The following question came up in a photography mailing list:
“…could you explain the thinking behind the evils of the antialiasing filter on DSLR’s? Why are they needed and what are their advantages/disadvantages and why don’t 2 1/4 digital backs have them, or do they?”
Since there seems to be general interest in this topic, I thought I’d take a crack at it and post it here.
First, my background as it relates to this question. From 1969 to 1989, I worked with one dimensional sampled data systems at Hewlett Packard, Rolm Corp., and IBM. From 1989 until 1995, I worked with digital photographic systems at IBM. With this history, I have to work to keep the math to a minimum. I will do so. I promise: no equations. I will have to talk about the concept of frequency and spectral content. I’ll do my best to relate what I’m saying to concepts understandable by most people. I will be doing a lot of simplification, and if I stray into oversimplification, either inadvertently or deliberately. I apologize in advance.
Before we talk about antialiasing, we need to understand what aliasing is. For that, we go to AT&T in 1924, where a guy named Harry Nyquist came up with a surprising idea, later generalized by the father of information theory, Claude Shannon.
Even though his ideas had broader applicability, Nyquist worked only with signals that varied with time, and I’m going to consider only those kinds of signals for the rest of this post; in the next one, I’ll get around to the two dimensional spatially sampled signals applicable to photography. Nyquist was most interested in the voice signals that AT&T got paid to transmit from place to place. These days, almost everything you hear has been sampled: CDs, wireline and cellphone telephone calls, satellite and HD radio, Internet audio, and iPod music.
The crux of Nyquist’s 1924 insight was that, given an idealized sampling system (perfect amplitude precision, no sampling jitter, infinitely small sampling window, etc.), if you took regularly spaced samples of the signal at a rate faster than twice the highest frequency in the signal, you had enough information to perfectly reconstruct the original signal. This came to be known as the Nyquist Criterion, and although I’ve worked with it for years, I still find it pretty amazing. It turns out that the Nyquist Criterion’s path from theory to practice has been pretty smooth; real systems, which don’t obey all the idealizing assumptions (some of which are pretty severe: pick any frequency you like, and any signal of finite duration has some frequency content above it) come very close to acting like the ideal case.
What if there is significant signal content at frequencies above half the sampling frequency? Let’s imagine a system which samples the input signal 20,000 times a second. We say the sampling frequency is 20 kilo Hertz, or 20 KHz. Let’s further say that this system is connected to a system to reconstruct the signal from the samples. Now let’s put a single frequency in the input, as see what we get at the output. If we put in 5 KHz, we get the same thing out. We turn the dial up towards 10 KHz, and we still get the same signal out as we put in. As we go above 10 KHz, a strange thing happens: the output frequency begins to drop. When we put 11 KHz in, we get a 9 KHz output. 12 KHz give us an output of 8 KHz. This continues to happen all the way to 20 KHz, where we get 0 Hz (dc) signal. 21 KHz in gives us 1 KHz out, 22 KHz in gives us 2 KHz out, etc.
Engineers, trying to relate this situation to everyday life, said that, at over half the sampling frequency, the input signal appears at the output, but “under an alias”. Thus, since in English there is no noun that cannot be verbed, we get aliased signals and aliasing.
Aliasing is almost always a bad thing. If aliasing is present, it’s impossible to tell whether the signals at the output of the reconstructive device were part of the original signal, or were aliased down in frequency from somewhere else in the spectrum. Therefore, in systems that sample, transmit or store, and reproduce time-varying signals in the real world, such as CDs, the audio part of DVDs, or telephone systems, place a filter in front of the sampler to diminish (attenuate is the engineering word) signal content at frequencies above half the sampling frequency. This filter is called an antialiasing (or AA, or, if you’re an engineer, “A-squared”) filter.
Robert Heinlein was fond of pointing out that “there ain’t no such thing as a free lunch.” The antialiasing filter is a good example. Real filters, the ones made up of resistors, capacitors, inductors, and amplifiers, can’t filter out a set of frequencies without affecting the characteristics of those they’re trying to pass unchanged. When Sony introduced the Compact Disc with the slogan “Perfect Sound Forever,” audiophiles raised a chorus of complaints about the sound quality. While the analog to digital conversion in the recording studio (with its fourteen to sixteen bits of resolution and lower accuracy), the digital to analog conversion in the player, and the sampling jitter (both pre analog to digital and post digital to analog conversion) all shared some of the blame, suspicions were focused early on the antialiasing filters, both the ones before the ADCs and the ones after the DACs. Over time, the analog filters got better, but the biggest improvements had to wait until the advent of oversampling. More on that in the last post in this series, where I will speculate on a possible way out of the current photographic antialiasing morass.
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