Dolby International Ab

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

Koninklijke Philips Corporation N.v.

Coding Technologies Ab

Dolby Laboratories Licensing Corporation

Coding Technologies Sweden Ab

Dolby Sweden Ab

Agere Systems Inc.

Voiceage Corporation

Dolby Sweden Ag

Conding Technologies Ab

Koninklijke Philips International N.v.

Fraunhofer-gesellschaft Zur F&#xd6;rderung Der Angewandten Forschung E.v.

Lars Villemoes

Heiko Purnhagen

Per Ekstrand

Kristofer Kjoerling

Jonas Engdegard

Per Hedelin

Pontus Carlsson

Leif Jonas Samuelsson

Sascha Disch

Frederik Nagel

Dirk Jeroen Breebaart

Erik Gosuinus Petrus Schuijers

Johannes Hilpert

Glenn N Dickins

Ralf Geiger

Stefan Bayer

Bernd Edler

Mark Stuart Vinton

Harald Mundt

Fredrik Henn

Karl Jonas Roeden

Juergen Herre

Nikolaus Rettelbach

Jürgen Herre

Andreas Hoelzer

Albertus Cornelis Den Brinker

Jeroen Koppens

Bernhard Grill

Christian Helmrich

Max Neuendorf

Christof Faller

Oliver Hellmuth

Matthias Neusinger

Ralph Sperschneider

Jan Plogsties

Richard James Cartwright

Redwan Salami

Jeremie Lecomte

Shen Huang

Jonas Roeden

Toni Hirvonen

Kristofer Kjörling

Barbara Resch

Stephan Wilde

Janusz Klejsa

Jeroen Dirk Breebaart

Heidi-Maria Lehtonen

Leif Sehlstrom

Kristofer Kjorling

Erik Schuijers

Gerard Herman Hotho

Arijit Biswas

Tom Baeckstroem

Cornelia Falch

Fransiscus Marinus Jozephus De Bont

Leonid Terentiev

Werner Oomen

Bruno Bessette

Philippe Gournay

Karsten Linzmeier

Roch Lefebvre

Robin Thesing

Francois Philippus Myburg

Roy M Fejgin

Cong Zhou

Andreas Hölzer

Karl Jonas Roden

Julien Robillard

Jonas Röden

Per Elstrand

Fransiscus Marinus Jozephus Bont

Jérémie Lecomte

Julien Robilliard

Jeroen Koninklijke Philips Elec NV Breebaart

Erik Koninklijke Philips Elec NV Schuijers

Werner Koninklijke Philips Elec NV Oomen

The present document relates an audio encoding and decoding system (referred to as an audio codec system). In some embodiments, a method of audio signal encoding comprises: receiving an input audio signal; transforming a sequence of samples of the input audio signal into a block of transform coefficients, the transform coefficients indicative of the spectral energy of the block; estimating a spectral envelope of the block from the transform coefficients; adjusting the transform coefficients using the spectral envelope and a spectral shaper, the spectral shaper including one or more parameters indicative of a fundamental frequency of a multi-sinusoidal signal model, where the fundamental frequency corresponds to a time domain delay; and entropy coding the adjusted transform coefficients.

Voice encoding and decoding using transform coefficients adjusted by spectral model and spectral shaper

Described herein is a method of waveform decoding, the method including the steps of: (a) receiving, by a waveform decoder, a bitstream including a finite bitrate representation of a source signal; (b) waveform decoding the finite bitrate representation of the source signal to obtain a waveform approximation of the source signal; (c) providing the waveform approximation of the source signal to a generative model that implements a probability density function, to obtain a probability distribution for a reconstructed signal of the source signal; and (d) generating the reconstructed signal of the source signal based on the probability distribution. Described are further a method and system for waveform coding and a method of training a generative model.

Methods and system for waveform coding of audio signals with a generative model by implementing a probability distribution

A method for decoding an encoded audio bitstream is disclosed. The method includes receiving the encoded audio bitstream and decoding the audio data to generate a decoded lowband audio signal. The method further includes extracting high frequency reconstruction metadata and filtering the decoded lowband audio signal with an analysis filterbank to generate a filtered lowband audio signal. The method also includes extracting a flag indicating whether either spectral translation or harmonic transposition is to be performed on the audio data and regenerating a highband portion of the audio signal using the filtered lowband audio signal and the high frequency reconstruction metadata in accordance with the flag.

Backward-compatible integration of high frequency reconstruction techniques for audio signals

A method for decoding an encoded audio bitstream is disclosed. The method includes receiving the encoded audio bitstream and decoding the audio data to generate a decoded lowband audio signal. The method further includes extracting high frequency reconstruction metadata and filtering the decoded lowband audio signal with an analysis filterbank to generate a filtered lowband audio signal. The method also includes extracting a flag indicating whether either spectral translation or harmonic transposition is to be performed on the audio data and regenerating a highband portion of the audio signal using the filtered lowband audio signal and the high frequency reconstruction metadata in accordance with the flag. The high frequency regeneration is performed as a post-processing operation with a delay of 3010 samples per audio channel.

Integration of high frequency audio reconstruction techniques

The invention provides methods and devices for stereo encoding and decoding using complex prediction in the frequency domain. In one embodiment, a decoding method, for obtaining an output stereo signal from an input stereo signal encoded by complex prediction coding and comprising first frequency-domain representations of two input channels, comprises the upmixing steps of:

MDCT-based complex prediction stereo coding

An apparatus for processing an input audio signal relies on a cascade of filterbanks, the cascade having a synthesis filterbank for synthesizing an audio intermediate signal from the input audio signal, the input audio signal being represented by a plurality of first subband signals generated by an analysis filterbank, wherein a number of filterbank channels of the synthesis filterbank is smaller than a number of channels of the analysis filterbank. The apparatus furthermore has a further analysis filterbank for generating a plurality of second subband signals from the audio intermediate signal, wherein the further analysis filterbank has a number of channels being different from the number of channels of the synthesis filterbank, so that a sampling rate of a subband signal of the plurality of second subband signals is different from a sampling rate of a first subband signal of the plurality of first subband signals.

Apparatus and method for processing an input audio signal using cascaded filterbanks

Backward-compatible integration of harmonic transposer for high frequency reconstruction of audio signals

Integration of high frequency reconstruction techniques with reduced post-processing delay

The present document relates to audio source coding systems which make use of linear prediction in combination with a filterbank. A method for estimating a sample of a subband signal from two or more previous samples of the subband signal is described. The subband signal corresponds to a plurality of subbands, having an equal subband spacing, of a subband-domain representation of an audio signal. The method comprises determining signal model data using a model parameter; determining a first prediction coefficient in response to the model parameter using a first lookup table and/or a first analytical function; determining a second prediction coefficient in response to the model parameter using a second lookup table and/or a second analytical function; and determining the estimate of the sample by applying the first prediction coefficient to the first previous sample and applying the second prediction coefficient to the second previous sample.

Model based prediction in a critically sampled filterbank

The invention provides an efficient implementation of cross-product enhanced high-frequency reconstruction (HFR), wherein a new component at frequency QΩ+rΩis generated on the basis of existing components at Ω and Ω+Ω. The invention provides a block-based harmonic transposition, wherein a time block of complex subband samples is processed with a common phase modification. Superposition of several modified samples has the net effect of limiting undesirable intermodulation products, thereby enabling a coarser frequency resolution and/or lower degree of oversampling to be used. In one embodiment, the invention further includes a window function suitable for use with block-based cross-product enhanced HFR. A hardware embodiment of the invention may include an analysis filter bank, a subband processing unit configurable by control data and a synthesis filter bank.

Cross product enhanced subband block based harmonic transposition

Embodiments relate to an audio processing unit that includes a buffer, bitstream payload deformatter, and a decoding subsystem. The buffer stores at least one block of an encoded audio bitstream. The block includes a fill element that begins with an identifier followed by fill data. The fill data includes at least one flag identifying whether enhanced spectral band replication (eSBR) processing is to be performed on audio content of the block. A corresponding method for decoding an encoded audio bitstream is also provided.

Decoding audio bitstreams with enhanced spectral band replication metadata in at least one fill element

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Lars Villemoes