Abstract: An electronics module for an electric guitar is provided. The electronics module includes a processor, a plurality of controls, an antenna, and a computer-readable medium. The processor receives an audio signal generated by a vibration of a plurality of strings of the electric guitar. The plurality of controls are operably coupled to the processor and provide a mechanism for adjusting a sound created from the audio signal. The computer-readable medium is operably coupled to the processor and configured to cause the electric guitar to determine a control of the plurality of controls associated with the received effects parameter; adjust a state of the determined control based on the received effects parameter; modify the audio signal based on the plurality of controls and on the received effects parameter; and output the modified audio signal through the antenna to a second device.

Abstract: Provided is an electronic musical instrument comprising: an input device for inputting sound generation instructions of tones at predetermined pitches; a tone generation device that generates tones with predetermined pitches based on sound generation instructions inputted by the input device; a specifying device that specifies a plurality of sound generation instructions inputted by the input device in a predetermined period as a sound generation instruction group; a sorting device that sorts the plurality of sound generation instructions composing the sound generation instruction group specified by the specifying device in a predetermined pitch order; and a control device that controls generation of tones by the tone generation device such that tones corresponding to the sound generation instruction group are generated in the order sorted by the sorting device.

Abstract: In one aspect, the present invention is directed to a modular electronic musical keyboard instrument, comprising: an array of separate keyboard segments (30, 32), each comprising a piano keyboard (20) in a range of one octave; a control system, for converting a keystroke of each key in the array of keyboard segments to a note sound in a level corresponding to the order of the key in the array of keyboard segments; octave order setting means (22), for determining to the control system the order of each keyboard segment in the array of keyboard segments; communication means of each of the keyboard segments with the control system; and connection means (12, 14) with another keyboard segment.

Abstract: A computer-implemented method comprises receiving musical data including reference timing data, and a succession of musical notes arranged with respect to the reference timing data, receiving input corresponding to a selection of a groove template, and altering the arrangement of the notes (shifting in a positive or negative direction) with respect to the reference timing data based on the selected groove template. Altering the arrangement of notes can further include adding additional musical notes to the succession of musical notes to add stylistic embellishments particular to the selected groove template, where one of the selectable groove templates includes adding a shuffle dynamic to the succession of musical notes by determining a positive offset associated with each of the musical notes along the musical bar, and applying the positive offset to each of the musical notes, wherein the positive offset corresponds to the position of the musical note.

Abstract: A percussion instrument including a first acoustic chamber housing having a tapered shape, a second acoustic chamber housing having the tapered shape, a sound board having the tapered shape arranged between the first acoustic chamber housing and the second acoustic chamber housing to form a first acoustic chamber defined by the first acoustic chamber housing and the sound board, and a second acoustic chamber defined by the second acoustic chamber housing and the sound board.

Abstract: Provided is a sound source control information generating apparatus, adapted for performing slapping techniques. According to the present invention, information based on an output value of a first sensor that detects striking on the housing is stored in a memory means. If striking on the struck head of a percussion instrument is detected based on an output value of a second sensor that detects striking on the struck head, whether an output value equal to or greater than a predetermined value is obtained from the first sensor in a predetermined time interval before a timing of detecting the striking on the struck head is determined based on the information stored in the memory means.

Abstract: An apparatus for creating a musical composition comprising an audio interface, and audio converter module, and a multi-track compositor module is disclosed. The audio interface operably receives audio from an audio input device and outputting audio to an audio output device. The audio converter module is operably connected to the audio interface to convert audio received via the audio interface into an audio track having one or more partitions. The multi-track compositor module is configured to receive a first audio track and a second audio track and automatically score each partition of the first and second audio tracks based on one or more criteria. The multi-track compositor module is then configured to construct a third audio track from the partitions of the first and second audio tracks based on the scores for each partition. A method is also provided.

Abstract: A method for transcoding music, according to various aspects of the present invention, includes in any practical order: (a) reading pitches and respective durations; (b) reading indicia of a quantity of beats per measure; (c) determining a word for each beat wherein: each word has one or more syllables, each syllable is associated with each pitch having duration that is within the duration of the beat; each syllable for a pitch, when preceded by a rest, comprises an initial consonant selected from the set consisting of ‘d’ and ‘t’; and each syllable comprises a vowel corresponding to an ordinal of the beat, wherein the vowel is selected from a set of vowels in accordance with the respective duration of the pitch associated with the syllable; and (d) outputting, for use by a music engraving engine, indicia of the pitches and words, in a manner that each syllable will be engraved in vertical alignment with the indicia of the associated pitch.

Abstract: A system and method for reproducing audio sound. Audio sound, based on an audio signal, is emitted within a space comprising an audio beamwidth. Modulated ultrasonic sound energy based on a modulated ultrasonic sound signal is emitted in an ultrasonic sound direction within an ultrasonic beamwidth that is less than and within the audio beamwidth. The modulated ultrasonic sound signal is generated such that the emitted modulated ultrasonic sound energy creates an audible cancellation sound with an amplitude substantially equal to an amplitude of the audio sound at a point within the ultrasonic beamwidth so as to combine with and cancel the audio sound at the point by being substantially out of phase with the audio sound along the ultrasonic sound direction.

Abstract: In an embodiment, a circuit arrangement for active noise cancellation, comprises a first input (E1) for supplying a playback signal (Spb), a second input (E2) for supplying a sensor signal (Sanc), a first and a second terminal (A1, A2) of an output that is designed for being connected to a loudspeaker (Lsp) and a compensating device for respectively generating a first and a second noise signal (Sanc1, Sanc2) as a function of the sensor signal (Sanc), wherein the first and the second input (E1, E2) are coupled to the first and the second terminal of the output (A1, A2) by means of the compensating device (Komp) in such a way that a virtual playback signal (Ssp1) is provided at the first terminal (A1) of the output (A1, A2) and a superposition signal (Ssp2) is provided at the second terminal (A2) of the output (A1, A2) such that a differential signal between the virtual playback signal (Ssp1) and the superposition signal (Ssp2) can be fed to the loudspeaker.

Abstract: A high quality speech is reproduced with a small data amount in speech coding and decoding for performing compression coding and decoding of a speech signal to a digital signal. In speech coding method according to a code-excited linear prediction (CELP) speech coding, a noise level of a speech in a concerning coding period is evaluated by using a code or coding result of at least one of spectrum information, power information, and pitch information, and various excitation codebooks are used based on an evaluation result.

Abstract: A screen reader software product for low-vision users, the software having a reader module collecting textual and non-textual display information generated by a web browser or word processor. Font styling, interface layout information and the like are communicated to the end user by sounds broadcast simultaneously rather than serially with the synthesized speech to improve the speed and efficiency in which information may be digested by the end user.

Abstract: A broadcast signal receiver comprises a text data receiver for receiving broadcast text data for display to a user in relation to a user interface; a text-to-speech (TTS) converter for converting received text data into an audio speech signal, the TTS converter being operable to detect whether a word for conversion is included in a stored list of words for conversion and, if so, to convert that word according to a conversion defined by the stored list; and if not, to convert that word according to a set of predetermined conversion rules; a conversion memory storing the list of words for conversion by the TTS converter; and an update receiver for receiving additional words and associated conversions for storage in the conversion memory.

Abstract: An input signal that includes linguistic content in a first language may be received by a computing device. The linguistic content may include text or speech. The computing device may associate the linguistic content in the first language with one or more phonemes from a second language. The computing device may also determine a phonemic representation of the linguistic content in the first language based on use of the one or more phonemes from the second language. The phonemic representation may be indicative of a pronunciation of the linguistic content in the first language according to speech sounds of the second language.

Abstract: The present disclosure discloses a speech recognition method and a terminal, which belong to the field of communications. The method comprises: receiving speech information inputted by a user; acquiring the current environment information, and judging whether the speech information needs to be played according to the current environment information; and recognizing the speech information as text information, when it is judged that the speech information needs not to be played. The terminal comprises an acquisition module, a judgment module and a recognition module. The present disclosure provides the speech receiver with a speech recognition function, when the speech information of the instant messaging is received by the terminal, it can help the receiver to normally acquire the content to be expressed by the speech sender under an inconvenient situation.

Abstract: A speech recognition system adaptively estimates a warping factor used to reduce speaker variability. The warping factor is estimated using a small window (e.g. 100 ms) of speech. The warping factor is adaptively adjusted as more speech is obtained until the warping factor converges or a pre-defined maximum number of adaptation is reached. The speaker may be placed into a group selected from two or more groups based on characteristics that are associated with the speaker's window of speech. Different step sizes may be used within the different groups when estimating the warping factor. VTLN is applied to the speech input using the estimated warping factor. A linear transformation, including a bias term, may also be computed to assist in normalizing the speech along with the application of the VTLN.

Abstract: A system and method are disclosed that improve automatic speech recognition in a spoken dialog system. The method comprises partitioning speech recognizer output into self-contained clauses, identifying a dialog act in each of the self-contained clauses, qualifying dialog acts by identifying a current domain object and/or a current domain action, and determining whether further qualification is possible for the current domain object and/or current domain action. If further qualification is possible, then the method comprises identifying another domain action and/or another domain object associated with the current domain object and/or current domain action, reassigning the another domain action and/or another domain object as the current domain action and/or current domain object and then recursively qualifying the new current domain action and/or current object. This process continues until nothing is left to qualify.

Abstract: A voice-responsive building management system is described herein. One system includes an interface, a dynamic grammar builder, and a speech processing engine. The interface is configured to receive a speech card of a user, wherein the speech card of the user includes speech training data of the user and domain vocabulary for applications of the building management system for which the user is authorized. The dynamic grammar builder is configured to generate grammar from a building information model of the building management system. The speech processing engine is configured to receive a voice command or voice query from the user, and execute the voice command or voice query using the speech training data of the user, the domain vocabulary, and the grammar generated from the building information model.

Abstract: Methods, systems, and apparatus, including computer programs encoded on a computer storage medium, for generating a set of training utterances. The methods, systems, and apparatus include actions of obtaining a target multi-dimensional distribution of characteristics in an initial set of candidate utterances and selecting a subset of the initial set of candidate utterances based on speech recognition confidence scores associated with the candidate utterances. Additional actions include selecting a particular candidate utterance from the subset of the initial set of utterances and determining that adding the particular candidate utterance to a set of training utterances reduces a divergence of a multi-dimensional distribution of the characteristics in the set of training utterances from the target multi-dimensional distribution. Further actions include adding the particular candidate utterance to the set of training utterances.

Abstract: Methods, systems, and apparatus, including computer programs encoded on a computer storage medium, for enhancing speech recognition accuracy. In one aspect, a method includes receiving voice queries, obtaining, for one or more of the voice queries, feedback information that references an action taken by a user that submitted the voice query after reviewing a result of the voice query, generating, for the one or more voice queries, a posterior recognition confidence measure that reflects a probability that the voice query was correctly recognized, wherein the posterior recognition confidence measure is generated based at least on the feedback information for the voice query, selecting a subset of the one or more voice queries based on the posterior recognition confidence measures, and adapting an acoustic model using the subset of the voice queries.

Abstract: Methods, systems, and apparatus, including computer programs encoded on a computer storage medium, for designating certain voice commands as hotwords. The methods, systems, and apparatus include actions of receiving a hotword followed by a voice command. Additional actions include determining that the voice command satisfies one or more predetermined criteria associated with designating the voice command as a hotword, where a voice command that is designated as a hotword is treated as a voice input regardless of whether the voice command is preceded by another hotword. Further actions include, in response to determining that the voice command satisfies one or more predetermined criteria associated with designating the voice command as a hotword, designating the voice command as a hotword.

Abstract: Deep recurrent neural networks applied to speech recognition. The deep recurrent neural networks (RNNs) are preferably implemented by stacked long short-term memory bidirectional RNNs. The RNNs are trained using end-to-end training with suitable regularization.

Abstract: A method and system of providing an interactive manual, including a speech engine to receive and process speech from a user, convert the speech into a word sequence, and identify meaning structures from the word sequence, a structured manual including information related to an operation of a device, a visual model to relate visual representation of the information, a dialog management arrangement to interpret the meaning structures in a context and to extract pertinent information and the visual representation from the structured manual and the visual model, and an output arrangement to output the information and visual representation.

Abstract: A facility and method for analyzing and classifying calls without transcription via keyword spotting is disclosed. The facility uses a group of calls having known outcomes to generate one or more domain- or entity-specific grammars containing keywords and related information that are indicative of particular outcome. The facility monitors telephone calls by determining the domain or entity associated with the call, loading the appropriate grammar or grammars associated with the determined domain or entity, and tracking keywords contained in the loaded grammar or grammars that are spoken during the monitored call, along with additional information. The facility performs a statistical analysis on the tracked keywords and additional information to determine a classification for the monitored telephone call.

Abstract: Systems and methods are provided for receiving speech and non-speech communications of natural language questions and/or commands, transcribing the speech and non-speech communications to textual messages, and executing the questions and/or commands. The invention applies context, prior information, domain knowledge, and user specific profile data to achieve a natural environment for one or more users presenting questions or commands across multiple domains. The systems and methods creates, stores and uses extensive personal profile information for each user, thereby improving the reliability of determining the context of the speech and non-speech communications and presenting the expected results for a particular question or command.

Abstract: An audio signal may be received, in a processor associated with a vehicle. Sound related vehicle information representing one or more sounds may be received by the processor. The sound related vehicle information may or may not include an audio signal. A speech recognition process or system may be modified based on the sound related vehicle information.

Abstract: Methods related to Generalized Mutual Interdependence Analysis (GMIA), a low complexity statistical method for projecting data in a subspace that captures invariant properties of the data, are implemented on a processor based system. GMIA methods are applied to the signal processing problem of voice activity detection and classification. Real-world conversational speech data are modeled to fit the GMIA assumptions. Low complexity GMIA computations extract reliable features for classification of sound under noisy conditions and operate with small amounts of data. A speaker is characterized by a slow varying or invariant channel that is learned and is tracked from single channel data by GMIA methods.

Abstract: Methods, systems, and apparatus, including computer programs encoded on a computer storage medium, for obtaining, for each of multiple words or sub-words, audio data corresponding to multiple users speaking the word or sub-word; training, for each of the multiple words or sub-words, a pre-computed hotword model for the word or sub-word based on the audio data for the word or sub-word; receiving a candidate hotword from a computing device; identifying one or more pre-computed hotword models that correspond to the candidate hotword; and providing the identified, pre-computed hotword models to the computing device.

Abstract: A method and device is disclosed for reducing and controlling stuttering. The method involves tactile feedback of the stutterer's own speech to reducing stuttering. In one embodiment, the device may detect speech by audible or mechanical means, and the feedback may be produced by vibration mechanisms.

Abstract: A computing device can capture video data of at least a portion of a mouth area (e.g., mouth, lips, tongue, chin, jaw) of a user of the device. The computing device can also capture sound data including a voice of the user as well as noise (e.g. background noise). The video data can be processed to detect a movement of the portion of the mouth area. The movement of the portion of the mouth area can be analyzed and compared with mouth area movement models characteristic of oral communication (e.g., speech, song). If the movement of the portion of the mouth area corresponds to at least one model characteristic of oral communication, then the movement indicates that the user is likely engaging in oral communication. Noise reduction can be applied and/or increased on the captured sound data to reduce noise and in turn enhance the user's voice.

Abstract: Concepts and technologies are described herein for multi-mode text input. In accordance with the concepts and technologies disclosed herein, content is received. The content can include one or more input indicators. The input indicators can indicate that user input can be used in conjunction with consumption or use of the content. The application is configured to analyze the content to determine context associated with the content and/or the client device executing the application. The application also is configured to determine, based upon the content and/or the contextual information, which input device to use to obtain input associated with use or consumption of the content. Input captured with the input device can be converted to text and used during use or consumption of the content.

Abstract: A distributed dictation/transcription system is provided. The system provides a dictation manager having a data port to receive and transmit audio signals. The dictation manager includes a dictation server selector to a dictation server to transcribe the audio based on whether the dictation server already has a user profile uploaded.

Abstract: A system that incorporates teachings of the present disclosure may include, for example, a server including a controller to receive audio signals and content identification information from a media processor, generate text representing a voice message based on the audio signals, determine an identity of media content based on the content identification information, generate an enhanced message having text and additional content where the additional content is obtained by the controller based on the identity of the media content, and transmit the enhanced message to the media processor for presentation on the display device, where the enhanced message is accessible by one or more communication devices that are associated with a social network and remote from the media processor. Other embodiments are disclosed.

Abstract: The subject matter of this specification can be implemented in, among other things, a computer-implemented method for correcting words in transcribed text including receiving speech audio data from a microphone. The method further includes sending the speech audio data to a transcription system. The method further includes receiving a word lattice transcribed from the speech audio data by the transcription system. The method further includes presenting one or more transcribed words from the word lattice. The method further includes receiving a user selection of at least one of the presented transcribed words. The method further includes presenting one or more alternate words from the word lattice for the selected transcribed word. The method further includes receiving a user selection of at least one of the alternate words. The method further includes replacing the selected transcribed word in the presented transcribed words with the selected alternate word.

Abstract: Various techniques are disclosed for improving packet loss concealment to reduce artifacts by using audio character measures of the audio signal. These techniques include attenuation to a noise fill instead of attenuation to silence, varying how long to wait before attenuating the extrapolation, varying the rate of attenuation of the extrapolation, attenuating periodic extrapolation at a different rate than non-periodic extrapolation, and performing period extrapolation on successively longer fill data based on the audio character measures, adjusting weighting between periodic and non-periodic extrapolation based on the audio character measures, and adjusting weighting between periodic extrapolation and non-periodic extrapolation non-linearly.

Abstract: A method is provided for allocating bits for quantifying spatial information parameters by frequency sub-band for parametric encoding/decoding of a multichannel audio stream representative of a soundstage consisting of a plurality of sound sources. The method includes a step of quantifying or inversely quantifying, by frequency sub-band, spatial information parameters for the sound sources of the soundscape. The method further includes: assessing a spatial resolution of the current sub-band on the basis of the spectral properties of the sub-band; and determining a number of bits to be allocated to the current sub-band, the number of bits to be allocated being inversely proportional to the estimated spatial resolution. Also provided is a device for allocating quantification bits implementing the above-described method.

Abstract: A method, system and program for decoding a speech signal. In some embodiments, the method comprises: receiving an encoded speech signal having quantization values; transforming the quantization values by adding simulated random-noise samples; and from the encoded speech signal, determining a parameter of the transformation that is usable to control the transformation of the quantization values.

Abstract: A method and system is disclosed for simultaneously determining glottal closure instants (GCIs), fundamental frequency (F0s), and voicing state of a speech signal. A speech signal may be processed to determine a sequence of candidate GCIs. For each candidate GCI, a set of candidate F0s may be determined. A lattice of hypotheses may be constructed, where each lattice point is a hypothesis of a concurrence of a candidate GCI, a candidate F0, and voicing state. Each given hypothesis may also include a score of the candidate GCI, F0, and voicing state for evaluating a cost of the given hypothesis and a cost of connections between the given hypothesis and other hypotheses of the lattice. Dynamic programming may be used to determine a least-cost path through the lattice, and backtracking across the path may be used to determine an optimal set of GCIs, F0s and voicing states of the speech signal.

Abstract: A method (1100) and apparatus (100) generate a candidate code-vector to code an information signal. The method can include producing (1110) a weighted target vector from an input signal. The method can include processing (1120) the weighted target vector through an inverse weighting function to create a residual domain target vector. The method can include performing (1130) a first search process on the residual domain target vector to obtain an initial fixed codebook code-vector. The method can include performing (1140) a second search process over a subset of possible codebook code-vectors for a low weighted-domain error to produce a final fixed codebook code-vector. The subset of possible codebook code-vectors can be based on the initial fixed codebook code-vector. The method can include generating (1150) a codeword representative of the final fixed codebook code-vector. The codeword can be for use by a decoder to generate an approximation of the input signal.

Abstract: A method for controlling an average encoding rate by an electronic device is described. The method includes obtaining a speech signal. The method also includes determining a first average rate. The method further includes determining a first threshold based on the first average rate. The method additionally includes controlling the average encoding rate by determining at least one other threshold based on the first threshold. The method also includes sending an encoded speech signal.

Abstract: Systems and methods for generating and performing a three-dimensional audio CAPTCHA are provided. One exemplary system can include a decoy signal database storing a plurality of decoy signals. The system also can include a three-dimensional audio simulation engine for simulating the sounding of a target signal and at least one decoy signal in an acoustic environment and outputting a stereophonic audio signal based on the simulation. One exemplary method includes providing an audio prompt to a resource requesting entity. The audio prompt can have been generated based on a three-dimensional audio simulation of the sounding of a target signal containing an authentication key and at least one decoy signal in an acoustic environment. The method can include receiving a response to the audio prompt from the resource requesting entity and comparing the response to the authentication key.

Abstract: A method of simultaneously transforming at least two input voice signals xi of a communications system (30), each input voice signal xi being received at a specific reception frequency Fi and corresponding to the voice of a remote party communicating with a user of the communications system (30). During an initialization stage, a transformation Ti is allocated to at least one reception frequency Fi of the input voice signals xi, and during a utilization stage, transformations Ti are applied simultaneously to the input voice signals xi as a function of the reception frequencies Fi, modifying at least one characteristic of each of the input voice signals xi. Thus, the voice of each remote party in communication with the user of the communications system (30) is modified artificially by a transformation Ti, thereby making it easier for the user to perceive and discriminate between simultaneous voices from the remote parties.

Abstract: An audio encoder has a window function controller, a windower, a time warper with a final quality check functionality, a time/frequency converter, a TNS stage or a quantizer encoder, the window function controller, the time warper, the TNS stage or an additional noise filling analyzer are controlled by signal analysis results obtained by a time warp analyzer or a signal classifier. Furthermore, a decoder applies a noise filling operation using a manipulated noise filling estimate depending on a harmonic or speech characteristic of the audio signal.

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: In a method for deep tagging a recording, a computer records audio comprising speech from one or more people. The computer detects a non-speech sound within the audio. The computer determines that the non-speech sound corresponds to a type of sound, and in response, associates a descriptive term with a time of occurrence of the non-speech sound within the recorded audio to form a searchable tag. The computer stores the searchable tag as metadata of the recorded audio.

Abstract: A system for classification of the emotional content of music is provided. An encoder receives a digital audio recording of a piece of music, and encodes it using musical notes and associated amplitudes. The artificial neural network is configured to take a plurality of encoded time slices and provide output indicative of the emotional content of the music.

Abstract: Methods and systems are provided for detecting chop in an audio signal. A time-frequency representation, such as a spectrogram, is created for an audio signal and used to calculate a gradient of mean power per frame of the audio signal. Positive and negative gradients are defined for the signal based on the gradient of mean power, and a maximum overlap offset between the positive and negative gradients is determined by calculating a value that maximizes the cross-correlation of the positive and negative gradients. The negative gradient values may be combined (e.g., summed) with the overlap offset, and the combined values then compared with a threshold to estimate the amount of chop present in the audio signal. The chop detection model provided is low-complexity and is applicable to narrowband, wideband, and superwideband speech.

Abstract: A voice activity detector (VAD) combines the use of an acoustic VAD and a vibration sensor VAD as appropriate to the conditions a host device is operated. The VAD includes a first detector receiving a first signal and a second detector receiving a second signal. The VAD includes a first VAD component coupled to the first and second detectors. The first VAD component determines that the first signal corresponds to voiced speech when energy resulting from at least one operation on the first signal exceeds a first threshold. The VAD includes a second VAD component coupled to the second detector. The second VAD component determines that the second signal corresponds to voiced speech when a ratio of a second parameter corresponding to the second signal and a first parameter corresponding to the first signal exceeds a second threshold.

Abstract: The invention relates to a method for disabling a discontinuous transmission node DTX of a speech encoder if a music signal is detected in a call input signal. The music signal is detected by determining an activity factor corresponding to the relation of sound signal periods relative to scheme signal periods. If the activity factor is higher than a specified activity factor, the DTX is disabled.

Abstract: The present invention provides a search method used to search for the reading order of a plurality of recording groups when the plurality of recording groups written on tape are continuously read by a tape drive which manages data on tape in recording units having a fixed data length for each recording. This search method includes the steps of: receiving information on a plurality of tape groups to be read; and sorting the plurality of recording groups to be read so the reading time is shortened. In the sorting step, the time required to sort the plurality of reading groups is reduced by combining two or more recording groups into a single object to be sorted in the sorting step when at least two or more contiguous recording groups have been assigned to the same region or are assigned across adjacent regions among the plurality of regions.