Abstract: Notwithstanding practical limitations imposed by mobile device platforms and applications, truly captivating musical instruments may be synthesized in ways that allow musically expressive performances to be captured and rendered in real-time. In some cases, synthetic musical instruments can provide a game, grading or instructional mode in which one or more qualities of a user's performance are assessed relative to a musical score. By constantly adapting to such modes to actual performance characteristics and, in some cases, to the level of a given user musician's skill, user interactions with synthetic musical instruments can be made more engaging and may capture user interest and economic opportunities (e.g., for in-app purchase and/or social networking) over generally longer periods of time.

Abstract: A programmable pickup director switching system for a musical instrument having a plurality of pickup coils and a pickup director control board communicatively interconnected to the plurality of pickup coils. The system also includes a push/pull potentiometer structured and arranged to activate different combinations of one or more of the plurality of pickup coils, a main multi-position switch having m positions, and a bank select switch having n positions. Additionally, the system includes a switching matrix configured to switch one or more of the plurality of pickup coils into a signal path based on a position of at least one of the main multi-position switch, the bank select switch and the push/pull potentiometer.

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

Abstract: A magnet or array of magnets is attached to the inner surface of a stringed musical instrument such an acoustic guitar, allowing an external electronic device equipped with one or more magnet or ferromagnetic elements to be softly and removably mounted to the outer surface of the instrument without adhering to or disturbing the outer surface of the instrument. The inner-mounted magnet or magnet array may be attached permanently or removably to the inner surface of the instrument, and may have its position adjusted for optimizing the position of the externally mounted device.

Abstract: The ultrasonic transducer device in accordance with the present invention includes a body unit and a fixing member. The body unit includes a housing including a space which is in a front surface of the housing and is for receiving an ultrasonic transducer, and a clip part provided to a side surface of the housing. The fixing member includes a fixing plate including a fixing surface to be attached to a rear surface of an exterior panel of a vehicle, and a through hole to communicate to an opening in the exterior panel, and an interlocked part removably connected to the clip part so that the space faces the through hole. The clip part includes an interlocking part, and a spring part keeping the interlocking part in a position in which the interlocking part is physically connected to the interlocked part.

Abstract: Disclosed herein are embodiments of an acoustic amplification system for a shoe which includes an amplifier configured to couple to a sole portion of the shoe. The amplifier includes a body; an open-ended cavity disposed within the body; and a sound wave-amplifying conduit disposed within the body, the sound wave-amplifying conduit having opposing inlet and outlet ports, whereby the inlet port acoustically couples to the open-ended cavity and the outlet port acoustically couples to an ambient environment. Following, sound waves entering the inlet port from a sound wave-generating device received within the open-ended cavity are amplified within the sound wave-amplifying conduit to generate amplified sound waves which exit from the outlet port into the ambient environment, the amplified sound waves providing amplified sound.

Abstract: An audio amplifier circuit may include a power amplifier, a charge pump power supply, and a control circuit. The power amplifier may have an audio input for receiving an audio input signal, an audio output for providing the output signal, and a power supply input. The charge pump power supply may provide a power supply voltage to the power supply input. The charge pump power supply may have a select input for selecting an operating mode of the power supply. In a first operating mode, the power supply voltage may equal to a first voltage, and in a second operating mode, the power supply voltage may be substantially equal to a second voltage which is a rational fraction of the first voltage. The control circuit may generate the select input based on a magnitude of anti-noise generated by an adaptive noise cancellation system associated with the audio transducer.

Abstract: A stepped acoustic structure having multiple degrees of acoustic freedom for reducing noise generated from a source. Acoustic septum caps are anchored at the same depth within the cells of the stepped acoustic structure. The multiple degrees of acoustic freedom that is usually provided by locating the septum portions of the septum caps at different depths within the cells is obtained by placing steps within the cells which reduce the cross-sectional area of the cell. Depth control portions are optionally included in the septum caps so that the septum portions of different septum caps are located at different depths within the stepped acoustic structure. Various combinations of steps and depth control portions are used to provide an acoustic structure having multiple degrees of acoustic freedom even though the acoustic septum caps are anchored at the same depth within the structure.

Abstract: A method for noise cancellation. The method includes providing a sound signal output device having a listening device output, a microphone input, and signal processing circuitry. The signal processing circuitry is adapted for causing the sound signal output device to perform in a noise-cancellation mode, including making available to the sound signal output device, at the microphone input, a noise-containing signal which is representative of ambient acoustical noise, deriving a noise-cancelling signal from the noise-containing signal, and making available to the personal listening device, at the listening device output, the noise-cancelling signal.

Abstract: The method is performed at an electronic device with one or more processors and memory storing one or more programs for execution by the one or more processors. A first speech input including at least one word is received. A first phonetic representation of the at least one word is determined, the first phonetic representation comprising a first set of phonemes selected from a speech recognition phonetic alphabet. The first set of phonemes is mapped to a second set of phonemes to generate a second phonetic representation, where the second set of phonemes is selected from a speech synthesis phonetic alphabet. The second phonetic representation is stored in association with a text string corresponding to the at least one word.

Abstract: Systems and processes for analyzing audio input for efficient speech and music recognition are provided. In one example process, an audio input can be received. A determination can be made as to whether the audio input includes music. In addition, a determination can be made as to whether the audio input includes speech. In response to determining that the audio input includes music, an acoustic fingerprint representing a portion of the audio input that includes music is generated. In response to determining that the audio input includes speech rather than music, an end-point of a speech utterance of the audio input is identified.

Abstract: Systems, methods, and computer-readable storage devices are for identifying a user profile for speech recognition. The user profile is selected from one of several user profiles which are all associated with a speaker, and can be selected based on the identity of the speaker, the location of the speaker, the device the speaker is using, or other relevant parameters. Such parameters can be hierarchical, having multiple layers, and can also be dependent or independent from one another. Using the parameters identified, the user profile is selected and used to recognize speech.

Abstract: A system is provided that includes an inspection instrument. The inspection instrument includes communications circuitry, configured to communicatively couple the inspection instrument with information services; an audio input device, configured to receive inspection audio; and a processor, configured to query the information services via sending a request to the information services based upon an analysis of keywords, phrases, or both interpreted from the inspection audio. A voice recognition system configured to analyze the inspection audio for the keywords, phrases, or both is also provided.

Abstract: Methods, systems, and apparatus, including computer programs encoded on computer storage media, for generating phoneme representations of acoustic sequences using projection sequences. One of the methods includes receiving an acoustic sequence, the acoustic sequence representing an utterance, and the acoustic sequence comprising a respective acoustic feature representation at each of a plurality of time steps; for each of the plurality of time steps, processing the acoustic feature representation through each of one or more long short-term memory (LSTM) layers; and for each of the plurality of time steps, processing the recurrent projected output generated by the highest LSTM layer for the time step using an output layer to generate a set of scores for the time step.

Abstract: A server and a guide sentence generating method are provided. The method includes receiving user speech, analyzing the user speech, determining a category of the user speech from among a plurality of categories, storing the user speech in the determined category, determining a usage frequency and a popularity of each of the plurality of categories, selecting a category from among the plurality of categories based on the usage frequency and the popularity, and generating a guide sentence corresponding to the selected category.

Abstract: An automated method is described for developing an automated speech input semantic classification system such as a call routing system. A set of semantic classifications is defined for classification of input speech utterances, where each semantic classification represents a specific semantic classification of the speech input. The semantic classification system is trained from training data from training data substantially without manually transcribed in-domain training data, and then operated to assign input speech utterances to the defined semantic classifications. Adaptation training data based on input speech utterances is collected with manually assigned semantic labels from at least one source of already collected language data. When the adaptation training data satisfies a pre-determined adaptation criteria, the semantic classification system is automatically retrained based on the adaptation training data.

Abstract: Described herein are systems and methods for the generation and maintenance of language models. Language models are developed based at least in part on transaction data from one or more users. These transactions may include purchases and other interactions between one or more users and one or more online merchants. The language models may be associated with a particular user or group of users. Ongoing transactions may modify the language models. The language models may be used to recognize spoken input from the one or more users.

Abstract: The present disclosure relates to a syntax parsing apparatus based on syntax preprocessing and a method thereof. In specific, the present disclosure parses syntaxes that can be parsed by rules and patterns without ambiguity by syntax parsing preprocessing, draws all possible syntax parsing results by applying syntax rules based on a result of syntax parsing preprocessing in which ambiguity is partially resolved, and resolves structural ambiguity by applying a statistic syntax parsing model learned from a syntax tree attachment learning corpus so as to reduce ambiguity in rule-based syntax parsing and to resolve ambiguity by a statistics-based scheme so that parsing correctness and processing efficiency in a syntax parsing method can be enhanced.

Abstract: A conversational, natural language voice user interface may provide an integrated voice navigation services environment. The voice user interface may enable a user to make natural language requests relating to various navigation services, and further, may interact with the user in a cooperative, conversational dialogue to resolve the requests. Through dynamic awareness of context, available sources of information, domain knowledge, user behavior and preferences, and external systems and devices, among other things, the voice user interface may provide an integrated environment in which the user can speak conversationally, using natural language, to issue queries, commands, or other requests relating to the navigation services provided in the environment.

Abstract: Methods, systems, and apparatus, including computer programs encoded on a computer storage medium, receiving audio data; determining that an initial portion of the audio data corresponds to an initial portion of a hotword; in response to determining that the initial portion of the audio data corresponds to the initial portion of the hotword, selecting, from among a set of one or more actions that are performed when the entire hotword is detected, a subset of the one or more actions; and causing one or more actions of the subset to be performed.

Abstract: A method of operation of a content delivery system includes: receiving a command phrase based on determining an utterance type according to a travel context; determining a trigger match with a control unit based on the command phrase matching a trigger phrase; and stopping a prompt according to a prompt type based on the trigger match for controlling the prompt presented by a device.

Abstract: Systems and methods may provide for determining a sound vibration condition of an ambient environment of a wearable device and determining a motion condition of the wearable device. In addition, one or more automated voice operations may be performed based at least in part on the sound vibration condition and the motion condition. In one example, two or more signals corresponding to the sound vibration condition and the motion condition may be combined.

Abstract: In one embodiment, a semantic classifier input and a corresponding label attributed to the semantic classifier input may be obtained. A determination may be made whether the corresponding label is correct based on logged interaction data. An entry of an adaptation corpus may be generated based on a result of the determination. Operation of the semantic classifier may be adapted based on the adaptation corpus.

Abstract: A method and system for monitoring video assets provided by a multimedia content distribution network includes testing closed captions provided in output video signals. A video and audio portion of a video signal are acquired during a time period that a closed caption occurs. A first text string is extracted from a text portion of a video image, while a second text string is extracted from speech content in the audio portion. A degree of matching between the strings is evaluated based on a threshold to determine when a caption error occurs. Various operations may be performed when the caption error occurs, including logging caption error data and sending notifications of the caption error.

Abstract: Systems and methods are provided for converting taxiway voice commands into taxiway textual commands. In various embodiments, the systems can comprise a radio receiver that is configured to receive the taxiway voice commands from an air traffic control center, a voice recognition processor coupled to the radio receiver that is configured to receive and convert the taxiway voice commands into the taxiway textual commands, and/or a taxiway clearance display coupled to the voice recognition processor that is configured to receive and display the taxiway textual commands.

Abstract: A minutes taking system includes a voice recognition apparatus and an image forming apparatus. The voice recognition apparatus receives voice signals from the image forming apparatus, performs voice recognition processing on the received voice signals, and returns voice recognition results to the image forming apparatus. The image forming apparatus includes voice input units, a storage unit, a voice transmission unit, a voice recognition result receiving unit, a minutes data creating unit, and an image forming unit. The storage unit accumulates the voice signals input from the voice input units. The voice transmission unit transmits a non-transmitted voice signal, among the voice signals stored in the storage unit, to the voice recognition apparatus when a predetermined condition is satisfied. The minutes data creating unit creates minutes data based on the voice recognition result received by the voice recognition result receiving unit.

Abstract: A vehicle based system and method for receiving voice inputs and determining whether to perform a voice recognition analysis using in-vehicle resources or resources external to the vehicle.

Abstract: Described is a technology by which speech is locally and remotely recognized in a hybrid way. Speech is input and recognized locally, with remote recognition invoked if locally recognized speech data was not confidently recognized. The part of the speech that was not confidently recognized is sent to the remote recognizer, along with any confidently recognized text, which the remote recognizer may use as context data in interpreting the part of the speech data that was sent. Alternative text candidates may be sent instead of corresponding speech to the remote recognizer.

Abstract: Some aspects of the invention may include a computer-implemented method for enrolling voice prints generated from audio streams, in a database. The method may include receiving an audio stream of a communication session and creating a preliminary association between the audio stream and an identity of a customer that has engaged in the communication session based on identification information. The method may further include determining a confidence level of the preliminary association based on authentication information related to the customer and if the confidence level is higher than a threshold, sending a request to compare the audio stream to a database of voice prints of known fraudsters. If the audio stream does not match any known fraudsters, sending a request to generate from the audio stream a current voice print associated with the customer and enrolling the voice print in a customer voice print database.

Abstract: In some embodiments, a system may process a user interface to identify textual or graphical items in the interface, and may prepare a plurality of audio files containing spoken representations of the items. As the user navigates through the interface, different ones of the audio files may be selected and played, to announce text associated with items selected by the user. A computing device may periodically determine whether a cache offering the interface to users stores audio files for all of the interface's textual items, and if the cache is missing any audio files for any of the textual items, the computing device may take steps to have a corresponding audio file created.

Abstract: A method of providing an input method for an electronic device includes receiving a command for switching a display image of the electronic device; determining whether at least two words correspond to at least two languages when at least the two words are in the display image; determining a word count of a first language and a word count of a second language when the two words correspond to the two languages; and enabling a first input method corresponding to the first language when the word count of the first language is greater than the word count of the second language, or enabling a second input method corresponding to the second language when the word count of the second language is greater than the word count of the first language.

Abstract: According to one of aspects, an electronic device includes a sound input module and a controller. The controller is configured to control a voice input through the sound input module to be activated when detecting occurrence of an event to be notified to a user.

Abstract: Systems, computer-implemented methods, and tangible computer-readable media are provided for voice authentication. The method includes receiving a speech sample from a user through an Internet browser for authentication as part of a request for a restricted-access resource, pre-authenticating the speech sample, transmitting an authentication request to an authentication server, which compares the speech sample to a previously established speech profile associated with the user, and providing access to the restricted access resource based on the response.

Abstract: The instant application includes computationally-implemented systems and methods that include managing adaptation data, the adaptation data is at least partly based on at least one speech interaction of a particular party, facilitating transmission of the adaptation data to a target device when there is an indication of a speech-facilitated transaction between the target device and the particular party, such that the adaptation data is to be applied to the target device to assist in execution of the speech-facilitated transaction, and facilitating acquisition of adaptation result data that is based on at least one aspect of the speech-facilitated transaction and to be used in determining whether to modify the adaptation data. In addition to the foregoing, other aspects are described in the claims, drawings, and text.

Abstract: An apparatus for coding a portion of an audio signal to obtain an encoded audio signal for the portion of the audio signal includes a transient detector for detecting whether a transient signal is located in the portion of the audio signal to obtain a transient detection result, an encoder stage for performing first and second encoding algorithms on the audio signal, the first and second encoding algorithms having differing first and second characteristics, respectively, a processor for determining which encoding algorithm results in an encoded audio signal being a better approximation to the portion of the audio signal with respect to the other encoding algorithm to obtain a quality result, and a controller for determining whether the encoded audio signal for the portion of the audio signal is to be generated by either the first or the second encoding algorithm based on the transient-detection and quality results.

Abstract: A system and method are provided for processing sound signals. The processing may include identifying individual harmonic sounds represented in sound signals, determining sound parameters of harmonic sounds, classifying harmonic sounds according to source, and/or other processing. The processing may include transforming the sound signals (or portions thereof) into a space which expresses a transform coefficient as a function of frequency and chirp rate. This may facilitate leveraging of the fact that the individual harmonics of a single harmonic sound may have a common pitch velocity (which is related to the chirp rate) across all of its harmonics in order to distinguish an the harmonic sound from other sounds (harmonic and/or non-harmonic) and/or noise.

Abstract: Systems and methods for adjusting audio levels in a plurality of audio signals are disclosed. One or more input signals containing one or more audio signals are depacketized. The audio signals are analyzed to determine if audio levels of the signals correspond to a desired audio level for at least some of the signals. The audio levels of signals that do not correspond are adjusted. Packets that do not require adjustment and the resulting adjusted packets are combined to form one or more output signals corresponding to the one or more input signals.

Abstract: The processing efficiency of a process used to decode frames of an enhanced AC-3 bit stream is improved by processing each audio block in a frame only once. Audio blocks of encoded data are decoded in block order rather than in channel order. Exemplary decoding processes for enhanced bit stream coding features such as adaptive hybrid transform processing and spectral extension are disclosed.

Abstract: An audio processing device includes an initial processing module to generate a stream of frequency coefficients based on input audio data, a watermarking module to embed a digital watermark into the stream of frequency coefficients to generate a modified stream of frequency coefficients, and a final processing module to process the modified stream of frequency coefficients to generate output audio data. In some implementations, the input audio data comprises unencoded audio data, the initial processing module comprises a frequency domain transform module to perform a time-to-frequency domain transform to generate the unencoded audio data, and the output audio data is encoded audio data. In other instances, the input audio data comprises encoded audio data, the initial processing module comprises an initial decoding module to partially decode the encoded audio data to generate the stream of frequency coefficients, and the output audio data is decoded audio data.

Abstract: A method includes determining, at a speech encoder, first gain shape parameters based on a harmonically extended signal and/or based on a high-band residual signal associated with a high-band portion of an audio signal. The method also includes determining second gain shape parameters based on a synthesized high-band signal and based on the high-band portion of the audio signal. The method further includes inserting the first gain parameters and the second gain shape parameters into an encoded version of the audio signal to enable gain adjustment during reproduction of the audio signal from the encoded version of the audio signal.

Abstract: An audio encoding 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 plurality of lobes based on a frequency signal constituting an audio signal; calculating a masking threshold value of the frequency signal; allocating an amount of bits per unit frequency region to be allocated for encoding of the frequency signal on a basis of the masking threshold value; selecting a main lobe on a basis of bandwidth and power of the lobes; and controlling the encoding by reducing the amount of bits in a first region including a maximum value of the power in the main lobe.

Abstract: Improved methods for coding an ensemble of pulse vectors utilize statistical models (i.e., probability models) for the ensemble of pulse vectors, to more efficiently code each pulse vector of the ensemble. At least one pulse parameter describing the non-zero pulses of a given pulse vector is coded using the statistical models and the number of non-zero pulse positions for the given pulse vector. In some embodiments, the number of non-zero pulse positions are coded using range coding. The total number of unit magnitude pulses may be coded using conditional (state driven) bitwise arithmetic coding. The non-zero pulse position locations may be coded using adaptive arithmetic coding. The non-zero pulse position magnitudes may be coded using probability-based combinatorial coding, and the corresponding sign information may be coded using bitwise arithmetic coding. Such methods are well suited to coding non-independent-identically-distributed signals, such as coding video information.

Abstract: In general, techniques are described for coding of vectors decomposed from higher-order ambisonic coefficients. A device comprising a memory and a processor may perform the techniques. The memory may be configured to store audio data. The processor may be configured to determine whether to perform vector dequantization or scalar dequantization with respect to a decomposed version of the plurality of HOA coefficients.

Abstract: The invention relates to a codec and a discriminator and methods therein for audio signal discrimination and coding. Embodiments of a method performed by an encoder comprises, for a segment of the audio signal: identifying a set of spectral peaks; determining a mean distance S between peaks in the set; and determining a ratio, PNR, between a peak envelope and a noise floor envelope. The method further comprises selecting a coding mode, out of a plurality of coding modes, based at least on the mean distance S and the ratio PNR; and applying the selected coding mode for coding of the segment of the audio signal.

Abstract: A method of coding/decoding of a digital audio signal comprising a succession of consecutive blocks of data, on the basis of a predictive filter. A modified predictive filter is used for the coding of at least one current block, the modified filter being constructed by the combination of: a rear filter calculated for a past block, preceding the current block, and enrichment parameters for the rear filter, which are determined as a function of the signal in the current block and comprising the coefficients of a modifying filter.

Abstract: A system for providing voice audio to an operator. The system includes: one or more audio inputs for receiving at least two received voice audio signals; a processing unit, a digital to analog converter, and a loudspeaker. The processing unit is connected to the one or more audio inputs, and configured to: adjust the pitch of a first voice audio signal of the at least two received voice audio signals to form a first adjusted voice audio signal; and combine the first adjusted voice audio signal with at least one other received voice audio signal or adjusted voice audio signal to form a composite audio signal. The pitch in the audio allows the listener to disambiguate one or more speakers or conveys to the listener attributes such as urgency or location information.

Abstract: Methods and apparatuses for addressing open space noise are disclosed. In one example, a method for masking open space noise includes outputting a test signal from a speaker, the test signal operable to measure a speech transmission quality of a transmission channel. The method includes receiving the test signal at a microphone, outputting a detected test signal, and processing the detected test signal to determine the speech transmission quality of the transmission channel. The method further includes adjusting an output level of a noise masking signal from the speaker responsive to the speech transmission quality.

Abstract: Techniques for providing self-voice feedback in a communications headset include processing signals carrying near-end speech in parallel digital and analog signal processing paths to produce a combined gain-adjusted near-end signal carrying the near-end speech for output to transducers of the communications device.

Abstract: A vehicle-to-vehicle communication device generates voice information that includes a voice message and added information regarding an output of the voice message. The voice information is transmitted in one direction of a subject vehicle via a transmission unit, and voice information from another vehicle is received via a reception unit. The vehicle-to-vehicle communication device plays the voice message of the voice information received by the reception unit based on the added information of the voice information. In such manner, information regarding a travel situation is appropriately transmitted by the vehicle-to-vehicle communication device.

Abstract: Presented are improvements for speech recognition systems used to control devices. Features include two-stage confirmation, two-stage limited speech recognition mode, and two-stage wake-up for speech driven applications and systems. A headset computer device includes such staged confirmation operation.