Abstract: The invention relates to the compression coding of digital signals such as multimedia signals (audio or video), and more particularly a method for multiple coding, wherein several encoders each comprising a series of functional blocks receive an input signal in parallel. Accordingly, a method is provided in which, a) the functional blocks forming each encoder are identified, along with one or several functions carried out of each block, b) functions which are common to various encoders are itemized and c) said common functions are carried out definitively for a part of at least all of the encoders within at least one same calculation module.

Abstract: A document translation method having extracting a term from a document, executing a substitute process and outputting the term in a predetermined manner. The process of executing a substitute process on the term is performed in a translating process when the term from the document matches a term predetermined by a user. The substitute process is corresponding to the term and predetermined by the user.

Abstract: A method for automatic code generation that includes obtaining a parse tree, wherein the parse tree represents at least one instruction and wherein the instruction is in a natural language, and wherein the at least one instruction is free-form, determining a plurality of sentence subparts from the parse tree, creating at least one instantiated concept for each of the plurality of sentence subparts, inserting the instantiated concept in an instantiated concept map, traversing the instantiated concept map, wherein traversing the instantiated concept map includes for each instantiated concept in the instantiated concept map obtaining a code output rule according to the instantiated concept, and outputting code according to the code output rule be completed upon finalization of claims.

Abstract: A system for multi-lingual speech recognition. The inventive system includes a speech modeling engine, a speech search engine, and a decision reaction engine. The speech modeling engine receives and transfers a mixed multi-lingual speech signal into speech features. The speech search engine locates and compares candidate data sets. The decision reaction engine selects resulting speech models from the candidate speech models and generates a speech command.

Abstract: A method of reducing noise in an audio signal, comprising the steps of: using a furrow filter to select spectral components that are narrow in frequency but relatively broad in time; using a bar filter to select spectral components that are broad in frequency but relatively narrow in time; analyzing the relative energy distribution between the output of the furrow and bar filters to determine the optimal proportion of spectral components for the output signal; and reconstructing the audio signal to generate the output signal. A second pair of time-frequency filters may be used to further improve intelligibility of the output signal. The temporal relationship between the furrow filter output and the bar filter output may be monitored so that the fricative components are allowed primarily at boundaries between intervals with no voiced signal present and intervals with voice components. A noise reduction system for an audio signal.

Abstract: A method for evaluating the processing delay of a speech signal contained in data packets received in a receiver terminal having a telephony module during a voice call to a terminal sending the data packets over a packet-switched network. The method includes the step of obtaining from the received data packets a stream of audio packets containing the speech signal. Within a predetermined decoding time, the stream of obtained audio packets is decoded and a first reconstituted speech signal is created. At least a portion of the speech reconstituted by the telephony module is duplicated to create a second reconstituted speech signal. The time difference between the first and the second reconstituted speech signals is determined. The processing delay of the speech signal in the receiver terminal is calculated from at least the determined time difference between the reconstituted first and second speech signals and the predetermined decoding time.

Abstract: Provides subtitle generation methods and apparatus which recognizes voice in a presentation to generate subtitles thereof, and retrieval apparatus for retrieving character strings by use of the subtitles. An apparatus of the present invention includes: a extraction unit for extracting text from presentation documents; an analysis unit for morphologically analyzing text to decompose it into words; a generation unit for generating common keywords by assigning weights to words; a registration unit for adding common keywords to a voice recognition dictionary; a recognition unit for recognizing voice in a presentation; a record unit for recording the correspondence between page and time by detecting page switching events; a regeneration unit for regenerating common keywords by further referring to the correspondence between page and time; a control unit for controlling the display of subtitles, common keywords, text and master subtitles; and a note generation unit for generating speaker notes from subtitles.

Abstract: A system and method for improving the accuracy of audio searching using multiple models to process an audio file or stream to obtain search tracks. The search tracks are processed to locate at least one search term and generate multiple search results. The number of search results is equivalent to the number of models used to process the audio stream. The search results are combined to generate a unified search result. The multiple models may represent different languages, dialects and accents.

Abstract: A system and method for facilitating transcription of an audio recording, with auditing. An IVR (Interactive Voice Response) unit makes a set of audio recordings of a caller's input during a call. The set of audio recordings is stored in a database and queued for transcription by remote agents. The recordings are forwarded to a first agent who listens to the audio and completes a form with text corresponding to the audio. The system then determines whether to audit the first agent's transcription by having another agent transcribe the same set of recordings. The recordings may be re-queued and transcribed multiple times, until two (or more) of the transcriptions match, or until some threshold number of transcriptions have been obtained. An agent may contact the caller (e.g., via telephone or electronic mail) if a recording cannot be fully transcribed.

Abstract: A method for developing a system for retrieving text related to a selected concept within a text corpus includes identifying a set of semantic classes which express the concept and identifying a set of keywords for each of the semantic classes to be used in text searching in a text corpus. Each set of keywords includes at least one keyword. A plurality of syntactic rules are established which are to be applied to retrieved text which includes keywords. Each of the syntactic rules identifies a first of the semantic classes and a second of the semantic classes. A rule is satisfied when a keyword from the first of the semantic classes is in a syntactic relationship with a keyword from the second of the semantic classes. The syntactic relationship can be any one of a plurality of syntactic relationships.

Abstract: Techniques and tools for prediction of spectral coefficients in encoding and decoding are described herein. For certain types and patterns of content, coefficient prediction exploits correlation between adjacent spectral coefficients, making subsequent entropy encoding more efficient. For example, an audio encoder predictively codes quantized spectral coefficients in the quantized domain and entropy encodes results of the predictive coding. Or, for a particular quantized spectral coefficient, an audio decoder entropy decodes a difference value, computes a predictor in the quantized domain, and combines the predictor and the difference value.

Abstract: The invention can include a token list generating unit 11 for decomposing a natural language text to be processed into tokens that are components of the natural language text and registering them on a token list, and a token string selecting unit 13 for selecting optimum token strings for composing the natural language text to be processed on the basis of the token list generated by the token list generating unit 11. The token list generating unit 11 registers, on the token list, tokens among the tokens obtained by decomposing the natural language text to be processed except tokens decomposable into smaller tokens according to conditions imposed on the morphological analysis.

Abstract: The invention relates to compression coding and/or decoding of digital signals, in particular by vector variable-rate quantisation defining a variable resolution. For this purpose an impulsion dictionary comprises: for a given dimension, increasing resolution dictionaries imbricated into each other and, for a given dimension, a union of: a totality (D?i<N>) of code-vectors produced, by inserting elements taken in a final set (A) into smaller dimension code-vectors according to a final set of predetermined insertion rules (F1) and a second totality of code-vectors (Y?) which are not obtainable by insertion into the smaller dimension code vectors according to said set of the insertion rules.

Abstract: Coding an audio signal wherein values of first parameters, which represent aspects of the audio signal at a first instant are calculated to obtain first calculated values and values of second parameters, which represent the aspects of the audio signal at a second, later, instant, are calculated to obtain second calculated values, wherein the number of the first parameters and the number of the second parameters differ. The values of the subset of the second parameters are coded based on a difference of this subset and a subset of the first calculated value associated with substantially a same particular portion of the frequency range. Thus the differentially coded values of the second parameters are obtained by coding the difference of the values of second parameters and first parameters which are associated with substantially the same frequency sub-range.

Abstract: A method and apparatus map a set of vocal tract resonant frequencies, together with their corresponding bandwidths, into a simulated acoustic feature vector in the form of LPC cepstrum by calculating a separate function for each individual vocal tract resonant frequency/bandwidth and summing the result to form an element of the simulated feature vector. The simulated feature vector is applied to a model along with an input feature vector to determine a probability that the set of vocal tract resonant frequencies is present in a speech signal. Under one embodiment, the model includes a target-guided transition model that provides a probability of a vocal tract resonant frequency based on a past vocal tract resonant frequency and a target for the vocal tract resonant frequency. Under another embodiment, the phone segmentation is provided by an HMM system and is used to precisely determine which target value to use at each frame.

Abstract: A method for analyzing audio data (AD) is provided wherein a signature (SAD) is generated based on a feature set (FS) comprising a plurality of time domain related features (TDFk). The signature (SAD) is formed in order to represent a plurality of temporal positions or ranges of a plurality of characteristics of time domain related features (TDFk) within said audio data (AD).

Abstract: There is provided a speech post-processor for enhancing a speech signal divided into a plurality of sub-bands in frequency domain. The speech post-processor comprises an envelope modification factor generator configured to use frequency domain coefficients representative of an envelope derived from the plurality of sub-bands to generate an envelope modification factor for the envelope derived from the plurality of sub-bands, where the envelope modification factor is generated using FAC=?ENV/Max+(1??), where FAC is the envelope modification factor, ENV is the envelope, Max is the maximum envelope, and ? is a value between 0 and 1, where ? is a different constant value for each speech coding rate. The speech post-processor further comprises an envelope modifier configured to modify the envelope derived from the plurality of sub-bands by the envelope modification factor corresponding to each of the plurality of sub-bands.

Abstract: The present invention proposes a new method for improving the performance of a real-valued filterbank based spectral envelope adjuster. By adaptively locking the gain values for adjacent channels dependent on the sign of the channels, as defined in the application, reduced aliasing is achieved. Furthermore, the grouping of the channels during gain-calculation, gives an improved energy estimate of the real valued subband signals in the filterbank.

Abstract: The present invention proposes a new method for improving the performance of a real-valued filterbank based spectral envelope adjuster. By adaptively locking the gain values for adjacent channels dependent on the sign of the channels, as defined in the application, reduced aliasing is achieved. Furthermore, the grouping of the channels during gain-calculation, gives an improved energy estimate of the real valued subband signals in the filterbank.

Abstract: A sentence is accessed and at least one query is generated based on the sentence. At least one query can be compared to text within a collection of documents, for example using a web search engine. Collocation errors in the sentence can be detected and/or corrected based on the comparison of the at least one query and the text within the collection of documents.