- Low-pass filter
A

**low-pass filter**is a filter that passes low-frequency signals butattenuate s (reduces theamplitude of) signals with frequencies higher than thecutoff frequency . The actual amount of attenuation for each frequency varies from filter to filter. It is sometimes called a**high-cut filter**, or**treble cut filter**when used in audio applications.The concept of a low-pass filter exists in many different forms, including electronic circuits (like a "hiss filter" used in audio), digital

algorithm s for smoothing sets of data, acoustic barriers, blurring of images, and so on. Low-pass filters play the same role insignal processing that moving averages do in some other fields, such as finance; both tools provide a smoother form of a signal which removes the short-term oscillations, leaving only the long-term trend.**Examples of low pass filters**Figure 1 shows a low pass

RC filter for voltage signals, discussed in more detail below. Signal "V_{out}" contains frequencies from the input signal, with high frequencies attenuated, but with little attentuation below thecorner frequency of the filter determined by its "RC"time constant . For current signals, a similar circuit using a resistor and capacitor in parallel works the same way. Seecurrent divider .**Acoustic**A stiff physical barrier tends to reflect higher sound frequencies, and so acts as a low-pass filter for transmitting sound. When music is playing in another room, the low notes are easily heard, while the high notes are attenuated.

**Electronic**Electronic low-pass filters are used to drive

subwoofer s and other types ofloudspeaker s, to block high pitches that they can't efficiently broadcast.Radio transmitters use lowpass filters to block

harmonic emissions which might cause interference with other communications.An

integrator is another example of a low-pass filter.DSL splitter s use low-pass and high-pass filters to separate DSL and POTS signals sharing the same pair of wires.Low-pass filters also play a significant role in the sculpting of sound for

electronic music as created by analoguesynthesiser s. "Seesubtractive synthesis ."**Ideal and real filters**An ideal low-pass filter completely eliminates all frequencies above the

cutoff frequency while passing those below unchanged. The transition region present in practical filters does not exist in an ideal filter. An ideal low-pass filter can be realized mathematically (theoretically) by multiplying a signal by therectangular function in the frequency domain or, equivalently,convolution with asinc function in the time domain.However, the ideal filter is impossible to realize without also having signals of infinite extent, and so generally needs to be approximated for real ongoing signals, because the sinc function's support region extends to all past and future times. The filter would therefore need to have infinite delay, or knowledge of the infinite future and past, in order to perform the convolution. It is effectively realizable for pre-recorded digital signals by assuming extensions of zero into the past and future, but even that is not typically practical.

Real filters for real-time applications approximate the ideal filter by truncating and windowing the infinite impulse response to make a

finite impulse response ; applying that filter requires delaying the signal for a moderate period of time, allowing the computation to "see" a little bit into the future. This delay is manifested as phase shift. Greater accuracy in approximation requires a longer delay.The

Whittaker–Shannon interpolation formula describes how to use a perfect low-pass filter to reconstruct acontinuous signal from a sampleddigital signal . Realdigital-to-analog converter s use real filter approximations.**Electronic low-pass filters**There are a great many different types of filter circuits, with different responses to changing frequency. The frequency response of a filter is generally represented using a

Bode plot .* A

**first-order filter**, for example, will reduce the signal amplitude by half (about –6 dB) every time the frequency doubles (goes up oneoctave ); more precisely, the rolloff approaches 20 dB per decade in the limit of high frequency. The magnitude Bode plot for a first-order filter looks like a horizontal line below thecutoff frequency , and a diagonal line above the cutoff frequency. There is also a "knee curve" at the boundary between the two, which smoothly transitions between the two straight line regions. If thetransfer function of a first-order lowpass filter has a zero as well as a pole, the Bode plot will flatten out again, at some maximum attenuation of high frequencies; such an effect is caused for example by a little bit of the input leaking around the one-pole filter; this one-pole–one-zero filter is still a first-order lowpass. "SeePole–zero plot andRC circuit ."* A

**second-order filter**attenuates higher frequencies more steeply. The Bode plot for this type of filter resembles that of a first-order filter, except that it falls off more quickly. For example, a second-orderButterworth filter will reduce the signal amplitude to one fourth its original level every time the frequency doubles (–12 dB per octave, or –40 dB per decade). Other all-pole second-order filters may roll off at different rates initially depending on theirQ factor , but approach the same final rate of –12 dB per octave; as with the first-order filters, zeroes in the transfer function can change the high-frequency asymptote. SeeRLC circuit .* Third- and higher-order filters are defined similarly. In general, the final rate of rolloff for an order-n all-pole filter is 6n dB per octave.

On any Butterworth filter, if one extends the horizontal line to the right and the diagonal line to the upper-left (the

asymptote s of the function), they will intersect at exactly the "cutoff frequency". The frequency response at the cutoff frequency in a first-order filter is –3 dB below the horizontal line. The various types of filters —Butterworth filter ,Chebyshev filter ,Bessel filter , etc. — all have different-looking "knee curves". Many second-order filters are designed to have "peaking" or resonance, causing their frequency response at the cutoff frequency to be "above" the horizontal line. "Seeelectronic filter for other types."The meanings of 'low' and 'high' — that is, the

cutoff frequency — depend on the characteristics of the filter. The term "low-pass filter" merely refers to the shape of the filter's response; a high-pass filter could be built that cuts off at a lower frequency than any low-pass filter – it is their responses that set them apart. Electronic circuits can be devised for any desired frequency range, right up through microwave frequencies (above 1000 MHz) and higher.**Passive electronic realization**One simple

electrical circuit that will serve as a low-pass filter consists of aresistor in series with a load, and acapacitor in parallel with the load. The capacitor exhibits reactance, and blocks low-frequency signals, causing them to go through the load instead. At higher frequencies the reactance drops, and the capacitor effectively functions as a short circuit. The combination of resistance and capacitance gives you thetime constant of the filter $au\; =\; RC$ (represented by the Greek lettertau ). The break frequency, also called the turnover frequency orcutoff frequency (in hertz), is determined by the time constant:$f\_mathrm\{c\}\; =\; \{1\; over\; 2\; pi\; au\; \}\; =\; \{1\; over\; 2\; pi\; R\; C\}$

or equivalently (in

radians per second):$omega\_mathrm\{c\}\; =\; \{1\; over\; au\}\; =\; \{\; 1\; over\; R\; C\}.$

One way to understand this circuit is to focus on the time the capacitor takes to charge. It takes time to charge or discharge the capacitor through that resistor:

* At low frequencies, there is plenty of time for the capacitor to charge up to practically the same voltage as the input voltage.

* At high frequencies, the capacitor only has time to charge up a small amount before the input switches direction. The output goes up and down only a small fraction of the amount the input goes up and down. At double the frequency, there's only time for it to charge up half the amount.Another way to understand this circuit is with the idea of reactance at a particular frequency:

* Since DC cannot flow through the capacitor, DC input must "flow out" the path marked $V\_mathrm\{out\}$ (analogous to removing the capacitor).

* Since AC flows very well through the capacitor — almost as well as it flows through solid wire — AC input "flows out" through the capacitor, effectivelyshort circuit ing to ground (analogous to replacing the capacitor with just a wire).It should be noted that the capacitor is not an "on/off" object (like the block or pass fluidic explanation above). The capacitor will variably act between these two extremes. It is the

Bode plot andfrequency response that show this variability.**Active electronic realization**Another type of electrical circuit is an "active" low-pass filter.

In the

operational amplifier circuit shown in the figure, the cutoff frequency (inhertz ) is defined as:$f\_mathrm\{c\}\; =\; \{1\; over\; 2\; pi\; R\_2\; C\; \}$

or equivalently (in radians per second):

$omega\_mathrm\{c\}\; =\; frac\{1\}\{R\_2\; C\}$

The gain in the passband is $frac\{-R\_2\}\{R\_1\}$, and the stopband drops off at −6 dB per octave, as it is a first-order filter.

Sometimes, a simple gain amplifier (as opposed to the very-high-gain operation amplifier) is turned into a low-pass filter by simply adding a feedback capacitor C. This feedback decreases the frequency response at high frequencies via the

Miller effect , and helps to avoid oscillation in the amplifier. For example, an audio amplifier can be made into a low-pass filter with cutoff frequency 100 kHz to reduce gain at frequencies which would otherwise oscillate. Since the audio band (what we can hear) only goes up to 20 kHz or so, the frequencies of interest fall entirely in thepassband , and the amplifier behaves the same way as far as audio is concerned.**Laplace notation**Continuous-time filters can also be described in terms of the

Laplace transform of their impulse response in a way that allows all of the characteristics of the filter to be easily analyzed by considering the pattern of poles and zeros of the Laplace transform in the complex plane (in discrete time, one can similarly consider theZ-transform of the impulse response).A first-order low-pass filter can be described in Laplace notation as

:$frac\{mathrm\{Output\{mathrm\{Input\; =\; frac\{1\}\{1\; +\; s\; au\}$

where "s" is the Laplace transform variable and "τ" is the filter

time constant .**Digital simulation**The effect of a low-pass filter can be simulated on a computer by analyzing its behavior in the time domain, and then discretizing the model.

From the circuit diagram to the right, according to Kirchoff's Laws and the definition of

capacitance ::$V\_\{in\}(t)\; -\; V\_\{out\}(t)\; =\; I(t)\; R$:$Q\_c(t)\; =\; C\; V\_\{out\}(t)$:$I(t)\; =\; frac\{d\; Q\_c\}\{d\; t\}$

Taking the time derivative of the second equation, $I(t)\; =\; C\; frac\{dV\_\{out\{dt\}$. Combining this with the first equation:

:$V\_\{in\}(t)\; -\; V\_\{out\}(t)\; =\; C\; left\; [frac\{dV\_\{out\{dt\}\; ight]\; R$

Now we may discretize the equation. Let us represent $V\_\{in\}$ by a series of samples $x\_\{1...n\}$. We will likewise represent $V\_\{out\}$ by a series of sample $y\_\{1...n\}$ at thesame points in time. For simplicity we assume that the samples are taken at evenly-spaced points in time separated by $Delta\; t$. Making these substitutions:

:$x\_i\; -\; y\_i\; =\; C\; left\; [\; frac\{y\_\{i\}-y\_\{i-1\{Delta\; t\}\; ight]\; R$

And rearranging terms:

:$y\_i\; =\; x\_i\; left(\; frac\{Delta\; t\}\{RC\; +\; Delta\; t\}\; ight)\; +\; y\_\{i-1\}\; left(\; frac\{RC\}\{RC\; +\; Delta\; t\}\; ight)$

or more succinctly,

:$y\_n\; =\; alpha\; x\_n\; +\; (1\; -\; alpha)\; y\_\{n-1\},$ :where $alpha\; =\; frac\{Delta\; t\}\{RC\; +\; Delta\; t\}$

This gives us a way to determine the output samples in terms of the input samples and the preceding output. The following algorithm will simulate the effect of a low-pass filter on a series of digital samples:

// Return RC low-pass filter output samples, given input samples, // time interval "dt", and time constant "RC"

**function**lowpass("real [0..n] " x, "real" dt, "real" RC)**var**"real [0..n] " y**var**"real" alpha := dt / (RC + dt) y [0] := x [0]**for**i**from**1**to**n y [i] := alpha * x [i] + (1-alpha) * y [i-1]**return**yEquivalently, more efficiently, and somewhat more intuitively (the change in filter output is proportional to the difference between the last output and the current input, which is the essence of exponential decay):

**for**i**from**1**to**n y [i] := y [i-1] + alpha * (x [i] - y [i-1] )**See also***

Baseband

*Digital filter : Another realization of a low-pass filter

*High-pass filter

*Band-stop filter

*Band-pass filter **External links*** [

*http://www.allaboutcircuits.com/vol_2/chpt_8/2.html Low-pass filter*]

* [*http://electronica.asm.ro/filters/low_pass_filter.html Low-pass visual simulation*]

* [*http://www.muzique.com/schem/filter.htm RC Filter Calculator*]

* [*http://electronica.asm.ro/filters/cutoff.php Cutoff frequency calculator*]

* [*http://www.st-andrews.ac.uk/~www_pa/Scots_Guide/experiment/lowpass/lpf.html Low Pass Filter java simulator*]

*Wikimedia Foundation.
2010.*