
Dear audio enthusiast,

    Here you have the Muleteer  DSP  —  a  rapidity maintainer module. This
DSP  designed  to  negate  unwanted effects, excessive mass exerts on sound
reproduction.  In  a  first  place,  it's  a  transducer  moving subsystem,
including  diaphragm  and a voice coil. Much of this overweight effectively
fought  by  negative  feedback  of  amplifier's  current  driver.  But this
approach  is  incapable  to  solve  this  problem completely, since perfect
solution  lays through indefinitely short feedback roundtrip and infinitely
fast amplifier slew rate.
    Muleteer DSP  uses  number  of  advantages  to  supply  right amount of
energy  just  at  the  right time,  overcoming critical drawbacks of modern
solutions.  First of all, it can look forward, to enable precise evaluation
required  amount  of action. Second, the mentioned excessive mass stays the
same all the time, making prediction of system response even more accurate.
So, at least mathematically, mentioned problem, gets a perfect solution.
    
The final notes:

    1. Activate "Rapidity Maintainer" from available DSP processors list.

    2. Guess your  system overweight.  The modern  systems have  this value
about  0.5-1.5.   So,  try   to   stay  close with these values. Only happy
owners of valve based systems can go beyond 2.0.

    3. Since Muleteer operates on  a  quite finer level than one considered
"enough"  by  engineers  failed  to  use  Claude Shannon findings (*), it's
highly  recommended  (as  essential condition of accurate work) to upsample
standard  definition  material  to  at  least  a double rate. Simply insert
resampler  to the DSP chain before Muleteer's module. Try to avoid any kind
of  dithering  features,  such as noise shaping and e.t.c. Especially after
the muleteer's work, since it's outcome, strictly, is not PCM anymore.

    4. Finetune module to desired withstand value.

    5. Enjoy.

CYa, copah (aka Vladimir Kopjov)
mailto: copah@yandex.ru

*)  Shannon-Kotelnikov  sampling theorem states: «Bandlimited analog signal
can  be perfectly reconstructed from an infinite sequence of samples if the
sampling rate exceeds 2B samples per second.»
Note: INIFINITE sequence of samples. This has very simple example. Let sine
wave  with  frequency  F at 2F samplerate have [1,-1,1,-1,1,-1,1] sequence.
So,  if  we  sample  this  sine  phase-shifted  by pi()/4 radian, we'll get
[0.7,-0.7,0.7,-0.7,0.7,-0.7,0.7].  But  we'll get same results if we sample
previous  sine  attenuated by 0.7. This evidence demonstrates impossibility
to discriminate amplitude and phase at frequency F on live (finite) signal.
For  Compact  Disc this means unavoidable decoding errors, since nobody nor
want,  nor  able play infinite note. On the other hand if samplerate equals
4F,  sampled  data  can  be  reconstructed  by  as  little  as  two samples
(equivalent  to  the  only  sample  at  2F),  since first derivative can be
calculated  as  per  2F  sample  with  perfect alignment. This way, on a 2F
samplerate, bandwidth 0F to 1/2F is error free. And bandwidth 1/2F to F has
harmonic error rate: E'f(t) = f/Fs*t if done math right… ;-)
