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Digital room correction (or DRC) is a process in the field of acoustics where digital filters designed to ameliorate unfavorable effects of a room's acoustics are applied to the input of a sound reproduction system. Modern room correction systems produce substantial improvements in the time domain and frequency domain response of the sound reproduction system.
The use of analog filters, such as equalizers, to normalize the frequency response of a playback system has a long history; however, analog filters are very limited in their ability to correct the distortion found in many rooms. Although digital implementations of the equalizers have been available for some time, digital room correction is usually used to refer to the construction of filters which attempt to invert the impulse response of the room and playback system, at least in part. Digital correction systems are able to use acausal filters, and are able to operate with optimal time resolution, optimal frequency resolution, or any desired compromise along the Gabor limit. Digital room correction is a fairly new area of study which has only recently been made possible by the computational power of modern CPUs and DSPs.
The configuration of a digital room correction system begins with measuring the impulse response of the room at a reference listening position, and sometimes at additional locations for each of the loudspeakers. Then, computer software is used to compute a FIR filter, which reverses the effects of the room and linear distortion in the loudspeakers. In low performance conditions, a few IIR peaking filters are used instead of FIR filters, which require convolution, a relatively computation-heavy operation. Finally, the calculated filter is loaded into a computer or other room correction device which applies the filter in real time. Because most room correction filters are acausal, there is some delay. Most DRC systems allow the operator to control the added delay through configurable parameters.
The most widely used test signal is a swept sine wave, also called chirp. This signal maximizes the measurement's signal-to-noise ratio, and the spectrum can be calculated by deconvolution, which is dividing the response's Fourier transform with the signal's Fourier transform. The spectrum is then smoothed, and a filter set is calculated, which equalizes the sound pressure levels at each frequency to the target curve. To calculate the delays and other time-domain corrections, an inverse Fourier transform is performed on the spectrum, which results in the impulse response. The impulse peak's distance from the start of the signal is its delay, and its sign is its polarity. The delay is corrected by subtracting each channel's delay from the system's peak delay, and applying this result as additional delay for the channel. This correction is sometimes provided to the user as distance from the speaker, which is calculated by multiplying the delay with the speed of sound. Inverse polarity (most likely caused by switching a speaker's + and - wires) could be fixed by multiplying each sample with -1 or swapping the speaker wire ends on one side of the cable, but this result is usually shown as a warning, as some speakers (e.g. Focal Kanta) do this intentionally. [1]
DRC systems are not normally used to create a perfect inversion of the room's response because a perfect correction would only be valid at the location where it was measured: a few millimeters away the arrival times from various reflections will differ and the inversion will be imperfect. The imperfectly corrected signal may end up sounding worse than the uncorrected signal because the acausal filters used in digital room correction may cause pre-echo. Room correction filter calculation systems instead favor a robust approach, and employ sophisticated processing to attempt to produce an inverse filter which will work over a usably large volume, and which avoid producing bad-sounding artifacts outside of that volume, at the expense of peak accuracy at the measurement location.
Room EQ Wizard, or REW for short is a free room measurement tool with SPL, phase, distortion, RT60, clarity, decay, waterfall, and spectrogram views. REW also features IR windowing, and SPL meter, room simulation for subwoofer placement, and peaking filter-based EQ generation for multiple platforms, DSPs, and AVRs with a target curve editor.
QuickEQ is part of Cavern, a free and open source spatial audio engine. QuickEQ supports multichannel measurements with multiple microphones, time and level alignment with multiple standards and target curves, IR windowing, multi-sub crossover, and experimental filters for increasing speech intelligibility and simulating other cabinet types. QuickEQ exports minimum- or 0-phase FIR filters or peaking EQs depending on the target device.
RePhase is a free EQ and crossover generation tool that also linearizes phase response. RePhase has multiple configurable filter sets available for manual filter composition, which then can be exported as a single FIR impulse.
Most new AVRs include room correction in their setup, and a microphone in the box. Dirac Live is a commercial software that is available for PC and select Onkyo, Pioneer, Integra, StormAudio, and other AVRs. Denon and Marantz AVRs use Audyssey, and more expensive models allow for more corrections, and since the 2023 model year, Dirac Live as an alternative. Anthem AVRs use a proprietary software called Anthem Room Correction, or ARC for short, Yamaha uses Yamaha Parametric room Acoustic Optimizer (YPAO), as well as Trinnov Audio with their optimizer solution.
DCI-compliant hardware that are used in commercial theaters, sometimes use commercially available room correction software. Notable examples are IMAX cinemas, which use Audyssey MultEQ XT32, [2] while Datasat processors (found in all DTS:X rooms) have Dirac software. Dolby's CP850 and CP950 processors (which support Dolby Atmos) use a proprietary solution called AutoEQ. AutoEQ measures 5 to 8 microphone positions simultaneously. It requires loudspeaker specifications manually entered for the room. Earlier Dolby processors, such as the CP750, used a 31-band equailzer for the 5 or 7 main channels, and a single peaking filter for correcting the subwoofers' largest peak. The CP750 didn't have a swept sine wave generator, and used pink noise for measurement.
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 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.
Linear filters process time-varying input signals to produce output signals, subject to the constraint of linearity. In most cases these linear filters are also time invariant in which case they can be analyzed exactly using LTI system theory revealing their transfer functions in the frequency domain and their impulse responses in the time domain. Real-time implementations of such linear signal processing filters in the time domain are inevitably causal, an additional constraint on their transfer functions. An analog electronic circuit consisting only of linear components will necessarily fall in this category, as will comparable mechanical systems or digital signal processing systems containing only linear elements. Since linear time-invariant filters can be completely characterized by their response to sinusoids of different frequencies, they are sometimes known as frequency filters.
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.
In signal processing, a digital filter is a system that performs mathematical operations on a 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.
In signal processing, pre-emphasis is a technique to protect against anticipated noise. The idea is to boost the frequency range that is most susceptible to noise beforehand, so that after a noisy process more information can be recovered from that frequency range. Removal of the distortion caused by pre-emphasis is called de-emphasis, making the output accurately reproduce the original input.
In signal processing, group delay and phase delay are two related ways of describing how a signal's frequency components are delayed in time when passing through a linear time-invariant (LTI) system. Phase delay describes the time shift of a sinusoidal component. Group delay describes the time shift of the envelope of a wave packet, a "pack" or "group" of oscillations centered around one frequency that travel together, formed for instance by multiplying a sine wave by an envelope.
Audio crossovers are a type of electronic filter circuitry that splits an audio signal into two or more frequency ranges, so that the signals can be sent to loudspeaker drivers that are designed to operate within different frequency ranges. The crossover filters can be either active or passive. They are often described as two-way or three-way, which indicate, respectively, that the crossover splits a given signal into two frequency ranges or three frequency ranges. Crossovers are used in loudspeaker cabinets, power amplifiers in consumer electronics and pro audio and musical instrument amplifier products. For the latter two markets, crossovers are used in bass amplifiers, keyboard amplifiers, bass and keyboard speaker enclosures and sound reinforcement system equipment.
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 a sufficient degree to make it useful.
In signal processing and electronics, the frequency response of a system is the quantitative measure of the magnitude and phase of the output as a function of input frequency. The frequency response is widely used in the design and analysis of systems, such as audio and control systems, where they simplify mathematical analysis by converting governing differential equations into algebraic equations. In an audio system, it may be used to minimize audible distortion by designing components so that the overall response is as flat (uniform) as possible across the system's bandwidth. In control systems, such as a vehicle's cruise control, it may be used to assess system stability, often through the use of Bode plots. Systems with a specific frequency response can be designed using analog and digital filters.
Audio system measurements are a means of quantifying system performance. These measurements are made for several purposes. Designers take measurements so that they can specify the performance of a piece of equipment. Maintenance engineers make them to ensure equipment is still working to specification, or to ensure that the cumulative defects of an audio path are within limits considered acceptable. Audio system measurements often accommodate psychoacoustic principles to measure the system in a way that relates to human hearing.
In signal processing, a finite impulse response (FIR) filter is a filter whose impulse response is of finite duration, because it settles to zero in finite time. This is in contrast to infinite impulse response (IIR) filters, which may have internal feedback and may continue to respond indefinitely.
In signal processing and control theory, the impulse response, or impulse response function (IRF), of a dynamic system is its output when presented with a brief input signal, called an impulse (δ(t)). More generally, an impulse response is the reaction of any dynamic system in response to some external change. In both cases, the impulse response describes the reaction of the system as a function of time (or possibly as a function of some other independent variable that parameterizes the dynamic behavior of the system).
In signal processing, linear phase is a property of a filter where the phase response of the filter is a linear function of frequency. The result is that all frequency components of the input signal are shifted in time by the same constant amount, which is referred to as the group delay. Consequently, there is no phase distortion due to the time delay of frequencies relative to one another.
In audio signal processing, pre-echo, sometimes called a forward echo, is a digital audio compression artifact where a sound is heard before it occurs. It is most noticeable in impulsive sounds from percussion instruments such as castanets or cymbals.
First-order hold (FOH) is a mathematical model of the practical reconstruction of sampled signals that could be done by a conventional digital-to-analog converter (DAC) and an analog circuit called an integrator. For FOH, the signal is reconstructed as a piecewise linear approximation to the original signal that was sampled. A mathematical model such as FOH (or, more commonly, the zero-order hold) is necessary because, in the sampling and reconstruction theorem, a sequence of Dirac impulses, xs(t), representing the discrete samples, x(nT), is low-pass filtered to recover the original signal that was sampled, x(t). However, outputting a sequence of Dirac impulses is impractical. Devices can be implemented, using a conventional DAC and some linear analog circuitry, to reconstruct the piecewise linear output for either predictive or delayed FOH.
BRP-PACU is a dual channel FFT audio analysis tool. It is designed to be used with an omnidirectional calibrated microphone to configure any sound system with an appropriate equalization and delay. It compares the output of the system to the input of the system to obtain the transfer function of the system. These data allow one to perform final equalization using just the input/output of the DSP or any other device used for Equalization.
In signal processing, a filter is a device or process that removes some unwanted components or features from a 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 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.
Audyssey Laboratories, Inc. (Audyssey) is an American-based company specializing in technologies that address acoustical problems in sound reproduction systems used in homes, cars, studios, and movie theaters.
Smaart is a suite of audio and acoustical measurements and instrumentation software tools introduced in 1996 by JBL's professional audio division. It is designed to help the live sound engineer optimize sound reinforcement systems before public performance and actively monitor acoustical parameters in real time while an audio system is in use. Most earlier analysis systems required specific test signals sent through the sound system, ones that would be unpleasant for the audience to hear. Smaart is a source-independent analyzer and therefore will work effectively with a variety of test signals including speech or music.
Equalization, or simply EQ, in sound recording and reproduction is the process of adjusting the volume of different frequency bands within an audio signal. The circuit or equipment used to achieve this is called an equalizer.