ginko synthese sampleslicer, full kit
ginko synthese sampleslicer, full kit
ginko synthese sampleslicer, full kit
The ultimate realtime sampler for your modular. No, it's not another sampleplayer! It's a realtime cv controlled sampleslice sequencer.
One with features that move it far beyond the scope of a regular sample player.
"I don't just like it, I LOVE it! This module is perfect. I would take it with me on a deserted island and will never let it go. Automatic speed matching to incoming clock is amazing! At first I wondered why it needs a clock. Then it amazed me in every aspect."
— Meng Qi
What does the Sampleslicer do?
The Sampleslicer is a real-time monophonic sampler, including a 16 step voltage-controlled sequencer.The incoming sound is chopped up automatically into 16 parts by the incoming clock signal and spread out over the16 steps in the internal sequencer.
How long can it sample?
The length of the recorded sample is set by a clock-divider, so the recording time is always linked to the incoming clock signal. You can choose between time divisions of 1, ½, ⅓, ¼ and ⅛.
If the divider is set to 1 the total sample time is equal to 16 incoming clock-pulses.
If set to ½ the sample length is equal to 8 incoming clock-pulses, spread over 16 slices (so half the time). It will produce even more glitchy fun when set to ⅓, as it will spread 6 steps over 16 slices. The maximum sample time is about 15 seconds and the minimum sample time is a fraction of a second for granular noises.
How does the sequencer work?
Inside the Sampleslicer there is a 16-step voltage controlled sequencer to play back the sound. Every step is dedicated to a 16th part of the recorded sound inside memory. When the 16 steps are full the sequencer starts playing from the chosen start point till the chosen amount of steps are played. This can be in one-shot mode or in loop mode for infinite looping.
The start point and play length is determined by potentiometers or CV input.
Last but not least the individual slices can be played back as notes via CV from a sequencer or keyboard, by setting the start point in V/oct mode and play length to 1.
How does the pitch work?
There is a control to pitch the slices up or down. The pitch control affects the overall pitch of the recorded material. It can be controlled via CV.
Why only realtime sampling?
Our vision on the modular world is that you get most fun out of realtime created sounds. Out of principle we chose to make this module without a memory card reader.
What about the soundquality?
The sample rate is 12bit, just like the good old sampling madness days. A lovely sound with a slightly raw character without being gritty. For comparison: CD digital sound quality is 16bit, gameboys are 8bit, the E-mu Emulator is 8bit, E-mu Emax is 12bit, the MPC60 is 12bit, the AKAI S612 is 12bit, the AKAI S900 is 12bit, Oberheim DPX1 is 12bit and the EMU sp-1200 is 12bit too…
In order to maximize sound quality and creative usage possibilities, we choose not to place an anti-aliasing filter in the audio output path. The digital-to-analog conversion process already gives a clean and rounded sound, and filtering would tend to take away some brightness and aliasing can give some extra to your sound.
It is designed to be part of a modular system so we leave it to you whether you like to patch the input/output through a filter or not.
What is aliasing?
If you experience noise in combination with some audio sources (but not all), then most probably you use digital audio sources with a lot of sharp corners in the audio. This causes aliasing, which might sound as noise. If this is the case with your setup, simply put a lowpass filter in front, slightly turn it on and all aliasing noise is gone. For example, if you use audio created with PWM, the aliasing sounds terribly noisy. This can be explained by the PWM being only "high" or "low". (The PWM I talk about is not the PWM on your analog VCO but digital created pseudo analog voltages by making a 5V pulse wider or smaller.) If you send digitally created audio through a lowpass filter the corners will be rounded off, the highest frequencies will be filtered and the aliasing/noise will be gone.
Analog audio will sound clean unfiltered and will have nice smooth aliasing.
Read more about aliasing on Wikipedia.
So, no, it is not 24bit as we know in the mastering studio’s. Instead, it will remind you of the good old “hip hop” sounds from the 80’s :) and it’s lovely! In the end it’s an instrument by itself not a mastering tool.
Be creative and try the module in any way you can think of! Feel free to share your results. I am excited to see what you come up with!
depth (including power cable): 25mm
The SAMPLESLICER can bring a bit of stress to people when they plug it in for the first time...
Don't stress there are some things you should check first:
What if your output is full of clicking sounds?
Don't stress it is a simple thing, you just have to recalibrate your module. To calibrate the sampleslicer turn of your modular. Turn the gain knob fully anti clockwise, now repower your system while holding down the "sample" and "pitch mode" button. Keep these buttons pressed down for a few seconds and the first LED will light up. Now release the buttons and the LEDs will light up one by one. When the calibration is finished you will see a binairy code on the LEDs. Just ignore this, this is the voltage it is calibrated to. Now restart the module and push the pitch mode after you powered up. Now restart one more time and have fun slicing again!
Why doesn't the SPEED knob work!
The SPEED knob works as a clock divider/multiplier for the incoming external clock. Therefor it only works when you feed the module an external clock.
I don't hear anything!
There are a few things which can cause a silent module.
If you don't hear anything while recording, just flip the little "auto/sample" switch to "auto". In "sample" mode the module only puts out the sound it recorded in it's buffer. In "auto" mode the module sents the incoming audio to the audio output when it does not play sound from it's buffer.
If you only hear the recorded audio one time after you pressed record, than just flip the "one shot/loop" switch to "loop". "One shot" is mainly used in combination with a gate or trigger input.
If you don't hear anything and the sequencer is not playing you might have plugged in a jack in the "gate" input. This will stop the sequencer (and no sound from the output as a result) when no gate signal or trigger is sent via this jack.
The PITCH knob does not work!
There are 4 pitch modes, which you can select with the little push button called "pitch mode". You will see another LED light up every time you press this button. After the 4th LED no LED will light up when pressing the button once. Now you are in a mode where the pitch modulation is disabled. The output will always have the same pitch as how it was recorded and the PITCH knob will not react. Just press on the little button again to come back to one of the 4 pitch modes.
The ATT knob does not seems to do anything!
The ATT knob is an attenuator for the incoming CV for modulating the pitch externally. This means that this knob only works when you feed the modula a CV signal on the pitch CV input.
The clock, gate and(or) sample trigger input do not work!
The kit is provided with BC547B transistors while I originally designed it with 2N3904 transistors in mind. The change in transistors is not noticed for many years and it works fluently for almost everybody. The BC547B is somewhat b-directional and therefor the CPU chip is sensing the voltage drop accurate in almost all cases. But although it mostly works I have to admit that it is a mistake I made. The BC547B should be soldered in reversed direction. So if you notice any hick-ups in one of these inputs, the best thing to do is to desolder the transistors and place them back reversed or replace them with a 2N3904.
If you come across any issue during the building process, please sent me an email at firstname.lastname@example.org. I am happy to help you.