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

Koninklijke Philips Corporation N.v.

Dolby International Ab

Other

Friedrich-alexander-universitaet Erlangen-nuernberg

Fraunhofer-Gesellschaft Zur Förderung Der Angewandten Forschung E.v.

Friedrich-Alexander-Universität Erlangen-Nürnberg

Stefan Doehla

Emmanuel Ravelli

Markus Multrus

Guillaume Fuchs

Harald Fuchs

Max Neuendorf

Nikolaus Faerber

Nikolaus Rettelbach

Markus Schnell

Manuel Jander

Christian Helmrich

Martin Dietz

Goran Markovic

Eleni Fotopoulou

Bernhard Grill

Daniel Fischer

Bernd Czelhan

Ingo Hofmann

Stefan Bayer

Wolfgang Jaegers

Jérémie Lecomte

Heiko Purnhagen

Jan Plogsties

Stephan Schreiner

Matthias Neusinger

Juergen Herre

Ralf Geiger

Manfred Lutzky

Ralph Sperschneider

Fabian Kuech

Jeremie Lecomte

Oliver Thiergart

Florin Ghido

Herbert Thoma

Stefan Reuschl

Simone Neukam

Frans De Bont

Grzegorz Pietrzyk

Anthony Lombard

Stefan Kraegeloh

Markus Prosch

Birgit Bartel-Kurz

Heinz Gerhaeuser

Albert Heuberger

Srikanth Korse

Andrea Eichenseer

Christian Forster

Andreas Mull

Ulf Jennehag

Jan Buethe

Franz Reutelhuber

Pallavi Maben

Wolfgang Jagers

Alexandre Boutheon

Simone Fug

Audio data processor, having: a receiver interface for receiving encoded audio data and metadata related to the encoded audio data; a metadata parser for parsing the metadata to determine an audio data manipulation possibility; an interaction interface for receiving an interaction input and for generating, from the interaction input, interaction control data related to the audio data manipulation possibility; and a data stream generator for obtaining the interaction control data and the encoded audio data and the metadata and for generating an output data stream, the output data stream having the encoded audio data, at least a portion of the metadata, and the interaction control data.

Data processor and transport of user control data to audio decoders and renderers

An audio decoder decodes a bit stream of encoded audio data, which bit stream represents a sequence of audio sample values and includes a plurality of frames, wherein each frame includes associated encoded audio sample values. The audio decoder includes a determiner configured to determine whether a frame of the encoded audio data is a special frame including encoded audio sample values associated with the special frame and additional information, wherein the additional information include encoded audio sample values of a number of frames preceding the special frame, wherein the encoded audio sample values of the preceding frames are encoded using the same codec configuration as the special frame, wherein the number of preceding frames is sufficient to initialize the decoder to be in a position to decode the audio sample values associated with the special frame if the special frame is the first frame upon start-up of the decoder.

Audio decoder, apparatus for generating encoded audio output data and methods permitting initializing a decoder

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

An apparatus for selecting one of a first encoding algorithm having a first characteristic and a second encoding algorithm having a second characteristic for encoding a portion of an audio signal to obtain an encoded version of the portion of the audio signal has a first estimator for estimating a first quality measure for the portion of the audio signal, which is associated with the first encoding algorithm, without actually encoding and decoding the portion of the audio signal using the first encoding algorithm. A second estimator is provided for estimating a second quality measure for the portion of the audio signal, which is associated with the second encoding algorithm, without actually encoding and decoding the portion of the audio signal using the second encoding algorithm. The apparatus has a controller for selecting the first or second encoding algorithms based on a comparison between the first and second quality measures.

Apparatus and method for selecting one of a first encoding algorithm and a second encoding algorithm

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

The invention provides an audio encoder including a combination of a linear predictive coding filter having a plurality of linear predictive coding coefficients and a time-frequency converter, wherein the combination is configured to filter and to convert a frame of the audio signal into a frequency domain in order to output a spectrum based on the frame and on the linear predictive coding coefficients; a low frequency emphasizer configured to calculate a processed spectrum based on the spectrum, wherein spectral lines of the processed spectrum representing a lower frequency than a reference spectral line are emphasized; and a control device configured to control the calculation of the processed spectrum by the low frequency emphasizer depending on the linear predictive coding coefficients of the linear predictive coding filter.

Low-frequency emphasis for LPC-based coding in frequency domain

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

Audio decoder device for decoding a bitstream, the audio decoder device including: a predictive decoder for producing a decoded audio frame from the bitstream, wherein the predictive decoder includes a parameter decoder for producing one or more audio parameters for the decoded audio frame from the bitstream and wherein the predictive decoder includes a synthesis filter device for producing the decoded audio frame by synthesizing the one or more audio parameters for the decoded audio frame; a memory device including one or more memories, wherein each of the memories is configured to store a memory state for the decoded audio frame, wherein the memory state for the decoded audio frame of the one or more memories is used by the synthesis filter device for synthesizing the one or more audio parameters for the decoded audio frame; and a memory state resampling device configured to determine the memory state for synthesizing the one or more audio parameters for the decoded audio frame, which has a sampling rate, for one or more of the memories by resampling a preceding memory state for synthesizing one or more audio parameters for a preceding decoded audio frame, which has a preceding sampling rate being different from the sampling rate of the decoded audio frame, for one or more of the memories and to store the memory state for synthesizing of the one or more audio parameters for the decoded audio frame for one or more of the memories into the respective memory.

Concept for switching of sampling rates at audio processing devices

Growing community of inventors

Erlangen, Germany

Stefan Doehla