can change the signal.

fra77z

i already sent you some k3 wavetables they are 8bits saves as linear 16bit format so only have 8 bits of wavedata

now the reason why they change is that they dont has sr compensation they only work accurate at 48khz

thers a big discussion of this and why simon shark made sr compensation modules so that if you swap palyback rates it compensates for this

this is concerning in pitch and filters which also use pitch frequency

i was doing some tune test and i was setting tun e frequency in the key of a for the yamaha gx1 pitch setting which gives you severa; pitch settings in actual organ pitch ranges and i was setting setting them up and notices the hz setting on the in[ut of the pitch was not the pitch i was getting out

i ask sharc about this and we talked about the sdk that tells you about this not all modules have sr compensation the mvc do

but most modules dont you need to know this as either you add sr compensation modules or you set them so they are correct by control rangers this is also for delays

so i would say that the modiv modules is not sr compensated if it is acting defferent between sample playback rates

also it really is point less in having filter high than 16k as most of the oscillators in the scope range only have about 12khz range
you will notice this with filters where you have a filter set to 24khz that from about 3 oclock the sound does not change what i do is i listen to where the filter sound is constant with no change and find its start position then i set that as the top frequency in a control ranger and a start frequency of 20-30hz this allows when you have the knob at the bottom of travel you just get sub bass frequencies rather than no sound you will then get a full sweeping filter for the whole spectrum of your synth creation

upsampling as soon as it doubled the rate of root sample after that you get overtones as soon as frequency content goes above nyquist

the best way to solve this is max uptranspose of 11 semitones or use root not as max frequency playback and only downsample this stope aliasing or you over sample the reason for oversampling is the error in computation is less by doing the calculations at much higher frequency rates the aliasing is out of the range of the frequency playback rate and so is filtered out

but to over come this you either bandlimit or like i try to do keep the sample playing at its root sample and only downsample as its only reading the memory out slower no nyquist can happen