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

Dolby International Ab

Christian Helmrich

Sascha Disch

Bernd Edler

Goran Markovic

Ralf Geiger

Frederik Nagel

Johannes Hilpert

Konstantin Schmidt

Jérémie Lecomte

Ralph Sperschneider

Sascha Dick

Guillaume Fuchs

Nikolaus Rettelbach

Christian Neukam

Andreas Hoelzer

Michael Schnabel

Stefan Bayer

Matthias Neusinger

Markus Multrus

Martin Dietz

Emmanuel Ravelli

Christian Ertel

Max Neuendorf

Markus Schnell

Stefan Doehla

Manuel Jander

Thomas Wiegand

Detlev Marpe

Heiko Schwarz

Achim Kuntz

Julien Robilliard

Benjamin Bross

Benjamin Schubert

Bernhard Grill

Marc Gayer

Eleni Fotopoulou

Andreas Niedermeier

Florin Ghido

Wolfgang Jaegers

Tung Nguyen

Mischa Siekmann

Lars Villemoes

Heiko Purnhagen

Juergen Herre

Valeri George

Jürgen Herre

Pontus Carlsson

Tobias Hinz

Balaji Nagendran Thoshkahna

Maria Luis Valero

Dominik Zorn

Stefan Reuschl

Tom Baeckstroem

Tom Backstrom

Andreas Holzer

Michael Fischer

Florian Schuh

Paul Keydel

Phan Hoang Tung Nguyen

Sebastian Bosse

Richard Fueg

Janine Sukowski

Stefan Döhla

Adam Wieckowski

Yoshikazu Yokotani

Arthur Tritthart

Julien Robillard

Markovic Goran

Stephan Wilde

Johannes Fischer

Tobias Schwegler

Vesa Ruoppila

Andreas Hölzer

Tom Bäckström

Richard Füg

Johannes Erfurt

Soeren Becker

Tung Hoang Nguyen

Wolfgang JÄGERS

Tom BÄCKSTRÖM

A device and method for manipulating an audio signal includes a windower for generating a plurality of consecutive blocks of audio samples, the plurality of consecutive blocks including at least one padded block of audio samples, the padded block having padded values and audio signal values. a first converter for converting the padded block into a spectral representation having spectral values, a phase modifier for modifying phases of the spectral values to obtain a modified spectral representation and a second converter for converting the modified spectral representation into a modified time domain audio signal.

Device and method for manipulating an audio signal

A frequency-domain audio codec is provided with the ability to additionally support a certain transform length in a backward-compatible manner, by the following: the frequency-domain coefficients of a respective frame are transmitted in an interleaved manner irrespective of the signalization signaling for the frames as to which transform length actually applies, and additionally the frequency-domain coefficient extraction and the scale factor extraction operate independent from the signalization. By this measure, old-fashioned frequency-domain audio coders/decoders, insensitive for the signalization, would be able to nevertheless operate without faults and with reproducing a reasonable quality. Concurrently, frequency-domain audio coders/decoders able to support the additional transform length would offer even better quality despite the backward compatibility. As far as coding efficiency penalties due to the coding of the frequency domain coefficients in a manner transparent for older decoders are concerned, same are of comparatively minor nature due to the interleaving.

Frequency-domain audio coding supporting transform length switching

A device and method for manipulating an audio signal includes a windower for generating a plurality of consecutive blocks of audio samples, the plurality of consecutive blocks including at least one padded block of audio samples, the padded block having padded values and audio signal values, a first converter for converting the padded block into a spectral representation having spectral values, a phase modifier for modifying phases of the spectral values to obtain a modified spectral representation and a second converter for converting the modified spectral representation into a modified time domain audio signal.

A picture in a data stream is subdivided into different size transform bocks. For a transform block, a transformation is selected by checking whether the block's size exceeds a threshold. If it does, a default technique is used to select the transformation. If it does not, a transformation is used from a list of transformations as identified by an index, the list including an identity transformation and non-identity transformations. A coefficients block in the data stream is decoded, and if the selected transformation is the identity transformation, then the coefficients block is used as a residual sample array, and if the selected transformation is not the identity transformation, the coefficients block is subjected to an inverse transformation corresponding to the selected transformation to obtain a residual sample array used to correct a prediction for the transform block.

Unified transform type signaling and transform type dependent transform coefficient level coding

An apparatus for decoding an encoded multichannel signal of a current frame to obtain three or more current audio output channels is provided. A multichannel processor is adapted to select two decoded channels from three or more decoded channels depending on first multichannel parameters. Moreover, the multichannel processor is adapted to generate a first group of two or more processed channels based on the selected channels. A noise filling module is adapted to identify for at least one of the selected channels, one or more frequency bands, within which all spectral lines are quantized to zero, and to generate a mixing channel using, depending on side information, a proper subset of three or more previous audio output channels that have been decoded, and to fill the spectral lines of frequency bands, within which all spectral lines are quantized to zero, with noise generated using spectral lines of the mixing channel.

Apparatus and method for stereo filling in multichannel coding

An audio decoder for providing at least four audio channel signals on the basis of an encoded representation is configured to provide a first residual signal and a second residual signal on the basis of a jointly encoded representation of the first residual signal and of the second residual signal using a multi-channel decoding. The audio decoder is configured to provide a first audio channel signal and a second audio channel signal on the basis of a first downmix signal and the first residual signal using a residual-signal-assisted multi-channel decoding. The audio decoder is configured to provide a third audio channel signal and a fourth audio channel signal on the basis of a second downmix signal and the second residual signal using a residual-signal-assisted multi-channel decoding. An audio encoder is based on corresponding considerations.

Audio encoder, audio decoder, methods and computer program using jointly encoded residual signals

An apparatus processing an encoded signal to acquire first and second frames comprising spectral values and an aliasing portion; applying a transform to the first frame using a first window function to acquire a first block of samples, applying another transform to a first portion of the second frame using a second window function, and applying another one or more transforms to a second portion of the second frame using one or more third window functions to acquire a second block of samples; and post-processing the second block of samples using a folding-out operation to acquire a post-processed second block of samples comprising a portion of the second block of samples overlapping with the first block of samples in a multi-overlap region, windowing the post-processed second block of samples using an auxiliary window function, and overlap-adding a windowed post-processed second block of samples and the first block of samples.

Apparatus and method for decoding an encoded audio or image signal using an auxiliary window function

In multichannel audio coding, an improved coding efficiency is achieved by the following measure: the noise filling of zero-quantized scale factor bands is performed using noise filling sources other than artificially generated noise or spectral replica. In particular, the coding efficiency in multichannel audio coding may be rendered more efficient by performing the noise filling based on noise generated using spectral lines from a previous frame of, or a different channel of the current frame of, the multichannel audio signal.

Noise filling in multichannel audio coding

A schematic block diagram of a decoder for decoding an encoded audio signal is shown. The decoder includes an adaptive spectrum-time converter and an overlap-add-processor. The adaptive spectrum-time converter converts successive blocks of spectral values into successive blocks of time values, e.g. via a frequency-to-time transform. Furthermore, the adaptive spectrum-time converter receives a control information and switches, in response to the control information, between transform kernels of a first group of transform kernels including one or more transform kernels having different symmetries at sides of a kernel, and a second group of transform kernels including one or more transform kernels having the same symmetries at sides of a transform kernel. Moreover, the overlap-add-processor overlaps and adds the successive blocks of time values to obtain decoded audio values, which may be a decoded audio signal.

Decoder for decoding an encoded audio signal and encoder for encoding an audio signal

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Erlangen, Germany

Christian Helmrich