Fraunhofer-Gesellschaft Zur Foerderung Der Angewandten Forschung E.V.

Dolby International Ab

Tom Baeckstroem

Guillaume Fuchs

Sascha Disch

Markus Multrus

Ralf Geiger

Christian Helmrich

Johannes Fischer

Martin Dietz

Emmanuel Ravelli

Achim Kuntz

Stefan Bayer

Konstantin Schmidt

Benjamin Schubert

Juha Vilkamo

Fabian Kuech

Wolfgang Jaegers

Grzegorz Pietrzyk

Christian Fischer Pedersen

Matthias Huettenberger

Alfonso Pino

Lars Villemoes

Max Neuendorf

Bernd Edler

Juergen Herre

Bernhard Grill

Nikolaus Rettelbach

Markus Schnell

Jürgen Herre

Florin Ghido

Vesa Ruoppila

Emma Jokinen

Jèrèmie Lecomte

An apparatus for encoding a speech signal by determining a codebook vector of a speech coding algorithm is provided. The apparatus includes a matrix determiner for determining an autocorrelation matrix R, and a codebook vector determiner for determining the codebook vector depending on the autocorrelation matrix R. The matrix determiner is configured to determine the autocorrelation matrix R by determining vector coefficients of a vector r, wherein the autocorrelation matrix R includes a plurality of rows and a plurality of columns, wherein the vector r indicates one of the columns or one of the rows of the autocorrelation matrix R, wherein R(i, j)=r(|i−j|), wherein R(i, j) indicates the coefficients of the autocorrelation matrix R, wherein i is a first index indicating one of a plurality of rows of the autocorrelation matrix R, and wherein j is a second index indicating one of the plurality of columns of the autocorrelation matrix R.

Apparatus for encoding a speech signal employing ACELP in the autocorrelation domain

A method and an apparatus for synthesizing an audio signal are described. A spectral tilt is applied to the code of a codebook used for synthesizing a current frame of the audio signal. The spectral tilt is based on the spectral tilt of the current frame of the audio signal. Further, an audio decoder operating in accordance with the inventive approach is described.

Apparatus and method for synthesizing an audio signal, decoder, encoder, system and computer program

An information encoder for encoding an information signal includes: a converter for converting the linear prediction coefficients of the predictive polynomial A(z) to frequency values f. . . fof a spectral frequency representation of the predictive polynomial A(z), wherein the converter is configured to determine the frequency values f. . . fby analyzing a pair of polynomials P(z) and Q(z) being defined as wherein m is an order of the predictive polynomial A(z) and I is greater or equal to zero, wherein the converter is configured to obtain the frequency values by establishing a strictly real spectrum derived from P(z) and a strictly imaginary spectrum from Q(z) and by identifying zeros of the strictly real spectrum derived from P(z) and the strictly imaginary spectrum derived from Q(z).

Concept for encoding of information

An apparatus for generating an audio output signal having two or more audio output channels from an audio input signal having two or more audio input channels includes a provider and a signal processor. The provider is adapted to provide first covariance properties of the audio input signal. The signal processor is adapted to generate the audio output signal by applying a mixing rule on at least two of the two or more audio input channels. The signal processor is configured to determine the mixing rule based on the first covariance properties of the audio input signal and based on second covariance properties of the audio output signal, the second covariance properties being different from the first covariance properties.

Optimal mixing matrices and usage of decorrelators in spatial audio processing

An apparatus for encoding a speech signal by determining a codebook vector of a speech coding algorithm is provided. The apparatus includes a matrix determiner for determining an autocorrelation matrix R, and a codebook vector determiner for determining the codebook vector depending on the autocorrelation matrix R. The matrix determiner is configured to determine the autocorrelation matrix R by determining vector coefficients of a vector r, wherein the autocorrelation matrix R includes a plurality of rows and a plurality of columns, wherein the vector r indicates one of the columns or one of the rows of the autocorrelation matrix R, wherein R(i, j)=r(|i&#x2212;j|), wherein R(i, j) indicates the coefficients of the autocorrelation matrix R, wherein i is a first index indicating one of a plurality of rows of the autocorrelation matrix R, and wherein j is a second index indicating one of the plurality of columns of the autocorrelation matrix R.

An information encoder for encoding an information signal includes: a converter for converting the linear prediction coefficients of the predictive polynomial A(z) to frequency values f. . . fof a spectral frequency representation of the predictive polynomial A(z), wherein the converter is configured to determine the frequency values f. . . fby analyzing a pair of polynomials P(z) and Q(z) being defined as()=()+() and()=()&#x2212;(),wherein m is an order of the predictive polynomial A(z) and l is greater or equal to zero, wherein the converter is configured to obtain the frequency values by establishing a strictly real spectrum derived from P(z) and a strictly imaginary spectrum from Q(z) and by identifying zeros of the strictly real spectrum derived from P(z) and the strictly imaginary spectrum derived from Q(z).

An apparatus for generating an audio output signal is provided. The audio output signal has two or more audio output channels and is generated from an audio input signal having two or more audio input channels. The apparatus includes a provider and a signal processor. The provider is adapted to provide first covariance properties of the audio input signal. The signal processor is adapted to generate the audio output signal by applying a mixing rule on at least two of the two or more audio input channels. The signal processor is configured to determine the mixing rule based on the first covariance properties of the audio input signal and based on second covariance properties of the audio output signal, the second covariance properties being different from the first covariance properties.

An apparatus for generating an audio signal envelope from one or more coding values is provided. The apparatus includes an input interface for receiving the one or more coding values, and an envelope generator for generating the audio signal envelope depending on the one or more coding values. The envelope generator is configured to generate an aggregation function depending on the one or more coding values, wherein the aggregation function includes a plurality of aggregation points. Furthermore, the envelope generator is configured to generate the audio signal envelope such that the envelope value of each of the envelope points of the audio signal envelope depends on the aggregation value of at least one aggregation point of the aggregation function.

Apparatus and method for audio signal envelope encoding, processing, and decoding by modelling a cumulative sum representation employing distribution quantization and coding

An audio encoder for providing an encoded representation on the basis of an audio signal, wherein the audio encoder is configured to obtain a noise information describing a noise included in the audio signal, and wherein the audio encoder is configured to adaptively encode the audio signal in dependence on the noise information, such that encoding accuracy is higher for parts of the audio signal that are less affected by the noise included in the audio signal than for parts of the audio signal that are more affected by the noise included in the audio signal.

Method for adaptively encoding an audio signal in dependence on noise information for higher encoding accuracy

An encoder for encoding an audio signal has a predictor, a factorizer, a transformer and a quantize and encode stage. The predictor is configured to analyze the audio signal to obtain prediction coefficients describing a spectral analog of the audio signal or a fundamental frequency of the audio signal and subject the audio signal to an analysis filter function dependent on the prediction coefficients to output a residual signal of the audio signal. The factorizer is configured to apply a matrix factorization onto an audiocorrelation or covariance matrix of synthesis filter function defined by the prediction coefficients to obtain factorized matrices. The transformer is configured to transform the residual signal based on the factorized matrices to obtain a transformed residual signal. The quantize and decode stage is configured to quantize the transformed residual signal to obtain a quantized transformed residual signal or an encoded quantized transformed residual signal.

Encoder, decoder and method for encoding and decoding

Disclosed is an apparatus for processing an input signal, having a perceptual weighter and a quantizer. The perceptual weighter has a model provider and a model applicator. The model provider provides a perceptual weighted model based on the input signal. The model applicator provides a perceptually weighted spectrum by applying the perceptual weighted model to a spectrum based on the input signal. The quantizer is configured to quantize the perceptually weighted spectrum and for providing a bitstream. The quantizer has a random matrix applicator and a sign function calculator. The random matrix applicator is configured for applying a random matrix to the perceptually weighted spectrum in order to provide a transformed spectrum. The sign function calculator is configured for calculating a sign function of components of the transformed spectrum in order to provide the bitstream. The invention further refers to an apparatus for processing an encoded signal and to corresponding methods.

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Nuremberg, Germany

Tom Baeckstroem