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Adaptive feedback cancellation is a common method of cancelling audio feedback in a variety of electro-acoustic systems such as digital hearing aids. The time-varying acoustic feedback leakage paths can only be eliminated with adaptive feedback cancellation. When an electro-acoustic system with an adaptive feedback canceller is presented with a correlated input signal, a recurrent distortion artifact, entrainment is generated. There is a difference between the system identification and feedback cancellation.
Adaptive feedback cancellation has its application in echo cancellation. The error between the desired and the actual output is taken and given as feedback to the adaptive processor for adjusting its coefficients to minimize the error.
In hearing aids, feedback arises when a part of the receiver (loudspeaker) signal is captured by the hearing aid microphone(s), gets amplified in the device and starts to loop around through the system. When feedback occurs, it results in a disturbingly loud tonal signal. Feedback is more likely to occur when the hearing aid volume is increased, when the hearing aid fitting is not in its proper position or when the hearing aid is brought close to a reflecting surface (e.g. when using a mobile phone). Adaptive feedback cancellation algorithms are techniques that estimate the transmission path between loudspeaker and microphone(s). This estimate is then used to implement a neutralizing electronic feedback path that suppresses the tonal feedback signal. [1] [2]
Adaptive feedback cancellation originated during the evolution of the hearing aid. The hearing aid became digital, and as such feedback cancellation was needed. In 1980 a directional microphone was introduced in the digital hearing aid, and adaptive feedback cancellation was created to block external noise that the microphone picked up. Today, adaptive feedback cancellation is in nearly every digital hearing aid and current research is still ongoing. [3]
Adaptive feedback cancellation follows the following process:
Hearing aids use adaptive feedback cancellation to improve the amount of gain. When hearing aids are misplaced or turned to loud volumes they can have large feedback. Noise picked up through the hearing aid's microphone creates this feedback, which is then amplified creating a ringing noise. Adaptive feedback cancellation cancels the noise with an adaptive filter. The product of the adaptive filter cancels the feedback from the microphone creating clear sound from the hearing aid. [4]
Echo cancellation is a form of adaptive feedback cancellation used in telephones and teleconferencing devices. Much like adaptive feedback cancellation in hearing aids, echo cancellation uses an adaptive filter to cancel echo reverberations from a microphone. [5]
Jammer suppression is a way to reject interference with large signals that are much stronger than traditional signals. This process uses an adaptive filter, and its algorithm has applications in all types of signal suppression. [6]
Prediction error method is an adaptive feedback cancellation algorithm that focuses on using audio signals instead of speech signals. This change proposes to improve adaptive feedback cancellation in objects like hearing aids and other audio applications. This approach works closely with echo cancellation, and looks to bring the techniques in echo cancellation to hearing aids. [7]
Sub-band feedback cancellation is a type of adaptive feedback cancellation that relies less on computations and algorithms, but rather uses the signals already in the device to optimize the feedback cancellation. Sub-band feedback cancellations purpose is to make adaptive feedback cancellation cheaper and more widespread.[ citation needed ]
Digital Signal Processing pledges to reduce the effect of adaptive feedback cancellation on sound quality with the use of feedback tests. The method also reports more gain in the hearing aid, and cites a figure of 10 decibels. [8]
Research from 2018 is ongoing into adaptive feedback cancellation on smartphone speakers and microphones. Current research intends to use digital signal processing to mimic the cancellation in hearing aids in smartphones. [9]
Audio signal processing is a subfield of signal processing that is concerned with the electronic manipulation of audio signals. Audio signals are electronic representations of sound waves—longitudinal waves which travel through air, consisting of compressions and rarefactions. The energy contained in audio signals or sound power level is typically measured in decibels. As audio signals may be represented in either digital or analog format, processing may occur in either domain. Analog processors operate directly on the electrical signal, while digital processors operate mathematically on its digital representation.
A microphone array is any number of microphones operating in tandem. There are many applications:
Digital audio is a representation of sound recorded in, or converted into, digital form. In digital audio, the sound wave of the audio signal is typically encoded as numerical samples in a continuous sequence. For example, in CD audio, samples are taken 44,100 times per second, each with 16-bit resolution. Digital audio is also the name for the entire technology of sound recording and reproduction using audio signals that have been encoded in digital form. Following significant advances in digital audio technology during the 1970s and 1980s, it gradually replaced analog audio technology in many areas of audio engineering, record production and telecommunications in the 1990s and 2000s.
A microphone, colloquially called a mic, or mike, is a transducer that converts sound into an electrical signal. Microphones are used in many applications such as telephones, hearing aids, public address systems for concert halls and public events, motion picture production, live and recorded audio engineering, sound recording, two-way radios, megaphones, and radio and television broadcasting. They are also used in computers and other electronic devices, such as mobile phones, for recording sounds, speech recognition, VoIP, and other purposes, such as ultrasonic sensors or knock sensors.
An adaptive filter is a system with a linear filter that has a transfer function controlled by variable parameters and a means to adjust those parameters according to an optimization algorithm. Because of the complexity of the optimization algorithms, almost all adaptive filters are digital filters. Adaptive filters are required for some applications because some parameters of the desired processing operation are not known in advance or are changing. The closed loop adaptive filter uses feedback in the form of an error signal to refine its transfer function.
Audio feedback is a positive feedback situation that may occur when an acoustic path exists between an audio output and its audio input. In this example, a signal received by the microphone is amplified and passed out of the loudspeaker. The sound from the loudspeaker can then be received by the microphone again, amplified further, and then passed out through the loudspeaker again. The frequency of the resulting howl is determined by resonance frequencies in the microphone, amplifier, and loudspeaker, the acoustics of the room, the directional pick-up and emission patterns of the microphone and loudspeaker, and the distance between them. The principles of audio feedback were first discovered by Danish scientist Søren Absalon Larsen, hence it is also known as the Larsen effect.
Active noise control (ANC), also known as noise cancellation (NC), or active noise reduction (ANR), is a method for reducing unwanted sound by the addition of a second sound specifically designed to cancel the first. The concept was first developed in the late 1930s; later developmental work that began in the 1950s eventually resulted in commercial airline headsets with the technology becoming available in the late 1980s. The technology is also used in road vehicles, mobile telephones, earbuds, and headphones.
A hearing aid is a device designed to improve hearing by making sound audible to a person with hearing loss. Hearing aids are classified as medical devices in most countries, and regulated by the respective regulations. Small audio amplifiers such as personal sound amplification products (PSAPs) or other plain sound reinforcing systems cannot be sold as "hearing aids".
A sound reinforcement system is the combination of microphones, signal processors, amplifiers, and loudspeakers in enclosures all controlled by a mixing console that makes live or pre-recorded sounds louder and may also distribute those sounds to a larger or more distant audience. In many situations, a sound reinforcement system is also used to enhance or alter the sound of the sources on the stage, typically by using electronic effects, such as reverb, as opposed to simply amplifying the sources unaltered.
In analog telephony, a telephone hybrid is the component at the ends of a subscriber line of the public switched telephone network (PSTN) that converts between two-wire and four-wire forms of bidirectional audio paths. When used in broadcast facilities to enable the airing of telephone callers, the broadcast-quality telephone hybrid is known as a broadcast telephone hybrid or telephone balance unit.
Beamforming or spatial filtering is a signal processing technique used in sensor arrays for directional signal transmission or reception. This is achieved by combining elements in an antenna array in such a way that signals at particular angles experience constructive interference while others experience destructive interference. Beamforming can be used at both the transmitting and receiving ends in order to achieve spatial selectivity. The improvement compared with omnidirectional reception/transmission is known as the directivity of the array.
The carbon microphone, also known as carbon button microphone, button microphone, or carbon transmitter, is a type of microphone, a transducer that converts sound to an electrical audio signal. It consists of two metal plates separated by granules of carbon. One plate is very thin and faces toward the speaking person, acting as a diaphragm. Sound waves striking the diaphragm cause it to vibrate, exerting a varying pressure on the granules, which in turn changes the electrical resistance between the plates. Higher pressure lowers the resistance as the granules are pushed closer together. A steady direct current is passed between the plates through the granules. The varying resistance results in a modulation of the current, creating a varying electric current that reproduces the varying pressure of the sound wave. In telephony, this undulating current is directly passed through the telephone wires to the central office. In public address systems it is amplified by an audio amplifier. The frequency response of most carbon microphones, however, is limited to a narrow range, and the device produces significant electrical noise.
An assistive listening device (ALD) is part of a system used to improve hearing ability for people in a variety of situations where they are unable to distinguish speech in noisy environments. Often, in a noisy or crowded room it is almost impossible for an individual who is hard of hearing to distinguish one voice among many. This is often exacerbated by the effect of room acoustics on the quality of perceived speech. Hearing aids are able to amplify and process these sounds, and improve the speech to noise ratio. However, if the sound is too distorted by the time it reaches the listener, even the best hearing aids will struggle to unscramble the signal. Assistive listening devices offer a more adaptive alternative to hearing aids, but can be more complex and cumbersome.
In live sound mixing, gain before feedback (GBF) is a practical measure of how much a microphone can be amplified in a sound reinforcement system before causing audio feedback. In audiology, GBF is a measure of hearing aid performance. In both fields the amount of gain is measured in decibels at or just below the point at which the sound from the speaker driver re-enters the microphone and the system begins to ring or feed back. Potential acoustic gain (PAG) is a calculated figure representing gain that a system can support without feeding back.
The first hearing aid was created in the 17th century. The movement toward modern hearing aids began with the creation of the telephone, and the first electric hearing aid was created in 1898. By the late 20th century, the digital hearing aid was distributed to the public commercially. Some of the first hearing aids were external hearing aids. External hearing aids direct sounds in front of the ear and block all other noises. The apparatus would fit behind or in the ear.
Echo suppression and echo cancellation are methods used in telephony to improve voice quality by preventing echo from being created or removing it after it is already present. In addition to improving subjective audio quality, echo suppression increases the capacity achieved through silence suppression by preventing echo from traveling across a telecommunications network. Echo suppressors were developed in the 1950s in response to the first use of satellites for telecommunications.
A feedback suppressor is an audio signal processing device which is used in the signal path in a live sound reinforcement system to prevent or suppress audio feedback.
Dereverberation is the process by which the effects of reverberation are removed from sound, after such reverberant sound has been picked up by microphones. Dereverberation is a subtopic of acoustic digital signal processing and is most commonly applied to speech but also has relevance in some aspects of music processing. Dereverberation of audio is a corresponding function to blind deconvolution of images, although the techniques used are usually very different. Reverberation itself is caused by sound reflections in a room and is quantified by the room reverberation time and the direct-to-reverberant ratio. The effect of dereverberation is to increase the direct-to-reverberant ratio so that the sound is perceived as closer and clearer.
Adaptive noise cancelling is a signal processing technique that is highly effective in suppressing additive interference or noise corrupting a received target signal at the main or primary sensor in certain common situations where the interference is known and is accessible but unavoidable and where the target signal and the interference are unrelated, that is, uncorrelated. Examples of such situations include:
Digital signal processing and machine learning are two technologies that are often combined.