User manual

54
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 D10 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 signifi-
cantly 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 dis-
advantage. 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 D10
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 D10 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 

 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 D10.
Frequency response and transient characteristics of
the Bezier interpolator