Abstract: 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 f1 . . . fn of a spectral frequency representation of the predictive polynomial A(z), wherein the converter is configured to determine the frequency values f1 . . . fn by analyzing a pair of polynomials P(z) and Q(z) being defined as P ? ( z ) = A ? ( z ) + z - m - l ? A ? ( z - 1 ) ? ? and Q ? ( z ) = A ? ( z ) - z - m - l ? A ? ( z - 1 ) , 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).

Abstract: 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.

Abstract: 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.

Abstract: 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.

Abstract: An apparatus for encoding an audio input signal to obtain an encoded audio signal is provided. The apparatus comprises a transformation module configured to transform the audio input signal from an original domain to a transform domain to obtain a transformed audio signal. Moreover, the apparatus comprises an encoding module, configured to quantize the transformed audio signal to obtain a quantized signal, and configured to encode the quantized signal to obtain the encoded audio signal. The transformation module is configured to transform the audio input signal depending on a plurality of predefined power values of quantization noise in the original domain.

Abstract: 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.

Abstract: 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.

Abstract: 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 f1 . . . fn of a spectral frequency representation of the predictive polynomial A(z), wherein the converter is configured to determine the frequency values f1 . . . fn by analyzing a pair of polynomials P(z) and Q(z) being defined as P ? ( z ) = A ? ( z ) + z - m - l ? A ? ( z - 1 ) ? ? and Q ? ( z ) = A ? ( z ) - z - m - l ? A ? ( z - 1 ) , 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).

Abstract: There are provided examples of decoders and decoding methods. One decoder includes: a bitstream reader to provide a version of an input signal as a sequence of frames, each frame subdivided into a plurality of bins, each bin having a sampled value; a context definer to define a context for one bin under process, the context including at least one additional bin in a predetermined positional relationship with the bin under process; a statistical relationship and information estimator to provide statistical relationships between the bin under process and the at least one additional bin; and a value estimator to process and acquire an estimate of the value of the bin. There is included a noise relationship and information estimator providing statistical relationships and information regarding noise, which includes a noise matrix estimating relationships among noise signals among the bin under process and the at least one additional bin.

Abstract: 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 f1 . . . fn of a spectral frequency representation of the predictive polynomial A(z), wherein the converter is configured to determine the frequency values f1 . . . fn by analyzing a pair of polynomials P(z) and Q(z) being defined as P(z)=A(z)+z?m?lA(z?1) and Q(z)=A(z)?z?m?lA(z?1), 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).

Abstract: 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.

Abstract: 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.

Abstract: There are provided examples of decoders and decoding methods. One decoder includes: a bitstream reader to provide a version of an input signal as a sequence of frames, each frame subdivided into a plurality of bins, each bin having a sampled value; a context definer to define a context for one bin under process, the context including at least one additional bin in a predetermined positional relationship with the bin under process; a statistical relationship and information estimator to provide statistical relationships between the bin under process and the at least one additional bin; and a value estimator to process and acquire an estimate of the value of the bin. There is included a noise relationship and information estimator providing statistical relationships and information regarding noise, which includes a noise matrix estimating relationships among noise signals among the bin under process and the at least one additional bin.

Abstract: 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.

Abstract: It is shown an encoder for encoding an audio signal with reduced background noise using linear predictive coding. The encoder includes a background noise estimator configured to estimate background noise of the audio signal, a background noise reducer configured to generate background noise reduced audio signal by subtracting the estimated background noise of the audio signal from the audio signal, and a predictor configured to subject the audio signal to linear prediction analysis to obtain a first set of linear prediction filter (LPC) coefficients and to subject the background noise reduced audio signal to linear prediction analysis to obtain a second set of linear prediction filter (LPC) coefficients. Furthermore, the encoder includes an analysis filter composed of a cascade of time-domain filters controlled by the obtained first set of LPC coefficients and the obtained second set of LPC coefficients.

Abstract: 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.

Abstract: An apparatus for encoding an audio input signal to obtain an encoded audio signal is provided. The apparatus comprises a transformation module configured to transform the audio input signal from an original domain to a transform domain to obtain a transformed audio signal. Moreover, the apparatus comprises an encoding module, configured to quantize the transformed audio signal to obtain a quantized signal, and configured to encode the quantized signal to obtain the encoded audio signal. The transformation module is configured to transform the audio input signal depending on a plurality of predefined power values of quantization noise in the original domain.

Abstract: 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.

Abstract: 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.

Abstract: 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 f1 . . . fn of a spectral frequency representation of the predictive polynomial A(z), wherein the converter is configured to determine the frequency values f1 . . . fn by analyzing a pair of polynomials P(z) and Q(z) being defined as P(z)=A(z)+z?m?lA(z?1) and Q(z)=A(z)?z?m?lA(z?1), 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).