Abstract: An acoustic echo cancellation unit includes: an audio processor configured to receive a multichannel audio signal and including: a first stage which is configured to process the multichannel audio signal to obtain a first set of spatial audio components; and a second stage which is configured to process the first set of spatial audio components to obtain a second set of spatial audio components; an echo cancellation processor configured to perform echo cancellation by use of the first set of spatial audio components or a deviated version of the first set of spatial audio components as reference signal.

Abstract: An apparatus for processing an information signal has: a feature extractor for extracting a set of features from the information signal, wherein the feature extractor has: a raw feature calculator for calculating raw feature results, each raw feature result having at least two raw feature components; and a raw feature compressor for performing a compression of a dynamic range to the at least two raw feature components to obtain at least two compressed raw feature components for each raw feature result, wherein the set of features has the compressed raw feature components; and a signal processor for processing the set of features to obtain the processed information signal, wherein the information signal has an audio signal, an image signal, or a radar signal.

Abstract: An apparatus for processing an information signal has: a feature extractor for extracting a set of features from the information signal, each feature having at least two feature components, and the set of features having a first subset with the first feature components and a second subset with the second feature components; and a neural network processor having: a first neural network for receiving, as input, the first subset and outputting a processed first subset; a combiner for combining the processed first subset and the second subset to obtain a combined subset; and a second neural network for receiving, as input, the combined subset and outputting a processed combined output, which represents a processed information signal, or the apparatus being configured to calculate the processed information signal using the processed combined output, and a complexity of the first neural network being greater than that of the second neural network.

Abstract: An apparatus for processing an information signal has: a feature extractor for extracting a set of features having a first dimension; a feature segmenter for segmenting into a first subset having a second dimension and a second subset having a third dimension, which overlap, both being lower than the first dimension; a neural network processor for processing the first and second subsets using a first and a second neural network to obtain a first and a second result, respectively; a feature combiner for combining the first and second results using a third neural network, having a third complexity lower than a first or a second complexity of the first and second neural network to obtain a result set of features having a result dimension; and an output post-processor for post-processing the result set of features to obtain a processed information signal.

Abstract: An apparatus for interference cancellation according to an embodiment is provided. The apparatus comprises a preprocessor configured for resampling a first audio signal to obtain a sampling-rate-adjusted first signal. Moreover, the apparatus comprises an interference estimator configured for estimating a first interference estimate depending on a first filter configuration and depending on the sampling-rate-adjusted first signal; and configured for estimating a second interference estimate depending on a second filter configuration and depending on a second audio signal.

Abstract: An apparatus for conducting acoustic echo cancellation by generating one or more error signals is provided. The apparatus is to generate a first echo estimate including one or more first echo estimation signals by filtering one or more reference signals using a first filter configuration, wherein the one or more reference signals correspond to one or more loudspeaker signals or are derived from the one or more loudspeaker signals. Moreover, the apparatus is to generate a second echo estimate including one or more second echo estimation signals by filtering the one or more first echo estimation signals using a second filter configuration; or, the apparatus is to generate the second echo estimate including the one or more second echo estimation signals by generating an intermediate signal from the one or more first estimation signals, and by filtering the intermediate signal using the second filter configuration.

Abstract: An apparatus for processing an audio signal includes an audio signal analyzer and a filter. The audio signal analyzer is configured to analyze an audio signal to determine a plurality of noise suppression filter values for a plurality of bands of the audio signal, wherein the analyzer is configured to determine a noise suppression filter value so that a noise suppression filter value is greater than or equal to a minimum noise suppression filter value and so that the minimum noise suppression value depends on a characteristic of the audio signal. The filter is configured for filtering the audio signal, wherein the filter is adjusted based on the noise suppression filter values.

Abstract: An acoustic echo cancellation unit includes: an audio processor configured to receive a multichannel audio signal and including: a first stage which is configured to process the multichannel audio signal to obtain a first set of spatial audio components; and a second stage which is configured to process the first set of spatial audio components to obtain a second set of spatial audio components; an echo cancellation processor configured to perform echo cancellation by use of the first set of spatial audio components or a deviated version of the first set of spatial audio components as reference signal.

Abstract: An apparatus for processing an audio signal includes an audio signal analyzer and a filter. The audio signal analyzer is configured to analyze an audio signal to determine a plurality of noise suppression filter values for a plurality of bands of the audio signal, wherein the analyzer is configured to determine a noise suppression filter value so that a noise suppression filter value is greater than or equal to a minimum noise suppression filter value and so that the minimum noise suppression value depends on a characteristic of the audio signal. The filter is configured for filtering the audio signal, wherein the filter is adjusted based on the noise suppression filter values.

Abstract: An apparatus for processing an audio signal includes an audio signal analyzer and a filter. The audio signal analyzer is configured to analyze an audio signal to determine a plurality of noise suppression filter values for a plurality of bands of the audio signal, wherein the analyzer is configured to determine a noise suppression filter value so that a noise suppression filter value is greater than or equal to a minimum noise suppression filter value and so that the minimum noise suppression value depends on a characteristic of the audio signal. The filter is configured for filtering the audio signal, wherein the filter is adjusted based on the noise suppression filter values.

Abstract: An apparatus for processing an input audio signal includes an evaluator for evaluating a loudness of the input audio signal to determine a loudness range and an actual loudness value. A calculator determines a compressor transfer function based on the determined loudness range, based on a target loudness range and based on the determined actual loudness value. The calculator further determines at least one loudness range control gain based on the determined compressor transfer function. An adjuster provides an output audio signal based on the input audio signal and based on the at least one determined loudness range control gain. The calculator is configured to determine the compressor transfer function such that a difference between a mean loudness of the output audio signal and a mean loudness of the input audio signal is minimized. A corresponding method is also provided.

Abstract: An apparatus for processing an input audio signal includes an evaluator for evaluating a loudness of the input audio signal to determine a loudness range and an actual loudness value. A calculator determines a compressor transfer function based on the determined loudness range, based on a target loudness range and based on the determined actual loudness value. The calculator further determines at least one loudness range control gain based on the determined compressor transfer function. An adjuster provides an output audio signal based on the input audio signal and based on the at least one determined loudness range control gain. The calculator is configured to determine the compressor transfer function such that a difference between a mean loudness of the output audio signal and a mean loudness of the input audio signal is minimized. A corresponding method is also provided.

Abstract: A partitioned block frequency domain adaptive filter device includes a frequency domain adaptive filter configured for filtering a frequency domain representation of a time domain input signal depending on a set of filter coefficients consisting of a plurality of blocks of filter coefficients in order to produce a filtered signal; a plurality of parallel arranged filter update blocks; wherein each of the filter update blocks includes an adaptation module configured for executing an adaptation sequence including the steps of calculating an approximation of a constrained gradient update for the filter coefficients of the respective block of filter coefficients, and calculating a cumulative error introduced on the unconstrained gradient update; wherein each of the filter update blocks includes a correction module configured for executing a correction sequence including the steps of calculating a corrected constrained gradient update for the filter coefficients of the respective block of filter coefficients.

Abstract: An apparatus for processing an audio signal includes an audio signal analyzer and a filter. The audio signal analyzer is configured to analyze an audio signal to determine a plurality of noise suppression filter values for a plurality of bands of the audio signal, wherein the analyzer is configured to determine a noise suppression filter value so that a noise suppression filter value is greater than or equal to a minimum noise suppression filter value and so that the minimum noise suppression value depends on a characteristic of the audio signal. The filter is configured for filtering the audio signal, wherein the filter is adjusted based on the noise suppression filter values.

Abstract: A partitioned block frequency domain adaptive filter device includes a frequency domain adaptive filter configured for filtering a frequency domain representation of a time domain input signal depending on a set of filter coefficients consisting of a plurality of blocks of filter coefficients in order to produce a filtered signal; a plurality of parallel arranged filter update blocks; wherein each of the filter update blocks includes an adaptation module configured for executing an adaptation sequence including the steps of calculating an approximation of a constrained gradient update for the filter coefficients of the respective block of filter coefficients, and calculating a cumulative error introduced on the unconstrained gradient update; wherein each of the filter update blocks includes a correction module configured for executing a correction sequence including the steps of calculating a corrected constrained gradient update for the filter coefficients of the respective block of filter coefficients.