Abstract: The invention relates to a linking unit 100, a parametric encoder 400 and a method for generating linking information L indicating components of consecutive extended segments sp and sc which may be linked together in order to form a sinusoidal track. The segments sp and sc approximate consecutive segments of a sinusoidal audio or speech signal s. The linking unit comprises a calculating unit 120 for generating a similarity matrix S(m,n) in response to received sinusoidal code data and an evaluating unit 140 for receiving and evaluating said similarity matrix S in order to generate said linking information by selecting those pairs of components m,n the similarity of which is maximal. According to the invention the calculating unit 120 is adapted to calculate the similarity matrix S by additionally considering information about the phase consistency between the components of the extended previous segment sp and the extended current segment sc.

Abstract: Apparatus for controlling noise characteristic estimation in a conferencing system, comprising a noise characteristic estimator for estimating a noise characteristic of a signal of interest transmitted in a first direction through the conferencing system, and a first voice activity detector for detecting audio signal activity in a signal transmitted through the conferencing system in a direction opposite to the signal of interest and in response disabling the noise characteristic estimator.

Abstract: A wireless, programmable, sound-activated and voice-operated remote control transmitter can be used to add hands-free speech control operation to a plurality of remotely controlled appliances manufactured by various manufacturers, each of which is normally controlled with one or more signals from an associated remote control transmitter. The present invention may be pre-programmed with a universal library of codes for controlling various appliance categories and appliances produced by various manufacturers within each category. The present invention may also be programmed using the controlled appliances' remote control transmitters and one or more operators' spoken commands. Once programming is complete, there is no need for the operator to manually operate the present invention, allowing true hands-free voice control of the remotely controlled products.

Abstract: A sound separation apparatus for separating a target signal from a mixed input signal, wherein the mixed input signal includes the target signal and one or more sound signals emitted from different sound sources. The sound separation apparatus according comprises a frequency analyzer for performing a frequency analysis on the mixed input signal and calculating spectrum and frequency component candidate points at each time. The apparatus further comprises feature extraction means for extracting feature parameters which are estimated to correspond with the target signal, comprising a local layer for analyzing local feature parameters using the spectrum and the frequency component candidate points and one or more global layers for analyzing global feature parameters using the feature parameters extracted by the local layer. The apparatus further comprises a signal regenerator for regenerating a waveform of the target signal using the feature parameters extracted by the feature extraction means.

Abstract: The present invention relates to coding and storage of phonetic features in order to search for strings of characters, whereby it is applied in particular to searching for a variety of names, identifiers, denotations and other character strings in a database. This is achieved by a method and system for coding and storing phonetic information representable as an original character sequence in which the phonetic information is coded in a bit code which does not comprise any characters. In some embodiments, tables are used and which comprise character groups that are found empirically and reflect the specific phonetics and method of spelling a name adapted to the actual language in use. This enables efficient coding of phonetic features associated with said groups and provides for adapting the coding method of the present invention to a plurality of different languages.

Abstract: A system and method of controlling the movement of a virtual agent while the agent is listening to a human user during a conversation is disclosed. The method comprises receiving speech data from the user, performing a prosodic analysis of the speech data and controlling the virtual agent movement according to the prosodic analysis.

Abstract: The present invention relates to audio data monitoring using speech recognition technology. In particular, the present invention uses business rules combined with unrestricted, natural speech recognition to monitor conversations in a customer interaction environment, literally transforming the spoken word to a retrievable data form. Implemented using the VorTecs Integration Platform (VIP), a flexible Computer Telephony Integration base, the present invention enhances quality monitoring by effectively evaluating conversations and initiating actionable events while observing for script adherence, compliance and/or order validation.

Abstract: A method and apparatus for speech recognition of the present application has a process to collate, with an input utterance, an acoustic model corresponding to a hypothesis to be expressed by the connection of utterance segments, such as phonemes or syllables, and developed according to a length of an input utterance by an inter-word connection rule thereby obtaining a recognition score. Within a word of the hypothesis, the similar hypotheses high in utterance score within a predetermined threshold from the maximum value of the score are all held to a word end irrespectively of the number of hypotheses. Meanwhile, at a word end of the hypotheses, the hypotheses are narrowed to a predetermined number of upper ranking in the order of higher score.

Abstract: A library of mouth shapes is created by separating speaker-dependent and speaker independent variability. Preferably, speaker dependent variability is modeled by a speaker space while the speaker independent variability (i.e. context dependency), is modeled by a set of normalized mouth shapes that need be built only once. Given a small amount of data from a new speaker, it is possible to construct a corresponding mouth shape library by estimating a point in speaker space that maximizes the likelihood of adaptation data and by combining speaker dependent and speaker independent variability. Creation of talking heads is simplified because creation of a library of mouth shapes is enabled with only a few mouth shape instances. To build the speaker space, a context independent mouth shape parametric representation is obtained. Then a supervector containing the set of context-independent mouth shapes is formed for each speaker included in the speaker space.

Abstract: A speech analyzing stage (12) and a method for analyzing a speech signal is described. The speech analyzing stage (12) is part of an automatic speech recognition system (10) and is adapted for analyzing in the spectral domain a speech signal sampled at one of at least two different system sampling rates. The speech analyzing stage (12) comprises a first spectral analyzer (18a) for analyzing the speech signal up to a first frequency (flowest) which is preferably derived from the lowest system sampling rate (2×flowest) and a second spectral analyzer (18b) for analyzing the speech signal at least above the first frequency (flowest).

Abstract: A method and a system for automatically finding one or more answers to a natural language question in a computer stored natural language text database is disclosed. The natural language text database has been analyzed with respect to syntactic functions of constituents, lexical meaning of word tokens and clause boundaries, and the natural language question comprises a question clause. A computer readable representation of the question clause is analyzed with respect to syntactic functions of its constituents and the lexical meaning of its word tokens. In response to the analysis a set of conditions for a clause in the natural language text database to constitute an answer to the question clause is defined. The conditions relate to the syntactic functions of constituents and the lexical meaning of word tokens in the clause.

Abstract: A speech synthesizing apparatus extracts small speech segments from a speech waveform as a prosody control target and adds inhibition information for inhibiting a predetermined prosody change process to a selected small speech segment in executing prosody control. Prosody control is performed by performing a predetermined prosody change process by using small speech segments of the extracted small speech segments other than small speech segments to which inhibition information is added. This makes it possible to prevent a deterioration in synthesized speech due to waveform editing operation.

Abstract: An encoder includes a segmentation unit for segmenting an audio or speech signal into at least one segment and a calculation unit for calculating sinusoidal code data in the form of frequency and amplitude data of a given extension from the segment such that the extension approximates the segment for a given criterion. The calculation of the sinusoidal code data ?ki, dji and eji for the segment x(n) is carried out according to the following extension {circumflex over (x)}: x ? = ? i = 1 L ? ? ? j = 0 J - 1 ? ? [ d j i ? f j ? ( n ) ? cos ? ( ? i ? ( n ) ) + e j i ? f j ? ( n ) ? sin ( ? i ? ( n ) ] . Fig . ? 1.

Abstract: The invention concerns a method and apparatus for performing packet loss or Frame Erasure Concealment (FEC) for a speech coder that does not have a built-in or standard FEC process. A receiver with a decoder receives encoded frames of compressed speech information transmitted from an encoder. A lost frame detector at the receiver determines if an encoded frame has been lost or corrupted in transmission, or erased. If the encoded frame is not erased, the encoded frame is decoded by a decoder and a temporary memory is updated with the decoder's output. A predetermined delay period is applied and the audio frame is then output. If the lost frame detector determines that the encoded frame is erased, a FEC module applies a frame concealment process to the signal. The FEC processing produces natural sounding synthetic speech for the erased frames.

Abstract: An administrative server interconnects user terminals through the communication network and transmits voice and/or text information submitted and received between the user terminals to a storage device. The server records in the storage device the voice and/or text information together with identification information of the user terminals that have submitted said voice and/or text information. The server retrieves words that disobey morals in the voice and/or text information recorded in the storage device. The server records disobedient data which indicates disobedience in morals in registration data of the user who submitted the voice and/or text information retrieved by the retrieval and presents the disobedient voice and/or text information and the identification information of the user at the submitting side to an administrator control table of the administrative server.

Abstract: A method and an apparatus for providing automated speech-to-text encoding and decoding for hearing-impaired persons. A broadband subscriber terminal interfaces to: (a) a network to convey speech packets thereover, (b) a telephone to convey speech information, and (c) a display device to display textual information of spoken words. A speech buffer in the subscriber terminal receives speech data and a processor decodes and displays textual representations of speech on the display device. A database stores voice and/or speech patterns that are used by a speech analyzer to recognize an incoming caller and to associate a name or characteristic (e.g., male or female) with the incoming call. A tonal and inflection analyzer analyzes speech to add punctuation to the displayed text. A detector, such as a DTMF detector, responds to subscriber inputs to activate/deactivate speech recognition or other functions.

Abstract: A method of and a device for output based objective speech quality assessment, wherein a degraded output speech signal comprising a speech information portion, is compared (5) with a reference signal retrieved from the output speech signal. The reference signal is provided by perceptual approximation of the speech information portion of the output speech signal using a speech recoder (2) producing a reference speech signal of finite bitrate. In a preferred embodiment, the speech recorder (2) is a speech codec.

Abstract: A language processing system includes a unified language model. The unified language model comprises a plurality of context-free grammars having non-terminal tokens representing semantic or syntactic concepts and terminals, and an N-gram language model having non-terminal tokens. A language processing module capable of receiving an input signal indicative of language accesses the unified language model to recognize the language. The language processing module generates hypotheses for the received language as a function of words of the unified language model and/or provides an output signal indicative of the language and at least some of the semantic or syntactic concepts contained therein.

Abstract: In general, this invention concerns speech encoding and decoding used in digital radio systems and a method by which the processing capacity required can be reduced in a telecommunication system using discontinuous transmission between a transmitter and receiver. In particular, the method according to the invention is used to match two telecommunication systems using different encoding methods between the transmitter and receiver. In the method, the signals transmitted by the transmitter are made suitable for the receiver in the signal path so that in the first step, at least one information parameter comprising at least two content identifiers is formed for each data frame of the data parameters (101) received. In the next step, data corresponding to the original data is synthesized from the data parameters (101) of the received frames, after which the synthesized data is transmitted for recoding with an encoding method suitable for the receiver.

Abstract: The invention relates to a method of checking the correct operation of a signal transformation wherein a input signal is transformed into an output signal. The method comprises: deriving a first robust feature from the input signal; deriving a second robust feature from the output signal; comparing said first and second robust features; in case of sufficient sumilarity, concluding a correct operation of said signal transformation, and in case of insufficient sumilarity, concluding a false operation of said signal transformation. In a special embodiment, the method is applied wherein the first robust feature is embedded in the input signal through watermark technology, the thus obtained signal being transmitted to a receiver so as to retrieve an output signal corresponding to said input signal.