Abstract: The present invention relates to a bit rate controller comprising: a light determination block configured to determine a light level in images of a captured scene; and a maximum bit rate setting block. The maximum bit rate setting block is configured to: upon the light level being determined to be a low light level, set a low-light maximum bit rate; upon the light level being determined to be an intermediate light level, set an intermediate-light maximum bit rate; or upon the light level being determined to be a high light level, set a high-light maximum bit rate. The low-light maximum bit rate is lower than the intermediate-light maximum bit rate. The high-light maximum bit rate is lower than the intermediate-light maximum bit rate.

Abstract: Displays, display components, image and video processing apparatus and related methods are described. A method for driving local-dimming displays comprises generating control values for driving pixels of a spatial light modulator from one image data component and generating control values for driving backlight elements from a second image data component. The first and second image data components may respectively comprise a tone map and a ratio image. Control values for the spatial light modulator and/or backlight may be obtained using cost effective hardware.

Abstract: An attachable article includes a flexible electronic display disposed thereon in a manner so that the display is bendable or conformable to a user's wrist or other curved surface thus enabling various images to be displayed on the electronic display in a manner that makes these images easily viewable to the user. The attachable article includes an adjustable band that can be fit to different sized wrists, for example, and includes a band orientation detection and calibration routine that enables messages or display screens to be placed at particular locations on the band with respect to a user's wrist. For example, the routine may be used to determine the portions or positions of the display that are at the top of the user's wrist and at the bottom of the user's wrist, and may thereafter calibrate the display to center display screens on the display at these particular locations.

Abstract: An electronic device comprises a display configured to display an image, at least one sensor configured to acquire information, and at least one controller. The at least one controller is configured to detect a first area of the display in contact with water and a second area of the display not in contact with water, and adjust the image so that the image is within the second area when both the first area and the second area are detected.

Abstract: Waking-up a violin or other instrument by determining naturally resonating features of a particular instrument by applying tones and analyzing the instrument's responses, and then applying vibrations (e.g., continuously and/or in patterns, such as alternating on/off, staccato, etc.) corresponding to the most responsive (resonant) portions of the instrument's responses (e.g., peaks of the analyzed instrument's response) fully breaking in the various parts of the instrument. The applied vibrations are continued until the instrument's amplitude response plateaus.

Abstract: A method and device for facilitating learning of musical scales on string instruments are provided. The device comprises a visual scheme consisting of blocks to be positioned just below notes of a scale when fixed on an instrument, each block being formed by the overlap of two rectangles in distinct sizes, which serve as space for accommodation of different didactic information; a manual providing information for the understanding and proper use of the visual scheme; and a musical background providing tracks that sound in the same frequency or tonality of the visual scheme. The device can further include a complementary visual scheme useful in practicing scales in different tonalities.

Abstract: The invention is based on a musical instrument case, in particular for storage of a string instrument, with at least one housing, with at least one storage region, which is delimited by the housing, for storage of a musical instrument, and with at least one fixing unit, which is provided to fixate a musical instrument in the storage region at least substantially fixedly as regards a position. It is proposed that the at least one fixing unit comprises at least one structural element of low flexural strength, which is in a fixated state of a musical instrument provided to contact a body of the musical instrument.

Abstract: A humidifier and digital hygrometer/thermometer device for use with a stringed, musical instrument having a sound hole includes: (a) a polyvinyl acetate sponge adapted to hold a specified volume of water, (b) a container having an opening adapted to accommodate the sponge and fit within the instrument's sound hole, (c) a case adapted to rest on the instrument's strings , (d) a relative humidity measuring means, (e) a temperature measuring means, (f) a digital display means, (g) a processing element adapted to process the data from the humidity and temperature measuring means and cause the digital display means to display appropriate humidity and temperature measurements, data and further adapted to fit within the case, (h) an adapter affixed to the case and adapted to detachably attach the container to the case.

Abstract: Providing feedback on a musical performance performed with a musical instrument. An instrument profile associated with the musical instrument used to perform the musical performance is identified. The instrument profile comprises information relating to one or more tuning characteristics of the instrument. The pitch of notes of the musical performance are analyzed based on the instrument profile to determine a measure of tuning of the musical performance. A feedback signal is generated based on the determined measure of tuning.

Abstract: Described is a trigger module that can be used with percussive instruments to generate an electric signal. The trigger module includes an enclosure with a strike plate. A piezoelectric transducer is affixed with the bottom of the strike plate. A striker having a protrusion is attached with the enclosure such that the protrusion rests proximate the strike plate. Thus, forcing the striker towards the strike plate causes the protrusion to impact the strike plate, thereby causing the piezoelectric transducer to generate an electric signal.

Abstract: A sound masking system for shaping the ambient noise level in a physical environment. The sound masking system comprises a networked and distributed system having a number of master units coupled together and to a control unit. One or more of the master units may include satellite sound masking units which function to reproduce the sound masking signal generated by the master sound masking unit. Each of the master units is addressable over the network by the control unit enabling the control unit to program the contour, spectral band, and gain characteristics of the sound masking output signal. The system may also include a remote control unit which provides the capability to tune and adjust each master sound masking unit in situ without requiring physical access through the ceiling installation.

Abstract: An active noise control (ANC) system includes a speaker and one or more processors. The one or more processors implement an adaptive subband filtered reference control algorithm that applies thresholds to reference and error feedback signal paths such that, in response to a series of broadband non-Gaussian impulsive reference signals indicative of road noise in the vehicle having an audible frequency range of 20 Hz to 20 kHz, weight coefficients defining an adaptive filter of the control algorithm converge and permit the ANC system to partially cancel the road noise via output of the speaker.

Abstract: Techniques for receiving a voice command from a user and, in response, providing audible content to the user via a first device and providing visual content for the user via a second device. In some instances, the first device includes a microphone for generating audio signals that include user speech, as well as a speaker for outputting audible content in response to identified voice commands from the speech. However, the first device might not include a display for displaying graphical content. As such, the first device may be configured to identify devices that include displays and that are proximate to the first device. The first device may then instruct one or more of these other devices to output visual content associated with a user's voice command.

Abstract: Speech recognition systems that use voice templates may create (or update) voice templates for a particular user by training (or re-training). If a training results in a vocabulary with similar voice templates, then the speech recognition system's performance may suffer. The present invention provides embraces methods for training a speech recognition system to prevent voice template similarity. In these methods, a trained word's voice template may be evaluated for similarity to other vocabulary templates prior to enrolling the voice template into the vocabulary. If template similarity is found, then a user may be prompted to retrain the system using an alternate word. Alternatively, the user may be prompted to retrain the system with the word spoken more clearly. This dynamic enrollment training analysis insures that all templates in the vocabulary are distinct.

Abstract: A language model for automatic speech processing, such as a finite state transducer (FST) may be configured to incorporate information about how a particular word sequence (N-gram) may be used in a similar manner from another N-gram. A score of a component of the FST (such as an arc or state) relating to the first N-gram may be based on information of the second N-gram. Further, the FST may be configured to have an arc between a state of the first N-gram and a state of the second N-gram to allow for cross N-gram back off, rather than backoff from a larger N-gram to a smaller N-gram during traversal of the FST during speech processing.

Abstract: Disclosed herein are systems, methods, and computer-readable storage media for a speech recognition application for directory assistance that is based on a user's spoken search query. The spoken search query is received by a portable device and portable device then determines its present location. Upon determining the location of the portable device, that information is incorporated into a local language model that is used to process the search query. Finally, the portable device outputs the results of the search query based on the local language model.

Abstract: Efficient empirical determination, computation, and use of an acoustic confusability measure comprises: (1) an empirically derived acoustic confusability measure, comprising a means for determining the acoustic confusability between any two textual phrases in a given language, where the measure of acoustic confusability is empirically derived from examples of the application of a specific speech recognition technology, where the procedure does not require access to the internal computational models of the speech recognition technology, and does not depend upon any particular internal structure or modeling technique, and where the procedure is based upon iterative improvement from an initial estimate; (2) techniques for efficient computation of empirically derived acoustic confusability measure, comprising means for efficient application of an acoustic confusability score, allowing practical application to very large-scale problems; and (3) a method for using acoustic confusability measures to make principled

Abstract: Computer-generated speech devices for site survey and maintenance, and methods of using the same are described herein. One computer-generated speech device includes a location engine to determine a location of the computer-generated speech device at a site, a solution engine to identify an action to perform associated with one of a plurality of nodes at the site using the location of the computer-generated speech device and input data associated with the plurality of nodes, and a speech engine to broadcast the identified action as computer-generated speech using a speaker component of the computer-generated speech device for a user to perform during a survey and/or maintenance of the site.

Abstract: An automatic speech recognition (ASR) system detects an endpoint of an utterance using the active hypotheses under consideration by a decoder. The ASR system calculates the amount of non-speech detected by a plurality of hypotheses and weights the non-speech duration by the probability of each hypotheses. When the aggregate weighted non-speech exceeds a threshold, an endpoint may be declared.

Abstract: A first microphone is operated in a low power sensing mode, and a buffer at the first microphone is used to temporarily store at least some of the phrase. Subsequently the first microphone is deactivated, then the first microphone is re-activated to operate in normal operating mode where the buffer is no longer used to store the phrase. The first microphone forms first data that does not include the entire phrase. A second microphone is maintained in a deactivated mode until the trigger portion is detected in the first data, and when the trigger portion is detected, the second microphone is caused to operate in normal operating mode where no buffer is used. The second microphone forms second data that does not include the entire phrase. A first electronic representation of the phrase as received at the first microphone and a second electronic representation of the phrase as received at the second microphone are formed from selected portions of the first data and the second data.

Abstract: Disclosed is an apparatus and method for determining which controllable device an audible command is directed towards, the method comprising: receiving at each of two or more controlling devices the audible command signal, the audible command being directed to control at least one of two or more controllable devices controlled by a respective one of the two or more controlling devices; digitizing each of the received audible command signals; attaching a unique identifier to each digitized audible command so as to uniquely correlate it to a respective controlling device; determining a magnitude of each of the digitized audible command; determining a digitized audible command with the greatest magnitude, and further determining to which controlling device the audible command is directed to on the basis of the unique identifier associated with the digitized audible command with the greatest magnitude; performing speech recognition on the digitized audible command with the greatest magnitude; and forwarding a com

Abstract: Contextual note taking is described. A note taking assistant can receive an indication of a specific presentation session. This indication can be used by the note taking assistant to access information or content related to the session. The note taking assistant can receive specific presentation session content, which includes identifiable context images. Identifiable context images are meant to define an individual page, an individual slide, or other atomic unit in the presentation. The note taking assistant operates by receiving a navigation message, changing the current assistant context image to a current presenter context image based on the navigation message; receiving a speech-to-text message comprising a unit of text; displaying the current presenter context image, and displaying the unit of text associated with the current presenter context image; and storing the unit of text associated with the current presenter context image.

Abstract: A computer-implemented system and method for performing distributed speech recognition is provided. Audio data is collected. A main grammar and secondary grammars are simultaneously provided for the audio data. Each secondary grammar includes an independent grammar. Speech recognition is simultaneously performed on the audio data using each secondary grammar. A new grammar is constructed for the audio data based on the main grammar template using results of the speech recognition. Further speech recognition is performed on the audio data using the new grammar.

Abstract: Disclosed herein are systems, methods, and non-transitory computer-readable storage media relating to speaker verification. In one aspect, a system receives a first user identity from a second user, and, based on the identity, accesses voice characteristics. The system randomly generates a challenge sentence according to a rule and/or grammar, based on the voice characteristics, and prompts the second user to speak the challenge sentence. The system verifies that the second user is the first user if the spoken challenge sentence matches the voice characteristics. In an enrollment aspect, the system constructs an enrollment phrase that covers a minimum threshold of unique speech sounds based on speaker-distinctive phonemes, phoneme clusters, and prosody. Then user utters the enrollment phrase and extracts voice characteristics for the user from the uttered enrollment phrase.

Abstract: A system and method for embedding digital audio watermarks in audio source information based at least upon identified video content are described. An audio/video processing system receives audiovisual data. A video content analyzer within the system analyzes video source information of the audiovisual data, determines video content depicted by data in the video source information, and generates an indication of the video content. An audio watermark embedder of the system receives the indication, and based at least in part on the indication, adjusts watermark embedding parameters used for embedding the audio watermark in the audio source information.

Abstract: A recording device, such as a video camera, emits audio watermarks (e.g., audible tones, inaudible tones) and records audio watermarks emitted by other recording devices. The audio watermarks provide information (e.g., data) such as an identifier of the camera and the identity of the operate. Audio watermarks may further include time information such as a timestamp indicating the time that the audio watermark was emitted by the camera. Audio watermarks may be emitted and recorded at intervals while recording an incident. A processing system, such as a digital video and audio processing system, may receive video information and recorded watermarks from multiple recording devices. The processing system may use the audio watermarks recorded by each recording device to determine which recording devices were in the vicinity of an incident. The processing system may also use the audio watermarks to align (e.g., synchronize) the video recorded by the various recording devices.

Abstract: There is provided methods and apparatuses for decoding and encoding of audio signals. In particular, a method for decoding includes receiving a waveform-coded signal having a spectral content corresponding to a subset of the frequency range above a cross-over frequency. The waveform-coded signal is interleaved with a parametric high frequency reconstruction of the audio signal above the cross-over frequency. In this way an improved reconstruction of the high frequency bands of the audio signal is achieved.

Abstract: A method for processing audio data includes determining a first common scalefactor value for representing quantized audio data in a frame. A second common scalefactor value is determined for representing the quantized audio data in the frame. A line equation common scalefactor value is determined from the first and second common scalefactor values.

Abstract: A gain adjustment apparatus (60) for use in decoding of audio that has been encoded with separate gain and shape representations includes an accuracy meter (62) configured to estimate an accuracy measure (A(b)) of the shape representation (Ñ(b)), and to determine a gain correction (gc(b)) based on the estimated accuracy measure (A(b)). It also includes an envelope adjuster (64) configured to adjust the gain representation (Ê(b)) based on the determined gain correction.

Abstract: Provided are an apparatus and a method for integrally encoding and decoding a speech signal and an audio signal. The encoding apparatus may include: an input signal analyzer to analyze a characteristic of an input signal; a first conversion encoder to convert the input signal to a frequency domain signal, and to encode the input signal when the input signal is an audio characteristic signal; a Linear Predictive Coding (LPC) encoder to perform LPC encoding of the input signal when the input signal is a speech characteristic signal; a frequency band expander for expanding a frequency band of the input signal whose output is transmitted to either the time-domain encoding module or the transform encoding module based on the input characteristic; and a bitstream generator to generate a bitstream using an output signal of the first conversion encoder and an output signal of the LPC encoder.

Abstract: In accordance with an example embodiment of the present invention, disclosed is a method and an apparatus thereof for formatting a payload for transmission of multi-mode speech/audio codec data. The method comprises deciding whether a header-less or a header-full payload format is used for transmission of a coded frame. The decision is based on a codec mode and a required functionality. The payload data is packetized with or without the payload header depending on the decision.

Abstract: A method and an apparatus for decoding a speech/audio bitstream are disclosed, where the method for decoding a speech/audio bitstream includes determining whether a current frame is a normal decoding frame or a redundancy decoding frame, obtaining a decoded parameter of the current frame by means of parsing when the current frame is a normal decoding frame or a redundancy decoding frame, performing post-processing on the decoded parameter of the current frame to obtain a post-processed decoded parameter of the current frame, and using the post-processed decoded parameter of the current frame to reconstruct a speech/audio signal.

Abstract: The techniques disclosed herein enable a system to coordinate audio objects that are generated by multiple applications. A system can receive contextual data from several applications and dynamically determine an allocation of a number of audio objects for each application based on the contextual data. The allocation can be based on a status of one or more applications, user interactions with one or more applications, and other factors. Policy data can also cause the system to allocate a number of audio objects to one or more applications based on an application type and other factors. For instance, a policy may cause a system to allocate more audio objects to a game application vs. a communications application. As a user interacts with an application, e.g., moves or resizes a user interface, closes an application, increases or decreases a level of interaction, the system can reallocate audio objects to individual applications.

Abstract: The invention relates to a codec and a signal classifier and methods therein for signal classification and selection of a coding mode based on audio signal characteristics. A method embodiment to be performed by a decoder comprises, for a frame m: determining a stability value D(m) based on a difference, in a transform domain, between a range of a spectral envelope of frame m and a corresponding range of a spectral envelope of an adjacent frame m?1. Each such range comprises a set of quantized spectral envelope values related to the energy in spectral bands of a segment of the audio signal. The method further comprises selecting a decoding mode, out of a plurality of decoding modes, based on the stability value D(m); and applying the selected decoding mode.

Abstract: A signaling processor is provided. The signaling processor includes a frequency domain processing module configured to generate a cut-off frequency of an input signal and to generate level information for adjusting a level of a high frequency recovery signal and a time domain processing module configured to receive the cut-off frequency and the level information from the frequency domain processing module, to generate a signal having a frequency greater than or equal to the cut-off frequency using part of a signal of a frequency lower than the cut-off frequency in the input signal, to generate the high frequency recovery signal by adjusting a level of the generated signal using the level information, and to synthesize the high frequency recovery signal with the input signal.

Abstract: Methods and systems are provided for optimizing call quality and improving network efficiency by reducing bandwidth requirements at the individual-voice-call level. Embodiments provided herein build vocal fingerprints that correspond to the frequency range of the human voice, as well as the frequency range of the voice of individual users. The vocal fingerprints are used minimize and reduce the transmission of background noise and ambient sound captured using HD voice while retaining the frequency range of a user's voice in HD voice. This filtered HD voice frequency range is then transmitted to recipients over the network. The reduced frequency range lowers bandwidth usage and conserves network resources, all while optimizing the call quality for individual users.

Abstract: A processing method includes acquiring an audio signal, magnifying one or more first portions of the audio signal in response to one or more magnitudes of the one or more first portions of the audio signal are lower than a predetermined threshold, deforming one or more second portions of the audio signal in response to one or more magnitudes of the one or more second portions of the audio signal are greater than the predetermined threshold, and performing a noise reduction to the one or more deformed second portions of the audio signal to reduce the one or more deformed second portions of the audio signal.

Abstract: An acoustic signal processing system and method. In accordance with an embodiment, the acoustic signal processing system includes an adaptive filter that filters a signal from a frequency band reservation module and generates a filter signal that is received by a subtractor. The subtractor generates an error signal that is used by a double-talk indicator module to generate a control signal that indicates the presence of double-talk.

Abstract: The invention relates to a method, comprising receiving a first audio signal during an application execution in an apparatus, determining a volume level of a volume control interface for controlling a volume level of the first audio signal, controlling processing the first audio signal by at least one digital signal processing algorithm determined on the basis of the determined volume level of the volume control interface for controlling the volume level of the first audio signal, and outputting the controlled processed first audio signal at the determined volume level of the volume control interface. The invention further relates to an apparatus and a computer program product that perform the method.

Abstract: A voice converting apparatus and a voice converting method are provided. The method of converting a voice using a voice converting apparatus including receiving a voice from a counterpart, analyzing the voice and determining whether the voice abnormal, converting the voice into a normal voice by adjusting a harmonic signal of the voice in response to determining that the voice is abnormal, and transmitting the normal voice.

Abstract: The present disclosure provides systems and methods for selecting and presenting content items based on user input. The method includes receiving first input intended to identify a desired content item among content items associated with metadata, determining that an input portion has an importance measure exceeding a threshold, and providing feedback identifying the input portion. The method further includes receiving second input, and inferring user intent to alter or supplement the first input with the second input. The method further includes, upon inferring intent to alter the first input, determining an alternative query by modifying the first input based on the second input, and, upon inferring intent to supplement the first input, determining an alternative query by combining the first input and the second input. The method further includes selecting and presenting a subset of content items based on comparing the alternative query and metadata associated with the subset.

Abstract: A speech-capture device can capture audio data during wakeword monitoring and use the audio data to determine if a user is present nearby the device, even if no wakeword is spoken. Audio such as speech, human originating sounds (e.g., coughing, sneezing), or other human related noises (e.g., footsteps, doors closing) can be used to detect audio. Audio frames are individually scored as to whether a human presence is detected in the particular audio frames. The scores are then smoothed relative to nearby frames to create a decision for a particular frame. Presence information can then be sent according to a periodic schedule to a remote device to create a presence “heartbeat” that regularly identifies whether a user is detected proximate to a speech-capture device.

Abstract: A magnetic write apparatus has a media-facing surface (MFS), a pole having leading and trailing surfaces, a trailing shield having a pole-facing surface, a write gap and coil(s). The pole's trailing surface has a portion adjoining the MFS and oriented at a nonzero, acute bevel angle from a direction perpendicular to the MFS. The pole-facing surface includes a first portion adjoining the MFS and oriented at a first angle substantially the same as the bevel angle, a second portion oriented at a second angle greater than the first trailing shield angle, and a third portion oriented at a third angle substantially the same as the first angle. The write gap has first, second and third thicknesses adjacent to the first, second and third portions of the pole-facing surface, respectively. The first thickness is constant. The second thickness varies. The third thickness is constant and greater than the first thickness.

Abstract: A write head includes a waveguide, a magnetic pole, and a near-field transducer. The near-field transducer includes an enlarged portion and a peg. The peg is separated from the magnetic pole in a downtrack direction by a dielectric gap. A peg coupler covers a bottom surface of the magnetic pole and is separated from the peg. The peg coupler is formed of a first plasmonic material. A pad extends from the peg coupler into part of the gap in the downtrack direction towards the peg. The pad is formed of a second plasmonic material and extends into the write head away from the media-facing surface a distance L that is less than a corresponding distance of the peg coupler.

Abstract: A magnetic recording head has a main magnetic pole that generates a recording magnetic field applied on a magnetic recording medium from an end surface that is one portion of an air bearing surface facing the magnetic recording medium; a trailing shield provided on the trailing side of the main magnetic pole with a write gap interposed therebetween; and a spin torque oscillator that is provided in the write gap to be interposed between the main magnetic pole and the trailing shield and that generates a magnetic field that exerts an interaction on the recording magnetic field applied on the magnetic recording medium.

Abstract: An apparatus, according to one embodiment, includes: an array of write transducers, each write transducer having: a first write pole having a pole tip extending from a media facing side of the first write pole, a second write pole having a pole tip extending from a media facing side of the second write pole, a nonmagnetic write gap between the pole tips of the write poles, and a high moment layer between the write gap and the pole tip of the second write pole. The high moment layer has a higher magnetic moment than a magnetic moment of the pole tip of the second write pole. Moreover, the high moment layer protrudes beyond a plane extending along a media facing side of the second pole tip. Other systems, methods, and computer program products are described in additional embodiments.

Abstract: A read sensor that includes a free layer having a magnetization that changes according to an external magnetic field. The read sensor also includes an additional magnetic layer and a non-magnetic layer. The non-magnetic layer may include a corrugated surface facing the additional magnetic layer. The corrugated surface is configured to enhance uniaxial anisotropy in the read sensor.

Abstract: In one embodiment, a system includes a sensor, the sensor having a free layer, a ferromagnetic spin sink layer spaced from the free layer, the spin sink layer being operative to reduce a spin-induced damping in the free layer during operation of the sensor, and a nonmagnetic spacer layer positioned between the free layer and the spin sink layer, the spacer layer having a long spin-diffusion length.

Abstract: A magnetic read apparatus includes a read sensor, a shield structure and a side magnetic bias structure. The read sensor includes a free layer having a side and a nonmagnetic spacer layer. The shield structure includes a shield pinning structure and a shield reference structure. The nonmagnetic spacer layer is between the shield reference structure and the free layer. The shield reference structure is between the shield pinning structure and the nonmagnetic spacer layer. The shield pinning structure includes a pinned magnetic moment in a first direction. The shield reference structure includes a shield reference structure magnetic moment weakly coupled with the pinned magnetic moment. The side magnetic bias structure is adjacent to the side of the free layer.

Abstract: An apparatus according to one embodiment includes an array of transducer structures arranged along a tape bearing surface of a common module. Each transducer structure includes a lower shield, an upper shield above the lower shield, a current-perpendicular-to-plane sensor between the upper and lower shields, at least one lead, and an insulating layer between the at least one lead and the shield closest thereto. The at least one lead is selected from a group including an upper electrical lead between the sensor and the upper shield, and a lower electrical lead between the sensor and the lower shield. The at least one lead is in electrical communication with the sensor. A width of one or more of the at least one lead in a cross track direction is about equal to a width of the sensor.