Dolby Laboratories Licensing Corporation

Dolby International Ab

Mark Stuart Vinton

Michael Mead Truman

Grant Allen Davidson

Louis Dunn Fielder

Zhiwei Shuang

Xiguang Zheng

Matthew Conrad Fellers

Alan Jeffrey Seefeldt

Charles Quito Robinson

Glenn N Dickins

Lars Villemoes

Heiko Purnhagen

Jeffrey Charles Riedmiller

Michael Schug

Vinay Melkote

Leif Sehlstrom

Arijit Biswas

Kenneth James Gundry

Steven Joseph Venezia

David S McGrath

Mark Franklin Davis

Kai Lin Li

Xuejing Sun

C Phillip Brown

Regunathan Radhakrishnan

Shen Huang

Kuan-Chieh Yen

Claus Bauer

Janusz Klejsa

Vivek Kumar

Timothy Neal

Farhad Farahani

Roy M Fejgin

Cong Zhou

David Mcgrath

Chih-Wei Wu

Matthew Aubrey Watson

Timothy J Neal

The present disclosure relates to the field of audio coding, in particular, it relates to a method for encoding audio signals through a masking model based on a hearing threshold of frequency intervals of the audio signal and a measured energy of the audio signal for the corresponding frequency intervals. The disclosure further relates to an encoder that is capable of carrying out the audio encoding method.

Psychoacoustic model for audio processing

The present disclosure relates to reverberation generation for headphone virtualization. A method of generating one or more components of a binaural room impulse response (BRIR) for headphone virtualization is described. In the method, directionally-controlled reflections are generated, wherein directionally-controlled reflections impart a desired perceptual cue to an audio input signal corresponding to a sound source location. Then at least the generated reflections are combined to obtain the one or more components of the BRIR. Corresponding system and computer program products are described as well.

Reverberation generation for headphone virtualization

An audio communication endpoint receives a bitstream containing spectral components representing spectral content of an audio signal, wherein the spectral components relate to a first range extending up to a first break frequency, above which any spectral components are unassigned. The endpoint adapts the received bitstream in accordance with a second range extending up to a second break frequency by removing spectral components or adding neutral-valued spectral components relating to a range between the first and second break frequencies. The endpoint then attenuates spectral content in a neighbourhood of the least of the first and second break frequencies for thereby achieving a gradual spectral decay. After this, reconstructing the audio signal is reconstructed by an inverse transform operating on spectral components relating to said second range in the adapted and attenuated received bitstream. At small computational expense, the endpoint may to adapt to different sample rates in received bitstreams.

Enabling sampling rate diversity in a voice communication system

A method for performing inter-channel encoding of a multi-channel audio signal comprising channel signals for N channels, with N being an integer, with N&#x3e;1, is described. The method comprises determining a basic graph comprising the N channels as nodes and comprising directed edges between at least some of the N channels. Furthermore, the method comprises determining an inter-channel coding graph from the basic graph, such that the inter-channel coding graph is a directed acyclic graph, and such that a cumulated a cumulated cost of the signals of the nodes of the inter-channel coding graph is reduced.

Method and system for inter-channel coding

According to an aspect of the present invention, a method for reconstructing an audio signal having a baseband portion and a highband portion is disclosed. The method includes obtaining a decoded baseband audio signal by decoding an encoded audio signal and obtaining a plurality of subband signals by filtering the decoded baseband audio signal. The method further includes generating a high-frequency reconstructed signal by copying a number of consecutive subband signals of the plurality of subband signals and obtaining an envelope adjusted high-frequency signal. The method further includes generating a noise component based on a noise parameter. Finally, the method includes adjusting a phase of the high-frequency reconstructed signal and obtaining a time-domain reconstructed audio signal by combining the decoded baseband audio signal and the combined high-frequency signal to obtain a time-domain reconstructed audio signal.

Methods, apparatus and systems for determining reconstructed audio signal

A method of encoding audio information for forward error correction reconstruction of a transmitted audio stream over a lossy packet switched network, the method including the steps of: (a) dividing the audio stream into audio frames; (b) determining a series of corresponding audio frequency bands for the audio frames; (c) determining a series of power envelopes for the frequency bands; (d) encoding the envelopes as a low bit rate version of the audio frame in a redundant transmission frame.

Adaptive forward error correction redundant payload generation

An audio communication endpoint receives a bitstream containing spectral components representing spectral content of an audio signal, wherein the spectral components relate to a first range extending up to a first break frequency, above which any spectral components are unassigned. The endpoint adapts the received bitstream in accordance with a second range extending up to a second break frequency by removing moving spectral components or adding neutral-valued spectral components relating to a range between the first and second break frequencies. The endpoint then attenuates spectral content in a neighbourhood of the least of the first and second break frequencies for thereby achieving a gradual spectral decay. After this, reconstructing the audio signal is reconstructed by an inverse transform operating on spectral components relating to said second range in the adapted and attenuated received bitstream. At small computational expense, the endpoint may to adapt to different sample rates in received bitstreams.

A system and method of blind bandwidth extension. The system selects a prediction model from a number of stored prediction models that were generated using an unsupervised clustering method (e.g., a k-means method) and a supervised regression process (e.g., a support vector machine), and extends the bandwidth of an input musical audio signal.

Blind bandwidth extension using K-means and a support vector machine

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San Francisco, CA, United States of America

Mark Stuart Vinton