Abstract: A noise reduction apparatus includes an analysis unit for converting input into a signal of a frequency area, a suppression unit for suppressing the signal, and a synthesis unit for synthesizing a signal of a time area. The apparatus further includes an estimation unit for estimating, using the output of the analysis unit, information corresponding to at least pure voice element excluding noise element in an input voice signal as voice information which is the basic voice information for calculation of a suppression gain of a signal, and a unit for calculating a suppression gain corresponding to the output of the estimation unit and the analysis unit and providing it for the suppression unit.

Abstract: A quality level analysis unit 504 specifies a coding method, communication line conditions, and a quality level, such as an S/N ratio and the like, of voices of an input channel, received by a corresponding receiving unit 501. A channel allocation/mixing unit 502 controls the allocation or mixing of voices of respective input channels to or into respective output channels 503 (output units 505) on the basis of the results of analysis by the quality level analysis unit 504. Consequently, loud speakers from which poor quality voices and good quality voices are output can be separated from one another, thus improving the total intelligibility of received voices.

Abstract: An active silencer includes: a speaker generating control sound which interferes with noise; a microphone detecting noise remaining after the interference as a remaining noise signal; a sound quality evaluation unit evaluating the sound quality of the remaining noise and output a result of the sound quality evaluation; an actuation signal determination unit determining, according to the result of the sound quality evaluation, the detection timing of the frequency component of the remaining noise signal to be used when the control sound is generated for a plurality of bands of the remaining noise, corresponding to the plurality of bands of a reference signal corresponding to the noise; and a control signal generation unit generating and output a control signal for generation of the control sound depending on a plurality of bands of the determined remaining noise signal and a plurality of bands of the reference signal corresponding to the noise.

Abstract: A sound processing method includes transforming an input signal from a time domain to a frequency domain to produce a spectrum, detecting a peak of the spectrum, calculating a target attenuation amount based on one of the input signal and the spectrum, calculating attenuation amounts of respective frequency components of the spectrum based on the target attenuation amount and the detected peak, correcting levels of the spectrum by attenuating the spectrum in response to the calculated attenuation amounts of respective frequency components, and performing inverse frequency transform with respect to the level-corrected spectrum to produce an output signal.

Abstract: A system for assessing a voice communication quality of a communication path between first and second nodes over a network, wherein coded data of voice communication signals are transferred in a stream of packets via the communication path, including: a capturing unit for capturing at the first node at least one packet containing coded data representing non voice signals among the packets of the coded data to be transferred from the first node to the second node; a replacing unit for replacing a part of the coded data representing non voice signals in the captured packet with a predetermined code before the captured packet is transferred from the first node; a retrieval unit for retrieving at the second node said at least one packet containing coded data representing non voice signals; and an assessment unit for assessing the voice communication quality of the communication path.

Abstract: A voice processing apparatus, which processes a first voice signal, includes: an acoustic analysis part which analyzes a feature quantity of an input second voice signal; a reference range calculation part which calculates a reference range based on the feature quantity; a comparing part which compares the feature quantity and the reference range and outputs a comparison result; and a voice processing part which processes and outputs the input first voice signal based on the comparison result.

Abstract: The invention relates to speech speed conversion, and provides a speech speed converting device and a speech speed converting method for changing a speed of voice without degrading the voice quality, without changing characteristics, regarding a signal containing voice. The speech speed converting device includes: a voice classifying unit that is input with voice waveform data and a voice code based on a linear prediction, and that classifies the input signal based on the characteristic of the input signal; and a speed adjusting unit that selects either one of or both a speed conversion processing using the voice waveform and a speed conversion processing using the voice code, based on the classification, and that changes a speech speed of the input signal using the selected speed converting method.

Abstract: A clipping detection device calculates an amplitude distribution of an input signal for each predetermined period, calculates a deflection degree of the distribution on the basis of the calculated amplitude distribution, and then detects clipping of a communication signal on the basis of the calculated deflection degree of the distribution.

Abstract: A voice communication system having, on a transmission side, a data embedding apparatus provided with an embedding allowability judgment unit (41) calculating an analysis parameter with respect to an input audio signal and judging based on the analysis parameter whether there is a part of the input audio signal allowing embedding of data and an embedding unit (42) outputting an audio signal having the data embedded in the allowable part when the result of judgment of the embedding allowability judgment unit is data can be embedded and outputting the audio signal as is when the result of judgment of the embedding allowability judgment unit is data cannot be embedded and having, on the receiving side, a data extraction apparatus provided with a data extraction apparatus extracting data by a reverse operation is provided, whereby data can be embedded in voice signals without causing an unallowable change in audio quality and a drop in amount of embedded data due to embedding data in parts unsuitable for embeddin

Abstract: In a voice packet communication system, a voice packet loss concealment device compensates for the deterioration of voice quality due to voice packet loss. In the device, a detecting section detects a loss of a voice packet and outputting information; an estimating section estimates the voice characteristics of the lost segment using a pre-loss voice packet received before the lost segment or a post-loss voice packet received after the lost segment; a pitch signal generating section generates a pitch signal having the voice characteristics; and a lost packet generating section outputs the pitch signal generated by the pitch signal generating section, with the voice characteristics estimated by the estimating section, which allows abnormal noise and feeling of mute, subjective deterioration of naturalness and continuity to be improved, and the voice packet loss concealment to be further improved.

Abstract: A disclosed speech sound enhancement device includes an SNR calculation unit for calculating an SNR which is a ratio of received speech sound to ambient noise; a first-frequency-range enhancement magnitude calculation unit for calculating, based on the SNR and frequency-range division information indicating a first and a second frequency range, enhancement magnitude of the first frequency range, the first frequency range contributing to an improvement of subjective intelligibility of the received speech sound, the second frequency range contributing to an improvement of subjective articulation of the received speech sound; a second-frequency-range enhancement magnitude calculation unit for calculating enhancement magnitude of the second frequency range based on the enhancement magnitude of the first frequency range; and a spectrum processing unit for processing spectra of the received speech sound using the enhancement magnitude of the first frequency range, the enhancement magnitude of the second frequency r

Abstract: As optimal candidate as a control signal (y*) for generating a control sound suppressing noise from a speaker is selected from among a plurality of control signal candidates (y1 to yn) by a selection function unit. For this selection, a residual noise estimation function unit receiving as input a residual noise signal (e) from an error microphone is introduced. The function unit first obtains an estimated value of a noise component using a first transfer characteristic simulating filter. Further, this noise component estimated value and filtered outputs from second transfer characteristic simulating filters are used to obtain residual noise estimated values for the control signal candidates (y1 to yn). Further, the single control signal candidate corresponding to the smallest of these residual noise estimated values is selected and used as the above control signal (y*).

Abstract: An active noise control apparatus that controls by a control sound a noise which is output from a noise source, includes: a control sound generating section which inputs a control signal, and produce the control sound; a residual noise detecting section which detects, as a residual noise signal, a noise remaining after the noise control by the control sound; a control signal generating section which inputs, as a reference signal, a signal concerning the noise or the generation state of the noise, and generates the control signal; and a controlling section which inputs the control signal and the residual noise signal, detects the components that cannot be identified in the control signal generating section, and controls the generation of the control signal in the control signal generating section.

Abstract: A code separation/decoding unit restores a vocal tract characteristic sp1 and a vocal source signal r1. A vocal tract characteristic modification unit modifies the vocal tract characteristic sp1 and outputs the modified vocal tract characteristic sp2. In this method, an emphasized vocal tract characteristic sp2 is generated to output by applying formant emphasis, using amplification ratios calculated based on estimated formants, directly to the vocal tract characteristic sp1 for instance. A signal synthesis unit synthesizes the modified vocal tract characteristic sp2 and the vocal source signal r1 to generate and output an output voice, s.

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: A method for identifying among a plurality of devices, through which a voice communication path is connected, a device transmitting an echo signal onto the voice communication path, includes transmitting a predetermined signal to each of the devices from a predetermined portion on the voice communication path, receiving a response signal for the predetermined signal from each of the devices, measuring a period of time for the predetermined signal to travel from the predetermined portion to each of the devices and for the response signal to travel back to the predetermined portion, monitoring a upstream signal and a downstream signal traveling on the voice communication path, extracting an eco component from the downstream signal to determine an echo delay time, and comparing the period of time and the echo delay time to determine a device which causes the echo signal.

Abstract: A communication apparatus for adjusting a received voice signal in accordance with an ambient noise, the communication apparatus includes: a microphone for receiving an ambient noise and input voice and outputting a voice input signal corresponding to a level of the input voice and the ambient noise; a receiver for receiving the voice signal; a processer for extracting a voice component originated by a sender and an ambient noise component originated by the ambient noise, determining the ratio between the voice component and the ambient noise component, and adjusting the amplitude of the received voice signal in accordance with the ratio; and a speaker for outputting a reception voice corresponding to the adjusted reception voice signal.

Abstract: A device for sleep apnea detection, includes an external sound recorder that records external sound in environment, a sound/silence determining unit that determines whether an audible sound or no sound is found in the external sound recorded in the external sound recorder, a breathing pace analyzing unit that analyzes, using information of intervals of the audible sound and no sound found by the sound/silence determining unit and the external sound, a breathing pace indicating a cycle of a breathing estimated interval during which breathing is estimated, and an apnea interval extracting unit that extracts a silent interval within the breathing estimated interval based on the breathing pace analyzed by the breathing pace analyzing unit.

Abstract: It is so arranged that a voice code can be converted even between voice encoding schemes having different subframe lengths. A voice code conversion apparatus demultiplexes a plurality of code components (Lsp1, Lag1, Gain1, Cb1), which are necessary to reconstruct a voice signal, from voice code in a first voice encoding scheme, dequantizes the codes of each of the components and converts the dequantized values of code components other than an algebraic code component to code components (Lsp2, Lag2, Gp2) of a voice code in a second voice encoding scheme. Further, the voice code conversion apparatus reproduces voice from the dequantized values, dequantizes codes that have been converted to codes in the second voice encoding scheme, generates a target signal using the dequantized values and reproduced voice, inputs the target signal to an algebraic code converter and obtains an algebraic code (Cb2) in the second voice encoding scheme.

Abstract: A voice synthesizer, which obtains a voice by emphasizing a specific part of a sentence, includes an emphasis degree deciding unit that extracts a word or a collocation to be emphasized from among respective words or respective collocations on the basis of an extracting reference with respect to the each word or the each collocation included in a sentence and deciding an emphasis degree of the extracted word or the extracted collocation, an acoustic processing unit that synthesizes a voice having an emphasis degree which is decided by the emphasis degree deciding unit applied to the word to be emphasized or the collocation to be emphasized, whereby the emphasized part of the word or the collocation can be obtained automatically on the basis of the extracting reference, such as a frequency of appearance and a level of importance of the word or the collocation.