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Patents

 
Interactive ring-shaped interface
Tue, 15 Apr 2014 08:00:00 EDT
A computer method and system for interactive visualization and modification of a user interest model via an interactive graphical user interface is described. The method and system comprise displaying at least one keyword in a circular sector on a ring on the interactive graphical user interface, wherein the keyword represents an interest of the user interest model, wherein a radial position of the keyword on the ring represents a degree of interest stored as part of the user interest model, and wherein the circular sector represents a semantic type of the user interest model. The method and system further enable a user to actively influence the modeling server and graphically modify/change the degree of interest via an interaction component of the interactive graphical user interface.
Foreign language translation tool
Tue, 15 Apr 2014 08:00:00 EDT
Embodiments of the present invention provide a foreign language translation tool that assists in the localization of an application. In particular, the foreign language translation tool enables translation in the context of a running application. In addition, the tool provides an editing mechanism so that the user may modify or correct the translation of the selected text and allows the user to easily and quickly verify the correctness of translations.
Data extraction system, terminal apparatus, program of the terminal apparatus, server apparatus, and program of the server apparatus for extracting prescribed data from web pages
Tue, 15 Apr 2014 08:00:00 EDT
This invention provides a terminal searching for web pages on the web and extracting the prescribed data from the web pages and a server verifying and accumulating the extracted data. The prescribed data can be extracted from the web pages on the web in a manner that the process relating to the data extraction is distributed between the terminal and the server. Therefore, necessary processes up to the data extraction are distributed, and the burden placed on each apparatus can be lessened. Further, new data not formerly found in the web pages can be found out and extracted from the web pages that has been updated or newly made.
Enabling multidimensional search on non-PC devices
Tue, 15 Apr 2014 08:00:00 EDT
Multidimensional search capabilities are enabled on a non-PC (personal computer) device being utilized by a user. An original query submitted by the user via the non-PC device is received. A structured data repository is accessed to extract structured data that is available for the original query, where the extracted structured data represents attributes of the original query. The extracted structured data is provided to the user in the form of a hierarchical menu which allows the user to interactively modify the original query, such modification resulting in a revised query.
Compressed domain encoding apparatus and methods for use with media signals
Tue, 15 Apr 2014 08:00:00 EDT
Apparatus, methods, and articles of manufacture for encoding a compressed media stream are disclosed. A disclosed method encodes frames associated with one or more compressed media streams within the media signal on a frame-by-frame basis and releases for transmission an encoded version of the media signal containing the encoded frames on a packet-by-packet basis.
Method and system for lossless value-location encoding
Tue, 15 Apr 2014 08:00:00 EDT
A method of encoding samples in a digital signal is provided that includes receiving a frame of N samples of the digital signal, determining L possible distinct data values in the N samples, determining a reference data value in the L possible distinct data values and a coding order of L−1 remaining possible distinct data values, wherein each of the L−1 remaining possible distinct data values is mapped to a position in the coding order, decomposing the N samples into L−1 coding vectors based on the coding order, wherein each coding vector identifies the locations of one of the L−1 remaining possible distinct data values in the N samples, and encoding the L−1 coding vectors.
Real-time versioning of device-bound content
Tue, 15 Apr 2014 08:00:00 EDT
Subject matter described herein relates to providing to a mobile device a version of content (e.g., music, video, text message, live call, etc.) that is consistent with a user's filter setting. That is, a user is allowed to specify content elements (e.g., words or images) that are proscribed from being presented on the mobile device, and the user's preferences are stored by a mobile telecommunications network. When the network receives content to be provided to the mobile device, the network edits the content in real time to prevent proscribed elements from being presented on the mobile device.
In-vehicle apparatus and information display system
Tue, 15 Apr 2014 08:00:00 EDT
An in-vehicle apparatus receives an image data representative of a screen image from a portable terminal with a touch panel. The apparatus extracts a text code data from the image data, and identifies a text-code display area in the screen image. The apparatus determines a command text based on a user-uttered voice command. The apparatus identifies a text-code display area as a subject operation area in the screen image of the portable terminal, based on the command text, the text code data extracted from image data, and information on the text-code display area corresponding to the text code data. An area of the screen image of the touch panel corresponding to the text-code display area is identified as the subject operation area, and a signal indicative of the subject operation area identified is transmitted to the portable terminal.
Systems and methods for recognizing sound and music signals in high noise and distortion
Tue, 15 Apr 2014 08:00:00 EDT
A method for recognizing an audio sample locates an audio file that closely matches the audio sample from a database indexing a large set of original recordings. Each indexed audio file is represented in the database index by a set of landmark timepoints and associated fingerprints. Landmarks occur at reproducible locations within the file, while fingerprints represent features of the signal at or near the landmark timepoints. To perform recognition, landmarks and fingerprints are computed for the unknown sample and used to retrieve matching fingerprints from the database. For each file containing matching fingerprints, the landmarks are compared with landmarks of the sample at which the same fingerprints were computed. If a large number of corresponding landmarks are linearly related, i.e., if equivalent fingerprints of the sample and retrieved file have the same time evolution, then the file is identified with the sample.
Preserving audio data collection privacy in mobile devices
Tue, 15 Apr 2014 08:00:00 EDT
Techniques are disclosed for using the hardware and/or software of the mobile device to obscure speech in the audio data before a context determination is made by a context awareness application using the audio data. In particular, a subset of a continuous audio stream is captured such that speech (words, phrases and sentences) cannot be reliably reconstructed from the gathered audio. The subset is analyzed for audio characteristics, and a determination can be made regarding the ambient environment.
Audio system and method for coordinating tasks
Tue, 15 Apr 2014 08:00:00 EDT
A system includes a hands free mobile communication device. Software stored on a machine readable storage device is executed to cause the hands free mobile communication device to communicate audibly with a field operator performing field operations. The operator receives instructions regarding operations to be performed. Oral communications are received from the operator and are processed automatically to provide further instructions in response to the received oral communications.
System and method for using semantic and syntactic graphs for utterance classification
Tue, 15 Apr 2014 08:00:00 EDT
Disclosed herein is a system, method and computer readable medium storing instructions related to semantic and syntactic information in a language understanding system. The method embodiment of the invention is a method for classifying utterances during a natural language dialog between a human and a computing device. The method comprises receiving a user utterance; generating a semantic and syntactic graph associated with the received utterance, extracting all n-grams as features from the generated semantic and syntactic graph and classifying the utterance. Classifying the utterance may be performed any number of ways such as using the extracted n-grams, a syntactic and semantic graphs or writing rules.
Unified treatment of data-sparseness and data-overfitting in maximum entropy modeling
Tue, 15 Apr 2014 08:00:00 EDT
A method of statistical modeling is provided which includes constructing a statistical model and incorporating Gaussian priors during feature selection and during parameter optimization for the construction of the statistical model.
System and method for efficient tracking of multiple dialog states with incremental recombination
Tue, 15 Apr 2014 08:00:00 EDT
Disclosed herein are systems, methods, and computer-readable storage media for tracking multiple dialog states. A system practicing the method receives an N-best list of speech recognition candidates, a list of current partitions, and a belief for each of the current partitions. A partition is a group of dialog states. In an outer loop, the system iterates over the N-best list of speech recognition candidates. In an inner loop, the system performs a split, update, and recombination process to generate a fixed number of partitions after each speech recognition candidate in the N-best list. The system recognizes speech based on the N-best list and the fixed number of partitions. The split process can perform all possible splits on all partitions. The update process can compute an estimated new belief. The estimated new belief can be a product of ASR reliability, user likelihood to produce this action, and an original belief.
Adaptive voice print for conversational biometric engine
Tue, 15 Apr 2014 08:00:00 EDT
A computer-implemented method, system and/or program product update voice prints over time. A receiving computer receives an initial voice print. A determining period of time is calculated for that initial voice print. This determining period of time is a length of time during which an expected degree of change in subsequent voice prints, in comparison to the initial voice print and according to a speaker's subsequent age, is predicted to occur. A new voice print is received after the determining period of time has passed, and the new voice print is compared with the initial voice print. In response to a change to the new voice print falling within the expected degree of change in comparison to the initial voice print, a voice print store is updated with the new voice print.
Subspace speech adaptation
Tue, 15 Apr 2014 08:00:00 EDT
Subspace speech adaptation may be utilized for facilitating the recognition of speech containing short utterances. Speech training data may be received in a speech model by a computer. A first matrix may be determined for preconditioning speech statistics based on the speech training data. A second matrix may be determined for representing a basis for the speech to be recognized. A set of basis matrices may then be determined from the first matrix and the second matrix. Speech test data including a short utterance may then be received by the computer. The computer may then apply the set of basis matrices to the speech test data to produce a transcription. The transcription may represent speech recognition of the short utterance.
Systems and methods for hands-free voice control and voice search
Tue, 15 Apr 2014 08:00:00 EDT
In one embodiment the present invention includes a method comprising receiving an acoustic input signal and processing the acoustic input signal with a plurality of acoustic recognition processes configured to recognize the same target sound. Different acoustic recognition processes start processing different segments of the acoustic input signal at different time points in the acoustic input signal. In one embodiment, initial states in the recognition processes may be configured on each time step.
Interface for setting confidence thresholds for automatic speech recognition and call steering applications
Tue, 15 Apr 2014 08:00:00 EDT
An interactive user interface is described for setting confidence score thresholds in a language processing system. There is a display of a first system confidence score curve characterizing system recognition performance associated with a high confidence threshold, a first user control for adjusting the high confidence threshold and an associated visual display highlighting a point on the first system confidence score curve representing the selected high confidence threshold, a display of a second system confidence score curve characterizing system recognition performance associated with a low confidence threshold, and a second user control for adjusting the low confidence threshold and an associated visual display highlighting a point on the second system confidence score curve representing the selected low confidence threshold. The operation of the second user control is constrained to require that the low confidence threshold must be less than or equal to the high confidence threshold.
Speech recognition of character sequences
Tue, 15 Apr 2014 08:00:00 EDT
A method of and a system for processing speech. A spoken utterance of a plurality of characters can be received. A plurality of known character sequences that potentially correspond to the spoken utterance can be selected. Each selected known character sequence can be scored based on, at least in part, a weighting of individual characters that comprise the known character sequence.
Generating speech data collection prompts
Tue, 15 Apr 2014 08:00:00 EDT
This document generally describes computer technologies relating to generating speech data collection prompts, such as textual scripts and/or textual scenarios. Speech data collection prompts for a particular language can be generated based on a variety of factors, including the frequency with which linguistic elements (e.g., phonemes, syllables, words, phrases) in the particular language occur in one or more corpora of textual information associated with the particular language. Textual prompts can also and/or alternatively be generated based on statistics for previously recorded speech data.
Transcription data extraction
Tue, 15 Apr 2014 08:00:00 EDT
A computer program product, for performing data determination from medical record transcriptions, resides on a computer-readable medium and includes computer-readable instructions for causing a computer to obtain a medical transcription of a dictation, the dictation being from medical personnel and concerning a patient, analyze the transcription for an indicating phrase associated with a type of data desired to be determined from the transcription, the type of desired data being relevant to medical records, determine whether data indicated by text disposed proximately to the indicating phrase is of the desired type, and store an indication of the data if the data is of the desired type.
Acoustic model adaptation using splines
Tue, 15 Apr 2014 08:00:00 EDT
Described is a technology by which a speech recognizer is adapted to perform in noisy environments using linear spline interpolation to approximate the nonlinear relationship between clean speech, noise, and noisy speech. Linear spline parameters that minimize the error the between predicted noisy features and actual noisy features are learned from training data, along with variance data that reflect regression errors. Also described is compensating for linear channel distortion and updating noise and channel parameters during speech recognition decoding.
Multi-stage speaker adaptation
Tue, 15 Apr 2014 08:00:00 EDT
A first gender-specific speaker adaptation technique may be selected based on characteristics of a first set of feature vectors that correspond to a first unit of input speech. The first set of feature vectors may be configured for use in automatic speech recognition (ASR) of the first unit of input speech. A second set of feature vectors, which correspond to a second unit of input speech, may be modified based on the first gender-specific speaker adaptation technique. The modified second set of feature vectors may be configured for use in ASR of the second unit of input speech. A first speaker-dependent speaker adaptation technique may be selected based on characteristics of the second set of feature vectors. A third set of feature vectors, which correspond to a third unit of speech, may be modified based on the first speaker-dependent speaker adaptation technique.
Speech-inclusive device interfaces
Tue, 15 Apr 2014 08:00:00 EDT
A user can provide input to a computing device through various combinations of speech, movement, and/or gestures. A computing device can analyze captured audio data and analyze that data to determine any speech information in the audio data. The computing device can simultaneously capture image or video information which can be used to assist in analyzing the audio information. For example, image information is utilized by the device to determine when someone is speaking, and the movement of the person's lips can be analyzed to assist in determining the words that were spoken. Any gestures or other motions can assist in the determination as well. By combining various types of data to determine user input, the accuracy of a process such as speech recognition can be improved, and the need for lengthy application training processes can be avoided.
Low complexity bandwidth expansion of speech
Tue, 15 Apr 2014 08:00:00 EDT
Audio signal bandwidth expansion is performed on a narrow bandwidth signal received from a far end source. The far end source may transmit the signal over the audio communication network. The narrow band signal bandwidth is expanded such that the bandwidth exceeds that of the audio communication network. The signal may be expanded by performing frequency folding on the signal. One or more features are determined for the narrow bandwidth signal, and the expanded signal is modified based on a feature. The feature may be signal band energy slope, narrow band signal energy, or some other feature. The modification may be performed by a shelf filter selected based on the feature. The modified signals are provided for additional processing. In some embodiments, a noise component is added to the narrow band signal prior to folding to create an excitation that reduces the appearance of a fully harmonic signal characteristic.
Operating method for voice activity detection/silence suppression system
Tue, 15 Apr 2014 08:00:00 EDT
A Voice Activity Detection/Silence Suppression (VAD/SS) system is connected to a channel of a transmission pipe. The channel provides a pathway for the transmission of energy. A method for operating a VAD/SS system includes detecting the energy on the channel, and activating or suppressing activation of the VAD/SS system depending upon the nature of the energy detected on the channel.
Systems and methods for model-based processing of linguistic user inputs using annotations
Tue, 15 Apr 2014 08:00:00 EDT
The present invention includes model-based processing of linguistic user inputs. In one embodiment, the present invention includes a computer-implemented method comprising receiving linguistic inputs, parsing the linguistic inputs, mapping the linguistic inputs to a formal representation used by a model, applying the formal representation against the model, where the model comprises said formal representation, and where the model specifies relationships between the elements of the formal representation and defines process information, and accessing software resources based on the formal representation of the user input and the relationships and process information in said model.
Audio transform coding using pitch correction
Tue, 15 Apr 2014 08:00:00 EDT
A processed representation of an audio signal having a sequence of frames is generated by sampling the audio signal within first and second frames of the sequence of frames, the second frame following the first frame, the sampling using information on a pitch contour of the first and second frames to derive a first sampled representation. The audio signal is sampled within the second and third frames, the third frame following the second frame in the sequence of frames. The sampling uses the information on the pitch contour of the second frame and information on a pitch contour of the third frame to derive a second sampled representation. A first scaling window is derived for the first sampled representation, and a second scaling window is derived for the second sampled representation, the scaling windows depending on the samplings applied to derive the first sampled representations or the second sampled representation.
Method and system for efficient transcoding of audio data
Tue, 15 Apr 2014 08:00:00 EDT
Methods and systems for transcoding input audio data in a first encoding format to generate audio data in a second encoding format, and filterbanks for use in such systems. Some such systems include a combined synthesis and analysis filterbank (configured to generate transformed frequency-band coefficients indicative of at least one sample of the input audio data by transforming frequency-band coefficients in a manner equivalent to upsampling the frequency-band coefficients and filtering the resulting up-sampled values to generate the transformed frequency-band coefficients, where the frequency-band coefficients are partially decoded versions of input audio data that are indicative of the at least one sample) and a processing subsystem configured to generate transcoded audio data in the second encoding format in response to the transformed frequency-band coefficients. Some such methods include the steps of: generating frequency-band coefficients indicative of at least one sample of input audio data by partially decoding frequency coefficients of the input audio data; generating transformed frequency-band coefficients indicative of the at least one sample of the input audio data by transforming the frequency-band coefficients in a manner equivalent to upsampling the frequency-band coefficients to generate up-sampled values and filtering the up-sampled values; and in response to the transformed frequency-band coefficients, generating the transcoded audio data so that the transcoded audio data are indicative of each sample of the input audio data.
Information processing apparatus, information processing method, and program
Tue, 15 Apr 2014 08:00:00 EDT
There is provided an information processing apparatus including: an acquiring unit acquiring a title of content; an analyzing unit dividing the title into tokens; a calculating unit calculating, for each token, an evaluation value based on a token length and weighted according to the token's position in the title; a mapping unit mapping, for each token, a token point shown by an ordinal number showing the token's position in the title and the evaluation value, onto a coordinate plane; a deciding unit deciding, based on the mapped token points, coordinates of a criterion point used as a criterion for extracting a series identifier and an extraction criterion based on the criterion point; an extracting unit extracting token points that conform to the extraction criterion out of the token points; and a generating unit generating the series identifier from the character strings included in tokens associated with the extracted token points.

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