Panasonic Intellectual Property Management Co., Ltd.

Matsushita Electric Industrial Co., Ltd.

Panasonic Corporation

Panasonic Intellectual Property Corporation of America

Panasonic Automotive Systems Co., Ltd.

Matsushita Electric Industrial

Yutaka Banba

Takeo Kanamori

Tomofumi Yamanashi

Shohei Taniguchi

Shinichi Yuzuriha

Masamichi Ohara

Misaki Tsujikawa

Masayuki Ito

Yasuhiro Terada

Yoshitaka Seto

An audio processing system includes at least one first microphone, at least one adaptive filter, and a processor. The at least one first microphone acquires a first audio signal and outputs a first signal based on the first audio signal. The first audio signal includes at least one of a first audio component generated at a first position and a second audio component generated at a second position different from the first position. The first signal is input to the at least one adaptive filter. The at least one adaptive filter outputs a passing signal based on the first signal. The processor, when executing a program stored in a memory, performs: making a determination of which of the first audio component and the second audio component the first audio signal includes more; and controlling a filter coefficient of the adaptive filter based on a result of the determination.

Audio processing system, audio processing device, and audio processing method

An audio processing system includes: a first microphone configured to output a first signal based on a first audio signal; one or more microphones each of which outputs a microphone signal based on an audio signal; one or more adaptive filters configured to respectively receive the microphone signals from the one or more microphones and output passing signals based on the microphone signals; and a processor configured to: determine whether the microphone signal includes uncorrelated noise; control one or more filter coefficients of the one or more adaptive filters; and subtract a subtraction signal based on the passing signals from the first signal. The one or more microphones include a second microphone that outputs a second signal. When determining that the second signal includes the uncorrelated noise, the processor is configured to set a level of the second signal input to the corresponding adaptive filter to zero.

Audio processing system and audio processing device

The speaker identification system has a voice acquisition unit that acquires voice information of a speaker, and a database management unit that determines whether or not the speaker corresponding to the acquired voice information matches a speaker corresponding to registered voice information in connection with content information on a content, that acquires content information on the content displayed on a device at the time of acquisition of the voice information and stores the acquired content information in connection with the registered voice information in a case where it is determined that the speaker corresponding to the acquired voice information matches the speaker corresponding to the registered voice information, and that stores the acquired voice information in the database as registered voice information in a case where it is determined that the speaker corresponding to the acquired voice information does not match the speaker corresponding to the registered voice information.

Speaker identification method, speaker identification apparatus, and information management method

A hearing aid capable of detecting contact vibration noise from a collected sound signal. The hearing aid is provided with two microphones, a vibration component extracting section that extracts from collected sound signals respectively obtained by the two microphones an uncorrelated component between two collected sound signals as a vibration component for each frequency band. Additionally, a vibration noise identifying section determines whether or not a contact noise occurs based on the vibration component for each frequency band extracted by the vibration component extracting section, an acoustic signal processing section, when generating an acoustic signal by hearing aid processing of the two collected sound signals, processes the acoustic signal depending on the presence or absence of the occurrence of the contact vibration noise, and a receiver converts the acoustic signal to sound.

Hearing aid and method of detecting vibration

A sound processing apparatus () is provided with: a directivity synthesis processing unit () for generating a first directivity sound pick-up signal by synthesizing a first sound pick-up signal and a relatively delayed second sound pick-up signal and a second directivity sound pick-up signal by synthesizing a relatively delayed first sound pick-up signal and a second sound pick-up signal; a comparison signal calculation unit () for generating a non-directivity level signal indicating the level of a sum of the directivity sound pick-up signals and a directivity level signal by adding the levels of the directivity sound pick-up signals; a level comparison unit () for acquiring the difference between the levels of the non-directivity level signal and the directivity level signal; and a delay control unit () for adjusting the delay amount such that the difference between the levels becomes smaller.

Sound processing device, and sound processing method

A sound processing apparatus which can improve precision of analyzes on ambient sounds, carries out analysis on the ambient sounds based upon collected sound signals acquired by two sound collectors. The sound processing apparatus is provided with a level signal converter that converts the collected sound signal into a level signal, which indicates an absolute value of the collected sound signal from which phase information is removed. A level signal synthesizer generates a synthesized level signal in which the level signals acquired from the collected sound signals of the two sound collectors are synthesized, and a detector/identifier carries out analysis on the ambient sounds, based upon the synthesized level signal.

Sound processing device and sound processing method

A noise cancelling device includes an extracting unit configured to extract a first noise from a signal, the signal being based on an input audio signal, a storing unit configured to store noise characteristic information on a second noise, the second noise remaining after subtracting the extracted first noise from the signal based on the audio signal. And the device further includes a cancelling unit configured to perform cancelling processing for cancelling a noise on the input audio signal based on the first noise and the noise characteristic information on the second noise.

Noise cancelling device

A power spectrum estimation unit () obtains an estimated sound power spectrum P(&#x3c9;), based on a power spectrum P(&#x3c9;) and on a first calculated value obtained by at least multiplying a power spectrum P(&#x3c9;) by a weight coefficient A(&#x3c9;). A coefficient update unit () updates the weight coefficient A(&#x3c9;) and a weight coefficient A(&#x3c9;) so that a second calculated value approximates to the power spectrum P(&#x3c9;). The second calculated value is obtained by adding at least two values obtained by multiplying the power spectrum P(&#x3c9;) and the estimated target sound power spectrum P(&#x3c9;) by the weight coefficient A(&#x3c9;) and the weight coefficient A(&#x3c9;), respectively.

Multi-input noise suppression device, multi-input noise suppression method, program thereof, and integrated circuit thereof

An audio encoder converts an input sound signal into a plurality of compressed frame data pieces in an sound signal compression coder, determines the importance of each bit in a classification unit of a transmission line coder based on the decoding quality in the presence of a transmission error, and classifies the bits into a plurality of classes. The audio encoder selects one of the three types of processing including convolution coding and addition of CRC check codes, convolution coding only, and no coding, in descending order of importance in the presence of a transmission error for each class. Then, the audio encoder adds preamble information and a synchronization signal in a multiplexer to generate a bit stream. It becomes possible to suppress degradation of a decoded sound signal without additional redundant bits.

Transmission line coding method, transmission line decoding method, and apparatus therefor

An audio signal coding device, an audio signal decoding device and a method to improve audio quality. The audio signal coding device and method include a quantizer that quantizes a given signal according to a number of assigned bits in order to generate a codeword. The coding device includes an extractor that extracts core bits from the generated codeword. The coding device also includes a determiner that determines an optimal value of the number of assigned bits based on an energy level corresponding to the extracted core bits. The audio signal decoding device and method include a dequantizer that dequantizes a given codeword according to the number of assigned bits to generate a decoded signal. The decoding device includes an extractor that extracts core bits from the given codeword. The decoding device also includes a determiner that determines an optimal value of the number of assigned bits used in the dequantizer.

Speech coding and decoding apparatus and method with number of bits determination

An apparatus having a band-separating filter bank for separating a digital signal into a plurality of sub-band signals, to be processed or transmitted, and a band-combining filter bank for subsequently combining the resultant sub-band signals into a single digital signal, wherein each of the band-separating filter bank and band-combining filter bank incorporates a FIR low pass filter having an asymmetric impulse response, as the prototype filter of the filter bank. A significant reduction can thereby be achieved in the amount of overall group delay that results from the processing performed by these filter banks.

Digital signal sub-band separating/combining apparatus achieving band-separation and band-combining filtering processing with reduced amount of group delay

An input digital audio signal is divided into sub-band signals in respective sub-bands. Scale factors of the respective sub-bands are determined on the basis of the sub-band signals for every frame. Calculation is made as to differences between the determined scale factors for a first frame and the determined scale factors for a second frame preceding the first frame. Absolute values of the calculated scale-factor differences are calculated, and data representative of the calculated absolute values are generated. The data representative of the calculated absolute values are encoded into data of a Huffman code. Sign bits are generated which represent signs of the calculated scale-factor differences. The sub-band signals are quantized in response to the determined scale factors for every frame to generate quantized samples of the sub-band signals. The Huffman-code data, the generated sign bits, and the quantized samples of the sub-band signals are combined into a bit stream.

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Kanagawa, Japan

Yutaka Banba