T+A R 1230 R, Руководство пользователя

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Technical description
Digital filters / Oversamplin
The audio data on CDs is stored at a sampling rate of
44.1 - i. e. for each second of music 44.100 sampled
values are available for each channel. In the
CD 1230 R
the audio data read from the CD is „multiplied“ to a
higher sampling rate (352,8 kHz) before it is converted
back into analogue music signals. This process delivers
a very much better, more finely graduated signal to the
converter, which can then be converted with corre-
spondingly higher precision. The raised sampling rate is
a calculating process for which there are many different
mathematical methods. In almost all digital audio devices
which exploit the advantages of increased digital sam-
pling rate a process known as a FIR filter is employed for
this purpose. At
 we have been carrying out re-
search for more than ten years, aimed at improving the
oversampling process, because the standard FIR
method has one drawback to set against its indisputable
advantages: it adds small pre- and post-echoes to the
music signals. At
 we have developed mathematical
processes (known as Bezier polynomial interpolators)
which do not share this disadvantage. For this reason
they should sound better and more natural than the
usual standard process. Since the calculating procedure
employed by us is considerably more complex than the
standard method, the
CD 1230 R features a high-
performance digital signal processor (DSP) which carries
out the over-sampling process with immense precision
(56 bit) using special algorithms developed by
 .
The freely programmable DSP which we use is capable
of carrying out the oversampling process using any
method of calculation. For this reason we have imple-
mented two slightly modified Bezier processes (filters 2
and 4) in the
CD 1230 R in addition to the pure Bezier
process (filter 3), together with two variants of the stan-
dard process (Standard FIR filter and filter 1). For more
information on the five different processes please refer to
the next section. You can switch between the various
algorithms, then decide for yourself which of the filters
gives the results you prefer.
Standard FIR filter
The long FIR filter is the standard oversampling process
in digital technology, offering extremely linear frequency
response, very high damping, linear phase characteris-
tics and constant group delays. The disadvantage is the
pre- and post-echoes which are added to the signal.
These „time range errors“ tend to affect the music sig-
nal’s dynamics, precision and naturalness, and reduce
spatial orientation.
Frequency response and transient characteristics of
the long FIR filter
Filter 1 (short FIR filter)
Shortening the filter (lower coefficient) reduces the time
range errors, albeit combined with a slight loss of linear-
ity in the frequency range and damping performance.
Frequency response and transient characteristics of
the short FIR filter
Filter 2 ( Bezier interpolator plus IIR filter)
In this process an ideal Bezier interpolator is combined
with what is known as an IIR filter. This eliminates the
problematic pre-echo of the FIR method. This process
produces highly „analogue“ system characteristics, with
a sound quality and measured performance similar to
those of good analogue disc players.
Frequency response and transient characteristics of
the Bezier interpolator plus IIR filter