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 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.
Signal processing is an electrical engineering subfield that focuses on analyzing, modifying and synthesizing signals, such as sound, images, potential fields, seismic signals, altimetry processing, and scientific measurements. Signal processing techniques are used to optimize transmissions, digital storage efficiency, correcting distorted signals, subjective video quality and to also detect or pinpoint components of interest in a measured signal.
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 for recording voice, speech recognition, VoIP, and for other purposes such as ultrasonic sensors or knock sensors.
Binaural recording is a method of recording sound that uses two microphones, arranged with the intent to create a 3-D stereo sound sensation for the listener of actually being in the room with the performers or instruments. This effect is often created using a technique known as dummy head recording, wherein a mannequin head is fitted with a microphone in each ear. Binaural recording is intended for replay using headphones and will not translate properly over stereo speakers. This idea of a three-dimensional or "internal" form of sound has also translated into useful advancement of technology in many things such as stethoscopes creating "in-head" acoustics and IMAX movies being able to create a three-dimensional acoustic experience.
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 which may occur when an acoustic path exists between an audio input and an audio output. 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.
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.
Noise shaping is a technique typically used in digital audio, image, and video processing, usually in combination with dithering, as part of the process of quantization or bit-depth reduction of a digital signal. Its purpose is to increase the apparent signal-to-noise ratio of the resultant signal. It does this by altering the spectral shape of the error that is introduced by dithering and quantization; such that the noise power is at a lower level in frequency bands at which noise is considered to be less desirable and at a correspondingly higher level in bands where it is considered to be more desirable. A popular noise shaping algorithm used in image processing is known as ‘Floyd Steinberg dithering’; and many noise shaping algorithms used in audio processing are based on an ‘Absolute threshold of hearing’ model.
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.
Acoustic quieting is the process of making machinery quieter by damping vibrations to prevent them from reaching the observer. Machinery vibrates, causing sound waves in air, hydroacoustic waves in water, and mechanical stresses in solid matter. Quieting is achieved by absorbing the vibrational energy or minimizing the source of the vibration. It may also be redirected away from the observer.
Audio signal flow is the path an audio signal takes from source to output. The concept of audio signal flow is closely related to the concept of audio gain staging; each component in the signal flow can be thought of as a gain stage.
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 directed sounds in front of the ear and blocked 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.
