Abstract: A method and system for addressing disambiguation issues in interactive applications by creating a disambiguation system for generating complex grammars that includes homonym detection and grouping, and provides optimization feedback that eliminates time-consuming and repetitive iterative steps during the grammar generation portion of the interactive application configuration.

Abstract: In this invention, a voice recognition engine 110 outputs to a controlling section 103 a matching state of a voice input signal as an error code. Then, the controlling section 103 determines the matching state based on the error code in the error determination section 105 and outputs to a voice synthesizing engine 113 guidance data according to the matching state based on a timing control by a guidance timing controlling section 107. According to such a configuration, this invention improves operatability by voice operation, while reducing a risk of erroneous recognition by maintaining a predetermined matching rate.

Abstract: Methods and systems for sample rate conversion convert a sampled signal to a higher data rate signal. Conversion pulses are received, having a conversion rate that is higher than the sample rate of the sampled signal. Sample points are then reconstructed from the sampled signal, in real time, on either side of a conversion pulse. An interpolation is performed between the reconstructed sample points, at the time of the conversion pulse. The interpolation results are outputted in real time. The process is repeated for additional conversion pulses. The outputted interpolated amplitudes form the higher data rate signal having a data rate equal to the conversion rate. Sample rate conversion is thus performed in real time according to the higher data rate clock, rather than with fixed ratios. As a result, when the higher data rate clock is affected by, for example, jitter or other frequency variations, the higher data rate samples immediately track the lower data rate samples.

Abstract: The disclosed solution includes a method for dynamically switching modalities based upon inferred conditions in a dialogue session involving a speech application. The method establishes a dialogue session between a user and the speech application. During the dialogue session, the user interacts using an original modality and a second modality. The speech application interacts using a speech modality only. A set of conditions indicative of interaction problems using the original modality can be inferred. Responsive to the inferring step, the original modality can be changed to the second modality. A modality transition to the second modality can be transparent the speech application and can occur without interrupting the dialogue session. The original modality and the second modality can be different modalities; one including a text exchange modality and another including a speech modality.

Abstract: A similarity degree estimation method is performed by two processes. In a first process, an inter-band correlation matrix is created from spectral data of an input voice such that the spectral data are divided into a plurality of discrete bands which are separated from each other with spaces therebetween along a frequency axis, a plurality of envelope components of the spectral data are obtained from the plurality of the discrete bands, and elements of the inter-band correlation matrix are correlation values between the respective envelope components of the input voice. In a second process, a degree of similarity is calculated between a pair of input voices to be compared with each other by using respective inter-band correlation matrices obtained for the pair of the input voices through the inter-band correlation matrix creation process.

Abstract: A noise-canceling device of a voice communication terminal that removes noise elements included in received voice signals. The device comprises: a digital filter array that exhibits filter qualities in response to a coefficient setting signal showing each supplied arrays of filter coefficients, and includes a first-stage filter that receives the received voice signals as well as multiple later-stage filters connected thereto in a straight line; a filter qualities designator that generates input designation that designates each qualities of the multiple digital filters forming the digital filters array; and a filter coefficient setter that retains multiple arrays of filter coefficients, extracts a filter coefficient array corresponding to the designation input from among the multiple filter coefficient arrays, and supplies to each multiple digital filters.

Abstract: There is disclosed a method and system for preparing a document to be read by a text-to-speech reader. The method can include identifying two or more voice types available to the text-to-speech reader, identifying the text elements within the document, grouping related text elements together, and classifying the text elements according to voice types available to the text-to-speech reader. The method of grouping the related text elements together can include syntactic and intelligent clustering. The classification of text elements can include performing latent semantic analysis on the text elements and characteristics of the available voice types.

Abstract: A system and method is provided for easily detecting a fraudulent user who attempts to obtain authentication using voice reproduced by a reproducer. A personal computer is provided with an audio data obtaining portion for picking up ambient sound around a person as a target of user authentication using voice authentication technology during a period before the person utters, and a fraud determination portion for calculating an intensity level showing intensity of the picked-up ambient sound per predetermined time for each of sections into which the period is divided and for determining that the person is a fraudulent user who attempts to obtain authentication using reproduced voice, when, of two of the calculated intensity levels, the intensity level of the later section is larger than a sum of the intensity level of the earlier section and a predetermined value.

Abstract: A speech processing apparatus includes a rule storing unit that stores therein a rule that correlates one another causes of errors in speech recognition, responding methods each of which is used when an error has occurred during the speech recognition, and responding users each of whom is one of a plurality of users and serving as a target of a response; a detecting unit that detects a cause of an error that has occurred during the recognition of the speech; a method selecting unit that selects one of the responding methods that is correlated with the detected cause of the error from the rule storing unit; a user selecting unit that selects one of the responding users that is correlated with the detected cause of the error from the rule storing unit; and an executing unit that executes the response by the selected responding method to the selected responding user.

Abstract: A spoken language understanding method and system are provided. The method includes classifying a set of labeled candidate utterances based on a previously trained classifier, generating classification types for each candidate utterance, receiving confidence scores for the classification types from the trained classifier, sorting the classified utterances based on an analysis of the confidence score of each candidate utterance compared to a respective label of the candidate utterance, and rechecking candidate utterances according to the analysis. The system includes modules configured to control a processor in the system to perform the steps of the method.

Abstract: Generic and specific C-to-E binaural cue coding (BCC) schemes are described, including those in which one or more of the input channels are transmitted as unmodified channels that are not downmixed at the BCC encoder and not upmixed at the BCC decoder. The specific BCC schemes described include 5-to-2, 6-to-5, 7-to-5, 6.1-to-5.1, 7.1-to-5.1, and 6.2-to-5.1, where “0.1” indicates a single low-frequency effects (LFE) channel and “0.2” indicates two LFE channels.

Abstract: A machine-implemented method of building a speech application includes generating a graphical user interface to enable a user to create and edit a speech application, and receiving user inputs directed to the graphical user interface, where the user inputs specify a set of dialog flows representing the speech application. The method further includes, based on the user inputs, automatically generating executable code representing the speech application and a functional specification document describing the speech application.

Abstract: Portions from time-domain speech segments are extracted. Feature vectors that represent the portions in a vector space are created. The feature vectors incorporate phase information of the portions. A distance between the feature vectors in the vector space is determined. In one aspect, the feature vectors are created by constructing a matrix W from the portions and decomposing the matrix W. In one aspect, decomposing the matrix W comprises extracting global boundary-centric features from the portions. In one aspect, the portions include at least one pitch period. In another aspect, the portions include centered pitch periods.

Abstract: The invention is generally directed to systems and methods for medical care, and more particularly to systems and methods for voice control of a medical device. A first embodiment includes a voice controlled surgical system, such as a phacoemulsification system, a microphone coupled to the surgical system, and a voice controlled computer interface coupled with the surgical system. The voice controlled interface is configured to receive a request to invoke a voice command via the microphone, to listen for a voice command upon receipt of a valid request to invoke a voice command, and to forward a valid voice command upon receipt of the valid voice command to the surgical system for execution.

Abstract: A method and device for updating statuses of synthesis filters are provided. The method includes: exciting a synthesis filter corresponding to a first encoding rate by using an excitation signal of the first encoding rate, outputting reconstructed signal information, and updating status information of the synthesis filter and a synthesis filter corresponding to a second encoding rate. In the present disclosure, the status of the synthesis filter corresponding to the current rate and the statuses of the synthesis filters at other rates are updated. Thus, synchronization between the statuses of the synthesis filters corresponding to different rates at the encoding terminal may be realized, thereby facilitating the consistency of the reconstructed signals of the encoding and decoding terminals when the encoding rate is switched, and improving the quality of the reconstructed signal of the decoding terminal.

Abstract: A weighted search program is disclosed. The weighted search program may be integrated into a translation program, or the weighted search program may be used independently with an available search engine. When integrated with the translation program, setting and weighting may be combined in a single search. In one embodiment, the weighting would be used in conjunction with a Pin Yin translation program so that a user could set some terms, and allocate a search weight to the remaining terms. The invention may be applied independently in Internet searching so that a user can apply weights to multiple elements of a search term.

Abstract: A tool, method, and system for use in the development of sentence-based test items are disclosed. The tool may include a user interface that may include a database selection field, a sentence pattern entry field, an option pane, and an output pane. The tool may search a database for one or more sentences and may generate one or more responses to the one or more sentences. The one or more sentences and one or more responses may be used to produce the sentence-based test items. The tool may allow test items to be developed more quickly and easily than manual test item authoring. Accordingly, test item development costs may be lowered and test security may be enhanced.

Abstract: A method for encoding sound signals on multiple channels includes extracting an arbitrary number of sine waves from each of the sound signals. The sine waves include at least a first sine wave, extracted from a first one of the channels and having first-channel information, and a second sine wave, extracted from a second one of the channels and having second-channel information. Using the first-channel information and one of the second-channel information and sine wave information corresponding to a predetermined sine wave, one of the second-channel information and the sine wave information corresponding to the predetermined sine wave is selected as a to-be-correlated object for encoding in a correlation with the first-channel information.

Abstract: Audio comparison using phoneme matching is described, including evaluating audio data associated with a file, identifying a sequence of phonemes in the audio data, associating the file with a product category based on a match indicating the sequence of phonemes is substantially similar to another sequence of phonemes, the file being stored, and accessing the file when a request associated with the product category is detected.

Abstract: A combination of a book and a voice phonation apparatus comprising a book having a plurality of pages, at least one of which is carrying a plurality of printed words and a plurality of specific codes associated with said words, said printed words being divided into a plurality of specific segments; and a voice phonation apparatus forming an integral part of said book and comprising (i) a housing with a plurality of switches and a plurality of keys, (ii) a voice output unit, and (iii) a control unit having a memory for storing data representing spoken words and connected to said switches, said keys, and said voice output unit.