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

Manuel Jander

Markus Schnell

Jérémie Lecomte

Markus Lohwasser

Max Neuendorf

Emmanuel Ravelli

Stefan Doehla

Ralf Geiger

Martin Dietz

Matthias Hildenbrand

Tobias Chalupka

Markus Multrus

Christian Helmrich

Ralph Sperschneider

Michael Schnabel

Bernhard Grill

Goran Markovic

Marc Gayer

Manfred Lutzky

Benjamin Schubert

Stefan Reuschl

Michael Schuldt

Matthias Neusinger

Sascha Disch

Nikolaus Rettelbach

Grzegorz Pietrzyk

Guillaume Fuchs

Frederik Nagel

Harald Popp

Ulrich Kraemer

Nikolaus Faerber

Virgilio Bacigalupo

Anthony Lombard

Yoshikazu Yokotani

Stefan Bayer

Bernhard Neugebauer

Stefan Döhla

Stephan Schreiner

Arne Borsum

Nikolaus Farber

Nikolaus Färber

Ulrich KRÄMER

The transient problem may be sufficiently addressed, and for this purpose, a further delay on the side of the decoding may be reduced if a new SBR frame class is used wherein the frame boundaries are not shifted, i.e. the grid boundaries are still synchronized with the frame boundaries, but wherein a transient position indication is additionally used as a syntax element so as to be used, on the encoder and/or decoder sides, within the frames of these new frame class for determining the grid boundaries within these frames.

Encoding an information signal

An apparatus for processing an audio signal having associated therewith a pitch lag information and a gain information, includes a domain converter for converting a first domain representation of the audio signal into a second domain representation of the audio signal; and a harmonic post-filter for filtering the second domain representation of the audio signal, wherein the post-filter is based on a transfer function including a numerator and a denominator, wherein the numerator includes a gain value indicated by the gain information, and wherein the denominator includes an integer part of a pitch lag indicated by the pitch lag information and a multi-tap filter depending on a fractional part of the pitch lag.

Apparatus and method for processing an audio signal using a harmonic post-filter

A method is described that processes an audio signal. A discontinuity between a filtered past frame and a filtered current frame of the audio signal is removed using linear predictive filtering.

Method and apparatus for processing an audio signal, audio decoder, and audio encoder to filter a discontinuity by a filter which depends on two fir filters and pitch lag

An apparatus for processing an audio signal includes a configurable first audio signal processor for processing the audio signal in accordance with different configuration settings to obtain a processed audio signal, wherein the apparatus is adapted so that different configuration settings result in different sampling rates of the processed audio signal. The apparatus furthermore includes n analysis filter bank having a first number of analysis filter bank channels, a synthesis filter bank having a second number of synthesis filter bank channels, a second audio processor being adapted to receive and process an audio signal having a predetermined sampling rate, and a controller for controlling the first number of analysis filter bank channels or the second number of synthesis filter bank channels in accordance with a configuration setting.

Resampling output signals of QMF based audio codecs

An audio decoder provides a decoded audio information on the basis of an encoded audio information including linear prediction coefficients (LPC) and includes a tilt adjuster to adjust a tilt of a noise using linear prediction coefficients of a current frame to acquire a tilt information and a noise inserter configured to add the noise to the current frame in dependence on the tilt information. Another audio decoder includes a noise level estimator to estimate a noise level for a current frame using a linear prediction coefficient of at least one previous frame to acquire a noise level information; and a noise inserter to add a noise to the current frame in dependence on the noise level information provided by the noise level estimator. Thus, side information about a background noise in the bit-stream may be omitted. Methods and computer programs serve a similar purpose.

Noise filling without side information for CELP-like coders

A method is described that processes an audio signal. A discontinuity between a filtered past frame and a filtered current frame of the audio signal is removed using linear predictive filter. Removing the discontinuity further comprises processing a beginning portion of the filtered current frame, wherein the beginning portion of the current frame comprises a predefined number of samples being less or equal than a total number of samples in the current frame, and wherein processing the beginning portion of the current frame comprises subtracting a beginning portion of a zero-input-response (ZIR) from the beginning portion of the filtered current frame.

Method and apparatus for processing an audio signal, audio decoder, and audio encoder for removing a discontinuity between frames by subtracting a portion of a zero-input-reponse

A time scaler for providing a time scaled version of an input audio signal is configured to compute or estimate a quality of a time scaled version of the input audio signal obtainable by a time scaling of the input audio signal. The time scaler is configured to perform the time scaling of the input audio signal in dependence on the computation or estimation of the quality of the time scaled version of the input audio signal obtainable by the time scaling. An audio decoder has such a time scaler.

Time scaler, audio decoder, method and a computer program using a quality control

Resampling output signals of QMF based audio codec

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Manuel Jander