T+A SACD 1250 R, Руководство пользователя

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Technical description
OVERSAMPLING
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
SACD 1260 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 significantly better, more finely
graduated signal to the converter, which can then be
converted with correspondingly 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 sampling rate a process
known as a FIR (Finite Impulse Response) filter is
employed for this purpose. At

we have been
carrying out research 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 con-siderably more
complex than the standard method, the SACD 1260 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 im-
plemented two slightly modified Bezier processes
(OVS 2) in the SACD 1260 R in addition to the pure
Bezier process (OVS 3), together with two variants of the
standard process (standard OVS FIR filter and OVS 1).
For more information on the different processes please
refer to the next section. You can switch between the
different algorithms using the FILTER button, and decide
for your-self which of the filters you prefer.
Standard OVS FIR filter
The long FIR filter is the standard oversampling process
in digital technology, offering extremely linear frequency
response, very high damping, linear phase charac-
teristics 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
signal’s dynamics, precision and naturalness, and reduce
spatial orientation.
Frequency response and transient characteristics of
the long FIR filter
OVS 1 (short FIR filter)
Shortening the filter (lower coefficient) reduces the time
range errors, albeit combined with a slight loss of linearity
in the frequency range and damping performance.
Frequency response and transient characteristics of
the short FIR filter
OVS 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
OVS 3 (pure Bezier interpolator)
This process delivers a perfect reconstruction of the
original music signal. It exhibits no pre- or post-echoes of
any kind, and does not add coloration or timing errors to
the original signal. In sonic terms this method offers an
impressive blend of naturalness, good dynamics and
accuracy. This is our preferred process due to its advan-
tages in respect of sound, and is the basic (default)
setting of the SACD 1260 R .
Frequency response and transient characteristics of
the Bezier interpolator