Abstract: The memory system has an overwrite operation and an operation control method thereof. A nonvolatile memory device has a plurality of memory blocks including a plurality of memory cells stacked in a direction perpendicular to a substrate. When data of memory cells connected to a word line of a selected memory block is read, the need of reclaim is determined based on an error bit level of the read data. In the case that memory cells having an erase state among the memory cells connected to the word line become a soft program state, the read data is overwritten in the memory cells connected to the word line of the selected memory block.

Abstract: An operation method of a flash memory system includes: obtaining first syndrome values to a codeword; obtaining locations of errors and the number of the locations of errors based on the first syndrome values; error-correcting the codeword by flipping bit values of error bits of the codeword based on the locations of errors to generate an error-corrected codeword; obtaining second syndrome values to the error-corrected codeword; determining whether an error is found in the error-corrected codeword based on the second syndrome values; changing the first syndrome values when it is determined that no error is found in the error-corrected codeword; and restoring the error-corrected codeword to the codeword by re-flipping the flipped bit values when it is determined that an error is found in the error-corrected codeword.

Abstract: A bit interleaving method involves applying a bit permutation process to bits of a QC-LDPC codeword made up of N cyclic blocks each including Q bits, and dividing the codeword after the permutation process into a plurality of constellation words each including M bits, the codeword being divided into F×N?/M folding sections (N? being a subset of N selected cyclic blocks and being a multiple of M/F), each of the constellation words being associated with one of the F×N?/M folding sections, and the bit permutation process being applied such that each of the constellation words includes F bits from each of M/F different cyclic blocks in a given folding section associated with a given constellation word.

Abstract: Amplifier circuitry that is operative to provide selective control of the amplitude of a musical signal as the signal transitions from the pre-amplifier to power amplifier components of an electric guitar amplifier. According to a preferred embodiment, at least one low variable power amplifier is integrated into the signal path from the pre-amplifier to the power amplifier components of the guitar amplifier. The low power amplifier allows for selective control of the amplitude of the musical signal such that the signal ranges from ‘zero volts’ to more than 100 DB. Such low power amplifiers may be selectively deployed in guitar amplifiers utilizing either Class A or Class A/B circuitry and enables a musician to have full control of the power amplifier from 0% gain to 100% gain at any tone selection and any power amp wattage, as may be desired.

Abstract: The invention relates to a system for masking a sound incident on a person. The system comprises a microphone sub-system for capturing the sound. The system further comprises a spectrum-analyzer for determining a power attribute of the sound captured by the multiple microphone sub-system, and a spatial analyzer for determining a directional attribute of the captured sound representative of a direction of incidence on the person. The system further comprises a generator sub-system for generating a masking sound under combined control of the power attribute and the spatial attribute, for masking the incident sound.

Abstract: The invention provides one sound producing system including a speaker, a driving circuit and an audio controller. The speaker is used for producing an audible sound. The driving circuit is connected to the speaker. The audio controller is configured to receive an input audio signal and pre-amplify the input audio signal with an amplify gain to obtain a pre-amplified audio signal; operate a multiband dynamic range control on the pre-amplified audio signal to obtain an output audio signal; convert the output audio signal to a driving voltage; provide the driving voltage to the speaker through the driving circuit; detect over the driving circuit to obtain at least one excursion parameter to determine an estimated excursion of the speaker in response to the driving voltage; and adjust the amplify gain according to the estimated excursion.

Abstract: A sensor system is provided. The sensor system includes a control device and a sensor device. The control device has an audio combo jack and selectively provides an audio signal to the audio combo jack. The sensor device is connected with the audio combo jack. The sensor device includes a transformer, a rectifier, a power supply circuit, and a sensor. The transformer receives the audio signal through the audio combo jack and amplifies the audio signal to generate an amplified audio signal. The rectifier, coupled to the transformer, receives the amplified audio signal and rectifies the amplified audio signal to generate a rectified voltage signal. The power supply circuit is controlled by the rectified voltage signal to provide a supply voltage. The sensor is powered by the supply voltage to perform a sensing operation. An output signal which carries information related the sensing operation is generated.

Abstract: Sound processors and systems including sound processors are disclosed.

Abstract: A low density parity check (LDPC) encoder, an LDPC decoder, and an LDPC encoding method are disclosed. The LDPC encoder includes first memory, second memory, and a processor. The first memory stores an LDPC codeword having a length of 16200 and a code rate of 4/15. The second memory is initialized to 0. The processor generates the LDPC codeword corresponding to information bits by performing accumulation with respect to the second memory using a sequence corresponding to a parity check matrix (PCM).

Abstract: The purpose of the present invention is to avoid ACK/NACK collision in a system in which E-PDCCH control information is transmitted, increase the utilization efficiency of ACK/NACK resources, and suppress unnecessary PUSCH band reduction. A wireless communications terminal having a configuration comprising: a reception unit that receives control signals including ACK/NACK indexes, via an expanded physical downlink control channel; a control unit that determines, on the basis of the ACK/NACK indexes, whether to use a dynamically allocated dynamic ACK/NACK resource or a specified resource specified beforehand, to send downlink data ACK/NACK signals; and a transmission unit that sends the ACK/NACK signals using the dynamic ACK/NACK resource or the specified resource, as determined.

Abstract: An audio processing apparatus and method are provided. The audio processing apparatus includes an envelope detector to detect an envelope of an input signal with respect to a low frequency band, and a signal restorer to restore the input signal including a high frequency band by performing frequency folding of frequency sub-band according the envelope.

Abstract: Methods and apparatuses relating to a hardware memory test unit to check a section of a data storage device for a transient fault before the data is stored in and/or loaded from the section of the data storage device are described. In one embodiment, an integrated circuit includes a hardware processor to operate on data in a section of a data storage device, and a memory test unit to check the section of the data storage device for a transient fault before the data is stored in the section of the data storage device, wherein the transient fault is to cause a machine check exception if accessed by the hardware processor.

Abstract: In at least one embodiment, an apparatus for performing an audio measurement is provided. The apparatus includes a controller that is programmed to generate an audio test signal based on a first frequency range that comprises a first noise spectrum, a second frequency range that comprises a second noise spectrum, and a third frequency range that comprises a third noise spectrum and to transmit the audio test signal to an audio device that generates an audio response signal responsive to the audio test signal. The controller is further programmed to receive the audio response signal and to determine audio related characteristics for the audio device in response to the audio response signal.

Abstract: Apparatuses, systems, methods, and computer program products are disclosed for reduced level cell solid-state storage. A method includes determining that an erase block of a non-volatile storage device is to operate in a reduced level cell (RLC) mode. The non-volatile storage device may be configured to store at least three bits of data per storage cell. A method includes instructing the non-volatile storage device to program first and second pages of the erase block with data. A method includes instructing the non-volatile storage device to program a third page of the erase block with a predefined data pattern. Programming of a predefined data pattern may be configured to adjust which abodes of the erase block are available to represent stored user data values.

Abstract: A system and method are disclosed for measurement and reporting of signal quality in an optical data link in terms of Bit Error Rate (BER). The system and method of optical fiber signal quality measurement is highly accurate in early determination of signal degradation, is computationally simple in implementation, and is immune to instability with variable temperature and other environmental factors. Counting a number of error free Cyclic Redundancy Checks (CRC) of a plurality of video signals, the computationally simple method limits the data analysis required to determine a link OK, a link degraded and a link fault status of the optical data link.

Abstract: An active noise reduction (ANR) circuit includes a digital feed-forward ANR pathway coupled to a feed-forward microphone, to detect environmental sounds in an environment external to a casing, and to a first acoustic driver to output sounds within the casing. The digital feed-forward ANR pathway applies a plurality of filters using a first set of coefficients to convert signals from the feed-forward microphone to feed-forward anti-noise sounds to reduce environmental sounds within the casing. In response to a stimulus, the digital feed-forward ANR pathway applies the plurality of filters using a second set of coefficients, which reduce the degree of feed-forward ANR to enable human speech sounds in the environment external to the casing to be conveyed from the feed-forward microphone to the acoustic driver with less reduction than provided by the first plurality of filters.

Abstract: Technical solutions are described for optimizing a set of test configurations used for testing an electronic circuit that includes latches. An example method includes receiving a test configuration that includes settings that initiate a set of predetermined input values and corresponding expected output values. The method also includes evaluating the test configuration by executing the electronic circuit according to the test configuration and recording parametric data during the execution, where the parametric data is representative of switching activity of the latches in the electronic circuit. The evaluation includes analyzing the parametric data to identify presence of a predetermined pattern in the switching activity and selecting the test configuration based on the predetermined pattern being absent/present in the switching activity.

Abstract: An echo cancellation system that detects and compensates for differences in sample rates between the echo cancellation system and a set of wireless speakers based on a frequency-domain analysis. The system generates Fourier transforms for a microphone signal and a reference signal and determines a series of angles for individual frames. For each tone in the Fourier transforms, the system determines the angles and uses linear regression to determine an individual frequency offset associated with the tone. Using the individual frequency offsets associated with the tones, the system uses linear regression to determine an overall frequency offset between the audio sent to the speakers and the audio received from a microphone. Based on the overall frequency offset, samples of the audio are added or dropped when echo cancellation is performed, compensating for the frequency offset.

Abstract: A transmission device transmitting a polarization-multiplexed optical signal includes: a frame encoder configured to encode an electric signal in accordance with a predetermined frame format; an error correction encoder configured to provide encoded signal data as a result of the encoding by the frame encoder with a predetermined error correction code; and a transmission loss information acquiring part configured to acquire transmission loss information based on a loss that the encoded signal data provided with the error correction code incurs when the encoded signal data is transmitted, from a receiving device as a transmission destination. The error correction encoder adjusts a redundancy of the error correction code given to the encoded signal data, based on the transmission loss information acquired by the transmission loss information acquiring part.

Abstract: A method of determining acoustical characteristics of a room or venue using a microphone unit having four omni-directional microphones placed at ends of a tetrahedro-mounting unit which are equidistant to a middle point of a mounting unit. The four microphones detect impulse responses for each of n sound sources. The detected impulse responses are analyzed: (1) by determining a direct-sound-component direction, delay, and frequency response; (2) by determining an early-reflection direction, delay, and frequency response of each m early reflection; and (3) in view of late-reverberation components by determining a delay and frequency responses. Direct-sound-transmission-function filter parameters are calculated based on the determined direct-sound-component direction, delay, and frequency response M early-reflection-transmission-function filters parameters are calculated based on the m determined directions, delays, and frequency responses of the m early-reflection components.