Abstract: The distance computation represents a central, constantly recurrent task in sample and speech recognition. It is used in speech recognition as a degree of similarity between a part of a speech utterance and a speech reference. In picture processing and sample recognition, it is used for data compression. The distance computation requires the longest computation time so that a reduction of the computation time results in a considerable efficiency improvement. A reduction of the computation time is achieved by the integration of the distance computation in a memory module in which particularly the reference data are stored. Due to this integration, the other components of the overall system are relieved of this constantly recurrent task and are available for more complex processes in this period of time. This integration makes the distance computation essentially shorter because the communication between memory sections and computation unit takes place directly without utilizing a busy system.

Abstract: A speech controlled computer user interface communicates between a user and at least one application program. The user interface has a speech layer, an utterance layer, and a discourse layer. The speech layer is in communication with the user and converts between speech messages and text messages. The utterance layer is in communication with the speech layer, and converts between text messages and semantic meaning messages. The discourse layer is in communication with the utterance layer and the at least one application program, and processes messages from the user and the at least one application program, and generates responsive messages to the user and the at least one application program.

Abstract: In a ringing-tone control device for a telephone set, an audio signal which is sent from a communication opposite party is converted into a corresponding bi-level signal. The bi-level signal is stored into the memory. The bi-level signal is read out from the memory. A ringing tone is generated in response to the bi-level signal read out from the memory.

Abstract: A method for performing noise suppression and channel equalization of a noisy voice signal comprising the steps of sampling the noisy voice signal at a predetermined sampling rate fs; segmenting the sampled voice signal into a plurality of frames having a predetermined number of samples per frame, over a predetermined temporal window; generating an N-point spectral sample representation of each of the sample signal frames; determining the magnitude of each of the N-point spectral samples and generating a histogram of the energy associated with each of the N-point spectral samples at a particular frequency; detecting a peak amplitude of the histogram which corresponds to a noise threshold Nf associated with the particular frequency; determining a channel frequency response Cf associated with the particular frequency by determining a geometric mean over all the spectral samples having magnitude exceeding the noise threshold Nf; subtracting from each of the magnitudes of the N point spectral samples the noise th

Abstract: A technique for obtaining an intermediate set of frequency dependant features from a speech signal for use in speech processing and in obtaining estimates of speech pitch. The technique utilizes multiple tapers derived from Slepian sequences to obtain a product of the speech signal and the Slepian functions. Multiple tapered Fourier transforms are then obtained from the product, from which the set of frequency dependent features are calculated. In a preferred embodiment, a derivative of the cepstrum of the speech signal is used as an estimate of speech signal pitch. In another preferred embodiment, the F-spectrum is calculated from the product and the F-cepstrum is obtained therefrom by calculating the Fourier transform of the smoothed derivative of the log of the F-spectrum. The maximum of the F-cepstrum also provides a pitch estimation.

Abstract: A computer program product residing on a computer readable medium for avoiding headers/footers while synthesizing speech for a reading machine includes instructions for causing the reading machine to determine if text in regions of a document correspond to text of a header or a footer and synthesize speech to read the document aloud to a user of the reading machine, while ignoring those portions of the document that correspond to a header or footer.

Abstract: A method and apparatus for compressing and decompressing an audio signal. The apparatus comprises an input for receiving an audio signal derived from a spoken utterance, the audio signal being contained into a plurality of successive data frames. A data frame holding a certain portion the audio signal is processed to generate a feature vector including a plurality of discrete elements characterizing at least in part the portion of the audio signal encompassed by the frame, the elements being organized in a certain sequence. The apparatus makes use of a compressor unit having a grouping processor for grouping elements of the feature vector into a plurality of sub-vectors on the basis of a certain grouping scheme, at least one of the sub-vectors including a plurality of elements from the feature vector, the plurality of elements being out of sequence relative to the certain sequence. The plurality of sub-vectors are then quantized by applying a vector quantization method.

Abstract: A method for separation of acoustical data into narrowband and broadband time series components includes the steps of: constructing a data matrix in which time series samples of acoustical data are arranged in a forward-backward linear predictor matrix; decomposing a subband of the matrix into an in-band component and an out-of-band component; estimating narrowband components; estimating out-of-band leakage; removing the in-band narrowband component and the out-of-band narrowband component; repeating sequentially the afore mentioned five steps until narrowband components from all of the subbands of the matrix are removed from the acoustical data; and reconstructing the broadband time series; whereby to provide a narrowband time series estimate indicative of the presence of a signal in the acoustical data.

Abstract: An apparatus, method and system are provided for personal telecommunication speed calling. The system includes a line unit couplable to customer premise equipment; a network signaling interface for receiving network signaling information; a network trunk interface for receiving corresponding network communications; a memory; and a processor coupled to the memory, to the line unit, to the network signaling interface, and to the network trunk interface. The processor includes instructions for generating a user specific affinity database, which contains names and corresponding telephone numbers of those people with whom the user has an affinity and therefore is likely to want to speed call. The affinity database is generated based upon affinity information derived from the user's outgoing calls and the user's incoming calls which have been answered and which have a minimum duration.

Abstract: A method and apparatus is disclosed for automatic segregation of signals of different origin, using models that statistically characterize a wave signal, more particularly including feature vectors consisting of a plurality of parameters extracted from a data stream of a known type for use in identifying data types by comparison, which can be Hidden Markov Model based methods, thereby enabling automatic data type identification and routing of received data streams to the appropriate destination device, thereby further enabling a user to transmit different data types over the same communication channel without changing communication settings.

Abstract: A system for conversant interaction includes a recognizer for receiving and processing input information and outputting a recognized representation of the input information. A dialog manager is coupled to the recognizer for receiving the recognized representation of the input information, the dialog manager having task-oriented forms for associating user input information therewith, the dialog manager being capable of selecting an applicable form from the task-oriented forms responsive to the input information by scoring the forms relative to each other. A synthesizer is employed for converting a response generated by the dialog manager to output the response. A program storage device and method are also provided.

Abstract: A method retrieves a multi-media segment from a signal stream having an audio component and a closed caption component. This includes separating the audio component and the closed caption text component from the signal stream, generating an audio pattern representative of the start of the multi-media segment, locating the audio pattern in the audio component, and temporally aligning the text from the closed caption text component with the audio pattern in the audio component. Locating the audio pattern in the audio component includes retrieving text from the closed caption text component; and comparing the text against one or more keywords delimiting the multi-media segment. Once located, the multi-media segment may be played on-demand. In addition, an apparatus retrieves a multi-media segment from a signal stream, the signal stream having an audio component and a closed caption text component.

Abstract: An improved speech coder takes advantage of the fact that any given pulse combination can be uniquely described by the following four properties: number of degenerate pulses, signs of pulses, positions of pulses, and pulse magnitudes. In accordance with the invention, a four stage iterative classification of the pulse combinations, where each stage groups the pulse combinations by one of these four properties, is performed. The process starts with the number of pulses, then determines the total number of possible sign combinations, pulse position combinations, and pulse magnitude combinations. This flexibility allows for the sign combinations to be grouped in the last stage. Since the number of sign combinations is always a power of two, leaving the sign combinations for last along with appropriately ordering the elements in the previous three stages allows the signs to be coded by independent bits, in turn allowing for error protection of those bits.

Abstract: In a speaker normalization processor apparatus, a vocal-tract configuration estimator estimates feature quantities of a vocal-tract configuration showing an anatomical configuration of a vocal tract of each normalization-target speaker, by looking up to a correspondence between vocal-tract configuration parameters and Formant frequencies previously determined based on a vocal tract model of the standard speaker, based on speech waveform data of each normalization-target speaker.

Abstract: A speech-rate converter slowing down input speech regularly monitors the data length of the input speech and the previously estimated extended output data length for the current rate scaling factor, computing new output data length estimates. The conversion rate is adaptively modified depending on the time lag between input and output speech so as to make input and output data lengths consistent without skipping any spoken input portions. Input signal power is monitored to discriminate speech and non-speech intervals, and the portions of input non-speech intervals exceeding a conversion-rate-dependent duration are deleted.

Abstract: A subscriber loop interface circuit is provided that includes a current directing element for directing current from the output stage of line interface amplifiers. The current is directed to a secondary power supply having a magnitude lower than the conventional primary power supply of a subscriber loop interface circuit in supplying power to a subscriber loop.

Abstract: An audio decoding device for decoding coded audio information with multiple channels includes a coded information memory section for storing the coded audio information; an information transmission section for reading the coded audio information stored at an arbitrary position in the coded information memory section; and an audio decoding section for decoding the coded audio information read by the information transmission section and outputting the resultant audio information in accordance with a time axis.

Abstract: A method and apparatus for encoding speech for communication to a decoder for reproduction of the speech where the speech signal is classified into steady state voiced (harmonic), stationary unvoiced, and “transitory” or “transition” speech, and a particular type of coding scheme is used for each class. Harmonic coding is used for steady state voiced speech, “noise-like” coding is used for stationary unvoiced speech, and a special coding mode is used for transition speech, designed to capture the location, the structure, and the strength of the local time events that characterize the transition portions of the speech. The compression schemes can be applied to the speech signal or to the LP residual signal.

Abstract: Dolby AC-3 and MPEG-2 audio permit the transmission of audio signals with more than two independent audio channels. If a reproduction device has only a two-channel audio decoder (DEC1) then an external multi-channel audio decoder (DEC2) can be used for multi-channel sound reproduction. If the audio reproduction at the same time accompanies video reproduction, then a synchronization method is required in order to achieve lip synchronism between picture and sound. According to the invention, for the synchronization of a first decoder (DEC1), which merely has two-channel compatibility, with a second decoder, which has multi-channel compatibility, a counting variable is allocated a value (F) which is produced from system parameters such as the data coding method used, the transmission speed and/or the data rate. The data are received by the first decoder and output by the latter to the second decoder, the counting variable being decremented or incremented respectively for a specific volume of data.

Abstract: A speech processing system receives multiple streams of speech frames. The system selects among concurrent ones of the frames a subset of those frames that are the most relevant, based on pre-assigned stream priorities and energy content of the frames. The selected frames are then decoded and rendered. The resulting signals are mixed. This architecture provides bandwidth scalability and/or processing power scalability.