摘要:
An optical fiber cable includes at least one buffer tube that includes a plurality of water-blocking plugs and an optical fiber. The water-blocking plugs can be spaced along the buffer tubes, substantially filling the cross-sectional space within the buffer tube not already filled by the optical fiber. The water-blocking plugs can provide a stronger bond between the optical fibers and the inner tube. This is reflected by a high normalized pullout force for the optical fiber, such as, above 5.0 N/m. Yet, the resulting fiber optic cable does not suffer from problems associated with a higher pullout force, such as attenuation.
摘要:
An optical fiber cable includes at least one buffer tube that includes a plurality of water-blocking plugs and an optical fiber. The water-blocking plugs can be spaced along the buffer tubes, substantially filling the cross-sectional space within the buffer tube not already filled by the optical fiber. The water-blocking plugs can provide a stronger bond between the optical fibers and the inner tube. This is reflected by a high normalized pullout force for the optical fiber, such as, above 5.0 N/m. Yet, the resulting fiber optic cable does not suffer from problems associated with a higher pullout force, such as attenuation.
摘要:
A composite cable for distributing electrical power to components in an optical fiber network and for transmitting optical signals between optical fiber network components includes at least one layer of insulated electrical conductors arranged in side-by-side relation to provide a layer of conductors of a thickness substantially equal to the thickness of the conductors. The conductors of the at least one layer of conductors are S-Z stranded and surround the optical fibers which are loosely contained in at least one plastic buffer tube to provide desirable structural and operational features to the cable and to an optical fiber network in which the cable can be included.
摘要:
The invention relates to the coding of audio signals that may include both speech-like and non-speech-like signal components. It describes methods and apparatus for code excited linear prediction (CELP) audio encoding and decoding that employ linear predictive coding (LPC) synthesis filters controlled by LPC parameters, a plurality of codebooks each having codevectors, at least one codebook providing an excitation more appropriate for non-speech-like signals and at least one codebook providing an excitation more appropriate for speech-like signals, and a plurality of gain factors, each associated with a codebook. The encoding methods and apparatus select from the codebooks codevectors and/or associated gain factors by minimizing a measure of the difference between the audio signal and a reconstruction of the audio signal derived from the codebook excitations. The decoding methods and apparatus generate a reconstructed output signal from the LPC parameters, codevectors, and gain factors.
摘要:
A vector excitation coder compresses vectors by using an optimum codebook designed off line, using an initial arbitrary codebook and a set of speech training vectors exploiting codevector sparsity (i.e., by making zero all but a selected number of samples of lowest amplitude in each of N codebook vectors). A fast-search method selects a number N.sub.c of good excitation vectors from the codebook, where N.sub.c is much smaller thaORIGIN OF INVENTIONThe invention described herein was made in the performance of work under a NASA contract, and is subject to the provisions of Public Law 96-517 (35 USC 202) under which the inventors were granted a request to retain title.
摘要:
The application relates to a method for determining at least one updated filter coefficient of an adaptive filter (22) adapted by an LMS algorithm. According to the method, filter coefficients of a first whitening filter (25′) are determined, in particular filter coefficients of an LPC whitening filter. The first whitening filter (25′) generates a filtered signal. A normalization value is determined based on one or more computed values obtained in the course of determining the filter coefficients of the first whitening filter (25′). The normalization value is associated with the energy of the filtered signal. At least one updated filter coefficient of the adaptive filter (22) is determined in dependency on the filtered signal and the normalization value. Preferably, updated filter coefficients for all filter coefficients of the adaptive filter (22) are determined.
摘要:
Analysis and synthesis filter banks such as those used in audio and video coding systems are each implemented by a hybrid transform that comprises a primary transform in cascade with one or more secondary transforms. The primary transforms for the filter banks implement an analysis/synthesis system in which time-domain aliasing artifacts are cancelled. The secondary transforms, which are in cascade with the primary transforms, are applied to blocks of transform coefficients. The length of the blocks is varied to adapt the time resolution of the analysis and synthesis filter banks.
摘要:
An electroacoustic channel soundfield is altered. An audio signal is applied by an electromechanical transducer to an acoustic space, causing air pressure changes therein. Another audio signal is obtained by a second electromechanical transducer, responsive to air pressure changes in the acoustic space. A transfer function estimate of the electroacoustic channel is established, responsive to the second audio signal and part of the first audio signal. The transfer function estimate is derived to be adaptive to temporal variations in the electroacoustic channel transfer function. Filters are obtained with transfer functions based on the transfer function estimate. Part of the first audio signal is filtered therewith.
摘要:
The invention relates to the coding of audio signals that may include both speech-like and non-speech-like signal components. It describes methods and apparatus for code excited linear prediction (CELP) audio encoding and decoding that employ linear predictive coding (LPC) synthesis filters controlled by LPC parameters, a plurality of codebooks each having codevectors, at least one codebook providing an excitation more appropriate for non-speech-like signals and at least one codebook providing an excitation more appropriate for speech-like signals, and a plurality of gain factors, each associated with a codebook. The encoding methods and apparatus select from the codebooks codevectors and/or associated gain factors by minimizing a measure of the difference between the audio signal and a reconstruction of the audio signal derived from the codebook excitations. The decoding methods and apparatus generate a reconstructed output signal from the LPC parameters, codevectors, and gain factors.