IP Library › Granted Patent US 12,394,425
Granted Patent B2
US 12,394,425 · App. 18/334,853 · Granted Aug 19, 2025

Methods, encoder and decoder for linear predictive encoding and decoding of sound signals upon transition between frames having different sampling rates

Inventors: Redwan Salami (Saint-Laurent, CA); Vaclav Eksler (Sherbrooke, CA)
Assignee: VOICEAGE EVS LLC
G10L19/12G10L19/06G10L19/167G10L19/173G10L19/24G10L19/26G10L25/06G10L2019/0002G10L2019/0004G10L2019/0016G10L19/07G10L21/038
View Patent ↗
Loading inventors, assignments & file history…
Monitor This Case
Get email alerts when status or documents change.
Order Certified Copies
Most orders are placed with the USPTO same day — all within 24 business hours.
Order via The Patent Place →
Pre-filled with this patent's details
Quick Facts
Patent No.
US 12,394,425
App. No.
18/334,853
Granted
Aug 19, 2025
Kind
B2
Abstract

Methods, an encoder and a decoder are configured for transition between frames with different internal sampling rates. Linear predictive (LP) filter parameters are converted from a sampling rate S 1 to a sampling rate S 2 . A power spectrum of a LP synthesis filter is computed, at the sampling rate S 1 , using the LP filter parameters. The power spectrum of the LP synthesis filter is modified to convert it from the sampling rate S 1 to the sampling rate S 2 . The modified power spectrum of the LP synthesis filter is inverse transformed to determine autocorrelations of the LP synthesis filter at the sampling rate S 2 . The autocorrelations are used to compute the LP filter parameters at the sampling rate S 2.

Claims (39)

1. A method for interpolating LP filter parameters in a current sound signal processing frame following a previous sound signal processing frame, the previous frame using an internal sampling rate S 1 and the current frame using an internal sampling rate S 2 and defining a number of subframes, comprising:

providing LP filter parameters of the previous frame at the internal sampling rate S 1 ;

providing LP filter parameters of the current frame at the internal sampling rate S 2 ;

converting the LP filter parameters of the previous frame from the internal sampling rate S 1 to the internal sampling rate S 2 , comprising:

computing, at the internal sampling rate S 1 , a power spectrum of an LP synthesis filter using the LP filter parameters of the previous frame;

modifying the power spectrum of the LP synthesis filter to convert it from the internal sampling rate S 1 to the internal sampling rate S 2 ;

inverse transforming the modified power spectrum of the LP synthesis filter to determine autocorrelations of the LP synthesis filter at the internal sampling rate S 2 ; and

using the autocorrelations to compute the LP filter parameters of the previous frame at the internal sampling rate S 2 ;

determining for at least one subframe of the current frame interpolated LP filter parameters by interpolation between the LP filter parameters of the current frame at the internal sampling rate S 2 and the LP filter parameters of the previous frame converted from the internal sampling rate S 1 to the internal sampling rate S 2 .

2. The method for interpolating LP filter parameters according to claim 1 , further comprising:

for determining the interpolated LP filter parameters, using a weighted sum of the LP filter parameters of the current frame at the internal sampling rate S 2 and the LP filter parameters of the previous frame at the internal sampling rate S 2 .

3. The method for interpolating LP filter parameters according to claim 1 , wherein the LP filter parameters are quantized LP filter parameters.

4. The method for interpolating LP filter parameters according to claim 1 , further comprising:

transforming the LP filter parameters in a quantization and interpolation domain.

5. The method for interpolating LP filter parameters according to claim 4 , wherein the quantization and interpolation domain is a line spectrum frequencies domain.

6. The method for interpolating LP filter parameters according to claim 1 , wherein modifying the power spectrum of the LP synthesis filter to convert it from the internal sampling rate S 1 to the internal sampling rate S 2 comprises:

if S 1 is less than S 2 , extending the power spectrum of the LP synthesis filter based on a ratio between S 1 and S 2 ;

if S 1 is larger than S 2 , truncating the power spectrum of the LP synthesis filter based on the ratio between S 1 and S 2 .

7. The method for interpolating LP filter parameters according to claim 1 , further comprising:

inverse transforming the modified power spectrum of the LP synthesis filter by using an inverse discrete Fourier Transform.

8. A device for interpolating LP filter parameters in a current sound signal processing frame following a previous sound signal processing frame, the previous frame using an internal sampling rate S 1 and the current frame using an internal sampling rate S 2 and defining a number of subframes, comprising:

at least one processor; and

a memory coupled to the processor and storing non-transitory instructions that when executed cause the processor to:

provide LP filter parameters of the previous frame at the internal sampling rate S 1 ;

provide LP filter parameters of the current frame at the internal sampling rate S 2 ;

for converting the LP filter parameters of the previous frame from the internal sampling rate S 1 to the internal sampling rate S 2 :

compute, at the internal sampling rate S 1 , a power spectrum of an LP synthesis filter using the LP filter parameters of the previous frame;

modify the power spectrum of the LP synthesis filter to convert it from the internal sampling rate S 1 to the internal sampling rate S 2 ;

inverse transform the modified power spectrum of the LP synthesis filter to determine autocorrelations of the LP synthesis filter at the internal sampling rate S 2 ; and

use the autocorrelations to compute the LP filter parameters of the previous frame at the internal sampling rate S 2 ;

determine for at least one subframe of the current frame interpolated LP filter parameters by interpolation between the LP filter parameters of the current frame at the internal sampling rate S 2 and the LP filter parameters of the previous frame converted from the internal sampling rate S 1 to the internal sampling rate S 2 .

9. The device for interpolating LP filter parameters according to claim 8 , wherein, to determine the interpolated LP filter parameters, the processor is configured to use a weighted sum of the LP filter parameters from the current frame at the internal sampling rate S 2 and the LP filter parameters from the previous frame at the internal sampling rate S 2 .

10. The device for interpolating LP filter parameters according to claim 8 , wherein the LP filter parameters are quantized LP filter parameters.

11. The device for interpolating LP filter parameters according to claim 8 , wherein the processor is configured to transform the LP filter parameters in a quantization and interpolation domain.

12. The device for interpolating LP filter parameters according to claim 11 , wherein the quantization and interpolation domain is a line spectrum frequencies domain.

13. The device for interpolating LP filter parameters according to claim 8 , wherein, to modify the power spectrum of the LP synthesis filter to convert it from the internal sampling rate S 1 to the internal sampling rate S 2 , the processor is configured to:

if S 1 is less than S 2 , extend the power spectrum of the LP synthesis filter based on a ratio between S 1 and S 2 ;

if S 1 is larger than S 2 , truncate the power spectrum of the LP synthesis filter based on the ratio between S 1 and S 2 .

14. The device for interpolating LP filter parameters according to claim 8 , wherein, to inverse transform the modified power spectrum of the LP synthesis filter, the processor is configured to use an inverse discrete Fourier Transform.

Assignments (2)
ASSIGNMENT OF ASSIGNOR'S INTEREST Recorded Jun 14, 2023
From: SALAMI, REDWAN; EKSLER, VACLAV
To: VOICEAGE CORPORATION
Reel/Frame 063951/0814 →
ASSIGNMENT OF ASSIGNOR'S INTEREST Recorded Jun 14, 2023
From: VOICEAGE CORPORATION
To: VOICEAGE EVS LLC
Reel/Frame 063951/0823 →
Continuity (7)
Continuation 17444799 · Aug 10, 2021
Continuation 16594245 · Oct 7, 2019
Continuation 15815304 · Nov 16, 2017
Continuation 15814083 · Nov 15, 2017
Continuation 14677672 · Apr 2, 2015
Provisional Application 61980865 · Apr 17, 2014
Related Publication 20230326472A1 · Oct 12, 2023
References Cited (185)
US 4980916A · Zinser · 1990 [cited by applicant]
US 5241692A · Harrison et al. · 1993 [cited by applicant]
US 5651090A · Moriya et al. · 1997 [cited by applicant]
US 5657350A · Hofmann · 1997 [cited by applicant]
US 5673286A · Lomp · 1997 [cited by applicant]
US 5673364A · Bialik · 1997 [cited by applicant]
US 5684920A · Iwakami et al. · 1997 [cited by applicant]
US 5864797A · Fujumoto · 1999 [cited by applicant]
US 5867814A · Yong · 1999 [cited by applicant]
US 5873059A · Iijima et al. · 1999 [cited by applicant]
US 5920832A · Wuppermann et al. · 1999 [cited by applicant]
US 6134518A · Cohen et al. · 2000 [cited by applicant]
US 6233550B1 · Gersho et al. · 2001 [cited by applicant]
US 6311154B1 · Gersho et al. · 2001 [cited by applicant]
US 6475245B2 · Gersho et al. · 2002 [cited by applicant]
US 6502069B1 · Grill et al. · 2002 [cited by applicant]
US 6636829B1 · Benyassine et al. · 2003 [cited by applicant]
US 6650258B1 · Kelly · 2003 [cited by applicant]
US 6691082B1 · Aguilar et al. · 2004 [cited by applicant]
US 6732070B1 · Rotola-Pukkita et al. · 2004 [cited by applicant]
US 6757654B1 · Westerlund et al. · 2004 [cited by applicant]
US 6873954B1 · Sundqvist et al. · 2005 [cited by applicant]
US 7106228B2 · Bessette et al. · 2006 [cited by applicant]
US 7337110B2 · Jasiuk · 2008 [cited by applicant]
US 7457742B2 · Kovesi et al. · 2008 [cited by applicant]
US 7529660B2 · Bessette et al. · 2009 [cited by applicant]
US 7693710B2 · Jelinek · 2010 [cited by applicant]
US 8315863B2 · Oshikiri · 2012 [cited by applicant]
US 8401843B2 · Eksler et al. · 2013 [cited by applicant]
US 8589151B2 · Chamberlain · 2013 [cited by applicant]
US 9053705B2 · Bessette · 2015 [cited by applicant]
US 9852741B2 · Salami et al. · 2017 [cited by applicant]
US 10431233B2 · Salami · 2019 [cited by examiner]
US 10468045B2 · Salami · 2019 [cited by examiner]
US 11282530B2 · Salami · 2022 [cited by examiner]
US 11721349B2 · Salami · 2023 [cited by examiner]
US 20010027390A1 · Rotola-Pukkila et al. · 2001 [cited by applicant]
US 20020123886A1 · Globerson · 2002 [cited by applicant]
US 20030177004A1 · Jabri · 2003 [cited by applicant]
US 20040071132A1 · Sundqvist et al. · 2004 [cited by applicant]
US 20060235685A1 · Nurminen et al. · 2006 [cited by applicant]
US 20060280271A1 · Oshikiri · 2006 [cited by applicant]
US 20080040105A1 · Wang et al. · 2008 [cited by applicant]
US 20080077401A1 · Jabri et al. · 2008 [cited by applicant]
US 20080079861A1 · Seo et al. · 2008 [cited by applicant]
US 20080120098A1 · Makinen et al. · 2008 [cited by applicant]
US 20090216527A1 · Oshikiri · 2009 [cited by applicant]
US 20090234644A1 · Reznik et al. · 2009 [cited by applicant]
US 20100161321A1 · Oshikiri · 2010 [cited by applicant]
US 20100250263A1 · Miseki · 2010 [cited by applicant]
US 20100280831A1 · Salami et al. · 2010 [cited by applicant]
US 20110010168A1 · Yu et al. · 2011 [cited by applicant]
US 20110200198A1 · Grill et al. · 2011 [cited by applicant]
US 20120095756A1 · Sung et al. · 2012 [cited by applicant]
US 20120095758A1 · Gibbs et al. · 2012 [cited by applicant]
US 20120116769A1 · Malah et al. · 2012 [cited by applicant]
US 20130151262A1 · Lohwasser et al. · 2013 [cited by applicant]
US 20130308792A1 · Gao · 2013 [cited by applicant]
US 20130332153A1 · Markovic et al. · 2013 [cited by applicant]
US 20140236588A1 · Subasingha et al. · 2014 [cited by applicant]
US 20140330415A1 · Ramo et al. · 2014 [cited by applicant]
US 20170053655A1 · Naka · 2017 [cited by examiner]
US 20170154635A1 · Doehla et al. · 2017 [cited by applicant]
CA 2979857 · 2013 [cited by applicant]
CN 1167308 · 1997 [cited by applicant]
CN 1391689 · 2003 [cited by applicant]
CN 1677492 · 2005 [cited by applicant]
CN 1701353 · 2005 [cited by applicant]
CN 101320566 · 2008 [cited by applicant]
CN 101578508 · 2009 [cited by applicant]
CN 101853240 · 2010 [cited by applicant]
CN 103187066 · 2013 [cited by applicant]
CN 103235288 · 2013 [cited by applicant]
EP 078083 · 1997 [cited by applicant]
EP 1785985 · 2007 [cited by applicant]
EP 2302345 · 2011 [cited by applicant]
EP 3136384 · 2017 [cited by applicant]
GB 1533337 · 1978 [cited by applicant]
JP S5994796 · 1994 [cited by applicant]
JP 2000206998 · 2000 [cited by applicant]
JP 2002251029 · 2002 [cited by applicant]
JP 2003108196 · 2003 [cited by applicant]
JP 2004289196 · 2004 [cited by applicant]
JP 2004320088 · 2004 [cited by applicant]
JP 2009508146 · 2009 [cited by applicant]
JP 2011247615 · 2011 [cited by applicant]
JP 2013541737 · 2013 [cited by applicant]
JP 2014090781 · 2014 [cited by applicant]
RU 2483365 · 2012 [cited by applicant]
WO 0057401 · 2000 [cited by applicant]
WO 2004010603 · 2004 [cited by applicant]
WO 2006129166 · 2006 [cited by applicant]
WO 2006130226 · 2006 [cited by applicant]
WO 2008049221 · 2008 [cited by applicant]
WO 2005104095 · 2009 [cited by applicant]
WO 2012103686 · 2012 [cited by applicant]
WO 2012110481 · 2012 [cited by applicant]
3GPP Technical Specification 26.190, 3rd Generation Partnership Project; Technical Specification Group Services and System Aspected; Speech codec speech processing functions; Adaptive Multi-Rate-Wideband (AMR-WB) speech… [cited by applicant]
3GPP TS 26.190 V6.1.1 (Jul. 2005), “Technical Specification Group Services and System Aspects; Speech codec speech processing functions; Adaptive Multi-Rate—Wideband (AMR-WB) speech codec; Transcoding functions”, Releas… [cited by applicant]
3GPP TS 26.190 V11.0.0 (Sep. 2012), “Technical Specification Group Services and System Aspects; Speech codec speech processing functions; Adaptive Multi-Rate—Wideband (AMR-WB) speech codec; Transcoding functions”, Relea… [cited by applicant]
Anandakumar et al., “Efficient, CELP-Based Diversity Schemes for VOIP”, IEEE, 2000, pp. 3682-3685. [cited by applicant]
“AMR Wideband speech codec”, 3GPP TS 26.190, Version 5.1.0, release 5, Dec. 2001, 3 sheets. [cited by applicant]
Bello et al., “A Tutorial on Onset Detection in Music Signals”, IEEE Transactions on Speech and Audio Processing, vol. 13, No. 5, Sep. 2005, pp. 1035-1047. [cited by applicant]
Bergström et al., “High Temporal Resolution in Multi-Pulse Coding”, IEEE, 1989, pp. 770-773. [cited by applicant]
Bergström et al., “Code-Book Driven Glottal Pulse Analysis”, IEEE, 1989, pp. 53-56. [cited by applicant]
Berouti et al., “Enhancement of Speech Corrupted by Acoustic Noise”, IEEE, 1979, pp. 208-211. [cited by applicant]
Bessette et al., “The Adaptive Multirate Wideband Speech Codec (AMR-WB)”, IEEE Transactions on Speech and Audio Processing, vol. 10, No. 8, Nov. 2002, pp. 620-636. [cited by applicant]
Bessette et al. “Proposed CE for extending the LPD mode in USAC”, International Organisation for Standardisation ISO/IEC JTC1/SC29/WG11 Coding of Moving Pictures and Audio, Oct. 2010, 4 sheets. [cited by applicant]
Bessette et al., “A Wideband Spech and Audio Codec at 16/24/32 Kbit/s Using Hybrid ACELP/TCX Techniques”, IEEE, 1999, pp. 7-9. [cited by applicant]
Bi et al., “Sampling Rate Conversion in the Frequency Domain”; DSP Tips & Tricks, IEEE Signal Processing Magazine, No. 140, May 2011, pp. 140-144. [cited by applicant]
Bhaskar, “Adaptive Predictive Coding with Transform Domain Quantization”, ISBN 0-7923-9345-7, Kluwer Academic Publishers, Speech and Audio Coding for Wireless and Network Applications, 1993, 7 sheets. [cited by applicant]
Brigham, “The Fast Fourier Transform and its Applications”, Prentice-Hall, Apr. 1988, pp. 198-199. [cited by applicant]
Brigham, “The Fast Fourier Transform and Its Applications,” Prentice-Hall International Editions, ISBN 0-13-307505-2, 1988, 8 sheets. [cited by applicant]
Boll, “Supression of Acoustic Noise in Speech Using Spectral Subtraction”, IEEE Trasactions on Acoustics, Speech, and Signal Processing, vol. ASSP-27, No. 2, Apr. 1979, pp. 113-120. [cited by applicant]
Cano et al., “A Review of Audio Fingerprinting”, Journal of VLSI Signal Processing 41, 2005, pp. 271-284. [cited by applicant]
“EVS Permanent Document#4 (EVS-4): EVS design constraints”, 3GPP TSG-SA4#74 meeting, Tdoc S4 (13)0778, Jul. 8-12, 2013, Dublin, Ireland, 7 sheets. [cited by applicant]
ETSI TS 126 190 V5.1.0 Technical Specification. Universal Mobile Telecommunications System (UMTS); Mandatory Speech Codec Speech Processing Functions AMR Wideband Speech Codecs; Transcoding Functions, 3GPP TS 26.190 Ver… [cited by applicant]
Foote, “Automatic Audio Segmentation Using a Measure of Audio Novelty”, IEEE, 2000, pp. 452-455. [cited by applicant]
Frohberg et al., “Pocket Book of Communication Engineering,” Specialist Book Publishing House Leipzig in the Carl Hanser Publishing House, ISBN 978-3-446-41602-4, 2008, 6 sheets. [cited by applicant]
Gersen et al., “Techniques for Improving the Performance of CELP-Type Speech Coders”, IEEE Journal on Selected Areas in Communications, vol. 10, No. 5, Jun. 1992, pp. 858-865. [cited by applicant]
Gersho, “Chapter 3, Speech Coding”, Center for Information Processing Research Dept. of Electrical & Computer Engineering, University of California, Santa Barbara, CA 93106, USA, 1992, pp. 73-100. [cited by applicant]
Gersho, “Concepts and Paradigms in Speech Coding”, Center for Information Processing Research, Department of Electrical and Computer Engineering, University of California, Santa Barbara, CA, California, 1995, pp. 369-38… [cited by applicant]
Hasegawa-Johnson et al., “Speech Coding: Fundamentals and Applications. Handbook on Telecommunications”, Copyright © 2003 by John Wiley and Sons, Inc., pp. 1-33. [cited by applicant]
Hawley, “Structure out of Sound”, Massachusetts Institute of Technology, 1993, 185 sheets. [cited by applicant]
Hosseinzadeh et al., “Combining Vocal Source and MFCC Features for Enhanced Speaker Recognition Performance Using GMMs”, IEEE, 2007, pp. 365-368. [cited by applicant]
Ince, “Digital Speech Processing—Speech Coding, Synthesis and Recognition”, ISBN 0-7923-9220-5, Kluwer Academic Publishers, 1992, 9 sheets. [cited by applicant]
Islam et al., “Partial-Energy Weighted Interpolation of Linear Prediction Coefficients”, Proc. IEEE Workshop Speech Coding, Delavan, WI, Sep. 2000, 3 sheets. [cited by applicant]
ITU-T Recommendations G.729, Series G: Transmission Systems and Media, Digital Systems and Networks, Digital terminal equipments—Coding of analogue signals by methods other than PCM, Coding of Speech at 8kbit/s using co… [cited by applicant]
ITU-T Recommendations G.729, Coding of speech at 8 kbit/s using conjugate structure algebraic-code-excited linear prediction (CS-ACELP), Jan. 2007, 146 sheets. [cited by applicant]
Jelinek et al., “Noise Reduction Method for Wideband Speech Coding”, 2004 12th European Signal Processing Conference, 2004, pp. 1959-1962. [cited by applicant]
Jury Trial Demanded, [cited by applicant]
Kim, “Adaptive Encoding of Fixed Codebook in CELP Coders”, The Journal of the Acoustical Society of Korea, vol. 16, No. 3E, 1997, pp. 44-49. [cited by applicant]
Kubin et al., “Speech Watermarking for Analog Flat-Fading Bandpass Channels”, IEEE Transactions on Audio Speech and Language Processing, Dec. 2009, 15 sheets. [cited by applicant]
Lebart et al., “A New Method Based on Spectral substraction for the Suppression of Late Reverbation from Speech Signals. Presented at the 105th Convention Sep. 26-29, 1998, San Francisco, California”, AES, 1998, 13 shee… [cited by applicant]
Lindén et al., “Investigation on the Audibility of Glottal Parameter Variations in Speech Synthesis”, Proceedings of Eusipco-94, 1994, 4 sheets. [cited by applicant]
Lindén et al., “A Glottal Vocoder Employing Vector Quantization”, Proc. NORSIG-94, 1994, 4 sheets. [cited by applicant]
Lu et al., “Content Analysis for Audio Classification and Segmentation”, IEEE Transactions on Speech and Audio Processing, vol. 10, No. 7, Oct. 2002, pp. 504-516. [cited by applicant]
Lyons, “How to Interpolate in the Time-Domain by Zero-Padding in the Frequency Domain”, Published at: https:/ / dspguru.com/dsp/howtos/how-to-interpolate-in-time-domain-by-zero-padding-in-frequency-domain/, Version ff M… [cited by applicant]
Makhoul, “Linear Prediction: A Tutorial Review”, Proc. IEEE, vol. 63, Issue 4, Apr. 1975, pp. 561-580. [cited by applicant]
Makhoul, “Spectral Linear Prediction: Properties and Applications”, IEEE Trans. Acoustics, Speech, Signal Processing, vol. 23, Issue 3, Jun. 1975, pp. 283-296. [cited by applicant]
Makhoul et al., “Vector Quantization in Speech Coding”, Proceeding of the IEEE, vol. 73, No. 11, Nov. 1985, pp. 1551-1588. [cited by applicant]
Makhoul et al., “High-Frequency Regeneration in Speech Coding Systems”, IEEE, 1979, 4 sheets. [cited by applicant]
Makhoul, “Selective Linear Prediction and Analysis-by-Synthesis in Speech Analysis,” Bolt Beranek and Newman Inc., Report No. 2578, A.I. Report No. 13, Apr. 1974, 66 sheets. [cited by applicant]
Markel et al., “Linear Prediction of Speech,” Springer-Verlag Berlin Heidelberg New York, ISBN 3-540-07563-1, 1976., 29 sheets. [cited by applicant]
Markel et al., “Linear Prediction of Speech,” Springer-Verlag Berlin Heidelberg New York, ISBN 13: 978-3-642-66288-1, 1976, 7 sheets. [cited by applicant]
Malenovsky et al., “Improving the Detection Efficiency of the VMR-WB Vad Algorithm on Music Signals”, 16th European Signal Processing Conference (EUSIPCO 2008), Lausane, Switzerland, Aug. 25-29, 2008, 5 sheets. [cited by applicant]
Martin, “Spectral Substraction based on Minimum Statistics”, Proc. EUSIPCO 1994, pp. 1182-1185. [cited by applicant]
Martin, “Noise Power Spectral Density Estimation Based on Optimal Smoothing and Minimum Statistics”, IEEE Transactions on Speech and Audio Processing, vol. 9, No. 5, Jul. 2001, pp. 504-512. [cited by applicant]
McElroy et al., “Wideband Speech Coding Using Multiple Codebooks and Glottal Pulses”, IEEE, 1995, 4 sheets. [cited by applicant]
Miki et al., “Pitch Synchrounous Innovation CELP (PSI-CELP)”, Eurospeech 93, Berlin, Germany, Sep. 1993, 4 sheets. [cited by applicant]
Moreau et al., “Mixed Excitation CELP Coder”, Eurospeech 89, Paris, France, Sep. 1989, 4 sheets. [cited by applicant]
Ooi et al., “A Computationally Efficient Wavelet Transform CELP Coder”, IEEE, 1994, 4 sheets. [cited by applicant]
Paksoy, “Variable Rate Speech Coding With Phonetic Classification. A dissertation submitted in partial satisfaction of the requirements for the degree of Doctor of Philosophy”, 1994, 145 sheets. [cited by applicant]
Paksoy et al., “A variable-rate multimodal speech coder with gain-matched analysis-by-synthesis”, 1997 IEEE International Conference on Acoustics, Speech, and Signal Processing, vol. 2, 1997, pp. 751-754. [cited by applicant]
Paliwal et al., “Efficient vector quantization of LPC parameters at 24 bits/frame”, IEEE, 1991, pp. 661-664. [cited by applicant]
Paliwal et al., “Efficient vector quantization of LPC parameters at 24 bits/frame”, IEEE Transactions on Speech and Audio Processing, vol. 1, No. 1, Jan. 1993, pp. 3-14. [cited by applicant]
Rabiner et al., “Digital Processing of Speech Signals”, ISBN 0-13-213603-1, Prentice-Hall Signal Processing Series, 1978, pp. 174-179, 324-325, and 398-413. [cited by applicant]
Rabiner et al., “Digital Processing of Speech Signals”, Prentice-Hall Signal Processing Series, 1978, pp. 1-115. [cited by applicant]
Ramachandran et al., “The Use of Pitch Prediction in Speech Coding”, Kluwer Academic Publishers, Modern Methods of Speech Processing, 1995, 30 sheets. [cited by applicant]
Ramalingam et al., “Gaussian Mixture Modeling Using Short Time Fourier Transform Features for Audio Fingerprinting”, IEEE, 2005, 4 sheets. [cited by applicant]
Saure et al., “Moisture Measurement by FT-IR-Spectroscopy”, Drying Technology, vol. 12, No. 6, 1994, pp. 1427-1444. [cited by applicant]
Schafer et al., “Digital Representations of Speech Signals”, Proceeding of the IEEE, vol. 63, No. 4, Apr. 1975, pp. 662-677. [cited by applicant]
Scheirer et al., “Constructions and Evaluation of a Robust Multifeature Speech/Music Discriminator”, 1997 IEEE International Conference on Acoustics, Speech, and Signal Processing, vol. 2, 1997, pp. 1331-1334. [cited by applicant]
Schnitzler et al., “Wideband Speech Coding Using Forward / Backward Adaptive Prediction with Mixed Time / Frequency Domain Excitation”, IEEE, 1999, 3 sheets. [cited by applicant]
Schroeder et al., “Code-Excited Linear Prediction (CELP): High Quality Speech at Very Low Bit Rates”, Proceedings—ICSASSP, IEEE International Conference on Acoustics, Speech Signal Processing, May 1985, 5 sheets. [cited by applicant]
Serra, A system for sound analysis/transformation/synthesis based on a deterministic plus stochastic decomposition. A dissertation sumbmitted in partial fulfillment of the requirements for the degree of Doctor of Philos… [cited by applicant]
Serra et al., “Spectral Modeling Synthesis: A Sound Analysis/Synthesis System Based on a Deterministic plus Stochastic Decomposition”, Computer Music Journal, vol. 14, No. 4, 1990, pp. 12-24. [cited by applicant]
Serra et al., “Spectral modeling synthesis”, Proceedings of the 1989 International Computer Music Conference; Nov. 2-5, 1989; 1989, pp. 281-284. [cited by applicant]
Skoglund, “Analysis and Quantization of glottal pulse shapes”, Speech Communications, vol. 24, 1998, 133-152. [cited by applicant]
Sohn et al., “A Voice Activity Detector Employing Soft Decision Based Noise Spectrum Adaptation”, IEEE, 1998, pp. 365-368. [cited by applicant]
Taddei et al., “Efficient Coding of Transitional Speech Segments in CELP”, IEEE, 2002, pp. 14-16. [cited by applicant]
Telecommunication Standardization Sector of ITU “Low-Complexity, Full-Band Audio Coding for High-Quality, Conversational Applications,” Recommendation ITU-T G.719, Jun. 2008, 58 sheets. [cited by applicant]
Tdoc S4 (13) 0778, “EVS Permanent Document #4 (EVS-4): EVS design constraints”, Version 1.2, 3GPP TSG-SA4 #74 Meeting, Jul. 8-12, 2013, pp. 1-7. [cited by applicant]
Tzanetakis et al., “Musical Genre Classification of Audio Signals”, IEEE Transactions on Speech and Audio Processing, vol. 10, No. 5, Jul. 2002, pp. 293-302. [cited by applicant]
Tzanetakis et al., “MARSYAS: a framework for audio analysis”, Organised Sound, Cambridge University Press, vol. 4, No. 3, 1999, pp. 169-175. [cited by applicant]
Varho, “New linear predictive methods for digital speech processing”, Helsinki University of Technology, Laboratory of Acoustics and Audio Signal Processing, Espoo 2001, Report 58, 2001, 68 sheets. [cited by applicant]
Valin, et al., “Bandwidth Extension of Narrowband Speech for Low Bit-Rate Wideband Coding”, Proc. IEEE Speech Coding Workshop (Scw), Feb. 2000, Doi:10.1109/Scft.2000.878425, 3 sheets. [cited by applicant]
Valin, “Spectral Extension of a Speech Signal of the Voice Band to the Am Band”, University Sherbrooke, Dec. 2001, 68 sheets. [cited by applicant]
Vaidyanathan, “The Theory of Linear Prediction”, Morgan & Claypool Publishers, ISBN 91-598-29575-6, 2008, 23 sheets. [cited by applicant]
Vary, et al., “Digital Speech Transmission. Enhancement, Coding and Error Concealment,” John Wiley & Sons, ISBN 0-471-56018-9, 2006, 5 sheets. [cited by applicant]
Wang et al., “Improved Excitation for Phonetically-Segmented VXC Speech Coding Below 4 Kb/s”, IEEE, 1990, pp. 0946-0950. [cited by applicant]
Wartewig et al., “IR and Raman Spectroscopy: Fundamental Processing”, ISBN 3-527-30245-X, Wiley-VCH Verlag GmbH & Co. KGaA, 2003, 67 sheets. [cited by applicant]
Westerlund et al., “Low Distorsion SNR-Based Speech Enhancement Employing Critical Band Filter Banks”, IEEE, 2003, pp. 129-133. [cited by applicant]
Zhang, “Code excited linear predicition with multi-pulse codebooks. A Thesis sumbmitted in partial fulfillment of the requirements for the degree of Master of Applied Science”, Simon Fraser University, 1997, 104 sheets. [cited by applicant]
Zhang et al., “A CELP variable rate speech codec with low average rate”, 1997 IEEE International Conference on Acoustics, Speech, and Signal Processing, vol. 2, 1997, pp. 735-738. [cited by applicant]