Technical Description Digital Filters / Oversampling - T+A HV Series User Manual

Technical description
Digital filters / Oversampling
Oversampling The audio data on for example CDs is stored at a sampling rate of 44.1 kHz -
i. e. for each second of music 44.100 sampled values are available for each
channel. In the SDV 3100 HV the audio data read from the CD is „multiplied"
to a higher sampling rate 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 considerably more complex than the standard method, the
SDV 3100 HV features a digtal high precision floating point signal processor
(DSP) which carries out the over-sampling process 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 implemented a slightly modified Bezier process (filters 3) in the
SDV 3100 HV 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 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.
FIR long (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 characteristics 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
63