Abstract: A noise estimation parameter learning device is provided according to which even in a large space causing a problem of the reverberation and the time frame difference, multiple microphones disposed at distant positions cooperate with each other, and a spectral subtraction method is executed, thereby allowing the target sound to be enhanced.

Abstract: A range finder 10 according to the present embodiment includes a first sensor 440 configured to detect an attitude of a body of the range finder, a correction lens 410 configured to be driven based on a shaking amount detected by a shaking detection sensor 442, a second sensor 450 configured to detect a position of the correction lens, and a processing unit 300 configured to determine an irradiating angle of light. The processing unit can determine, based on a detection result of the attitude of the body by the first sensor and a detection result of the position of the correction lens by the second sensor, the irradiating direction (i.e. measurement direction) in which the light deflected by the correction lens which position is corrected according to the shaking amount of the attitude of the body determined from the detection result of the second sensor is actually irradiated.

Abstract: A noise estimation parameter learning device is provided according to which even in a large space causing a problem of the reverberation and the time frame difference, multiple microphones disposed at distant positions cooperate with each other, and a spectral subtraction method is executed, thereby allowing the target sound to be enhanced.

Abstract: A range finder including a sighting optical system that forms an optical image of a sighting target by sighting a target object and includes a correcting member in an optical path thereof that is driven to correct image blur of an optical image; a driving section that drives the correcting member based on a shaking amount applied to the sighting optical system; a light transmitting section that emits measurement light to a sighting target; a light receiving section that receives returned light from the sighting target and outputs a received light signal; a distance calculating section that calculates distance to the sighting target based on a timing when the measurement light is output and a timing when the return light is received by the light receiving section; and a power changing section that controls a focal distance of the light receiving section according to driving of the correcting member.

Abstract: A distance measuring apparatus includes: a light-transmitting unit that transmits signal light from a light source toward a target object; a light-receiving unit that has a light-receiving element and receives the signal light from the target object on an optical axis different from that of the light-transmitting unit; a distance measuring unit that measures a distance to the target object based on propagation time from light transmission to light reception of the signal light; and a compensating unit that, while either optical path of an optical path from the light-transmitting unit to the light-receiving unit or an optical path from the target object to the light-receiving element is fixed, compensates an optical path of the signal light by displacing the other optical path at the light-transmitting unit or the light-receiving unit.

Abstract: A range finder including a sighting optical system that forms an optical image of a sighting target by sighting a target object and includes a correcting member in an optical path thereof that is driven to correct image blur of an optical image; a driving section that drives the correcting member based on a shaking amount applied to the sighting optical system; a light transmitting section that emits measurement light to a sighting target; a light receiving section that receives returned light from the sighting target and outputs a received light signal; a distance calculating section that calculates distance to the sighting target based on a timing when the measurement light is output and a timing when the return light is received by the light receiving section; and a power changing section that controls a focal distance of the light receiving section according to driving of the correcting member.

Abstract: A distance measuring apparatus includes: a light-transmitting unit that transmits signal light from a light source toward a target object; a light-receiving unit that has a light-receiving element and receives the signal light from the target object on an optical axis different from that of the light-transmitting unit; a distance measuring unit that measures a distance to the target object based on propagation time from light transmission to light reception of the signal light; and a compensating unit that, while either optical path of an optical path from the light-transmitting unit to the light-receiving unit or an optical path from the target object to the light-receiving element is fixed, compensates an optical path of the signal light by displacing the other optical path at the light-transmitting unit or the light-receiving unit.

Abstract: The present application provides a distance measuring apparatus that can measure a distance with accuracy even when a light receiving level of a reflected light becomes saturated. A light emitting section of the distance measuring apparatus emits a measuring light toward an object to be measured. A light receiving section receives the reflected light being reflected from the object to be measured. A distance calculating section obtains a distance to the object to be measured based on an elapsed time which is from a point the measuring light is emitted until a point a light receiving level of the reflected light indicates a peak thereof. A distance correcting section corrects a value of the distance to the object to be measured depending on a length of saturation time of the light receiving level when the light receiving level becomes saturated and the peak is impossible to identify.

Abstract: A coding method with a small error is provided. In the coding method of the present invention, a normalization value obtained from an input signal is corrected for an error calculated from an input and output in vector quantization and is then quantized. The coding method includes a normalization stage of normalizing the input signal in accordance with the normalization value of the input signal, calculated in each frame; a dividing stage of dividing the normalized frame into divided input signal sequences in accordance with a predetermined rule; a vector quantization stage of applying vector quantization to the divided input signal sequences to generate a vector quantization index; and a normalization value correction stage of correcting the normalization value of the input signal for the error obtained from the input and output in the vector quantization stage.

Abstract: The present application provides a distance measuring apparatus that can measure a distance with accuracy even when a light receiving level of a reflected light becomes saturated. A light emitting section of the distance measuring apparatus emits a measuring light toward an object to be measured. A light receiving section receives the reflected light being reflected from the object to be measured. A distance calculating section obtains a distance to the object to be measured based on an elapsed time which is from a point the measuring light is emitted until a point a light receiving level of the reflected light indicates a peak thereof. A distance correcting section corrects a value of the distance to the object to be measured depending on a length of saturation time of the light receiving level when the light receiving level becomes saturated and the peak is impossible to identify.

Abstract: When acoustic signal packets are communicated over an IP communication network, data corresponding to an acoustic signal (acoustic signal corresponding data) has been included and transmitted in a packet different from a packet containing the acoustic signal. However, conventionally, a packet in which the acoustic signal corresponding data is to be included must be determined beforehand and cannot dynamically be changed. According to the present invention, the amount of delay of acoustic signal corresponding data with respect to an acoustic signal is contained in an acoustic signal packet as delay amount control information. Furthermore, the conditions of a communication network are detected from the number of packets lost in a burst loss or jitters and the number of the packets to be stored and the amount of delay at the receiving end are thereby determined.

Abstract: A coding method with a small error is provided. In the coding method of the present invention, a normalization value obtained from an input signal is corrected for an error calculated from an input and output in vector quantization and is then quantized. The coding method includes a normalization stage of normalizing the input signal in accordance with the normalization value of the input signal, calculated in each frame; a dividing stage of dividing the normalized frame into divided input signal sequences in accordance with a predetermined rule; a vector quantization stage of applying vector quantization to the divided input signal sequences to generate a vector quantization index; and a normalization value correction stage of correcting the normalization value of the input signal for the error obtained from the input and output in the vector quantization stage.

Abstract: The present invention prevents a receiving buffer from becoming empty by: storing received packets in the receiving buffer; detecting the largest arrival delay jitter of the packets and the buffer level of the receiving buffer by a state detecting part; obtaining an optimum buffer level for the largest delay jitter using a predetermined table by a control part; determining, based on the detected buffer level and the optimum buffer level, the level of urgency about the need to adjust the buffer level; expanding or reducing the waveform of a decoded audio data stream of the current frame decoded from a packet read out of the receiving buffer by a consumption adjusting part to adjust the consumption of reproduction frames on the basis of the urgency level, the detected buffer level, and the optimum buffer level.

Abstract: Input speech is coded in an encoder (11), the coded speech is decoded in a decoder (12), compensatory speech which compensates the speech of the current frame is generated in a compensatory speech generating part (20) by using past decoded speech, the quality of the compensatory speech is evaluated by using the input speech and the compensatory speech and a duplication level is generated the value of which increases incrementally with decreasing speech quality evaluation value in a speech quality evaluating part (40), and as many identical packets as the number specified by the duplication level is generated for the coded speech in a packet generating part (15), and the packets are transmitted, thereby reducing the possibility that packet loss will occur at the receiving end.

Abstract: When acoustic signal packets are communicated over an IP communication network, data corresponding to an acoustic signal (acoustic signal corresponding data) has been included and transmitted in a packet different from a packet containing the acoustic signal. However, conventionally, a packet in which the acoustic signal corresponding data is to be included must be determined beforehand and cannot dynamically be changed. According to the present invention, the amount of delay of acoustic signal corresponding data with respect to an acoustic signal is contained in an acoustic signal packet as delay amount control information. Furthermore, the conditions of a communication network are detected from the number of packets lost in a burst loss or jitters and the number of the packets to be stored and the amount of delay at the receiving end are thereby determined.

Abstract: The present invention prevents a receiving buffer from becoming empty by: storing received packets in the receiving buffer; detecting the largest arrival delay jitter of the packets and the buffer level of the receiving buffer by a state detecting part; obtaining an optimum buffer level for the largest delay jitter using a predetermined table by a control part; determining, based on the detected buffer level and the optimum buffer level, the level of urgency about the need to adjust the buffer level; expanding or reducing the waveform of a decoded audio data stream of the current frame decoded from a packet read out of the receiving buffer by a consumption adjusting part to adjust the consumption of reproduction frames on the basis of the urgency level, the detected buffer level, and the optimum buffer level.

Abstract: Input speech is coded in an encoder (11), the coded speech is decoded in a decoder (12), compensatory speech which compensates the speech of the current frame is generated in a compensatory speech generating part (20) by using past decoded speech, the quality of the compensatory speech is evaluated by using the input speech and the compensatory speech and a duplication level is generated the value of which increases incrementally with decreasing speech quality evaluation value in a speech quality evaluating part (40), and as many identical packets as the number specified by the duplication level is generated for the coded speech in a packet generating part (15), and the packets are transmitted, thereby reducing the possibility that packet loss will occur at the receiving end.

Abstract: An excitation vector of the previous frame stored in an adaptive codebook is cut out with a selected pitch period. The excitation vector thus cut out is repeated until one frame is formed, by which a periodic component codevector is generated. An optimum pitch period is searched for so that distortion of a reconstructed speech obtained by exciting a linear predictive synthesis filter with the periodic component codevector is minimized. Thereafter, a random codevector selected from a random codebook is cut out with the optimum pitch period and is repeated until one frame is formed, by which a repetitious random codevector is generated. The random codebook is searched for a random codevector which minimizes the distortion of the reconstructed speech which is provided by exciting the synthesis filter with the repetitious random codevector.