General Description; Power Supply; Analog Signal Paths - Alesis MicroVerb Service Manual

1.0 General Description

The MicroVerb and MicroVerb II digital reverbs achieve their results by slicing analog signals
into segments, and then converting them to a numeric value corresponding to the amplitude of the
signal at that particular instant. These values are then mathematically manipulated and stored at
various locations in a memory "loop" for eventual playback. By varying the placement and amplitude
of incoming samples, discrete time delays are achieved. When mixed together, and converted back
into analog, these delays simulate the reflections associated with natural reverb, as well as non
natural effects such as reverse reverb, and gated reverbs. This service manual covers both the
Alesis MicroVerb and MicroVerb II, since both units are virtually the same. The differences between
these units are the algorithms programmed into the EPROM (U1), the digital clock rate, and the
input filter values.

2.0 Power Supply

The power supply begins with the 9 Volt A.C., adapter (Alesis P2 type). Input from J6 is R.F.
filtered by C1. From there it is split for the +12V, -12V, and +5V rails. The +12V rail consists of a
voltage doubler (C5, C6, and D2, D4), a 7812 regulator (VR2), and filter capacitors (C10, C21). The
-12V rail is a "mirror" of the +12V rail, consisting of voltage doubler (C4, C7, and D3, D5), a 7912
regulator (VR3), and filter capacitors (C11, and C22). The +5V rail consists of a rectifier diode (D1),
filter capacitors (C2, C3, C9), a 7805 regulator (VR1), and a multitude of 0.1uF bypass capacitors.

3.0 Analog Signal Paths

The inputs (stereo) from J1 and J2 pass through the A.C. coupling capacitors (C35, C36) and
have their impedances fixed at 1M by R5 and R7. While operating the unit monauraly (right input
only), the input impedance is fixed at 500K (R5, and R7, in parallel). From there, the inputs are
buffered by U16, and passed through the input potentiometers. The stereo signal is then sent to a
low pass filter/buffer (U16 etc.) and on to the dry side of the mix potentiometer, as well as summed
to mono (Via R12, R13, and defeat jack J5).
The summed stereo signal is sent to the anti-aliasing filter consisting of most of U15, and
associated resistors and capacitors. There are several important features in the filter to be aware of.
The first is the use of the LSTMSB (See section 4.1 for a description) signal from the ASIC. This
signal is injected into the signal path at U15 pin 5. A signal diode/capacitor combination (D8, C32)
at U10 pin 12 shunts any incoming signal exceeding 5V to the power supply of U9, preventing
damage to the analog switch.
The input sample and hold circuit consists of 1/3 of the 4053 analog switch (U9B), the input
sample capacitor (C20), a buffer amplifier (U10), and a comparator (U8).
The signal beyond this point is purely digital, until the DAC output cycle of the DASP 16. At
the appropriate time, the DAC will output the processed left and right signals. This action is
coordinated with the two output sample and hold circuits (U9 A&C, 2 op amps of U10, C18, C19),
so that each receives the correct, separate signal for stereo output. After passing through low pass
(anti aliasing) filters and buffering (2 op amps of U15, Misc. Resistors & Capacitors), the signals are
sent through the wet side of the mix potentiometer. From here, they pass through R13, R14 to the
output potentiometer, and finally on to the output jacks (J3, J4).
MicroVerb/MicroVerb II Service Manual 1.00 1 3/7/2003