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Filter Theory
Filter design is the process of designing a signal processing filter that satisfies a set of requirements, some of which may be conflicting. The purpose is to find a realization of the filter that meets each of the requirements to an acceptable degree. The filter design process can be described as an optimization problem. Certain parts of the design process can be automated, but an experienced designer may be needed to get a good result. The design of digital filters is a complex topic. Although filters are easily understood and calculated, the practical challenges of their design and implementation are significant and are the subject of advanced research. Typical design requirements Typical requirements which may be considered in the design process are: * Frequency response * Phase shift or group delay * impulse response * Causal filter required? * Stable filter required? * Finite (in duration) impulse response required? * Computational complexity * Technology The frequ ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Filter (signal Processing)
In signal processing, a filter is a device or process that removes some unwanted components or features from a Signal (electronics), signal. Filtering is a class of signal processing, the defining feature of filters being the complete or partial suppression of some aspect of the signal. Most often, this means removing some frequency, frequencies or frequency bands. However, filters do not exclusively act in the frequency domain; especially in the field of image processing many other targets for filtering exist. Correlations can be removed for certain frequency components and not for others without having to act in the frequency domain. Filters are widely used in electronics and telecommunication, in radio, television, audio recording, radar, control systems, music synthesis, image processing, computer graphics, and structural dynamics. There are many different bases of classifying filters and these overlap in many different ways; there is no simple hierarchical classification. Fil ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
All-pass Filter
An all-pass filter is a signal processing filter that passes all frequencies equally in gain, but changes the phase relationship among various frequencies. Most types of filter reduce the amplitude (i.e. the magnitude) of the signal applied to it for some values of frequency, whereas the all-pass filter allows all frequencies through without changes in level. Common applications A common application in electronic music production is in the design of an effects unit known as a " phaser", where a number of all-pass filters are connected in sequence and the output mixed with the raw signal. It does this by varying its phase shift as a function of frequency. Generally, the filter is described by the frequency at which the phase shift crosses 90° (i.e., when the input and output signals go into quadrature – when there is a quarter wavelength of delay between them). They are generally used to compensate for other undesired phase shifts that arise in the system, or for mi ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Digital Filter
In signal processing, a digital filter is a system that performs mathematical operations on a Sampling (signal processing), sampled, discrete-time signal to reduce or enhance certain aspects of that signal. This is in contrast to the other major type of electronic filter, the analog filter, which is typically an electronic circuit operating on continuous-time analog signals. A digital filter system usually consists of an analog-to-digital converter (ADC) to sample the input signal, followed by a microprocessor and some peripheral components such as memory to store data and filter coefficients etc. Program Instructions (software) running on the microprocessor implement the digital filter by performing the necessary mathematical operations on the numbers received from the ADC. In some high performance applications, an FPGA or ASIC is used instead of a general purpose microprocessor, or a specialized digital signal processor (DSP) with specific paralleled architecture for expedi ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Analog Sampled Filter
An analog sampled filter an electronic filter that is a hybrid between an analog and a digital filter. The input is an analog signal, and usually stored in capacitors. The time domain is discrete, however. Distinct analog samples are shifted through an array of holding capacitors as in a bucket brigade. Analog adders and amplifiers do the arithmetic in the signal domain, just as in an analog computer. Note that these filters are subject to aliasing phenomena just like a digital filter, and anti-aliasing filters will usually be required. See . Companies such as Linear Technology and Maxim produce integrated circuits that implement this functionality. Filters up to the 8th order may be implemented using a single chip. Some are fully configurable; some are pre-configured, usually as low-pass filters. Due to the high filter order that can be achieved in an easy and stable manner, single chip analog sampled filters are often used for implementing anti-aliasing filters for ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Analog Filter
Analogue Filter (signal processing), filters are a basic building block of signal processing much used in electronics. Amongst their many applications are the separation of an audio signal before application to bass (music), bass, mid-range speaker, mid-range, and tweeter loudspeakers; the combining and later separation of multiple telephone conversations onto a single channel; the selection of a chosen radio station in a radio receiver and rejection of others. Passive linear electronic analogue filters are those filters which can be described with linear differential equations (linear); they are composed of capacitors, inductors and, sometimes, resistors (passive component, passive) and are designed to operate on continuously varying analogue signals. There are many linear filters which are not analogue in implementation (digital filter), and there are many electronic filters which may not have a passive topology – both of which may have the same transfer function of th ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Transient (acoustics)
In acoustics and audio, a transient is a high amplitude, short-duration sound at the beginning of a waveform that occurs in phenomena such as musical sounds, noises or speech. Transients do not necessarily directly depend on the frequency of the tone they initiate. It contains a high degree of non-periodic components and a higher magnitude of high frequencies than the harmonic content of that sound. Transients are more difficult to encode with many audio compression algorithms, causing pre-echo. See also * Prefix (acoustics) * Impulse function * Onset (audio) * Transient response – a common electrical engineering term that may be the source of the idea of an acoustic "transient" References {{reflist Acoustics Sonar de:Einschwingvorgang ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Steady State
In systems theory, a system or a process is in a steady state if the variables (called state variables) which define the behavior of the system or the process are unchanging in time. In continuous time, this means that for those properties ''p'' of the system, the partial derivative with respect to time is zero and remains so: : \frac = 0 \quad \text t. In discrete time, it means that the first difference of each property is zero and remains so: : p_t-p_=0 \quad \text t. The concept of a steady state has relevance in many fields, in particular thermodynamics, economics, and engineering. If a system is in a steady state, then the recently observed behavior of the system will continue into the future. In stochastic systems, the probabilities that various states will be repeated will remain constant. For example, see ' for the derivation of the steady state. In many systems, a steady state is not achieved until some time after the system is started or initiated. This initial sit ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
High-resolution Audio
High-resolution audio is a term for music files with bit depth greater than 16-bit and sampling frequency higher than 44.1 kHz or 48 kHz used in CD and DVD formats. The Audio Engineering Society (AES), Consumer Technology Association (CTA) and Japan Audio Society (JAS) set 24-bit/96 kHz as the minimum requirement to fulfill the standard. The Recording Academy Producers & Engineers Wing also cites 24-bit/96 kHz as the preferred resolution for tracking, mixing and mastering audio. It is supported by media formats such as DVD-Audio, DualDisc and High Fidelity Pure Audio, download stores like Bandcamp, HDtracks and Qobuz, and streaming platforms including Apple Music, Amazon Music and Tidal. Research into high-resolution audio began in the late 1980s and recordings were made available on the consumer market in 1996. Other bit depth/sample rate combinations that are often marketed as "high-resolution" include 1-bit/2.8224 MHz ( DSD), 20-bit/44.1 kHz ( HDCD), 24-bit/44 ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Fourier Transform
In mathematics, the Fourier transform (FT) is an integral transform that takes a function as input then outputs another function that describes the extent to which various frequencies are present in the original function. The output of the transform is a complex-valued function of frequency. The term ''Fourier transform'' refers to both this complex-valued function and the mathematical operation. When a distinction needs to be made, the output of the operation is sometimes called the frequency domain representation of the original function. The Fourier transform is analogous to decomposing the sound of a musical chord into the intensities of its constituent pitches. Functions that are localized in the time domain have Fourier transforms that are spread out across the frequency domain and vice versa, a phenomenon known as the uncertainty principle. The critical case for this principle is the Gaussian function, of substantial importance in probability theory and statist ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Digital Delay Line
A digital delay line (or simply delay line, also called delay filter) is a discrete element in a digital filter, which allows a signal to be delayed by a number of Sample (signal), samples. Delay lines are commonly used to delay audio signals feeding loudspeakers to compensate for the speed of sound in air, and to align video signals with accompanying audio, called audio-to-video synchronization. Delay lines may compensate for Latency (audio), electronic processing latency so that multiple signals leave a device simultaneously despite having different pathways. Digital delay lines are widely used building blocks in methods to simulate room acoustics, musical instruments and Effects unit, effects units. Digital waveguide synthesis shows how digital delay lines can be used as sound synthesis methods for various musical instruments such as string instruments and wind instruments. If a delay line holds a non-integer value smaller than one, it results in a fractional delay line (also ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Hilbert Transform
In mathematics and signal processing, the Hilbert transform is a specific singular integral that takes a function, of a real variable and produces another function of a real variable . The Hilbert transform is given by the Cauchy principal value of the convolution with the function 1/(\pi t) (see ). The Hilbert transform has a particularly simple representation in the frequency domain: It imparts a phase shift of ±90° (/2 radians) to every frequency component of a function, the sign of the shift depending on the sign of the frequency (see ). The Hilbert transform is important in signal processing, where it is a component of the Analytic signal, analytic representation of a real-valued signal . The Hilbert transform was first introduced by David Hilbert in this setting, to solve a special case of the Riemann–Hilbert problem for analytic functions. Definition The Hilbert transform of can be thought of as the convolution of with the function , known as the Cauchy ker ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Phaser (effect)
A phaser is an electronic sound processor used to filter a signal by creating a series of peaks and troughs in the frequency spectrum. The position of the peaks and troughs of the waveform being affected is typically modulated by an internal low-frequency oscillator so that they vary over time, creating a sweeping effect. Phasers are often used to give a "synthesized" or electronic effect to natural sounds, such as human speech. Process The electronic phasing effect is created by splitting an audio signal into two paths. One path treats the signal with an all-pass filter, which preserves the amplitude of the original signal and alters the phase. The amount of change in phase depends on the frequency. When signals from the two paths are mixed, the frequencies that are out of phase will cancel each other out, creating the phaser's characteristic notches. Changing the mix ratio changes the depth of the notches; the deepest notches occur when the mix ratio is 50%. The def ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
