Abstract: A method for improving the effective signal-to-noise ratio (“SNR”) of an analog to digital converter (“ADC”) for active loudspeakers uses the two available channels of a stereo ADC to separately process the low- and high-frequency components of an audio signal. Because the power spectral density of music approximates a pink noise spectrum, the high-frequency component of the signal has peak levels low enough to avoid exceeding the maximum ADC input level. The audio signal is analog high-pass filtered and the resulting high-frequency signal component is sent directly to a first ADC channel without attenuation. The remaining low-frequency component is attenuated and sent to a second ADC channel. The digital signals are processed, converted back to analog, amplified, and reproduced by loudspeaker drivers. Noise and distortion at low frequencies is less audible than higher frequencies, so the improved SNR at higher frequencies yields a significant practical improvement in audio fidelity.

Abstract: Systems and methods are generally described for generation and transmission of musical performance data. In some examples, the musical input device may generate a first command encoding a first musical event. In various further examples, the musical input device may generate a first message corresponding to the first command, the first message encoding a first acoustic attribute type of the first musical event. In some examples, the musical input device may generate a second message corresponding to the first command, the second message encoding a second acoustic attribute type of the first musical event. In various examples, the musical input device may generate timestamp data denoting a time of an occurrence of the first musical event. In some examples, the musical input device may send the timestamp data, the first command, the first message and the second message to a computing device.

Abstract: Methods and apparatuses according the present embodiments derive the sound of a Hammond tonewheel organ from the equal-tempered tuning of its tonewheels and drawbar registration design, as well as its vibrato/chorus processing and pickup distortion. In embodiments, as a reverberation effect, the modal processor simulates a room response as the sum of resonant filter responses, providing precise, independent and interactive control over the frequency, damping, and complex amplitude of each mode. As an effects processor, the modal processor provides pitch shifting and distortion by simple manipulations of the mode output sinusoids.

Abstract: A reduced power consumption actuator drive circuit that includes separate circuit power paths for different portions of the signal spectrum for applications in which lower frequencies have high amplitudes. The low frequency circuit paths use higher power supply voltages at lower currents and the high frequency circuit paths use lower power supply voltages at higher currents. In one embodiment, the drive circuit drives a nutator that employs a resonating circuit that maintains actuator motion with reduced energy supplied by the power supply.

Abstract: An electric guitar includes a neck and a body connected with the neck. The neck includes a mounting plate, a plurality of touch keys and a display screen. The touch keys and the display screen are mounted on the mounting plate. The touch keys are positioned at an upper edge or lower edge of the display screen. The body includes an outer shell, a human-machine interface, a control circuit board, a speaker and a power source. The human-machine interface is mounted on an upper surface of the outer shell. The control circuit board, the speaker and the power source are mounted in the outer shell; the touch keys, the display screen, the speaker and the human-machine interface are electrically connected with the control circuit board. The mounting plate is positioned at a right side surface of the outer shell.

Abstract: A tone control for string instruments includes a pair of potentiometers each coupled in series relationship with a respective filter capacitor. The pair of potentiometers are mechanically coupled one to the other for concurrent mechanical travel of respective displaceable contacts thereof to provide selective filtering of one or more pick-up sensors of the instrument. Responsive to selective positioning of the displaceable contacts of the pair of potentiometers, signals input thereto are high pass filtered, low pass filtered or unfiltered. The tone control can serve as a master tone control where separate tone controls are connected to series coupled pairs of pickup coils and coupled to the master tone control through a blend configuration control.

Abstract: An electronic signal processor for processing signals includes a complex first filter, one or more gain stages and a second filter. The first filter is characterized by a frequency response curve that includes multiple corner frequencies, with some corner frequencies being user selectable. The first filter also has at least two user-preset gain levels which may be alternately selected by a switch. Lower frequency signals are processed by the first filter with at least 12 db/octave slope, and preferably with 18 db/octave slope to minimize intermodulation distortion products by subsequent amplification in the gain stages. A second filter provides further filtering and amplitude control. The signal processor is particularly suited for processing audio frequency signals.

Abstract: A signal processing device is composed of a signal acquisition unit and a signal processing unit. The signal acquisition unit acquires a signal corresponding to a vibration propagated from a string attached to a stringed instrument from a pickup element that picks up the signal corresponding to the vibration. The signal processing unit includes a filter that performs convolution operation using a filter coefficient set in the filter, the signal processing unit applying the convolution operation to the acquired signal through the filter and outputting a processed signal.

Abstract: An electronic signal processor for processing signals includes a complex first filter, one or more gain stages and a second filter. The first filter is characterized by a frequency response curve that includes multiple corner frequencies, with some corner frequencies being user selectable. The first filter also has at least two user-preset gain levels which may be alternately selected by a switch. Lower frequency signals are processed by the first filter with at least 12 db/octave slope, and preferably with 18 db/octave slope to minimize intermodulation distortion products by subsequent amplification in the gain stages. A second filter provides further filtering and amplitude control. The signal processor is particularly suited for processing audio frequency signals.

Abstract: A signal processing apparatus 30 includes a FIR filter 33 that generates a signal corresponding to an instrument sound including a resonance of a body caused by a vibration of a string, based upon a pickup signal from a pickup sensor 21. The signal processing apparatus 30 guides the pickup signal from the pickup sensor 21 to an adder circuit 35 via a low-pass filter 32, guides the signal generated by the FIR filter 33 to the adder circuit 35 via a high-pass filter, and adds and mixes these signals in the adder circuit 35. The cut-off frequency of the high-pass filter 34 is equal to or higher than the cut-off frequency of the low-pass filter 32.

Abstract: Disclosed is a stringed instrument that includes a plurality of strings and a pickup to which each of the plurality of strings is respectively coupled that is connectable to a computer to implement DSP modeling. A serial interface circuit is coupled to the pickup and to a digital connector that formats each digital string vibration signal received from the pickup into a digital serial protocol. A computer is coupled by a serial link to the digital connector such that the computer receives each serially formatted digital string signal (SFDSS). The computer operates at least one audio DSP-based software module to process each received SFDSS wherein each SFDSS is processed in order to emulate a corresponding string tone of one of a plurality of stringed instruments to create an emulated digital string tone signal (EDSTS). Each EDSTS is then transmitted back over the serial link to the stringed instrument for playback.

Abstract: A signal processing apparatus 30 includes a FIR filter 33 that generates a signal corresponding to an instrument sound including a resonance of a body caused by a vibration of a string, based upon a pickup signal from a pickup sensor 21. The signal processing apparatus 30 guides the pickup signal from the pickup sensor 21 to an adder circuit 35 via a low-pass filter 32, guides the signal generated by the FIR filter 33 to the adder circuit 35 via a high-pass filter, and adds and mixes these signals in the adder circuit 35. The cut-off frequency of the high-pass filter 34 is equal to or higher than the cut-off frequency of the low-pass filter 32.

Abstract: From analog music information in which the sounds of a plurality of musical instruments are mixed, sound data corresponding to the register of the reproduced sound of a bass guitar and sound data corresponding to the register of the reproduced sound of a drum are extracted using a band-pass filter. A drive pulse with a low frequency is generated within the periods of data sections in which the former sound data reaches a predetermined level or higher, and a drive pulse with a high frequency is generated within the periods of data sections in which the latter sound data reaches a predetermined level or higher. A vibrating body in a vibration mechanism unit is resonated by two drive pulses with frequencies, thereby causing vibration according to the reproduced sound of music.

Abstract: An electronic signal processor for processing signals includes a complex first filter, one or more gain stages and a second filter. The first filter is characterized by a frequency response curve that includes multiple corner frequencies, with some corner frequencies being user selectable. The first filter also has at least two user-preset gain levels which may be alternately selected by a switch. Lower frequency signals are processed by the first filter with at least 12 db/octave slope, and preferably with 18 db/octave slope to minimize intermodulation distortion products by subsequent amplification in the gain stages. A second filter provides further filtering and amplitude control. The signal processor is particularly suited for processing audio frequency signals.

Abstract: An electronic signal processor for processing signals includes a complex first filter, one or more gain stages and a second filter. The first filter is characterized by a frequency response curve that includes multiple corner frequencies, with some corner frequencies being user selectable. The first filter also has at least two user-preset gain levels which may be alternately selected by a switch. Lower frequency signals are processed by the first filter with at least 12 db/octave slope, and preferably with 18 db/octave slope to minimize intermodulation distortion products by subsequent amplification in the gain stages. A second filter provides further filtering and amplitude control. The signal processor is particularly suited for processing audio frequency signals.

Abstract: A signal processing device is composed of a signal acquisition unit and a signal processing unit. The signal acquisition unit acquires a signal corresponding to a vibration propagated from a string attached to a stringed instrument from a pickup element that picks up the signal corresponding to the vibration. The signal processing unit includes a filter that performs convolution operation using a filter coefficient set in the filter, the signal processing unit applying the convolution operation to the acquired signal through the filter and outputting a processed signal.

Abstract: An electroacoustic transducer system for converting sound waves propagating from a musical instrument drum to an electric signal is generally provided. The system includes a housing, an electroacoustic transducer, and a cover. The housing includes a concave inner surface and a convex outer surface. In addition, the housing defines a housing cavity with a mouth. The electroacoustic transducer is disposed in the housing cavity and faces outwardly from the concave inner surface and towards the mouth to receive sound waves propagating from the drum. The electroacoustic transducer receives and converts the sounds waves in the housing cavity to the electric signal. The cover has a generally planar outer surface and extends over the mouth to at least partially enclose the electroacoustic transducer.

Abstract: A system for control signal generation using detected dynamic characteristics of components of an incoming electronic signal. Fixed or adjustable bandpass filters are coupled to signal parameter measurement elements. Each filter isolates a particular overtone component from the incoming electronic signal for isolated signal parameter measurement. Pitch and amplitude of a plurality of overtones may be individually measured. A control signal processor generates one or more outgoing control signals based upon one or more isolated overtone parameter signals. The control signal processor may use these to generate individual associated output signals, subject to optional mathematical operations and warpings, or combine measured parameters of several overtones via mathematical operations to other output signals.

Abstract: An electronic signal processor for processing signals includes a complex first filter, one or more gain stages and a second filter. The first filter is characterized by a frequency response curve that includes multiple corner frequencies, with some corner frequencies being user selectable. The first filter also has at least two user-preset gain levels which may be alternately selected by a switch. Lower frequency signals are processed by the first filter with at least 12 db/octave slope, and preferably with 18 db/octave slope to minimize intermodulation distortion products by subsequent amplification in the gain stages. A second filter provides further filtering and amplitude control. The signal processor is particularly suited for processing audio frequency signals.

Abstract: Methods and apparatus for simulating the sound of an acoustic percussion instrument. A first stored waveform signal representative of the impulse response of an acoustic percussion instrument is convolved with a waveform produced by a sensor circuit attached to a physical playing surface. Undesirable response characteristics of the playing surface and sensor circuit may be filtered out by deconvolving sensor output, or the stored waveform representing the acoustic instrument, with the impulse response of the combination of the playing surface and the sensor circuit.

Abstract: An audio effects control for and method of controlling the application of special audio effects applied to an audio signal, comprises a sensor configured to sense movement associated with the generation of the audio signal, wherein the sensor produces a control signal in response to detecting the movement, and the control signal is transmitted to an audio effects unit to control application of an audio effect on an audio signal.

Abstract: An amplifier for electric guitar including a plurality of foot switch selectable preamplifier channels which can be set to provide differing sonic characteristics, and also including a power amplifier section for driving a loudspeaker that can be switched between at least two different power or performance capabilities. User operable switching dedicated to at least one of the preamplifier channels is provided to select preferred power amplifier parameters in conjunction with differing channel voicings. Thus, differing preferred power amplifier characteristics may be pre-set and assigned to individual preamplifier channels and called up automatically in conjunction with the remote channel selection.

Abstract: An electronic signal processor for processing signals includes a complex first filter, one or more gain stages and a second filter. The first filter is characterized by a frequency response curve that includes multiple corner frequencies, with some corner frequencies being user selectable. The first filter also has at least two user-preset gain levels which may be alternately selected by a switch. Lower frequency signals are processed by the first filter with at least 12 db/octave slope, and preferably with 18 db/octave slope to minimize intermodulation distortion products by subsequent amplification in the gain stages. A second filter provides further filtering and amplitude control. The signal processor is particularly suited for processing audio frequency signals.

Abstract: Disclosed is a stringed instrument with embedded DSP modeling capabilities to model an acoustic stringed instrument. The stringed instrument has a body and a plurality of strings and each of the plurality of strings is respectively coupled to a pickup to detect a vibration signal for each string. An A/D converter converts the detected vibration signal of a string into a digital string vibration signal. A DSP is located within the body of the stringed instrument to process the digital string vibration signal and to implement an acoustic modeling system to process the digital string vibration signal in order to emulate a corresponding string tone of one of a plurality of selectable acoustic stringed instruments. Acoustic modeling includes acoustic string and body modeling, microphone placement modeling, and pick-sound modeling. The emulated acoustic digital tone signal is then converted to analog form for output to an amplification device.

Abstract: A device for mixing the outputs of two sensors. The device includes a first input for receiving a signal from at least one first sensor, a second input for receiving a signal from at least one second sensor, a low pass filter for passing signal components of the first input signal below a first frequency, a high pass filter for passing signal components of the second input signal above a second frequency and a mixing circuit for combining the signals passed by the low pass filter and the high pass filter to form a combined output signal. The mixing device may be incorporated into a pre-amplifier for use with an electrified acoustic guitar, or other acoustic stringed instrument, having an under saddle sensor and a second sensor attached to a body portion of the guitar, such as the soundboard. In this way, lower frequencies are taken from the under saddle sensor and higher frequencies are taken from the body/soundboard sensor.

Abstract: A selector switch for musical instruments, such as electric guitars. is provided having a single, manually operated toggle member adapted to perform greater functionality. A selector switch is disclosed which indexes longitudinally for the electrical connection of pickup(s) for resultant amplification but will additionally provide further associated connectivity by means of a transverse indexing motion thus availing expanded switching function from a single switch apparatus. This inventive step is referred to as a compound selector switch.

Abstract: A method is provided for designing an acoustic correction filter applicable to a stringed instrument, which is composed of a string member operable to undergo a vibration, a support member for supporting the string member, a body member responsive to the vibration transmitted through the support member for generating a natural sound and a mute attachment for muting the natural sound. The acoustic correction filter is operable when the natural sound is muted by the mute attachment for filtering a signal derived from the vibration so as to create an artificial sound instead of the muted natural sound.

Abstract: An electric violin has a combination of two sorts of vibration detectors for transverse vibration components and vertical vibration components, a low-pass filter connected to the vibration detector for the transverse vibration components and a high-pass filter connected to the vibration detector for the vertical vibration components, piezoelectric transducers connected in parallel to a signal processing system, or vibration detectors connected to filter circuits different in frequency characteristics for producing right-channel and left-channel signals so that the electric violin generate electric tones close to acoustic violin tones.

Abstract: A method is provided for designing an acoustic correction filter applicable to a stringed instrument, which is composed of a string member operable to undergo a vibration, a support member for supporting the string member, a body member responsive to the vibration transmitted through the support member for generating a natural sound and a mute attachment for muting the natural sound. The acoustic correction filter is operable when the natural sound is muted by the mute attachment for filtering a signal derived from the vibration so as to create an artificial sound instead of the muted natural sound.

Abstract: An electric violin has a combination of two sorts of vibration detectors for transverse vibration components and vertical vibration components, a low-pass filter connected to the vibration detector for the transverse vibration components and a high-pass filter connected to the vibration detector for the vertical vibration components, piezoelectric transducers connected in parallel to a signal processing system, or vibration detectors connected to filter circuits different in frequency characteristics for producing right-channel and left-channel signals so that the electric violin generate electric tones close to acoustic violin tones.

Abstract: A system and a method of measuring, decomposing, processing and uniquely recombining forces and vibrations acting on stringed musical instruments (SMI). The system utilizes a digital signal processor and reproduces the musical sound characteristics of an acoustic instrument into high fidelity electrical signals for amplification, processing and/or filtering and reproduction of musical sounds by uniquely exploiting, through measurements and subsequent signal processing, the vector nature of string excitation forces (SEF) and body vibrations of stringed musical instruments. A signal processing system of the current invention also utilizes a plurality of sensors, each responsive to at least one of force, displacement, velocity or acceleration indicative of the vibrational energy of the strings, to produce a sensor signal vector, which is then processed and transformed by a plurality of re-creation filters into a transformed signal vector, and then resynthesized into an output signal.

Abstract: A combination of limited data, 3D linear interpolation, and a unique simulation circuit are combined to provide a programmable tone control that can simulate any desired tone stack. In accordance with one aspect of the invention, it has been found that a reduced number of data points plus interpolation provides sufficiently accurate coefficients for simulation. The data points are obtained from measurements of the actual operation of the analog tone stacks to be simulated. A second aspect of the invention is the simulation circuit controlled by the coefficients to accurately reproduce the operation of any given tone stack. The simulation circuit uses two programmable filters and three variable gain amplifiers to simulate a tone stack. The cut-off frequency of each filter and the gain of each amplifier are determined by coefficients extracted by interpolation from stored data.

Abstract: A signal processing circuit, for each string of a musical instrument which string is individually tuned to a frequency, includes a transducer adjacent to the string and an equalizer connected to an output circuit. To emulate an acoustic instrument, the equalizer is set to emphasize a signal present in a frequency range of the tuned frequency of the string and/or its harmonics. It also de-emphasizes at least low end frequencies below the operating range of the string. A mid-range frequency device is provided in the equalizer to produce characteristic mid-range response for a specific acoustic instrument. If the transducer produces an inherent resonance at a characteristic frequency, the equalizer will also de-emphasize the inherent resonance produced by the transducer at the characteristic frequency of the transducer.

Abstract: A music synthesizer has one or more sensors that generate a respective plurality of sensor signals, at least one of which is an audio frequency sensor signal. Electronic circuitry, such as a specialized circuit or a programmed digital signal processor or other microprocessor, implements a physical model. The electronic circuitry includes an excitation signal input port for continuously receiving the audio frequency sensor signal as well as a control signal port for continuously receiving a control signal corresponding to the audio frequency sensor signal. The-control signal can have much lower bandwidth than the audio frequency sensor signal.

Abstract: The signal from a piezo-electric transducer in an electro-acoustic guitar is modified by a variable depth notch filter. The notch filter includes two paths between an input and a summation circuit. The signals in the two paths are 180.degree. out of phase and one of the paths includes a bandpass filter. The notch is located at approximately 5 khz. The depth of the notch depends upon either the broadband amplitude or the narrowband amplitude of the signal from the transducer and the depth of the notch is controlled using either feedback or feedforward control.

Abstract: A sound monitoring system includes a resonator platform comprised of musician support plate, a base plate, and at least one transducer. The musician support plate acts as a surface upon which the user may stand or sit. In response to high-frequency filtered and amplified electrical signals from a sound source, the transducer of the resonator platform imparts vibrational energy to the musician support plate, which in turn acts as a diaphragm to impart low frequency sounds to the musician by touch without generating appreciable audible energy. The base and musician support plates may have various geometric shapes and sizes, and are composed of materials suitable for transmitting vibrational energy.

Abstract: The device is designed to receive at its input the analog AC output signal of an electric guitar either directly from the guitar, its amplifier or a speaker. The incoming signal is fed to three series connected circuits which successively boost the bass and treble frequencies, and then reduce the signal level between the bass and treble peaks before applying the processed signal to an output circuit. A high frequency adder circuit connected in parallel with the three series connected circuits permits frequencies above a certain level (10 KHz) to be added back to the processed signal at the output circuit. One switch is operable in the bass boost circuit to introduce different resistance selectively to raise or lower the frequency center and gain of the circuit, and another switch at the input of the device is operable to add resistance when the input signal is from a speaker.

Abstract: A circuit for multiplexing three or more audio band electrical signals into two audio band electrical signals which can be provided to a stereo input and a circuit and computer method for demultiplexing the signals. In particular, six signals from the six strings of a guitar are multiplexed for stereo audio analog input to a sound card in a personal computer where the signals are digitized and then, using software running on the computer, demultiplexed into six signals from which the frequencies above 4.5 kHz have been lost.

Abstract: A musical tone generating apparatus is provided for a keyboard electronic musical instrument. The apparatus consists of a musical tone source, a coefficient generator, a digital filter and so on. The coefficient generator generates at least one time dependence coefficient, and applies It to output of the digital filter. The parameter controller changes at least one parameter, which designates tone color of the musical tone, in accordance with the time dependent coefficient to thereby realize time-variant changing rate of envelope of musical tone. And, the coefficient generator and the parameter controller are used by means of time sharing technique. That is, the coefficient is used as plural units, wherein each unit generates a coefficient in each stage, and the digital filter is also used as plural units. Each digital filter unit changes a parameter of tone color of the musical tone in accordance with the coefficient supplied thereto.

Abstract: A tone controller includes: a string counterpart member; a bow; a friction sound detector for detecting a friction sound generated by rubbing together the bow and the string counterpart member; and a signal processor for generating a tone control signal representing a bow pressure and a bow speed from an output of the friction sound detector.

Abstract: An emphasize system for an electronic musical instrument including filter and control device. The filter extract and output predetermined frequency range component of an input musical tone signal. The control device control the level of an output from the filter at least on the basis of an output from an envelope generator which controls the level of the input musical tone signal, or on the basis of the level of another frequency range component of the input musical tone signal. At least the level controlled signal is outputted from the system.

Abstract: A musical tone signal forming device for an electric string music instrument having at least one pickup for detecting and delivering vibration of a string as an electric musical sound signal, including at least one comb filter having the electric musical sound signal as an input thereof; and structure for producing parameters corresponding to a specific position, which differs from a position of the pickup with respect to the string, at which an imaginary pickup is disposed and a reference vibration frequency of the string and supplying the parameters to the at least one comb filter to determine a frequency characteristic of the at least one comb filter, wherein the frequency characteristic of the at least one comb filter simulates a frequency characteristic of the electric musical sound signal delivered from the imaginary pickup disposed at the specific position.

Abstract: A musical tone synthesizing apparatus conducts musical tone synthesis based on the tone generation mechanism of an acoustic musical instrument, and generates quickly and accurately desired musical tones in response to the operation of the beginning of tone generation. The musical tone synthesizing apparatus includes a performance data generation mechanism which generates performance data in accordance with performance operations, an excitation circuit, which generates an excitation signal in correspondence with the performance data, and a signal loop circuit which delays the excitation signal by a fixed period and repeatedly cycles the excitation signal. The performance data generation mechanism, furthermore, supplies an initial excitation control signal, which is for the purpose of exciting the signal loop circuit, to the excitation circuit during the initial generation of a musical tone.

Abstract: A preconditioned signal that is applied to the input of an audio power amplifier for providing a perceived control of reactance simulation (or the Damping Factor) of the amplifier. A low frequency fundamental resonance is combined with a slowly rising high frequency preemphasis to mimic or simulate an audible interaction with the impedance plot of the loudspeaker, without increasing the output impedance of the amplifier.

Abstract: An apparatus for synthesizing musical tones is provided with an excitation device and time delay device so as to form a loop circuit, and also, a noise generation device for generating a blowing noise signal when blowing a wind instrument, and noise mixing device for mixing the blowing noise signal with excitation signal corresponding to musical information output from the wind instrument are incorporated in the loop circuit, thereby the excitation signal is circulated in the loop circuit and mixed with the blowing noise signal.

Abstract: A guitar control system using multiple string processing channels for developing pitch and peak signals for a multiple of string vibrations. Pitch data is multiplexed into a microprocessor along with the peak data and the programmed microprocessor outputs processed sound information on an address/data buss. Midi input/output, analog input/output and counter circuitry are interactively connected with the data buss, and final audio processing is derived from the analog output.

Abstract: A guitar control system using multiple string processing channels for developing pitch and peak signals for a multiple of string vibrations. Pitch data is multiplexed into a microprocessor along with the peak data and the programmed microprocessor outputs processed sound information on an address/data buss. Midi input/output, analog input/output and counter circuitry are interactively connected with the data buss, and final audio processing is derived from the analog output.

Abstract: In an apparatus for modulation of the timbre of an electrical amplifier, in particular for guitars, a cylindric coil (2), which can be disposed downstream of an amplifier output, and a bar magnet (3) inside the cylindric coil (2), which can be deflected against restoring forces within a limited amplitude range out of its position of equilibrium, are provided for achieving a tonal effect corresponding to the direct connection of a loudspeaker.

Abstract: Waveform data is read out from a memory which stores the waveform data at a rate corresponding to a designated pitch. The readout data is supplied to digital filters, and two or more different filtering operations are executed in the digital filters. Two or more waveform data obtained as a filtering result are synthesized at a rate determined in response to a breath input signal output from a breath sensor, and the synthesized data is output as a musical tone waveform signal. The filter performs the filtering operation while a cut-off frequency is changed in response to the breath input signal.

Abstract: The tone signal from a transducer monitoring the music vibrations of a bowed musical instrument contains an objectionable friction noise component which is removed by passing the tone signal through a high-pass filter having a variable cutoff frequency which is controlled by a pitch detector monitoring the fundamental frequency of a played note in the tone signal. The cutoff frequency of the filter is kept slightly below the fundamental frequency of the played note to avoid altering the tone color of the tone signal. When no fundamental frequency is detected by the pitch detector, the cutoff frequency of the filter remains at a pre-determined minimum which may be adjustable by the musician for best effect when strings are both bowed and plucked. The filtered tone signal may be applied to the pitch detector in order to increase the accuracy of the pitch detection process and thus increase the accuracy of the noise reducing characteristics of the system.