Dolby Laboratories Licensing Corporation

Dolby International Ab

University of California

Vinay Melkote

Heiko Purnhagen

Kristofer Kjoerling

Jeffrey Charles Riedmiller

Karl Jonas Roeden

Leif Sehlstrom

Michael Schug

Arijit Biswas

Grant Allen Davidson

Per Hedelin

Kuan-Chieh Yen

Mark Stuart Vinton

Matthew Conrad Fellers

Malcolm James Law

Roy M Fejgin

Michael C Ward

David S McGrath

Mark Franklin Davis

Charles Quito Robinson

Rhonda J Wilson

Freddie Sanchez

Kenneth Rose

Vivek Kumar

Andy Jaspar

Simon Plain

Jingning Han

Malcolm J Law

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

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.

Methods for detecting whether a rendered version of a specified seamless connection ('SSC') at a connection point between two audio segment sequences results in an audible discontinuity, and methods for analyzing at least one SSC between audio segment sequences to determine whether a renderable version of each SSC would have an audible discontinuity at the connection point when rendered, and in appropriate cases, for a SSC having a renderable version which is determined to have an audible discontinuity when rendered, correcting at least one audio segment of at least one segment sequence to be connected in accordance with the SSC in an effort to ensure that rendering of the SSC will result in seamless connection without an audible discontinuity. Other aspects are editing systems configured to implement any of the methods, and storage media and rendering systems which store audio data generated in accordance with any of the methods.

Audio discontinuity detection and correction

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 (a) 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.

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 (a) from the bitstream, where 1&#x2264;m&#x3c;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.

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 (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.

Methods for generating encoded audio programs indicative of N channels of discontinuity-corrected, encoded audio content, including by applying discontinuity correction values to multi-channel audio content, and for rendering such a program (e.g., to generate a discontinuity-corrected M-channel mix of content indicated by the program). Other aspects are systems or devices (e.g., encoders or decoders, or rendering systems) configured to implement any of the methods.

Audio encoding and rendering with discontinuity compensation

A method of encoding adaptive audio, comprising receiving N objects and associated spatial metadata that describes the continuing motion of these objects, and partitioning the audio into segments based on the spatial metadata. The method encodes adaptive audio having objects and channel beds by capturing a continuing motion of a number N objects in a time-varying matrix trajectory comprising a sequence of matrices, coding coefficients of the time-varying matrix trajectory in spatial metadata to be transmitted via a high-definition audio format for rendering the adaptive audio through a number M output channels, and segmenting the sequence of matrices into a plurality of sub-segments based on the spatial metadata, wherein the plurality of sub-segments are configured to facilitate coding of one or more characteristics of the adaptive audio.

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Bengaluru, India

Vinay Melkote