Filter devices and methods

Details Filter devices and methods Filter devices and methods

2001-11-30

2004-09-14

Mai, Tan V. (Department: 2124)

Electrical computers: arithmetic processing and calculating

Electrical digital calculating computer

Particular function performed

C708S300000

Reexamination Certificate

active

06792444

ABSTRACT:

The invention relates to a filtering method, including: generating a primary filtered signal of an input signal; rounding the primary filtered signal to a primary rounded signal; and combining the primary rounded signal with said input signal for calculating a primary filter error or coded signal.
From general filter theory, it is known that it may be advantageous to replace a primary filter by a cascade of secondary filters performing parts of the function of the primary filter. However, this cascading is not necessarily advantageous.
For example, in predictive filtering this imposes several problems. A method for such prediction filtering is known from: A. A. M. L. Bruekers, A. W. J. Oomen, R. J. v.d. Vleuten and L. M. van de Kerkhof, “
Lossless coding for DVD audio
”, AES 101
st
convention, Los Angeles (Preprint 4358 November 1996) and from: A. A. M. L. Bruekers, A. W. J. Oomen, R. J. v.d. Vleuten and L. M. van de Kerkhof, “
Improved lossless coding of
1-
bit audio signals
”, AES 103
rd
convention, New York (Preprint 4563 September 1997).
In the method described in the first ‘Bruekers’ article, a signal is encoded by prediction filtering. The prediction filtering described in this prior art document includes the steps of: generating a primary prediction signal of an input signal, based on preceding input signals; rounding said primary prediction signal to a primary rounded signal; and subtracting said primary rounded signal from said input signal for calculating a primary filter error or coded signal. In the method described in the second ‘Bruekers’ article the input signal is a 1-bit signal. Simply applying a cascade of secondary filters imposes several problems in these methods. Firstly, the least significant digit (lsd) of the prediction error or coded signal has to be at least as significant as the lsd of the input signal in order to prevent the coded signal to have less significants lsd's. Therefor, the secondary prediction signal has to be rounded to a certain lsd. However, this rounding causes a distortion of the prediction error or coded signal, whereby subsequent prediction of the distorted prediction error or coded signal by a next secondary filter is disturbed.
Secondly, the prediction signal itself is needed for further coding processes like Direct Stream Transfer (DST) as is used in the Super Audio CD (SACD) standard. Thirdly, if a recursive filter structure or Infinite Impulse Response (IIR) filter, as is commonly known in filtering, is used, the rounding step may cause limit cycles, i.e. if there is no input signal, there should be no prediction signal but the rounding step may result in a non zero rounded signal and therefore in a prediction error. Although identical limit cycles in the decoder guarantee that there is no output signal or decoded signal, the efficiency of the coding is effected negatively. At the decoding side this will result in a decoded signal, though there was no input signal at all at the encoding side.
The invention intends to solve these problems. Therefore, in accordance with the invention a filtering method as indicated above is characterised in that the step of generating the primary filtered signal includes: at least one secondary filtering step including the steps of: generating a secondary filtered signal of a source signal; rounding the secondary filtered signal to a secondary rounded signal; and combining the secondary rounded signal with said source signal for calculating a secondary filter error.
Such a method makes it possible to round each secondary filtered signal to any bit size required by the specific application of the method, because only the primary filtered signal has to be rounded to the required lsd, for that is the signal that is used to calculate the filter error or coded signal. The steps of rounding the secondary filtered signals are reversible steps because the only required rounding that is limited by the further application of the signals is the rounding of the primary filtered signal, and it is possible to reconstruct this primary rounding in the decoding process. Furthermore, the primary filtered signal is obtained, so this signal can be used in further filtering processes. Since the rounding of the secondary filtered signal can chosen to be of any resolution, a rounding step can be applied that reduces the limit cycles.
The invention also provides a reversed filtering method as is claimed in claim
11
. Such a method allows reverse filtering or decoding of an primary filtered signal.
Furthermore, the invention provides a filter device for performing a method according to the invention, as is claimed in claim
12
. The invention also provides a filter device as is claimed in claim
22
.


REFERENCES:
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patent: 6650255 (2003-11-01), Bruekers et al.
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patent: 2002/0161806 (2002-10-01), Shaikh
patent: 1052622 (2000-11-01), None
patent: WO0041313 (2000-07-01), None
Chen et al, “A fixed-point 16 kb/s LD-CELP Algorithm”, Inter. Confer. on Acoustics, Speech, and Signal Processing, IEEE, 1991, pp. 21-24.*
Chen et al, “Improving the performance of the 16 kb/s LD-CELP speech coder”, Inter. Confer. on Acoustics, Speech, and Signal Processing, IEEE, 1992, pp. 69-72.

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