Abstract: Provided is a charging scan and charge sharing scan double output GOA circuit to combine the time sequence and circuit. The nth stage GOA unit circuit receives the first, the second low frequency clock signals (LC1, LC2), the direct current low voltage signal (Vss), the Mth, M?2th high frequency clock signals (CK(M), CK(M?2)), a stage transfer signal (ST(n?2)) generated by the n?2th stage GOA unit circuit, a charging scan signal (CG(n?2)) generated by the n?2th stage GOA unit circuit and a stage transfer signal (ST(n+2)) generated by the n+2th stage GOA unit circuit, the charging scan signal (CG(n)), a charge sharing scan signal (SG(n?2)) generated by the n?2th stage GOA unit circuit and the stage transfer signal (ST(n)) are respectively outputted with different TFTs; the nth stage GOA unit circuit comprises a transmission module (100), a transfer regulation module (200), an output module (300), a rapid pull-down module (400) and a pull-down holding module (500).

Abstract: Provided is a charging scan and charge sharing scan double output GOA circuit to combine the time sequence and circuit. The nth stage GOA unit circuit receives the first, the second low frequency clock signals (LC1, LC2), the direct current low voltage signal (Vss), the Mth, M?2th high frequency clock signals (CK(M), CK(M?2)), a stage transfer signal (ST(n?2)) generated by the n?2th stage GOA unit circuit, a charging scan signal (CG(n?2)) generated by the n?2th stage GOA unit circuit and a stage transfer signal (ST(n+2)) generated by the n+2th stage GOA unit circuit, the charging scan signal (CG(n)), a charge sharing scan signal (SG(n?2)) generated by the n?2th stage GOA unit circuit and the stage transfer signal (ST(n)) are respectively outputted with different TFTs; the nth stage GOA unit circuit comprises a transmission module (100), a transfer regulation module (200), an output module (300), a rapid pull-down module (400) and a pull-down holding module (500).

Abstract: A liquid crystal display includes a display unit having a plurality of pixels in a plurality of first rows and second rows that are alternately arranged. A gate driver supplies a same scan signal to a plurality of pixels of a first row and a second row that are adjacent to each other among the plurality of first rows of pixels and the plurality of second rows of pixels at a first frame period and a second frame period that are continuous. A plurality of scan signals are respectively supplied to the plurality of second rows of pixels at the second frame period. A data driver generates a plurality of data voltages respectively corresponding to the plurality of first rows of pixels at the first frame period and generating a plurality of data voltages respectively corresponding to the plurality of second rows of pixels at the second frame period.

Abstract: According to an aspect, a display apparatus includes: a display device that displays an image; and a drive circuit that drives the display device. The display device includes a regulator that supplies power based on a predetermined set voltage, a plurality of functional blocks that are operated by the supply of the power from the regulator and are related to operation of the display device, a voltage monitoring circuit that determines whether a level of a power supply voltage of at least one of the functional blocks is high or low based on a predetermined threshold voltage, and a voltage controller that increases the set voltage when the voltage monitoring circuit determines that the power supply voltage of the functional block is lower than the threshold voltage.

Abstract: A multiplexer is provided herein. The multiplexer has a plurality of first driving units and a plurality of second driving units. Each of the first driving units has a first data voltage input terminal, and each of the second driving units has a second data voltage input terminal. The first data voltage input terminal and the second data voltage input terminal are configured to receive pixel voltage signals with different polarities. In the first driving unit, a voltage difference between a gate and a drain of a transistor is controlled by a first reset signal, wherein the transistor of the first driving unit is coupled to the first data voltage input terminal and a first data line. In the second driving unit, a voltage difference between a gate and a drain of a transistor is controlled by a second reset signal, wherein the transistor of the second driving unit is coupled to the second data voltage input terminal and a second data line.

Abstract: In an embodiment, the system includes a direct input device (the “client”) executing a client application (“app”), a secondary device (the “server”) executing a server app, and a network link between the server and the client. The server app transmits to the client app, via the network link, a portion of the content displayed on the server's computer screen. The client app may display this portion. Whenever the user interacts with the client device, the client app transmits those interactions to the server app over the network link. The server app sends an acknowledgement to the client app and provides the received interactions to the spatial input software executing on the server, and the spatial input software consumes the interactions as if the user performed the interactions on the server device itself.

Abstract: Aspects of the present disclosure relate to a method for improving color rendition in images displayed to a user of a head-worn computer as digital images with colored subpixels, the method including determining the colors associated with each color filter in the color filter array, determining the subpixels positioned under each color filter in the color filter array along the chief ray angles sampled by the lens, shifting the digital code values between subpixels in the digital image to provide a modified digital image, wherein digital code values within the modified image are positioned at the determined subpixels positioned under each color filter along the chief ray angles sampled by the lens in correspondence to the colored subpixels in the digital image, and displaying the modified digital image in the head-worn computer.

Abstract: The present application discloses a method for compensating mura defects in a display image, the method includes obtaining a plurality of display data and a plurality of address data corresponding to a plurality of display pixels; and determining if a pixel is a sampling pixel based on a data table comprising a plurality of compensation data associated with a plurality of compensation points for compensating a plurality of sampling pixels, each compensation point corresponding to a group of at least one sampling pixel, the plurality of sampling pixels comprising a plurality of display pixels having mura defects and constituting a portion of the plurality of display pixels.

Abstract: An electronic apparatus and a method of outputting content of the electronic apparatus outputs a synchronized content to a display apparatus and a projection-type display apparatus and a method of outputting a content of the electronic apparatus. The electronic apparatus includes an input/output unit connected with a display apparatus and a projection-type display apparatus, and a control unit configured to control the input/output unit, wherein the control unit outputs a synchronized content to the display apparatus and the projection-type display apparatus through the input/output unit.

Abstract: Provided is an image displaying apparatus including: a receiver configured to receive a first image group including an initial image and a final image; a display configured to display, on an image display window, one of a plurality of images included in the first image group; and a controller configured to control the display to display, on the image display window, the initial image, and control the display to change an image displayed on the image display window from the initial image to the final image, based on at least one of a degree to which the image display window is scrolled on the display and an elapsed time from when the initial image is displayed on the image display window.

Abstract: An information processing apparatus that has a display unit and a plurality of graphic controllers, the apparatus comprises: a detection unit that detects an event that is generated by a user operation on the display unit; a determination unit that determines, when the event is detected by the detection unit, which of the plurality of graphic controllers an application displaying a screen on the display unit uses; and a transmission unit that, based on a resolution of the display unit and information of the application for which the determination is made by the determination unit, converts coordinate information of the user operation that the event indicates to a value of a coordinate system corresponding to the graphic controller that the application uses, and transmits a result of the conversion to the application.

Abstract: An driving method for a display panel, the display panel having a first resolution. The driving method comprises: converting RGB signals of an input image having a second resolution into YUV signals corresponding to an output image, a resolution of the output image being not smaller than the second resolution; converting the YUV signals into RGB signals corresponding to the output image; converting the RGB signals corresponding to the output image into driving signals for driving the display panel; and outputting the driving signals to the display panel. Display panels having the resolution of 10240×4320 can be driven by using the driving method.

Abstract: One example includes a mission monitoring system. The system includes a data store configured to store map data associated with a geographic region of interest and a plurality of graphical elements corresponding to at least one mission asset and to mission constraints. Each of the plurality of graphical elements includes a layer assignment associated with one of a plurality of graphical layers. The system also includes a human-machine interface (HMI) configured to facilitate user inputs to selectively activate the plurality of graphical layers associated with the plurality of graphical elements and to display a graphical rendering of the selectively activated graphical layers superimposed over a graphical virtual environment of the geographic region of interest via a display system.

Abstract: The present inventive disclosures are generally directed to various improved versions of locking compensated nuts for fretted stringed musical instruments. Many embodiments are especially directed to improved locking compensated nuts that can be more-easily retrofitted into existing guitars and other stringed musical instruments without requiring any major reconfigurations or modifications of the base instruments, and some variations are directed to a detachably coupled tuner-mount device that will facilitate enhanced tuning capabilities because of improved vibrational communication between the instrument strings and the tuner. In addition, some improved locking compensated nuts described herein feature interchangeable compensated inserts to allow for easy adaption to alternate string sets.

Abstract: Disclosed is a new-type native American flute with a new finger hole design. One end of a flute head is provided with a mouthpiece, and the other end of the flute head is connected with a flute body. The front of the flute head is provided with a flute nose and a sound outlet. The flute body is provided with 7-10 finger holes. The finger holes may include main finger holes and an auxiliary finger hole, and the auxiliary finger hole is on the side of the main finger holes. The main finger holes include front main finger holes and a back main finger hole. Arranging an auxiliary finger hole on the surface of a native American flute in the present disclosure may broaden the range, increase altered tones as well, and further realize fine tuning for the altered tones.

Abstract: A keyed wind musical instrument (100, 101, 102) including an elongate tubular body (200) having a plurality of tone holes (201) extending through a surface of the elongate tubular body (200); at least one key member (202) attached to the elongate tubular body (200) adjacent at least one of said plurality of tone holes (201), the at least one key member (202) including: a closure portion (202B) movable between at least one of a closed configuration in which the closure portion (202B) substantially seals the tone hole (201), and, an opened configuration in which the closure portion (202B) is spaced apart from the tone hole (201) wherein the tone hole (201) is opened; and an attachment portion (202A) integrally formed with the closure portion (202B), the attachment portion (202A) being configured for attaching the key member (202) to the elongate tubular body (200); and wherein the integrally formed attachment portion (202A) and closure portion (202B) are configured to bias the closure portion (202B) towards the

Abstract: The present disclosure presents a novel approach to automatic transcription of piano music in a context-dependent setting. Embodiments described herein may employ an efficient algorithm for convolutional sparse coding to approximate a music waveform as a summation of piano note waveforms convolved with associated temporal activations. The piano note waveforms may be pre-recorded for a particular piano that is to be transcribed and may optionally be pre-recorded in the specific environment where the piano performance is to be performed. During transcription, the note waveforms may be fixed and associated temporal activations may be estimated and post-processed to obtain the pitch and onset transcription.

Abstract: An adjustable guitar effects pedalboard for anchoring guitar effects pedals. More specifically, a pedalboard having at least two telescoping sections, wherein the pedalboard is configured to allow users to adjust the width of the pedalboard to fit their needs. The pedalboard can have an upper level and a lower, telescoping level, and the upper level and the lower, telescoping level may be curved and further contain a plurality of holes on which users can use cable ties to anchor their guitar effects pedals. The plurality of holes in each of the at least two telescoping sections may align with each other in various positions of expansion and contraction.

Abstract: A sound absorbing mat comprising a first polyurethane open cell polyester layer, wherein the first polyurethane open cell polyester layer is at least one quarter of an inch thick, a polyvinyl chloride layer having a top surface and a bottom surface, wherein the polyvinyl chloride layer is at least a quarter inch thick and the bottom surface is fused to the first polyurethane open cell polyester, a second polyurethane open cell polyester layer having a top surface and a bottom surface, wherein the second polyurethane open cell polyester layer is at least three quarters of an inch and the bottom surface of the second polyurethane open cell polyester layer is fused with the top surface of the polyvinyl chloride layer, and a urethane layer, wherein the urethane layer is at least a quarter inch thick and is mated to the top surface of the second polyurethane open cell polyester layer.

Abstract: A noise-reducing fan system, comprising a motor, a fan body, a plurality of magnetic-inducing elements, a magnetic field generator and a noise-reducing sound source device, is provided. Here, the fan body is mounted on the motor. The fan body comprises a plurality of blades, on which the plurality of magnetic-inducing elements are disposed, respectively. The magnetic field generator, which may generate a magnetic field, is employed for driving the plurality of magnetic-inducing elements to vibrate the plurality of blades and generate a vibration sound, so that at least one portion of the noise emitted from the fan body as rotating may be counterbalanced. The noise-reducing sound source device is disposed on a predetermined position and may send out a noise-reducing sound, so that the noise-reducing sound may counterbalance at least the other portion of the noise emitted from the fan body as rotating.

Abstract: A method and an apparatus for generating a g2p model based on AI are provided. The method includes: during performing a grapheme-to-phoneme conversion training by a neural network on each word in training data, screening nodes in a hidden layer of the neural network randomly according to a preset node ratio so as to obtain retaining nodes for training each word; training each word with a sub-neural network corresponding to the retaining nodes and updating a weight of each retaining node of the sub-neural network; and performing a mean processing on the weights of the retaining nodes of respective sub-neural networks, so as to generate the grapheme-to-phoneme model.

Abstract: A voice input comprising a command word, one or more media variable instances, and one or more zone variable instances is received. A media playback system command which corresponds to the command word is determined. Media content which corresponds to the one or more media variable instances is identified. The media playback system is caused to execute the media playback system command on the media content based on the one or more zone variable instances.

Abstract: The present disclosure relates generally to improving audio processing using an intelligent microphone and, more particularly, to techniques for processing audio received at a microphone with integrated analog-to-digital conversion, digital signal processing, acoustic source separation, and for further processing by a speech recognition system. Embodiments of the present disclosure include intelligent microphone systems designed to collect and process high-quality audio input efficiently. Systems and method for audio processing using an intelligent microphone include an integrated package with one or more microphones, analog-to-digital converters (ADCs), digital signal processors (DSPs), source separation modules, memory, and automatic speech recognition. Systems and methods are also provided for audio processing using an intelligent microphone that includes a microphone array and uses a preprogrammed audio beamformer calibrated to the included microphone array.

Abstract: An electronic device and a voice processing method of the electronic device are provided. The electronic device includes a microphone array including a plurality of microphones facing specified directions; a sensor module configured to sense a user located near the electronic device; and a processor configured to select one of a plurality of users sensed near the electronic device, process a voice received from a direction in which the selected user is located, as a user input, and process a voice received from another direction, as noise.

Abstract: An electronic apparatus and a service providing method thereof are provided. The service providing method includes receiving a sound generated from outside through at least one microphone, extracting a noise feature from the received sound, and providing a service corresponding to the extracted noise feature by comparing the extracted noise feature and a pre-stored noise feature.

Abstract: A speech recognition system, which continuously recognizes speech uttered by at least one user and controls a navigation system on the basis of a recognition result, includes: a speech-operation-intention determination unit for determining—whether or not the user has made a recognized speech with the intention of operating the navigation system through speech; and a control mode altering unit for changing, when the speech-operation-intention determination unit determines that the user has no operation intention, the control mode of the navigation system in such a manner that the user is less aware of or pays less attention to the control mode than the case in which the speech-operation-intention determination unit determines that the user has an intention of operating the navigation system.

Abstract: Techniques for leveraging a combination of audio-based and vision-based cues for voice command-and-control are provided. In one embodiment, an electronic device can identify one or more audio-based cues in a received audio signal that pertain to a possible utterance of a predefined trigger phrase, and identify one or more vision-based cues in a received video signal that pertain to a possible utterance of the predefined trigger phrase. The electronic device can further determine a degree of synchronization or correspondence between the one or more audio-based cues and the one or more vision-based cues. The electronic device can then conclude, based on the one or more audio-based cues, the one or more vision-based cues, and the degree of synchronization or correspondence, whether the predefined trigger phrase was actually spoken.

Abstract: The assistive hearing device implementations described herein assist hearing impaired users of the device by using automated speech transcription to generate text representing speech received in audio signals which can then be read in a synthesized voice tailored to overcome a user's hearing deficiencies. A speech recognition engine recognizes speech in received audio and converts the speech of the received audio to text. Once the speech is converted to text, a text-to-speech engine can convert the text to synthesized speech that can be enhanced and output in a voice that compensates for the hearing loss profiles of a user of the assistive hearing device. By transcribing received speech into text the assistive hearing device implementations described herein eliminate background noise from the audio signal. By converting the transcribed text into a synthesized voice that is easier to understand to hearing impaired persons, their hearing deficiencies can be remedied.

Abstract: A system and method for speech-to-text translation. The method includes determining, based on at least one audio input in a first language, at least one original language concept; identifying, based on the determined at least one original language concept, the first language of the at least one audio input; determining, for each original language concept, a matching translated concept, wherein each matching translated concept is associated with a second language, wherein the second language is different from the first language; generating a textual output based on the determined at least one translated concept.

Abstract: Methods employ sensors in portable devices (e.g., smartphones) both to sense content information (e.g., audio and imagery) and context information. Device processing is desirably dependent on both. For example, some embodiments activate certain processor intensive operations (e.g., content recognition) based on classification of sensed content and context. The context can control the location where information produced from such operations is stored, or control an alert signal indicating, e.g., that sensed speech is being transcribed. Some arrangements post sensor data collected by one device to a cloud repository, for access and processing by other devices. Multiple devices can collaborate in collecting and processing data, to exploit advantages each may have (e.g., in location, processing ability, social network resources, etc.). A great many other features and arrangements are also detailed.

Abstract: A particular apparatus includes a first buffer that is configured to store multiple audio data samples and a second buffer that is configured to store the multiple audio data samples. The first buffer is coupled to a first processor that is configured to analyze audio data samples to detect a keyword. The second buffer is coupled to a second processor that is configured to initialize a speech recognition engine (SRE) based on the multiple audio data samples. The first buffer has less storage capacity that the second buffer.

Abstract: Provided are methods, systems, and apparatuses for detecting, processing, and responding to audio signals, including speech signals, within a designated area or space. A platform for multiple media devices connected via a network is configured to process speech, such as voice commands, detected at the media devices, and respond to the detected speech by causing the media devices to simultaneously perform one or more requested actions. The platform is capable of scoring the quality of a speech request, handling speech requests from multiple end points of the platform using a centralized processing approach, a de-centralized processing approach, or a combination thereof, and also manipulating partial processing of speech requests from multiple end points into a coherent whole when necessary.

Abstract: Systems and methods disclosed herein include (i) receiving a voice command via a networked microphone device configured to receive voice commands for a media playback system, wherein the media playback system comprises a primary media playback device configured to perform a function and/or provide a first feedback in response to voice commands received via the networked microphone device, (ii) determining that the primary media playback device is not available to perform a function and/or provide the first feedback in response to the voice command, and (iii) in response to determining that the primary media playback device is not available to perform a function and/or provide the first feedback in response to the voice command, instructing a fallback device to perform a function and/or provide a second feedback in response to the voice command.

Abstract: A speech recognition method, an electronic device and a speech recognition system are provided. When a local device is not connected to Internet is determined, a voiceprint comparison between the received voice data and the history voice data stored in the voice database is executed to obtain the corresponding history voice data, and an associated history text data is found from a result database of the local device according to the obtained history voice data.

Abstract: The invention is suited for improving a low bit rate compressed and decompressed Higher Order Ambisonics HOA signal representation of a sound field, wherein the decompression provides a spatially sparse decoded HOA representation and a set of indices of coefficient sequences of this representation. From reconstructed signals of the original HOA representation a number of modified phase spectra signals are created using de-correlation filters, which modified phase spectra signals are uncorrelated with the signals of said original representation. The modified phase spectra signals are mixed with each other using predetermined mixing parameters, in order to provide a replicated ambient HOA component. Finally the spatially sparse decoded HOA representation is enhanced with the replicated time domain HOA representation.

Abstract: A method (600) for generating a bitstream (500) indicative of an object based audio program is described. The bitstream (500) comprises a sequence of containers (501). A first container (501) of the sequence of containers (501) comprises a plurality of substream entities (520) for a plurality of substreams (411, 412, 413, 414) of the object based audio program and a presenta tion section (510). The method (600) comprises determining (601) a set of object channels (424). The method (600) further comprises providing (602) a set of object related metadata (434, 444) for the set of object channels (424). In addition, the method (600) comprises inserting (603) a first set of object channel frames and a first set of object related metadata frames into a respective set of substream entities (520) of the first container (501). Furthermore, the method (600) comprises inserting (604) presentation data into the presentation section (510).

Abstract: A microphone receives speech data to be transmitted. A business wireless communication unit transmits the received speech data. A processing unit causes a storage unit to store the speech data and time information for identifying blocks produced by dividing the received speech data into a plurality of blocks. The business wireless communication unit receives a request signal for requesting transmission of the speech data stored in the storage unit, the request signal including the time information. The processing unit acquires from the storage unit the speech data for the block identified based on the time information included in the received request signal, and causes the business wireless communication unit to transmit the acquired speech data.

Abstract: For an efficient encoding of subband configuration data the first, penultimate and last subband groups are treated differently than the other subband groups. Further, subband group bandwidth difference values are used in the encoding. The number of subband groups NSB is coded using a fixed number of bits representing NSB?1. The bandwidth value BSB[1] of the first subband group is coded using a unary code representing BSB[1]?1. No bandwidth value BSB[g] is coded for the last subband g=NSB. For subband groups g=2, . . . , NSB?2 bandwidth difference values ?BSB [g]=BSB [g]?BSB[g?1] are coded using a unary code, and the bandwidth difference value ?BSB[NSB?1] for subband group g=NSB?1 is coded using a fixed number of bits.

Abstract: The present invention proposes a new method for improving the performance of a real-valued filterbank based spectral envelope adjuster. By adaptively locking the gain values for adjacent channels dependent on the sign of the channels, as defined in the application, reduced aliasing is achieved. Furthermore, the grouping of the channels during gain-calculation, gives an improved energy estimate of the real valued subband signals in the filterbank.

Abstract: A speech/audio coding apparatus is provided that includes a receiver that receives a time-domain speech input signal and a processor. The processor transforms a time-domain speech input signal into a frequency-domain spectrum, and divides a frequency region of the spectrum in an extended band into a plurality of bands. The processor also sets a limited band for each divided band in the current frame, when a difference between a first frequency with a first maximum amplitude in a spectrum of the divided band in a preceding frame and a second frequency with a second maximum amplitude in a spectrum of the divided band in a current frame is below a threshold. The processor further encodes the spectrum in the limited band within each divided band in the current frame, and does not encode a spectrum outside the limited band within each divided band in the current frame.

Abstract: An audio signal (X) is represented by a bitstream (B) segmented into frames. An audio processing system (500) comprises a buffer (510) and a decoding section (520). The buffer joins sets of audio data (D1; D2, . . . , DN) carried by N respective frames (F1, F2, . . . , FN) into one decodable set of audio data (D) corresponding to a first frame rate and to a first number of samples of the audio signal per frame. The frames have a second frame rate corresponding to a second number of samples of the audio signal per frame. The first number of samples is N times the second number of samples. The decoding section decodes the decodable set of audio data into a segment of the audio signal by at least employing signal synthesis, based on the decodable set of audio data, with a stride corresponding to the first number of samples of the audio signal.

Abstract: Embodiments are directed processing adaptive audio content by determining an audio type as one of channel-based audio and object-based audio for each audio segment of an adaptive audio bitstream, tagging the each audio segment with a metadata definition indicating the audio type of the corresponding audio segment, processing audio segments tagged as channel-based audio in a channel audio renderer component, and processing audio segments tagged as object-based audio in an object audio renderer component that is distinct from the channel audio renderer component. Object-based audio is rendered through an object audio renderer interface that dynamically adjusts processing block sizes of the object audio segments based on timing and alignment of metadata updates and maximum/minimum block size parameters.

Abstract: A sound processing system, method and program product for estimating parameters from binaural audio data.

Abstract: A gain control system for controlling gain applied to an audio signal includes a power estimator configured to estimate the power of a digital signal derived from the audio signal, a digital gain estimator configured to determine, in dependence on the estimated power, a digital gain which would modify the power of the digital signal so as to reach a target power level, and a gain controller configured to adjust an analogue gain applied to the audio signal in dependence on the determined digital gain.

Abstract: The present invention relates to a method for a user to adjust the contents of the closed-captioning based on a user's language skill and other needs. The method comprises the steps of: receiving closed captioning contents; determining sub-difficulty levels of the closed captioning contents by comparing the closed captioning contents with at least one classification database; grouping sub-difficulty levels based on preset ranges into difficulty levels; and displaying and adjusting closed captioning contents at difficulty levels chosen by the user.

Abstract: A wearable device is provided that includes a microphone, a display, and controller. The controller analyzes audio information picked up by the microphone, and, when audio corresponding to a predetermined verbal address phrase has been detected in the acquired audio information, causes the display to display an indication of an utterance of a verbal address on the display.

Abstract: The invention relates to monitoring behavior of listening to broadcast, and it is presented a device and a method for monitoring behavior of listening to broadcast. The device comprises a transmitter transmitting a characteristic audio signal which is modulated on a carrier signal; a collector collecting an ambient sound; a detector detecting whether the collected ambient sound contains the characteristic audio signal; and a determiner determining whether a broadcast receiving apparatus receives the broadcasting channel corresponding to frequency of the carrier signal according to the result of the detector.

Abstract: An electronic device includes a first microphone that receives a sound generated for a specific time period, from the outside, a second microphone, which is disposed at a location spaced apart from the first microphone and which receives the sound, an audio converter comprising audio converting circuitry, and a processor electrically connected with the first microphone, the second microphone, and the audio converter. The processor is configured to convert the sound obtained from the first microphone, into a first signal and to convert the sound obtained from the second microphone, into a second signal, using the audio converter, and to determine the sound, which is generated for the specific time period, as a voice or a noise based on a frequency-related correlation between the first signal and the second signal.

Abstract: A method comprises creating calibration data using a first control circuitry of an apparatus, replacing the first control circuitry with a second control circuitry in the apparatus, and operating the apparatus with the second control circuitry using the calibration data. As an example, the apparatus may be a disc drive. The second control circuitry may be substantially similar to the first control circuitry such that calibration measurements using the first control circuitry are applicable to the second control circuitry. The first control circuitry may be included in a circuit board that is replaced with a second circuit board including the second control circuitry. In an exemplary embodiment, the second circuit board may include different and/or additional components relative to the first circuit board, such as integrated video inputs and/or video control circuitry.

Abstract: According to one embodiment, a magnetic recording and reproducing device includes a magnetic recording medium, a magnetic head, and a processor. The magnetic head includes a first reproducing element portion and a second reproducing element portion. The processor is configured to acquire a first signal and a second signal, and to output an output signal according to either one of the first signal and the second signal. The first signal is obtained by reproducing information recorded on a first recording region by the first reproducing element portion. The second signal is obtained by reproducing the information recorded on the first recording region by the second reproducing element portion.