Dolby International Ab

Dolby Laboratories Licensing Corporation

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

Coding Technologies Ab

Sk Telecom Company, Ltd.

Dolby Sweden Ab

Michael Schug

Harald Mundt

Arijit Biswas

Jeffrey Charles Riedmiller

Martin Wolters

Per Hedelin

Vinay Melkote

Holger Hoerich

Juergen Herre

Mark Stuart Vinton

Pontus Carlsson

Phillip A Williams

Robin Thesing

Wolfgang Alexander Schildbach

Bernhard Grill

Nikolaus Rettelbach

Martin Dietz

Leif Jonas Samuelsson

Grant Allen Davidson

Christof Fersch

Thomas Sporer

Karsten Linzmeier

Alexander Groeschel

Andreas Schneider

Bodo Teichmann

Tobias R Wagenblass

Andreas Ehret

Johannes Hilpert

Max Neuendorf

Stefan Geyersberger

Matthias Neusinger

Ralph Sperschneider

Fredrik Henn

Regunathan Radhakrishnan

Karlheinz Brandenburg

Thomas Ziegler

Daniel Homm

Farhad Farahani

Luca Baradel

Pierre Lauber

Reinhold Boehm

Wolfgang A Schildbach

Marco Fink

Jonas Samuelsson

Mark S Vinton

Oliver Watzke

Andrea Ehret

Danilo Hollosi

Jonas Lief Samuelsson

Per Hendrik Hedelin

Harald H Mundt

Jonas Leif Samuelsson

Pontus Jan Carlsson

Jeffrey C Riedmiller

Per Henrik Hedelin

Grant A Davidson

Many portable playback devices cannot decode and playback encoded audio content having wide bandwidth and wide dynamic range with consistent loudness and intelligibility unless the encoded audio content has been prepared specially for these devices. This problem can be overcome by including with the encoded content some metadata that specifies a suitable dynamic range compression profile by either absolute values or differential values relative to another known compression profile. A playback device may also adaptively apply gain and limiting to the playback audio. Implementations in encoders, in transcoders and in decoders are disclosed.

System and method for non-destructively normalizing loudness of audio signals within portable devices

Embodiments are directed to a companding method and system for reducing coding noise in an audio codec. A compression process reduces an original dynamic range of an initial audio signal through a compression process that divides the initial audio signal into a plurality of segments using a defined window shape, calculates a wideband gain in the frequency domain using a non-energy based average of frequency domain samples of the initial audio signal, and applies individual gain values to amplify segments of relatively low intensity and attenuate segments of relatively high intensity. The compressed audio signal is then expanded back to the substantially the original dynamic range that applies inverse gain values to amplify segments of relatively high intensity and attenuating segments of relatively low intensity. A QMF filterbank is used to analyze the initial audio signal to obtain a frequency domain representation.

Companding system and method to reduce quantization noise using advanced spectral extension

The present disclosure relates to methods and apparatus for audio coding. A method of encoding a portion of an audio signal comprises determining whether the portion of the audio signal is likely to contain dense transient events, and if it is determined that the portion of the audio signal is likely to contain dense transient events, quantizing the portion of the audio signal using a quantization 5 mode that applies a substantially constant signal-to-noise ratio over frequency for the portion of the audio signal. The present disclosure further relates to a method of detecting dense transient events in a portion of an audio signal.

Low complexity dense transient events detection and coding

The present document describes a method for allocating bits to a frame of a sequence of frames to yield a bitstream having a constant average bitrate, wherein the frame comprises audio data and metadata. The method comprises maintaining an overall bit reservoir and maintaining a virtual bit reservoir being a subset of the overall bit reservoir, such that bits for the metadata of the frame are allocated from the virtual bit reservoir and such that bits for the audio data of the frame are allocated from the overall bit reservoir.

Method and system for bit reservoir control in case of varying metadata

The present document describes a method () for allocating bits to a frame () of a sequence of frames () to yield a bitstream having a constant average bitrate, wherein the frame () comprises audio data and metadata. The method () comprises maintaining () an overall bit reservoir () and maintaining () a virtual bit reservoir () being a subset of the overall bit reservoir (), such that bits for the metadata of the frame () are allocated from the virtual bit reservoir () and such that bits for the audio data of the frame () are allocated from the overall bit reservoir ().

The present document relates to the technical field of audio coding, decoding and processing. It specifically relates to methods of recovering high frequency content of an audio signal from low frequency content of the same audio signal in an efficient manner. A method for determining a first banded tonality value for a first frequency subband of an audio signal is described. The first banded tonality value is used for approximating a high frequency component of the audio signal based on a low frequency component of the audio signal. The method comprises determining a set of transform coefficients in a corresponding set of frequency bins based on a block of samples of the audio signal; determining a set of bin tonality values for the set of frequency bins using the set of transform coefficients, respectively; and combining a first subset of two or more of the set of bin tonality values for two or more corresponding adjacent frequency bins of the set of frequency bins lying within the first frequency subband, thereby yielding the first banded tonality value for the first frequency subband.

Methods and systems for efficient recovery of high frequency audio content

Growing community of inventors

Erlangen, Germany

Michael Schug