Abstract: Recognition of proper nouns by an automated speech recognition system is improved by augmenting the pronunciation of each proper noun or name in the natural language of the speech recognition system with at least one “native” pronunciation in another natural language. To maximize recognition, preferably the pronunciations are predicted based on information not available to the speech recognition system. Prediction of pronunciation may be based on a location derived from a telephone number or postal address associated with the name and the language or dialect spoken in the country or region of that location. The “native” pronunciation(s) may be added to a dictionary of the speech recognition system or directly to the grammar used for recognizing speech.

Abstract: A system, method and computer program product for classification of an analog electrical signal using statistical models of training data. A technique is described to quantize the analog electrical signal in a manner which maximizes the compression of the signal while simultaneously minimizing the diminution in the ability to classify the compressed signal. These goals are achieved by utilizing a quantizer designed to minimize the loss in a power of the log-likelihood ratio. A further technique is described to enhance the quantization process by optimally allocating a number of bits for each dimension of the quantized feature vector subject to a maximum number of bits available across all dimensions.

Abstract: A system and method for identifying audio command prompts for use in a voice response environment is provided. A signature is generated for audio samples each having preceding audio, reference phrase audio, and trailing audio segments. The trailing segment is removed and each of the preceding and reference phrase segments are divided into buffers. The buffers are transformed into discrete fourier transform buffers. One of the discrete fourier transform buffers from the reference phrase segment that is dissimilar to each of the discrete fourier transform buffers from the preceding segment is selected as the signature. Audio command prompts are processed to generate a discrete fourier transform. Each discrete fourier transform for the audio command prompts is compared with each of the signatures and a correlation value is determined. One such audio command prompt matches one such signature when the correlation value for that audio command prompt satisfies a threshold.

Abstract: A training device comprises a first regenerating unit regenerates at least one of an image and a voice for training during the training courses which lead the user to train the operation of an input device, an operation accepting unit accepts the user operation for at least one of the image and the voice for training from a simulated user interface which simulates a user interface of the input device during training, a second regenerating unit regenerates at least one of the image and the voice for training when the training is ended, and a normal operation instructing unit instructs a normal operation to the user by outputting at least one of the image and the voice of the normal operation of the user, which show at least one of the image and the voice for training, which is synchronous with the regeneration of the second regenerating unit.

Abstract: A method, system, and computer program for class detection and time mediated averaging of class dependent models. A technique is described to take advantage of gender information in training data and how obtain female, male, and gender independent models from this information. By using a probability value to average male and female Gaussian Mixture Models (GMMs), dramatic deterioration in cross gender decoding performance is avoided.

Abstract: A method and apparatus for remote access to a target application is disclosed where a system administrator may establish telephonic contact with an interactive voice response system and obtain access to the target application by speech communication. The interactive response system may authenticate the system administrator by implementing various measures including biometric measures. Once access is granted, the interactive response system may broker a communication between the target application using text/data and the system administrator using natural language.

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 an audio signal that corresponds to an utterance recorded by a mobile device, determining a geographic location associated with the mobile device, adapting one or more acoustic models for the geographic location, and performing speech recognition on the audio signal using the one or more acoustic models model that are adapted for the geographic location.

Abstract: Provided herein are systems and methods for using context-sensitive speech recognition logic in a computer to create a software program, including context-aware voice entry of instructions that make up a software program, automatic context-sensitive instruction formatting, and automatic context-sensitive insertion-point positioning.

Abstract: Disclosed herein is a method for emotion recognition based on a minimum classification error. In the method, a speaker's neutral emotion is extracted using a Gaussian mixture model (GMM), other emotions except the neutral emotion are classified using the Gaussian Mixture Model to which a discriminative weight for minimizing the loss function of a classification error for the feature vector for emotion recognition is applied. In the emotion recognition, the emotion recognition is performed by applying a discriminative weight evaluated using the Gaussian Mixture Model based on minimum classification error to feature vectors of the emotion classified with difficult, thereby enhancing the performance of emotion recognition.

Abstract: A handheld electronic device includes a reduced QWERTY keyboard and is enabled with disambiguation software. The device provides output in the form of a default output and a number of variants. The output is based largely upon the frequency, i.e., the likelihood that a user intended a particular output, but various features of the device provide additional variants that are not based solely on frequency and rather are provided by various logic structures resident on the device. The device enables editing during text entry and also provides a learning function that allows the disambiguation function to adapt to provide a customized experience for the user. The disambiguation function can be selectively disabled and an alternate keystroke interpretation system provided.

Abstract: The present invention can include a method of call processing using a distributed voice browser including allocating a plurality of service processors configured to interpret parsed voice markup language data and allocating a plurality of voice markup language parsers configured to retrieve and parse voice markup language data representing a telephony service. The plurality of service processors and the plurality of markup language parsers can be registered with one or more session managers. Accordingly, components of received telephony service requests can be distributed to the voice markup language parsers and the parsed voice markup language data can be distributed to the service processors.

Abstract: A plurality of prompting layers configured to provide varying levels of detailed assistance in prompting a user are maintained. A prompt from a current prompting layer is presented to a user. Input is received from the user. A level of detail in prompting the user is adaptively changed based on user behavior. Upon the user making a hesitant verbal gesture that reaches a threshold duration, a transition is made from the current prompting layer to a more detailed prompting layer. Upon the user interrupting the prompt with a valid input, a transition is made from the current prompting layer to a less detailed prompting layer.

Abstract: A method for real-time speaker recognition including obtaining speech data of a speaker, extracting, using a processor of a computer, a coarse feature of the speaker from the speech data, identifying the speaker as belonging to a pre-determined speaker cluster based on the coarse feature of the speaker, extracting, using the processor of the computer, a plurality of Mel-Frequency Cepstral Coefficients (MFCC) and a plurality of Gaussian Mixture Model (GMM) components from the speech data, determining a biometric signature of the speaker based on the plurality of MFCC and the plurality of GMM components, and determining in real time, using the processor of the computer, an identity of the speaker by comparing the biometric signature of the speaker to one of a plurality of biometric signature libraries associated with the pre-determined speaker cluster.

Abstract: A method and apparatus for updating a speech model on a multi-user speech recognition system with a personal speech model for a single user. A speech recognition system, for instance in a car, can include a generic speech model for comparison with the user speech input. A way of identifying a personal speech model, for instance in a mobile phone, is connected to the system. A mechanism is included for receiving personal speech model components, for instance a BLUETOOTH connection. The generic speech model is updated using the received personal speech model components. Speech recognition can then be performed on user speech using the updated generic speech model.

Abstract: During the replaying of audio data stored in a, which audio data corresponds to text data from a text composed of words, the replaying of the audio data in forward and reverse modes is controlled. Starting from particular momentary replay position in the audio data, a backward jump over a return distance corresponding to the length of about at least two words, to a target position, is automatically initiated for the replaying of the audio data in the reverse mode. Then, starting from the particular target position, a replay of the audio data in the forward sequence for just one part of the return distance is undertaken.

Abstract: A voice recognition apparatus recognizes speaker's voice collected by a microphone, determines whether a telephone number is grouped into categories based on an inclusion of vocabulary in the telephone number that divides the telephone number into groups such as an area code, a city code and a subscriber number, and displays the telephone number in a display part in a grouped form of the area code, city code and subscriber number.

Abstract: Acoustic models to provide features to a speech signal are created based on speech features included in regions where similarities of acoustic models created based on speech features in a certain time length are equal to or greater than a predetermined value. Feature vectors acquired by using the acoustic models of the regions and the speech features to provide features to speech signals of second segments are grouped by speaker.

Abstract: A voice processing apparatus has a storage device that stores registration information containing a characteristic parameter of a given voice. The voice processing apparatus is further provided with a judgment unit, a management unit and a notification unit. The judgment unit judges whether an input voice is appropriate or not for creating or updating the registration information based on a degree of a difference between an inter-band correlation matrix of an input voice acquired this time and an inter-band correlation matrix of another input voice that is judged as being appropriate last time. The management unit creates or updates the registration information based on a characteristic parameter of the input voice when the judgment unit judges that the input voice is appropriate. The notification unit notifies a speaker of the input voice when the judgment unit judges that the input voice is inappropriate.

Abstract: A method and device for verification of an identity of a subscriber of a communication service on a telecommunications network is provided. The communication service requires authentication of the subscriber. The verification includes comparing a reference biometric with at least one biometric characteristic detected from a biometric sample of the subscriber, in order to provide the subscriber with access to the restricted communication service. The reference biometric can be adapted and used for verification purposes based on the different security requirements of the various communication services provided on the telecommunications network.

Abstract: An apparatus and method for recognizing paraphrases of uttered phrases, such as place names. At least one keyword contained in a speech utterance is recognized. Then, the keyword(s) contained in the speech utterance are re-recognized using a phrase including the keyword(s). Based on both recognition results, it is determined whether a paraphrase could have been uttered. If a paraphrase could have been uttered, a phrase corresponding to the paraphrase is determined as a result of speech recognition of the speech utterance.

Abstract: An electronic appliance includes a speaker which outputs a first sound wave based on a first voice signal generated from the electronic appliance, and a microphone to detect a second sound wave on which a sound wave generated for control of the electronic appliance is superimposed to output a second voice signal. A first waveform generator generates a first waveform signal based on the first voice signal, and a second waveform generator generates a second waveform signal based on the second voice signal. A waveform shaping unit outputs a third waveform signal in which the first waveform signal is enlarged in a time axis direction, and a subtracter subtracts the third waveform signal from the second waveform signal.

Abstract: A method of aiding a speech recognition program developer by grouping calls passing through an identified question-answer (QA) state or transition into clusters based on causes of problems associated with the calls is provided. The method includes determining a number of clusters into which a plurality of calls will be grouped. Then, the plurality of calls is at least partially randomly assigned to the different clusters. Model parameters are estimated using clustering information based upon the assignment of the plurality of calls to the different clusters. Individual probabilities are calculated for each of the plurality of calls using the estimated model parameters. The individual probabilities are indicative of a likelihood that the corresponding call belongs to a particular cluster. The plurality of calls is then re-assigned to the different clusters based upon the calculated probabilities. These steps are then repeated until the grouping of the plurality of calls achieves a desired stability.

Abstract: Methods, systems, and computer program products are provided for modifying a grammar of a hierarchical multimodal menu that include monitoring a user invoking a speech command in a first tier grammar, and adding the speech command to a second tier grammar in dependence upon the frequency of the user invoking the speech command. Adding the speech command to a second tier grammar may be carried out by adding the speech command to a higher tier grammar or by adding the speech command to a lower tier grammar. Adding the speech command to a second tier grammar may include storing the speech command in a grammar cache in the second tier grammar.

Abstract: A system and method are described for recognizing repeated audio material within at least one media stream without prior knowledge of the nature of the repeated material. The system and method are able to create a screening database from the media stream or streams. An unknown sample audio fragment is taken from the media stream and compared against the screening database to find if there are matching fragments within the media streams by determining if the unknown sample matches any samples in the screening database.

Abstract: An answering machine detection module is used to determine whether a call recipient is an actual person or an answering machine. The answering machine detection module includes a speech recognizer and a call analysis module. The speech recognizer receives an audible response of the call recipient to a call. The speech recognizer processes the audible response and provides an output indicative of recognized speech. The call analysis module processes the output of the speech recognizer to generate an output indicative of whether the call recipient is a person or an answering machine.

Abstract: Improved systems and methods are described which simplify the individual's interaction with speech recognition software, expand the database of spoken point names that can be recognized, and increase the quality and therefore likelihood of success of speech recognition applications. The present systems and methods apply to various uses, such as providing driving directions, finding the nearest location based service, and finding the nearest “Where Am I?” type of location based services.

Abstract: Exemplary methods, systems, and products are disclosed for synchronizing visual and speech events in a multimodal application, including receiving from a user speech; determining a semantic interpretation of the speech; calling a global application update handler; identifying, by the global application update handler, an additional processing function in dependence upon the semantic interpretation; and executing the additional function. Typical embodiments may include updating a visual element after executing the additional function. Typical embodiments may include updating a voice form after executing the additional function. Typical embodiments also may include updating a state table after updating the voice form. Typical embodiments also may include restarting the voice form after executing the additional function.

Abstract: Embodiments of the invention provide a system for generating an Interactive Voice Response (IVR) database. The system comprises a processor; and a memory coupled to the processor. The memory comprises a list of telephone numbers associated with one or more destinations implementing IVR, wherein the destinations are grouped based on at least one category; instructions executable by the processor for automatically communicating with at least one user; and instructions executable by the processor for at least one personal record from the at least one user and for storing the at least one personal record in the IVR database.

Abstract: A handheld electronic device includes a reduced QWERTY keyboard and is enabled with disambiguation software that is operable to disambiguate compound text input. The device is able to assemble language objects in the memory to generate compound language solutions. The device is able to generate compound language solutions by employing different groupings of data sources to generate different portions of the compound language solutions.

Abstract: One-to-many comparisons of callers' voice prints with known voice prints to identify any matches between them. When a customer communicates with a particular entity, such as a customer service center, the system makes a recording of the real-time call including both the customer's and agent's voices. The system segments the recording to extract at least a portion of the customer's voice to create a customer voice print, and it formats the segmented voice print for network transmission to a server. The server compares the customer's voice print with multiple known voice prints to determine any matches, meaning that the customer's voice print and one of the known voice prints are likely from the same person. The identification of any matches can be used for a variety of purposes, such as determining whether to authorize a transaction requested by the customer.

Abstract: The present invention refers to a method for verifying the identity of a speaker based on the speakers voice comprising the steps of: a) receiving a voice utterance; b) using biometric voice data to verify (10) that the speakers voice corresponds to the speaker the identity of which is to be verified based on the received voice utterance; and c) verifying (12, 13) that the received voice utterance is not falsified, preferably after having verified the speakers voice; d) accepting (16) the speakers identity to be verified in case that both verification steps give a positive result and not accepting (15) the speakers identity to be verified if any of the verification steps give a negative result. The invention further refers to a corresponding computer readable medium and a computer.

Abstract: A system and method for enabling two computer systems to communicate over an audio communications channel, such as a voice telephony connection. Such a system includes a software application that enables a user's computer to call, interrogate, download, and manage a voicemail account stored on a telephone company's computer, without human intervention. A voicemail retrieved from the telephone company's computer can be stored in a digital format on the user's computer. In such a format, the voicemail can be readily archived, or even distributed throughout a network, such as the Internet, in a digital form, such as an email attachment. Preferably a computationally efficient audio recognition algorithm is employed by the user's computer to respond to and navigate the automated audio menu of the telephone company's computer.

Abstract: Method and apparatus for microphone matching for wearable directional hearing assistance devices are provided. An embodiment includes a method for matching at least a first microphone to a second microphone, using a user's voice from the user's mouth. The user's voice is processed as received by at least one microphone to determine a frequency profile associated with voice of the user. Intervals are detected where the user is speaking using the frequency profile. Variations in microphone reception between the first microphone and the second microphone are adaptively canceled during the intervals and when the first microphone and second microphone are in relatively constant spatial position with respect to the user's mouth.

Abstract: Provided are a speech recognition system, a method and a storage medium capable of, even in a case where plural speakers input superimposed speeches, recognizing a speech of an individual each speaker and making a single application program sharable among the speakers in execution. In a speech recognition system receiving speeches of plural speakers to execute a predetermined application program, the received speeches are separated according to the respective speakers if necessary, the received speeches of individual speakers are speech-recognized, results of speech recognition are matched with data items necessary for executing the application program, one of results of recognition of plural speeches which are found as a result of the matching to be overlapping is selected, and the results of recognition of plural speeches which are found as a result of the matching not to be overlapping are linked to the selected result of speech recognition.

Abstract: A method for dynamically modifying an outgoing message language includes receiving a message from a sender. A language associated with the received message is identified and an outgoing message language is automatically set to the identified language associated with the received message.

Abstract: A method for a predictive interactive voice recognition system includes receiving a voice call, associating said voice call with a behavioral pattern, and invoking a service context responsive to said behavioral pattern. The system provides advantages of improved voice recognition and more efficient use of the voice user interface to obtain services.

Abstract: In one embodiment, the present system recognizes a user's speech input using an automatically generated probabilistic context free grammar for street names that maps all pronunciation variations of a street name to a single canonical representation during recognition. A tokenizer expands the representation using position-dependent phonetic tokens and an intersection classifier classifies an intersection, despite the presence of recognition errors and incomplete street names.

Abstract: A system and method for providing enhanced security through the monitoring of communications. In one embodiment, the monitoring process is aided through an automatic speech recognition process that is focused on the recognition of words from a limited vocabulary.

Abstract: A user term information extraction unit extracts term information of a user out of information that has been input by the user to an application for use other than speech recording beforehand, and a speech recognition dictionary management unit expands a vocabulary of a speech recognition dictionary according to the term information of the user. Next, the user inputs speech via a speech input unit, and a speech recognition unit executes speech recognition using the speech recognition dictionary. A representative term information selection unit extracts the term information of the user contained in the speech recognition result, and selects one or a plurality of pieces of representative term information from the term information of the user. A speech file recording unit records the speech data as a speech file, and renders a file name of the speech file according to the representative term information.

Abstract: A handheld electronic device includes a reduced QWERTY keyboard and is enabled with disambiguation software. The device provides output in the form of a default output and a number of variants. The output is based largely upon the frequency, i.e., the likelihood that a user intended a particular output, but various features of the device provide additional variants that are not based solely on frequency and rather are provided by various logic structures resident on the device. The device enables editing during text entry and also provides a learning function that allows the disambiguation function to adapt to provide a customized experience for the user. The disambiguation function can be selectively disabled and an alternate keystroke interpretation system provided.

Abstract: The vocabulary size of a speech recognizer for a large task is reduced by providing a recognizer only for the most common vocabulary items. Uncommon items are catered for by providing aliases from the common items. This allows accuracy to remain high while also allowing uncommon items to be recognized when necessary.

Abstract: Example embodiments provide a speaker authentication technology that compensates for mismatches between enrollment process conditions and test process conditions using correction parameters or correction models, which allow for correcting one of the test voice characterizing parameter set and the enrollment voice characterizing parameter set according to a mismatch between the test process conditions and the enrollment process conditions, thereby obtaining values for the test voice characterizing parameter set and the enrollment voice characterizing parameter set that are based on the same or at least similar process conditions. Alternatively, each of the enrollment and test voice characterizing parameter sets may be normalized to predetermined standard process conditions by using the correction parameters or correction models.

Abstract: Provided herein are systems and methods for using context-sensitive speech recognition logic in a computer to create a software program, including context-aware voice entry of instructions that make up a software program, automatic context-sensitive instruction formatting, and automatic context-sensitive insertion-point positioning.

Abstract: A handheld electronic device includes a reduced QWERTY keyboard and is enabled with disambiguation software that is operable to disambiguate compound text input. The device is able to assemble language objects in the memory to generate compound language solutions. The device is able to prioritize compound language solutions according to various criteria.

Abstract: In detection systems, such as speaker verification systems, for a given operating point range, with an associated detection “cost”, the detection cost is preferably reduced by essentially trading off the system error in the area of interest with areas essentially “outside” that interest. Among the advantages achieved thereby are higher optimization gain and better generalization. From a measurable Detection Error Tradeoff (DET) curve of the given detection system, a criterion is preferably derived, such that its minimization provably leads to detection cost reduction in the area of interest. The criterion allows for selective access to the slope and offset of the DET curve (a line in case of normally distributed detection scores, a curve approximated by mixture of Gaussians in case of other distributions). By modifying the slope of the DET curve, the behavior of the detection system is changed favorably with respect to the given area of interest.

Abstract: A handheld electronic device includes a reduced QWERTY keyboard and is enabled with disambiguation software that is operable to disambiguate compound text input. The device is able to assemble language objects in the memory to generate compound language solutions. The device is able to prioritize compound language solutions according to various criteria, including the degree of completeness of the text components of a compound language solution.

Abstract: Exemplary methods, systems, and products are disclosed for synchronizing visual and speech events in a multimodal application, including receiving from a user speech; determining a semantic interpretation of the speech; calling a global application update handler; identifying, by the global application update handler, an additional processing function in dependence upon the semantic interpretation; and executing the additional function. Typical embodiments may include updating a visual element after executing the additional function. Typical embodiments may include updating a voice form after executing the additional function. Typical embodiments also may include updating a state table after updating the voice form. Typical embodiments also may include restarting the voice form after executing the additional function.

Abstract: A system comprises a computer system comprising a central processing unit coupled to a memory and resource management application. A plurality of different automatic speech recognition (ASR) engines is coupled to the computer system. The computer system is adapted to select ASR engines to analyze a speech utterance based on resources available on the system.

Abstract: An apparatus with a speech control unit includes a microphone array having multiple microphones for receiving respective audio signals, and a beam forming module for extracting a speech signal of a user, from the audio signals. A keyword recognition system recognizes a predetermined keyword that is spoken by the user and which is represented by a particular audio signal and is arranged to control the beam forming module, on basis of tie recognition. A speech recognition unit creates an instruction for the apparatus based on recognized speech items of the speech signal. As a consequence, the speech control unit is more selective for those parts of the audio signals for speech recognition which correspond to speech items spoken by the user.

Abstract: A handheld electronic device includes a reduced QWERTY keyboard and is enabled with disambiguation software that is operable to disambiguate compound text input. The device is able to assemble language objects in the memory to generate compound language solutions. The device is able to generate compound language solutions by employing different groupings of data sources to generate different portions of the compound language solutions.