Interactive graphic equalizer

2019

This paper outlines the design of a high-precision graphic audio equalizer with digital filters in parallel, along with its implementation in MATLAB App Designer. The equalizer is comprised of 31 b ...

2015

This paper describes the implementation of a recently introduced parallel graphic equalizer (PGE) in a heterogeneous way. The control and audio signal processing parts of the PGE are distributed to a PC and to a signal processor, of WaveCore architecture, respectively. This arrangement is particularly suited to the algorithm in question, benefiting from the low-latency characteristics of the audio signal processor as well as general purpose computing power for the more demanding filter coefficient computation. The design is achieved cleanly in a high-level language called Kronos, which we have adapted for the purposes of heterogeneous code generation from a uniform program source.

A straight-forward design of graphic equalizers with minimumphase behavior based on recently developed higher-order bandshelving filters is presented. Due to the high filter order, the gain in one band is almost completely independent from the gain in the other bands. Although no special care will be taken to design filters with complementary edges except for a suitable definition of the cut-off frequencies, the resulting amplitude deviation in the transitional region between the bands will be sufficiently low for many applications.

2000

This paper describes the design and implementation of an FIR based 20-band stereo audio equalizer that offers good spectral selectivity and phase response. The underlying theory of fast filtering in the frequency domain is addressed, the architecture of the DSP system using a single TI TMS320C31 processor is presented, the functionality of the interface, either software-based or hardware-based is described, and a few test results are presented and discussed.

IEEE Signal Processing Letters, 2017

A graphic equalizer is a high-order filter controlling the gain of several frequency bands. For good accuracy, graphic equalizers consisting of cascaded IIR filters have been of very high order. A previously proposed parallel graphic equalizer entailing twice as many second-order filter sections as there are bands can have a maximum approximation error of less than 1 dB, but its design is complicated. This letter proposes a cascade graphic equalizer having an accuracy comparable to the best parallel graphic equalizer, although only one second-order section is assigned per command gain. A key idea is to use band filters whose interaction with the two neighboring filters at their center frequency is exactly controlled. The filter gains are obtained using the least-squares method with one iteration step, which involves linear interpolation of the target gain vector, inversion of a square matrix, and a few matrix multiplications. The proposed method is compared with previous designs and is shown to be the most accurate one. The new graphic equalizer is widely useful for audio and music processing.

2016 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP), 2016

A graphic equalizer is an adjustable filter in which the command gain of each frequency band is practically independent of the gains of other bands. Designing a graphic equalizer with a high precision requires evaluating a target response that interpolates the magnitude response at several frequency points between the command gains. Good accuracy has been previously achieved by using polynomial interpolation methods such as cubic Hermite or spline interpolation. However, these methods require large computational resources, which is a limitation in real-time applications. This paper proposes an efficient way of computing the target response without sacrificing the approximation accuracy. This new approach called Linear Interpolation with Constant Segments (LICS) reduces the computing time of the target response by 55% and has an intrinsic parallel structure. Performance of the LICS method is assessed on an ARM Cortex-A7 core, which is commonly used in embedded systems.

Parametric equalizers alter audio frequency response which is defined by centre frequency, Q-factor and maximum Gain. In this paper, we present the design of a 6 band parametric equalizer by cascading of shelving and peaking filters. The main characteristic of the filters is to amplify or attenuate certain frequency band by the required gain while leaving the other frequency bands unaltered. The results of the equalization were verified using psychoacoustic test and graphical interpretation.

1994

Filters arecommonplace elements in audiosignalprocessing. Most well-known filter topologies generate fixed responses with fLxedelement values, oruse variableresistors to altertheirresponses underuser control.This paper explores solutions tovarying the responseof filters and equalizersusing digitalcontrol techniques suitable for computer-controlled systems. High-pass, low-pass, band-pass, and parametric topologies are covered using VCAs, resistor ladders, and multiplying DACs as controlelements.

For some years, real-time general-purpose digital audio systems, based around specialized hardware, have been used by composers and researchers in the field of electronic music, and by professionals in various audio-related fields. During the past decade, these systems have gradually replaced many of the audio-processing devices used in amateur and professional configurations. Today, with the significant increase in computing power available on the desktop, the audio community is witnessing an important shift away from these systems, which required specialized hardware, towards general purpose desktop computing systems featuring high-level digital signal processing (DSP) software programming environments. Introduction An new real-time DSP programming environment called Max Signal Processing (MSP) [Zicarelli, 1997], was released this past year for the Apple Macintosh PowerPC platform. This software offers a suite of signal processing objects as an extension to the widely used MAX software environment, and provides new opportunities for musicians and engineers wishing to explore professional-quality real-time DSP. Most important, MSP provides a number of frequency-domain processing primitives that allow for the development of sophisticated frequency-domain signal processing applications.

This paper is an implementation of FPGA interfaced with different devices. The project focuses on two protocols for establishing communication between them: the RS232 and parallel port. Here, a Digital to Analog Converter was interfaced with an FPGA, which in turn was connected to a computer. In this system, one can enter the expected voltage level as numerical input, along with the desired channel the voltage is expected on, when prompted on the computer screen, using the Hyper Terminal software. The system will then pass these values via RS232 to the FPGA. The FPGA is programmed in Verilog HDL. It selects DAC channel and gives appropriate output.