Technical Description Digital Filters / Oversampling - T+A DAC 8 User Manual

Technical description
Digital filters / Oversampling
Oversampling
The audio data on for example 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 DAC 8 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
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 DAC 8 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 implemented a slightly modified Bezier process (filters 3) in the DAC 8
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.
Oversampling 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 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
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