Patents by Inventor Tadashi Amada
Tadashi Amada has filed for patents to protect the following inventions. This listing includes patent applications that are pending as well as patents that have already been granted by the United States Patent and Trademark Office (USPTO).
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Publication number: 20230409203Abstract: A memory system includes a nonvolatile memory and a controller. The controller is configured to determine a first predicted read address as a subsequent read address following an input read address from which data is to be read, based on the input read address and a preset write sequence rule, determine a second predicted read address as the subsequent read address, based on the input read address and a read sequence history, select one of read addresses including the first and second predicted read addresses as a target read address, and read data from the target read address of the nonvolatile memory.Type: ApplicationFiled: March 2, 2023Publication date: December 21, 2023Inventor: Tadashi AMADA
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Publication number: 20200303016Abstract: A memory reading method includes reading data from a memory cell array of a nonvolatile memory where data is written after randomization, using a first read voltage, counting a total number of bits set to either 1 or 0 in the read data, performing an error correction on the read data, and in case of failure of the error correction, retrying reading the data using a second read voltage only when the counted total number of bits falls within a first predetermined range.Type: ApplicationFiled: August 30, 2019Publication date: September 24, 2020Inventor: Tadashi AMADA
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Patent number: 9865279Abstract: According to one embodiment, a method performed by an electronic device includes: receiving an audio signal comprising voice and background sound via a microphone; receiving a user's operation to set a loudness of the voice or the background sound; setting a balance between a first gain of the voice and a second gain of the background sound according to the user's operation; separating the input audio signal into a first signal of the voice and a second signal of the background sound; amplifying the first signal according to the first gain; amplifying the second signal according to the second gain; and outputting the first signal and the second signal at least partially overlapping each other via a speaker.Type: GrantFiled: February 22, 2016Date of Patent: January 9, 2018Assignee: Kabushiki Kaisha ToshibaInventors: Tadashi Amada, Hirokazu Takeuchi
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Publication number: 20160210983Abstract: According to one embodiment, a method performed by an electronic device includes: receiving an audio signal comprising voice and background sound via a microphone; receiving a user's operation to set a loudness of the voice or the background sound; setting a balance between a first gain of the voice and a second gain of the background sound according to the user's operation; separating the input audio signal into a first signal of the voice and a second signal of the background sound; amplifying the first signal according to the first gain; amplifying the second signal according to the second gain; and outputting the first signal and the second signal at least partially overlapping each other via a speaker.Type: ApplicationFiled: February 22, 2016Publication date: July 21, 2016Inventors: Tadashi AMADA, Hirokazu TAKEUCHI
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Patent number: 9375143Abstract: According to one embodiment, an electronic apparatus includes a biological sensor, an extraction module, a state detector, and a transmission controller. The biological sensor generates first biological data. The extraction module extracts one or more first features from the first biological data. The state detector detects a communication state between a communication module and a server. The transmission controller transmits at least one of the first biological data and the one or more first features to the server, based on the detected communication state.Type: GrantFiled: November 6, 2013Date of Patent: June 28, 2016Assignee: Kabushiki Kaisha ToshibaInventors: Takaya Matsuno, Shingo Suzuki, Yusaku Kikugawa, Koji Yamamoto, Kentaro Takeda, Takashi Sudo, Tadashi Amada, Yasuhiro Kanishima, Chikashi Sugiura
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Publication number: 20140320307Abstract: According to one embodiment, an electronic apparatus includes a biological sensor, an extraction module, a state detector, and a transmission controller. The biological sensor generates first biological data. The extraction module extracts one or more first features from the first biological data. The state detector detects a communication state between a communication module and a server. The transmission controller transmits at least one of the first biological data and the one or more first features to the server, based on the detected communication state.Type: ApplicationFiled: November 6, 2013Publication date: October 30, 2014Applicant: KABUSHIKI KAISHA TOSHIBAInventors: Takaya Matsuno, Shingo Suzuki, Yusaku Kikugawa, Koji Yamamoto, Kentaro Takeda, Takashi Sudo, Tadashi Amada, Yasuhiro Kanishima, Chikashi Sugiura
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Patent number: 8873766Abstract: According to one embodiment, a sound signal processor includes: a connector; an input module; and a generator. The connector is connectable with an earphone. The input module receives and processes a plurality of sound signals corresponding to sound of a plurality of times output from the earphone, respectively. The generator generates, by using first data indicating a frequency characteristic of a first sound signal among the received and processed sound signals and second data indicating a frequency characteristic of a second sound signal among the received and processed sound signals, correction data correcting a frequency characteristic of the earphone to be a target frequency characteristic set as a target. The first data is used for a first frequency band lower than or equal to a reference. The second data is used for a second frequency band higher than the reference.Type: GrantFiled: January 6, 2012Date of Patent: October 28, 2014Assignee: Kabushiki Kaisha ToshibaInventors: Toshifumi Yamamoto, Tadashi Amada
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Patent number: 8630850Abstract: In one embodiment, a signal processing method is disclosed. The method can perform filter processing of convoluting a tap coefficient in a first signal sequence to generate a second signal sequence. The method can subtract the second signal sequence from a third signal sequence to generate a fourth signal sequence. The third signal sequence includes an echo signal of the first signal sequence. The method can correct the tap coefficient in accordance with an amount of correction determined using a function. The function includes at least one of a first region and a second region, and has values limited. The first region is included in a negative value region of the fourth signal sequence. The second region is included in a positive value region of the fourth signal sequence.Type: GrantFiled: September 22, 2011Date of Patent: January 14, 2014Assignee: Kabushiki Kaisha ToshibaInventors: Kaoru Suzuki, Tadashi Amada
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Publication number: 20130329899Abstract: According to one embodiment, a measuring apparatus includes: an output module configured to temporally exclusively output, to a measuring target system, a first output signal corresponding to a first measuring signal for sweeping frequency and a second output signal corresponding to a second measuring signal for sweeping frequency and having a different amplitude characteristic from an amplitude characteristic of the first measuring signal; and a frequency characteristic computation module configured to synthesize a first frequency amplitude spectrum obtained from a first reception signal when a sound output from the measuring target system based on the first output signal is received and a second frequency amplitude spectrum obtained from a second reception signal when a sound output from the measuring target system based on the second output signal is received to generate frequency characteristic data representing an acoustic characteristic.Type: ApplicationFiled: February 27, 2013Publication date: December 12, 2013Applicant: KABUSHIKI KAISHA TOSHIBAInventors: Kimio Miseki, Tadashi Amada, Toshifumi Yamamoto
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Publication number: 20130223636Abstract: According to one embodiment, a measurement apparatus is configured to supply, to a output device, a second measurement signal to sweep the frequency, to receive a second received signal output from the output device when the second measurement signal is supplied to the output device, and to calculate an impulse response by convolving the second received signal and an inverted measurement signal having an inverted characteristic of the second measurement signal. The second measurement signal has a signal characteristic obtained by multiplying the first measurement signal by a characteristic of a characteristic of the amplitude spectrum.Type: ApplicationFiled: November 8, 2012Publication date: August 29, 2013Inventors: Tadashi Amada, Toshifumi Yamamoto, Kimio Miseki
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Patent number: 8522263Abstract: In one embodiment, there is provided an audio signal processor. The processor includes: a person position detector configured to detect each position of one or more persons present in a specific space; a grouping module configured to allocate the detected persons to one or more groups, wherein the number of the groups is less than a given number; a plurality of directionality controllers configured to control directionality of a microphone array; and a directionality setting module configured to set directionality of each of the groups in a corresponding one of the directionality controllers.Type: GrantFiled: June 29, 2011Date of Patent: August 27, 2013Assignee: Kabushiki Kaisha ToshibaInventor: Tadashi Amada
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Patent number: 8503697Abstract: According to one embodiment, a pickup signal processing apparatus includes microphones, a sound determining unit, a signal level calculating unit, a setting unit, and a calculating unit. The sound determining unit determines whether pickup signals picked up by the microphones are signals from a neighboring sound source or a background noise signal. The signal level calculating unit calculates the signal levels for the microphones. The setting unit sets a gain value of at least one microphone and reduces a difference between the signal levels for the microphones on the basis of the signal levels for the microphones, when determined that the pickup signal is the background noise signal. The calculating unit multiplies the pickup signal of the at least one microphone by the gain value set by the setting unit.Type: GrantFiled: August 29, 2011Date of Patent: August 6, 2013Assignee: Kabushiki Kaisha ToshibaInventor: Tadashi Amada
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Publication number: 20130028429Abstract: According to one embodiment, an information processing apparatus includes a measurement module and a correction filter design module. The measurement module measures frequency characteristics of an earphone connected to an output terminal using a measurement audio signal output from the output terminal and collected by a microphone. The correction filter design module designs a correction filter in association with one range of a treble range higher than a crossover frequency range and a bass range lower than the crossover frequency range, based on the measured frequency characteristics of the earphone and predetermined goal frequency characteristics.Type: ApplicationFiled: May 2, 2012Publication date: January 31, 2013Applicant: KABUSHIKI KAISHA TOSHIBAInventors: Tadashi Amada, Toshifumi YAMAMOTO, Kimio MISEKI
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Patent number: 8363821Abstract: According to one embodiment, in response to a first acoustic signal output, a second acoustic signal is input. A filter unit is configured to generate a third acoustic signal by convoluting the first acoustic signal with coefficients. A subtraction unit is configured to generate a fourth acoustic signal by subtracting the third acoustic signal from the second acoustic signal. An estimation unit is configured to decide whether a sound volume of the first acoustic signal is below a predetermined threshold, and to set a sound volume of the second acoustic signal as a non-echo sound level when the sound volume of the first acoustic signal is below the predetermined threshold. A determination unit is configured to determine a step size to correct the coefficients using the non-echo sound level. A correction unit is configured to correct the coefficients using the step size.Type: GrantFiled: September 13, 2010Date of Patent: January 29, 2013Assignee: Kabushiki Kaisha ToshibaInventors: Kaoru Suzuki, Koichi Yamamotop, Tadashi Amada
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Patent number: 8363850Abstract: An audio signal processing method for processing input audio signals of plural channels includes calculating at least one feature quantity representing a difference between channels of input audio signals, selecting at least one weighting factor according to the feature quantity from at least one weighting factor dictionary prepared by learning beforehand, and subjecting the input audio signals of plural channels to signal processing including noise suppression and weighting addition using the selected weighting factor to generate output an output audio signal.Type: GrantFiled: June 9, 2008Date of Patent: January 29, 2013Assignee: Kabushiki Kaisha ToshibaInventor: Tadashi Amada
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Patent number: 8364484Abstract: An input voice detect is detected after starting a voice input waiting state; the detected voice is recognized; an elapsed time from the start of the voice input waiting state is counted; an informative sound which urges a user to input the voice is outputted when the elapsed time reaches a preset output set time; and the output of the informative sound is stopped when the elapsed time at the time of inputting the voice is shorter than the output set timedetect.Type: GrantFiled: April 14, 2009Date of Patent: January 29, 2013Assignee: Kabushiki Kaisha ToshibaInventors: Takehide Yano, Tadashi Amada, Kazunori Imoto, Koichi Yamamoto
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Publication number: 20120275616Abstract: According to one embodiment, a sound signal processor includes: a connector; an input module; and a generator. The connector is connectable with an earphone. The input module receives and processes a plurality of sound signals corresponding to sound of a plurality of times output from the earphone, respectively. The generator generates, by using first data indicating a frequency characteristic of a first sound signal among the received and processed sound signals and second data indicating a frequency characteristic of a second sound signal among the received and processed sound signals, correction data correcting a frequency characteristic of the earphone to be a target frequency characteristic set as a target. The first data is used for a first frequency band lower than or equal to a reference. The second data is used for a second frequency band higher than the reference.Type: ApplicationFiled: January 6, 2012Publication date: November 1, 2012Inventors: Toshifumi Yamamoto, Tadashi Amada
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Publication number: 20120219161Abstract: According to one embodiment, a playback apparatus includes a generator, a corrector, and an audio signal output module. The generator is configured to generate correction data used for correcting a frequency characteristic of sound output from a headphone based on first data and second data, the first data indicating the frequency characteristic of sound output from the headphone, and the second data indicating a target frequency characteristic. The corrector is configured to correct audio data based on the correction data. The audio signal output module is configured to output an audio signal based on the corrected audio data to the headphone.Type: ApplicationFiled: November 11, 2011Publication date: August 30, 2012Inventors: Tadashi Amada, Toshifumi Yamamoto, Kazuyuki Saito
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Patent number: 8229739Abstract: A speech processing apparatus includes a plurality of microphones which receive speech produced by a first sound source to obtain first speech signals for a plurality of channels having one-to-one correspondence with the plurality of microphones, a calculation unit configured to calculate a first characteristic amount indicative of an inter-channel correlation of the first speech signals, a storage unit configured to store in advance a second characteristic amount indicative of an inter-channel correlation of second speech signals for the plurality of channels obtained by receiving speech produced by a second sound source by the plurality of microphones, and a collation unit configured to collate the first characteristic amount with the second characteristic amount to determine whether the first sound source matches with the second sound source.Type: GrantFiled: July 21, 2008Date of Patent: July 24, 2012Assignee: Kabushiki Kaisha ToshibaInventor: Tadashi Amada
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Publication number: 20120124603Abstract: In one embodiment, there is provided an audio signal processor. The processor includes: a person position detector configured to detect each position of one or more persons present in a specific space; a grouping module configured to allocate the detected persons to one or more groups, wherein the number of the groups is less than a given number; a plurality of directionality controllers configured to control directionality of a microphone array; and a directionality setting module configured to set directionality of each of the groups in a corresponding one of the directionality controllers.Type: ApplicationFiled: June 29, 2011Publication date: May 17, 2012Inventor: Tadashi Amada