Abstract: A decoding apparatus decodes a first encoded data that is encoded from a low-frequency component of an audio signal, and a second encoded data that is used when creating a high-frequency component of an audio signal from a low-frequency component and encoded in accordance with a certain bandwidth, into the audio signal. In the decoding apparatus, a high-frequency component detecting unit divides the high-frequency component into bands with a certain interval range correspondingly to the certain bandwidth, and detects magnitude of the high-frequency components corresponding to each of the bands. A high-frequency component compensating unit compensates the high-frequency components based on the magnitude of the high-frequency components corresponding to each of the bands detected by the high-frequency component detecting unit.

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: 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: 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-analogue converter that based on Fplay converts a sound source signal that is a digital signal into a reproduction signal that is an analogue signal, an analogue-to-digital converter that based on Frec converts a recording signal that is an analogue 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: In a signal processing method and apparatus, a predetermined correcting signal having a same frame length as a second frame signal in which predetermined processing is performed to a frequency spectrum of a first frame signal of a frame length to which a predetermined window function is performed and is converted into a time domain is adjusted so that amplitudes of both ends of the correcting signal become equal to amplitudes of both or one of frame ends of the second frame signal, and a corrected frame signal is obtained by subtracting an adjusted correcting signal from the second frame signal.

Abstract: A signal 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 speech of a speaker as a first signal; detecting an expiration period included in the first signal; extracting a number of phonemes included in the expiration period; and controlling, a second signal, which is an output to the speaker, on the basis of the number of phonemes and a length of the expiration period.

Abstract: An audio coding 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 a frequency spectrum characteristic of an input signal; defining a scale factor that is used to quantize a frequency spectrum converted from the input signal, based on the frequency spectrum characteristic, for each of a plurality of bands; and quantizing the frequency spectrum based on the scale factor.

Abstract: An audio decoding apparatus and method are provided. The audio decoding apparatus includes a spectrum converting part configured to divide the first frequency spectrum in each channel of the first audio signal in a time direction or in a frequency direction to calculate a first signal sequence having the same time resolution and the same frequency resolution in all the channels of the first audio signal, a down-mixing part configured to perform weighted addition on the signals at the same time and within the same frequency band included in the first signal sequence in all the channels to calculate a second signal sequence having channels of a second number different from the first number of channels.

Abstract: An audio signal processing system including a time-frequency conversion unit which converts an audio signal in time domain into frequency domain in frame units so as to calculate a frequency spectrum of the audio signal, a spectral change calculation unit which calculates an amount of change between a frequency spectrum of a first frame and a frequency spectrum of a second frame before the first frame based on the frequency spectrum of the first frame and the frequency spectrum of the second frame, and a judgment unit which judges the type of the noise which is included in the audio signal of the first frame in accordance with the amount of spectral change.

Abstract: A data embedding device includes a processor; and a memory which stores a plurality of instructions, which when executed by the processor, cause the processor to execute, extracting, from a code book storing a plurality of prediction coefficients, a plurality of candidates of prediction coefficients of a channel signal out of a plurality of channel signals, each candidate having a predictive error falling within a specific range of predictive error of predictive encoding of two other channel signals; and selecting from the extracted candidates a prediction coefficient 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: An audio decoding device includes a processor; and a memory which stores a plurality of instructions, which when executed by the processor, cause the processor to execute, decoding, using a first channel signal and a second channel signal included in a plurality of channels of an audio signal having a first frequency range and a second frequency range, a first prediction coefficient of the first frequency range and a second prediction coefficient of the second frequency range, both selected from a code book when prediction-encoding a third channel signal that is not subjected to prediction encoding and that is included in the plurality of channels; decoding a residual signal included in the first frequency range, the residual signal representing an error occurring in prediction encoding; and prediction-decoding the third channel signal subjected to prediction-encoding in the second frequency range from the first channel signal, the second channel signal.

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 decoded sound analysis unit (104) calculates, regarding the frequency-region stereo signals L(b) and R(b) decoded by the PS decoding unit (103), a second degree of similarity (109) and a second intensity difference (110) from the decoded sound signals. A spectrum correction unit (105) detects a distortion added by the parametric-stereo conversion by comparing the second degree of similarity (109) and the second intensity difference (110) calculated at the decoding side with the first degree of similarity (107) and the first intensity difference (108) calculated and transmitted from the encoding side, and corrects the spectrum of the frequency-region stereo decoded signals L(b) and R(b).

Abstract: An SBR encoder includes a filter bank that receives an input signal, a time/frequency grid generator that controls a number of bits of various parameters, a parameter calculator that calculates various parameters, a parameter coding unit that encodes the parameters, an upper-limit number-of-bit storage unit that stores an upper limit of the number of bit of encoded data of high-frequency component finally generated in a high-pass encoding process, and a number-of-bit controller. The number-of-bit controller controls the high-pass encoding process by preferentially encoding a parameter having a large influence to sound quality and not encoding a parameter having a small influence to the sound quality relative to a plurality of parameters, so that the number of bits of the encoded data of high-frequency component finally generated in the high-pass encoding process becomes equal to or less than the upper limit to be stored in the upper-limit number-of-bit storage unit.

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 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.

Abstract: A portable terminal device includes a vibrator configured to vibrate a housing; an analysis section configured to execute frequency analysis on an input sound from a microphone; a calculating section configured to calculate an amount of temporal change of a spectrum obtained by the analysis section for the input sound when the housing vibrates; and a control section configured to execute incoming-call control depending on the amount of temporal change calculated by the calculating section.

Abstract: An acoustic signal processing apparatus includes a processor; and a memory. The processor executes: detecting an acoustic signal that is contained in a sound input signal and that is generated by a human body; calculating an effective volume indicating an effective volume of the acoustic signal detected at the detecting the acoustic signal, based on the input signal; calculating an accumulated effective volume by accumulating effective volumes calculated at the calculating the effective volume; and determining whether the accumulated effective volume calculated at the calculating the accumulated effective volume has reached a predetermined threshold.

Abstract: A sensing apparatus which controls the detection timing of a sensor by a prescribed guidance signal includes a guidance signal analyzing unit for analyzing the prescribed guidance signal, and a sensing control unit for determining, based on the result of the detection, an effective detection period during which an output of the sensor is effective.

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.