Dolby International Ab

Dolby Laboratories Licensing Corporation

Other

Koninklijke Philips Corporation N.v.

Coding Technologies Ab

Karl Jonas Roeden

Heiko Purnhagen

Kristofer Kjoerling

Leif Sehlstrom

Jeffrey Charles Riedmiller

Harald Mundt

Vinay Melkote

Jonas Engdegard

Scott Gregory Norcross

Robin Thesing

Jens Popp

Lars Gustaf Liljeryd

Fredrik Henn

Lars Villemoes

Michael C Ward

Erik Gosuinus Petrus Schuijers

Kristofer Kjörling

Gerard Herman Hotho

Alexander Groeschel

Dirk Jeroen Breebaart

Lie Lu

Michael D Ward

Alexander Stahlmann

Holger Hoerich

Toni Hirvonen

Martin Wolters

Karsten Linzmeier

Daniel Arteaga

Phillip A Williams

Tobias Friedrich

Wolfgang Alexander Schildbach

Machiel Willem Van Loon

Holger Horich

Leif Sehlström

Hans Magnus Kristofer Kjoerling

Eduard Aylon Pla

Alex Cabrer Manning

Kristopher Kjoerling

Jenns Popp

There is provided methods and apparatuses for decoding and encoding of audio signals. In particular, a method for decoding includes receiving a waveform-coded signal having a spectral content corresponding to a subset of the frequency range above a cross-over frequency. The waveform-coded signal is interleaved with a parametric high frequency reconstruction of the audio signal above the cross-over frequency. In this way an improved reconstruction of the high frequency bands of the audio signal is achieved.

Audio encoder and decoder for interleaved waveform coding

Embodiments are directed to a method and system for receiving, in a bitstream, metadata associated with the audio data, and analyzing the metadata to determine whether a loudness parameter for a first group of audio playback devices are available in the bitstream. Responsive to determining that the parameters are present for the first group, the system uses the parameters and audio data to render audio. Responsive to determining that the loudness parameters are not present for the first group, the system analyzes one or more characteristics of the first group, and determines the parameter based on the one or more characteristics.

System and method for optimizing loudness and dynamic range across different playback devices

A method for decoding an encoded audio bitstream in an audio processing system is disclosed. The method includes extracting from the encoded audio bitstream a first waveform-coded signal comprising spectral coefficients corresponding to frequencies up to a first cross-over frequency for a time frame and performing parametric decoding at a second cross-over frequency for the time frame to generate a reconstructed signal. The second cross-over frequency is above the first cross-over frequency and the parametric decoding uses reconstruction parameters derived from the encoded audio bitstream to generate the reconstructed signal. The method also includes extracting from the encoded audio bitstream a second waveform-coded signal comprising spectral coefficients corresponding to a subset of frequencies above the first cross-over frequency for the time frame and interleaving the second waveform-coded signal with the reconstructed signal to produce an interleaved signal for the time frame.

Audio decoder for interleaving signals

An audio processing system () accepts an audio bitstream having one of a plurality of predefined audio frame rates. The system comprises a front-end component (), which receives a variable number of quantized spectral components, corresponding to one audio frame in any of the predefined audio frame rates, and performs an inverse quantization according to predetermined, frequency-dependent quantization levels. The front-end component may be agnostic of the audio frame rate. The audio processing system further comprises a frequency-domain processing stage () and a sample rate converter (), which provide a reconstructed audio signal sampled at a target sampling frequency independent of the audio frame rate. By its frame-rate adaptability, the system can be configured to operate frame-synchronously in parallel with a video processing system that accepts plural video frame rates.

Method, apparatus and systems for audio decoding and encoding

On the basis of a bitstream (P), an n-channel audio signal (X) is reconstructed by deriving an m-channel core signal (Y) and multichannel coding parameters (α) from the bitstream, where 1≤m<n. Also derived from the bitstream are pre-processing dynamic range control, DRC, parameters (DRC) quantifying an encoder-side dynamic range limiting of the core signal. The n-channel audio signal is obtained by parametric synthesis in accordance with the multichannel coding parameters and while cancelling any encoder-side dynamic range limiting based on the pre-processing DRC parameters. In particular embodiments, the reconstruction further includes use of compensated post-processing DRC parameters quantifying a potential decoder-side dynamic range compression. Cancellation of an encoder-side range limitation and range compression are preferably performed by different decoder-side components. Cancellation and compression may be coordinated by a DRC pre-processor.

System for maintaining reversible dynamic range control information associated with parametric audio coders

On the basis of a bitstream (P), an n-channel audio signal (X) is reconstructed by deriving an m-channel core signal (Y) and multichannel coding parameters (&#x3b1;) from the bitstream, where 1&#x2264;m&#x3c;n. Also derived from the bitstream are pre-processing dynamic range control, DRC, parameters (DRC2) quantifying an encoder-side dynamic range limiting of the core signal. The n-channel audio signal is obtained by parametric synthesis in accordance with the multichannel coding parameters and while cancelling any encoder-side dynamic range limiting based on the pre-processing DRC parameters. In particular embodiments, the reconstruction further includes use of compensated post-processing DRC parameters quantifying a potential decoder-side dynamic range compression. Cancellation of an encoder-side range limitation and range compression are preferably performed by different decoder-side components. Cancellation and compression may be coordinated by a DRC pre-processor.

Described herein is a method for creating object-based audio content from a text input for use in audio books and/or audio play, the method including the steps of: a) receiving the text input; b) performing a semantic analysis of the received text input; c) synthesizing speech and effects based on one or more results of the semantic analysis to generate one or more audio objects; d) generating metadata for the one or more audio objects; and e) creating the object-based audio content including the one or more audio objects and the metadata. Described herein are further a computer-based system including one or more processors configured to perform said method and a computer program product comprising a computer-readable storage medium with instructions adapted to carry out said method when executed by a device having processing capability.

Method and system for creating object-based audio content

Audio decoder utilizing sample rate conversion for frame synchronization

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Solna, Sweden

Karl Jonas Roeden