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Bilinear transform

The bilinear transform, also known as Tustin's method, is a mathematical mapping technique in digital signal processing (DSP) and discrete-time control theory that converts continuous-time systems represented in the Laplace domain (s-plane) to discrete-time systems in the z-domain.^[1]^[2] Introduced by Arnold Tustin in 1947 for analyzing linear systems via time series, it employs the substitution s = \frac{2}{T} \frac{1 - z^{-1}}{1 + z^{-1}}, where T is the sampling period, to derive discrete equivalents while mapping the imaginary axis in the s-plane (continuous frequencies) to the unit circle in the z-plane (discrete frequencies).^[1]^[3] This transform is particularly valued in IIR filter design, where it enables the direct conversion of analog prototypes—such as Butterworth, Chebyshev, or elliptic filters—into stable digital filters without aliasing, as the entire frequency axis is compressed into the range from 0 to the Nyquist frequency.^[2]^[4] Key properties include stability preservation, since the left-half s-plane maps to the interior of the unit circle in the z-plane, and a one-to-one correspondence between poles and zeros, ensuring no information loss from partial fraction expansions.^[3]^[4] However, it induces nonlinear frequency warping, described by \Omega = \frac{2}{T} \tan\left(\frac{\omega T}{2}\right), where \Omega is the analog frequency and \omega is the digital frequency, which distorts the response unless mitigated by prewarping specific band-edge frequencies before design.^[2]^[4] Beyond filter design, the bilinear transform finds applications in control systems for discretizing controllers and in audio signal processing for frequency-domain mappings, such as Bark-scale warping, due to its conformal nature that maps lines and circles to lines and circles while fixing DC (s=0 to z=1) and infinite frequency (to z=-1).^[3]^[5] Its advantages over methods like impulse invariance include avoiding aliasing and simplifying implementation in tools like MATLAB, though the warping limitation makes it most suitable for bandlimited signals or lowpass/highpass filters with narrow transition bands.^[2]^[4]

Introduction

Definition and Purpose

The bilinear transform is a mathematical mapping technique employed in digital signal processing to convert continuous-time systems represented in the s-plane of the Laplace domain to equivalent discrete-time systems in the z-plane of the z-transform domain.^[3] This substitution enables the discretization of linear time-invariant (LTI) systems while approximating their dynamic behavior.^[2] The primary purpose of the bilinear transform is to design infinite impulse response (IIR) digital filters by starting from well-established analog prototypes, such as Butterworth or Chebyshev filters, and transforming them into digital implementations that retain core performance attributes.^[6] By avoiding the need to derive digital filters from scratch, it streamlines the development process in applications requiring precise frequency-selective processing.^[3] Beyond filter design, the bilinear transform is widely applied in discrete-time control systems to adapt continuous-time controllers for implementation on digital hardware, ensuring compatibility with sampled-data environments.^[7] The transformation explicitly incorporates the sampling period T as a scaling parameter, which relates the continuous-time frequency response to the discrete-time domain.^[2] This method is fundamentally based on the trapezoidal rule for numerical integration.^[7]

Historical Development

The bilinear transform, also known as Tustin's method, was first introduced by British engineer Arnold Tustin in the mid-1940s as a technique for analyzing linear systems using time series data in control applications. Tustin developed it during World War II to model the behavior of continuous-time systems through discrete approximations, particularly for servo mechanisms and feedback control in time-domain simulations. His seminal work appeared in the 1947 paper "A Method of Analysing the Behaviour of Linear Systems in Terms of Time Series," published in the Journal of the Institution of Electrical Engineers, where he derived the transformation from the trapezoidal rule of numerical integration to approximate differential equations with difference equations.^[8] Parallel to Tustin's contributions, German engineers Rudolf Oldenbourg and Hans Sartorius independently advanced similar ideas using difference equations to model dynamic systems during the same wartime period. Their work focused on sampled-data control and the discretization of continuous processes for automatic regulation, laying groundwork for digital approximations in industrial control. This was detailed in their 1948 book The Dynamics of Automatic Controls, which emphasized practical applications in mechanical and electrical engineering contexts. Following the war, the method gained traction in control theory during the 1950s for numerical simulations and early digital control systems, such as those implemented in industrial processes. With the proliferation of digital computers in the 1960s, it transitioned into digital signal processing (DSP), where it became a cornerstone for designing infinite impulse response (IIR) filters by mapping analog prototypes to discrete-time equivalents. By the 1970s, its role in early numerical integration evolved into a standard tool in DSP, as evidenced by its prominent inclusion in foundational texts like Oppenheim and Schafer's Digital Signal Processing (1975), which popularized it for filter design amid the field's rapid growth.^[9]

Mathematical Derivation

Trapezoidal Rule Approximation

The bilinear transform arises from the discretization of continuous-time systems, particularly through the approximation of the integrator, which is fundamental to linear time-invariant (LTI) systems described by differential equations.^[10] In continuous time, an integrator computes the output y(t) as the integral of the input u(t), i.e., y(t) = \int u(t) \, dt. To discretize this for digital implementation with sampling period T, numerical integration methods approximate the integral over each interval [nT, (n+1)T]. The trapezoidal rule provides a second-order accurate approximation by assuming a linear variation of the integrand between samples, using the average of the input values at the endpoints of the interval.^[4] The step-by-step derivation begins with the integral over one sampling interval:
y((n+1)T) = y(nT) + \int_{nT}^{(n+1)T} u(\tau) \, d\tau.
Applying the trapezoidal rule, the integral is approximated as the area of a trapezoid:
\int_{nT}^{(n+1)T} u(\tau) \, d\tau \approx \frac{T}{2} \left( u(nT) + u((n+1)T) \right).
Thus, the difference equation for the discrete integrator becomes
y(n+1) = y(n) + \frac{T}{2} \left( u(n) + u(n+1) \right),
where y(n) = y(nT) and u(n) = u(nT). This implicit relation solves for the output at the next step using both current and future input values, making it suitable for offline computation or when combined with other equations in a system.^[10]^[11] To connect this to the s-domain, consider the continuous-time integrator in the Laplace domain: Y(s) = \frac{1}{s} U(s), where s represents differentiation. The trapezoidal approximation implies a mapping from the continuous derivative to a discrete difference. Taking the z-transform of the difference equation yields
Y(z) = \frac{T}{2} \frac{1 + z^{-1}}{1 - z^{-1}} U(z),
which rearranges to approximate the continuous relation as
s Y(s) \approx \frac{2}{T} \frac{1 - z^{-1}}{1 + z^{-1}} Y(z).
This substitution, known as the bilinear mapping, effectively replaces s with \frac{2}{T} \frac{1 - z^{-1}}{1 + z^{-1}} in the transfer function of the continuous system.^[4]^[11] The trapezoidal rule is preferred over simpler methods like backward Euler (s \approx \frac{1 - z^{-1}}{T}) or forward Euler (s \approx \frac{1 - z^{-1}}{T z^{-1}}) because it provides a more accurate frequency response, particularly at higher frequencies. While Euler methods introduce significant distortion and potential instability for certain mappings, the trapezoidal approximation symmetrically averages the endpoints, yielding a mapping that better preserves the phase and gain characteristics up to the Nyquist frequency, albeit with nonlinear frequency warping.^[10]^[11]

Transformation Formula

The bilinear transform maps the continuous-time Laplace variable s to the discrete-time z-transform variable z via the substitution

s = \frac{2}{T} \frac{1 - z^{-1}}{1 + z^{-1}},

where T denotes the sampling interval of the discrete-time system.^[4] This expression originates from the trapezoidal rule for numerical integration.^[2] To derive the discrete-time transfer function H(z) from its continuous-time counterpart H(s), the substitution is applied directly, yielding the equivalent form

H(z) = H\left( s \bigg|_{s = \frac{2}{T} \frac{z-1}{z+1}} \right).

The mapping \frac{z-1}{z+1} is algebraically identical to \frac{1 - z^{-1}}{1 + z^{-1}}, facilitating computation in either forward or backward shift operator notations.^[5] Geometrically, the bilinear transform constitutes a conformal mapping—a special case of a Möbius transformation—that wraps the entire left half of the s-plane onto the interior of the unit disk in the z-plane, ensuring stability preservation under the mapping. The imaginary axis (s = j\Omega) in the s-plane corresponds precisely to the unit circle (|z| = 1) in the z-plane, providing a one-to-one frequency correspondence without aliasing.^[12] The choice of T influences the frequency scaling; in practice, it is often normalized to T = 1 for simplified filter design when the sampling rate is not critical, or set to T = 2 to align the frequency warping expression with \Omega = \tan(\omega/2), aiding analytical computations.^[13]

Key Properties

Stability Preservation

The bilinear transform preserves the stability of causal continuous-time systems when converting them to discrete-time equivalents. Specifically, it maps the open left half of the s-plane, where Re(s) < 0 corresponds to stable continuous-time poles, to the interior of the unit disk in the z-plane, where |z| < 1 denotes stable discrete-time poles.^[14] The imaginary axis in the s-plane, which forms the stability boundary for continuous-time systems, is mapped to the unit circle |z| = 1 in the z-plane.^[14] This stability-preserving property arises from the form of the transformation, given by s = \frac{2}{T} \frac{1 - z^{-1}}{1 + z^{-1}}, where T is the sampling period. To outline the proof, express z in polar form as z = r e^{j \omega} with $0 \leq r < \infty and -\pi < \omega \leq \pi. Substituting yields the real part of s as \sigma = \frac{2}{T} \frac{r^2 - 1}{r^2 + 1 + 2 r \cos \omega}. The denominator is always positive, so the sign of σ depends on the numerator: when r < 1, r^2 - 1 < 0, hence σ < 0; when r = 1, σ = 0; and when r > 1, σ > 0. Thus, the entire left half-plane Re(s) < 0 maps precisely to |z| < 1.^[14] The mapping also preserves causality by being one-to-one and onto between the stable regions of the s-plane and z-plane, ensuring that stable, causal continuous-time systems yield stable, causal discrete-time systems without introducing extraneous poles or zeros that could violate causality.^[14] In contrast, methods like the forward Euler approximation, s = \frac{1 - z^{-1}}{T}, do not unconditionally preserve stability, as they map the left half s-plane to the unit disk only for low frequencies; high-frequency stable poles may be mapped outside |z| = 1, potentially destabilizing the discrete-time system.^[15]

Minimum-Phase Preservation

Minimum-phase systems in the continuous-time domain are defined as those where all poles and zeros lie within the stable region, specifically the open left half of the s-plane where the real part of s is negative.^[16] The bilinear transform preserves the minimum-phase property by conformally mapping the entire left half of the s-plane onto the interior of the unit disk in the z-plane. Zeros located in the left half s-plane are thus transformed to positions inside the unit circle in the z-plane, ensuring that the resulting discrete-time system remains minimum-phase without introducing any zeros outside the unit disk or in unstable regions.^[17] This preservation implies that analog prototypes designed as minimum-phase filters, such as Butterworth or Chebyshev low-pass filters, produce digital infinite impulse response (IIR) filters via the bilinear transform that retain their minimum-phase characteristics, thereby upholding the minimal phase lag for a given magnitude response and facilitating efficient causal realizations.^[17] Unlike some approximation methods, the bilinear transform does not inherently introduce all-pass factors into the transfer function, avoiding additional phase alterations that could compromise the minimum-phase nature of the system.^[17]

Frequency Response Mapping

The bilinear transform maps the imaginary axis in the s-plane (jΩ, where Ω is the analog angular frequency in radians per second) to the unit circle in the z-plane (z = e^{jω}, where ω is the digital angular frequency in radians per sample). This mapping ensures a one-to-one correspondence without aliasing, as the entire infinite analog frequency range is compressed onto the finite digital frequency range from -π to π.^[18] The specific frequency warping relation is given by \omega = 2 \arctan\left( \frac{\Omega T}{2} \right), where T is the sampling period in seconds per sample. Equivalently, the analog frequency can be expressed as \Omega = \frac{2}{T} \tan\left( \frac{\omega}{2} \right). As Ω approaches infinity, ω approaches π (the Nyquist frequency), demonstrating the nonlinear compression of high frequencies into the upper end of the digital spectrum.^[18]^[4] For low frequencies, where \frac{\Omega T}{2} \ll 1, the approximation \tan(x) \approx x holds, yielding \omega \approx \Omega T, which provides a linear mapping and faithful reproduction of the analog response. However, at higher frequencies, the tangent function causes increasing compression, such that a large range of analog frequencies is squeezed near ω = π, potentially distorting the filter's frequency response by altering the relative spacing of features like passband and stopband edges.^[18]^[4] This warping effect is often illustrated graphically by plotting the normalized digital frequency ω/π against the normalized analog frequency ΩT/ (2π), revealing a curve that starts linearly near the origin but asymptotically approaches 1 as the analog frequency increases, highlighting the progressive compression. Such visualizations underscore the transform's suitability for bandlimited signals but emphasize the need for careful frequency scaling in design to mitigate distortions.^[4]

Applications to Filter Design

General LTI System Transformation

The bilinear transform provides a method to convert a continuous-time linear time-invariant (LTI) system into an equivalent discrete-time LTI system while preserving key properties such as stability.^[4]^[19] The general procedure begins with obtaining the analog transfer function H(s) of the continuous-time system, which describes the input-output relationship in the s-domain.^[20] Next, substitute the bilinear mapping s = \frac{2}{T} \frac{z-1}{z+1} (or equivalently s = \frac{2}{T} \frac{1 - z^{-1}}{1 + z^{-1}} in terms of the delay operator) into H(s) to yield the discrete-time transfer function H(z).^[4]^[19] Simplify the resulting rational expression in z algebraically to express H(z) in a standard form, such as a ratio of polynomials in z^{-1} for direct form implementation.^[20] Finally, derive the corresponding difference equation from H(z) by performing polynomial long division or inverse z-transform, enabling digital implementation via recursive computation.^[4] Direct substitution of the bilinear mapping into H(s) preserves the order of the system, as the transformation is a rational function that maps poles and zeros from the s-plane to the z-plane without introducing additional singularities.^[19] However, for practical implementation, especially in higher-order systems, partial fraction expansion of H(z) may be necessary to decompose it into simpler cascaded or parallel second-order sections, facilitating fixed-point arithmetic and reducing quantization errors in digital hardware.^[4] This handling ensures that the discrete-time system accurately emulates the continuous-time dynamics within the primary frequency band of interest.^[20] The choice of sampling period T is critical, as it scales the frequency warping inherent in the transformation; T should be selected small relative to the system's bandwidth to ensure the warped frequency response closely approximates the original analog response.^[19]^[20] For systems represented in state-space form, the bilinear transform can be applied directly to the continuous-time matrices. Consider the state-space model \dot{x}(t) = A x(t) + B u(t), y(t) = C x(t) + D u(t); the discrete-time equivalent is obtained by substituting the mapping into the state equations, yielding updated matrices such as A_d = (I - \frac{T}{2} A)^{-1} (I + \frac{T}{2} A), B_d = (I - \frac{T}{2} A)^{-1} T B, and similarly for C_d and D_d, assuming a specific normalization for the input scaling. This matrix-based approach is particularly useful for multi-input multi-output (MIMO) LTI systems or when simulating differential equations numerically.^[20]

First-Order Filter Transformation

The bilinear transform provides a method to convert a general first-order continuous-time low-pass filter, expressed as H(s) = \frac{a}{s + b} where a > 0 and b > 0, into a discrete-time equivalent that preserves key properties such as stability and DC gain.^[7]^[21] This transformation substitutes s = \frac{2}{T} \frac{1 - z^{-1}}{1 + z^{-1}}, where T is the sampling period, directly into the analog transfer function to obtain H(z). To derive the discrete transfer function, begin with the substitution into H(s):

H(z) = \frac{a}{\frac{2}{T} \frac{1 - z^{-1}}{1 + z^{-1}} + b}.

Multiplying the numerator and denominator by $1 + z^{-1} clears the inner fraction, yielding

H(z) = \frac{a (1 + z^{-1})}{\frac{2}{T} (1 - z^{-1}) + b (1 + z^{-1})} = \frac{a (1 + z^{-1}) }{ \left( \frac{2}{T} + b \right) + \left( b - \frac{2}{T} \right) z^{-1} }.

Normalizing the denominator by dividing the numerator and denominator by \frac{2}{T} + b gives the standard form

H(z) = \frac{ \frac{a T / 2}{1 + b T / 2} (1 + z^{-1}) }{ 1 + \frac{b T / 2 - 1}{b T / 2 + 1} z^{-1} },

which aligns with common implementations where the leading denominator coefficient is unity.^[7]^[21] The corresponding difference equation is obtained by cross-multiplying in the z-domain and taking the inverse z-transform:

y + \frac{b T / 2 - 1}{b T / 2 + 1} y[n-1] = \frac{a T / 2}{1 + b T / 2} \left( x + x[n-1] \right).

Rearranging for y,

y = \frac{a T / 2}{1 + b T / 2} \left( x + x[n-1] \right) - \frac{b T / 2 - 1}{b T / 2 + 1} y[n-1],

where the coefficients are \alpha = \frac{a T / 2}{1 + b T / 2} for the input terms and \beta = -\frac{b T / 2 - 1}{b T / 2 + 1} = \frac{1 - b T / 2}{1 + b T / 2} for the output feedback. This form ensures the filter is implementable as a first-order infinite impulse response (IIR) structure.^[7]^[21] The transformation preserves the DC gain, as evaluating H(z) at z = 1 yields H(1) = \frac{a}{b}, matching the analog filter's low-frequency response at s = 0.^[7] Additionally, the analog pole at s = -b maps to a discrete pole at z = \frac{1 - b T / 2}{1 + b T / 2}, which lies inside the unit circle for b > 0 and finite T > 0, thereby preserving stability. This pole shift approximates the exponential decay of the continuous-time response for sufficiently small T relative to $1/b.^[7]^[21]

Second-Order Biquad Transformation

The bilinear transform applied to second-order continuous-time filters yields digital biquad structures, which serve as essential components for realizing complex infinite impulse response (IIR) filters in discrete-time systems. A general second-order analog transfer function takes the form

H(s) = \frac{a_2 s^2 + a_1 s + a_0}{s^2 + b_1 s + b_0},

where the numerator coefficients a_2, a_1, a_0 define the zeros and the denominator coefficients b_1, b_0 (with leading coefficient normalized to 1) characterize the poles, influencing properties such as resonance and damping.^[2] This quadratic form is common in analog filter prototypes like Butterworth, Chebyshev, or elliptic designs, where complex conjugate poles enable selective frequency responses.^[21] Applying the bilinear substitution s = \frac{2}{T} \frac{1 - z^{-1}}{1 + z^{-1}}, where T is the sampling period, transforms H(s) into a discrete-time counterpart H(z). The resulting expression simplifies to the biquad form

H(z) = \frac{\sum_{k=0}^2 c_k z^{-k}}{1 + \sum_{k=1}^2 d_k z^{-k}},

with the coefficients c_k and d_k derived through polynomial division and clearing the denominator after substitution; these coefficients implicitly incorporate frequency scaling factors like \tan(\omega T / 2) to align the digital response with the analog prototype up to the Nyquist frequency.^[22] This process ensures the digital filter retains a second-order structure, mapping the analog poles to locations inside the unit circle in the z-plane while preserving the relative pole-zero configuration.^[2] The biquad H(z) is typically implemented using the Direct Form II realization, which employs two delay elements shared between numerator and denominator paths for computational efficiency. This structure preserves the analog filter's resonance (determined by pole magnitude and angle) and damping (via pole real part mapping), allowing the digital version to mimic peaking or notch behaviors in applications such as audio equalization or control systems.^[23] One key advantage is the retention of quadratic factors, which facilitates modular digital implementation: biquads can be cascaded with minimal coefficient quantization sensitivity, enabling higher-order filters without excessive numerical instability on fixed-point processors.^[2]

Practical Considerations

Frequency Warping

The bilinear transform introduces a nonlinear distortion known as frequency warping, which alters the frequency response of the digitized filter compared to the original analog prototype. This effect arises because the transform maps the infinite extent of the analog frequency axis onto the finite digital frequency range spanning 0 to the Nyquist frequency (half the sampling rate).^[24] High-frequency compression is the primary consequence, where frequencies far beyond the Nyquist limit in the analog domain are "squeezed" into the upper end of the digital spectrum, leading to a disproportionate scaling that accelerates as frequencies increase. For instance, an analog filter with a response extending to infinite frequency—such as an ideal high-pass filter—maps to a digital version that passes signals up to the Nyquist frequency, but with the transition band and higher-order behaviors severely compressed and shifted toward the Nyquist edge. This distortion significantly impacts the design of high-pass and band-pass filters, where passbands or transitions located at elevated analog frequencies result in digital responses that appear broader or less selective than intended, potentially compromising selectivity in applications like audio equalization or communications.^[25]^[26] In contrast, low-pass filters with cutoffs confined to the baseband experience minimal warping, as the nonlinearity is negligible at lower frequencies, preserving much of the original response shape for baseband signal processing tasks. However, band-stop (notch) filters and differentiators, which depend heavily on accurate high-frequency rejection or amplification, exhibit more severe alterations; the warped response can widen stopbands or attenuate high-frequency details unexpectedly, making these filter types particularly challenging to implement faithfully in the digital domain. A quantitative illustration of this effect occurs in low-pass filter design: for an analog cutoff frequency of ω_c = 100 rad/s and a sampling period T = 0.01 s (Nyquist frequency ≈ 50 Hz or 314 rad/s), the corresponding digital cutoff warps to approximately 92.7 rad/s (about 14.8 Hz), rather than a linearly approximated 100 rad/s (about 15.9 Hz), demonstrating how the effective passband shrinks nonlinearly and necessitates careful specification adjustment to achieve the desired digital performance. This warping represents a fundamental trade-off in the bilinear method: while an ideal linear frequency mapping would scale frequencies proportionally without distortion, such an approach fails to preserve stability by potentially mapping stable analog poles outside the unit circle in the z-plane, whereas the bilinear transform guarantees stability preservation for all stable analog systems at the expense of frequency accuracy.^[24]^[25]

Prewarping Techniques

Prewarping is a technique used to mitigate the nonlinear frequency mapping, or warping, inherent in the bilinear transform, ensuring that critical frequencies in the digital filter design align accurately with their analog counterparts. By adjusting the frequency specifications of the analog prototype filter before applying the transform, prewarping preserves the desired response at specified points, such as cutoff frequencies. This method is particularly valuable in infinite impulse response (IIR) filter design, where maintaining precise frequency selectivity is essential.^[7]^[4] The core of prewarping involves scaling the desired digital frequencies to obtain equivalent analog frequencies for the prototype design. Given a desired digital angular frequency \omega_d (in radians per second) and sampling period T, the prewarped analog frequency \omega_a' is computed as

\omega_a' = \frac{2}{T} \tan\left(\frac{\omega_d T}{2}\right).

^[7]^[21] This formula derives from the inverse of the bilinear frequency mapping relation, ensuring that the warped digital response matches the analog specification at the prewarped point. The design procedure then follows these steps: (1) Specify the critical digital frequencies, such as passband and stopband edges \omega_d; (2) Compute the corresponding prewarped analog frequencies \omega_a' using the formula above; (3) Design the analog prototype filter (e.g., Butterworth or Chebyshev) using these \omega_a' as the specifications; (4) Apply the bilinear transform s = \frac{2}{T} \frac{1 - z^{-1}}{1 + z^{-1}} to obtain the digital filter transfer function H(z). This process guarantees stability preservation and eliminates aliasing while aligning key response features.^[7]^[4]^[21] In the context of a low-pass filter, prewarping simplifies the bilinear constant adjustment. For a prototype with cutoff \omega_c, the standard constant \frac{2}{T} is replaced by k = \frac{2}{T} \tan\left(\frac{\omega_c T}{2}\right), yielding the substitution s = k \frac{1 - z^{-1}}{1 + z^{-1}}. This ensures the digital filter's cutoff aligns exactly with \omega_c, as demonstrated in designs where the analog prototype is scaled accordingly before transformation. For instance, in a Butterworth low-pass filter, prewarping the cutoff frequency maintains the -3 dB point precisely in the digital domain.^[7]^[4] Despite its effectiveness, prewarping has limitations, primarily that it only corrects the frequency response at the specified critical points, such as a single cutoff, leading to residual distortion elsewhere due to the nonlinear warping. For multi-band filters like bandpass or bandstop, where multiple edges must be matched, prewarping the individual frequencies may not fully preserve the analog shape between bands, often necessitating iterative refinement or approximation techniques to optimize the overall response. This shape distortion arises from the inherent frequency compression in the bilinear mapping, which prewarping cannot eliminate globally.^[21]^[7]

References

[1]
A method of analysing the behaviour of linear systems in terms of ...
A method of analysing the behaviour of linear systems in terms of time series. Author: A. TustinAuthors Info & Affiliations. Publication: Journal of the ...
[2]
Bilinear transformation method for analog-to-digital filter conversion
Bilinear transformation is a method for converting analog filters to discrete equivalents by mapping the s-plane to the z-plane.
[3]
Bilinear Transformation - Stanford CCRMA
The bilinear transform is often used to design digital filters from analog prototype filters [323]. An on-line introduction is given in [426].
[4]
[PDF] Digital Signal Processing IIR Filter Design via Bilinear Transform
Bilinear transform converts a CT filter Hc(s) to a DT filter H(z) by substituting s = 2 Td (1 - z-1 / 1 + z-1), using the formula H(z) = Hc ( 2 Td (1 - z-1 1 + ...
[5]
The Bilinear Transform | Spectral Audio Signal Processing
Bilinear transformations map circles and lines into circles and lines (lines being viewed as circles passing through the point at infinity).
[6]
Bilinear Transformation | Physical Audio Signal Processing
The bilinear transform is often used to design digital filters from analog prototype filters [343]. An on-line introduction is given in [449]. Finite Difference ...
[7]
[PDF] 2.161 Signal Processing: Continuous and Discrete
The bilinear transform method uses the truncated series approximation. 1. 2. z ... In particular, when A = 2/T the method is known as. Tustin's method. With ...<|control11|><|separator|>
[8]
https://doi.org/10.1049/ji-2a.1947.0020
[9]
[PDF] 4. A History of Automatic Control
Tustin in the UK developed the bilinear transformation for time series models, while. Oldenbourg and Sartorius also used difference equa- tions to model such ...
[10]
Trapezoidal Rule for Numerical Integration - Stanford CCRMA
The trapezoidal rule approximates a one-sample integral by the area under a trapezoid, using a first-order approximation of the time derivative.
[11]
[PDF] Lecture 6: Discrete Equivalents - Creating Web Pages in your Account
The first method is to use numerical integration to approximate a system described by a set of linear differential equations. The second method uses the ...
[12]
The Bilinear Transform - Stanford CCRMA
Bilinear transformations map circles and lines into circles and lines (lines being viewed as circles passing through the point at infinity).
[13]
How To Digitize an Analog Filter with the Bilinear Transform
Jan 15, 2022 · The bilinear transform maps the analog filter's transfer function from the s-plane to the z-plane, mapping the frequency axis to the unit ...
[14]
[PDF] INTRODUCTION TO DIGITAL SIGNAL PROCESSING
Hence the mapping maintains stability. ... Design a single-pole Butterworth lowpass digital filter with 3dB bandwidth of 0.2π, using the bilinear transform.
[15]
[PDF] Approximate C(s) with a discrete-time controller, C(z).
Controller stability: If C(s) is stable, is C(z) stable? • Closed-loop ... Tustin/bilinear transform. The line Ω = ωT is the equivalent sampled ...Missing: proof | Show results with:proof
[16]
[PDF] All-Pass Systems - MIT OpenCourseWare
1. Page 2. Minimum-Phase Systems. The basic definition of a minimum-phase system is as follows: Stable & causal and has a stable & causal inverse.
[17]
[PDF] Digital Signal Processing Impulse Invariance vs. Bilinear Transform
It can be shown that, in general, the bilinear transform preserves minimum phase. Suppose we have an all-pass CT filter and we wish to convert this filter to ...
[18]
Lecture 15: Design of IIR Digital Filters, Part 2 - MIT OpenCourseWare
Digital Signal Processing. Menu. More Info. Introduction · Readings · Study ... Topics covered: Digital filter design using the bilinear transformation, frequency ...
[19]
[PDF] Digital Signal Processing Lecture 8 - Filter Design - IIR - UTK-EECS
Stable analog filter mapped to stable digital filter? Important properties of bilinear transformation. Left-side of the s-plane → interior of the unit circle;.Missing: procedure | Show results with:procedure
[20]
[PDF] Two Approaches to Physical Modeling - Stanford CCRMA
Jun 27, 2020 · (“Forward Euler”)in place of a backward difference. 16. Page 9 ... • Bilinear Transform, or Trapezoid Rule (TR) fn = 2. T. (pn − pn−1) ...
[21]
[PDF] Lecture 8 - IIR Filters (II) - Colorado State University
Note that the bilinear transform maps the entire left-hand s-plane to the interior of the unit circle of the z-plane, and that higher frequencies along the ...
[22]
Bilinear Transform - Pieter's Pages
Aug 23, 2024 · The bilinear transform is a transformation from continuous-time systems (in the Laplace domain) to discrete-time systems (in the Z-domain).
[23]
The bilinear z transform - EarLevel Engineering
Mar 2, 2003 · The bilinear transform is the most popular method of converting analog filter prototypes in the s domain to the z domain so we can implement them as digital ...
[24]
Frequency Warping | Introduction to Digital Filters
### Summary of Frequency Warping from "Introduction to Digital Filters" by Julius O. Smith III
[25]
Frequency Warping - Stanford CCRMA
Frequency Warping. It is easy to check that the bilinear transform gives a one-to-one, order-preserving, conformal map [57] between the analog frequency axis ...Missing: "julius | Show results with:"julius
[26]
Digitizing Analog Filters with the Bilinear Transformation
Some kind of frequency warping is obviously unavoidable in any one-to-one map because the analog frequency axis is infinite while the digital frequency axis is ...