T+A DVD 820 M, Руководство пользователя

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Technical description
Digital filters / Oversampling
The audio data on CDs and DVDs is stored at a sampling
rate of 44.1 or 48 kHz - i.e. for each second of music
44.100 or 48.000 sampled values are available for each
channel. In the
DVD 820 M the audio data read from the
DVD or CD is „multiplied“ to a higher sampling rate
(384 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 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 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
DVD 820 M 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
(filters 3 and 5) in the
DVD 820 M in addition to the pure
Bezier process (filter 4), together with two variants of the
standard process (filter 1 and filter 2). For more
information on the five 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.
Filter 1 ( 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 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
Filter 2 (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
Filter 3 ( 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