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

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Dolby Laboratories Licensing Corporation

Technische Universitaet Ilmenau

Technische Universitat Ilmenau

Martin Dietz

Emmanuel Ravelli

Guillaume Fuchs

Markus Multrus

Goran Markovic

Markus Schnell

Benjamin Schubert

Matthias Neusinger

Eleni Fotopoulou

Sascha Disch

Christian Helmrich

Stefan Doehla

Manuel Jander

Wolfgang Jaegers

Bernhard Grill

Ralph Sperschneider

Michael Schnabel

Jérémie Lecomte

Stefan Bayer

Karlheinz Brandenburg

Anthony Lombard

Oliver Kunz

Bernhard Neugebauer

Stefan Döhla

Jan Büthe

Uwe Gbur

Konstantin Schmidt

Marc Gayer

Tom Baeckstroem

Grzegorz Pietrzyk

Heinz Gerhauser

J&uuml;rgen Herre

Daniel Homm

Tom Backstrom

Juergen Herre

Christian Neukam

Conrad Benndorf

Srikanth Korse

Stefan Reuschl

Bodo Teichmann

Andreas Ehret

Pierre Lauber

Reinhold Böhm

Ralf Geiger

Manfred Lutzky

Stephan Wilde

Adrian Tomasek

Bernd Edler

James David Johnston

Jürgen Herre

Michael Schug

Panji Setiawan

Franz Reutelhuber

Heinz Gerh&auml;user

Pallavi Maben

Maximilian Schlegel

Rainer Buchta

Martin Sieler

Ali Nowbakht-Irani

Charles Quito Robinson

Grant Allen Davidson

Thomas Sporer

Schuyler Reynier Quackenbush

Heinz Gerhaeuser

Wolfgang Alexander Schildbach

Jan Buethe

Alexander Tschekalinskij

Roland Bitto

Arthur Tritthart

Tom Bäckström

J&uuml;rgen Zeller

Reinhold Boehm

Marina Bosi

Jurgen Zeller

Tom BÄCKSTRÖM

Wolfgang JÄGERS

Jurgen Herre

Jeremie Lecomte

A decoder for decoding a frame to reconstruct a signal portion of a signal is provided. The signal portion is encoded within the frame, wherein the frame includes a bitstream payload and two or more redundancy bits, wherein the bitstream payload includes a plurality of payload bits, wherein each of the payload bits exhibits a position within the bitstream payload. The decoder includes a channel decoding module, being configured to detect, depending on the two or more redundancy bits, whether the bitstream payload includes one or more corrupted bits being one or more of the payload bits that are distorted or that are likely to be distorted. Moreover, the decoder includes a source decoding module.

Decoder and decoding method selecting an error concealment mode, and encoder and encoding method

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

Described are an encoder for coding speech-like content and/or general audio content, wherein the encoder is configured to embed, at least in some frames, parameters in a bitstream, which parameters enhance a concealment in case an original frame is lost, corrupted or delayed, and a decoder for decoding speech-like content and/or general audio content, wherein the decoder is configured to use parameters which are sent later in time to enhance a concealment in case an original frame is lost, corrupted or delayed, as well as a method for encoding and a method for decoding.

Encoder, decoder and method for encoding and decoding audio content using parameters for enhancing a concealment

An apparatus for estimating an inter-channel time difference between a first channel signal and a second channel signal, includes a signal analyzer for estimating a signal characteristic of the first channel signal or the second channel signal or both signals or a signal derived from the first channel signal or the second channel signal; a calculator for calculating a cross-correlation spectrum for a time block from the first channel signal in the time block and the second channel signal in the time block; a weighter for weighting a smoothed or non-smoothed cross-correlation spectrum to obtain a weighted cross correlation spectrum using a first weighting procedure or using a second weighting procedure depending on a signal characteristic estimated by the signal analyzer, wherein the first weighting procedure is different from the second weighting procedure; and a processor for processing the weighted cross-correlation spectrum to obtain the inter-channel time difference.

Apparatus, method or computer program for estimating an inter-channel time difference

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

An audio encoder for encoding audio input data has: a preprocessor for preprocessing the audio input data to obtain audio data to be coded; a coder processor for coding the audio data to be coded; and a controller for controlling the coder processor so that, depending on a first signal characteristic of a first frame of the audio data to be coded, a number of audio data items of the audio data to be coded by the coder processor for the first frame is reduced compared to a second signal characteristic of a second frame, and a first number of information units used for coding the reduced number of audio data items for the first frame is stronger enhanced compared to a second number of information units for the second frame.

Audio encoder with a signal-dependent number and precision control, audio decoder, and related methods and computer programs

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 audio encoder for encoding an audio signal has: a first encoding processor for encoding a first audio signal portion in a frequency domain, having: a time frequency converter for converting the first audio signal portion into a frequency domain representation; an analyzer for analyzing the frequency domain representation to determine first spectral portions to be encoded with a first spectral resolution and second regions to be encoded with a second resolution; and a spectral encoder for encoding the first spectral portions with the first spectral resolution and encoding the second portions with the second resolution; a second encoding processor for encoding a second different audio signal portion in the time domain; a controller for analyzing and determining, which portion of the audio signal is the first audio signal portion encoded in the frequency domain and which portion is the second audio signal portion encoded in the time domain; and an encoded signal former for forming an encoded audio signal having a first encoded signal portion for the first audio signal portion and a second encoded signal portion for the second portion.

Audio encoder and decoder using a frequency domain processor with full-band gap filling and a time domain processor

A codec allowing for switching between different coding modes is improved by, responsive to a switching instance, performing temporal smoothing and/or blending at a respective transition.

Concept for coding mode switching compensation

FIG.illustrates a decoder for decoding a current frame to reconstruct an audio signal according to an embodiment. The audio signal is encoded within the current frame. The current frame includes a current bitstream payload. The current bitstream payload includes a plurality of payload bits. The plurality of payload bits encodes a plurality of spectral lines of a spectrum of the audio signal. Each of the payload bits exhibits a position within the current bitstream payload. The decoder includes a decoding module and an output interface. The decoding module is configured to reconstruct the audio signal. The output interface is configured to output the audio signal.

Decoder and decoding method for LC3 concealment including full frame loss concealment and partial frame loss concealment

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

An apparatus for encoding audio information is provided. The apparatus for encoding audio information includes a selector for selecting a comfort noise generation mode from two or more comfort noise generation modes depending on a background noise characteristic of an audio input signal, and an encoding unit for encoding the audio information, wherein the audio information includes mode information indicating the selected comfort noise generation mode.

Apparatus and method for comfort noise generation mode selection

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.

Apparatus for encoding a speech signal employing ACELP in the autocorrelation domain

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Nuremberg, Germany

Martin Dietz