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Patents

 
Software framework and development platform for multi-sensor systems
Tue, 21 Oct 2014 08:00:00 EDT
The disclosed software framework and development platform facilitates software development for multi-sensor systems. In some implementations, developers can select a sensor board that includes a desired combination of sensor devices. The sensor board can be coupled to a development board that includes a target processor and other circuitry to facilitate development and testing of a system that includes the target processor and the sensors. Various software support tools are provided including an Application Programming Interface (API) that provides API abstractions for software drivers for the sensors on the sensor board. By using the abstractions of the API, a software developer does not have to write code (“glue”) to interact with the various software drivers. Additionally, the API provides access to a variety of software library functions for performing data scaling, unit conversion and mathematical functions and algorithms.
Identification of semantic relationships within reported speech
Tue, 21 Oct 2014 08:00:00 EDT
Methods and computer-readable media for associating words or groups of words distilled from content, such as reported speech or an attitude report, of a document to form semantic relationships collectively used to generate a semantic representation of the content are provided. Semantic representations may include elements identified or parsed from a text portion of the content, the elements of which may be associated with other elements that share a semantic relationship, such as an agent, location, or topic relationship. Relationships may also be developed by associating one element that is in relation to, or is about, another element, thereby allowing for rapid and effective comparison of associations found in a semantic representation with associations derived from queries. The semantic relationships may be determined based on semantic information, such as potential meanings and grammatical functions of each element within the text portion of the content.
Parallel processing of data sets
Tue, 21 Oct 2014 08:00:00 EDT
Systems, methods, and devices are described for implementing learning algorithms on data sets. A data set may be partitioned into a plurality of data partitions that may be distributed to two or more processors, such as a graphics processing unit. The data partitions may be processed in parallel by each of the processors to determine local counts associated with the data partitions. The local counts may then be aggregated to form a global count that reflects the local counts for the data set. The partitioning may be performed by a data partition algorithm and the processing and the aggregating may be performed by a parallel collapsed Gibbs sampling (CGS) algorithm and/or a parallel collapsed variational Bayesian (CVB) algorithm. In addition, the CGS and/or the CVB algorithms may be associated with the data partition algorithm and may be parallelized to train a latent Dirichlet allocation model.
System and method for phrase identification
Tue, 21 Oct 2014 08:00:00 EDT
A phrase identification system and method are provided. The method comprises: identifying one or more phrase candidates in the electronic document; selecting one of the phrase candidates; numerically representing features of the selected phrase candidates to obtain a numeric feature representation associated with that phrase candidate; and inputting the numeric feature representation into a machine learning classifier, the machine learning classifier being configured to determine, based on each numeric feature representation, whether the phrase candidate associated with that numeric feature representation is a phrase.
Audio decoder and decoding method using efficient downmixing
Tue, 21 Oct 2014 08:00:00 EDT
A method, an apparatus, a computer readable storage medium configured with instructions for carrying out a method, and logic encoded in one or more computer-readable tangible medium to carry out actions. The method is to decode audio data that includes N.n channels to M.m decoded audio channels, including unpacking metadata and unpacking and decoding frequency domain exponent and mantissa data; determining transform coefficients from the unpacked and decoded frequency domain exponent and mantissa data; inverse transforming the frequency domain data; and in the case M
Audio signal bandwidth extension in CELP-based speech coder
Tue, 21 Oct 2014 08:00:00 EDT
A method for decoding an audio signal having a bandwidth that extends beyond a bandwidth of a CELP excitation signal in an audio decoder including a CELP-based decoder element. The method includes obtaining a second excitation signal having an audio bandwidth extending beyond the audio bandwidth of the CELP excitation signal, obtaining a set of signals by filtering the second excitation signal with a set of bandpass filters, scaling the set of signals using a set of energy-based parameters, and obtaining a composite output signal by combining the scaled set of signals with a signal based on the audio signal decoded by the CELP-based decoder element.
Recognition dictionary creation device and voice recognition device
Tue, 21 Oct 2014 08:00:00 EDT
A recognition dictionary creation device identifies the language of a reading of an inputted text which is a target to be registered and adds a reading with phonemes in the language identified thereby to the target text to be registered, and also converts the reading of the target text to be registered from the phonemes in the language identified thereby to phonemes in a language to be recognized which is handled in voice recognition to create a recognition dictionary in which the converted reading of the target text to be registered is registered.
Methods, devices, and computer program products for providing real-time language translation capabilities between communication terminals
Tue, 21 Oct 2014 08:00:00 EDT
A language translation device includes a network transceiver configured to provide communication between first and second communication terminals, a language recognition unit, and a language translation unit. The language recognition unit is configured to receive a signal representing speech and/or text in a first human language from the first communication terminal and map the received signal to intermediate data. The language translation unit is configured to generate a translated signal representing speech and/or text in a second human language, which is different from the first human language, in response to the intermediate data and provide the translated signal to the second communication terminal. The speech and/or text represented by the translated signal has a meaning corresponding to that of the speech and/or text represented by the received signal. Related devices and methods of operation are also discussed.
Method and device for storing audio data
Tue, 21 Oct 2014 08:00:00 EDT
A method for storing audio data is disclosed, including: recording basic information of a versatile audio data storage file into the versatile audio data storage file; storing Versatile Audio Codec (VAC) frame data into the versatile audio data storage file sequentially; recording payload information of the versatile audio data storage file into the versatile audio data storage file; and recording index information of VAC frames stored in the versatile audio data storage file into the versatile audio data storage file. A device for storing the audio data is also disclosed, including: a basic information record module, a VAC frame data storage module, a payload information record module and an index information record module. The file generated with this method is simple and is easy to read and access, which can be applied to various applications of the versatile audio frequently.
Integration of embedded and network speech recognizers
Tue, 21 Oct 2014 08:00:00 EDT
A method, computer program product, and system are provided for performing a voice command on a client device. The method can include translating, using a first speech recognizer located on the client device, an audio stream of a voice command to a first machine-readable voice command and generating a first query result using the first machine-readable voice command to query a client database. In addition, the audio stream can be transmitted to a remote server device that translates the audio stream to a second machine-readable voice command using a second speech recognizer. Further, the method can include receiving a second query result from the remote server device, where the second query result is generated by the remote server device using the second machine-readable voice command and displaying the first query result and the second query result on the client device.
System and method for updating information in electronic calendars
Tue, 21 Oct 2014 08:00:00 EDT
Systems and methods for updating electronic calendar information. Speech is received from a user at a vehicle telematics unit (VTU), wherein the speech is representative of information related to a particular vehicle trip. The received speech is recorded in the VTU as a voice memo, and data associated with the voice memo is communicated from the VTU to a computer running a calendaring application. The data is associated with a field of the calendaring application, and stored in association with the calendaring application field.
Screen reader with focus-based speech verbosity
Tue, 21 Oct 2014 08:00:00 EDT
The amount of speech output to a blind or low-vision user using a screen reader application is automatically adjusted based on how the user navigates to a control in a graphic user interface. Navigation by mouse presumes the user has greater knowledge of the identity of the control than navigation by tab keystroke which is more indicative of a user searching for a control. In addition, accelerator keystrokes indicate a higher level of specificity to set focus on a control and thus less verbosity is required to sufficiently inform the screen reader user.
System and method for providing network coordinated conversational services
Tue, 21 Oct 2014 08:00:00 EDT
A system and method for providing automatic and coordinated sharing of conversational resources, e.g., functions and arguments, between network-connected servers and devices and their corresponding applications. In one aspect, a system for providing automatic and coordinated sharing of conversational resources includes a network having a first and second network device, the first and second network device each comprising a set of conversational resources, a dialog manager for managing a conversation and executing calls requesting a conversational service, and a communication stack for communicating messages over the network using conversational protocols, wherein the conversational protocols establish coordinated network communication between the dialog managers of the first and second network device to automatically share the set of conversational resources of the first and second network device, when necessary, to perform their respective requested conversational service.
Interactive voice response data collection object framework, vertical benchmarking, and bootstrapping engine
Tue, 21 Oct 2014 08:00:00 EDT
A method, a system, and computer readable medium comprising instructions for analyzing data of a speech application are provided. The method comprises defining a set of data collection objects for a call flow in a speech application, collecting data using the set of data collection objects during execution of the speech application, and analyzing the data using a benchmarking and bootstrapping engine, storing the data in a repository, and presenting the data for analysis.
System and method for controlling access to resources with a spoken CAPTCHA test
Tue, 21 Oct 2014 08:00:00 EDT
Systems and methods for controlling access to resources using spoken Completely Automatic Public Turing Tests To Tell Humans And Computers Apart (CAPTCHA) tests are disclosed. In these systems and methods, entities seeking access to resources are required to produce an input utterance that contains at least some audio. That utterance is compared with voice reference data for human and machine entities, and a determination is made as to whether the entity requesting access is a human or a machine. Access is then permitted or refused based on that determination.
Storing a representative speech unit waveform for speech synthesis based on searching for similar speech units
Tue, 21 Oct 2014 08:00:00 EDT
According to one embodiment, a method for editing speech is disclosed. The method can generate speech information from a text. The speech information includes phonologic information and prosody information. The method can divide the speech information into a plurality of speech units, based on at least one of the phonologic information and the prosody information. The method can search at least two speech units from the plurality of speech units. At least one of the phonologic information and the prosody information in the at least two speech units are identical or similar. In addition, the method can store a speech unit waveform corresponding to one of the at least two speech units as a representative speech unit into a memory.
Methods and systems for identifying errors in a speech recognition system
Tue, 21 Oct 2014 08:00:00 EDT
A method for model adaptation for a speech recognition system includes determining an error rate, corresponding to either recognition of instances of a word or recognition of instances of various words, without using a transcript of words input to the system. The method may further include adjusting an adaptation, of the model for the word or various models for the various words, based on the error rate. Apparatus are disclosed for identifying possible errors made by a speech recognition system without using a transcript of words input to the system. An apparatus for model adaptation for a speech recognition system includes a processor adapted to estimate an error rate, corresponding to either recognition of instances of a word or recognition of instances of various words, without using a transcript of words input to the system. The apparatus may further include a controller adapted to adjust an adaptation of the model for the word or various models for the various words, based on the error rate.
Continuous speech transcription performance indication
Tue, 21 Oct 2014 08:00:00 EDT
A method of providing speech transcription performance indication includes receiving, at a user device data representing text transcribed from an audio stream by an ASR system, and data representing a metric associated with the audio stream; displaying, via the user device, said text; and via the user device, providing, in user-perceptible form, an indicator of said metric. Another method includes displaying, by a user device, text transcribed from an audio stream by an ASR system; and via the user device, providing, in user-perceptible form, an indicator of a level of background noise of the audio stream. Another method includes receiving data representing an audio stream; converting said data representing an audio stream to text via an ASR system; determining a metric associated with the audio stream; transmitting data representing said text to a user device; and transmitting data representing said metric to the user device.
Generalizing text content summary from speech content
Tue, 21 Oct 2014 08:00:00 EDT
A text content summary is created from speech content. A focus more signal is issued by a user while receiving the speech content. The focus more signal is associated with a time window, and the time window is associated with a part of the speech content. It is determined whether to use the part of the speech content associated with the time window to generate a text content summary based on a number of the focus more signals that are associated with the time window. The user may express relative significance to different portions of speech content, so as to generate a personal text content summary.
Receiver intelligibility enhancement system
Tue, 21 Oct 2014 08:00:00 EDT
Embodiments of the invention provide a communication device and methods for enhancing audio signals. A first audio signal buffer and a second audio signal buffer are acquired. Thereafter, the magnitude spectrum calculated from the Fast Fourier Transform (FFT) of the second audio signal is processed based on the Linear Predictive Coding (LPC) spectrum of the first audio signal to generate an enhanced second audio signal.
Handset intelligibility enhancement system using adaptive filters and signal buffers
Tue, 21 Oct 2014 08:00:00 EDT
Embodiments of the invention provide a communication device and methods for enhancing audio signals. A first audio signal buffer and a second audio signal buffer are acquired. Thereafter, the second audio signal is processed based on the linear predictive coding coefficients and gains based on noise power of the first audio signal to generate an enhanced second audio signal.
Apparatus and method for cancelling echo in joint time domain and frequency domain
Tue, 21 Oct 2014 08:00:00 EDT
Disclosed in the present invention is a method for cancelling echo in joint time domain and frequency domain. The method includes: receiving an input receiver signal and an input transmitter signal; implementing echo cancellation on the received transmitter signal, based on the received receiver signal, by using a first echo canceller which is either a time domain echo canceller or a frequency domain echo canceller, to obtain a first echo-cancelled signal; implementing echo cancellation again on the first echo-cancelled signal, based on the received receiver signal, by using a second echo canceller which is the other one of the time domain echo canceller and the frequency domain echo canceller, to obtain a second echo-cancelled signal; wherein, and the first and second echo canceller respectively include the corresponding first and second filters, and the filter parameter of the second filter is updated based on the input receiver signal, the second echo-cancelled signal and the filter parameter of the first filter. By using said method in the present invention, fast response to echo reflecting environment can be achieved with little residual echo, thus the effect of echo cancellation is entirely improved.
Discontinuous transmission control based on vocoder and voice activity
Tue, 21 Oct 2014 08:00:00 EDT
A method and system is disclosed for control of discontinuous transmission based on vocoder and voice activity. An access terminal (AT) may engage in a communication session via an encoder-decoder in a network device in a wireless network. During silence intervals of the communication session, when the AT has no data to transmit, the AT may transmit periodic silence frames at a silence-frame rate to the encoder-decoder. The silence frames may contain parameters for generation of audio noise by the network device. Upon determining that the encoder-decoder has ceased transmitting data to the AT in response to a prolonged absence of transmissions from the AT, the AT may increase the silence-frame rate so as to reduce the duration of the absence of transmissions from the AT, and correspondingly cause the encoder-decoder to begin transmitting audio data to the AT.
Audio signal processing device with enhancement of low-pitch register of audio signal
Tue, 21 Oct 2014 08:00:00 EDT
An audio signal processing device is designed to enhance the low-pitch register of an audio signal by generating harmonics causing a missing fundamental effect with a light load of processing but without damaging an audio waveform. The audio signal processing device includes a filtering part (e.g. a band-pass filter configured of a high-pass filter and a low-pass filter) that extracts a low-pitch signal from an audio signal input thereto; a dynamic range compression part that compresses a dynamic range of the low-pitch signal by use of a time-variant gain relative to a peak of the low-pitch signal, which is detected via a peak hold operation using a predetermined time constant, thus producing a compressed signal; and an adder that adds the compressed signal to the audio signal so as to produce a processed audio signal including harmonics.
Accelerometer vector controlled noise cancelling method
Tue, 21 Oct 2014 08:00:00 EDT
A telecommunication device is disclosed, comprising: a microphone array comprising a plurality of microphones, wherein each microphone receives an analogue acoustic signal; a position sensing device for determining how the telecommunication device is positioned in three-dimensions with respect to a user's mouth; at least one analogue/digital converter for converting each analogue acoustic signal into a digital signal; a digital signal processor for performing signal processing on the received digital signals comprising a controller, a plurality of delay circuits for delaying each received signal based on an input from the controller and a plurality of preamplifiers for adjusting the gain of each received signal based on a gain input from the controller, wherein the controller selects the appropriate delay and gain values applied to each received signal to remove noise from the received signals based on the determined position of the telecommunication device. A method for creating and controlling a location of a virtual microphone near a telecommunication device so as to reduce background noise in a speech signal is also disclosed.
Pitch-correction of vocal performance in accord with score-coded harmonies
Tue, 21 Oct 2014 08:00:00 EDT
Despite many practical limitations imposed by mobile device platforms and application execution environments, vocal musical performances may be captured and continuously pitch-corrected for mixing and rendering with backing tracks in ways that create compelling user experiences. In some cases, the vocal performances of individual users are captured on mobile devices in the context of a karaoke-style presentation of lyrics in correspondence with audible renderings of a backing track. Such performances can be pitch-corrected in real-time at a portable computing device (such as a mobile phone, personal digital assistant, laptop computer, notebook computer, pad-type computer or netbook) in accord with pitch correction settings. In some cases, pitch correction settings include a score-coded melody and/or harmonies supplied with, or for association with, the lyrics and backing tracks. Harmonies notes or chords may be coded as explicit targets or relative to the score coded melody or even actual pitches sounded by a vocalist, if desired.
Non-dialogue-based and dialogue-based learning apparatus by substituting for uttered words undefined in a dictionary with word-graphs comprising of words defined in the dictionary
Tue, 21 Oct 2014 08:00:00 EDT
The invention provides a dialogue-based learning apparatus through dialogue with users comprising: a speech input unit (10) for inputting speeches; a speech recognition unit (20) for recognizing the input speech; and a behavior and dialogue controller (30) for controlling behaviors and dialogues according to speech recognition results, wherein the behavior and dialogue controller (30) has a topic recognition expert (34) to memorise contents of utterances and to retrieve the topic that best matches the speech recognition results, and a mode switching expert (35) to control mode switching in accordance with a user utterance, wherein the mode switching expert switches modes in accordance with a user utterance, wherein the topic recognition expert registers a plurality words in the utterance as topics in first mode, performs searches from among the registered topics, and selects the maximum likelihood topic in second mode. Finally for every uttered word which is not found in a predetermined dictionary, it substitutes the said word with a word-graph comprising of either a known or a combination of known words in the said dictionary.
Evaluating transcriptions with a semantic parser
Tue, 21 Oct 2014 08:00:00 EDT
In some implementations, audio data for an utterance is provided over a network. At a client device and over the network, information is received that indicates candidate transcriptions for the utterance and semantic information for the candidate transcriptions. A semantic parser is used at the client device to evaluate each of at least a plurality of the candidate transcriptions. One of the candidate transcriptions is selected based on at least the received semantic information and the output of the semantic parser for the plurality of candidate transcriptions that are evaluated.
Systems and methods for word offensiveness processing using aggregated offensive word filters
Tue, 21 Oct 2014 08:00:00 EDT
Computer-implemented systems and methods are provided for identifying language that would be considered obscene or otherwise offensive to a user or proprietor of a system. A first plurality of offensive words are received, and a second plurality of offensive words are received. A string of words are received, where one or more detected offensive words are selected from the string of words that matches words from the first plurality of offensive words or the second plurality of offensive words. The string of words is processed based upon the detection of offensive words in the string of words.
Language processor
Tue, 21 Oct 2014 08:00:00 EDT
A referring expression processor which uses a probabilistic model and in which referring expressions including descriptive, anaphoric and deictic expressions are understood and generated in the course of dialogue is provided. The referring expression processor according to the present invention includes: a referring expression processing section which performs at least one of understanding and generation of referring expressions using a probabilistic model constructed with a referring expression Bayesian network, each referring expression Bayesian network representing relationships between a reference domain (D) which is a set of possible referents, a referent (X) in the reference domain, a concept (C) concerning the referent and a word (W) which represents the concept; and a memory which stores data necessary for constructing the referring expression Bayesian network.

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