Abstract: A context-free grammar can be represented by a weighted finite-state transducer. This representation can be used to efficiently compile that grammar into a weighted finite-state automaton that accepts the strings allowed by the grammar with the corresponding weights. The rules of a context-free grammar are input. A finite-state automaton is generated from the input rules. Strongly connected components of the finite-state automaton are identified. An automaton is generated for each strongly connected component. A topology that defines a number of states, and that uses active ones of the non-terminal symbols of the context-free grammar as the labels between those states, is defined. The topology is expanded by replacing a transition, and its beginning and end states, with the automaton that includes, as a state, the symbol used as the label on that transition. The topology can be fully expanded or dynamically expanded as required to recognize a particular input string.

Abstract: The invention disclose: Chinese spelling scheme, Chinese alphabetic writing and Phonetic Symbols scheme. The Chinese spelling scheme which is used English letters to mark the tones, it is reduced to 5 sound symbols of the monosyllabic words from 6 spelling letters of other schemes at most. Between the syllables is clearly demarcated. It can be used as marks in sound and tone for Chinese characters, and also can form an alphabetic writing independently which follows the law of international languages. It's highly integrated with word, sound, and code which can be displayed each other and converted each other. It can be translated directly with the Chinese-language sentence and foreign languages. The Phonetic Symbols do not go beyond the range of 26 letters and symbols of common used English keyboard, which can not only be used to mark in English and other languages, but also can be used as a phonetic symbol written language independently.

Abstract: A method of modifying the operation of the encoder function and/or the decoder function of a perceptual coding system in accordance with supplemental information, such as a watermark, so that the supplemental information may be detectable in the output of the decoder function. One or more parameters are modulated in the encoder function and/or the decoder function in response to the supplemental information.

Abstract: A system and method for lossless and/or progressive to lossless data coding (e.g., audio and/or image) is provided. The system and method employ a multiple factorization reversible transform component that provides quantized coefficients based, at least in part, upon a multiple factorization reversible transform. The multiple factorization reversible transform component can employ an N-point modulated lapped transform in accordance with one aspect of the present invention. The multiple factorization reversible transform component can comprise a modulation stage, a pre-FFT rotation stage, a complex FFT stage and a post-FFT rotation stage.

Abstract: A novel method and microprocessor-controlled apparatus are provided for improving research and/or communication interaction between human and animals, such as dolphins and the like, or between normal humans and speech-impaired humans, by playing back audible pre-recorded human-language phrases corresponding to and descriptive of the meaning of sounds and/or gestures or responses produced by the animal or the speech-impaired human and in response to stimuli, conditions or environmental events to which the animal (or impaired human) is subjected, wherein the pre-recorded phrases represent what the human would say or describe if subjected to such stimuli, conditions or environmental events; such that during actual real-time interacting with the animals (or such impaired humans) one actually hears “from” the animal or speech-impaired human, spoken language phrases descriptive of their condition.

Abstract: A method and device for performing some preprocessing on voice transmissions depending upon the intended destination of the transmission. The device includes a receiving component configured to receive a voice signal from a source over a network. The device also includes a processing component configured to determine a destination address associated with the received signal, determine a signal processing algorithm from a plurality of signal processing algorithms based on the determined address, and process the voice signal according to the specified algorithm. The device further includes a delivery component configured to send the processed signal to the associated address.

Abstract: A method and system for automatically generating a scoring model for scoring a speech sample are disclosed. One or more training speech samples are received in response to a prompt. One or more speech features are determined for each of the training speech samples. A scoring model is then generated based on the speech features. At least one of the training speech samples may be a high entropy speech sample. An evaluation speech sample is received and a score is assigned to the evaluation speech sample using the scoring model. The evaluation speech sample may be a high entropy speech sample.

Abstract: A speech recognition system for processing voice inputs from a user to select a list element from a list or group of list elements. Recognition procedures are carried out on the voice input of the user. One recognition procedure separates the voice input of a whole word into at least one sequence of speech subunits to produce a vocabulary of list elements. Another recognition procedure compares the voice input of the whole word with the vocabulary of list elements.

Abstract: Primary and alternate optimization procedures are used to improve the ITU-T G.723.1 speech coding standard (the “Standard”) by replacing the Hamming window of the Standard with an optimized window, with two windows, or with two windows and an additional performance of an autocorrelation method. When two windows replace the Hamming window, at least one of which is an optimized window, generally the first is used to determine optimized unquantized LP coefficients which are used to define an optimized perceptual weighting filter, and the second is used to determine optimized unquantized LP coefficients which are used to determine optimized synthesis coefficients. Optimized windows created using the primary and alternate optimization procedures and used in the Standard yield improvements in the objective and subjective quality of synthesized speech produced by the Standard. The improved Standard, methods, and window can all be implemented as computer readable software code.

Abstract: A method, apparatus, and computer instructions for testing software programs running on a data processing system. Text is translated from the source language to the target language to form translated text in response to a user input, containing the text in a source language. The text is entered through a computer interface in the data processing system. The translated text is inserted into a user interface of the software program to be tested to form inserted, translated text. The software program is written using the target language. A determination is made as to whether the software program functions correctly using the inserted, translated text.

Abstract: Parallelization of word alignment for a text-to-text operation. The training data is divided into multiple groups, and training is carried out of each group on separate processors. Different techniques can be carried out to increase the speed of the processing. The hookups can be done only once for all of multiple different iterations. Moreover, parallel operations can apply only to the counts, since this may be the most time-consuming part.

Abstract: A system and method for classifying a voice signal to one of a set of predefined categories, based upon a statistical analysis of features extracted from the voice signal. The system includes an acoustic processor and a classifier. The acoustic processor extracts features that are characteristic of the voice signal and generates feature vectors using the extracted spectral features. The classifier uses the feature vectors to compute the probability that the voice signal belongs to each of the predefined categories and classifies the voice signal to a predefined category that is associated with the highest probability.

Abstract: A method for unifying speech user interface and graphic user interface commands includes the steps of receiving (52) user entered text via a GUI, processing (54) the user-entered text, monitoring (60) the user-entered text, adding input context (62) to the user-entered text, and, updating (74, 76, and 78) a speech recognizer with the user-entered text and semantic information. Updating the speech recognizer can include the step of accepting new text information and input context to update a speech grammar (74) and recognition vocabulary of the speech recognizer. The method can include the step of updating the GUI (72) by updating GUI directives (68) and elements (70) to maintain the GUI unified with the speech grammar. The method can further include the step of forming a window (402) enabling the display of a speech interface command as a user constructs the speech interface command using the GUI (400).

Abstract: A user interface, and associated techniques, that permit a fast and efficient way of correcting speech recognition errors, or of diminishing their impact. The user may correct mistakes in a natural way, essentially by repeating the information that was incorrectly recognized previously. Such a mechanism closely approximates what human-to-human dialogue would be in similar circumstances. Such a system fully takes advantage of all the information provided by the user, and on its own estimates the quality of the recognition in order to determine the correct sequence of words in the fewest number of steps.

Abstract: A system and method for identifying language attributes through probabilistic analysis is described. A set of language classes and a plurality of training documents are defined, Each language class identifies a language and a character set encoding. Occurrences of one or more document properties within each training document are evaluated. For each language class, a probability for the document properties set conditioned on the occurrence of the language class is calculated. Byte occurrences within each training document are evaluated. For each language class, a probability for the byte occurrences conditioned on the occurrence of the language class is calculated.

Abstract: A computer method, system and code, for representing a natural-language document in a vector form suitable for text manipulation operations are disclosed. The method involves determining (a) for each of a plurality of terms selected from one of (i) non-generic words in the document, (ii) proximately arranged word groups in the document, and (iii) a combination of (i) and (ii), a selectivity value of the term related to the frequency of occurrence of that term in a library of texts in one field, relative to the frequency of occurrence of the same term in one or more other libraries of texts in one or more other fields, respectively. The document is represented as a vector of terms, where the coefficient assigned to each term includes a function of the selectivity value determined for that term.

Abstract: A method is provided for optimizing an objective measure used to estimate mean opinion score or naturalness of synthesized speech from a speech synthesizer. The method includes using an objective measure that has components derived directly from textual information used to form synthesized utterances. The objective measure has a high correlation with mean opinion score such that a relationship can be formed between the objective measure and corresponding mean opinion score. The objective measure is altered to provide a different function of textual information derived from the utterances so as to improve the relationship between the scores of the objective measure and subjective ratings of the synthesized utterances.

Abstract: A pitch adaptive circuit (200) includes an equalizer control circuit (206) that evaluates the pitch of the speech signals that are being processed and depending on the pitch information, the equalizer control circuit (206) selects an equalizer (208, 210) to shape the decoded speech signals. By selecting the best equalizer (208 or 210) to use based on the pitch information, improvements in audio quality are provided automatically without user intervention.

Abstract: A method and apparatus converts input data such as short notes into a global text realization to provide semantically-coherent grammatical text. In various exemplary embodiments, an individual inputs short notes into a computer system, the computer system associates local text realizations with the short notes. Subsequently, the user may select the appropriate local text realizations, which may be converted to semantic representations and to semantically coherent grammatical text or a global text realization.

Abstract: In a disclosed method for interpreting automatic speech recognition (ASR) performance data, a data processing system may receive user input that selects a log file to be processed. The log file may contain log records produced by an ASR system as a result of verbal interaction between an individual and the ASR system. In response to receiving the user input, the data processing system may automatically interpret data in the log records and generate interpretation results. The interpretation results may include a duration for a system prompt communicated to the individual by the ASR system, a user response to the system prompt, and a duration for the user response. The user response may include a textual representation of a verbal response from the individual, obtained through ASR. The interpretation results may also include an overall duration for the telephone call.