Abstract: A greeting message is recorded and stored at a first rate for use in a system. The system includes a command indicating the desired rate to be used in coding on the system. The greeting message is then reconstructed and re-encoded based on the desired rate. Alternatively, the greeting message may be stored in a plurality of rate formats. The appropriate rate format may then be selected by the system.

Abstract: A method of controlling access of a speaker to one of a service and a facility, the method comprising the steps of: (a) receiving first spoken utterances of the speaker, the first spoken utterances containing indicia of the speaker; (b) decoding the first spoken utterances; (c) accessing a database corresponding to the decoded first spoken utterances, the database containing information attributable to a speaker candidate having indicia substantially similar to the speaker; (d) querying the speaker with at least one question based on the information contained in the accessed database; (e) receiving second spoken utterances of the speaker, the second spoken utterances being representative of at least one answer to the at least one question; (f) decoding the second spoken utterances; (g) verifying the accuracy of the decoded answer against the information contained in the accessed database serving as the basis for the question; (h) taking a voice sample from the utterances of the speaker and processing the voic

Abstract: A security method compares a present verbal utterance with a previously recorded verbal utterance by comparing frequency domain representations of the utterances, with multiple repeat utterances forming a basis for determining a variation in repetitious performance by an individual, and similar differences between enrollment and challenge utterances forming a basis for a similar analysis of variance between enrollment and challenge utterances. In one embodiment a set of enrollment data is searched by each challenge until either a match is made, indicating an action, possibly dependent upon the specific match, or no match is made indicating an abort.

Abstract: A family of low-complexity, high quality CELP speech coders are described which use two new techniques: Backward and Inverse Filtered Target (BIFT) for fixed codebook excitation search; and Tree-Structured Multitap adaptive codebook search. Incorporation of these new techniques resulted in very low complexity CELP coders at less than 16 Kb/s. The three coefficients for linear combination of the adaptive codebook are chosen from a tree-structured tap codebook. The best tap index in the primary codebook points to a secondary codebook where the search is further conducted. This procedure may be repeated many times, wherein each subsequent tap codebook points to yet another subsequent tap codebook, which points to yet another subsequent tap codebook, etc. A fixed ternary excitation codebook using a new technique called Backward and Inverse Filtered Target matching (BIFT), is used to encode the portion of the target signal that is left behind after the adaptive codebook contribution has been subtracted.

Abstract: A network interface unit with a base and cover having two layers of equipment with the upper layer being hinged to the base and the cover being hinged to an extension of the base. The hinged arrangement allows the cover and one of the layers of the equipment to be swung out to facilitate maintenance.

Abstract: A network interface unit having a plurality of modular layers mountable on each other in a consecutive manner with each modular layer having provisions for being hinged.

Abstract: A system and method for providing user feedback for coupling the speaker and microphone of a communications device with the mouthpiece and earpiece of a telephone handset is provided in which the device may indicate to the user when a poor coupling exists causing a poor signal quality value to be generated. The system may provide feedback to the user, which may be visual or audible, so that the user may correct the interface between the communications device and the telephone handset until a predetermined quality signal is generated indicating that communications may occur between the communications device and the telephone handset.

Abstract: A signal including speech and background noise is encoded by first decomposing the signal into speech and noise components. A first speech encoding algorithm is then used to generate codebook indices for the speech component and a second algorithm is applied to generate codebook indices for the noise component. The speech encoding algorithm performs better since it faces clean speech, while a simple, very low bit rate algorithm may be used to encode the noise.

Abstract: A recognition test matches a speech segment supplied to the system with a set of parameters associated with a reference and memorized in a dictionary. A provisional version of each set of parameters to be memorized in the dictionary in association with a reference is estimated on the basis of one or more segments of speech, after which the provisional version is memorized in the dictionary in association with the reference. At least one repetition of the speech segment is submitted to a recognition test, after which depending on whether it has matched the speech segment with the provisional version, the provisional version is modified and the modified provisional version is memorized.

Abstract: A computer telephony system including a telephone communication unit operable for initiating and receiving voice calls and a computer operable for providing and receiving voice data. Circuitry is provided for coupling voice data between the computer and the telephone communication unit simultaneously with the voice calls, and for isolating the voice data from the voice calls. In another embodiment of the invention, there is provided a computer telephony amplifier for use with a telephone communication unit operable for initiating and receiving voice calls and a computer operable for providing and receiving voice data. The amplifier includes circuitry for transmitting and receiving voice data between the computer and the telephone communication unit simultaneously with the voice calls, and for isolating the voice data from the voice calls.

Abstract: A voice encoder using a VOX (voice operated transmission) control has a pitch analyzer and a high-efficiency encoder. When a voiced state is detected in an input audio signal, the input audio signal and pitch information extracted therefrom are encoded by the high-efficiency encoder and transmitted to a voice decoder. When an unvoiced state is detected, the high-efficiency encoder encodes the input audio signal without a gain of the pitch information. The encoded data without using the gain information is transmitted after a post-amble signal to obtain natural background noise.

Abstract: An apparatus for reproducing recorded signals by using a recorded medium and a method for reproducing recorded signals. The learning of language is done by using the general recorded medium such as cassette tape and video tape with movies or music recorded thereon, thereby improving the learning efficiency. In the method for reproducing recorded signals of a recording medium in a language learning apparatus, the operation is carried out in the following manner. A control section switches a first switch in accordance with a reproduction command of a reproduction key inputting section, so that the audio signals of an audio signal processing section would be supplied to a speaker. Further, the control section turns on a second switch in accordance with a voice recognition command of a voice recognition key inputting section, so that the voices of a voice detecting section would be supplied and stored to a voice recognizing section.

Abstract: An audio decoder is provided with a programmable and re-configurable downmixing process. In one embodiment, the audio decoder includes a control module and a data path. The data path is configured to read, scale, add, and write audio samples to and from various audio channel frame buffers. The control module implements state diagrams which specify various control signals for directing the operations of the data path. The control module implements state diagrams for directing windowing and downmixing operations. The order in which these operations are performed may be reconfigurable, i.e. downmixing may be performed before or after windowing. This reconfigurability advantageously permits the system designer to trade a slight audio quality enhancement for a decreased memory requirement for some speaker configurations. The downmixing operation requires scaling coefficients which are provided by the control module.

Abstract: The system described herein facilitates the control, by spoken input, of multiple user application programs running under an operating system providing multi-tasking capability. A speech recognition program runs as one of the multi-tasked programs. The recognition program is operative to provide translation, either text or program commands, of spoken words in accordance with a translation table. Each of the other user application programs which is to be controlled is provided with a respective translation table section. In accordance with the invention, the recognition program includes means for creating a database for each of the other user application programs which is run and stores, in the respective database, a record reflecting each translation which is sent to a running one of the other application programs. These records identify the speech event which was translated and also identifies alternate possible translations of the speech event.

Abstract: An adaptive linear predictor is used to predict samples, and residuals from such predictions are encoded using Golomb-Rice encoding. Linear prediction of samples of a signal which represents digitized sound tends to produce relatively low residuals and those residuals tend to be distributed exponentially. Accordingly, linear prediction combined with Golomb-Rice encoding produces particularly good compression rates with very efficient and simple implementation. The accuracy of the linear predictor is improved by including, in the prediction of a current sample of a first channel of the digitized signal, look-ahead sample data from a corresponding second channel of the digitized signal. For example, prediction of a right channel sample of a digitized, stereo, audio signal is improved by inclusion of look-ahead left channel sample data in the right channel sample predictor.

Abstract: An audio decoder is provided with a programmable and re-configurable downmixing process. In one embodiment, the audio decoder includes a control module and a data path. The data path is configured to read, scale, add, and write audio samples to and from various audio channel frame buffers. The control module implements state diagrams which specify various control signals for directing the operations of the data path. The control module implements state diagrams for directing windowing and downmixing operations. The order in which these operations are performed may be reconfigurable, i.e. downmixing may be performed before or after windowing. This reconfigurability advantageously permits the system designer to trade a slight audio quality enhancement for a decreased memory requirement for some speaker configurations. The downmixing operation requires scaling coefficients which are provided by the control module.

Abstract: Noise is suppressed in an input signal that carries a combination of noise and speech. The input signal is divided into signal blocks, which are processed to provide an estimate of a short-time perceptual band spectrum of the input signal. A determination is made at various points in time as to whether the input signal is carrying noise only or a combination of noise and speech. When the input signal is carrying noise only, the corresponding estimated short-time perceptual band spectrum of the input signal is used to update an estimate of an long term perceptual band spectrum of the noise. A noise suppression frequency response is then determined based on the estimate of the long term perceptual band spectrum of the noise and the short-time perceptual band spectrum of the input signal, and used to shape a current block of the input signal in accordance with the noise suppression frequency response.

Abstract: A method and apparatus for matching at least a first input identifier with a reference identifier. A user provides an input identifier into a system, and the system produces a recognized identifier based on the input identifier. The system of the present invention perform a check-sum operation to determine whether the recognized identifier was recognized correctly. If the check-sum operation reveals that the recognized identifier is incorrect, the system of the present invention generates a plurality of substitute identifiers. The substitute identifiers are compared to a set of pre-stored reference identifiers. If a match is found between a reference identifier and a substitute identifier, the matched reference identifier is selected as corresponding to the input identifier provided by the user.

Abstract: The present invention improves upon electronic speech synthesis using pre-recorded segments of speech to fill in for other missing segments of speech. The formalized aliasing approach of the present invention overcomes the ad hoc aliasing approach of the prior art which oftentimes generated less than satisfactory speech synthesis sound output. By formalizing the relationship between missing speech sound samples and available speech sound samples, the present invention provides a structured approach to aliasing which results in improved synthetic speech sound quality. Further, the formalized aliasing approach of the present invention can be used to lessen storage requirements for speech sound samples by only storing as many sound samples as memory capacity can support.

Abstract: An audio decoder is described which supports simple sound-effect generation. The audio decoder includes a direct access pulse code modulation (PCM) first-in-first-out buffer (FIFO) to support simple sound effect generation. In one embodiment, the audio decoder additionally includes an input buffer, a decoding module, and an output interface. The input buffer buffers incoming data frames for the decoding module to retrieve and convert to a sequence of decoded audio samples. The FIFO is configured to receive and buffer audio sound effect samples from a control component external to the audio decoder. The output interface is configurable to retrieve decoded audio samples from the decoding module and audio sound effect samples from the FIFO. Any retrieved audio sound effect samples are included in a digital audio output signal provided by the output interface. The digital audio output signal may be provided directly to a digital-to-analog converter for sound reproduction.