NEW MODULE
MATH
"divscale"
Multiply (attenuate) with a constant value, then divide by an integer.
To be used with the divide-outputs of the table-oscillators/LFO's to scale the input back to maximal quantification 4 steps, whatever the amount of quantification of the quantizer of the oscillator/LFO.
NEW MODULE
DELAY
"tappedDelay"
multi-tap delay with lowpass filter on each tap.
You enter the tap-velocity/timing/cutoff for each tap by external tapping.
eg. when using a midi note module:
connect gate-output of midi module to the tap-input of this module
connect note-output of midi module to the cutoft-input of this module
connect velocity-output of midi module to the vel-input of this module
first tap you play will reset the timing-counter to the "zero" position->incoming note
taps after this will set the timing/velocity/cutoff of the delays.
Up to 16 taps can be recorded.
when you want to set new timings for the delay, just hit "rst" and play a new rhythm.
on-module cutoff-knob is added to the recorded cutoff.
there is an onboard control for feedback, but be careful with this one. Full feedback will be 100% feedback for ALL taps, thus making the signal louder and louder as taps will be fed back into each other and multiply the volume..
NEW MODULE
ENV
"hahdEnv"
delay/Attack/hold/decay envelope, linear attack, exponential decay.
triggered, not gated
NEW MODULE
OSC
"2DHQtable"
updated version of the "2DtableMorph" oscillator.
Added interpolation for the readout of the tables and an internal sine generator to be able to fade out the table-harmonics using an external envelope.
NEW MODULE
DELAY
"loop4"
16bit 4 channel audio looper with channel "summing2one"
Allows you to record 4 seperate audio loops.
Syncs to external clocks.
Recorded loops can be saved/loaded to/from SDcard.
Channels can be combined using the combine-function.
-rstCount: send a trigger to this input to restart the audio loop. This resets the counter for recording/reading the audio buffer to zero.
-playChnl1 to 4: mute/unmute the audio for this channel.
-EditChnl: input between 0 and 3, selects the internal channel-buffer to save the incoming audio to.
-rec: overwrites the current buffer-position of the selected channel with the incoming audio.
-dub: sums to the current buffer-position of the selected channel with the incoming audio.
-delete: deletes the current buffer-position of the selected channel.
-clear: clears the whole channel, though at a low rate to prevent hick-ups. This can cause the channel to still sound until it re-loops again.
-combine2channel: selects to which channel the current selected edit-channel will be summed to. (0, 1, 2 or 3)
-combine: combines the selected channels into the combine2Chnl-channel.
-save: saves the current audio buffers to your SDcard.
-load: loads a saved buffer from your SDcard into the audio-buffer
-filename: use this to set the filename of your audioloop-file
the next two are for adjusting the play-position or to shuffle the loop by an external CV-pattern
-offset: this enables you to quick-offset the reading position.
-size: this sets the division/quantification of the offset. Higher numbers will divide the total loop-length (of the rstCount-timing) into smaller parts, allowing you to offset the audio in smaller steps. So, setting this to 4, while "offset" is 1, will offset the read-out of the audio-buffer a quarter of it's looping length.
-size attribute: sets the maximum size of a single channel. Total table-size will be 4 times this big!
ps. I experienced some "bugs" having audio recorded into other channels. If the sync/reset time is longer then the buffer, it starts recording the extra time into the next channel. I let this be for the moment, as it might be useful somehow.. if you don't want this to happen, just make sure the reset is within the table-size.
NEW MODULE
CTRL
"cfgKnobF" and "cfgKnobI"
knob with a text-config for configuring the width of the knob with a minimum/maximum value.
Expects integer numbers between -128 and +128 and scales it to the right 32-bit range for the cfgKnobF module.
how to config?
eg. for a knob output range between -32 and 64. (don't forget the ; at the end of each line!)
press the "config" attribute and type:
max=64;
min=-32;
the "cfgKnobI" module outputs integers and expects just any integer min/max values.
so with this one you'ld just add like:
max=56252315;
min=234;
NEW MODULE
DIST
"softComp"
direct-audio-compression with soft-treshold feature. No internal envelope follower, attack or decay, so it's more like a distortion, but based on the native dynamics module
NEW MODULE
DYN
"compressor2"
compressor with soft-threshold function, attack/release, ratio and gain.
very nice distortion with short attack/decay rates
has readouts for input/output volume, compression amount and treshold reach
NEW MODULE
DIST
"digiana"
digital LQ remake of a comparing amplifier/filter.
module compares incoming audio with the filtered signal and adds/subtracts a digital value to/from the filtered signal input depending on whether the incoming audio signal is higher/lower then the filtered signal.
saturate/volume control the clipping and volume of the signal
rate controls the frequency of the lowpass filters.
NEW MODULE
DELAY
"taildelay"
Delay with a precise feedback-time control.
delay-time functions the same as the other normal delays, but..
the tailtime is a direct representation of the time (in seconds) the feedback will take to die out, whatever the delaytime. So when you set the control to 30, it will take 30 seconds die out, whether it's at a delaytime of 100ms or 2 seconds..
NEW MODULE
DELAY
"swelldelay"
Swell-delay with a precise peak-time control (lowest tailtime)
delay-time functions the same as the other normal delays, but..
the tailtime is a direct representation of the time (in seconds) the feedback will take to die out, whatever the delaytime. So when you set to control to 30, it will take 30 seconds die out, whether it's at a delaytime of 100ms or 2 seconds..
As it's a "swell" delay, it features two audio-buffers using the exact same timing of writing/reading, but using different feedback times. As one is subtracted by the other, a volume peak in the delay will happen when either one dies out while the other rings on.
As we know both the tailtimes, we can also calculate what the volume-drop will be at the moment of the volume-peak (remaining volume of the signal that hasn't died out yet). So this is internally normalised to input level and goes through a gain stage.
A switch is added so you can have up to 2x gain for the delay, making it up to 2 times the volume of the input level (so watch it!)
As last, an input-attenuator is added to control the dry amount.
Though, I just figured, as the peak is normalised, even though the peak is at the timing set by the lowest tailtime, the duration of the entire delay will be longer, as there can be added quite a lot of gain to get the peak to the right level. This forces the delay to take the long-delay time as extra time after it hits the peak volume. So a tailtime of 3 seconds and one of 7 seconds will create a delay of 10 seconds with a volume peak at 3 seconds. Also, the two tailtimes may not be too close to each other.
NEW MODULE
FILTER
"BPstack"
Up 16x stackable BP-filter
-pitch controls the center frequency of all filters
-Pspread spreads the frequencies of all the filters over the entire spectrum
-reso sets the resonance amount
-rate sets the frequency modulation rate
-spread sets adds a phase-offset of each next filter-LFO
-depth sets the modulation depth
-stack sets the amount of filters that will be stacked (max=16)
Been using two 'syncKing' oscillators just fine up until I've press 'Sync Libraries' in main window - boom - constant pitch, not responding to any controls, did I miss something? 
Update: thought it could be an issue caused by 1.0.10 so tried switching to 1.0.11, after installation got an error about 'firmware version mismatch', flashed new, could not get anything live (error similar to Problem with 1.0.11 and uploading on sd card, switched back do 1.0.10 ('Repair' installation), 'firmware version mismatch' again, flashed new, can not get patch contacting 'syncKing' live at all, patch containing factory oscillator (sine) loads fine 
Edit2: just realise I've overlooked the new 'active' switch, but for some reason any patch containing syncKing won't switch live, other oscillators ie: 'nativeSaw' goes live just fine
I've just updated both the "waveformGenerator" and "guitarTable" modules, so they can be used together in a patch.
Just load in both the modules and refer the "guitarTable" module to the "waveformGenerator", no "guitarAllocation" module is needed as the "waveformGenerator" has it's own allocation inside.
The "waveformGenerator" will set the amount of maximum waveforms that can be created and will automatically create a sine,triangle, saw, damped-saw, pulse, 2 other standard waveforms and the remaining waveforms at random when the patch starts, within the harmonic bounds set by the control on the module.
After this, you can change any of the waveforms using the "guitarTable" module by selecting a waveform/preset, adjusting the settings and hitting the "do" button.
If you like your waveform-table, be sure to have connected a "filename" module, so you'll be able to save your waveform-table.
You can use both the 2Dtablemorph as the guitarOsc to play the waveforms.
The 2D tablemorph is heavy on CPU but allows lots of morphings between waveforms while the guitarOsc is simple (wave is selected) and asks as less CPU as possible (4%).
An in-between oscillator is coming up as well as a "simple-slow-design" oscillator.
NEW MODULE
FILTER
"3xPara"
3 parallel SVF filters in BP mode that are summed with the original signal to make a 3 band parametric resonant equaliser.
automatable cutoff, band and gain for each stage and a master gain-control
NEW MODULE
FILTER
"twinMorphSVF"
same a dualMorphSVF, but has 1 audio input instead of 3.
not sure what you'ld use it for, but might come in handy... (like, to use as part of a code in another module)
NEW MODULE
MATH
"GoldRat"
Golden Ratio calculator
Set the total size with the "size" parameter
the outputs will give the size of each next golden division of the smallest part of the previous division
The buttons "next" and "prev" can be used to get to even smaller divisions based on the current size.
Next will take the next smaller division and resets the control to the next smaller division, taking you "down".
Prev will take the current size and multiplies it with the golden ratio+1, taking you "up".
ps. at first sight you might wonder why only the smallest part is taken as a division, but if you'ld take the golden ratio of the biggest part, it again gives you the smallest part of the previous division..... doubling the amount of calculations needed, while getting the same numbers and all of them twice..
a demo of a couple of my latest modules and some older ones
drum-triggering is provided by the "rndWeightSeq" (patt/)
drums are from the "DR1" (edrum/)
modulation done by the "qtsFLO4" (lfo/)
melody by "rndWeightNote" (patt/)
waveforms are generated by the "waveformGenerator" and "guitarTable" (osc/)
bass is done by the "2DHQtable" (osc/)
drone is done by the "1DHQtable" (osc/ UPCOMING)
bass-filtering is done by "mostFilter" and "tripple peak" (filter/)
bass-delay is done by "swellDelay" (delay/)
delay on drone is done by "shimmerDelay" (delay/)
distortion is done on several places by the "softComp" (dist/)
end-compression is done by "compressor2" (dyn/)
