Fraunhofer-gesellschaft Zur Foerderung Der Angewandten Forschung E.v.

Bernhard Neugebauer

Michael Kratschmer

Michael Meier

Christian Uhle

Jan Plogsties

Jérémie Lecomte

Martin Dietz

Goran Markovic

Michael Schnabel

Fabian Kuech

Fabian Küch

Juha Vilkamo

Arne Borsum

Stephan Schreiner

Andreas Silzle

Juergen Herre

Johannes Hilpert

Matthias Neusinger

Jürgen Herre

Fabian Kuch

Manuel Jander

Harald Popp

Harald Mundt

Markus Lohwasser

Emanuel Habets

Christian Borss

Hanne Stenzel

Felix Fleischmann

Edwin Mabande

Sebastian Schlecht

An apparatus for determining an estimated pitch lag is provided. The apparatus includes an input interface for receiving a plurality of original pitch lag values, and a pitch lag estimator for estimating the estimated pitch lag. The pitch lag estimator is configured to estimate the estimated pitch lag depending on a plurality of original pitch lag values and depending on a plurality of information values, wherein for each original pitch lag value of the plurality of original pitch lag values, an information value of the plurality of information values is assigned to the original pitch lag value.

Apparatus and method for improved concealment of the adaptive codebook in a CELP-like concealment employing improved pitch lag estimation

An audio processor for processing an audio signal includes: an audio signal modifier for modifying the audio signal in response to a user input; a loudness controller for determining a loudness compensation gain based on a reference loudness or a reference gain and a modified loudness or a modified gain, where the modified loudness or the modified gain depends on the user input; and a loudness manipulator for manipulating a loudness of a signal using the loudness compensation gain.

Loudness control for user interactivity in audio coding systems

The invention provides a concept for combined dynamic range compression and guided clipping prevention for audio devices. An audio decoder for decoding an audio bitstream and a metadata bitstream related to the audio bitstream according to the concept includes an audio processing chain including a plurality of adjustment stages including a dynamic range control stage for adjusting a dynamic range of the audio output signal and a guided clipping prevention stage for preventing clipping of the audio output signal; and a metadata decoder configured to receive the metadata bitstream and to extract dynamic range control gain sequences and guided clipping prevention gain sequences from the metadata bitstream, at least a part of the dynamic range control gain sequences being supplied to the dynamic range control stage, and at least a part of the guided clipping prevention gain sequences being supplied to the guided clipping prevention stage.

Concept for combined dynamic range compression and guided clipping prevention for audio devices

Apparatus and method for improved concealment of the adaptive codebook in ACELP-like concealment employing improved pitch lag estimation

An audio encoder device includes an audio encoder configured for producing an encoded audio bitstream from an audio signal having consecutive audio frames; a dynamic range control encoder configured for producing an encoded dynamic range control bitstream fromanadynamic range control sequence corresponding to the audio signal and having consecutive dynamic range control frames, wherein each dynamic range control frame of the dynamic range control frames has one or more nodes, wherein each node of the one or more nodes has gain information for the audio signal and time information indicating to which point in time the gain information corresponds.

Audio encoder device and an audio decoder device having efficient gain coding in dynamic range control

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.

Apparatus for processing an input audio signal and corresponding method

An equalization filter coefficient determinator continuously or quasi-continuously fades between a plurality of different equalizer settings in dependence on one or more setting parameters, to obtain a current set of equalization filter target coefficients describing a current equalizer setting. A number of setting parameters is smaller than a number of equalization filter target coefficients of current set of equalization filter target coefficients. An equalization filter coefficient determinator linearly combines a plurality of equalization filter target coefficient set components in dependence on one or more setting parameters. An equalization filter coefficient determinator obtains the current set of equalization filter target coefficients in dependence on position information obtained using a two-dimensional or three-dimensional user input device. An apparatus includes a user interface, an equalization filter coefficient determinator and an equalizer. An equalization filter coefficient processor may provide sets of basis equalization filter target coefficients. A system is also described.

Equalization filter coefficient determinator, apparatus, equalization filter coefficient processor, system and methods

An apparatus for reconstructing a frame including a speech signal as a reconstructed frame is provided, the apparatus including a determination unit and a frame reconstructor being configured to reconstruct the reconstructed frame, such that the reconstructed frame completely or partially includes the first reconstructed pitch cycle, such that the reconstructed frame completely or partially includes a second reconstructed pitch cycle, and such that the number of samples of the first reconstructed pitch cycle differs from a number of samples of the second reconstructed pitch cycle.

Apparatus and method for improved concealment of the adaptive codebook in ACELP-like concealment employing improved pulse resynchronization

An audio encoder device includes an audio encoder configured for producing an encoded audio bitstream from an audio signal having consecutive audio frames; a dynamic range control encoder configured for producing an encoded dynamic range control bitstream from an dynamic range control sequence corresponding to the audio signal and having consecutive dynamic range control frames, wherein each dynamic range control frame of the dynamic range control frames has one or more nodes, wherein each node of the one or more nodes has gain information for the audio signal and time information indicating to which point in time the gain information corresponds.

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Erlangen, Germany

Bernhard Neugebauer