Abstract: An acoustic processing device includes: a plurality of loudspeaker units that output sound; and a mechanism that is configured to be capable of changing a distance between the loudspeaker units.

Abstract: An acoustic processing device includes: a plurality of loudspeaker units that output sound; and a mechanism that is configured to be capable of changing a distance between the loudspeaker units.

Abstract: A loudspeaker device includes: an output control unit that causes a sound to be output from a loudspeaker in a case where a headphone device and the loudspeaker device are connected to each other. The sound is generated based on sound data transmitted from the headphone device that has received the sound data through wireless communication from a sound output device. The headphone device is configured to output a sound from a headphone.

Abstract: A speaker apparatus includes speaker units, a first inputting section to which one channel audio signal is supplied, a second inputting section to which at least L-channel and R-channel audio signals are supplied, an acoustic effect imparting section that performs a first signal processing on the audio signal supplied to the first inputting section to output L channel and R channel audio signals, an outputting section that outputs the audio signal supplied to the first inputting section to at least one of the speaker units, and a widening effect imparting section that performs a second signal processing for imparting a sound image widening effect on both the L-channel and R-channel audio signals and the at least L-channel and R-channel audio signals and supplies processed audio signals to speaker units for corresponding channels respectively.

Abstract: A signal processing device is designed to automatically calculate a transfer characteristic representing sound-box resonance of a guitar due to acoustic excitation of vibration which may occur due to white noise. White noise emitted toward the guitar causes vibration propagating via strings so as to produce an audio signal via a pickup. A transfer characteristic is calculated based on an audio signal and a white-noise signal. A filter performs convolution, using the transfer characteristic, on audio data representing user's playing sound of the guitar, thus reproducing sound-box resonance indicating distinctive peaks which may appear in a low-frequency range of guitar's sound. It is possible to store a plurality of transfer characteristics in memory, whereby any user may be allowed to select a desired transfer characteristic among transfer functions stored in memory or to utilize a transfer characteristic actually calculated by the signal processing device.

Abstract: In a coefficient measurement apparatus, a line input terminal receives a pickup signal that is generated based on a string vibration of a musical instrument. A microphone input terminal receives a microphone signal acquired by a microphone that collects sounds of the musical instrument. An adaptive filter estimates a transfer function associated to resonance of the musical instrument and a transfer function of an acoustic space formed from the musical instrument to the microphone, generates an output signal by processing the pickup signal using the estimated transfer function, and updates the transfer function using a difference between the output signal and the microphone signal as a reference signal.

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: 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: A speaker apparatus includes speaker units, a first inputting section to which one channel audio signal is supplied, a second inputting section to which at least L-channel and R-channel audio signals are supplied, an acoustic effect imparting section that performs a first signal processing on the audio signal supplied to the first inputting section to output L channel and R channel audio signals, an outputting section that outputs the audio signal supplied to the first inputting section to at least one of the speaker units, and a widening effect imparting section that performs a second signal processing for imparting a sound image widening effect on both the L-channel and R-channel audio signals and the at least L-channel and R-channel audio signals and supplies processed audio signals to speaker units for corresponding channels respectively.

Abstract: A signal processing device is composed of an acquiring unit, a filter unit and a changing unit. The acquiring unit acquires a signal indicating vibration of a string. The filter unit performs convolution operation on the signal acquired by the acquiring unit according to a filter coefficient and outputs a resulting signal. The filter coefficient is set such that the resulting signal has a frequency response containing a plurality of peak waveforms associated with resonance of a body of a stringed instrument within a specific frequency range. The changing unit changes the filter coefficient so as to change a peak value of each of the peak waveforms while maintaining a width of each of the peak waveforms in the frequency response.

Abstract: In a coefficient measurement apparatus, a line input terminal receives a pickup signal that is generated based on a string vibration of a musical instrument. A microphone input terminal receives a microphone signal acquired by a microphone that collects sounds of the musical instrument. An adaptive filter estimates a transfer function associated to resonance of the musical instrument and a transfer function of an acoustic space formed from the musical instrument to the microphone, generates an output signal by processing the pickup signal using the estimated transfer function, and updates the transfer function using a difference between the output signal and the microphone signal as a reference signal.

Abstract: A sound generating system produces tones from an audio signal as if tones originally produced from an acoustic guitar are converted to an electric signal through a microphone; the electric signal is produced from vibrations of the strings through a pickup unit, and is converted to digital signals; a digital signal processor simulates reverberation generated through a resonator of the guitar and direct tones directly produced from the strings through plural sorts of digital data processing; the signal component representative of the reverberation is added to the signal component representative of the direct tones for producing the audio signal; the digital signal processor simulates the direct tones through the function of an all-pass filter so that the reproduced sound has the timbre approximate to the timbre of the original tones.

Abstract: Reverberation imparting functions as are obtained by a mechanical type reverberation apparatus employing coil spring delaying elements are simulated by an electric circuit configuration. Three delay loop circuits are individually provided, each including a first and a second delay circuit and delaying and circulating an input signal to thereby simulate the propagation of vibrations through each individual coil spring. Multiplier circuits are provided to individually supplying the out puts from the first and the second delay circuit in one delay loop circuit to the input sides of the second and the first delay circuit in two other delay loop circuits after controlling the signal characteristic to thereby simulate the propagation of vibrations through the supporting plates for the coil springs.

Abstract: A sound generating system produces tones from an audio signal as if tones originally produced from an acoustic guitar are converted to an electric signal through a microphone; the electric signal is produced from vibrations of the strings through a pickup unit, and is converted to digital signals; a digital signal processor simulates reverberation generated through a resonator of the guitar and direct tones directly produced from the strings through plural sorts of digital data processing; the signal component representative of the reverberation is added to the signal component representative of the direct tones for producing the audio signal; the digital signal processor simulates the direct tones through the function of an all-pass filter so that the reproduced sound has the timbre approximate to the timbre of the original tones.

Abstract: The audio signal processor has a harmonics modifier for processing an input audio signal to produce an output audio signal. The harmonics modifier is comprised of translating means for sequentially translating the input audio signal into an address signal according to a sampled amplitude of the input audio signal. Memory means is provided for storing an input/output conversion table containing amplitude values in addressable manner. Conversion means operates for accessing the memory means in response to the address signal to effect input/output conversion to read out a sequence of the amplitude values which form the output audio signal containing desired harmonic components according to results of the input/output conversion.