Graphic Equalizer Research Articles

Implementation of Music Equalization Simulink Model on DSK6713

Audio equalization is a technique which consists of boosting or cutting certain frequency components of a given signal for sound quality enhancement. Equalizer is an electronic device or type of software that increases and decreases the power of sound waves. The paper deals with the analysis of music signals to develop insight on its frequency bands and auditory perception, through a case study of audio equalization. The graphic equalizer is designed through a Simulink model and is implemented in TMS320C6713 DSP board. This project is developed in order to deliver an insight on how audio equalization can be implemented with Matlab / Simulink for educational purposes.

Digital self-adapting graphic equalizer and method

A self-adaptive graphic equalizer 30 operable to equalize the effects of an audio system on an audio signal includes an adaptive graphic equalizer 34 having a plurality of equalizing filters 38, 40, where the plurality of equalizing filters have different center frequencies equidistant from one another and spanning a predetermined audio bandwidth. Each equalizing filter 38, 40 is operable to filter an ith sub-band of the audio signal. A plurality of first filters are coupled to the audio system, each first filter is operable to filter an ith sub-band of an output signal of the audio system. A plurality of second filters are operable to filter an ith sub-band of the audio signal. A gain adjuster 42 is operable to adjust the ith sub-band of the adaptive graphic equalizer in response to a difference in the ith sub-band of the filtered output signal from the plurality of first filters and the ith sub-band of the filtered audio signal from the plurality of second filters.

Automatic sound field correcting system and a sound field correcting method

A filter coefficient of a graphic equalizer GEQ is corrected based on detection results of reproduced sounds generated by supplying a noise to all frequency band loudspeakers 6 FL to 6 RR and a low frequency band exclusively reproducing loudspeaker 6 WF via the graphic equalizer GEQ. Then, attenuation factors of channel-to-channel attenuators ATG 1 to ATG 5 are corrected based on the detection results of the reproduced sounds generated by supplying the noise to the loudspeakers 6 FL to 6 RR via the graphic equalizer. Then, delay times of delay circuits DLY 1 to DLY k are corrected based on the detection results of the reproduced sounds generated by supplying the noise to the loudspeakers 6 FL to 6 WF via the graphic equalizer. Then, an attenuation factor of a channel-to-channel attenuator ATG k is corrected based on the detection results of the reproduced sounds generated by supplying the noise to the loudspeakers 6 FL to 6 RR via the graphic equalizer and the detection result of the reproduced sound generated by supplying the noise to the loudspeaker 6 WF via the graphic equalizer, whereby levels of the reproduced sounds reproduced by the loudspeakers 6 FL to 6 WF are adjusted to be made flat over the audio frequency band.

A cochlear implant user's guide to assistive devices and telephones

A cochlear implant user's guide to assistive devices and telephones

Circuit for providing visual indication of feedback

A compound differential circuit detects the highest relative signal of a plurality of input signals. Peak detector circuits including a sensing diode and a capacitor, are employed to produce a control signal. A switch receives the corresponding control signal. One switch is active at a time in response to the highest control signal. An LED connected to the activated switch is illuminated so that a visual indication of the highest input signal is displayed. In another embodiment, a feedback sensor for a graphic equalizer is disclosed.

A novel octave graphic equalizer

An octave constant Q graphic equalizer using a new class of active bandpass filters is presented. It covers all 11 standard octaves in a frequency range from 16 Hz to 20 kHz. The level in each octave can be changed within ±20 dB with respect to the reference level which corresponds to the zero dB gain of the equalizer. The advantages of the proposed bandpass biquads are the use of only one operational amplifier per octave, a very small resistors spread ratio, and equal values of the resistors in all octaves. The computer simulation has been used to verify the design of the novel circuit. The new design has been compared to the existing solutions reported in the literature. The comparison shows excellent agreement with the published results. The effects of the finite gain and bandwidth of the real operational amplifier have also been investigated. The proposed new equalizer shows to be robust against these amplifier imperfections.

High-speed CMOS continuous-time complex graphic equalizer for magnetic recording

A high-speed continuous-time CMOS analog adaptive equalizer for use in magnetic recording read channels is presented. The equalizer is implemented as the summation of several bandpass filters covering different frequency bands as in a graphic equalizer. The outputs from each filter are weighted by a complex coefficient and summed, which results in a linear combiner structure guaranteed to converge under least mean square (LMS) adaptation. System-level simulations of our "complex graphic equalizer (CGE)" show that its performance is comparable to that of a ten-tap finite impulse response (FIR) equalizer following a fourth-order low-pass filter when tested with two different sequence detectors: EPR4-MLSD and fixed delay tree search with decision feedback (FDTS/DF). A five-band tunable CGE has been fabricated using a 0.8-/spl mu/m CMOS technology. The highest band of the fabricated CGE was centered at 80 MHz (corresponding to channel data rate of about 200 Msymbols/s). Measured dynamic range was 68 dB, and measured total harmonic distortion was only -75 dB while consuming 97 mW at 3.3 V. The measured CGE performance agreed within 0.2 dB with the simulation results for an FDTS/DF system with an ideal CGE operating at 2.5 user bits/PW50.

Analog complex graphic equalizer for EPR4 channels

A continuous-time analog adaptive equalizer for use in EPR4 channels in magnetic recording is presented. The equalizer is implemented as the summation of several bandpass filters covering different frequency bands as in a graphic equalizer. The outputs from each filter are weighted by a complex coefficient and summed, which results in a linear combiner structure guaranteed to minimize MSE under LMS adaptation. System-level simulations of our 'Complex Graphic Equalizer (CGE)' shows that its performance is comparable to that of a 10-tap FIR equalizer on an EPR4 channel.

EIN: A Signal Processing Scratchpad

Computer systems designed for music synthesis usually encapsulate signal-processing algorithms as macros or functions, and thus provide a modular interface which facilitates the development of complex structures. This is the thinking behind Mathews’ original concept of the ‘‘unit generator’’ (Mathews 1969) and most subsequent software synthesis languages use this approach. While it has proven to be an effective method it has not generally provided a means by which the users of these systems, often more musically prepared than wise in the ways of digital signal processing, can gain an intuitive understanding of the mechanisms used to modify and create digital signals. Indeed, there is little conceptual difference between patching a signal through a bank of two-pole resonating filters/unit generators and tweaking the sliders on a graphic equalizer. EIN is an attempt to provide an interface in which the user has direct control over every add, multiply and store applied to each sample, and can gain a more direct understanding of the machinery of digital signal processing. While its main use has been instructional, it also provides a way to experiment with digital filters and design more complex instruments and algorithms. It is, in effect, a kind of low-level circuit design kit for signal processing. EIN Syntax EIN provides the routine machinery for calling the user’s code, executing it, writing the resulting output sound samples to a file, playing the file, and displaying and analyzing it in the time and frequency domains. The user provides a script in a language that is a superset of C. The EIN system then compiles it and provides a wrapper that calls the script for each time sample from t = 0 up to the specifiednumber of samples, nsamps; computes the next output sample y (for the mono case), or the variables left and right (for the stereo case); and writes the output samples to an output sound file,formated for the sampling rate of sr samples/sec. The int variables t and nsamps, and the float variables y, left, right, and sr, are reserved by EIN, and should not be used for other purposes. The mono/stereo option, and the values of sr and nsamps are selected with radio buttons on the interface. The remaining reserved names are described in Appendix 3. As an example, here is a one-line EIN script that produces a sine wave at 440 Hz:

Adaptive continuous-time equalization and FDTS/DF sequence detection

An adaptive analog continuous-time forward equalizer and a Fixed Delay Tree Search with Decision Feedback (FDTS/DF) detector can be used to implement high speed low power sequence detection for a magnetic recording system. An analog continuous-time forward equalizer offers a low power, high speed solution to the equalization problem, although it cannot achieve the degree of pulse shape control provided by a long FIR filter. However, an FDTS/DF detector can be easily adapted to cope with the resulting imprecisely equalized pulses. Earlier work demonstrated the suitability of a Complex Graphic Equalizer (CGE) for the adaptive continuous time forward equalizer. System simulations indicated the viability of a CGE-FDTS/DF detector in a 2/3(1,7) coded channel. New results explore the performance of a CGE-FDTS/DF system for an 8/9(0,3) coded channel. Additionally, the feasibility of a CGE-EPR4 detection system is demonstrated.

Audio apparatus having electronic graphic equalizer

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A 12-band microprocessor controlled graphic equalizer display filter IC

A 12-band microprocessor-controlled graphic equalizer display filter IC, designed for use in audio applications, is described. The device is implemented in a 2-micron CMOS process and utilizes switched-capacitor technology to replace several discrete components (op amps, resistors, capacitors, and diodes) and multiplexers. The chip occupies 18000 square mils and dissipates 100 mW.< <ETX xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">&gt;</ETX>

Method and apparatus for measuring and correcting acoustic characteristic in sound field

The acoustic transmission characteristics of a sound field, such as in the passenger compartment of a vehicle, are measured by placing microphones (1-3L,1-3R) at the positions of ears of a dummy mannequin (1-4) having a sound absorbing characteristics similar to that of a clothed adult human. The transmission characteristics are then corrected, such as with an automatic graphic equalizer (3-2), to provide optimized listening conditions inside the sound field for an actual listener. The use of the invention results in a listener's perception of completely natural sound.

A 7-point stereo graphic equalizer display filter IC

A seven-point graphic equalizer display filter IC for use in audio applications is described. The device is implemented in a three-micron CMOS process, and utilizes switched capacitor circuit technology to replace several discrete components (opamps, diodes, resistors, capacitors) which occupy significant board space. The chip occupies 17000 square mils and dissipates about 120 mW. Typical performance measurements of the device are given. >

A digital signal processor for digital audio use

The authors describe the D/sup 2/SP digital signal processor which is designed with a dual processor architecture, and is especially effective for stereo audio signal processing. The D/sup 2/SP has two independent processor units, the left processor and right processor. The D/sup 2/SP contains an audio data interface, external dynamic random-access memory (DRAM) interface, microcomputer interface, and a sequence controller. The single chip D/sup 2/SP is capable of performing audio signal processing needed for graphic equalizers, surround, and sound power spectrum calculation. The dual processor mainly consists of a 24-bit multiplier with a 108-ns cycle time and is assembled on a single chip fabricated from 1.2- mu m CMOS technology. The total number of integrated transistors is approximately 300 K. Applications, such as use as a seven-band graphic equalizer, seven-band power spectrum calculator, and surround sound processing, are briefly discussed. >

Frequency response displays for use with a graphic equalizer

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A high-quality digital audio filter set designed by silicon compiler CATHEDRAL-1

The semi-automatic design and custom integration of a high-quality digital audio preamplifier filter set are described. The set consists of an offset filter, ten graphic equalizer sections, and a scratch filter with a worst-case overall signal-to-noise ratio of 10 dB. The silicon compiler CATHEDRAL-1 supports the design from specifications to layout. The combination of efficient synthesis tools with optimisation at all design levels leads to a very compact silicon integration, compared with a general-purpose signal processor approach. An experimental chip is described on which the offset filter, three equalizer sections, and the scratch filter are integrated. The silicon area is 243 mm/SUP 2/ in a conservative 6-/spl mu/m NMOS technology or 8 mm/SUP 2/ when scaling down to 3-/spl mu/m technology, allowing for functional densities up to 0.2 mm/SUP 2/ pole zero, comparable with area requirements for typical switched-capacitor filters. The total filter set requires only 20 mm/SUP 2/ in a 3-/spl mu/m NMOS technology, which demonstrates the potential for low-cost digital high-fidelity signal processing.

Sound Reinforcement for a Touring Musical Show

A sound-reinforcement system for a touring musical show was designed using an adjustable central theater-type loudspeaker cluster, supplementary time-delayed distributed loudspeakers, a frequency shifter, a custom-designed control console with vertical attenuators, a graphic equalizer, and directional condenser microphones. The objective was to provide speech reinforcement and reproduction of sound effects equal to those provided by state-of-the-art permanent systems. This paper discusses the system design, plus installation and operational problems and results in seven large auditoriums having varied configurations and acoustical environments.