Abstract: Provided is a sound source localization device which can detect a source location of an extraction sound, including at least two microphones; an analysis unit (103) which (i) analyze frequencies of the mixed sound including the noise and received by each microphone, and (ii) generates frequency signals; and an extraction unit (105) which, for each source location candidate, (a) adjusts time axes of the frequency signals corresponding to the microphones, so that there is no time difference between when the mixed sound reaches one microphone from the source location candidate and when the mixed sound reaches another microphone from the source location candidate, and (b) determines frequency signals having a difference distance equal to or smaller than a threshold value, from among the frequency signals corresponding to the microphones with the time axis having been adjusted, the difference distance representing a degree of a difference in the frequency signals between the microphones, and (c) extracts the sourc

Abstract: An audio restoration apparatus is provided which restores an audio to be restored having a missing audio part and being included in a mixed audio. The audio restoration apparatus includes: a mixed audio separation unit which extracts the audio to be restored included in the mixed audio; an audio structure analysis unit which generates at least one of a phoneme sequence, a character sequence and a musical note sequence of the missing audio part; an unchanged audio characteristic domain analysis unit which segments the extracted audio to be restored into time domains in each of which an audio characteristic remains unchanged; an audio characteristic extraction unit which identifies a time domain where the missing audio part is located, and extracts audio characteristics of the identified time domain in the audio to be restored; and an audio restoration unit which restores the missing audio part in the audio to be restored.

Abstract: A mixed audio separation system (100) which separates a specific audio from among a mixed audio (S100) includes a local frequency information generation unit (105) which obtains pieces of local frequency information (S103) corresponding to local reference waveforms (S102), based on the local reference waveforms (S102) and an analysis waveform which is the waveform of the mixed audio (S100). Each of the local reference waveforms (S102) (i) constitutes a part of a reference waveform for analyzing a predetermined frequency, (ii) has a predetermined temporal/spatial resolution and (iii) includes at least one of an amplification spectrum and a phase spectrum in the predetermined frequency.

Abstract: A sound identification apparatus which reduces the chance of a drop in the identification rate, including: a frame sound feature extraction unit which extracts a sound feature per frame of an inputted audio signal; a frame likelihood calculation unit which calculates a frame likelihood of the sound feature in each frame, for each of a plurality of sound models; a confidence measure judgment unit which judges a confidence measure based on the frame likelihood; a cumulative likelihood output unit time determination unit which determines a cumulative likelihood output unit time based on the confidence measure; a cumulative likelihood calculation unit which calculates a cumulative likelihood in which the frame likelihoods of the frames included in the cumulative likelihood output unit time are cumulated, for each sound model; a sound type candidate judgment unit which determines, for each cumulative likelihood output unit time, a sound type corresponding to the sound model that has a maximum cumulative likelihood

Abstract: A target sound analysis apparatus capable of distinguishing between a sound having the same fundamental period as a target sound but which differs therefrom and the target sound and analyzing whether or not the target sound is contained in an evaluation sound is an target sound analysis apparatus that analyzes whether or not a target sound is included in an evaluation sound, and includes: a target sound preparation unit that prepares a target sound that is an analysis waveform to be used for analyzing a fundamental period; an evaluation sound preparation unit that prepares an evaluation sound that is an analyzed waveform in which its fundamental period will be analyzed; and an analysis unit that temporally shifts the target sound with respect to the evaluation sound to sequentially calculate differential values of the evaluation sound and the target sound at corresponding points in time, calculate an iterative interval between the points in time where the differential value is equal to or lower than a predete

Abstract: The present invention provides a speech recognition apparatus which appropriately performs speech recognition by generating, in real time, language models adapted to a new topic even in the case where topics are changed.

Abstract: A sound identification apparatus which reduces the chance of a drop in the identification rate, including: a frame sound feature extraction unit which extracts a sound feature per frame of an inputted audio signal; a frame likelihood calculation unit which calculates a frame likelihood of the sound feature in each frame, for each of a plurality of sound models; a confidence measure judgment unit which judges a confidence measure based on the frame likelihood; a cumulative likelihood output unit time determination unit which determines a cumulative likelihood output unit time based on the confidence measure; a cumulative likelihood calculation unit which calculates a cumulative likelihood in which the frame likelihoods of the frames included in the cumulative likelihood output unit time are cumulated, for each sound model; a sound type candidate judgment unit which determines, for each cumulative likelihood output unit time, a sound type corresponding to the sound model that has a maximum cumulative likelihood

Abstract: With respect to audio signal coding and decoding apparatuses, there is provided a coding apparatus that enables a decoding apparatus to reproduce an audio signal even through it does not use all of data from the coding apparatus, and a decoding apparatus corresponding to the coding apparatus. A quantization unit constituting a coding apparatus includes a first sub-quantization unit comprising sub-quantization units for low-band, intermediate-band, and high-band; a second sub-quantization unit for quantizing quantization errors from the first sub-quantization unit; and a third sub-quantization unit for quantizing quantization errors which have been processed by the first sub-quantization unit and the second sub-quantization unit.

Abstract: An audio restoration apparatus which restores an audio to be restored having a missing audio part and being included in a mixed audio.

Abstract: To provide a speech recognition apparatus which appropriately performs speech recognition by generating, in real time, language models adapted to a new topic even in the case where topics are changed.

Abstract: The speech recognition apparatus (1) is equipped with the garbage acoustic model storage unit (110) storing the garbage acoustic model which learned the collection of the unnecessary words; the feature value calculation unit (101) which calculates the feature parameter necessary for recognition by acoustically analyzing the unidentified input speech including the non-language speech per frame which is a unit for speech analysis; the garbage acoustic score calculation unit (111) which calculates the garbage acoustic score by comparing the feature parameter and the garbage acoustic model; the garbage acoustic score correction unit (113) which corrects the garbage acoustic score calculated by the garbage acoustic score calculation unit (111) so as to raise it in the frame where the non-language speech is inputted; and the recognition result output unit (105) which outputs, as the recognition result of the unidentified input speech, the word string with the highest cumulative score of the language score, the word

Abstract: With respect to audio signal coding and decoding apparatuses, there is provided a coding apparatus that enables a decoding apparatus to reproduce an audio signal even through it does not use all of data from the coding apparatus, and a decoding apparatus corresponding to the coding apparatus. A quantization unit constituting a coding apparatus includes a first sub-quantization unit comprising sub-quantization units for low-band, intermediate-band, and high-band; a second sub-quantization unit for quantizing quantization errors from the first sub-quantization unit; and a third sub-quantization unit for quantizing quantization errors which have been processed by the first sub-quantization unit and the second sub-quantization unit.

Abstract: With respect to audio signal coding and decoding apparatuses, there is provided a coding apparatus that enables a decoding apparatus to reproduce an audio signal even through it does not use all of data from the coding apparatus, and a decoding apparatus corresponding to the coding apparatus. A quantization unit constituting a coding apparatus includes a first sub-quantization unit comprising sub-quantization units for low-band, intermediate-band, and high-band; a second sub-quantization unit for quantizing quantization errors from the first sub-quantization unit; and a third sub-quantization unit for quantizing quantization errors which have been processed by the first sub-quantization unit and the second sub-quantization unit.

Abstract: An input speech utterance is segmented into a prefixed time length to make frames, to extract an acoustic feature parameter of each frame. The acoustic feature parameter is frequency-converted by using pluralfrequency conversion coefficients previously defined. By using all combinations of plural post-conversion feature parameters obtained by the frequency conversion and at least one standard phonemic model, to compute plural similarities or distances of between the post-conversion feature parameters of each of the frames and the standard phonemic model. A frequency converting condition for normalizing the input utterance is decided by using the pluralsimilarities or distances. By using the frequency converting condition, the input utterance is normalized. With this method, even in case there is change of the speaker making a speech utterance, the individual difference of input utterance can be corrected thereby improving the performance of speech recognition.

Abstract: An audio signal compression apparatus for compressively coding an input audio signal comprises a time-to-frequency transformation unit for transforming the input audio signal to a frequency domain signal; a spectrum envelope calculation unit for calculating a spectrum envelope having different resolutions for different frequencies, from the input audio signal, using a weighting function on frequency based on human auditory characteristics; a normalization unit for normalizing the frequency domain signal using the spectrum envelope to obtain a residual signal; a power normalization unit for normalizing the residual signal by the power; an auditory weighting calculation unit for calculating weighting coefficients on frequency, based on the spectrum of the input audio signal and human auditory characteristics; and a multi-stage quantization device having plural stages of vector quantizers connected in series, to which the normalized residual signal is input, and at least one of the vector quantizers quantizing t

Abstract: An audio signal compression apparatus for compressively coding an input audio signal comprises a time-to-frequency transformation unit for transforming the input audio signal to a frequency domain signal; a spectrum envelope calculation unit for calculating a spectrum envelope having different resolutions for different frequencies, from the input audio signal, using a weighting function on frequency based on human auditory characteristics; a normalization unit for normalizing the frequency domain signal using the spectrum envelope to obtain a residual signal; a power normalization unit for normalizing the residual signal by the power; an auditory weighting calculation unit for calculating weighting coefficients on frequency, based on the spectrum of the input audio signal and human auditory characteristics; and a multi-stage quantization means having plural stages of vector quantizers connected in series, to which the normalized residual signal is input, and at least one of the vector quantizers quantizing th

Abstract: An audio signal compression apparatus for compressively coding an input audio signal comprises a time-to-frequency transformation unit for transforming the input audio signal to a frequency domain signal; a spectrum envelope calculation unit for calculating a spectrum envelope having different resolutions for different frequencies, from the input audio signal, using a weighting function on frequency based on human auditory characteristics; a normalization unit for normalizing the frequency domain signal using the spectrum envelope to obtain a residual signal; a power normalization unit for normalizing the residual signal by the power; an auditory weighting calculation unit for calculating weighting coefficients on frequency, based on the spectrum of the input audio signal and human auditory characteristics; and a multi-stage quantization device having plural stages of vector quantizers connected in series, to which the normalized residual signal is input, and at least one of the vector quantizers quantizing t

Abstract: Apparatus for expanding the bandwidth of speech signals such that a narrowband speech signal is input and digitized, the spectral envelope information and residual information are extracted from the digitized signal by linear predictive coding analysis, the spectral envelope information is expanded into wideband information by a spectral envelope converter, the residual information is expanded into wideband information by a residual converter, the converted spectral envelope information and residual information are combined to produce a wideband speech signal, frequency information not contained in the input signal is extracted from the obtained wideband speech signal by a filter, and the resulting signal is added to the original digitized input signal, and the obtained signal is converted into an analog signal as the output signal of the apparatus.

Abstract: The speech detection apparatus comprises: a reference model maker for extracting a plurality of parameters for a speech detection from training data, and for making a reference model based on the parameters; a parameter extractor for extracting the plurality of parameters from each frame of an input audio signal; and a decision device for deciding whether or not the audio signal is speech, by comparing the parameters extracted from the input audio signal with the reference model. The reference model maker makes the reference model for each phoneme.

Abstract: A circuit for generating a pulse-like timing signal driving a stepping motor which is used to drive for example a magnetic tape to run in an audio tape or video tape recording and reproducing apparatus. A rewritable memory stores output pattern data and time information for generating the timing signal. This timing signal generating circuit comprises a rewrite control circuit responsive to an operation mode instructing signal, for performing a predetermined operational processing on basic pattern data and basic time information and storing the resulting output pattern data and uniquely related time information in the rewritable memory.