Patents by Inventor Stanislaw Gorlow
Stanislaw Gorlow has filed for patents to protect the following inventions. This listing includes patent applications that are pending as well as patents that have already been granted by the United States Patent and Trademark Office (USPTO).
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Patent number: 11929082Abstract: The present disclosure relates to the field audio coding, an in particular to an audio decoder having at least two decoding modes, and associated decoding methods and decoding software for such audio decoder. In one of the decoding modes, at least one dynamic audio object is mapped to a set of static audio objects, the set of static audio objects corresponding to a predefined speaker configuration. The present disclosure further relates to a corresponding audio encoder, and associated encoding methods and encoding software for such audio encoder.Type: GrantFiled: October 30, 2019Date of Patent: March 12, 2024Assignee: DOLBY INTERNATIONAL ABInventors: Tobias Friedrich, Heiko Purnhagen, Stanislaw Gorlow, Celine Merpillat
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Publication number: 20230198488Abstract: The present document describes a dynamic range control unit (210) configured to apply dynamic range control, referred to as DRC, to an audio signal (211). The DRC unit (210) is configured to downsample a subband signal (212) derived from the audio signal (211), to provide a downsampled subband signal (321), to determine a DRC gain (329) based on the downsampled subband signal (321), and to apply the DRC gain (329) to the subband signal (212), to provide a compressed subband signal (213) of a compressed audio signal (214).Type: ApplicationFiled: May 17, 2021Publication date: June 22, 2023Applicant: DOLBY INTERNATIONAL ABInventors: Stanislaw GORLOW, Robin THESING
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Publication number: 20220199101Abstract: Dialogue enhancement of an audio signal, comprising obtaining a set of time-varying parameters configured to estimate a dialogue component present in said audio signal, estimating the dialogue component from the audio signal, applying a compressor only to the estimated dialogue component, to generate a processed dialogue component, applying a user-determined gain to the processed dialogue component, to provide an enhanced dialogue component. The processing of the estimated dialogue may be performed on the decoder side or encoder side. The invention enables an improved dialogue enhancement.Type: ApplicationFiled: April 15, 2020Publication date: June 23, 2022Applicant: DOLBY INTERNATIONAL ABInventors: Stanislaw Gorlow, Leif Jonas Samuelsson, Holger Hoerich, Tobias Friedrich
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Patent number: 11227621Abstract: The present disclosure provides new variants of non-negative matrix factorization suitable for separating desired audio content from undesired audio content. In certain embodiments, a multi-dimensional non-negative representation of an audio signal is decomposed into desired content and undesired content by performing convolutional non-negative matrix factorization (CNMF) on multiple layers, each layer having a respective non-negative matrix representation. In certain embodiments, the desired content is represented by a first dictionary and the undesired content is represented by a second dictionary, and sparsity is imposed on activations of basic elements of the first or the second dictionary, wherein a degree of sparsity is controlled by setting a minimum number of components with significant activations of the first or second dictionary, respectively.Type: GrantFiled: September 16, 2019Date of Patent: January 18, 2022Assignee: Dolby International ABInventors: Pedro Jafeth Villasana Tinajero, Stanislaw Gorlow
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Publication number: 20220005484Abstract: The present disclosure relates to the field audio coding, an in particular to an audio decoder having at least two decoding modes, and associated decoding methods and decoding software for such audio decoder. In one of the decoding modes, at least one dynamic audio object is mapped to a set of static audio objects, the set of static audio objects corresponding to a predefined speaker configuration. The present disclosure further relates to a corresponding audio encoder, and associated encoding methods and encoding software for such audio encoder.Type: ApplicationFiled: October 30, 2019Publication date: January 6, 2022Applicant: DOLBY INTERNATIONAL ABInventors: Tobias FRIEDRICH, Heiko PURNHAGEN, Stanislaw GORLOW, Celine MERPILLAT
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Patent number: 10924078Abstract: The present disclosure relates to a method of determining parameters for use by a first dynamic range control, DRC, model. The method comprises feeding a first audio signal to a second DRC model and receiving a second audio signal from the second DRC model, the second audio signal being a dynamic range controlled version of the first audio signal, rule-based selecting one or more pairs of samples of the first audio signal and corresponding samples of the second audio signal, and determining parameters of a first set of parameters among the parameters for use by the first DRC model based on the one or more selected pairs of samples. The present disclosure further relates to a method of reversing DRC of a dynamic range controlled audio signal and to a method of declipping a clipped audio signal that has been clipped by a DRC model. The present disclosure yet further relates to corresponding apparatus and computer-readable media.Type: GrantFiled: March 19, 2018Date of Patent: February 16, 2021Assignee: Dolby International ABInventor: Stanislaw Gorlow
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Publication number: 20200090677Abstract: The present disclosure provides new variants of non-negative matrix factorization suitable for separating desired audio content from undesired audio content. In certain embodiments, a multi-dimensional non-negative representation of an audio signal is decomposed into desired content and undesired content by performing convolutional non-negative matrix factorization (CNMF) on multiple layers, each layer having a respective non-negative matrix representation. In certain embodiments, the desired content is represented by a first dictionary and the undesired content is represented by a second dictionary, and sparsity is imposed on activations of basic elements of the first or the second dictionary, wherein a degree of sparsity is controlled by setting a minimum number of components with significant activations of the first or second dictionary, respectively.Type: ApplicationFiled: September 16, 2019Publication date: March 19, 2020Applicant: DOLBY INTERNATIONAL ABInventors: PEDRO JAFETH VILLASANA TINAJERO, STANISLAW GORLOW
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Publication number: 20200067476Abstract: The present disclosure relates to a method of determining parameters for use by a first dynamic range control, DRC, model. The method comprises feeding a first audio signal to a second DRC model and receiving a second audio signal from the second DRC model, the second audio signal being a dynamic range controlled version of the first audio signal, rule-based selecting one or more pairs of samples of the first audio signal and corresponding samples of the second audio signal, and determining parameters of a first set of parameters among the parameters for use by the first DRC model based on the one or more selected pairs of samples. The present disclosure further relates to a method of reversing DRC of a dynamic range controlled audio signal and to a method of declipping a clipped audio signal that has been clipped by a DRC model. The present disclosure yet further relates to corresponding apparatus and computer-readable media.Type: ApplicationFiled: March 19, 2018Publication date: February 27, 2020Applicant: Dolby International ABInventor: Stanislaw GORLOW
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Patent number: 9936295Abstract: An electronic device comprising a processing unit arranged to determine an estimation signal (y(k)) based on an input signal (x(k)) and based on a non-stationary reference signal (s0(k)).Type: GrantFiled: July 11, 2016Date of Patent: April 3, 2018Assignee: Sony CORPORATIONInventors: Stanislaw Gorlow, Mathieu Ramona, Francois Pachet
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Publication number: 20170026748Abstract: An electronic device comprising a processing unit arranged to determine an estimation signal (y(k)) based on an input signal (x(k)) and based on a non-stationary reference signal (s0(k)).Type: ApplicationFiled: July 11, 2016Publication date: January 26, 2017Applicant: Sony CorporationInventors: Stanislaw GORLOW, Mathieu RAMONA, Francois PACHET
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Patent number: 9437199Abstract: The invention relates to a method and the associated device 1 for separating one or more particular digital audio source signals (si) contained in a mixed multichannel digital audio signal (smix) obtained by mixing a plurality of digital audio source signals (s1, . . . , sp). According to the invention: the modulus of the amplitude or the normalized power of the particular source signal(s) (si) is determined from representative values of said particular source signal(s) contained in the mixed signal; and then linearly constrained minimum variance spatial filtering is performed on the mixed signal in order to obtain each particular source signal (s?i), said filtering being based on the distribution of said particular source signal between at least two channels of the mixed signal, and the modulus of the amplitude or the normalized power of said particular source signal is used as a linear constraint of the filter.Type: GrantFiled: September 25, 2013Date of Patent: September 6, 2016Assignees: UNIVERSITÉ BORDEAUX 1, CENTRE NATIONAL DE LAS RECHERCHE SCIENTIFIQUE (CInventors: Sylvain Marchand, Stanislaw Gorlow
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Publication number: 20150243290Abstract: The invention relates to a method and the associated device 1 for separating one or more particular digital audio source signals (si) contained in a mixed multichannel digital audio signal (smix) obtained by mixing a plurality of digital audio source signals (s1, . . . , sp). According to the invention: the modulus of the amplitude or the normalized power of the particular source signal(s) (si) is determined from representative values of said particular source signal(s) contained in the mixed signal; and then linearly constrained minimum variance spatial filtering is performed on the mixed signal in order to obtain each particular source signal (s?i), said filtering being based on the distribution of said particular source signal between at least two channels of the mixed signal, and the modulus of the amplitude or the normalized power of said particular source signal is used as a linear constraint of the filter.Type: ApplicationFiled: September 25, 2013Publication date: August 27, 2015Applicant: CENTRE NATIONAL DE LA RECHERCHE SCIENTFIQUE (CNRS)Inventors: Sylvain Marchand, Stanislaw Gorlow