Hello! Welcome to the final lesson of this module on z-domain analysis.
In our last session, we established a crucial result: a causal, discrete-time LTI system is stable if and only if all the poles of its transfer function, , lie inside the unit circle. We briefly touched upon the mapping as the bridge between the continuous-time s-plane and the discrete-time z-plane, noting how the stable left half of the s-plane maps to the stable interior of the unit circle.
Today, we will formalize this connection by exploring a foundational technique for digital filter design called impulse invariance. Our goal is to derive the relationship between the s-plane poles of an analog system and the z-plane poles of its discrete-time counterpart via this method. This lesson serves as a bridge from the analysis of abstract discrete systems to the practical design of digital filters based on well-established analog prototypes.
1. Why Map from Analog to Digital?
Before we delve into the mathematics, it's worth asking why we would want to convert an analog filter design into a digital one. Why not just design digital filters from scratch? As you appreciate the historical context behind concepts, there are two primary motivations.
Lecture 23, Mapping Continuous-Time Filters to Discrete-Time Filters | MIT RES.6.007
This video from MIT OpenCourseWare provides a concise overview of the motivations for mapping continuous-time filters to discrete-time filters.
Please watch the segment from 19:49 to 23:49. Focus on the two main reasons presented: 1) for simulating continuous-time systems, and 2) for leveraging the rich history and established design procedures of analog filters.
As the video explains, the two key drivers are:
- Leveraging a Rich Heritage: The theory and design of analog filters (like Butterworth, Chebyshev, and Elliptic filters, which we will study later) were developed and refined over many decades. It is often more efficient to adapt these proven designs to the digital domain than to start from a blank slate.
- Simulation: To simulate a continuous-time physical system on a computer, we must create a discrete-time model that accurately represents it. This requires a principled way to convert the system's continuous-time description to a discrete-time one.
2. The Principle of Impulse Invariance
The impulse invariance method is one of the most intuitive approaches to this conversion. The core idea is to create a digital filter whose impulse response is a sampled version of the original analog filter's impulse response.
Specifically, if the analog filter has an impulse response , we want the digital filter's impulse response, , to be "invariant" at the sampling instants.
Here, is the sampling period. By preserving the shape of the impulse response at the sampling points, the hope is to also preserve the filter's characteristics.
3. Deriving the Pole Mapping
Now, let's perform the derivation, which will take us from this simple time-domain relationship to a direct mapping of poles in the complex plane. This process elegantly combines the Laplace and z-transforms.
The derivation follows these steps:
- Start with a general analog transfer function, , and expand it into first-order terms using a partial fraction expansion.
- Take the inverse Laplace transform to find the continuous-time impulse response, .
- Sample at intervals of to get the discrete-time impulse response, .
- Take the z-transform of to find the digital transfer function, .
- Compare the forms of and to see how the poles and residues have been transformed.
The following resource provides a clear walkthrough of this mathematical process.
The webpage 'Impulse Invariant Method' from Stanford's CCRMA provides a very clear and concise mathematical derivation of the method. We will walk through its key steps.
Please read the section 'Derivation'. It starts with the analog transfer function Hₐ(s) and its partial fraction expansion. Follow the steps down to the final digital transfer function Hₓ(z) and the resulting pole mapping zᵢ = e^(sᵢT).
Let's summarize the key steps from that derivation.
We start with the analog transfer function expressed in its partial fraction expansion, assuming it has distinct poles with corresponding residues :
The inverse Laplace transform gives us the causal analog impulse response:
Next, we sample this response at , which is the defining step of impulse invariance:
We can rewrite this as:
Finally, we take the z-transform. Recognizing the form of a geometric series (), we get the digital transfer function :
By comparing the initial and final forms, we arrive at the central conclusion:
- An analog pole at is mapped to a digital pole at .
- The residues are preserved in the partial fraction expansion.
This elegant result forms the heart of the impulse invariance method.
Test your understanding!
A simple first-order analog low-pass filter has the transfer function:
This filter is to be converted to a digital filter using the impulse invariance method with a sampling period of seconds.
- Where is the s-plane pole?
- What is the resulting digital transfer function ?
- Where is the z-plane pole? Is it inside the unit circle?
Show answer
- The s-plane pole is at . Since this is in the left-half plane, the analog filter is stable.
- The digital transfer function will have the form . The residue is 2.
- The z-plane pole is at . Since , the pole is inside the unit circle, and the resulting digital filter is stable, as expected.
4. Implications of the Pole Mapping
The relationship has profound consequences for both stability and the frequency response.
Stability Preservation
As we saw in the previous lesson, a causal analog system is stable if all its poles have a negative real part, . Let's see how this translates to the z-plane.
The magnitude of the corresponding digital pole is:
Since .
If the analog filter is stable, then . Because is always positive, , which means . Therefore, .
This proves a vital property of impulse invariance: it always maps a stable analog filter to a stable digital filter. The stable left half of the s-plane is mapped to the interior of the unit circle in the z-plane, perfectly aligning with the stability criteria we established in the last lesson.

Frequency Response and Aliasing
While the pole mapping is direct and stability is preserved, the effect on the frequency response is more complex. Recall from the sampling theorem that when we sample a signal, its spectrum becomes periodic. The same happens here, but with the filter's frequency response. The relationship is:
where is the normalized discrete-time frequency.
This equation tells us that the digital filter's frequency response is a superposition of scaled and shifted replicas of the analog filter's frequency response. If the analog filter is not perfectly band-limited (i.e., for ), these replicas will overlap, causing aliasing.
Lecture 23, Mapping Continuous-Time Filters to Discrete-Time Filters | MIT RES.6.007
Let's return to the MIT video, which now explains this aliasing effect and shows what it looks like.
Watch from 39:20 to 48:10. The first part (until 44:01) explains the concept of aliasing in the frequency response. The second part provides a concrete example showing how an analog resonant peak is mapped to the digital domain and how aliasing slightly distorts the shape of the frequency response.
This aliasing is the primary drawback of the impulse invariance method. It distorts the frequency response, especially near the Nyquist frequency (). The method works best for analog filters that are already mostly band-limited, such as sharp low-pass or band-pass filters, where the response is negligible outside the main band, minimizing the overlap.
Visualizing the Mapping
To build a deeper intuition for how continuous-time system characteristics translate to the discrete-time domain, it is useful to see how entire regions and lines in the s-plane are mapped. Given your background in control systems, visualizing how things like damping and natural frequency are transformed is particularly insightful.
Digital control 14: Mapping poles from the s-plane to the z-plane
This short video provides an excellent dynamic visualization of the s-plane to z-plane mapping, showing what happens to poles as you move them around the s-plane.
Watch the entire video (5:55). Pay attention to how: The imaginary axis in the s-plane maps to the unit circle. Lines of constant damping in the s-plane become spirals in the z-plane. Semicircles of constant natural frequency in the s-plane map to curves in the z-plane.
Conclusion
In this lesson, we have formally bridged the gap between analog and digital systems by deriving the principles of the impulse invariance method. We saw how a simple and intuitive time-domain requirement—preserving the impulse response at sampling instants—leads to a direct and powerful relationship in the complex frequency domains.
Key Takeaways:
- Impulse Invariance: A technique to design a digital filter by sampling the impulse response of an analog prototype, i.e., .
- Pole Mapping: The core result of this method is that an analog pole at is mapped to a digital pole at . The residues of the partial fraction expansion are preserved.
- Stability Preservation: This pole mapping ensures that a stable analog filter (poles in the left-half s-plane) will always result in a stable digital filter (poles inside the unit circle).
- Limitation: Aliasing: The main drawback is that the frequency response of the digital filter suffers from aliasing unless the analog filter is perfectly band-limited, which is never the case in practice. This limits its use to filters where out-of-band energy is minimal.
Preview of the Next Lesson:
We have now concluded our foundational exploration of discrete-time signals and systems in the z-domain. We've covered the z-transform, its properties, stability analysis, and now a method for filter design. The next major topic in our course is the analysis of signals in the frequency domain. While the DTFT provides the theoretical framework, it is a continuous function of frequency and cannot be computed directly.
In our next module, we will introduce its practical, computable counterpart: the Discrete Fourier Transform (DFT). We will begin by tracing the fascinating history of the algorithm that makes it efficient—the Fast Fourier Transform (FFT)—from its surprising origins with Gauss in the early 19th century to its modern rediscovery by Cooley and Tukey.