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Patents

 
Voice-controlled navigation device utilizing wireless data transmission for obtaining maps and real-time overlay information
Tue, 25 Nov 2014 08:00:00 EST
A navigation system and method involving wireless communications technology and speech processing technology is presented. In accordance with an embodiment of the invention, the navigation system includes a subscriber unit communicating with a service provider. The subscriber unit includes a global positioning system mechanism to determine subscriber position information and a speech processing mechanism to receive destination information spoken by a subscriber. The subscriber unit transmits the subscriber position and destination information to the service provider, which gathers navigation information, including a map and a route from the subscriber position to the specified destination. The service provider transmits the navigation information to the subscriber unit. The subscriber unit conveys the received navigation information to the subscriber via an output mechanism, such as a speech synthesis unit or a graphical display.
Audio user interface
Tue, 25 Nov 2014 08:00:00 EST
An audio user interface that provides audio prompts that help a user interact with a user interface of an electronic device is disclosed. The audio prompts can provide audio indicators that allow a user to focus his or her visual attention upon other tasks such as driving an automobile, exercising, or crossing a street, yet still enable the user to interact with the user interface. An intelligent path can provide access to different types of audio prompts from a variety of different sources. The different types of audio prompts may be presented based on availability of a particular type of audio prompt. As examples, the audio prompts may include pre-recorded voice audio, such as celebrity voices or cartoon characters, obtained from a dedicate voice server. Absent availability of pre-recorded or synthesized audio data, non-voice audio prompts may be provided.
Systems and methods for generating markup-language based expressions from multi-modal and unimodal inputs
Tue, 25 Nov 2014 08:00:00 EST
When using finite-state devices to perform various functions, it is beneficial to use finite state devices representing regular grammars with terminals having markup-language-based semantics. By using markup-language-based symbols in the finite state devices, it is possible to generate valid markup-language expressions by concatenating the symbols representing the result of the performed function. The markup-language expression can be used by other applications and/or devices. Finite-state devices are used to convert strings of words and gestures into valid markup-language, for example, XML, expressions that can be used, for example, to provide an application program interface to underlying system applications.
Method, artificially intelligent system and networked complex for facilitating group interactions
Tue, 25 Nov 2014 08:00:00 EST
An artificially intelligent or rule-based system to assist teams or groups become more effective by improving the communication process between members of the team or group. The system helps members share information, negotiate more effectively and make better group decisions. The system is designed to allow users to provide feedback to the system regarding undetected emotional feelings of any one user to all users of the system.
Audio encoder, audio decoder, method for encoding and audio information, method for decoding an audio information and computer program using a modification of a number representation of a numeric previous context value
Tue, 25 Nov 2014 08:00:00 EST
An audio decoder includes an arithmetic decoder for providing decoded spectral values on the basis of an arithmetically-encoded representation of the spectral values and a frequency-domain-to-time-domain converter for providing a time-domain audio representation using the decoded spectral values. The arithmetic decoder selects a mapping rule describing a mapping of a code value onto a symbol code in dependence on a context state described by a numeric current context value, and determines the numeric current context value in dependence on a plurality of previously-decoded spectral values. The arithmetic decoder modifies a number representation of a numeric previous context value, describing a context state associated with one or more previously decoded spectral values, in dependence on a context subregion value, to acquire a number representation of a numeric current context value describing a context state associated with one or more spectral values to be decoded. An audio encoder uses a similar concept.
Subband block based harmonic transposition
Tue, 25 Nov 2014 08:00:00 EST
The present document relates to audio source coding systems which make use of a harmonic transposition method for high frequency reconstruction (HFR), as well as to digital effect processors, e.g. exciters, where generation of harmonic distortion add brightness to the processed signal, and to time stretchers where a signal duration is prolonged with maintained spectral content. A system and method configured to generate a time stretched and/or frequency transposed signal from an input signal is described. The system comprises an analysis filterbank (101) configured to provide an analysis subband signal from the input signal; wherein the analysis subband signal comprises a plurality of complex valued analysis samples, each having a phase and a magnitude. Furthermore, the system comprises a subband processing unit (102) configured to determine a synthesis subband signal from the analysis subband signal using a subband transposition factor Q and a subband stretch factor 5″. The subband processing unit (102) performs a block based nonlinear processing wherein the magnitude of samples of the synthesis subband signal are determined from the magnitude of corresponding samples of the analysis subband signal and a predetermined sample of the analysis subband signal. In addition, the system comprises a synthesis filterbank (103) configured to generate the time stretched and/or frequency transposed signal from the synthesis subband signal.
Multi-lingual text-to-speech system and method
Tue, 25 Nov 2014 08:00:00 EST
A multi-lingual text-to-speech system and method processes a text to be synthesized via an acoustic-prosodic model selection module and an acoustic-prosodic model mergence module, and obtains a phonetic unit transformation table. In an online phase, the acoustic-prosodic model selection module, according to the text and a phonetic unit transcription corresponding to the text, uses at least a set controllable accent weighting parameter to select a transformation combination and find a second and a first acoustic-prosodic models. The acoustic-prosodic model mergence module merges the two acoustic-prosodic models into a merged acoustic-prosodic model, according to the at least a controllable accent weighting parameter, processes all transformations in the transformation combination and generates a merged acoustic-prosodic model sequence. A speech synthesizer and the merged acoustic-prosodic model sequence are further applied to synthesize the text into an L1-accent L2 speech.
Configurable speech recognition system using multiple recognizers
Tue, 25 Nov 2014 08:00:00 EST
Techniques for combining the results of multiple recognizers in a distributed speech recognition architecture. Speech data input to a client device is encoded and processed both locally and remotely by different recognizers configured to be proficient at different speech recognition tasks. The client/server architecture is configurable to enable network providers to specify a policy directed to a trade-off between reducing recognition latency perceived by a user and usage of network resources. The results of the local and remote speech recognition engines are combined based, at least in part, on logic stored by one or more components of the client/server architecture.
Identifying candidate passwords from captured audio
Tue, 25 Nov 2014 08:00:00 EST
A computing device configured to request a password from a user, capture audio after issuing the request, and determine a number of alternative candidate passwords most likely represented by the audio. After identifying the number of candidate passwords, the computing device may submit these candidate passwords, one at a time, to an entity until the entity grants the device access to an account associated with the user or until the device has submitted each candidate password. The account may comprise a network account (e.g., a wired or wireless network account), an online account (e.g., an email account, an account an online merchant, etc.), or the like.
Method for converting speech to text, performing natural language processing on the text output, extracting data values and matching to an electronic ticket form
Tue, 25 Nov 2014 08:00:00 EST
A system and method for extracting data values from a conversation to complete an electronic trade ticket over a communications network. The system comprises a plurality of client devices and a processor based server. A digital switchboard routes an incoming call from a first user to a second user to the server over the communications network. The telephone numbers of the users are verified against the stored telephone numbers in a database. A sample representing a predetermined period of the recorded conversation are utilized to identify the users. The conversation between the users are transcribed onto an electronic text file and processed to extract terms to populate data fields of an electronic trade ticket.
Strained-rough-voice conversion device, voice conversion device, voice synthesis device, voice conversion method, voice synthesis method, and program
Tue, 25 Nov 2014 08:00:00 EST
A strained-rough-voice conversion unit (10) is included in a voice conversion device that can generate a “strained rough” voice produced in a part of a speech when speaking forcefully with excitement, nervousness, anger, or emphasis and thereby richly express vocal expression such as anger, excitement, or an animated or lively way of speaking, using voice quality change. The strained-rough-voice conversion unit (10) includes: a strained phoneme position designation unit (11) designating a phoneme to be uttered as a “strained rough” voice in a speech; and an amplitude modulation unit (14) performing modulation including periodic amplitude fluctuation on a speech waveform. The amplitude modulation unit (14) generates, according to the designation of the strained phoneme position designation unit (11), the “strained rough” voice by performing the modulation including periodic amplitude fluctuation on the part to be uttered as the “strained rough” voice, in order to generate a speech having realistic and rich expression uttering forcefully with excitement, nervousness, anger, or emphasis.
Distributed user input to text generated by a speech to text transcription service
Tue, 25 Nov 2014 08:00:00 EST
A particular method includes receiving, at a representational state transfer endpoint device, a first user input related to a first speech to text conversion performed by a speech to text transcription service. The method also includes receiving, at the representational state transfer endpoint device, a second user input related to a second speech to text conversion performed by the speech to text transcription service. The method includes processing of the first user input and the second user input at the representational state transfer endpoint device to generate speech to text adjustment information.
Source code adaption based on communication link quality and source coding delay
Tue, 25 Nov 2014 08:00:00 EST
Method and arrangement in a network node for adapting a property of source coding to the quality of a communication link in packet switched conversational services in a communication system. The method comprises obtaining (404) information related to the quality of a communication link. The method further comprises selecting (406) a source coding mode with an associated source coding delay, based on the obtained information and the associated source coding delay. The selected source coding mode is selected from a set of at least two source coding modes associated with different source coding delays, and is to be used when source coding voice data to be transmitted over the communication link.
LPC residual signal encoding/decoding apparatus of modified discrete cosine transform (MDCT)-based unified voice/audio encoding device
Tue, 25 Nov 2014 08:00:00 EST
Disclosed is an LPC residual signal encoding/decoding apparatus of an MDCT based unified voice and audio encoding device. The LPC residual signal encoding apparatus analyzes a property of an input signal, selects an encoding method of an LPC filtered signal, and encode the LPC residual signal based on one of a real filterbank, a complex filterbank, and an algebraic code excited linear prediction (ACELP).
Systems, methods, and apparatus for voice activity detection
Tue, 25 Nov 2014 08:00:00 EST
Systems, methods, apparatus, and machine-readable media for voice activity detection in a single-channel or multichannel audio signal are disclosed.
Encoding apparatus, decoding apparatus and methods thereof
Tue, 25 Nov 2014 08:00:00 EST
Disclosed is an encoding apparatus that can efficiently encode a signal that is a broad or extra-broad band signal or the like, thereby improving the quality of a decoded signal. This encoding apparatus includes a band establishing unit (301) that generate, based on the characteristic of the input signal, band establishment information to be used for dividing the band of the input signal to establish a first band part of lower frequency side and a second band part of higher frequency side; a lower frequency encoding unit (302) for encoding, based on the band establishment information, the input signal of the first band part to generate encoded lower frequency part information; and a higher frequency encoding unit (303) for encoding, based on the band establishment information, the input signal of the second band part to generate encoded higher frequency part information.
System and method for generating a separated signal by reordering frequency components
Tue, 25 Nov 2014 08:00:00 EST
The present invention relates to blind source separation. More specifically certain embodiments relate to the blind source separation using frequency domain processes. Aspects of the invention relate to methods and systems for receiving a set of frequency-domain first signals, and then separating the set of frequency-domain first signals into a set of frequency-domain second signals. The frequency-domain second signals may have a set of separated frequency-domain second signal elements corresponding to individual frequencies wherein each frequency-domain second signal element is assigned an identifier. The identifier may indicate which of the set of frequency-domain second signals includes the frequency-domain second signal element. Some aspects also include reordering the identifiers corresponding to at least one frequency to improve coherence of the frequency-domain second signals and to produce a set of frequency-domain third signals.
Voice quality conversion device and voice quality conversion method for converting voice quality of an input speech using target vocal tract information and received vocal tract information corresponding to the input speech
Tue, 25 Nov 2014 08:00:00 EST
A voice quality conversion device including: a target vowel vocal tract information hold unit holding target vowel vocal tract information of each vowel indicating target voice quality; a vowel conversion unit (i) receiving vocal tract information with phoneme boundary information of the speech including information of phonemes and phoneme durations, (ii) approximating a temporal change of vocal tract information of a vowel in the vocal tract information with phoneme boundary information applying a first function, (iii) approximating a temporal change of vocal tract information of the same vowel held in the target vowel vocal tract information hold unit applying a second function, (iv) calculating a third function by combining the first function with the second function, and (v) converting the vocal tract information of the vowel applying the third function; and a synthesis unit synthesizing a speech using the converted information.
Determining and conveying contextual information for real time text
Tue, 25 Nov 2014 08:00:00 EST
Aspects relate to machine recognition of human voices in live or recorded audio content, and delivering text derived from such live or recorded content as real time text, with contextual information derived from characteristics of the audio. For example, volume information can be encoded as larger and smaller font sizes. Speaker changes can be detected and indicated through text additions, or color changes to the font. A variety of other context information can be detected and encoded in graphical rendition commands available through RTT, or by extending the information provided with RTT packets, and processing that extended information accordingly for modifying the display of the RTT text content.
Encoding device, decoding device, and methods therein
Tue, 25 Nov 2014 08:00:00 EST
An encoding device, a decoding device, and related methods are provided that eliminate the loss of synchronization of the adaptive filters of a terminal at the encoding end and a terminal at the decoding end caused by transmission errors. Deterioration of the sound quality is suppressed when a multiple channel signal is encoded with high efficiency using an adaptive filter. In the terminal at the encoding end, a buffer stores updated filter coefficients. When packet loss detection information indicating whether there is any packet loss in the terminal at the decoding end indicates that there is packet loss, a switch outputs the past filter coefficients of the previous (NX+1) frames from the buffer to an adaptive filter. The adaptive filter uses the past filter coefficients of the previous (NX+1) frames to conduct filtering.
Systems and methods for training statistical speech translation systems from speech utilizing a universal speech recognizer
Tue, 25 Nov 2014 08:00:00 EST
An iterative language translation system. The system includes a first automatic speech recognition component adapted to recognize spoken language in a source language and to create a source language hypothesis and a first machine translation component adapted to translate the source language hypothesis into a target language. The system also includes a second universal automatic speech recognition component adapted to recognize spoken languages in plurality of target languages spoken by a translator, and wherein the second automatic speech recognition component is further adapted to create a target language hypothesis. The system further includes a second machine translation component adapted to translate the target language hypothesis into the source language, wherein the translation of the target language hypothesis into the source language is used to adapt the first automatic speech recognition component, wherein the translation of the source language hypothesis into the target language is used to adapt the second automatic speech recognition component, wherein the source language hypothesis is used to adapt the first machine translation component and the second machine translation component, and wherein the target language hypothesis is used to adapt the first machine translation component and the second machine translation component.
System and method of adjusting the sound of multiple audio objects directed toward an audio output device
Tue, 25 Nov 2014 08:00:00 EST
Embodiments of the present invention include methods and apparatuses for adjusting audio content when more multiple audio objects are directed toward a single audio output device. The amplitude, white noise content, and frequencies can be adjusted to enhance overall sound quality or make content of certain audio objects more intelligible. Audio objects are classified by a class category, by which they are can be assigned class specific processing. Audio objects classes can also have a rank. The rank of an audio objects class is used to give priority to or apply specific processing to audio objects in the presence of other audio objects of different classes.
Denoising an audio signal using local formant information
Tue, 25 Nov 2014 08:00:00 EST
A system, method, and computer program product are provided for cleaning an audio segment. For a given audio segment, an offset amount is calculated where the audio segment is maximally correlated to the audio segment as offset by the offset amount. The audio segment and the audio segment as offset by the offset amount are averaged to produce a cleaned audio segment, which has had noise features reduced while having signal features (such as voiced audio) enhanced.
Method and apparatus for estimating spectrum density of diffused noise
Tue, 25 Nov 2014 08:00:00 EST
Provided are a method for estimating a spectrum density of diffused noises. Also provided is a processor for implementing the method. The processor includes at least two sound receiving units and a spectrum density estimating unit for estimating spectrum density.
Reconfigurable orthogonal frequency division multiplexing (OFDM) chip supporting single weight diversity
Tue, 25 Nov 2014 08:00:00 EST
A method and system for a reconfigurable orthogonal frequency division multiplexing (OFDM) chip supporting single weight diversity are provided. The reconfigurable OFDM chip may be configured to process signals such as IEEE 802.11, 802.16, and digital video broadcasting (DVB). The OFDM chip may generate channel weights to be applied to signals received in receive antennas. The weighted signals may be combined into a single received signal and channel estimates may be generated from the single received signal. Updated channel weights may be generated from the generated channel estimates. Updates to the channel weights may be performed dynamically. The configurable OFDM chip may be utilized to provide collaborative cellular and OFDM-based communication. The reconfigurable OFDM chip and the cellular chip may communicate data and/or control information via a memory coupled to a common bus.
Information processing device and mobile terminal
Tue, 25 Nov 2014 08:00:00 EST
There is a need to enable decompression of a speech signal even if no network synchronizing signal is output from a baseband processing portion. For this purpose, an information processing device includes a first serial interface. The first serial interface includes a notification signal generation circuit that generates a notification signal each time compressed data incorporated from the baseband processing portion reaches a predetermined data quantity, and notifies a speech processing portion of this state using the notification signal. The speech processing portion includes a synchronizing signal generation circuit that generates a network synchronizing signal based on the notification signal. A clock signal for PCM communication is generated based on the network synchronizing signal. A speech signal can be decompressed even if no network synchronizing signal is output from the baseband processing portion.
Systematic framework for application protocol field extraction
Tue, 25 Nov 2014 08:00:00 EST
A computer-implemented system is provided for implementing application protocol field extraction. The system includes: an automata generator configured to receive the extraction specification that specifies data elements to be extracted from data packets and generate a counting automaton; and a field extractor configured to receive a data flow and operates to extract data elements from the data packets in accordance with the counting automaton. The extraction specification is expressed in terms of a context-free grammar, where the grammar defines grammatical structures of data packets transmitted in accordance with an application protocol and includes counters used to chronicle parsing history of production rules comprising the grammar.

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