Abstract: A driver and an electronic device include a capacitor driving circuit and a capacitor circuit having a plurality of capacitors provided between a plurality of capacitor driving nodes and a data voltage output terminal. The capacitor driving circuit has a plurality of driving units that output capacitor driving voltages, and in the case where a capacitance of one of the plurality of capacitors is the highest, after a driving unit that drives that capacitor has outputted a capacitor driving voltage, the next driving unit outputs the next capacitor driving voltage.

Abstract: Aspects of the present disclosure include a method and system for the delivery of virtual reality content that enables a more immersive experience to the user by adaptively changing display parameters of virtual reality content using resolution-defined projections. The resolution-defined projections are two-dimensional and virtual reality content agnostic. According to some aspects, the projections can be used to generate virtual reality content images and video that can be encoded and/or compressed for processing and transmission with lower or without increasing bandwidth requirements, and to be perceived of a higher resolution. Additional aspects include, the adaptive focus via the resolution-defined projections using the user's viewing direction and/or orientation, optimization data from a feedback loop and/or crowdsourcing data.

Abstract: A clock and data recovery circuit in accordance with an embodiment of the inventive concept includes a phase locked loop configured to receive a data stream into which an additional bit is inserted at every reference period to generate parallelized data and a clock signal, and a first detector circuit configured to determine whether the parallelized data is locked based on a bit-conversion of the data stream according to an insertion of the additional bit. The bit-conversion is executed with respect to the additional bits according to a predetermined protocol, or is executed with respect to at least one bit from among data of the data stream between a current one of the additional bits and a next one of the additional bits.

Abstract: An image output apparatus includes a light emitting unit, a power supplying unit, and a control unit. The light emitting unit is configured to emit a laser beam. The power supplying unit is configured to supply power to the light emitting unit at a predetermined voltage. The control unit is configured to control the power supplying unit to supply the power to the light emitting unit in a predetermined period at the voltage associated with a peak light emission amount of the light emitting unit in image data for the predetermined period.

Abstract: A handheld imaging device has a data receiver that is configured to receive reference encoded image data. The data includes reference code values, which are encoded by an external coding system. The reference code values represent reference gray levels, which are being selected using a reference grayscale display function that is based on perceptual non-linearity of human vision adapted at different light levels to spatial frequencies. The imaging device also has a data converter that is configured to access a code mapping between the reference code values and device-specific code values of the imaging device. The device-specific code values are configured to produce gray levels that are specific to the imaging device. Based on the code mapping, the data converter is configured to transcode the reference encoded image data into device-specific image data, which is encoded with the device-specific code values.

Abstract: An instrument cluster assembly for a vehicle includes a vehicle instrument that is viewable by an occupant of the vehicle. A transparent protective cover covers the vehicle instrument such that the vehicle instrument is viewable by the occupant through the transparent protective cover. A transparent display overlay system includes an overlay display comprising a transparent display layer connected to the transparent protective cover by an optical adhesive layer. A display controller displays an image on the transparent display layer based on vehicle information. The vehicle instrument is viewable by the occupant through the transparent protective cover and the transparent display layer adjacent the image.

Abstract: Apparatuses, methods and storage media associated with display of visual assets on a device are described. Specifically, the device may include a display screen. The device may further include a visual asset scheduler. The visual asset scheduler may include a screen predictor, a queue, and a visual asset loader. Other embodiments may be described and/or claimed.

Abstract: An apparatus, system, or method wirelessly sends an audio component of a multimedia channel to one or more wireless computing devices while a visual component of the multimedia channel is playing on a video monitor. The audio component may be packetized into data packets and broadcast via a network such that the data packets can be received and played by the wireless computing devices in synchronization with the visual component playing on the video monitor. A channel listing may inform the wireless computing devices what channels are available. A visual content that is not the visual component of the multimedia channel may be displayed on the wireless computing device while an audio component of the multimedia channel is streamed to the wireless computing device.

Abstract: The disclosed subject matter generally relates to image presentation control methods and image presentation control apparatuses. The disclosed subject matter can comprise acquiring at least one piece of background information corresponding to the field of view of at least one eye of a user; adjusting, according to the at least one piece of background information, at least one presentation parameter corresponding to at least one image; and presenting, by means of near-to-eye presentation, the at least one image to the at least one eye according to the at least one presentation parameter that adjusted. In an embodiment of the disclosed subject matter, a presentation parameter corresponding to a to-be-presented image can be adjusted with reference to a content attribute of the image by analyzing a background environment feature of near-to-eye presentation.

Abstract: In an aspect of the disclosure, a method, a computer-readable medium, and an apparatus are provided. The apparatus may be a device. The device receives content to be displayed on a screen of the device. The device generates a picture based on the content. The device sends the picture to an on-screen display driver bypassing a thin client operating system of the device. The device replaces at least a portion of pixel data in a frame buffer of the device with the picture. The pixel data is presented on the screen. The device transfers the pixel data in the frame buffer to a display driver of the device. The display driver generates display signals based on the pixel data.

Abstract: A sound producing apparatus includes: a plurality of sound producing members with differing oscillation frequencies; dampers each corresponding to a respective sound producing member and suppressing vibration by contacting the corresponding sound producing members; a damper operation mechanism that moves each damper, thereby controlling a state of contact of the damper with a corresponding sound producing members; a soundboard that undergoes sympathetic resonance with sound producing members; a vibrator that vibrates the soundboard; an acquisition unit configured to acquire an audio signal; a signal output unit configured to: generate a drive signal for driving the vibrator with a vibration corresponding to the acquired audio signal; and output the drive signal to the vibrator; and a damper control unit configured to drive the damper operation mechanism to change a state of contact between the sound producing members and their corresponding dampers, based on frequency distribution of acquired audio signal.

Abstract: A capo device for selectively altering the pitch range of a stringed instrument, the capo device being adapted to engage a variety of stringed instrument neck sizes and/or geometries while maintaining desired sound and pitch quality.

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: A frog for a stringed instrument bow is disclosed which has a hollow metal body, which may be made, for example, of direct laser metal sintered titanium. The hollow body has a top side adapted to engage a bow stick of the stringed instrument bow, and a bottom side adapted to support a hair bundle of the stringed instrument bow. A back stop is positioned proximate the bottom side in the interior of the hollow metal body and a cam attached to the bottom side of the hollow metal body is adapted to slide or rotate with respect to the hollow metal body, from an open position in which a hair bundle can be inserted in the hollow metal body and a closed position in which the cam forms a pinch point with the back stop to secure the hair bundle.

Abstract: Exemplary embodiments described herein are directed at instruments and instrument accessories and their associated designs and methods of manufacture based on contoured surfaces and their acoustic properties. Exemplary embodiments may be used to broaden the basic design of stringed instruments or stringed instrument picks or the associated accessories to include forms in three dimensions or using and optimizing the acoustic and/or strength properties inherent in an infinite variety of shapes and sizes.

Abstract: A mute equipment capable of controlling a breath pressure and reducing the volume of the sound during a performance state in normal manner, an adapter thereof and the wind instruments equipped with them, are characterized in that it the mute equipment capable of controlling a breath pressure is to be attached to the blowing section of the wind instruments, wherein the equipment has a hole to release the breath and a section of restricting a wind way width. The mute equipment capable of controlling a breath pressure may be further characterized in that the equipment has a section of adjusting a breath volume to adjust the size of the hole to release the breath. The mute equipment capable of controlling a breath pressure may be further characterized in that the section of adjusting a breath volume can use as the section of restricting the wind way width.

Abstract: A drum, including: a shell; at least one head; and a vibration damping portion including at least one of (a) an outer-surface facing surface which is disposed in substantially parallel with and is closely opposed to an outer surface of an outer peripheral portion of the at least one head without contacting the outer surface and (b) an inner-surface facing surface which is disposed in substantially parallel with and is closely opposed to an inner surface of the outer peripheral portion of the at least one head without contacting the inner surface.

Abstract: It is inter alia disclosed a method comprising: determining a divergence measure between a statistical distribution of audio features of a first audio track and a statistical distribution of audio features of at least one further audio track; determining a divergence measure threshold value from at least the divergence measure between the statistical distribution of audio features of a first audio track and the statistical distribution of audio features of the at least one further audio track; and comparing the divergence measure with the divergence measure threshold value.

Abstract: A system and method eliminates the drag associated with CPU drag on experienced by combine audio streams from multiple sources. The embodiments receive audio signals from two separate audio sources and apply a MIDI based timing signature to the audio signals. The audio signals are synchronized according to the MTC signature signal and combined, when synchronized, at a summing module for output as a combined audio stream.

Abstract: A method includes performing a harmonic decomposition on a target engine sound, thereby to extract each of N harmonics of the target engine sound over an RPM range; and out of the extracted harmonics, extracting phase and shape information for each of the N harmonics over the RPM range for reproducing the target engine sound. The method also includes configuring an engine harmonic enhancement (EHE) system to utilize the extracted phase and shape information to generate an engine harmonic enhancement signal, he(t), to be added on top of a baseline engine sound.

Abstract: A recording method acquires, from a first musical instrument which outputs playing data which represents playing information, playing data of playing by the first musical instrument; generates a first waveform signal according to the played sounds of the first musical instrument which correspond to the playing data; generates a second waveform signal according to a sound including a sound emitted from the first musical instrument and other sounds; generates a third waveform signal wherein the first waveform signal is subtracted from the second waveform signal; generates audio data from the third waveform signal; and records the audio data.

Abstract: A performance enhancing device includes a first glove configured to be worn on a first hand of a user. The first glove includes a plurality of finger pockets, at least one pressure sensor secured proximate a distal end of a respective finger pocket and configured to produce an output signal, a processor coupled to the at least one pressure sensor and configured to receive the output signal and translate the output signal to a wireless output signal, and an antenna coupled to the processor and configured to transmit the wireless signal. The performance enhancing device also includes a jacket configured to be worn by the user, where the jacket includes a receiver to receive the wireless output signal, an amplifier coupled to the receiver, and a speaker integrated within the jacket and coupled to the amplifier and configured to produce an audible sound.

Abstract: A sound wave metamaterial amplifies a sound wave, and includes a plate. The plate has a plurality of sound wave guides passing through both surfaces of the plate and having a predetermined pattern. The sound wave guides are spaced apart from each other by a predetermined distance and face each other, with respect to a central point, a central axis or a central surface.

Abstract: In accordance with embodiments of the present disclosure, a processing system may include multiple selectable processing paths for processing an analog signal in order to reduce noise, increase dynamic range, and mask audio artifacts associated with a change in noise floor. Techniques are employed to transition between processing paths and calibrate operational parameters of the two paths in order to reduce or eliminate artifacts caused by switching between processing paths.

Abstract: A flexible acoustic barrier for suspending from a support frame wherein the flexible acoustic barrier comprises a portion of acoustic insulation material and at least one loudspeaker that is configured to emit an audible sound-masking signal to mask sound from a sound source. Another aspect of the invention comprises a corresponding method of mitigating noise pollution.

Abstract: A noise-cancelling headphone is provided that avoids an influence of the wind and prevents degradation in the sound quality of the reproduced sound output from a driver unit. The noise-cancelling headphone includes an ear piece including a housing unit having an interior and an exterior, a driver unit attached to the housing unit, and a microphone collecting external sounds at the exterior of the housing unit. The housing unit includes an accommodating portion accommodating the microphone and a sound collecting hole establishing the communication between the accommodating portion and the exterior of the housing unit. The accommodating portion is disposed in an upper portion of the housing unit of the noise-cancelling headphone when worn by the user. The sound collecting hole is open toward the upper side of the housing unit of the noise-cancelling headphone when worn by the user.

Abstract: The various embodiments set forth an active noise cancellation system that includes a source separation algorithm. The source separation algorithm enables the identification of acoustic inputs from a particular sound source based on a reference signal generated with one or more microphones. Consequently, the identified acoustic inputs can be cancelled or damped in a targeted listening location via an acoustic correction signal, where the acoustic correction signal is generated based on a sound source separated from the reference signal. Advantageously, the reference signal can be generated with a microphone, even though such a reference signal may include a combination of multiple acoustic inputs. Thus, noise sources that cannot be individually measured, for example with an accelerometer mounted to a vibrating structure, can still be identified and actively cancelled.

Abstract: Noises that are to be emitted by an aerial vehicle during operations may be predicted using one or more machine learning systems, algorithms or techniques. Anti-noises having equal or similar intensities and equal but out-of-phase frequencies may be identified and generated based on the predicted noises, thereby reducing or eliminating the net effect of the noises. The machine learning systems, algorithms or techniques used to predict such noises may be trained using emitted sound pressure levels observed during prior operations of aerial vehicles, as well as environmental conditions, operational characteristics of the aerial vehicles or locations of the aerial vehicles during such prior operations. Anti-noises may be identified and generated based on an overall sound profile of the aerial vehicle, or on individual sounds emitted by the aerial vehicle by discrete sources.

Abstract: A method for automated speech recognition includes generating first and second pluralities of candidate speech recognition results corresponding to audio input data using a first general-purpose speech recognition engine and a second domain-specific speech recognition engine, respectively. The method further includes generating a third plurality of candidate speech recognition result including a plurality of words included in one of the first plurality of speech recognition results and at least one word included in another one of the second plurality of speech recognition results, ranking the third plurality of candidate speech recognition results using a pairwise ranker to identify a highest ranked candidate speech recognition result, and operating the automated system using the highest ranked speech recognition result as an input from the user.

Abstract: A speech recognition apparatus based on a deep-neural-network (DNN) sound model includes a memory and a processor. As the processor executes a program stored in the memory, the processor generates sound-model state sets corresponding to a plurality of pieces of set training speech data included in multi-set training speech data, generates a multi-set state cluster from the sound-model state sets, and sets the multi-set training speech data as an input node and the multi-set state cluster as output nodes so as to learn a DNN structured parameter.

Abstract: A method, which is performed by an electronic device, for obtaining a speaker-independent keyword model of a keyword designated by a user is disclosed. The method may include receiving at least one sample sound from the user indicative of the keyword. The method may also generate a speaker-dependent keyword model for the keyword based on the at least one sample sound, send a request for the speaker-independent keyword model of the keyword to a server in response to generating the speaker-dependent keyword model, and receive the speaker-independent keyword model adapted for detecting the keyword spoken by a plurality of users from the server.

Abstract: In some implementations, systems and methods are capable of performing location-based voice query recognition. Audio corresponding to an utterance of the user and data indicating a location associated with a user are initially received. An initial transcription of the utterance is then obtained. A set of n-grams that are associated with a region are selected for each of multiple regions that are associated with the location. One or more additional candidate descriptions are generated for one or more of the multiple geographic regions. A representative transcription of the utterance is selected for one or more of the multiple geographic regions. The representative transcription is finally provided to an automated speech recognizer.

Abstract: The present invention provides information processing method and electronic apparatus. The method is applied in an electronic apparatus having voice recognition service, and the method includes: obtaining first voice information; recognizing the first voice information by a first recognition model to obtain a first recognition result; deciding whether the first recognition result conforms to a first preset condition; recognizing the first voice information by a second recognition model different from the first recognition model to obtain a second recognition result when the first recognition result conforms to the first preset condition; and controlling the electronic apparatus to execute a corresponding control instruction based on the second recognition result.

Abstract: A semantic network storage unit stores multiple nodes necessary for performing a task of generating a response sentence to user's speech while associating the nodes with each other. A natural language processor acquires speech information indicating content of the user's speech and identifies a primary node corresponding to the speech information from multiple nodes. A conversation generation unit selects one secondary node from multiple secondary nodes associated with the identified primary node, based on weight values each related to one of the multiple secondary nodes and generates a response sentence corresponding to the selected one secondary node.

Abstract: An electronic device may capture a voice command from a user. The electronic device may store contextual information about the state of the electronic device when the voice command is received. The electronic device may transmit the voice command and the contextual information to computing equipment such as a desktop computer or a remote server. The computing equipment may perform a speech recognition operation on the voice command and may process the contextual information. The computing equipment may respond to the voice command. The computing equipment may also transmit information to the electronic device that allows the electronic device to respond to the voice command.

Abstract: Domain specific programming is facilitated through the use of a conversational interface using natural language commands from the user and natural language cues to the user. The natural language conversation provides the actual program and thus can be saved and edited for future use.

Abstract: Features are disclosed for processing a user utterance with respect to multiple subject matters or domains, and for selecting a likely result from a particular domain with which to respond to the utterance or otherwise take action. A user utterance may be transcribed by an automatic speech recognition (“ASR”) module, and the results may be provided to a multi-domain natural language understanding (“NLU”) engine. The multi-domain NLU engine may process the transcription(s) in multiple individual domains rather than in a single domain. In some cases, the transcription(s) may be processed in multiple individual domains in parallel or substantially simultaneously. In addition, hints may be generated based on previous user interactions and other data. The ASR module, multi-domain NLU engine, and other components of a spoken language processing system may use the hints to more efficiently process input or more accurately generate output.

Abstract: A system and method of speech recognition involving a mobile device. Speech input is received (202) on a mobile device (102) and converted (204) to a set of phonetic symbols. Data relating to the phonetic symbols is transferred (206) from the mobile device over a communications network (104) to a remote processing device (106) where it is used (208) to identify at least one matching data item from a set of data items (114). Data relating to the at least one matching data item is transferred (210) from the remote processing device to the mobile device and presented (214) thereon.

Abstract: Aspects relate to computer implemented methods, systems, and processes to automatically generate audio-based display indicia of media content including receiving, by a processor, a plurality of media content categories including at least one feature, receiving a plurality of categorized speech recognition algorithms, each speech recognition algorithm being associated with a respective one or more of the plurality of media content categories, determining a media content category of a current media content based on at least one feature of the current media content, selecting one speech recognition algorithm from the plurality of categorized speech recognition algorithms based on the determination of the media content category of the current media content, and applying the selected speech recognition algorithm to the current media content.

Abstract: A method for generating voice dictionary is disclosed which makes it possible to improve the accuracy of speaker identification. A method according to an aspect of the present disclosure includes: acquiring voices of a plurality of unspecified speakers; acquiring noise in a predetermined place; superimposing the noise onto the voices of the plurality of unspecified speakers; and generating, on the basis of the features of the voices of the plurality of unspecified speakers, unspecified speaker voice dictionary that is used for generating personal voice dictionary for identifying a target speaker.

Abstract: A method and a system for identity authentication are presented. In one example embodiment, audio data (e.g. a sound wave) may be received from a user. The audio data may be used to establish an identity of an entity to the user. The audio data may be stored at a storage location; and be presented to the user to establish the identity of the entity when the entity participates in an electronic communication with the user. In another example embodiment, a server (e.g., a web client or client application server) may present a plurality of audio files to a user; receive a user selection of selected audio data from the plurality of audio files; responsive to the user selection, the server may communicate, via a network, the selected audio data to another server. The selected audio data may be used as an identity authentication.

Abstract: In general, techniques are described for specifying spherical harmonic coefficients in a bitstream. A device comprising one or more processors may perform the techniques. The processors may be configured to identify, from the bitstream, a plurality of hierarchical elements describing a sound field that are included in the bitstream. The processors may further be configured to parse the bitstream to determine the identified plurality of hierarchical elements.

Abstract: In general, techniques are described for closed loop quantization of HOA coefficients that provide a three-dimensional representation of the sound field. An audio encoding device may perform closed loop quantization of an audio object based at least in part on a result of performing quantization of directional information associated with the audio object. An audio decoding device may obtain an audio object that has been closed loop quantized based at least in part on a result of performing quantization of directional information associated with the audio object, and may dequantize the audio object.

Abstract: A device includes a receiver and a decoder. The receiver is configured to receive stereo parameters encoded, by an encoder, based on a plurality of windows having a first length of overlapping portions between the plurality of windows. The decoder is configured to perform an upmix operation using the stereo parameters to generate at least two audio signals. The at least two audio signals are generated based on a second plurality of windows used in the upmix operation. The second plurality of windows has a second length of overlapping portions between the second plurality of windows. The second length is different from the first length.

Abstract: An audio processing unit (APU) is disclosed. The APU includes a buffer memory configured to store at least one frame of an encoded audio bitstream, where the encoded audio bitstream includes audio data and a metadata container. The metadata container includes a header and one or more metadata payloads after the header. The one or more metadata payloads include dynamic range compression (DRC) metadata, and the DRC metadata is or includes profile metadata indicative of whether the DRC metadata includes dynamic range compression (DRC) control values for use in performing dynamic range compression in accordance with at least one compression profile on audio content indicated by at least one block of the audio data.

Abstract: A context-based arithmetic encoding apparatus and method and a context-based arithmetic decoding apparatus and method are provided. The context-based arithmetic decoding apparatus may determine a context of a current N-tuple to be decoded, determine a Most Significant Bit (MSB) context corresponding to an MSB symbol of the current N-tuple, and determine a probability model using the context of the N-tuple and the MSB context. Subsequently, the context-based arithmetic decoding apparatus may perform a decoding on an MSB based on the determined probability model, and perform a decoding on a Least Significant Bit (LSB) based on a bit depth of the LSB derived from a process of decoding on an escape code.

Abstract: In general, techniques are described for coding higher-order ambisonic coefficients during multiple transitions. A device comprising a processor and a memory coupled to the processor may be configured to perform the techniques. The processor may be configured to obtain a multi-transition indication of whether an ambient HOA coefficient is in transition during a same frame of the bitstream as a foreground audio signal is in transition. The processor may also be configured to obtain a vector that describes a spatial characteristic of a corresponding foreground audio signal based on the multi-transition indication, both the vector and the corresponding HOA audio signal decomposed from the HOA audio data. The memory may be configured to store the vector.

Abstract: A voice processing device includes: a sound pickup to receive sounds respectively from a plurality of locations; and circuitry to: identify, from the plurality of locations, a sound source location at which a sound source exists, as a current sound source location; specify at least one location of the plurality of locations other than the current sound source location, from at least one sound source location that has been identified as the sound source location during a past predetermined time period, as a specified source location at which a sound to be enhanced exists; enhance a sound from the current sound source location, and a sound from the specified source location; and output an audio signal including the enhanced sounds.

Abstract: Howling or oscillation in an in-vehicle communications system is prevented when the speaker gain or microphone gain is detected as giving rise to a loop gain equal to or greater than one for a particular frequency or band of frequencies or when the rate of gain of a frequency or band of frequencies increases at a rate that indicating howling will occur. Howling is prevented and not just stopped or suppressed, by adjusting the gain of all frequencies prior to the howling actually starting.

Abstract: The method and system for measuring low-noise acoustical impulse responses at high sampling rates of the present invention utilizes two exponential sine sweeps (ESSs) to measure the impulse responses. The first ESS is a quick sweep up to the Nyquist frequency to provide an estimate of the system response and sample the ambient noise. This measurement is used to algorithmically determine an appropriate pass-band of the system. A second, slower sweep through the pass-band alone is then executed and a corresponding band-pass filter is applied to the resulting output signal to suppress noise. The result is a measured impulse response with an improved signal-to-noise ratio and a much-reduced pre-response.