Abstract: In one embodiment, a signal processing method is disclosed. The method can perform filter processing of convoluting a tap coefficient in a first signal sequence to generate a second signal sequence. The method can subtract the second signal sequence from a third signal sequence to generate a fourth signal sequence. The third signal sequence includes an echo signal of the first signal sequence. The method can correct the tap coefficient in accordance with an amount of correction determined using a function. The function includes at least one of a first region and a second region, and has values limited. The first region is included in a negative value region of the fourth signal sequence. The second region is included in a positive value region of the fourth signal sequence.

Abstract: A plurality of sound receiving units is installed onto an equipment body. An initial information memory stores an initial direction of the equipment body in a terminal coordinate system based on the equipment body. An orientation detection unit detects an orientation of the equipment body in a world coordinate system based on a real space. A lock information output unit outputs lock information representing to rock the orientation. An orientation information memory stores the orientation detected when the lock information is output. A direction conversion unit converts the initial direction to a target sound direction in the world coordinate system by using the orientation stored in the orientation information memory. A directivity forming unit forms a directivity of the plurality of sound receiving units toward the target sound direction.

Abstract: An acoustic treatment apparatus obtains a first output signal by performing filtering for forming a directivity in a first direction for received sound signals of sound receivers, obtains a second output signal by performing filtering for forming a directivity in a second direction different from the first direction for received sound signals of sound receivers, obtains a strength ratio between a strength of the first output signal and a strength of the second output signal, and estimates a sound source direction on the basis of the strength ratio.

Abstract: According to one embodiment, a pickup signal processing apparatus includes microphones, a sound determining unit, a signal level calculating unit, a setting unit, and a calculating unit. The sound determining unit determines whether pickup signals picked up by the microphones are signals from a neighboring sound source or a background noise signal. The signal level calculating unit calculates the signal levels for the microphones. The setting unit sets a gain value of at least one microphone and reduces a difference between the signal levels for the microphones on the basis of the signal levels for the microphones, when determined that the pickup signal is the background noise signal. The calculating unit multiplies the pickup signal of the at least one microphone by the gain value set by the setting unit.

Abstract: According to one embodiment, a speech dialog apparatus includes a speech detection unit that detects a start and an end of echo removed speech obtained by removing an echo of response speech contained in input speech; a response interruption control unit that outputs a response interruption command if the end is not yet detected when a predetermined period from the detection of the start passes; and a dialog control unit that causes a response speech output unit to interrupt output of the response speech upon receipt of the response interruption command from the response interruption control unit.

Abstract: According to one embodiment, in response to a first acoustic signal output, a second acoustic signal is input. A filter unit is configured to generate a third acoustic signal by convoluting the first acoustic signal with coefficients. A subtraction unit is configured to generate a fourth acoustic signal by subtracting the third acoustic signal from the second acoustic signal. An estimation unit is configured to decide whether a sound volume of the first acoustic signal is below a predetermined threshold, and to set a sound volume of the second acoustic signal as a non-echo sound level when the sound volume of the first acoustic signal is below the predetermined threshold. A determination unit is configured to determine a step size to correct the coefficients using the non-echo sound level. A correction unit is configured to correct the coefficients using the step size.

Abstract: According to one embodiment, a speech detection apparatus includes a first acoustic signal analyzing unit configured to analyze a frequency spectrum of a first acoustic signal, and a feature extracting unit configured to remove a frequency spectrum of the first acoustic signal from a third acoustic signal, which is obtained by suppressing an echo component of the first acoustic signal contained in a second acoustic signal, so as to extract a feature of a frequency spectrum of the third acoustic signal.

Abstract: A sound signal processing method includes calculating a difference between every few ones of input multiple channel sound signals to obtain a plurality of characteristic quantities each indicating the difference, selecting a weighting factor from a weighting factor dictionary containing a plurality of weighting factors of a plurality of channels corresponding to the characteristic quantities, weighting the sound signals by using the selected weighting factor, and adding the weighted input sound signals to generate an output sound signal.

Abstract: A noise estimation unit estimates a noise signal in an input signal. A section decision unit distinguishes a target signal section from a noise signal section in the input signal. A noise suppression unit suppresses the noise signal based on a first suppression coefficient from the input signal. A noise excess suppression unit suppresses the noise signal based on a second suppression coefficient from the input signal. The second suppression coefficient is larger than the first suppression coefficient. A switching unit switches between an output signal from the noise suppression unit and an output signal from the noise excess suppression unit based on a decision result of the section decision unit.

Abstract: A directional setting apparatus according to the present invention, comprising: a voice recognition unit which detects a certain voice included in a sound signal outputted from a microphone array having a plurality of microphones and a directional determination period indicating a detection period of said certain voice; a voice direction detector which detects occurrence direction of said certain voice in said directional determination period; and a directional controller which controls directivity of a prescribed apparatus based on the sound signals outputted from said plurality of microphones in said directional determination period.

Abstract: An input voice detect is detected after starting a voice input waiting state; the detected voice is recognized; an elapsed time from the start of the voice input waiting state is counted; an informative sound which urges a user to input the voice is outputted when the elapsed time reaches a preset output set time; and the output of the informative sound is stopped when the elapsed time at the time of inputting the voice is shorter than the output set timedetect.

Abstract: An acoustic treatment apparatus obtains a first output signal by performing filtering for forming a directivity in a first direction for received sound signals of sound receivers, obtains a second output signal by performing filtering for forming a directivity in a second direction different from the first direction for received sound signals of sound receivers, obtains a strength ratio between a strength of the first output signal and a strength of the second output signal, and estimates a sound source direction on the basis of the strength ratio.

Abstract: An audible signal process method includes preparing, in at least one dictionary, a plurality of weighting factors each learned to optimize evaluation function established by a weighted learning audible signal and a target speech signal corresponding to the learning audible signal and used for weighting, estimating a noise component included in the input audible signal, calculating a feature quantity depending upon the noise component of the input audible signal, selecting a weighting factor corresponding to the feature quantity from the dictionary, and weighting the input audible signal using the selected weighting factor to generate a processed output audible signal.

Abstract: A speech processing apparatus includes a plurality of microphones which receive speech produced by a first sound source to obtain first speech signals for a plurality of channels having one-to-one correspondence with the plurality of microphones, a calculation unit configured to calculate a first characteristic amount indicative of an inter-channel correlation of the first speech signals, a storage unit configured to store in advance a second characteristic amount indicative of an inter-channel correlation of second speech signals for the plurality of channels obtained by receiving speech produced by a second sound source by the plurality of microphones, and a collation unit configured to collate the first characteristic amount with the second characteristic amount to determine whether the first sound source matches with the second sound source.

Abstract: An audio signal processing method for processing input audio signals of plural channels includes calculating at least one feature quantity representing a difference between channels of input audio signals, selecting at least one weighting factor according to the feature quantity from at least one weighting factor dictionary prepared by learning beforehand, and subjecting the input audio signals of plural channels to signal processing including noise suppression and weighting addition using the selected weighting factor to generate output an output audio signal.

Abstract: A plurality of sound receiving units is installed onto an equipment body. An initial information memory stores an initial direction of the equipment body in a terminal coordinate system based on the equipment body. An orientation detection unit detects an orientation of the equipment body in a world coordinate system based on a real space. A lock information output unit outputs lock information representing to rock the orientation. An orientation information memory stores the orientation detected when the lock information is output. A direction conversion unit converts the initial direction to a target sound direction in the world coordinate system by using the orientation stored in the orientation information memory. A directivity forming unit forms a directivity of the plurality of sound receiving units toward the target sound direction.

Abstract: A sound signal processing method includes calculating a difference between every few ones of input multiple channel sound signals to obtain a plurality of characteristic quantities each indicating the difference, selecting a weighting factor from a weighting factor dictionary containing a plurality of weighting factors of a plurality of channels corresponding to the characteristic quantities, weighting the sound signals by using the selected weighting factor, and adding the weighted input sound signals to generate an output sound signal.

Abstract: A noise estimation unit estimates a noise signal in an input signal. A section decision unit distinguishes a target signal section from a noise signal section in the input signal. A noise suppression unit suppresses the noise signal based on a first suppression coefficient from the input signal. A noise excess suppression unit suppresses the noise signal based on a second suppression coefficient from the input signal. The second suppression coefficient is larger than the first suppression coefficient. A switching unit switches between an output signal from the noise suppression unit and an output signal from the noise excess suppression unit based on a decision result of the section decision unit.

Abstract: An input speech signal to an input terminal is supplied to a speech synthesizer section through a speech analyzer section and frequency parameter quantizer section to form a synthesis filter, and the input speech signal is expressed by quantized LPC coefficients representing the characteristics of the synthesis filter and an excitation signal for exciting the synthesis filter. In this case, in a pulse excitation section, a pulse position selector selects pulse position candidates from the integer pulse positions and non-integer pulse positions stored in a pulse position codebook, and an integer position pulse generator and non-integer position pulse generator respectively generate integer position pulses set at sampling points of the excitation signal and non-integer position pulses set at positions located between sampling points. These pulses are synthesized into a pulse train serving as a source of an excitation signal.

Abstract: A directional setting apparatus according to the present invention, comprising: a voice recognition unit which detects a certain voice included in a sound signal outputted from a microphone array having a plurality of microphones and a directional determination period indicating a detection period of said certain voice; a voice direction detector which detects occurrence direction of said certain voice in said directional determination period; and a directional controller which controls directivity of a prescribed apparatus based on the sound signals outputted from said plurality of microphones in said directional determination period.