Patents by Inventor Zoran Fejzo
Zoran Fejzo 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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Publication number: 20150255076Abstract: A post-encoding bitrate reduction system and method for generating one more scaled compressed bitstreams from a single encoded plenary file. The plenary file contains multiple audio object files that were encoded separately using a scalable encoding process having fine-grained scalability. Activity in the data frames of the encoded audio object files at a time period are compared with each other to obtain a data frame activity comparison. Bits from an available bitpool are assigned to all of the data frames based on the data frame activity comparison and corresponding hierarchical metadata. The plenary file is scaled down by truncating bits in the data frames to conform to the bit allocation. In some embodiments frame activity is compared to a silence threshold and the data frame contains silence if the frame activity is less than or equal to the threshold and minimal bits are used to represent the silent frame.Type: ApplicationFiled: March 6, 2014Publication date: September 10, 2015Applicant: DTS, INC.Inventor: Zoran Fejzo
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Publication number: 20150230041Abstract: Devices and methods are adapted to characterize a multi-channel loudspeaker configuration, to correct loudspeaker/room delay, gain and frequency response or to configure sub-band domain correction filters.Type: ApplicationFiled: April 20, 2015Publication date: August 13, 2015Applicant: DTS, INC.Inventors: Zoran Fejzo, James D. Johnston
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Publication number: 20150170657Abstract: A multiplet-based spatial matrixing codec and method for reducing channel counts (and thus bitrates) of high-channel count (seven or more channels) multichannel audio, optimizing audio quality by enabling tradeoffs between spatial accuracy and basic audio quality, and converting audio signal formats to playback environment configurations. An initial N channel count is reduced to M channels by spatial matrix mixing to a lower number of channels using multiplet pan laws. The multiplet pan laws include doublet, triplet, and quadruplet pan laws. For example, using a quadruplet pan law one of the N channels can be downmixed to four of the M channels to create a quadruplet channel. Spatial information as well and audio content is contained in the multiplet channels. During upmixing the downmixed channel is extracted from the multiplet channels using the corresponding multiplet pan law. The extracted channel then is rendered at any location within a playback environment.Type: ApplicationFiled: November 26, 2014Publication date: June 18, 2015Applicant: DTS, INC.Inventors: Jeffrey Kenneth Thompson, Zoran Fejzo
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Patent number: 9031268Abstract: Devices and methods are adapted to characterize a multi-channel loudspeaker configuration, to correct loudspeaker/room delay, gain and frequency response or to configure sub-band domain correction filters.Type: GrantFiled: May 9, 2011Date of Patent: May 12, 2015Assignee: DTS, Inc.Inventors: Zoran Fejzo, James D. Johnston
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Publication number: 20140350944Abstract: The present invention provides a novel end-to-end solution for creating, encoding, transmitting, decoding and reproducing spatial audio soundtracks. The provided soundtrack encoding format is compatible with legacy surround-sound encoding formats, so that soundtracks encoded in the new format may be decoded and reproduced on legacy playback equipment with no loss of quality compared to legacy formats.Type: ApplicationFiled: March 15, 2012Publication date: November 27, 2014Applicant: DTS, Inc.Inventors: Jean-Marc Jot, Zoran Fejzo, James D. Johnston
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Publication number: 20140270263Abstract: There are disclosed automatic mixers and methods for creating a surround audio mix. A set of rules may be stored in a rule base. A rule engine may select a subset of the set of rules based, at least in part, on metadata associated with a plurality of stems. A mixing matrix may mix the plurality of stems in accordance with the selected subset of rules to provide three or more output channels.Type: ApplicationFiled: March 12, 2014Publication date: September 18, 2014Applicant: DTS, Inc.Inventors: Zoran Fejzo, Fred Maher
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Publication number: 20130182852Abstract: There is disclosed methods and apparatus for decomposing a signal having a plurality of channels into direct and diffuse components. The correlation coefficient between each pair of signals from the plurality of signals may be estimated. A linear system of equations relating the estimated correlation coefficients and direct energy fractions of each of the plurality of channels may be constructed. The linear system may be solved to estimate the direct energy fractions. A direct component output signal and a diffuse component output signal may be generated based in part on the direct energy fractions.Type: ApplicationFiled: September 12, 2012Publication date: July 18, 2013Inventors: Jeff Thompson, Brandon Smith, Aaron Warner, Zoran Fejzo, Jean-Mar Jot
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Patent number: 8374858Abstract: An audio codec losslessly encodes audio data into a sequence of analysis windows in a scalable bitstream. This is suitably done by separating the audio data into MSB and LSB portions and encoding each with a different lossless algorithm. An authoring tool compares the buffered payload to an allowed payload for each window and selectively scales the losslessly encoded audio data, suitably the LSB portion, in the non-conforming windows to reduce the encoded payload, hence buffered payload. This approach satisfies the media bit rate and buffer capacity constraints without having to filter the original audio data, reencode or otherwise disrupt the lossless bitstream.Type: GrantFiled: March 9, 2010Date of Patent: February 12, 2013Assignee: DTS, Inc.Inventor: Zoran Fejzo
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Publication number: 20120288124Abstract: Devices and methods are adapted to characterize a multi-channel loudspeaker configuration, to correct loudspeaker/room delay, gain and frequency response or to configure sub-band domain correction filters.Type: ApplicationFiled: May 9, 2011Publication date: November 15, 2012Inventors: Zoran Fejzo, James D. Johnston
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Patent number: 8239210Abstract: A lossless audio codec segments audio data within each frame to improve compression performance subject to a constraint that each segment must be fully decodable and less than a maximum size. For each frame, the codec selects the segment duration and coding parameters, e.g., a particular entropy coder and its parameters for each segment, that minimizes the encoded payload for the entire frame subject to the constraints. Distinct sets of coding parameters may be selected for each channel or a global set of coding parameters may be selected for all channels. Compression performance may be further enhanced by forming M/2 decorrelation channels for M-channel audio. The triplet of channels (basis, correlated, decorrelated) provides two possible pair combinations (basis, correlated) and (basis, decorrelated) that can be considered during the segmentation and entropy coding optimization to further improve compression performance.Type: GrantFiled: December 19, 2007Date of Patent: August 7, 2012Assignee: DTS, Inc.Inventor: Zoran Fejzo
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Publication number: 20110224991Abstract: An audio codec losslessly encodes audio data into a sequence of analysis windows in a scalable bitstream. This is suitably done by separating the audio data into MSB and LSB portions and encoding each with a different lossless algorithm. An authoring tool compares the buffered payload to an allowed payload for each window and selectively scales the losslessly encoded audio data, suitably the LSB portion, in the non-conforming windows to reduce the encoded payload, hence buffered payload. This approach satisfies the media bit rate and buffer capacity constraints without having to filter the original audio data, reencode or otherwise disrupt the lossless bitstream.Type: ApplicationFiled: March 9, 2010Publication date: September 15, 2011Inventor: Zoran Fejzo
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Publication number: 20110106546Abstract: An audio codec losslessly encodes audio data into a sequence of analysis windows in a scalable bitstream. This is suitably done by separating the audio data into MSB and LSB portions and encoding each with a different lossless algorithm. An authoring tool compares the buffered payload to an allowed payload for each window and selectively scales the losslessly encoded audio data, suitably the LSB portion, in the non-conforming windows to reduce the encoded payload, hence buffered payload. This approach satisfies the media bit rate and buffer capacity constraints without having to filter the original audio data, reencode or otherwise disrupt the lossless bitstream.Type: ApplicationFiled: March 9, 2010Publication date: May 5, 2011Inventor: Zoran Fejzo
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Patent number: 7930184Abstract: A lossless audio codec encodes/decodes a lossless variable bit rate (VBR) bitstream with random access point (RAP) capability to initiate lossless decoding at a specified segment within a frame and/or multiple prediction parameter set (MPPS) capability partitioned to mitigate transient effects. This is accomplished with an adaptive segmentation technique that fixes segment start points based on constraints imposed by the existence of a desired RAP and/or detected transient in the frame and selects a optimum segment duration in each frame to reduce encoded frame payload subject to an encoded segment payload constraint. In general, the boundary constraints specify that a desired RAP or detected transient must lie within a certain number of analysis blocks of a segment start point.Type: GrantFiled: January 30, 2008Date of Patent: April 19, 2011Assignee: DTS, Inc.Inventor: Zoran Fejzo
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Publication number: 20100082352Abstract: An audio codec losslessly encodes audio data into a sequence of analysis windows in a scalable bitstream. This is suitably done by separating the audio data into MSB and LSB portions and encoding each with a different lossless algorithm. An authoring tool compares the buffered payload to an allowed payload for each window and selectively scales the losslessly encoded audio data, suitably the LSB portion, in the non-conforming windows to reduce the encoded payload, hence buffered payload. This approach satisfies the media bit rate and buffer capacity constraints without having to filter the original audio data, reencode or otherwise disrupt the lossless bitstream.Type: ApplicationFiled: November 5, 2009Publication date: April 1, 2010Inventor: Zoran Fejzo
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Patent number: 7668723Abstract: An audio codec losslessly encodes audio data into a sequence of analysis windows in a scalable bitstream. This is suitably done by separating the audio data into MSB and LSB portions and encoding each with a different lossless algorithm. An authoring tool compares the buffered payload to an allowed payload for each window and selectively scales the losslessly encoded audio data, suitably the LSB portion, in the non-conforming windows to reduce the encoded payload, hence buffered payload. This approach satisfies the media bit rate and buffer capacity constraints without having to filter the original audio data, reencode or otherwise disrupt the lossless bitstream.Type: GrantFiled: August 14, 2007Date of Patent: February 23, 2010Assignee: DTS, Inc.Inventor: Zoran Fejzo
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Publication number: 20090164223Abstract: A lossless audio codec segments audio data within each frame to improve compression performance subject to a constraint that each segment must be fully decodable and less than a maximum size. For each frame, the codec selects the segment duration and coding parameters, e.g., a particular entropy coder and its parameters for each segment, that minimizes the encoded payload for the entire frame subject to the constraints. Distinct sets of coding parameters may be selected for each channel or a global set of coding parameters may be selected for all channels. Compression performance may be further enhanced by forming M/2 decorrelation channels for M-channel audio. The triplet of channels (basis, correlated, decorrelated) provides two possible pair combinations (basis, correlated) and (basis, decorrelated) that can be considered during the segmentation and entropy coding optimization to further improve compression performance.Type: ApplicationFiled: December 19, 2007Publication date: June 25, 2009Inventor: Zoran Fejzo
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Publication number: 20090164224Abstract: A lossless audio codec segments audio data within each frame to improve compression performance subject to a constraint that each segment must be fully decodable and less than a maximum size. For each frame, the codec selects the segment duration and coding parameters, e.g., a particular entropy coder and its parameters for each segment, that minimizes the encoded payload for the entire frame subject to the constraints. Distinct sets of coding parameters may be selected for each channel or a global set of coding parameters may be selected for all channels. Compression performance may be further enhanced by forming M/2 decorrelation channels for M-channel audio. The triplet of channels (basis, correlated, decorrelated) provides two possible pair combinations (basis, correlated) and (basis, decorrelated) that can be considered during the segmentation and entropy coding optimization to further improve compression performance.Type: ApplicationFiled: December 19, 2007Publication date: June 25, 2009Inventor: Zoran Fejzo
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Publication number: 20080215317Abstract: A lossless audio codec encodes/decodes a lossless variable bit rate (VBR) bitstream with random access point (RAP) capability to initiate lossless decoding at a specified segment within a frame and/or multiple prediction parameter set (MPPS) capability partitioned to mitigate transient effects. This is accomplished with an adaptive segmentation technique that fixes segment start points based on constraints imposed by the existence of a desired RAP and/or detected transient in the frame and selects a optimum segment duration in each frame to reduce encoded frame payload subject to an encoded segment payload constraint. In general, the boundary constraints specify that a desired RAP or detected transient must lie within a certain number of analysis blocks of a segment start point.Type: ApplicationFiled: January 30, 2008Publication date: September 4, 2008Inventor: Zoran Fejzo
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Patent number: 7392195Abstract: A lossless audio codec segments audio data within each frame to improve compression performance subject to a constraint that each segment must be fully decodable and less than a maximum size. For each frame, the codec selects the segment duration and coding parameters, e.g., a particular entropy coder and its parameters for each segment, that minimizes the encoded payload for the entire frame subject to the constraints. Distinct sets of coding parameters may be selected for each channel or a global set of coding parameters may be selected for all channels. Compression performance may be further enhanced by forming M/2 decorrelation channels for M-channel audio. The triplet of channels (basis, correlated, decorrelated) provides two possible pair combinations (basis, correlated) and (basis, decorrelated) that can be considered during the segmentation and entropy coding optimization to further improve compression performance.Type: GrantFiled: August 4, 2004Date of Patent: June 24, 2008Inventor: Zoran Fejzo
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Publication number: 20080021712Abstract: An audio codec losslessly encodes audio data into a sequence of analysis windows in a scalable bitstream. This is suitably done by separating the audio data into MSB and LSB portions and encoding each with a different lossless algorithm. An authoring tool compares the buffered payload to an allowed payload for each window and selectively scales the losslessly encoded audio data, suitably the LSB portion, in the non-conforming windows to reduce the encoded payload, hence buffered payload. This approach satisfies the media bit rate and buffer capacity constraints without having to filter the original audio data, reencode or otherwise disrupt the lossless bitstream.Type: ApplicationFiled: August 14, 2007Publication date: January 24, 2008Inventor: Zoran Fejzo