Abstract: An Electronic Bass Drum includes one or more loudspeakers, an internal amplifier system, a removable electronic drum module and control panel, at least one impact sensitive electronic kick pad, an attachment for mounting a bass drum pedal, mounting hardware for tom toms or other acoustic or electronic instruments, legs for stabilization, one or more headphone jacks for silent play, input jacks for other instruments and/or microphones, and an MP3 player cradle for playing along with or recording music.

Abstract: A handheld device for controlling a vibration of a resonant musical member of a musical instrument includes a handheld housing; at least one transducer on the housing that is configured to sense a proximity and/or vibration of a resonant musical member and output an actuating vibratory electromagnetic force based on the vibration of the resonant musical member; and a haptic component on the housing that is configured to output a tactile feedback responsive to a location and/or vibration of a resonant musical member.

Abstract: An information handling system has a haptic generation module to generate haptic effects including haptic noise and a haptic noise reduction module. The haptic noise reduction module receives characteristics of sound representative of haptic noise generated by a haptic generation module of a device and entering an audio input module of the device, the characteristics including frequencies and timing. It also detects the generation of haptic effects, the generations occurring after the receiving characteristics. It also reduces the effects of haptic noise on digital data representing audio input to the device based upon the received characteristics of the sound. It may reduce the effects by subtracting amplitudes of audio waves representing the haptic noise from amplitudes of audio waves representing the audio input.

Abstract: A time-division-multiplexing (TDM) based noise cancelation headphone is presented herein. A headphone can include an earbud including a speaker, and a TDM based bus that electrically couples the earbud to a portable electronic device. Further, the headphone can include a first micro-electro-mechanical system (MEMS) microphone that is configured to receive a first set of acoustic waves outside of an ear canal, generate first microphone information based on the first set of acoustic waves, and send, utilizing the TDM based bus, the first microphone information directed to the portable electronic device. The speaker is configured to receive, utilizing the TDM based bus, feedforward noise cancelation information associated with the first microphone information from the portable electronic device, and generate, based on the feedforward noise cancelation information, sound within a portion of the ear canal.

Abstract: A wearable computing device can receive, via at least one input transducer, a first audio signal associated with ambient sound from an environment of the device. The device can then process the first audio signal so as to determine a second audio signal that is out of phase with the first audio signal and effective to substantially cancel at least a portion of the first audio signal. The device may then generate a noise-cancelling audio signal based on the second audio signal, based on a third audio signal, and based on one or more wearer-specific parameters, where the third audio signal is representative of a sound to be provided by the device. The device may then cause a bone conduction transducer (BCT) to vibrate so as to provide to an ear a noise-cancelling sound effective to substantially cancel at least a portion of the ambient sound.

Abstract: According to an embodiment, a transliteration work support device includes an analysis unit, a storage unit, an estimation unit, a construction unit, a correction unit, and an update unit. The analysis unit performs language analysis on document data and creates transliteration auxiliary information representing a way of transliteration of a word or a phrase in the document data. The storage unit stores a correction history representing a way of transliteration corrected in the past of the word or the phrase. The estimation unit estimates a correction place and a correction candidate of the document data or the transliteration auxiliary information from the history. The construction unit constructs work list information including work items corresponding to types of corrections according to the correction candidate and progress information. The correction unit corrects the document data or the transliteration auxiliary information.

Abstract: The voice recognition processing apparatus includes a voice acquirer, a first voice recognizer, a storage device, and a recognition result determiner. The voice acquirer acquires a user's voice, and outputs voice information. The first voice recognizer converts the voice information into first information. The storage device previously stores a dictionary in which an exclusion vocabulary is registered. The recognition result determiner compares the first information with the exclusion vocabulary to determine whether the first information includes a word that agrees with a word included in the exclusion vocabulary. The recognition result determiner determines that the first information is information to be rejected, when the first information includes the word that agrees with a word included in the exclusion vocabulary, and determines that the first information is information to be executed, when the first information does not include the word that agrees with a word included in the exclusion vocabulary.

Abstract: Disclosed is the technology for dynamic and intelligent generation of dialog recommendations for the users of chat information systems based on multiple criteria. An example method may include receiving a speech-based user input, recognizing at least a part of the speech-based user input to generate a recognized input, and providing at least one response to the recognized input. The method may further include identifying at least one triggering event, generating at least one dialog recommendation based at least in part on the identification, and presenting the at least one dialog recommendation to a user via a user device.

Abstract: A system and method for an integrated, multi-modal, multi-device natural language voice services environment may be provided. In particular, the environment may include a plurality of voice-enabled devices each having intent determination capabilities for processing multi-modal natural language inputs in addition to knowledge of the intent determination capabilities of other devices in the environment. Further, the environment may be arranged in a centralized manner, a distributed peer-to-peer manner, or various combinations thereof. As such, the various devices may cooperate to determine intent of multi-modal natural language inputs, and commands, queries, or other requests may be routed to one or more of the devices best suited to take action in response thereto.

Abstract: Methods, systems, and software for converting the audio input of a user of a handheld client device or mobile phone into a textual representation by means of a backend server accessed by the device through a communications network. The text is then inserted into or used by an application of the client device to send a text message, instant message, email, or to insert a request into a web-based application or service. In one embodiment, the method includes the steps of initializing or launching the application on the device; recording and transmitting the recorded audio message from the client device to the backend server through a client-server communication protocol; converting the transmitted audio message into the textual representation in the backend server; and sending the converted text message back to the client device or forwarding it on to an alternate destination directly from the server.

Abstract: A speech recognition apparatus and a speech recognition method are provided. In the invention, whether an original voice sampling signal corresponding to a target voice frame is a consonant signal is determined according to at least one of a ratio of an energy of a low-pass sampling signal to an energy of the original voice sampling signal and a ratio value of an energy of a second consonant frequency band signal.

Abstract: A system and method is presented for performing dual mode speech recognition, employing a local recognition module on a mobile device and a remote recognition engine on a server device. The system accepts a spoken query from a user, and both the local recognition module and the remote recognition engine perform speech recognition operations on the query, returning a transcription and confidence score, subject to a latency cutoff time. If both sources successfully transcribe the query, then the system accepts the result having the higher confidence score. If only one source succeeds, then that result is accepted. In either case, if the remote recognition engine does succeed in transcribing the query, then a client vocabulary is updated if the remote system result includes information not present in the client vocabulary.

Abstract: The present disclosure relates to an electronic device and a method for operating same, which can divide a recognition section for voice recognition, process different types of voice recognition corresponding to the divided recognition sections, and improve a voice recognition rate. According to an embodiment, a method for recognizing a voice by an electronic device may include: detecting a voice input; determining the direction of the voice and a beamforming direction. Voice recognition is based on the voice when the direction of the voice and the beamforming direction correspond to each other. Different types of voice recognition may be performed for voices input in a first section and a second section, based on divided types of voice input sections, when the direction of the voice and the beamforming direction do not correspond to each other.

Abstract: Methods, systems, and apparatus, including computer programs encoded on a computer storage medium, for hotword detection on multiple devices are disclosed. In one aspect, a method includes the actions of receiving, by a first computing device, audio data that corresponds to an utterance. The actions further include determining a first value corresponding to a likelihood that the utterance includes a hotword. The actions further include receiving a second value corresponding to a likelihood that the utterance includes the hotword, the second value being determined by a second computing device. The actions further include comparing the first value and the second value. The actions further include based on comparing the first value to the second value, initiating speech recognition processing on the audio data.

Abstract: A voice processing device includes: a processor; and a memory which stores a plurality of instructions, which when executed by the processor, cause the processor to execute: detecting a first utterance section included in a first voice and a second utterance section included in a second voice; specifying an overlapping section within which the first utterance section and the second utterance section overlap with each other; calculating a first utterance continuation section from a start point of the overlapping section to an end point of the first utterance section; and evaluating an impression regarding the first voice at least on the basis of information relating to a length of the first utterance continuation section.

Abstract: Disclosed herein are methods of diarizing audio data using first-pass blind diarization and second-pass blind diarization that generate speaker statistical models, wherein the first pass-blind diarization is on a per-frame basis and the second pass-blind diarization is on a per-word basis, and methods of creating acoustic signatures for a common speaker based only on the statistical models of the speakers in each audio session.

Abstract: Disclosed herein are methods of diarizing audio data using first-pass blind diarization and second-pass blind diarization that generate speaker statistical models, wherein the first pass-blind diarization is on a per-frame basis and the second pass-blind diarization is on a per-word basis, and methods of creating acoustic signatures for a common speaker based only on the statistical models of the speakers in each audio session.

Abstract: There is provided an information processing apparatus including a circuitry configured to initiate a voice recognition upon a determination that a user gaze has been made towards a first region within which a display object is displayed, and initiate an execution of a process based on the voice recognition.

Abstract: A method and an electronic device are provided for voice recognition. A method includes receiving a first voice by one of a first voice recognition device and a second voice recognition device; when the first voice is received by the first voice recognition device and the first voice recognition device recognizes a predetermined command in the first voice, transferring a received second voice to an external electronic device and recognizing the received second voice through the external electronic device; and when the first voice is received by the second voice recognition device and the second voice recognition device recognizes the predetermined command in the first voice, recognizing a second command in the received second voice, and performing an operation based on the recognized second command.

Abstract: A digital wireless audio transmission system having a wireless receiving unit for wirelessly receiving a high-frequency signal. An HF analysis unit analyzes the received high-frequency data, detects errors in the transmission within a time window, and outputs the received data and first items of information in respect of the detected errors. A decoding unit converts/decodes the received high-frequency data into audio data. An error detection unit checks errors in the conversion of the decoding unit within a previously established time window and outputs second items of information in respect of errors during decoding of the received high-frequency data. An error rate is determined based on the first and second items of information. If the error rate exceeds a first threshold value then no audio signal is outputted. Only if the error rate falls below a second value, lower than the first value, is an audio output is then again effected.

Abstract: Encoding and decoding devices for encoding the channels of an audio system having at least four channels are disclosed. The decoding device has a first stereo decoding component which subjects a first pair of input channels to a first stereo decoding, and a second stereo decoding component which subjects a second pair of input channels to a second stereo decoding. The results of the first and second stereo decoding components are crosswise coupled to a third and a fourth stereo decoding component which each performs stereo decoding on one channel resulting from the first stereo decoding component, and one channel resulting from the second stereo decoding component.

Abstract: In watermark symbol detection for watermarked audio signals a correlation and statistical detection is used, which is computationally complex. Therefore a downsampling can be used prior to the correlation. However, if the watermarked audio signals are transmitted over an acoustic path, without downsampling the detection rate is considerably higher than the detection rate when including downsampling of the correlation input signals. There is a trade-off between calculation complexity and detection robustness. According to the invention, an interpolation of the correlation result values is carried out for input to the statistical detector, in order to approximate the detection robustness of correlation without downsampling.

Abstract: A machine may be configured to generate one or more audio fingerprints of one or more segments of audio data. The machine may access audio data to be fingerprinted and divide the audio data into segments. For any given segment, the machine may generate a spectral representation from the segment; generate a vector from the spectral representation; generate an ordered set of permutations of the vector; generate an ordered set of numbers from the permutations of the vector; and generate a fingerprint of the segment of the audio data, which may be considered a sub-fingerprint of the audio data. In addition, the machine or a separate device may be configured to determine a likelihood that candidate audio data matches reference audio data.

Abstract: The present disclosure provides methods, devices and computer program products for non-uniform quantization of parameters relating to parametric spatial coding of audio signals. The disclosure further relates to a method and apparatus for reconstructing an audio object in an audio decoding system taking the non-uniformly quantized parameters into account. According to the disclosure, such an approach renders it possible to reduce bit consumption without substantially reducing the quality of the reconstructed audio object.

Abstract: The present invention relates to a method for processing an audio signal, comprising: a step of performing a frequency conversion process on an audio signal to obtain a plurality of frequency transform coefficients; a step of selecting either a general mode or a non-general mode, on the basis of a pulse ratio, for the frequency transform coefficients having a high frequency band from among the plurality of frequency transform coefficients; and a step of performing, if the non-general mode is selected, the following steps: extracting a predetermined number of pulses from the frequency transform coefficients having the high frequency band, and generating pulse information; generating an original noise signal from the frequency transform coefficients having the high frequency band, excluding the pulses; generating a reference noise signal using the frequency transform coefficient having a low frequency band from among the plurality of frequency transform coefficients; and generating noise position information an

Abstract: The subject disclosure is directed towards dynamically computing anti-aliasing filter coefficients for sample rate conversion in digital audio. In one aspect, for each input-to-output sampling rate ratio (pitch) obtained, anti-aliasing filter coefficients are interpolated based upon the pitch (e.g., using the fractional part of the ratio) from two filters (coefficient sets) selected based upon the pitch (e.g., using the integer part of the ratio). The interpolation provides for fine-grained cutoff frequencies, and by re-computation for each pitch, smooth anti-aliasing with dynamically changing ratios.

Abstract: A signal processing apparatus for effectively detecting an abrupt change in a signal, in particular, an abrupt change different from an ideal pulse includes a converter that converts an input signal into a phase component signal in a frequency domain, a first calculator that calculates a phase gradient for each of a plurality of frequencies of the phase component signal as a first phase gradient, a weighting unit that weights the first phase gradient with an amplitude or a power of the input signal, thereby calculating a weighted first phase gradient, a second calculator that calculates a second phase gradient based on the weighted first phase gradient, and a determiner that determines presence of an abrupt change in the input signal based on the first phase gradient and the second phase gradient.

Abstract: Systems and methods are disclosed for capturing sound for communication by mounting one or more intra-oral microphones to capture sound; and mounting a mouth wearable communicator in the oral cavity to communicate sound with a remote unit.

Abstract: A voice signal processing apparatus and a voice signal processing method are provided. Calculate a value of an interpolation parametric function corresponding to a sampling signal frame according to three consecutive sample values in the sampling signal frame, and calculate an interpolated value between two adjacent sampling points in a frequency-lowered signal frame according to the value of the interpolation parametric function.

Abstract: A voice signal processing apparatus and a voice signal processing method are provided. Each frequency-lowered signal window included in a frequency-lowered sampling voice signal is divided into a first sub signal window that is faded-in and a second sub signal window that is faded-out. The first sub signal window and the second sub signal window that are adjacent to each other and belong to the different frequency-lowered signal windows are overlapped in order to generate an overlapping voice signal. The overlapping voice signal and the sampling voice signal are combined to generate an output signal.

Abstract: A multi-party conversation analyzer and logger uses a variety of techniques including spectrographic voice analysis, absolute loudness measurements, directional microphones, and telephonic directional separation to determine the number of parties who take part in a conversation, and segment the conversation by speaking party. In one aspect, the invention monitors telephone conversations in real time to detect conditions of interest (for instance, calls to non-allowed parties or calls of a prohibited nature from prison inmates). In another aspect, automated prosody measurement algorithms are used in conjunction with speaker segmentation to extract emotional content of the speech of participants within a particular conversation, and speaker interactions and emotions are displayed in graphical form.

Abstract: A speech recognition apparatus and a speech recognition method are provided. In the invention, whether an original voice sampling signal corresponding to a target voice frame is a noise signal is determined according to a ratio of an energy of a first consonant frequency band signal to an energy of a second consonant frequency band signal, a ratio of an energy of the first consonant frequency band signal to an energy of the original voice sampling signal and a ratio of an energy of the second consonant frequency band signal to an energy of the original voice sampling signal.

Abstract: According to an embodiment, a speech processing apparatus includes an enhancer, a converter, a filter, and an inverter. The enhancer is configured to generate a frequency spectrum in which a harmonic component included in input sound is enhanced. The converter is configured to convert the frequency spectrum into a first signal in a modulation frequency domain. The filter is configured to filter the first signal to pass human speech. The inverter is configured to convert the filtered first signal into a second signal in a frequency domain.

Abstract: A described read request may be received. An object and a descriptor may be returned in response to the described read request. The object may be retrieved from a volume. The object may be analyzed to generate the descriptor, which may indicate whether the object is a tape mark or a data record.

Abstract: A bio-disc detection device is provided. A spindle motor is disposed on a chassis for rotating a bio-disc. The bio-disc has a detection groove disposed thereon. The chassis has an optical pick-up head disposed thereon for emitting a light beam to irradiate the detection groove. A receiver receives a light flux passing through the detection groove. A magnetic field generator generates a magnetic field which uniformly covers the detection groove. The chassis has a bracket disposed thereon for fixing the receiver and the magnetic field generator, such that the receiver, the magnetic field generator, the spindle motor and the optical pick-up head are fixed on the chassis of the same foundation member.

Abstract: Methods that include forming at least a portion of a near field transducer (NFT) structure; depositing a material onto at least one surface of the portion of the NFT to form a metal containing layer; and subjecting the metal containing layer to conditions that cause diffusion of at least a portion of the material into the at least one surface of the portion of the NFT; and devices formed thereby.

Abstract: In one embodiment, a magnetic head includes a lower magnetic shield layer positioned at a media-facing surface, a pinned layer positioned above the lower magnetic shield layer at the media-facing surface, at least two MR elements extending in an element height direction by a first length positioned above the pinned layer and separated in a cross-track direction by an inner layer, bias layers extending in the element height direction by a second length positioned on outside edges of the MR elements and the pinned layer, and current paths positioned above and in electrical communication with the bias layers on either side of the inner layer, each current path extending in the element height direction away from the media-facing surface by a third length.

Abstract: A head gimbal assembly for a disk drive includes a flexure tail terminal region having flexure bond pads in electrical communication with the head. Each of the flexure bond pads is aligned with one of a plurality of flexible printed circuit (FPC) bond pads. An anisotropic conductive film is disposed between the FPC and the flexure tail terminal region. The flexure tail terminal region overlaps the anisotropic conductive film in a bonding area. Each of the flexure bond pads is bonded to one of the plurality of FPC bond pads by the anisotropic conductive film in a bonding area. A conductive layer of the flexure tail terminal region includes an exposed conductive ground pad that is disposed outside of the bonding area, and/or disposed outside of an application area of a thermode tool that is pressed against the flexure tail terminal region during the bonding process.

Abstract: A magnetic disk apparatus of one of the embodiments stores read position dependency information on read signal quality of a data region at a first track and measures the read signal quality at a predetermined radial position in a second data region of a second track different from the first track. A positioning error of the second data region is determined based on the read position dependency information and the read signal quality at the predetermined radial position. Data is recorded in a recording target data region in a shingled recording so as to prevent data written in the second data region from being overwritten by data in a recording target data region adjacent to the second data region by using the determined positioning error.

Abstract: According to one embodiment, a magnetic medium includes a substrate, and a magnetic recording layer positioned above the substrate, the magnetic recording layer including an ordered alloy having a L10-type structure, where the ordered alloy comprises a plurality of ferromagnetic crystal grains surrounded by non-magnetic grain boundaries, and where the ordered alloy comprises Fe, Ni and Pt.

Abstract: A hexagonal ferrite magnetic powder for a magnetic recording medium, containing magnetic powder contains hexagonal ferrite particles having coated on the surface thereof an aluminum hydroxide material, having a Ba/Fe molar ratio of 0.080 or more, a Bi/Fe molar ratio of 0.025 or more, and an Al/Fe molar ratio of from 0.030 to 0.200. The magnetic powder preferably has an activation volume Vact of from 1,300 to 2,000 nm3. The magnetic powder particularly preferably has a coercive force Hc of from 159 to 287 kA/m (approximately from 2,000 to 3,600 Oe) and a coercivity distribution SFD of from 0.3 to 1.0. The magnetic powder may contain one or two or more kinds of a divalent transition metal M1 and a tetravalent transition metal M2, as an element that replaces Fe of the hexagonal ferrite. The magnetic powder has improved magnetic characteristics including SNR and durability.

Abstract: A method for manufacturing a glass substrate according to which surface roughnesses of main surfaces of a glass substrate can be reduced more than with currently available methods is provided. After the main surfaces of the glass substrate used in a magnetic disk are mirror-polished (final finishing-polished) using a polishing liquid containing organic-based particles made of, for example, a styrene-based resin an acrylic resin, or a urethane-based resin as polishing abrasive particles, by cleaning the glass substrate using an organic-based cleaning agent, surface roughnesses of the main surfaces of the substrate can be reduced more than with currently available methods.

Abstract: According to an optical disc, a track, where data is recorded in predetermined block units, is formed by wobbling at a spatial period in accordance with a radial position, and the track is divided into zones in a radial direction. Data is recorded in each of the zones at a predetermined line density based on a ratio n/m of the number of bits of a bit pattern of recording data to one wavelength of wobbling. The optical disc includes a first area having one of the zones, a second area having one of the zones different from the one of the zones in the first area, and a line density of data different from a line density of data in the first area, and a BCA where a zone format identifier specifying a zone table defining a boundary position of each of the zones and the ratio n/m is recorded.

Abstract: According to one embodiment, a system for processing data includes a processor and logic integrated with and/or executable by the processor. The logic is configured to individually set, for each of one or more range-constrained finite impulse response (FIR) filter taps configured for use in a FIR filter, a predetermined range of values suitable for controlling an equalizer response. The logic is also configured to pass data through the equalizer comprising the FIR filter to obtain equalized data. Each of the one or more range-constrained FIR filter taps are individually adaptive within its predetermined range of values. Also, the data is read from a magnetic storage medium.

Abstract: An apparatus and associated method contemplating a head gimbal assembly (HGA) and a bridge circuit including first and second microactuators attached to the HGA. Computation logic is connected to the bridge circuit and configured to independently measure an electrical output of each microactuator and to sum the electrical outputs to derive a value related to a vertically dominant modulation mode of the HGA.

Abstract: Provided is a video management apparatus and a method for event recording using the same, which are capable of reducing loss of video data in an event recording. The video management apparatus performs an event recording in cooperation with a network camera to transmit a first frame with basic information of video data and a plurality of second frames with changed information of the video data. The video management apparatus includes a buffer unit to store the first and the plurality of second frames, and eliminate the stored frames according to an external control signal. The video management apparatus further includes an event recording unit to save the first frame and at least one of the second frames stored in the buffer unit, and when detecting an occurrence of events, and frames transmitted from the network camera after the detection of the occurrence of events on a storage medium.

Abstract: Methods and apparatus associated with data cold storage are described. Example apparatus include an array of data storage devices arranged in rows and columns. Columns of the array are orthogonal to rows. A row has an associated row-centric power supply, and a column has an associated column-centric local electronics module (LEM) that controls a data storage device in the column independently of other data storage devices in the array. Example apparatus include logics that control a power mode of a data storage device independently of other data storage devices in the array, that control a power mode of an LEM, that adaptively regulate the level of data stored in a buffer, and that determine whether a data object will be stored in the buffer or stored on a data storage device in the array, based on the probability the data object will be accessed within a threshold period of time.

Abstract: An accessor includes a main body unit configured to convey a recording medium in a first direction, the main body unit being attached to a housing of a library device to be movable in the first direction, a picker configured to convey the recording medium in a second direction intersecting with the first direction, the picker being mounted on the main body unit to be movable in the second direction, and a cleaner mounted on a surface of the picker along the second direction.

Abstract: The present invention relates to data application technologies in the field of video surveillance, and discloses an intelligent playback method for video records based on a motion information and an apparatus thereof. The method comprises the following steps: searching for all motion detection additional information conforming to a time period of a playback request in a hard disc, and storing the motion detection additional information as a motion information in a motion information linked list in terms of a time sequence; obtaining a current playback mode and a current playback time, and looking up a nearest motion information under the current playback mode and the current playback time from the motion information linked list; matching the found nearest motion information with the current playback time; controlling a playback speed according to a matching result, so as to implement an automatic control of an intelligent playback speed.

Abstract: Providing a method for creating and displaying portions of videos called video previews. The video previews may be associated with a predefined duration of a video, such that the video preview is generated from a portion of the video. The video previews are configured to play a series of images associated with images from the portion of the video when the video preview is activated.