Abstract: The improved AES processing method provides an efficient alternative to both Mips intensive multiplication and to conventional table lookup, used to multiply terms over a Galois field (GF). The improved method takes advantage of the fact that in the GF, any non zero element X can be represented by a power of a primitive element P. The improved method thereby results in a 2 by 256 table. The log base P of the terms being multiplied are looked up and summed, and the anti-log of the sum is looked up in the same table.

Abstract: A identity of a remote speaker is verified by receiving compressed audio formants from a remote Internet telephony client and comparing the compressed audio formants with sample compressed audio formants known to represent the person the remote speaker purports to be. The compressed audio formants include energy and pitch data characterizing the residue of the speaker uttering a predetermined pass phrase and a plurality of formant coefficients characterizing the resonance of the speaker.

Abstract: The improved AES processing method provides an efficient alternative to both Mips intensive multiplication and to conventional table lookup, used to multiply terms over a Galois field (GF). The improved method takes advantage of the fact that in the GF, any non zero element X can be represented by a power of a primitive element P. The improved method thereby results in a 2 by 256 table. The log base P of the terms being multiplied are looked up and summed, and the anti-log of the sum is looked up in the same table.

Abstract: A microphone for a full duplex speaker phone comprises a loudspeaker, a uni-directional microphone, a omni-directional microphone, and a background noise suppression circuit. The uni-directional microphone is a directional microphone such that the intensity of an acoustic source detected at the uni-directional microphone is dependent on the angular direction of the acoustic source relative to a reference direction. The omni-directional microphone is a non-directional microphone such that the intensity of the acoustic source detected at the omni-directional microphone is substantially independent of the angular direction of the acoustic source relative to the reference direction. The background noise suppression circuit determines the direction of the acoustic source by calculating a quotient between a signal strength of the acoustic source as detected by the two microphones and provides for suppressing an acoustic source that is not within a threshold angle of the reference angle.

Abstract: A identity of a remote speaker is verified by receiving compressed audio formants from a remote Internet telephony client and comparing the compressed audio formants with sample compressed audio formants known to represent the person the remote speaker purports to be. The compressed audio formants include energy and pitch data characterizing the residue of the speaker uttering a predetermined pass phrase and a plurality of formant coefficients characterizing the resonance of the speaker.

Abstract: A method and apparatus for improving burst error correction. The method and apparatus involve transmitting block codes with interleaving, and predicting a burst error in the received block codes. Burst error prediction includes decoding received block codes and determining if the decoding is successful. The method includes receiving a code having an error correction capability, an erasure detection capability and a burst error length associated with it. A decoding step is performed to find an error in the code. The error has a length associated with it. The decoding is successful if the error length found is less than twice the error correction capability. If an initial decode attempt is not successful, then a burst error length is associated with the code and another decode is performed. The burst error length has an initial value that is less than the error correction capability.

Abstract: A data signal compression technique for real-time voice and data processing where the digitized signal is first compressed to obtain a first compressed signal, and the first compressed signal is then compressed again to obtain a second compressed signal. Within a digital signal processor, digital signals first undergo time scale compression after which the compressed signals undergo audio compression to achieve multiple-compressed signals. Upon reception in a second digital signal processor, the multiple-compressed signals are correspondingly decompressed to achieve a high-quality estimation of the original digital signals.