Cascaded Integrator–comb Filter
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Cascaded Integrator–comb Filter
In digital signal processing, a cascaded integrator–comb (CIC) is a computationally efficient class of low-pass finite impulse response (FIR) filter that chains N number of integrator and comb filter pairs (where N is the filter's order) to form a decimator or interpolator. In a decimating CIC, the input signal is first fed through N integrator stages, followed by a down-sampler, and then N comb stages. An interpolating CIC (e.g. Figure 1) has the reverse order of this architecture, but with the down-sampler replaced with a zero-stuffer (up-sampler). Operation CIC filters were invented by Eugene B. Hogenauer in 1979 (published in 1981), and are a class of FIR filters used in multi-rate digital signal processing. Unlike most FIR filters, it has a down-sampler or up-sampler in the middle of the structure, which converts between the high sampling rate of f_s used by the integrator stages and the low sampling rate of \tfrac used by the comb stages. Transfer function At ...
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Digital Signal Processing
Digital signal processing (DSP) is the use of digital processing, such as by computers or more specialized digital signal processors, to perform a wide variety of signal processing operations. The digital signals processed in this manner are a sequence of numbers that represent Sampling (signal processing), samples of a continuous variable in a domain such as time, space, or frequency. In digital electronics, a digital signal is represented as a pulse train, which is typically generated by the switching of a transistor. Digital signal processing and analog signal processing are subfields of signal processing. DSP applications include Audio signal processing, audio and speech processing, sonar, radar and other sensor array processing, spectral density estimation, statistical signal processing, digital image processing, data compression, video coding, audio coding, image compression, signal processing for telecommunications, control systems, biomedical engineering, and seismology ...
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Right-shift Operator
In computer programming, a bitwise operation operates on a bit string, a bit array or a binary numeral (considered as a bit string) at the level of its individual bits. It is a fast and simple action, basic to the higher-level arithmetic operations and directly supported by the central processing unit, processor. Most bitwise operations are presented as two-operand instructions where the result replaces one of the input operands. On simple low-cost processors, typically, bitwise operations are substantially faster than division, several times faster than multiplication, and sometimes significantly faster than addition. While modern processors usually perform addition and multiplication just as fast as bitwise operations due to their longer instruction pipelines and other computer architecture, architectural design choices, bitwise operations do commonly use less power because of the reduced use of resources. Bitwise operators In the explanations below, any indication of a bit's p ...
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Latency (engineering)
Latency, from a general point of view, is a time delay between the Causality, cause and the effect of some physical change in the system being observed. Lag (video games), Lag, as it is known in Gaming culture, gaming circles, refers to the latency between the input to a simulation and the visual or auditory response, often occurring because of network delay in online games. The original meaning of “latency”, as used widely in psychology, medicine and most other disciplines, derives from “latent”, a word of Latin origin meaning “hidden”.  Its different and relatively recent meaning (this topic) of “lateness” or “delay” appears to derive from its superficial similarity to the word “late”, from the old English “laet”. Latency is physically a consequence of the limited velocity at which any Event (relativity), physical interaction can propagate. The magnitude of this velocity is always less than or equal to the speed of light. Therefore, every physical s ...
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Settling Time
In control theory the settling time of a dynamical system such as an amplifier or other output device is the time elapsed from the application of an ideal instantaneous step input to the time at which the amplifier output has entered and remained within a specified error band. Settling time includes a propagation delay, plus the time required for the output to slew to the vicinity of the final value, recover from the overload condition associated with slew, and finally settle to within the specified error. Systems with energy storage cannot respond instantaneously and will exhibit transient responses when they are subjected to inputs or disturbances. Definition Tay, Mareels and Moore (1998) defined settling time as "the time required for the response curve to reach and stay within a range of certain percentage (usually 5% or 2%) of the final value." Mathematical detail Settling time depends on the system response and natural frequency. The settling time for a second or ...
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Half-power Point
The half-power point is the condition at which a system's output power has dropped to half of its peak value; that is, at a level of approximately . In filters, optical filters, and electronic amplifiers, the half-power point is also known as half-power bandwidth and is a commonly used definition for the cutoff frequency. In the characterization of antennas the half-power point is also known as half-power beamwidth and relates to measurement position as an angle and describes directionality. Amplifiers and filters This occurs when the output voltage has dropped to \tfrac \approx \text of the filter's nominal passband voltage and the power has dropped by half. A bandpass amplifier will have two half-power points, while a low-pass amplifier or a high-pass amplifier will have only one. The bandwidth of a filter or amplifier is usually defined as the difference between the lower and upper half-power points. This is, therefore, also known as the bandwidth. There is no lower hal ...
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Delta-sigma Modulation
Delta-sigma (ΔΣ; or sigma-delta, ΣΔ) modulation is an oversampling method for encoding signals into low bit depth digital signals at a very high sample-frequency as part of the process of delta-sigma analog-to-digital converters (ADCs) and digital-to-analog converters (DACs). Delta-sigma modulation achieves high quality by utilizing a negative feedback loop during quantization to the lower bit depth that continuously corrects quantization errors and moves quantization noise to higher frequencies well above the original signal's bandwidth. Subsequent low-pass filtering for demodulation easily removes this high frequency noise and time averages to achieve high accuracy in amplitude, which can be ultimately encoded as pulse-code modulation (PCM). Both ADCs and DACs can employ delta-sigma modulation. A delta-sigma ADC (e.g. Figure 1 top) encodes an analog signal using high-frequency delta-sigma modulation and then applies a digital filter to demodulate it t ...
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Sinc Filter
In signal processing, a sinc filter can refer to either a sinc-in-time filter whose impulse response is a sinc function and whose frequency response is rectangular, or to a sinc-in-frequency filter whose impulse response is rectangular and whose frequency response is a sinc function. Calling them according to which domain the filter resembles a sinc avoids confusion. If the domain is unspecified, sinc-in-time is often assumed, or context hopefully can infer the correct domain. Sinc-in-time Sinc-in-time is an ideal filter that removes all frequency components above a given cutoff frequency, without attenuating lower frequencies, and has linear phase response. It may thus be considered a ''brick-wall filter'' or ''rectangular filter.'' Its impulse response is a sinc function in the time domain: \frac while its frequency response is a rectangular function: :H(f) = \operatorname \left( \frac \right) = \begin 0, & \text , f, > B, \\ \frac, & \text , f, = B, \\ ...
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Low-pass Filter
A low-pass filter is a filter that passes signals with a frequency lower than a selected cutoff frequency and attenuates signals with frequencies higher than the cutoff frequency. The exact frequency response of the filter depends on the filter design. The filter is sometimes called a high-cut filter, or treble-cut filter in audio applications. A low-pass filter is the complement of a high-pass filter. In optics, high-pass and low-pass may have different meanings, depending on whether referring to the frequency or wavelength of light, since these variables are inversely related. High-pass frequency filters would act as low-pass wavelength filters, and vice versa. For this reason, it is a good practice to refer to wavelength filters as ''short-pass'' and ''long-pass'' to avoid confusion, which would correspond to ''high-pass'' and ''low-pass'' frequencies. Low-pass filters exist in many different forms, including electronic circuits such as a '' hiss filter'' used in audio, ...
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Unit Circle
In mathematics, a unit circle is a circle of unit radius—that is, a radius of 1. Frequently, especially in trigonometry, the unit circle is the circle of radius 1 centered at the origin (0, 0) in the Cartesian coordinate system in the Euclidean plane. In topology, it is often denoted as because it is a one-dimensional unit -sphere. If is a point on the unit circle's circumference, then and are the lengths of the legs of a right triangle whose hypotenuse has length 1. Thus, by the Pythagorean theorem, and satisfy the equation x^2 + y^2 = 1. Since for all , and since the reflection of any point on the unit circle about the - or -axis is also on the unit circle, the above equation holds for all points on the unit circle, not only those in the first quadrant. The interior of the unit circle is called the open unit disk, while the interior of the unit circle combined with the unit circle itself is called the closed unit disk. One may also use other notions of "dis ...
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DC Bias
In signal processing, when describing a periodic function in the time domain, the DC bias, DC component, DC offset, or DC coefficient is the mean value of the waveform. A waveform with zero mean or no DC bias is known as a ''DC balanced'' or ''DC free'' waveform. Origin The term originates in electronics, where ''DC'' refers to a direct current voltage. In contrast, various other non-DC frequencies are analogous to superimposed alternating current (AC) voltages or currents, hence called AC component or AC coefficients. Applications In the design of electronic amplifier circuits, every active device has biasing to set its ''operating point'', the steady state current and voltage on the device when no signal is applied. In bipolar transistor biasing, for example, a network of resistors is used to apply a small amount of DC to the base terminal of the transistor. The AC signal is applied at the same terminal and is amplified. The bias network is designed to preserve the applied AC ...
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Decibel
The decibel (symbol: dB) is a relative unit of measurement equal to one tenth of a bel (B). It expresses the ratio of two values of a Power, root-power, and field quantities, power or root-power quantity on a logarithmic scale. Two signals whose level (logarithmic quantity), levels differ by one decibel have a power ratio of 101/10 (approximately ) or root-power ratio of 101/20 (approximately ). The strict original usage above only expresses a relative change. However, the word decibel has since also been used for expressing an Absolute scale, absolute value that is relative to some fixed reference value, in which case the dB symbol is often suffixed with letter codes that indicate the reference value. For example, for the reference value of 1 volt, a common suffix is "#Voltage, V" (e.g., "20 dBV"). As it originated from a need to express power ratios, two principal types of scaling of the decibel are used to provide consistency depending on whether the scaling refer ...
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Group Delay And Phase Delay
In signal processing, group delay and phase delay are functions that describe in different ways the delay times experienced by a signal’s various sinusoidal frequency components as they pass through a linear time-invariant (LTI) system (such as a microphone, coaxial cable, amplifier, loudspeaker, communications system, ethernet cable, digital filter, or analog filter). Unfortunately, these delays are sometimes frequency dependent, which means that different sinusoid frequency components experience different time delays. As a result, the signal's waveform experiences distortion as it passes through the system. This distortion can cause problems such as poor fidelity in analog video and analog audio, or a high bit-error rate in a digital bit stream. Background Frequency components of a signal Fourier analysis reveals how signals in time can alternatively be expressed as the sum of sinusoidal frequency components, each based on the trigonometric function \sin(x) with a fixed ...
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