Welcome to Extended Education

Speech and Audio Coding for Communications and Multimedia

Submitted by sputtagu on Tue, 2010-10-19 15:56
Program Summary
Program Length: 
2 days
Program Fee: 

$895

Location: 
Arizona State University - Tempe Arizona
Overview: 

This two-day course describes the fundamental principles, techniques, and algorithms used in current applications; including a detailed discussion of current speech coding and telecommunication standards. The course starts with a discussion on the basic speech representation methods, the performance measures used to evaluate coded speech, and the role of the standards. Algorithm descriptions include ADPCM, sub-band coding, adaptive transform coding sinusoidal transform coding (STC) multiband excitation (MBE) coding linear predictive coding (LPC) residual excited LPC analysis-by-synthesis LPC (multipulse LPC, regular pulse LPC, code excited LPC).The program includes lecture, audio, and computer demonstrations of recent speech coding standards including voice-over IP algorithms. Descriptions of wideband audio standards such as the AC-3, the MPEG (MP3) audio layers, and the Sony systems algorithms will also be covered. Participants will get a copy of all the viewgraphs as well as a package of detailed notes for the course. Bonus: Participants will also get an opportunity to experiment with software for the FS1015 LPC-10 and the FS1016 CELP algorithms. Participants take back a copy of this software to the work place.

Students will achieve the following learning outcomes;

  • IS-xx series vocoders for digital cellular applications

  • Speech and Audio Coding Standards for Multimedia, Teleconferencing, and Voice-Over-IP (VoIP)

  • G.xxx and H.xxx series ITU Vocoder Standards for Communications and Multimedia Applications

  • Audio Standards for Hi-Fi Applications, Streaming Audio, Cinema, MP3, AC-3, SDDS etc

  • the CELP, ACELP, RCELP, SMV, EVRC, LPC, STC, IMBE (IRIDIUM) algorithms

  • New standards CDMA 2000 - SMV, Adaptive Multirate GSM, ITU-4

Who Should Attend

The course is designed for engineers and managers who need to understand emerging speech coding techniques and telecommunication standards. The course should be of particular interest to engineers of telecommunication and computer companies who are evaluating new digital systems for mobile telephones, multimedia computing, and wireless teleconferencing. Participants should have an understanding of basic engineering mathematics. Individuals may want to take the DSP course to obtain a broader perspective on signal and speech processing.

Topics

  • General: characterizations of voiced and unvoiced speech, analysis/synthesis models, performance measures and complexity issues

  • DSP Background: digital filters for speech processing, random signals, autocorrelation and covariance methods, FFT processing of speech

  • Waveform Coders: scalar and vector quantization, the ADPCM ITU G.726, sub-band and transform Coders, the ITU G.722 sub-band coder

  • Speech Coding Using Sinusoidal Analysis/Synthesis Models: the sinusoidal transform coder (STC), the multiband excitation coder (MBE), The Inmarsat Standard, IRIDIUM IBME/AMBE

  • Vocoder Methods: the channel, formant, and homomorphic vocoders, linear predictive vocoders, the LPC-10 Algorithm, mixed excitation LPC, RELP

  • Analysis-by-Synthesis Linear Predictive Coders: multi-pulse excited linear prediction, the Skyphone standard, Regular Pulse Excitation (RPE) coders, the Full-Rate, half-rate, and Enhanced full-rate GSM standards, code excited linear prediction, Algebraic CELP, Relaxed CELP, the IS-54 and IS-641 (PCS) standards, the Federal Standard 1016 CELP, the ITU G.728 low-delay CELP coder, the ACELP ITU G.729 and G.729A,, the CDMA QCELP (IS-96), the CDMA EVRC (IS-127), True Speech G723.1, G.4, CDMA 2000 - SMV, Adaptive Multirate Coder for GSM and the new Federal Standard 1017 2.4 kbit/s Mixed Excitation Coder. Info also on proposed vocoders for CDMA 2000, ITU-4, and Adaptive Multi-rate GSM standard.

  • Audio coding topics include introduction to perceptual coding, the MPEGs (MP3), SDDS, DTS, AC-3, and other algorithms embedded in recent streaming audio, cinema, and digital audio products.

Instructors

Dr. Andreas Spanias is Professor in the Department of Electrical Engineering Fulton School of Engineering at Arizona State University (ASU). His research interests are in the areas of adaptive signal processing and speech processing. While at ASU, he has developed and taught courses in DSP, adaptive signal processing, and speech coding. He has also developed and taught continuing education short courses and web courses in digital signal processing and speech coding. Andreas Spanias has been the principal investigator on research contracts from Intel Corporation, Sandia National Labs, Motorola Inc., and Active Noise and Vibration Technologies.

He has also consulted with Inter-Tel Communications, Intel Corporation, Motorola, Texas Instruments, DTC, and the Cyprus Institute of Neurology and Genetics. In his work with Intel Corporation he contributed to the development of architectures with signal processing capabilities and received an award from Intel-Chandler for "his leadership and contributions to the development of the Intel 60172 processor architecture" and a corporate award by the Intel NDTC committee for his support of the Intel research program.

He recently published refereed papers in Perceptual Coding of Digital Speech and Audio, Adaptive Beamforming, Genomic Signal Processing, and DSP Java tools. He and his student team developed the computer simulation software Java-DSP (J-DSP - ISBN 0-9724984-0-0) which is used in the ASU DSP courses. He received the 2003 teaching award from the IEEE Phoenix section for the development of J-DSP. Andreas Spanias is associate director of the ASU Arts, Media, and Engineering (AME) center where he heads a program on sound localization for smart stages using microphone arrays. He is involved extensively in IEEE scientific activities.

He is member of the DSP Committee of the IEEE Circuits and Systems society, and has served as a member in the technical committee on Statistical Signal and Array Processing of the IEEE Signal Processing society (SPS). He has also served as Associate Editor of the IEEE Transactions on Signal Processing and as General Co-chair of the 1999 International Conference on Acoustics Speech and Signal Processing (ICASSP-99) in Phoenix. He served as the IEEE Signal Processing Vice-President for Conferences and the Chair of the Conference Board. He served as a member of the IEEE Signal Processing Executive Committee and as Associate Editor of the IEEE Signal Processing Letters.

EDUCATION

Ph.D. 1988, Dept. of Electrical and Computer Eng, WVU

Registration, Refunds and Cancellations

The registration fees for Center for Professional Development and Distance Education courses held at Arizona State University include instruction, handouts, refreshment breaks and meals as noted in the schedules found in the course schedule (agenda). Hotel accommodations are not included.

Fees may be paid by check, money order or purchase order. Please make all remittances payable, in U.S. funds, to Arizona State University. Payments by VISA, MasterCard and American Express also are accepted. Seating at the course is limited. Fax or e-mail the enrollment form as soon as possible to assure your space, even though payment may come later. Do not rely on your buyer or business office to send the form. Putting the enrollment form in the U.S. mail is only necessary if you will be enclosing a check or purchase order form. A confirmation letter will be faxed or mailed to you shortly after receipt of your enrollment.

Should you register and then need to cancel, please note that there is a cancellation fee. The rate of the fee is determined by how far in advance of the program/module start date the written request for cancellation is received by the Center for Professional Development and Distance Education (please see below). Written requests for cancellation may be received via either mail or fax.

  • Four or more weeks prior - 10% of program fee

  • Three weeks prior - 50% of program fee

  • Within two weeks - 75% of program fee

Transfer to another program or module is subject to a $250 administrative fee if made within six weeks of the program/module start date. Registrants who do not attend and do not cancel are subject to the complete fee. Participant substitutes may be made by submitting in advance a written request. The Center reserves the right to change instructors or cancel or reschedule a program in the event of insufficient enrollment or unforeseen circumstances.

Program Fee

$895 - Program fee includes all course material, software and book; Audio Signal Processing and Coding (Hardcover) by Andreas Spanias, Ted Painter and Venkatraman Atti

For more information, contact:

Layla Reitmeier
Coordinator-Professional and Executive Programs
layla@asu.edu
480-965-8515