Dolby International Ab

Dolby Laboratories Licensing Corporation

Alexander Stahlmann

Jeffrey Charles Riedmiller

Jeroen Koppens

Kristofer Kjoerling

Heiko Purnhagen

Alan Jeffrey Seefeldt

Michael C Ward

Holger Hoerich

Scott Gregory Norcross

Christof Fersch

Leif Jonas Samuelsson

Karl Jonas Roeden

Rhonda J Wilson

David Matthew Cooper

Jens Popp

Dirk Jeroen Breebaart

Kristofer Kjörling

Leif Sehlstrom

Karsten Linzmeier

Robin Thesing

Reinhold Boehm

Gael Lassure

Simon Plain

Leif Sehlström

Mark C Leddy

Vinay Mathew

Jan Mueller

Rhonda Joy Wilson

Mark Leddy

A method for generating a bitstream indicative of an object based audio program is described. The bitstream comprises a sequence of containers. A first container of the sequence of containers comprises a plurality of substream entities for a plurality of substreams of the object based audio program and a presentation section. The method comprises determining a set of object channels. The method further comprises providing a set of object related metadata for the set of object channels. In addition, the method comprises inserting a first set of object channel frames and a first set of object related metadata frames into a respective set of substream entities of the first container. Furthermore, the method comprises inserting presentation data into the presentation section.

Methods, apparatus and system for rendering an audio program

Audio content coded for a reference speaker configuration is downmixed to downmix audio content coded for a specific speaker configuration. One or more gain adjustments are performed on individual portions of the downmix audio content coded for the specific speaker configuration. Loudness measurements are then performed on the individual portions of the downmix audio content. An audio signal that comprises the audio content coded for the reference speaker configuration and downmix loudness metadata is generated. The downmix loudness metadata is created based at least in part on the loudness measurements on the individual portions of the downmix audio content.

Loudness adjustment for downmixed audio content

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

In an audio encoder, for audio content received in a source audio format, default gains are generated based on a default dynamic range compression (DRC) curve, and non-default gains are generated for a non-default gain profile. Based on the default gains and non-default gains, differential gains are generated. An audio signal comprising the audio content, the default DRC curve, and differential gains is generated. In an audio decoder, the default DRC curve and the differential gains are identified from the audio signal. Default gains are re-generated based on the default DRC curve. Based on the combination of the re-generated default gains and the differential gains, operations are performed on the audio content extracted from the audio signal.

Dynamic range control for a wide variety of playback environments

A method for encoding an input audio stream including the steps of obtaining a first playback stream presentation of the input audio stream intended for reproduction on a first audio reproduction system, obtaining a second playback stream presentation of the input audio stream intended for reproduction on a second audio reproduction system, determining a set of transform parameters suitable for transforming an intermediate playback stream presentation to an approximation of the second playback stream presentation, wherein the transform parameters are determined by minimization of a measure of a difference between the approximation of the second playback stream presentation and the second playback stream presentation, and encoding the first playback stream presentation and the set of transform parameters for transmission to a decoder.

Audio encoding and decoding using presentation transform parameters

The present document discloses a method for fading discontinued audio feeds for replay by a speaker. In particular, the method may first comprise receiving an input audio feed comprising a plurality of samples. The method may further comprise determining whether the input audio feed is discontinued. And, when discontinuity of the input audio feed is detected, the method may comprise generating an intermediate audio signal comprising a plurality of samples based on the discontinued input audio feed. In particular, the intermediate audio signal may be generated based on a last portion of the discontinued input audio feed that has been output for replay. In addition, the method may further comprise applying a fadeout function to the intermediate audio signal to generate a fadeout audio signal. Finally, the method may comprise outputting the fadeout audio signal for replay by the speaker.

Methods, apparatus and systems for low latency audio discontinuity fade out

A technique including receiving and decoding a coded bitstream encoded with audio content including first audio objects corresponding to a first media content type of two consecutive media content types and second audio objects corresponding to a second media content type of the two consecutive media content types, and audio metadata corresponding to the audio content. The audio metadata including first and second audio object gains, for the first and second audio objects, generated in part based on a first fading curve of the first media content type and a second fading curve of the second media content type, respectively. The technique further includes applying the first and second audio object gains to the first and second audio objects, and rendering a sound field represented by the first audio object with the applied first audio object gain and the second audio object with the applied second audio object gain.

Using metadata to aggregate signal processing operations

Growing community of inventors

Bubenreuth, Germany

Alexander Stahlmann