Yes Axo is much more low level than G2. In G2 you can only work with the objects that are there and that is it. But we can build our own objects Axo. Very nice, but also more complicated.
Technobear, if you didnt se the video Sputnki posted with clips of Alchemy you should watch it. It basicly has got all the paramters that would be great to implement in the sample manipulator.
Maybe you get some ideas on how to implement samples and modulations from that video.
here... this is how simple it can be in Axoloti
the bottom row was an attempt to play the recorded data in a raw way, but unfortunately this wont work (I think) as the oscillator doesn't go high enough, we'd need 48khz - @johannes any tips?
(btw: small bug, I think at 24khz it goes silent)
playing with this, I thought it might be nice if the table/play and record object had a trigger output, when they get to the end. (I recognise this is at k-rate, so could be 1/3000 sec later for a retrig, but better than nothing)
poly expressive... oh all sorts of possibilities, and of course you can drive these changes programatically (e.g. via a sequencer) per voice. actually a cool thing i saw recently (not sure where), was someone using a pad, and then using that to create wider detunes on different voices, worked really well
I'll have a look at the tutorial - thanks I've not really done much with manipulating samples/granular synthesis but always interested to try new things.
(I played with Kaivo from Madrona Labs which is great, but couldn't really get anything musical from it ... due to my skills, or lack of )
24kHz = nyquist frequency at 48kHz sample rate.
your osc/phasor lin freqency should correspond with table length/duration for untransposed playback.
Also no need for bipolar2unipolar in this case, the phasor is unipolar (hinted by the + sign at its outlet).
phasor unipolar, oops got caught out by scope, its presumably does a conversion to bipolar, as when I fed it in, it showed a bipolar signal.
24kHz - yeah, I was confusing the rate at which we drive the table (which is always audio rate) and rate at which the ramp needs to be..
hmm, going to need to think about this... for raw
so it we have 96k samples, that 2sec .... which means i need a frequency of 1/2sec - 0.5 hz, over the whole table i.e. to got from 0 to 1. therefore I need a smooth ramp at audio rate at 0.5hz...
should be easy enough, just need to create an audio rate LFO... which in the case is just the phasor lin, but with units not scaled ... but when I try that i still don't get what I need.... do I lack resolution? because of the way table read works?
(_USAT(inleta,27)>>(27-attr_table.LENGTHPOW)) ... does this mean i only have 5 bits of resolution?
hmm, I knew this was going to get into 'new territory for me' 
Looks great, Thanks WIll definatly take a look at it tommorrow, when I get the Axoloties.
by design, Axoloti patches are easily reproduced from a screenshot.
here's an attempt at that: samplethetechnobear_fromscreenshot.axp (5.5 KB)
(haven't tested)
hej everyone, just trying out this sampler example (thanks for putting this up technobear). I still haven't figured out where to change the parameters you were talking about. If I understood correctly:
change sample rate to 48kHz
delete unipilar2bipolar
where do I have to put 24kHz?
best!
amelie
This is something i'm also very interesting in.
Looping/sampling/audio manipulation.
My digital electronics knowledge is sound... my programming? Non-existent. So I'm looking at patches to try to understand what is going on.
I have recently bought some AVR programming texts as I plan to understand programming like I can understand schematics. 
I've found axoloti's tools to be good so far.
Would slicing/splicing a sample be doable.
I'm assuming it's just a case of highlighting start position?
