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Direct-Form Filter Parameter Quantization

Direct-Form Filter Parameter Quantization

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Published by Brian Neunaber
The effect of coefficient quantization on audio filter parameters using the direct-form filter implementation is analyzed. An expression for estimating the maximum error in frequency and Q resolution is developed. Due to coefficient quantization, appreciable error in the DC gain of some types of second-order direct-form filters may result. Simple techniques are developed for reducing or eliminating this error without increasing filter complexity or coefficient precision.
The effect of coefficient quantization on audio filter parameters using the direct-form filter implementation is analyzed. An expression for estimating the maximum error in frequency and Q resolution is developed. Due to coefficient quantization, appreciable error in the DC gain of some types of second-order direct-form filters may result. Simple techniques are developed for reducing or eliminating this error without increasing filter complexity or coefficient precision.

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Published by: Brian Neunaber on Oct 14, 2008
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05/09/2014

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Parameter Quantization in Direct-Form Recursive Audio Filters
Brian NeunaberQSC Audio Products1675 MacArthur Blvd.Costa Mesa, CA 92626
Abstract – The effect of coefficient quantization on audio filter parameters using the direct-formfilter implementation is analyzed. An expression for estimating the maximum error in frequency andQ resolution is developed. Due to coefficient quantization, appreciable error in the DC gain of sometypes of second-order direct-form filters may result. Simple techniques are developed for reducingor eliminating this error without increasing filter complexity or coefficient precision.
0 Introduction
The direct-form I (DF1) filter topology is preferred for recursive audio filtering [1], [2], and its efficiencyof implementation is hard to beat. However, one disadvantage of the DF1 topology is its poor coefficientsensitivity [3], [4]. Recent trends to increase sampling rates further degrade coefficient sensitivity. Usinghigher-precision coefficients often comes at an expense, such as increased hardware cost or reducedperformance from double-precision arithmetic. We analyze how coefficient quantization affects filterparameters and introduce the concept of 
 parameter quantization
. We develop methods for minimizingthese effects without increasing filter complexity or coefficient precision.
1 Background
For high quality audio using a fixed-point DF1 implementation, a minimum of 24-bit signal precision with48-bit accumulator precision and some form of error feedback is recommended. With truncation errorcancellation, the DF1 has noise performance sufficient for the most demanding audio applications. TheDF1 filter topology with truncation error cancellation is mathematically equivalent to double-precision inthe signal feedback paths using single-precision coefficients. As a result, truncation error cancellationgreatly improves signal-to-quantization noise of the DF1 but does nothing for coefficient sensitivity. Formore information on error feedback and truncation error cancellation, the reader is referred to [1] and [2].High order recursive filters may be broken down into parallel or cascade first- and second-order sections,and there are good reasons to do so. Cascade implementation of first- and second-order sections has bettercoefficient sensitivity than direct implementation and is easier to analyze [4]. Many audio equalizationfilters are implemented as parametric first- or second-order sections, such as shelving or boost/cut (alsocalled
 peak 
or
 presence
) filters; and graphic equalizers are implemented as either parallel or cascadedsecond-order sections. Therefore, we limit our analysis of coefficient quantization to first- and second-order sections only. Higher order filters may be constructed from these basic structures.
1.1 Recursive Filter Transfer Function
Given the parameters of gain, frequency, and Q (in the second-order case), we first develop the coefficientsfor several types of audio filters.
1.1.1 First-order Case 
The general bilinear transfer function,
 H 
1
(s)
, of a first-order filter is written as
( )
1
 H L
V s  H ss
ω  ω  
+=+
(1)where
 H 
is the high-pass gain (at the Nyquist frequency),
 L
is the low-pass gain (at DC), and
ω
=2
π·
 f 
[5].
 f 
is the cutoff frequency (for high- and low-pass filters) or center frequency (for all-pass and shelf filters).
 
 2To convert Equation (1) to the digital domain, we use the bilinear transform. We make the followingsubstitutions [3]:11tan
s
 zs z f  f 
ω π  
=+
→ Ω =
(2)The sampling rate is
 f 
s
. After the substitutions, we simplify the general bilinear transfer function to thefollowing:
( )( ) ( )( ) ( )
111
11
 L H L
V V V V z H z z
+ + =Ω+ + Ω−
(3)Given the first-order transfer function,
 H 
1
(z)
, in the form
( )
101111
1
a a z H zb z
+=+
(4)The coefficients of 
 H 
1
(z)
become the following:
0
1
 L
V a
+=+
(5)
1
1
 L
V a
=+
(6)
1
11
b
=+
(7)The parameters for common first-order audio filter types are shown in Table 1. The functions
min(x, y)
and
max(x, y)
return the minimum and maximum (respectively) of their arguments.
High-pass Low-pass
 
All-pass High Shelf Low Shelf 
 
 
 
 
( )
min,1
 H 
 1max,1
 L
 
V
L
 
0 1 1 1 V
L
 
V
H
1 0 -1 V
H
1
Table 1. First-order parameters for common audio filters.
1.1.2 Second-order Case 
The general biquadratic transfer function,
 H 
2
(s)
, of a second-order filter is written as
( )
22222
 H B L
V s V s Q H ss sQ
ω  ω  ω  ω  
+ +=+ +
(8)
 
 3where
 H 
is the high-pass gain,
 B
is the band-pass gain (at
 f 
),
 L
is the low-pass gain, and
ω
=2
π·
 f 
[5].
 f 
 is the cutoff frequency (for high- and low-pass filters) or center frequency (for all-pass, shelf and boost/cutfilters). To convert Equation (8) to the digital domain, we again use the bilinear transform. After thesubstitutions, we get the general digital biquadratic transfer function
( )
( )( )
22122222122
21211
 L B H L H L B
V V V V V z V V V zQ Q H z z zQ Q
+ + + + +
=
+ + + + +
(9)We want
 H 
2
(z)
in the form
( )
1201221212
1
a a z a z H zb z b z
+ +=+ +
(10)So, the coefficients become
202
1
 L B
V V QaQ
+ +=+ +
(11)
( )
212
21
 L
V aQ
=+ +
(12)
222
1
 L B
V V QaQ
+=+ +
(13)
( )
212
211
bQ
=+ +
(14)
222
11
QbQ
+=+ +
(15)The parameters for common second-order audio filter types are shown in Table 2. The boost/cut filter(based on [6]) is designed such that its frequency response is symmetrical about unity gain forcomplementary boost and cut gains.

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