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Zoran Fejzo

from Los Angeles, CA
Age ~61
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Also known as:
Fejzo Zoran
Phone and address:
12206 Tweed Ln, Los Angeles, CA 90049
(310) 471-4352
Related to:
Jennifer Rachael Childs, 54Maya Feyzo PearlmanJasenka Feyzo, 66
Connected to:
Edona Fejzullahu, 42Mandy Feka, 59Michelle A Fekade, 45Dawit Fekadu, 42Christine L Fekay, 61Meron Z Fekede, 45Stacey W Feken, 50Adam P Fekete, 46Althea F Fekete, 83Annette M Fekete, 58

Zoran Fejzo Phones & Addresses

  • 12206 Tweed Ln, Los Angeles, CA 90049 (310) 471-4352
  • San Francisco, CA
  • Boston, MA
  • Austin, TX
  • Berkeley, CA

Work

Company: Dts, inc. Oct 2015 Position: Senior fellow, r and d

Education

Degree: Doctorates, Doctor of Philosophy School / High School: Northeastern University 1990 to 1994 Specialities: Electronics, Electronics Engineering, Philosophy

Skills

Consumer Electronics • Audio Processing • Embedded Software • Embedded Systems • Audio Codecs • Digital Signal Processing • Acoustics • System Architecture • Firmware • Audio Signal Processing • Adaptive Signal Processing • Electronics • Product Management • C • Design and Specification of Audio Techol...

Industries

Consumer Electronics

Resumes

Resumes

Zoran Fejzo Photo 1

Senior Fellow, R And D

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Location:
Los Angeles, CA
Industry:
Consumer Electronics
Work:
Dts, Inc.
Senior Fellow, R and D
Dts, Inc. Jan 2010 - Oct 2015
Vice President Research and Development
Dts, Inc. Jan 2004 - Dec 2009
Chief Technologist
Dts, Inc. Oct 1998 - Dec 2003
Principal R and D Engineer
Tcsi Aug 1995 - Aug 1998
Senior R and D Engineer
Education:
Northeastern University 1990 - 1994
Doctorates, Doctor of Philosophy, Electronics, Electronics Engineering, Philosophy University of Belgrade 1983 - 1988
Bachelors, Electronics Engineering, Electronics
Skills:
Consumer Electronics
Audio Processing
Embedded Software
Embedded Systems
Audio Codecs
Digital Signal Processing
Acoustics
System Architecture
Firmware
Audio Signal Processing
Adaptive Signal Processing
Electronics
Product Management
C
Design and Specification of Audio Techologies

Publications

Us Patents

Scalable Lossless Audio Codec And Authoring Tool

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US Patent:
7272567, Sep 18, 2007
Filed:
Aug 4, 2004
Appl. No.:
10/911062
Inventors:
Zoran Fejzo - Los Angeles CA, US
International Classification:
H04B 1/66
H03M 7/30
US Classification:
704500, 341 51, 375241, 704501
Abstract:
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.

Lossless Multi-Channel Audio Codec

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US Patent:
7392195, Jun 24, 2008
Filed:
Aug 4, 2004
Appl. No.:
10/911067
Inventors:
Zoran Fejzo - Los Angeles CA, US
International Classification:
H04R 5/02
H04B 1/66
US Classification:
704500, 381 2, 381 20
Abstract:
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.

Scalable Lossless Audio Codec And Authoring Tool

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US Patent:
7668723, Feb 23, 2010
Filed:
Aug 14, 2007
Appl. No.:
11/891905
Inventors:
Zoran Fejzo - Los Angeles CA, US
Assignee:
DTS, Inc. - Calabasas CA
International Classification:
H04B 1/66
G10L 19/00
US Classification:
704500, 704501, 370477
Abstract:
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.

Multi-Channel Audio Coding/Decoding Of Random Access Points And Transients

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US Patent:
7930184, Apr 19, 2011
Filed:
Jan 30, 2008
Appl. No.:
12/011899
Inventors:
Zoran Fejzo - Los Angeles CA, US
Assignee:
DTS, Inc. - Calabasas CA
International Classification:
G10L 19/04
US Classification:
704500, 704216, 704219, 381 2, 370470
Abstract:
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.

Lossless Multi-Channel Audio Codec

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US Patent:
8239210, Aug 7, 2012
Filed:
Dec 19, 2007
Appl. No.:
12/004334
Inventors:
Zoran Fejzo - Los Angeles CA, US
Assignee:
DTS, Inc. - Calabasas CA
International Classification:
G10L 21/04
G10L 21/00
US Classification:
704504, 704503
Abstract:
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.

Scalable Lossless Audio Codec And Authoring Tool

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US Patent:
8374858, Feb 12, 2013
Filed:
Mar 9, 2010
Appl. No.:
12/720365
Inventors:
Zoran Fejzo - Los Angeles CA, US
Assignee:
DTS, Inc. - Calabasas CA
International Classification:
G10L 19/00
G10L 19/14
G10L 11/04
G10L 19/02
G10L 21/04
US Classification:
704229, 7042001, 704205, 704206, 704212, 704501, 704502, 704503, 704504
Abstract:
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.

Lossless Multi-Channel Audio Codec

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US Patent:
20090164223, Jun 25, 2009
Filed:
Dec 19, 2007
Appl. No.:
12/004333
Inventors:
Zoran Fejzo - Los Angeles CA, US
International Classification:
G10L 19/00
US Classification:
704500, 704E19004
Abstract:
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.

Scalable Lossless Audio Codec And Authoring Tool

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US Patent:
20100082352, Apr 1, 2010
Filed:
Nov 5, 2009
Appl. No.:
12/613316
Inventors:
Zoran Fejzo - Los Angeles CA, US
International Classification:
G10L 19/00
G10L 21/00
US Classification:
704500, 375240, 370477, 704E19004, 704E19044
Abstract:
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.
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Zoran Fejzo from Los Angeles, CA, age ~61 Get Report