module a-187-1 is a dsp based effects module. four parameters of the selected digital effect are voltage controlled. the main effect (e.g. reverb, delay, pitch-shifter, equalizer) is selected by the two small
buttons effect up/down. the upper row of the display shows the effect that is currently selected.in the lower row the four parameters are shown as well as a small bar left from the abbreviation that displays
the current parameter value. each parameter can be adjusted manually (upper potentiometer row) and modified by external control voltages (lower row of the potentiometers and upper row of the sockets). the lower
row of the sockets containes the two audio inputs and outputs.
another button (bypass) is used to turn the effect on/off. when bypass is chosen the upper line of the display shows in turn "bypass" and the name of the pre-selected effect. in the bypass mode another effect
can be pre-selected and called-up by pressing the bypass button again. even the effect parameters can be adjusted and are displayed with the bar graphs. but they become effective not before the bypass mode is
left.
a list with all effects and the voltage controlled parameters is shown below.
the module is equipped with two audio inputs and two audio outputs because the dsp board features stereo audio processing. the audio inputs can process usual a-100 signal levels without clipping/distortion. for
higher levels external attenuators (e.g.. a-183-1) or vcas may be used.
a ready made dsp board is used inside the a-187-1 module. consequently only these effects and parameters are available that are supported by the dsp module. some parameters can be changed only in steps or with
audible artefacts. the dsp board has 20 bit da and ad converter with 32 khz sample rate available. the inter sound processing uses 24 bit. some audio examples can be found below.
table of available effects:
effect name
parameter 1
parameter 2
parameter 3
parameter 4
equalizer 1
31 hz
62 hz
125 hz
250 hz
equalizer 2
250 hz
500 hz
1 khz
2 khz
equalizer 3
2 khz
4 khz
8 khz
12 khz
distortion
input
resonance
vcf
depth
pitch shifter
shift
resonance
vcf
balance
reverb
predelay
reverb time
high dump
volume
echo
time (*)
feedback
high dump
volume
chorus/flanger/echo 1
delay (*)
feedback
rate
depth
chorus/flanger/echo 2
delay (*)
feedback
rate
volume
delay & reverb 1
delay (*)
feedback
time
volume
delay & reverb 2
feedback
volume
time
volume
chorus & delay
feedback
volume
feedback
volume
remarks/technical datas:
reverb predelay ( around 0 .. 50 ms), reverb time ( about 40 ms up to 500 ms) / chorus/flanging (1 ms to 41 ms) , delay ( 1 ms .. 165 ms) - all in 128 steps
chorus/flaging rate about 0,025 hz till around 12,5 hz in 128 steps
pitch shifter: -12 half tones .... + 12 half tone - in half tone steps
all equalizer parameters in steps of 1 db from about -24db to about +12db
(*) possible artefacts - audibility depends highly on input signal and the other parameters
the sound source is a simple 2-note sequence generated by two a-110 vcos processed by an a-131 vca which is controlled by an adsr. the original is heard in the first cycle. then the reverb time (parameter 3)
is increased little by little until the maximum is reached. the sequence stops to hear the pure reverb. the sequence starts again and the reverb time is set to a lower value. then the pre-delay time
(parameter 2) is modulated by the same adsr that is used to control the vca. the resulting effect sounds a bit like a spring reverb. after a while the reverb time is increased again and the sequence
stops.
a simple 3-note sequence generated by two a-110 vcos processed by an a-106-6 vcf with adsr is the sound source. this original sequence is heard in the first cycle. then the delay effect is added in the second
cycle by changing the effect volume (parameter 1). in the following cycles the delay time (parameter 2) and feedback (parameter 3) are increased. at the end the feedback is set to maximum and the audio input
signal is removed (i.e. only the delay loop is heard).
a male voice loop is used as audio source. the unchanged original signal is heard in the first cycle. in the second cycle the pitch-shifted signal is added to original signal by changing the signal balance
(parameter 4) and the pitch shift is changed manually (parameter 1). in the following cycles pitch shift (parameter 1) is modulated by an triangle lfo and then by a rectangle lfo.
a short sequence generated by two a-110 vcos with an a-131 vca controlled by an adsr is the sound source. in the first cycles feedback, rate and depth (parameters 2, 3 and 4) are changed manually. then the
internal lfo is turned off (parameter 4 / depth = 0) and the delay time (parameter 1) and feedback (parameter 2) are controlled by a cv output (delay time) and a gate output (feedback) of the sequencer. in
this patch each note of the sequence has a different delay time and feedback.
nothing but noise (a-118) processed by the a-187-1. feedback and depth (parameters 2 and 4) are at maximum, rate (parameter 3) at a medium setting. the delay time (parameter 1) is changed manually from minimum
to maximum - a little bit more after each lfo cycle. at the end of the example the rate of the internal lfo (parameter 3) is lowered.
another short sequence generated by two a-110 processed by an a-106-6 and a-131 with adsrs is the sound source. this original sequence is heard in the first cycle. in the second cycle the effect is added
(parameter 2) and the echo feedback (parameter 4) is increased. then the echo time (parameter 3) is increased. after a while the pitch shift is added (parameter 1). the pseudo-polyphonic sound comes only from
the echoed pitch-shifted signal! the pitch of the two a-110 is not changed. the pitch shift (parameter 1) is modulated by a s&h which is fed by a random signal (a-118) and triggered at step 1 of the sequence,
i.e. the pitch shift changes with each cycle of the sequence.
this module has a maximum current draw of 200ma. it requires 18 te/hp worth of space to fit in a eurorack frame.
