Abstract: The present disclosure teaches a filtering unit configured to perform filtering on input audio signals in accordance with coefficients set for a plurality of audio outputs, respectively, and a plurality of speakers configured to output audio in response to audio signals filtered for the plurality of audio outputs. The filtering unit performs the filtering on the input audio signals in accordance with the coefficients adjusted so that the sound pressures and the sound pressure gradients at a plurality of control points set on a boundary plane of an enclosed space surrounding the plurality of speakers reach values corresponding to a desired sound field. This allows producing a desired sound field on the boundary plane of the enclosed space surrounding the sound sources by making use of the properties of the Kirchhoff-Helmholtz integral equation that requires that the sound sources are present in an enclosed space.

Abstract: A digital tuner in a broadcasting receiving unit outputs audio data at a sample rate of Fs+dr (Hz), and an ASRC rate-converts audio data to audio data at a sample rate of Fs+ds (Hz) and transmits the resulting audio data to an audio processing unit. A DAC in the audio processing unit analog-converts the received audio data at an output rate of Fs+da (Hz) and outputs the resulting audio data to a speaker. The sample rate Fs+ds and the output rate Fs+da are synchronized with a SYNC transmitted from the broadcasting receiving unit to the audio processing unit on a 125 ms cycle. For the sample rate Fs+ds and the output rate Fs+da, a relationship of (da?ds)×0.125<1 is assured.

Abstract: A digital tuner in a broadcasting receiving unit outputs audio data at a sample rate of Fs+dr (Hz), and an ASRC rate-converts audio data to audio data at a sample rate of Fs+ds (Hz) and transmits the resulting audio data to an audio processing unit. A DAC in the audio processing unit analog-converts the received audio data at an output rate of Fs+da (Hz) and outputs the resulting audio data to a speaker. The sample rate Fs+ds and the output rate Fs +da are synchronized with a SYNC transmitted from the broadcasting receiving unit to the audio processing unit on a 125 ms cycle. For the sample rate Fs+ds and the output rate Fs+da, a relationship of (da?ds)×0.125<1 is assured.

Abstract: A sound field control apparatus includes at least two main microphones; for each main microphone, a set of at least two sub microphones arranged such that the at least two sub microphones are placed in different axis directions about each of the main microphones; a filtering unit; and a filter coefficient calculating unit configured to calculate a filter coefficient for the filtering unit. A filter coefficient used to control sound pressure levels and air particle velocities of an output audio signal is calculated on the basis of a sound pressure level detected by each main microphone and the difference between the sound pressure level detected by the main microphone and that detected by each of the corresponding sub microphones.

Abstract: An audio apparatus has a function of correcting an audio signal in response to a noise level. The audio apparatus includes a correction unit that corrects an input audio signal on the basis of a weighting factor, an output unit that produces a played-back audio sound on the basis of the corrected audio signal, a microphone for receiving an external sound that includes the played-back audio sound and noise, a noise-extracting unit that extracts a noise signal from an external sound signal, the noise-extracting unit including a speech-removing unit that removes a speech signal from the noise signal on the basis of noise spectrum data, and a weighting factor calculation unit that calculates the weighting factor on the basis of the extracted noise signal and supplies the calculated weighting factor to the correction unit.

Abstract: A sound field control apparatus includes at least two main microphones; for each main microphone, a set of at least two sub microphones arranged such that the at least two sub microphones are placed in different axis directions about each of the main microphones; a filtering unit; and a filter coefficient calculating unit configured to calculate a filter coefficient for the filtering unit. A filter coefficient used to control sound pressure levels and air particle velocities of an output audio signal is calculated on the basis of a sound pressure level detected by each main microphone and the difference between the sound pressure level detected by the main microphone and that detected by each of the corresponding sub microphones.

Abstract: In a sound field control apparatus including multiple speakers, multiple microphones gathering sound radiated from the multiple speakers, a mode decomposition filter that performs mode decomposition on a sound pressure distribution, and a control filter that controls the input signals to be input to the multiple speakers such that the mode amplitudes of the modes decomposed by the mode decomposition filter can have a predetermined value, a sound pressure distribution in the acoustic space is measured, and the sound pressure distribution in the acoustic space is expressed by using a sinusoidal function and cosine function of a space frequency of the mode to be controlled in amplitude. The mode space frequency is corrected such that the expressed sound pressure distribution can be equal to the measured sound pressure distribution, and the filter coefficient for the mode decomposition filter is determined based on the mode space frequency obtained by the correction (corrected mode space frequency).

Abstract: In a sound field control apparatus including multiple speakers, multiple microphones gathering sound radiated from the multiple speakers, a mode decomposition filter that performs mode decomposition on a sound pressure distribution, and a control filter that controls the input signals to be input to the multiple speakers such that the mode amplitudes of the modes decomposed by the mode decomposition filter can have a predetermined value, a sound pressure distribution in the acoustic space is measured, and the sound pressure distribution in the acoustic space is expressed by using a sinusoidal function and cosine function of a space frequency of the mode to be controlled in amplitude. The mode space frequency is corrected such that the expressed sound pressure distribution can be equal to the measured sound pressure distribution, and the filter coefficient for the mode decomposition filter is determined based on the mode space frequency obtained by the correction (corrected mode space frequency).

Abstract: A first band analyzer divides an acoustic signal received from a sound playback system through an input unit into frequency bands, and generates a first band level. An acoustic signal estimator estimates the band level of the original acoustic signal at the input unit, and generates a second band level for each band. A processor extracts an external noise component which is contained in the acoustic signal using the first band level and the second band level. The external noise can be accurately estimated with less computation than in the related art.

Abstract: An apparatus for estimating an amount of noise is disclosed. The apparatus includes a FET processor that analyzes frequency components of an audio signal input to a speaker in a vehicle, a FET processor that analyzes frequency components of a signal output from a microphone in the vehicle, a coherence function calculator which detects a ratio of the audio signal included in the signal output from the microphone by calculating a magnitude squared coherence function based on the frequency components of the two signals analyzed by the first and second frequency analyzers, and a multiplier and adder which calculates an amount of external noise reaching the microphone separately from an audio sound corresponding to the audio signal on the basis of the signal output from the microphone and the ratio of the audio signal detected by the coherence function calculator.

Abstract: An audio apparatus has a function of correcting an audio signal in response to a noise level. The audio apparatus includes a correction unit that corrects an input audio signal on the basis of a weighting factor, an output unit that produces a played-back audio sound on the basis of the corrected audio signal, a microphone for receiving an external sound that includes the played-back audio sound and noise, a noise-extracting unit that extracts a noise signal from an external sound signal, the noise-extracting unit including a speech-removing unit that removes a speech signal from the noise signal on the basis of noise spectrum data, and a weighting factor calculation unit that calculates the weighting factor on the basis of the extracted noise signal and supplies the calculated weighting factor to the correction unit.

Abstract: In an adaptive audio equalizer apparatus, a signal that is output from an adaptive filter 13 is fed to a speaker and to a delaying unit. The signal fed to the delaying unit is delayed for a predetermined time, and is multiplied by a scaling factor in a multiplying unit. A first calculation unit calculates a difference between an output of a microphone and a target response signal, and outputs the result as an error signal. A second calculation unit adds the output of the multiplying unit to the error signal, and outputs the result to a filter coefficient setting unit of the adaptive filter.

Abstract: A first band analyzer divides an acoustic signal received from a sound playback system through an input unit into frequency bands, and generates a first band level. An acoustic signal estimator estimates the band level of the original acoustic signal at the input unit, and generates a second band level for each band. A processor extracts an external noise component which is contained in the acoustic signal using the first band level and the second band level. The external noise can be accurately estimated with less computation than in the related art.

Abstract: A designing system for adaptively characterizing an audio transmitting system has a white noise generating unit for generating a white noise signal. A speaker radiates the white noise generated by the white noise generating unit into an acoustic space. A microphone is placed at a predetermined position in the acoustic space and collects sound radiated from the speaker. A FIR adaptive filter receives the above white noise signal. An LMS algorithm processing unit updates each tap coefficient of the adaptive filter by using the LMS algorithm. A computation unit calculates the difference between a detection signal output from the microphone and an output of the adaptive filter and outputs the difference as an error signal &egr;. By using a white noise signal having an average power of one, the range of the step size parameter of the LMS algorithm required for stably operating the adaptive filter is fixed.

Abstract: In an adaptive audio equalizer apparatus, a signal that is output from an adaptive filter 13 is fed to a speaker and to a delaying unit. The signal fed to the delaying unit is delayed for a predetermined time, and is multiplied by a scaling factor in a multiplying unit. A first calculation unit calculates a difference between an output of a microphone and a target response signal, and outputs the result as an error signal. A second calculation unit adds the output of the multiplying unit to the error signal, and outputs the result to a filter coefficient setting unit of the adaptive filter.