Session T1C
Work in Progress - Teaching Speech Signal
Processing and Coding using LabVIEWTM
Andreas Spanias1, Karthikeyan Natesan1, Jayaraman Jayaraman1, Photini Spanias2
1
Dept. Electrical Engineering, SenSIP Center, 2College of Education, Arizona State University
spanias@asu.edu
Abstract - This paper presents a visual programming and
visualization tool for use in covering signal processing
aspects of speech coding in a DSP class. The tool is based
on the National Instruments LabVIEW™ environment.
The framework of this tool was built using existing C code
as a library along with LabVIEW’s native functionalities.
Standardized linear predictive coding algorithms have
been implemented; these are used to demonstrate in our
DSP classes how digital filters and signal modeling is
utilized in cellular and military communications.
Experiments covered include the introduction of speech
synthesis models, parameterization of speech in terms of
filter and excitation parameters, and robustness of speech
parameters to additive and channel noise. The tool
provides capabilities for both quantitative and subjective
assessment of the synthesized speech signal. An
assessment process for the tool is in place and preliminary
results will be presented at the conference.
Index Terms – educational software, LabVIEW, real-time
DSP, speech coding.
INTRODUCTION
Speech coding is concerned with compact digital
representations of voice signals for the purpose of efficient
transmission or storage [1-6]. There have been several
standardized speech coders that use the linear prediction
representation of speech [7,8,9]. In particular source-system
analysis-synthesis systems that use all-pole filters to represent
the vocal tract are to this day used in most standardized
algorithms. In fact, the Levinson-Durbin linear prediction
algorithm is embedded in every cell phone. We choose the
linear prediction source-system paradigm for implementation
because it can be connected to several concepts covered in
DSP classes including digital filter theory, estimation of
periodicity, autocorrelation computation, filter stability, and
estimation of non-stationary signal parameters. In our study
we worked with the FS-1015 LPC (open loop Linear
Predictive Coding), FS-1016 CELP (closed loop or Code
Excited Linear Prediction) standards, and the full rate ETSI
GSM standard [1,7]. Although the Federal Standards 1015
and 1016 have been around for some time, they were chosen
because they are open source and capture the basic aspects of
speech compression.
The choice of LabView [10] as a visual programming
environment was motivated by the real time signal acquisition
capabilities of this environment and the presence of several
1-4244-1084-3/07/$25.00 ©2007 IEEE
37th ASEE/IEEE Frontiers in Education Conference
T1C-22
native graphical functions that enable users to visualize
different aspects and parameters of a speech coding algorithm.
The block diagram approach also enables the users to
understand the basic blocks of a speech analysis-synthesis
system. This speech coding tool is scalable in the sense that
additional options and capabilities could be added.
Exercises that expose students to the non-stationarity of the
speech signal, the all-pole spectral modeling performed by
LPC analysis-synthesis, the estimation of the pitch have been
developed. We note that the tool is of value not only to
undergraduate and graduate students but also to DSP
practitioners. Also with some simplifications, the tool can be
used for demonstrations in high school science classes.
Assessment instruments have been developed and preliminary
results from pre- and post-quizzes and summaries of student
interviews will be presented at the conference.
THE LABVIEW SPEECH CODING TOOL
The graphical user interface of the LabVIEW speech coding
tool is shown in Figure 1. Presently, the FS-1015 LPC 10e and
the FS-1016 CELP 4.8 kbps standards have been implemented
by interfacing the shared C libraries [10] with the framework
developed in LabVIEW. The speech coder is represented
using individual blocks for analysis (encoder) and synthesis
(decoder). The tool also has additional features for real-time
DSP, signal playback and graphical plots of different sets of
parameters.
The software can access either an audio (‘.wav’) file or
real-time speech input. The user also has options to change
certain speech parameters to analyze the performance and
behavior of the algorithm under different conditions. The
preprocessed input speech is displayed and processed on a
frame-by-frame basis. Frame-by-frame display is also used to
view the excitation, the filter parameters, the spectrum of the
preprocessed and decoded speech, the quantized LPC spectral
envelopes, the pitch estimates, pole-zero plots of the synthesis
filter, formants, speech synthesis waveforms SNRs, etc. The
software has options to save and read the input data and the
coded data. The user can also analyze the subjective quality
of these algorithms by listening to the synthesized speech with
the aid of the playback feature.
UTILITY IN EDUCATION
The capabilities of this tool allow the students to experiment
with a wide range of speech data thereby understanding the
several aspects of the algorithms that are not immediately
October 10 – 13, 2007, Milwaukee, WI

FIGURE 1 USER INTERFACE OF THE LABVIEW SPEECH CODING TOOL a) FS1015, b) FS1016
evident from standard documentation. One significant
extension planned is the integration of a digital signal
processor interface with the LabVIEW tool. Such an addition
will enable students to experiment real-time with fixed-point
implementation issues without having to code the entire
algorithm on the processor chip. In addition to the currently
implemented FS-1015 LPC and FS-1016 standards (Figure 1),
the GSM RPE-LTP algorithm is also being implemented.
ASSESSMENT
A collection of experiments that utilize this LabVIEW tool
was designed. We had a select group of students from our
DSP class evaluate this tool and fill out assessment forms.
Preliminary results will be presented at the conference.
REFERENCES
[1] A. Spanias, “Speech Coding: A Tutorial Review”,
Proceedings of the IEEE, Vol.82, Issue 10, Oct 1994.
[2] V. Atti, “A Simulation Tool For Introducing Algebraic
CELP (ACELP) Coding Concepts In A DSP Course,” IEEE
2002 DSP Workshop, Callaway, Georgia, October 2002.
1-4244-1084-3/07/$25.00 ©2007 IEEE
37th ASEE/IEEE Frontiers in Education Conference
T1C-23
Session T1C
[3] Spanias A., Painter E.M., “A Software Tool for
Introducing Speech Coding Fundamentals in a DSP Course”,
Education, IEEE Trans on, Vol.39,2, pp.143-152, May 1996.
[4] A. Spanias, T Painter, V. Atti, Audio Signal Processing
and Coding, ISBN: 0-471-79147-4, Wiley, February 2007.
[5] A. Spanias, Digital Signal Processing; An Interactive
Approach, ISBN: 978-1-4243-2524-5, January 2007.
[6] V. Atti, “Interactive On-line Undergraduate Laboratories
Using J-DSP,” IEEE Trans. on Education Special Issue on
Web-based Instruction, vol. 48, no. 4, pp. 735-749, Nov. 2005.
[7] Andreas Spanias, Chapter 3: Speech Coding Standards,
pp. 25-44, Invited. Academic Press, Ed: G. Gibson, ISBN
2000 0-12- 282160-2.
[8] “FS-1015 LPC C Code Implementation”, Available at
World Wide Web: http://www.arl.wustl.edu/~jaf/lpc/
[9] “FS-1016 CELP C Code Implementation”, Available at
World Wide Web: ftp://svr-ftp.eng.cam.ac.uk/ comp.speech/
coding/celp_3.2a.tar.Z.
[10] “LabVIEW Fundamentals”, Available at World Wide
Web: http://www.ni.com/pdf/manuals/374029a.pdf
ACKNOWELEDGEMENT
Portions of this work have been sponsored by the ASU
SenSIP center and the NSF CCLI award 0443137.
October 10 – 13, 2007, Milwaukee, WI