Abstract: An apparatus for speech processing: calculates a pitch gain indicating a magnitude of periodicity of an audio signal for each frame, the audio signal representing speaker's voice to be divided into the frames each having a predetermined length; determines that a speech production interval has started, when the pitch gain becomes equal or greater than a first threshold after a non-speech production interval; sets a second threshold that is lower than the first threshold by a reduction amount corresponding to a value acquired by subtracting a second representative value of the pitch gain in an interval preceding the start of the speech production interval from a first representative value of the pitch gain in the speech production interval; and determines that the speech production interval has terminated, when the pitch gain becomes smaller than the second threshold after the speech production interval has started.

Abstract: A device for determining a sound source direction determines a direction in which a source of a reached sound exists, based on at least one of a sound pressure difference between a first sound pressure that is a sound pressure of a first frequency component of a first part of the reached sound acquired by a first microphone and a second sound pressure that is a sound pressure of the first frequency component of a second part of the reached sound acquired by a second microphone, and a phase difference between a first phase that is a phase of a second frequency component of the first part of the reached sound and a second phase that is a phase of the second frequency component of the second part of the reached sound.

Abstract: An audio coding device includes a filter configured to extract a low-band signal having a first frequency component from an input signal, a memory, and a processor coupled to the memory and configured to extract envelope information relating to an envelope of a high-band signal having a second frequency component which is higher than the first frequency component in the input signal, detect tone information that is information on a tone signal included in a high-band signal spectrum from the input signal, correct the envelope information based on a difference between frequency of the tone signal and frequency of a peak of the envelope, and code the low-band signal, the tone information, and the envelope information that is corrected.

Abstract: A voice section determination method including determining, for each of a plurality of sound frames, whether each of the plurality of sound frames corresponds to an utterance section, calculating a background noise for a target sound frame in the plurality of sound frames based on the plurality of sound frames prior to the target sound frame, the plurality of sound frames being included in a silence section, calculating a signal-to-noise ratio by using the calculated background noise, determining which does the target sound frame correspond to a first sound section of a first sound, or a second sound section of a second sound, the second sound being generated by transforming the first sound, and when the target sound frame is determined to correspond to the first sound section, determining whether the target sound frame corresponds to a voice section based on a pitch gain.

Abstract: There is provided an audio encoding apparatus including a memory, and a processor coupled to the memory and the processor configured to determine whether a tone is included in a boundary between a low-frequency that is a frequency bandwidth below a predetermined frequency of an input signal and a high-frequency that is a frequency bandwidth above the predetermined frequency of the input signal, suppress a tone in one of the low-frequency and the high-frequency, encode the input signal having the low-frequency to generate a low-frequency code, encode the input signal having the high-frequency to generate a high-frequency code, and generate an encoded stream by multiplexing the low-frequency code and the high-frequency code.

Abstract: An audio coding device includes a filter configured to extract a low-band signal having a first frequency component from an input signal, a memory, and a processor coupled to the memory and configured to extract envelope information relating to an envelope of a high-band signal having a second frequency component which is higher than the first frequency component in the input signal, detect tone information that is information on a tone signal included in a high-band signal spectrum from the input signal, correct the envelope information based on a difference between frequency of the tone signal and frequency of a peak of the envelope, and code the low-band signal, the tone information, and the envelope information that is corrected.

Abstract: A method for utterance section detection includes: executing pitch gain calculation processing that includes calculating a pitch gain indicating an intensity of periodicity of an audio signal expressing a voice of a speaker for each of frames that are obtained by dividing the audio signal and that each have a predetermined length; and executing utterance section detection processing that includes determining that an utterance section on the audio signal starts when the pitch gain becomes greater than or equal to a first threshold value after a non-utterance section on the audio signal lasts, wherein the utterance section detection processing further includes determining that the utterance section ends when the pitch gain becomes less than a second threshold value lower than the first threshold value after the utterance section lasts.

Abstract: A plurality of candidates of prediction coefficients of a channel signal out of a plurality of channel signals are extracted from a code book storing a plurality of prediction coefficients, each candidate having a predictive error falling within a specific range of predictive error of predictive encoding of two other channel signals, and a prediction coefficient is selected from the extracted candidates as a result of the predictive encoding in accordance with a specific data embedding rule and embedding embed target data into the selected prediction coefficient.

Abstract: A device for embedding data upon a prediction coding of a multi-channel signal includes a storage unit to store a code book that includes a plurality of prediction parameter sets, each of the plurality of prediction parameter sets including a plurality of kinds of prediction parameters for a processing regarding the prediction coding. The device extracts a plurality of candidates of a prediction parameter set for the multi-channel signal from the code book, wherein the plurality of candidates are capable of suppressing a prediction error in the prediction coding within a predetermined range, converts an embedding object that is at least part of the data in accordance with a number corresponding to a number of the candidates, selects, from the plurality of candidates, the prediction parameter set corresponding to the converted embedding object, and multiplexes the selected prediction parameter set with coded data which are down-mixed from the multi-channel signal.

Abstract: An apnea episode determination device includes, a processor; and a memory which stores a plurality of instructions, which when executed by the processor, cause the processor to execute, detecting a breathing segment and a midway segment from a sound signal during sleep, the breathing segment being considered to include a breathing sound, the midway segment existing in between the breathing segments; calculating an acoustic feature based on a background noise component and a signal component excluding the background noise component, which are included in the midway segment; and determining that the midway segment is an apnea episode when the acoustic feature meets a preset condition.

Abstract: An audio coding device that performs predictive coding on a third-channel signal included in a plurality of channels in an audio signal according to a first-channel signal and a second-channel signal, which are included in the plurality of channels, and to a plurality of channel prediction coefficients included in a coding book, the device includes a processor; and a memory which stores a plurality of instructions, which when executed by the processor, cause the processor to execute, selecting channel prediction coefficients corresponding to the first-channel signal and the second-channel signal so that an error, which is determined by a difference between the third-channel signal before predictive coding and the third-channel signal after predictive coding, is minimized; and controlling the first-channel signal or the second-channel signal so that the error is further reduced.

Abstract: A centering device for a plate-shaped workpiece comprises: a table (21) having a conveyor mechanism (23) for conveying a plate-shaped workpiece (10) in the horizontal direction; a camera (56) for taking an image of the plate-shaped workpiece (10) placed on the table (21); a table movement mechanism (28) for moving the table (21) horizontally in the direction orthogonal to the conveyance direction; a table rotation mechanism (60) for rotating the table (21) about a vertical axis; a computation unit (58) for comparing target central position information and information acquired by the camera (56) and computing the movement amount of the conveyor mechanism (23), the movement amount of the table movement mechanism (28), and the movement amount of the table rotation mechanism (60); and a control unit (59) for controlling the conveyor mechanism (23), the table movement mechanism (28), and the table rotation mechanism (60) on the basis of the movement amounts calculated by the computation unit (58).

Abstract: An audio processing device includes a setting section that sets a reproduction sampling frequency Fplay and a recording sampling frequency Frec higher than Fplay, a digital-to-analog converter that based on Fplay converts a sound source signal that is a digital signal into a reproduction signal that is an analog signal, an analog-to-digital converter that based on Frec converts a recording signal that is an analog signal converter into an input signal that is a digital signal, a signal separator that separates the input signal into a low region signal contained in a band of less than Fplay and a high region signal contained in a band of the Fplay and higher, and a breakup detector that detects whether or not breakup is occurring in the reproduced sound based on power of the high region signal.

Abstract: An audio signal encoding apparatus includes a processor to quantize an audio signal, to obtain a characteristic of reverberation masking that is exerted on a sound represented by the audio signal by reverberation of the sound generated in a reproduction environment by reproducing the sound, and to control a quantization step size of the audio signal that quantized based on the characteristic of the reverberation masking.

Abstract: An audio processing device includes a reverb property estimating unit that estimates a reverb property at each frequency on the basis of a first audio signal and a second audio signal representing sounds of the first audio signal output by an audio output unit and collected by an audio input unit, a gain calculating unit that determines an attenuating ratio for a component of the first audio signal at each frequency such that the larger the reverb property at the frequency is, the larger the attenuating ratio for the component at the frequency becomes, and a correcting unit that attenuates the first audio signal at the each frequency in accordance with the attenuating ratio determined for each frequency.

Abstract: A voice processing device includes: a processor; and a memory which stores a plurality of instructions, which when executed by the processor, cause the processor to execute, receiving a first signal including a plurality of voice segments; controlling such that a non-voice segment with a length equal to or greater than a predetermined first threshold value exists between at least one of the plurality of voice segments; and outputting a second signal including the plurality of voice segments and the controlled non-voice segment.

Abstract: A voice control device includes a calculation section configured to calculate a response time representing a time difference between a voice in a received signal and a voice in a sending signal; a hearing estimate section configured to estimate hearing of a user based on the calculated response time; and a voice control section configured to control the received signal by a compensation quantity responsive to the estimated hearing.

Abstract: A sleep apnea syndrome testing apparatus includes: an analyzing unit that analyzes every unit time a sound signal resulting from a subject during sleep; a determining unit that determines whether a unit time of the sound signal includes a breath sound or a characteristic sound produced when a patient with sleep apnea syndrome recovers from an apneic state into a breathing state, and the determining unit that determines that the sleep state of the subject in the unit time is any one of “breathing restored state,” “state with breathing,” and “state without breathing;” a storage unit in which a sleep state of the subject in each unit time is stored; and a detecting unit that detects an apneic state of the subject if a history of the sleep states of the subject indicates at least a transition from the “state without breathing” to the “breathing restored state.

Abstract: An audio encoding device includes a processor; and a memory which stores a plurality of instructions, which when executed by the processor, cause the processor to execute, calculating first phases indicating phases of a first channel signal and a second channel signal included in audio signals of a plurality of channels; and performing, on the basis of the first phases, either first predictive coding in which a third channel signal included in the audio signals of the plurality of channels is predicted using the first channel signal and the second channel signal or second predictive coding in which the second channel signal is predicted using the first channel signal.

Abstract: A voice control device includes a hearing estimate section configured to estimate hearing of a user based on a sending/received sound ratio representing a ratio of the volume of a sending sound to the volume of a received sound; a compensation-quantity calculating section configured to calculate a compensation quantity for a received signal of the received sound responsive to the estimated hearing; and a compensation section configured to compensate the received signal based on the calculated compensation quantity.