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 method, an apparatus, a computer readable storage medium configured with instructions for carrying out a method, and logic encoded in one or more computer-readable tangible medium to carry out actions. The method is to decode audio data that includes N.n channels to M.m decoded audio channels, including unpacking metadata and unpacking and decoding frequency domain exponent and mantissa data; determining transform coefficients from the unpacked and decoded frequency domain exponent and mantissa data; inverse transforming the frequency domain data; and in the case M
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 coeffi-cients 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 normaliza-tion 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 normaliza-tion value. Preferably, updated filter coefficients for all filter coefficients of the adaptive filter (22) are determined.
摘要:
A method derives at least three audio signals, each associated with a direction, from two input audio signals. In response to the two input signals, a passive matrix generates a plurality of passive matrix audio signals, including two pairs of passive matrix audio signals, a first pair of passive amtrix audio signals represent directions lying on a first axis and a second pair of passive matrix audio signals represent directions lying on a second axis, the first and second axes being substantially at ninety degrees to ach other. The pairs of passive matrix audio signals are processed to derive a plurality of matrix coefficients therefrom. Th e processing includes deriving a pair of intermediate signals and urging each pair of intermediate signals toward equality in response to a respective error signal. At least three ouput signals are produced by matrix multiplying the two input signals by the matrix coefficients.
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:
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 coeffi-cients 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 normaliza-tion 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 normaliza-tion value. Preferably, updated filter coefficients for all filter coefficients of the adaptive filter (22) are determined.
摘要:
A method derives at least three audio signals, each associated with a direction, from two input audio signals. In response to the two input signals, a passive matrix generates a plurality of passive matrix audio signals, including two pairs of passive matrix audio signals, a first pair of passive amtrix audio signals represent directions lying on a first axis and a second pair of passive matrix audio signals represent directions lying on a second axis, the first and second axes being substantially at ninety degrees to ach other. The pairs of passive matrix audio signals are processed to derive a plurality of matrix coefficients therefrom. Th e processing includes deriving a pair of intermediate signals and urging each pair of intermediate signals toward equality in response to a respective error signal. At least three ouput signals are produced by matrix multiplying the two input signals by the matrix coefficients.