Abstract: An architecturally and cost efficient method and system are disclosed for accessing a voice messaging platform. An exemplary method for accessing the platform includes: (1) forwarding a telephone call to the platform over foreign exchange lines when the telephone call involves a caller who, upon call initiation to a called party subscriber, encountered a busy or ring-no-answer condition, so that the caller may leave a message for the called party in the called party's mailbox; and (2) routing a telephone call to the platform over a trunk group local to the platform when the telephone call involves a caller-subscriber initiating a telephone call directly to the platform to access his/her mailbox to retrieve messages, administer greetings, and/or send voice messages to other subscribers. For calls forwarded over foreign exchange lines, the platform is provided with a wide variety of call information transmitted via an SMDI link.

Abstract: A communications system permits a personal mobile telecommunications device user, such as a cellular telephone subscriber, to complete a connection to a calling party that has attempted to reach the subscriber using the subscriber's assigned telephone number, even when the subscriber is roaming outside of his or her home area. The system includes a messaging and bridging complex having an associated database, which receives a call placed to the subscriber. If the subscriber is out of his or her home area so that the call cannot be completed, information stored in the database is used to initiate transmission of a paging signal to the subscriber. The connection to the calling party is held in the complex for a predetermined period after the paging signal is transmitted. The mobile telephone user, upon receiving the paging signal, may initiate a call from their personal mobile telecommunications device to the messaging and bridging complex, which bridges the subscriber and the calling party together.

Abstract: In a speech decoder which experiences frame erasure, the pitch delay associated with the first of consecutive erased frames is incremented. The incremented value is used as the pitch delay for the second of consecutive erased frames. Pitch delay associated with the first of consecutive erased frames may correspond to the last correctly received pitch delay information from a speech encoder (associated with a non-erased frame), or it may itself be the result of an increment added to a still previous value of pitch delay (associated with a still previous erased frame).

Abstract: An access control system stores two codes for each user that is authorized to access a resource, a primary code and a secondary code. When a user desiring access inputs the primary code to the system, the code is compared with the stored code for that user. If the primary code is valid, the user is allowed to access the resource. However, if the primary code is entered incorrectly, a count of the number of invalid attempts for that user is incremented, and if the count does not exceed a first threshold, the user can try again. When the number of invalid access attempts for the user exceeds the first threshold, the system requires the user to correctly input both the primary and secondary codes, before access to the resource is allowed. A second count is also maintained of the number of failed attempts in providing both codes. When the number of failed attempts exceeds a second threshold, the user is "locked out", i.

Abstract: A multimedia telecommunications network provides telephone service which may involve one or more of audio, video, and data communications. In one example of the invention, the multimedia telecommunications network contains a virtual meeting services complex which provides multimedia telecommunications service to subscribers of the network. Multimedia telephone calls can be completed automatically by the virtual meeting services complex in a manner similar to that now used to complete conventional audio telephone calls. The multimedia telecommunications network containing this virtual meeting services complex is able to connect together users having diverse equipment and is able to handle integrated and nonintegrated network access mechanisms.

Abstract: An image-processing system for perceptual coding of an image is disclosed. Coding is accomplished through an analysis of human visual sensitivity to noise in halftone images and an analysis of one or more signals representing the image to be coded. These analyses determine levels of noise. A first image is encoded so as to produce encoded values without introducing noise which exceeds a determined level of noise. Analysis of human visual sensitivity is carried out under a set of user determined conditions comprising viewing distance and lighting. The encoded image is communicated and decoded to produce representation of the first image for display. As part of displaying the representation of the first image, a halftoning process is carried out. The halftoning process may comprise the interpolation of data values of the representation of the first image as well as the addition of micro-dither to the representation.

Abstract: A new technique for the determination of the masking effect of an audio signal is employed to provide transparent compression of an audio signal at greatly reduced bit rates. The new technique employs the results of recent research into the psycho-physics of noise masking in the human auditory system. This research suggests that noise masking is a function of the uncertainty in loudness as perceived by the brain. Measures of loudness uncertainty are employed to determine the degree to which audio signals are "tone-like" (or "noise-like"). The degree of tone-likeness, referred to as "tonality," is used to determine masking thresholds for use in the compression of audio signals. Tonality, computed in accordance with the present invention, is used in conventional and new arrangements to achieve compression of audio signals.

Abstract: A telephone call is received at a relay center for routing to a forward party. A platform at the relay center can detect a call processing event in the relay center during processing of the telephone call. The relay center platform generates a context-dependent message based on the call processing event for review by the communications assistant. The communications assistant reviews and sends the message to a relay party using a message button. The relay center platform can automatically transmit messages to the caller, the forward party, a third party, and the communications assistant based on detected call processing events. The message button condenses responses to commonly occurring processing events discerned during call routing. The platform can automatically enter data determined from call processing into blanks in stored message templates. Such stored message templates can be activated and executed in response to detected call processing events.

Abstract: An automated call routing system and method which operates on a call-routing objective of a calling party expressed in natural speech of the calling party. The system incorporates a speech recognition function, as to which a calling party's natural-speech call routing objective provides an input, and which is trained to recognize a plurality of meaningful phrases, each such phrase being related to a specific call routing objective. Upon recognition of one or more of such meaningful phrases in a calling party's input speech, an interpretation function then acts on such calling party's routing objective request to either implement the calling party's requested routing objective or to enter into a dialog with the calling party to obtain additional information from which a sufficient confidence level can be attained to implement that routing objective.

Abstract: A facility is provided for allowing a caller to place a telephone call by merely uttering a label identifying a desired called destination and to charge the telephone call to a particular billing account by merely uttering a label identifying that account. Alternatively, the caller may place the call by dialing or uttering the telephone number of the called destination or by entering a speed dial code associated with that telephone number. The facility includes a speaker verification system which employs cohort normalized scoring. Cohort normalized scoring provides a dynamic threshold for the verification process making the process more robust to variation in training and verification utterences. Such variation may be caused by, e.g., changes in communication channel characteristics or speaker loudness level.

Abstract: A speech coding system employing an adaptive codebook model of periodicity is augmented with a pitch-predictive filter (PPF). This PPF has a delay equal to the integer component of the pitch-period and a gain which is adaptive based on a measure of periodicity of the speech signal. In accordance with an embodiment of the present invention, speech processing systems which include a first portion comprising an adaptive codebook and corresponding adaptive codebook amplifier and a second portion comprising a fixed codebook coupled to a pitch filter, are adapted to delay the adaptive codebook gain; determine the pitch filter gain based on the delayed adaptive codebook gain, and amplify samples of a signal in the pitch filter based on said determined pitch filter gain. The adaptive codebook gain is delayed for one subframe. The pitch filter gain equals the delayed. adaptive codebook gain, except when the adaptive codebook gain is either less than 0.2 or greater than 0.8.

Abstract: A method and apparatus for recognizing connected and degraded text is presented. A text enhancement process which may be used in combination with text recognition comprises filtering a scanned image to determine whether a binary image value of an image pixel should be complemented, determining whether complementing the value of the pixel reduces the sharpness of wedge-like figures in the image, and complementing the binary value of the pixel when doing so does not reduce sharpness. Image preprocessing may also provide a line adjacency graph (LAG) based on a set of image pixels and the removal of a path from a LAG when the path is located either at the top or bottom of the set of image pixels and is of degree (1,0) or (0,1), respectively, and connected to a junction of the LAG.

Abstract: When a collect call is initiated by a calling party, and the call is routed to an operator services position, an automated attendant, or other system in the telephone network. In addition to playing an announcement to the called party to determine if he or she will accept the call, information is furnished to the called party, preferably by an announcement, indicating (a) the anticipated cost for the call, and, optionally, (b) the relative cost of the call, as compared to the cost that would be imposed if another carrier carried the call. This information may be obtained using the dialed number and information identifying the calling party location, and by querying a database containing (a) rate information, and optionally (b) comparative information setting forth the rates of different carriers offering collect call service in the same calling area.

Abstract: A speech coding system robust to frame erasure (or packet loss) is described. Illustrative embodiments are directed to a modified version of CCITT standard G.728. In the event of frame erasure, vectors of an excitation signal are synthesized based on previously stored excitation signal vectors generated during non-erased frames. This synthesis differs for voiced and non-voiced speech. During erased frames, linear prediction filter coefficients are synthesized as a weighted extrapolation of a set of linear prediction filter coefficients determined during non-erased frames. The weighting factor is a number less than 1. This weighting accomplishes a bandwidth-expansion of peaks in the frequency response of a linear predictive filter. Computational complexity during erased frames is reduced through the elimination of certain computations needed during non-erased frames only.

Abstract: A method of making a speech recognition model database is disclosed. The database is formed based on a training string utterance signal and a plurality of sets of current speech recognition models. The sets of current speech recognition models may include acoustic models, language models, and other knowledge sources. In accordance with an illustrative embodiment of the invention, a set of confusable string models is generated, each confusable string model comprising speech recognition models from two or more sets of speech recognition models (such as acoustic and language models). A first scoring signal is generated based on the training string utterance signal and a string model for that utterance, wherein the string model for the utterance comprises speech recognition models from two or more sets of speech recognition models. One or more second scoring signals are also generated, wherein a second scoring signal is based on the training string utterance signal and a confusable string model.

Abstract: A security node disposed in the telecommunications network connecting calling and called parties transforms information (which can be voice, data, facsimile, video and other types of calls or messages) encrypted in a first format to (a) encrypted information in a different format or to (b) non-encrypted information, and vice-versa. The node is accessible from any location connected to the network. By routing calls or messages originated by the calling party and destined for the called party via the security node, and providing appropriate control signals to the node, the information may be encrypted only over a portion of the transmission path between the parties, and clear over the remainder of the transmission path. Alternatively, the information may be encrypted in different portions of the path using different encryption algorithms.

Abstract: A method and apparatus for performing a Modified Discrete Cosine Transform on an audio signal is disclosed which utilizes a Discrete Fourier Transform. Illustratively, the MDCT spectral coefficients for the signal are generated from the real FFT spectral coefficients.

Abstract: A method and apparatus for providing a differential microphone with a desired frequency response are disclosed. The desired frequency response is provided by operation of a filter, having an adjustable frequency response, coupled to the microphone. The frequency response of the filter is set by operation of a controller, also coupled to the microphone, based on signals received from the microphone. The desired frequency response may be determined based upon the distance between the microphone and a source of sound, and may comprise both a relative frequency response and absolute output level. The frequency response of the filter may comprise the substantial inverse of the frequency response of the microphone to provide a flat response. Furthermore, the filter may comprise a Butterworth filter.

Abstract: A system pattern-based speech recognition, e.g., a hidden Markov model (HMM) based speech recognizer using Viterbi scoring. The principle of minimum recognition error rate is applied by the present invention using discriminative training. Various issues related to the special structure of HMMs are presented. Parameter update expressions for HMMs are provided.

Abstract: A speech coding system robust to frame erasure (or packet loss) is described. Illustrative embodiments are directed to a modified version of CCITT standard G.728. In the event of frame erasure, vectors of an excitation signal are synthesized based on previously stored excitation signal vectors generated during non-erased frames. This synthesis differs for voiced and non-voiced speech. During erased frames, linear prediction filter coefficients are synthesized as a weighted extrapolation of a set of linear prediction filter coefficients determined during non-erased frames. The weighting factor is a number less than 1. This weighting accomplishes a bandwidth-expansion of peaks in the frequency response of a linear predictive filter. Computational complexity during erased frames is reduced through the elimination of certain computations needed during non-erased frames only.