Abstract: A method for determining mantissa bit allocation of frequency domain audio data to be encoded, including by performing adaptive low frequency compensation on each frequency band of a set of low frequency bands of the data. The low frequency compensation includes steps of: performing tonality detection on the audio data to generate compensation control data indicative of whether each frequency band in the set has prominent tonal content; and performing low frequency compensation on each frequency band in the set having prominent tonal content, including by correcting a preliminary masking value for each frequency band having prominent tonal content, but not performing low frequency compensation on the audio data in any other frequency band in the set. Other aspects are audio encoding methods including such tonality detection and low frequency compensation steps, and a system configured to perform any embodiment of the inventive method.

Abstract: A method and apparatus for enhancing a desired audio signal for delivery through an electroacoustic channel include obtaining a noise estimate attributable to an external disturbance, applying the noise estimate to a dynamic noise compensation (DNC) process to thereby condition the desired audio signal as a function of the spectral characteristics of the noise estimate, applying the noise estimate to an adaptive equalization (AEQ) process to thereby condition the desired audio signal as a function of the electroacoustic response of the electroacoustic channel, and applying the noise estimate to an active noise cancellation (ANC) process configured to generate anti-noise for delivery into the electroacoustic channel.

Abstract: 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.

Abstract: 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.

Abstract: 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.

Abstract: 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.

Abstract: 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.

Abstract: 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.

Abstract: 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.

Abstract: A method of encoding time-discrete audio signals comprises the steps of weighting the time-discrete audio signal by means of window functions overlapping each other so as to form blocks, the window functions producing blocks of a first length for signals varying weakly with time and blocks of a second length for signals varying strongly with time. A start window sequence is selected for the transition from windowing with blocks of the first length to windowing with blocks of the second length, whereas a stop window sequence is selected for the opposite transition. The start window sequence is selected from at least two different start window sequences having different lengths, whereas the stop window sequence is selected from at least two different stop window sequences having different lengths. A method of decoding blocks of encoded audio signals selects a suitable inverse transformation as well as a suitable synthesis window as a reaction to side information associated with each block.

Abstract: The invention relates in general to low bit-rate encoding and decoding of information such as audio information. More particularly, the invention relates to computationally efficient adaptive bit allocation and quantization of encoded information useful in high-quality low bit-rate coding systems.In one embodiment, an audio split-band encoder splits an input signal into frequency subband signals, quantizes the subband signals according to values established by an allocation function, and assembles the quantized subband signals into an encoded signal. The allocation function establishes allocation values in accordance with psychoacoustic principles based upon a masking threshold.

Abstract: A signal processing system comprising components such as split-band perceptual coders can receive a peak amplitude limited input audio signal and can process the signal in such a manner that the processed signal preserves the apparent loudness of the input signal but is no longer peak-amplitude limited. In one embodiment, upsampling is used in estimating the resultant peak amplitude and gain factors established in response to the estimated peak amplitude are applied to one or more frequency subbands of the processed signal.

Abstract: The invention relates to formatting encoded audio information in a form suitable for transmission or storage. Audio information is encoded into a binary form, using an invariant number of bits to represent at least some but not all of the encoded information. The information represented by an invariant number of bits is assembled into pre-established positions within a formatted frame.

Abstract: The invention relates in general to low bit-rate encoding and decoding of information such as audio information. More particularly, the invention relates to the suppression of quantizing noise which results from low bit-rate encoding and decoding of signal events such as transients. An encoder identifies intervals of an input signal in which a decoder is likely to generate audible artifacts caused by quantizing errors. Indications of these intervals are passed in the encoded signal to the decoder. The decoder uses these indications to eliminate quantizing artifacts by suppressing corresponding intervals in the output signal.

Abstract: The invention relates in general to high-quality low bit-rate digital transform coding and decoding of information corresponding to audio signals such as music signals. More particularly, the invention relates to signal analysis/synthesis in coding and decoding. The invention can optimize the trade off in transform coders between time resolution and frequency resolution by adaptively selecting the transform block length for each sampled audio segment, and/or can optimize coding gain by adaptively selecting the transform and/or by adaptively selecting the analysis window or the analysis/synthesis window pair.

Abstract: The invention relates in general to digital encoding and decoding of information. More particularly, the invention relates to efficient implementation of digital analysis and synthesis filter banks used in encoding and decoding. The invention permits the length of a digital transform used to implement critically-sampled analysis and synthesis filter banks to be adaptively selected.

Abstract: A transform encoder, a transform decoder, and a transform encoder/decoder system utilize complex pre- and post-transform multiplication of input signal samples to implement concurrent application of a modified Discrete Cosine Transform and a modified Discrete Sine Transform according to the Evenly-Stacked Time Domain Aliasing Cancellation technique against two channels of input signal samples, and to reduce the computational complexity of a digital filter bank of a modified Discrete Cosine Transform.

Abstract: A transform encoder, a transform decoder, and a transform encoder/decoder system employ adaptive bit allocation wherein each code word representing spectral information is allocated a fixed number of bits and an adaptive number of bits, except that at least some but not all code words are allocated a fixed number of bits.

Abstract: Quantizing noise is lessened in a speech signal system by using adaptive bit allocation for subband channels, wherein subband information of digital words is represented in block-floating-point form, and normalized mantiss as may allow dropping a sign bit.

Abstract: 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.