Audio filter - Audio filter An audio filter is a type of filter used for processing sound signals. Many types of filters exist for applications including graphic equalizers, synthesizers, sound effects, CD players and virtual reality systems. An audio filter is typically designed to pass some frequency regions through unattenuated while significantly attenutating others. In some applications, such as in the design of graphic equalizers or CD players, the filters are designed according to a set of objective criteria such as pass band, pass band attenutation, stop band and stop band attenutation, where the pass bands are the frequency ranges for which audio is attenuated less than a specified maximum, and the stop bands are the frequency ranges for which the audio must be attenuated by a specified minimum..
Filter - Filter A filter is a membrane or layer that is designed to block certain things (objects or substances) whilst letting others through. Filters are often used to remove harmful substances from air or water, for example to reduce air pollution or to make contaminated air breathable with a gas mask. Very common is the coffee filter. In mathematics, a filter is a subset of power set with certain properties. See filter (mathematics). A filter is a short-form of mail filter. An electronic filter is an electrical circuit whose frequency response or transfer function is non-uniform. In other words, its gain or attenuation depends on frequency. An analog filter is a form of filter (usually electronic) that uses analog circuitry in its implementation. A digital filter is.
Electronic filter - Electronic filter An electronic filter eliminates unwanted frequencies from an electronic signal. A low-pass filter passes low frequencies. A high-pass filter passes high frequencies. A band-pass filter passes a limited range of frequencies. A band-stop filter passes all frequencies except a limited range. A notch filter is a type of band-stop filter that acts on a particularly narrow range of frequencies. Band-stop and band-pass filters can be constructed by combining low-pass and high-pass filters. A popular form of 2 pole filter is the Sallen-Key type. This is able to provide low-pass, band-pass, and high pass versions. Table of contents showTocToggle("show","hide") 1 Passive Filters 2 Active Filters 3 Other filters Passive Filters The simplest electronic filters are based on combinations of resistors, inductors and capacitors. Since resistance has.
Audio data compression - Audio data compression Note: This article is about audio data compression, which reduces the data rate of digital audio signals. This should not be confused with audio level compression (also known as companding), which reduces the dynamic range of audio signals. Audio compression is a form of data compression designed to reduce the size of audio data files. Audio compression algorithms are typically referred to as audio codecs. As with other specific forms of data compression, there exist many "lossless" and "lossy" algorithms to achieve the compression effect. Table of contents showTocToggle("show","hide") 1 Lossless Compression 2 Examples 3 Lossy Compression 4 Examples 5 See also Lossless Compression In contrast to image compression, lossless audio compression algorithms are not nearly as widely used. The primary users of.
Audio signal processing - Audio signal processing Audio signal processing, sometimes referred shortly to as audio processing or just audio, is the processing of some representation of auditory signals, that is sound. The representation can be digital or analog. An analog representation is usually electrical where the voltage level implies the pressure level (that is, the sound). Similarly, a digital representation expresses the pressure wave-form in a binary sequence. The focus in audio signal processing is most typically in an analysis of which parts ofa signal is audible. For example, a signal can be modified for different purposes such that the modification is controlled in the auditory domain. Which parts of the signal are heard and which are not, is not decided merely by physiology of the human hearing system,.
Audio mastering - Audio mastering Audio mastering is the process of preparing audio for playback on a wide range of playback devices. The main changes to a track during mastering are: tonal balancing and level adjustment. The former is primarily done with equalisation while the latter is primarily achieved with compression. Other effects that can be used during mastering are: audio enhancers, audio exciters and stereo expansion. The steps taken in mastering a number of tracks are: timeline all the tracks in the desired order. Leave a 2 second gap between tracks when producing a CD. apply noise reduction to eliminate hum and hiss normalize the tracks to set the highest peaks in audio volume to a preset level; the overall audio should never exceed 0 dBfs equalize audio.
Sinc filter - Sinc filter In signal processing, the sinc filter strips high-frequency data from a wave. If you find this article confusing, you may want to read about aliasing and signal processing as well as the Fourier transform and convolutions. Many physical processes are subject to noise. For instance, reception of an ordinary music radio is rarely crystal-clear, telephones don't transmit a perfect sound and old pictures get scratched or lose some of their colors. One method of minimizing such defects is to filter the sounds and images, to remove obvious noises and scratches. At this point it is unfortunately impossible to create filters that restore a signal to its pristine original self, but we have a starting point, and that is the sinc filter. The sinc filter presumes.
Low-pass filter - Low-pass filter A low-pass filter is an electronic filter that allows low frequencies to pass through, but attenuates high frequencies. Low-pass filters are used to block unwanted high-frequency signals, whilst passing the lower frequencies. They are the opposite of high-pass filters. Examples of low-pass filters A physical barrier acts as a low-pass filter for sound waves. When music is playing in another room, the low notes are easily heard, while the high notes are largely filtered out. Similarly, very loud music played in one car is heard as a low throbbing by occupants of other cars, because the closed vehicles (and air gap) function as a very low-pass filter. Low-pass filters are also used in subwoofers and other types of loudspeaker, to block high pitches that they can't.
HILN - or Harmonic and Individual Lines and Noise is a parametric audio codec for audio. The basic premise of the encoder is that most audio, and particularly speech, can be synthesized from only sinusoids and noise. The encoder describes individual sinusses with amplitude and frequency, harmonic tones by fundamental fequency, amplitude and the sepctral envelope of the partials, and the noise by amplitude and spectral envelope. This type of encoder is capable of encoding audio to between 6 and 16 kilobits per second for a typical audio bandwidth of 8 kHz. The framelength of this encoder is 32 msec. A typical codec extracts sinusoid information from the samples by applying a short fourier transform to the samples and using that to find the important harmonic content of a single frame. By matching.
Flanging - Flanging Flanging is an audio effect that occurs when a sound is echoed for a very short, but slowly varying, period of time. The delayed signal, usually smaller than 10 ms (milliseconds), is added to, or mixed with, the original signal. It gives sound a 'comb filter' effect that changes over time. A flanger is a device dedicated to creating this sound effect. According to one story, the effect was given its name by none other than Beatle John Lennon in the early 1960s. The name originates from the original implementation which was created by playing the same recording on two synchronized tape recorders, and then mixing the signals together. As long as the machines were synchronized, the mix would sound more-or-less normal, but if the operator placed.
FM radio - the development of, FM radio. On March 1, 1945 W47NV began operations in Nashville, Tennessee becoming the first modern commercial FM radio station. FM stereo technology New technology was added to FM radio in the early 1960s to allow FM stereo transmissions, where the frequency modulated radio signal is used to carry stereophonic sound, using the pilot-tone multiplex system. This multiplexes the left and right audio signal channels in a manner that is compatible with mono sound, using a sum-and-difference technique to produce a single "mono-compatible" signal, which has a baseband part that is equal to the sum of the left and right channels (L+R), and a higher-frequency part that is the difference of the left and right channels (L-R) amplitude modulated on a 38 kHz subcarrier. A 19 kHz pilot.
Electrical engineering - put to every day use are antenna design for use in mobile phones, and controlling the form of the electromagnetic field in an MRI scanner by the exact placement and alignment of its electromagnets. Another technology made possible by electromagnetism is the microwave oven. The field of high-power radio-frequency (RF) engineering was once feared to be a lost art. Because of the trend for low-power, miniaturized circuitry, there is a perception that the need for high-power radio engineering and engineers is diminishing. On the contrary, the need for engineers and technicians in this particular field has never been greater, and the need will only increase in the foreseeable future. Theories and tools The theories and tools an electrical engineer can consult include mathematics and physics in general, the theory of electromagnetism,.
Electronic mixer - types of mixer. Additive mixers add two signals together, and are used for such applications as audio mixing. Multiplying mixers multiply the signals together, and produce an output containing both original signals, and new signals that have the sum and difference of the frequency of the original signals. Additive mixers are usually resistor networks, surrounded by impedance matching and amplification stages. Multiplying mixers have been done in a wide variety of ways. The most popular are diode mixers, gilbert cell mixers, diode ring mixers and switching mixers. A diode mixer has two or more signals going into a diode. Whenever any signal pushes the voltage above the threshold of the diode, current will flow to the other side, but not back. If the inputs are the right voltages, the result is.
Electronic amplifier - and employ in practice. If the amplifying element is linear, then the output will be faithful copy of the input, only larger and inverted. In practice, transistors are not linear, and the output will only approximate the input. This is the origin of distortion within an amplifier. Which class of amplifier (A, B, AB or C) depends on how the amplifying device is biased - in the diagrams the bias circuits are omitted for clarity. Table of contents showTocToggle("show","hide") 1 Class A 2 Class B and AB 3 Class C 4 Negative Feedback 5 A Practical Circuit 6 Class D and E 7 See Also Class A Class A amplifiers amplify over the whole of the input cycle. They are the usual means of implementing small-signal amplifiers. They are not very.
Envelope detector - be written in the following form x(t) = R(t)cos(ωt+φ(t)) In the case of AM, φ(t) is constant and can be ignored, so all the information in the signal is in R(t), which is called the envelope of the signal. Given that an AM signal is given by the equation with m(t) representing the original audio frequency message, x(t) = (C + m(t))cos(ωt) R(t) is then equal to (C + m(t). So, if the envelope of the AM signal can be extracted, the original message can be recovered. Diode Detector The simplest form of envelope detector is the diode detector. To construct a diode detector, simply connect a diode between the input and output of a circuit, and connect a resistor and capacitor in parallel from the output of the circuit to.
Digital signal processing - these signals. DSP and analog signal processing are subsets of signal processing. It has three major subfields: audio signal processing, digital image processing and speech processing. In DSP, engineers most commonly study digital signals in one of the following domains: time domain (one-dimensional signals), spatial domain (multidimensional signals), frequency domain, autocorrelation domain, and wavelet domains. They choose the domain in which to process a signal by making an educated guess (or trying out different possibilities) as to which domain best represents the essential characteristics of the signal. A sequence of samples from a measuring device produces a time or spatial domain representation, whereas a discrete Fourier transform produces the frequency domain information. The autocorrelation is, loosely speaking, defined as the expected value of correlation of the signal with itself on some.
Digital waveguide synthesis - synthesis Digital waveguide synthesis is the synthesis of audio using a digital waveguide. Digital waveguides are efficient computational models for real life media through which acoustic waves propagate. For this reason, digital waveguides constitute a major part of most modern physical modelling synthesizers. A basic digital waveguide (likely of a string) with a rigid termination on one end (left) and a frequency-dependent attenuating filter at the other (right). Digital waveguide models are comprised of delay lines to represent the geometry of the waveguide, digital filters to represent the frequency-dependent losses and dispersion in the medium, and often include non-linear elements. Losses incurred throughout the medium are generally consolidated so that they can be calculated once at the termination of a delay line, rather than many times throughout. Digital waveguide synthesis was.
A1000 - Hz NTSC TV output by default versions available 50/60Hz mode switchable by software, although switching a PAL Amiga to NTSC mode produces a 60Hz PAL display, and vice versa (this would usually suffice for software relying on a particular format) hardware-switchable low-pass audio filter (cut/join a track, many people built switches); software-switchable on later models IRQ sharing (like the PCI bus) IRQ system had 7 priority levels of interrupts No limit on number of interrupts available Resources handled by Autoconfig, very similar to ACPI, resources were not numbered or labelled, just given as amounts and addresses No specific I/O ports, instead using memory-mapped I/O space separately for each hardware device.
A500 - (24-bit external address bus) OCS/ECS chipset 50 Hz PAL and 60 Hz NTSC TV output by default versions available; 50/60Hz mode switchable by software software-switchable low-pass audio filter (power LED shows filter status, darker when off) IRQ sharing (like the PCI bus) IRQ system had 7 priority levels of interrupts No limit on number of interrupts available Resources handled by Autoconfig, very similar to ACPI, resources were not numbered or labelled, just given as amounts and addresses No specific I/O ports, instead using memory mapped I/O space separately for each hardware device (thanks to Jay Miner) The A500 designation was also used on an internal Acorn Archimedes development machine..
Aliasing - revolution, but now each image will have lapsed 25 hours. The apparent period of the sun is then 600 hours (24x25), which is 25 real days. A similar temporal aliasing effect may occur filming a spoked wheel. This effect can be used to cause a repetitive action to appear to slow down by using a strobe light. The term "aliasing" derives from the usage in radio engineering, where a radio signal could be picked up at two different positions on the radio dial in a superheterodyne radio: one where the local oscillator was above the radio frequency, and one where it was below. This is analogous to the frequency-space "wrapround" that is one way of understanding aliasing. The qualitative effects of aliasing can be heard in the following audio demonstration. Four.
