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

Goran Markovic

Christian Helmrich

Jérémie Lecomte

Michael Schnabel

Martin Dietz

Emmanuel Ravelli

Markus Schnell

Ralph Sperschneider

Guillaume Fuchs

Stefan Doehla

Eleni Fotopoulou

Wolfgang Jaegers

Stefan Bayer

Bernd Edler

Sascha Disch

Markus Multrus

Manuel Jander

Stefan Reuschl

Grzegorz Pietrzyk

Bernhard Neugebauer

Matthias Neusinger

Marc Gayer

Maria Luis Valero

Ralf Geiger

Janine Sukowski

Sascha Dick

Juergen Herre

Bernhard Grill

Nikolaus Rettelbach

Stefan Geyersberger

Jürgen Herre

Manfred Lutzky

Benjamin Schubert

Stefan Döhla

Franz Reutelhuber

Srikanth Korse

Pallavi Maben

Alexander Tschekalinskij

Jan Frederik Kiene

Wolfgang JÄGERS

An audio quantizer for quantizing a plurality of audio information items has: a first stage vector quantizer for quantizing the plurality of audio information items to determine a first stage vector quantization result and a plurality of intermediate quantized items corresponding to the first stage vector quantization result; a residual item determiner for calculating a plurality of residual items from the plurality of intermediate quantized items and the plurality of audio information items; and a second stage vector quantizer for quantizing the plurality of residual items to obtain a second stage vector quantization result, wherein the first stage vector quantization result and the second stage vector quantization result are a quantized representation of the plurality of audio information items.

Audio quantizer and audio dequantizer and related methods

A multisignal encoder for encoding at least three audio signals, including: a signal preprocessor for individually preprocessing each audio signal to obtain at least three preprocessed audio signals, wherein the preprocessing is performed so that a preprocessed audio signal is whitened with respect to the signal before preprocessing; an adaptive joint signal processor for performing a processing of the at least three preprocessed audio signals to obtain at least three jointly processed signals or at least two jointly processed signals and an unprocessed signal; a signal encoder for encoding each signal to obtain one or more encoded signals; and an output interface for transmitting or storing an encoded multisignal audio signal including the one or more encoded signals, side information relating to the preprocessing and side information relating to the processing.

Multisignal audio coding using signal whitening as processing

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 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

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 apparatus for decoding an encoded audio signal to obtain a reconstructed audio signal is provided. The apparatus includes a receiving interface, a delay buffer and a sample processor for processing the selected audio signal samples to obtain reconstructed audio signal samples of the reconstructed audio signal. The sample selector is configured to select, if a current frame is received by the receiving interface and if the current frame being received by the receiving interface is not corrupted, the plurality of selected audio signal samples from the audio signal samples being stored in the delay buffer depending on a pitch lag information being included by the current frame.

Apparatus and method realizing improved concepts for TCX LTP

An apparatus for estimating an inter-channel time difference between a first channel signal and a second channel signal, includes: 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 spectral characteristic estimator for estimating a characteristic of a spectrum of the first channel signal or the second channel signal for the time block; a smoothing filter for smoothing the cross-correlation spectrum over time using the spectral characteristic to obtain a smoothed cross-correlation spectrum; and a processor for processing the smoothed cross-correlation spectrum to obtain the inter-channel time difference.

Apparatus and method for estimating an inter-channel time difference

An apparatus for decoding an audio signal includes a receiving interface, wherein the receiving interface is configured to receive a first frame and a second frame. Moreover, the apparatus includes a noise level tracing unit for determining noise level information being represented in a tracing domain. Furthermore, the apparatus includes a first reconstruction unit for reconstructing a third audio signal portion of the audio signal depending on the noise level information and a second reconstruction unit for reconstructing a fourth audio signal portion depending on noise level information being represented in the second reconstruction domain.

Apparatus and method for improved signal fade out for switched audio coding systems during error concealment

An apparatus for encoding a first channel and a second channel of an audio input signal including two or more channels to obtain an encoded audio signal according to an embodiment includes a normalizer configured to determine a normalization value for the audio input signal depending on the first channel of the audio input signal and depending on the second channel of the audio input signal. Moreover, the apparatus includes an encoding unit configured to generate a processed audio signal having a first channel and a second channel. The encoding unit is configured to encode the processed audio signal to obtain the encoded audio signal.

Apparatus and method for MDCT M/S stereo with global ILD with improved mid/side decision

The invention provides an audio encoder for encoding an audio signal so as to produce therefrom an encoded signal, the audio encoder including: a framing device configured to extract frames from the audio signal; a quantizer configured to map spectral lines of a spectrum signal derived from the frame of the audio signal to quantization indices, wherein the quantizer has a dead-zone, in which the input spectral lines are mapped to quantization index zero; and a control device configured to modify the dead-zone; wherein the control device includes a tonality calculating device configured to calculate at least one tonality indicating value for at least one spectrum line or for at least one group of spectral lines, wherein the control device is configured to modify the dead-zone for the at least one spectrum line or the at least one group of spectrum lines depending on the respective tonality indicating value.

Low-complexity tonality-adaptive audio signal quantization

An apparatus for generating an encoded signal includes: a window sequence controller for generating a window sequence information for windowing an audio or image signal, the window sequence information indicating a first window for generating a first frame of spectral values, a second window function and at least one third window function for generating a second frame of spectral values, wherein the first window function, the second window function and the one or more third window functions overlap within a multi-overlap region; a preprocessor for windowing a second block of samples corresponding to the second window function and the at least one third window functions using an auxiliary window function to acquire a second block of windowed samples, a spectrum converter for applying an aliasing-introducing transform; and a processor for processing the first frame and the second frame to acquire encoded frames of the audio or image signal.

Apparatus and method for generating an encoded signal or for decoding an encoded audio signal using a multi overlap portion

An apparatus for encoding an audio or image signal, includes: a controllable windower for windowing the audio or image signal to provide the sequence of blocks of windowed samples; a converter for converting the sequence of blocks of windowed samples into a spectral representation including a sequence of frames of spectral values; a transient location detector for identifying a location of a transient within a transient look-ahead region of a frame; and a controller for controlling the controllable windower to apply a specific window having a specified overlap length to the audio or image signal in response to an identified location of the transient, wherein the controller is configured to select the specific window from a group of at least three windows, wherein the specific window is selected based on the transient location.

Apparatus and method for encoding or decoding an audio signal using a transient-location dependent overlap

The coding efficiency of an audio codec using a controllable&#x2014;switchable or even adjustable&#x2014;harmonic filter tool is improved by performing the harmonicity-dependent controlling of this tool using a temporal structure measure in addition to a measure of harmonicity in order to control the harmonic filter tool. In particular, the temporal structure of the audio signal is evaluated in a manner which depends on the pitch. This enables to achieve a situation-adapted control of the harmonic filter tool so that in situations where a control made solely based on the measure of harmonicity would decide against or reduce the usage of this tool, although using the harmonic filter tool would, in that situation, increase the coding efficiency, the harmonic filter tool is applied, while in other situations where the harmonic filter tool may be inefficient or even destructive, the control reduces the appliance of the harmonic filter tool appropriately.

Harmonicity-dependent controlling of a harmonic filter tool

In methods and systems for filtering an information input signal, a system may have: a first filter unit filtering an input signal at an initial subinterval in a current update interval according to parameters associated to the preceding update interval, the parameters being scaled by a first scaling factor changing towards 0; and a second filter unit filtering a second filter input signal, based on the output of the first filter unit, at the initial subinterval, according to parameters associated to the current update interval, the parameters being scaled by a second scaling factor changing from 0, or a value close to 0, toward a value more distant from 0.

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Nuremberg, Germany

Goran Markovic