Abstract: A system for timbral change, capable of changing timbres included in an existing music audio signal to arbitrary timbres. Replaced harmonic peak parameters are created by replacing a plurality of harmonic peaks included in harmonic peak parameters, which are stored in a separated audio signal analyzing and storing section 3 and indicate relative amplitudes of n-th order harmonic components of each tone generated by a musical instrument of a first kind, with harmonic peaks included in harmonic peak parameters, which are stored in a replacement parameter storing section 6 and indicate relative amplitudes of n-th order harmonic components of each tone generated by a musical instrument of a second kind and corresponding to each tone generated by the musical instrument of the first kind. A synthesized separated audio signal generating section 7 generates a synthesized separated audio signal for each tone using parameters other than the harmonic peak parameters and the replaced harmonic peak parameters.

Abstract: A thereminist robot has a characteristic model of theremin and is capable of performing in response to an environment of theremin performance by calibrating the characteristic model before the performance. A robot 10 has a first arm 12, a second arm 11, and a pitch model for indicating an arm position corresponding to a pitch of the theremin The robot 10 plays the theremin by moving the first arm 12 to the arm position corresponding to a musical note based on the target note and the pitch model. The robot further has a parameter adjustment unit for adjusting parameters of the pitch model that change depending on environments surrounding the theremin.

Abstract: A sound processing device includes a separation unit configured to separate at least a music signal and a speech signal from a recorded audio signal, a noise suppression unit, a music feature value estimation unit, a speech recognition unit, a noise-processing confidence calculation unit, a music feature value estimation confidence calculation unit, a speech recognition confidence calculation unit, and a control unit configured to calculate at least one behavioral decision function of a speech behavioral decision function associated with speech and a music behavioral decision function associated with music based on a noise-processing confidence value, a music feature value estimation confidence value, and a speech recognition confidence value and to determine behavior corresponding to the calculated behavioral decision function.

Abstract: A robot may include a driving control unit configured to control a driving of a movable unit that is connected movably to a body unit, a voice generating unit configured to generate a voice, and a voice output unit configured to output the voice, which has been generated by the voice generating unit. The voice generating unit may correct the voice, which is generated, based on a bearing of the movable unit, which is controlled by the driving control unit, to the body unit.

Abstract: A beat tracking apparatus includes: a filtering unit configured to perform a filtering process on an input acoustic signal and to accentuate an onset; a beat interval reliability calculating unit configured to perform a time-frequency pattern matching process employing a mutual correlation function on the acoustic signal of which the onset is accentuated and to calculate a beat interval reliability; and a beat interval estimating unit configured to estimate a beat interval on the basis of the calculated beat interval reliability.

Abstract: A speech recognition system according to the present invention includes a sound source separating section which separates mixed speeches from multiple sound sources from one another; a mask generating section which generates a soft mask which can take continuous values between 0 and 1 for each frequency spectral component of a separated speech signal using distributions of speech signal and noise against separation reliability of the separated speech signal; and a speech recognizing section which recognizes speeches separated by the sound source separating section using soft masks generated by the mask generating section.

Abstract: A speech understanding apparatus includes a speech recognition unit which performs speech recognition of an utterance using multiple language models, and outputs multiple speech recognition results obtained by the speech recognition, a language understanding unit which uses multiple language understanding models to perform language understanding for each of the multiple speech recognition results output from the speech recognition unit, and outputs multiple speech understanding results obtained from the language understanding, and an integrating unit which calculates, based on values representing features of the speech understanding results, utterance batch confidences that numerically express accuracy of the speech understanding results for each of the multiple speech understanding results output from the language understanding unit, and selects one of the speech understanding results with a highest utterance batch confidence among the calculated utterance batch confidences.

Abstract: A musical score position estimating apparatus includes a sound feature quantity generating unit configured to generate a feature quantity of an input sound signal, and, a score position estimating unit configured to calculate a weight coefficient based on the feature quantity of the sound signal and a feature quantity of musical score information and estimates a musical score position using a virtual musical score position and a virtual tempo corresponding to the weight coefficient.

Abstract: In a sound source localization system using a light emitting device for visualizing sound information, including: a light emitting device (40) including a microphone for receiving sound from a sound source (1, 2) and a light emitting means for emitting light based on the sound from the microphone; a generating section for generating light emitting information for the light emitting device (40); and a sound source localization section (60) for determining a position of the sound source based on the light emitting information from the generating section.

Abstract: The speech recognition system of the present invention includes: a sound source separating section which separates mixed speeches from multiple sound sources; a mask generating section which generates a soft mask which can take continuous values between 0 and 1 for each separated speech according to reliability of separation in separating operation of the sound source separating section; and a speech recognizing section which recognizes speeches separated by the sound source separating section using soft masks generated by the mask generating section.

Abstract: A reverberation suppressing apparatus separating sound source signals based on input signals output from microphones collecting the plurality of sound source signals, includes a sound signal output unit generating sound signals and outputting the generated sound signals, a sound acquiring unit acquiring the input signals from microphones, a first evaluation function calculation unit calculating a separation matrix, the input signals, and the sound source signals, and calculating a first evaluation function, a reverberation component suppressing unit calculating an optimal separation matrix, and suppressing a reverberation component by separating the sound source signals other than the generated sound signals, and a separation matrix updating unit dividing a step-size function, approximating each segment to a linear function, calculating step sizes based on the approximated linear functions, and repeatedly updating the separation matrix so that the degree of separation of the sound source signals exceeds the p

Abstract: In accordance with one aspect of the invention, the sound source position determining system comprises a detector (101) for detecting sound data, and a computing unit (103) for computing MUSIC spectrum for each direction and time. The system includes a state transition model that describes state and transition of the state according to existence or absence of sound sources in each direction, and a model parameter estimating unit (105) that determines an observation model describing MUSIC spectrum observed in a state where one or more sound sources exist and in a state where no sound sources exist and estimates posterior distribution of model parameters of the observation model and the state transition model based on temporal data of MUSIC spectrum.

Abstract: A musical piece recommendation system that allows instantaneous registration of a new user and a new musical piece without retraining in a basic training section. A first incremental training section monitors a rating history storage section, and each time a change is made to a rating history or a new user is added, performs updating of or addition of the topic selection probability for the user for which the change is made to the rating history or for the new user such that the likelihood determined by a basic training section is kept maximized. A second incremental training section monitors an acoustic feature storage section, and each time a new musical piece is added to perform addition to acoustic features, adds the musical piece selection probability related to the added musical piece such that the likelihood determined by the basic training section is kept maximized.

Abstract: A sound separating device includes a first microphone that converts input sound into a first sound signal, a second microphone that converts input sound into a second sound signal and has characteristics of a larger distance attenuation ratio than the first microphone, and a sound signal processing portion that optimizes a separating matrix by independent component analysis based on the first sound signal and the second sound signal that are supplied, and uses the optimized separating matrix so as to separate a third sound signal as a sound signal from a near field sound source and to separate a fourth sound signal as a sound signal from a far field sound source.

Abstract: A language understanding device includes: a language understanding model storing unit configured to store word transition data including pre-transition states, input words, predefined outputs corresponding to the input words, word weight information, and post-transition states, and concept weighting data including concepts obtained from language understanding results for at least one word, and concept weight information corresponding to the concepts; a finite state transducer processing unit configured to output understanding result candidates including the predefined outputs, to accumulate word weights so as to obtain a cumulative word weight, and to sequentially perform state transition operations; a concept weighting processing unit configured to accumulate concept weights so as to obtain a cumulative concept weight; and an understanding result determination unit configured to determine an understanding result from the understanding result candidates by referring to the cumulative word weight and the cumul

Abstract: An audio signal produced by playing a plurality of musical instruments is separated into sound sources according to respective instrument sounds. Each time a separation process is performed, the updated model parameter estimation/storage section 114 estimates parameters respectively contained in updated model parameters such that updated power spectrograms gradually change from a state close to initial power spectrograms to a state close to a plurality of power spectrograms most recently stored in a power spectrogram separation/storage section. Respective sections including the power spectrogram separation/storage section 112 and an updated distribution function computation/storage section 118 repeatedly perform process operations until the updated power spectrograms change from the state close to the initial power spectrograms to the state close to the plurality of power spectrograms most recently stored in the power spectrogram separation/storage section 112.

Abstract: A result of a sound source direction measurement based on an output of an REMA (first microphone array) (11) and a result of a sound source position measurement based on an output of an IRMA (second microphone array) (12) are integrated through a particle filter or in space. Thus, the different microphones, i.e., the REMA (11) and the IRMA (12) can cancel mutual defects or ambiguities with each other. Therefore, from views of improvement in accuracy and robustness a performance of sound source localization can be improved.

Abstract: A system for timbral change, capable of changing timbres included in an existing music audio signal to arbitrary timbres. Replaced harmonic peak parameters are created by replacing a plurality of harmonic peaks included in harmonic peak parameters, which are stored in a separated audio signal analyzing and storing section 3 and indicate relative amplitudes of n-th order harmonic components of each tone generated by a musical instrument of a first kind, with harmonic peaks included in harmonic peak parameters, which are stored in a replacement parameter storing section 6 and indicate relative amplitudes of n-th order harmonic components of each tone generated by a musical instrument of a second kind and corresponding to each tone generated by the musical instrument of the first kind. A synthesized separated audio signal generating section 7 generates a synthesized separated audio signal for each tone using parameters other than the harmonic peak parameters and the replaced harmonic peak parameters.

Abstract: A voice recognition system (10) for improving the toughness of voice recognition for a voice input for which a deteriorated feature amount cannot be completely identified. The system comprises at least two sound detecting means (16a, 16b) for detecting a sound signal, a sound source localizing unit (21) for determining the direction of a sound source based on the sound signal, a sound source separating unit (23) for separating a sound by the sound source from the sound signal based on the sound source direction, a mask producing unit (25) for producing a mask value according to the reliability of the separation results, a feature extracting unit (27) for extracting the feature amount of the sound signal, and a voice recognizing unit (29) for applying the mask to the feature amount to recognize a voice from the sound signal.

Abstract: A reverberation suppressing apparatus, includes: a sound acquiring unit which acquires a sound signal; a reverberation data computing unit which computes reverberation data from the acquired sound signal; a reverberation characteristics estimating unit which estimates reverberation characteristics based on the computed reverberation data; a filter length estimating unit which estimates a filter length of a filter which is used to suppress a reverberation based on the estimated reverberation characteristics; and a reverberation suppressing unit which suppresses the reverberation based on the estimated filter length.