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DEVELOPMENT OP AN 8000 BPS VOICE CODEC FOR AvSat 


JOSEPH F. CLARK, AvSat Program, Aeronautical Radio, Inc., United States 


AERONAUTICAL RADIO, INC. 
2551 Riva Road 
Annapolis, Maryland 21401 
United States 


ABSTRACT 

Air -mobile speech communication applications 
share robustness and noise immunity requirements with 
other mobile applications. The quality requirements 
are stringent, especially in the cockpit where air 
safety is involved. Based on these considerations, a 
decision was made to test an intermediate data rate 
such as 8.0 and 9.6 kb/s as proven technologies. 

A number of vocoders and codec technologies were 
investigated at rates ranging from 2.4 kb/s up to and 
including 9.6 kb/s. The "proven" vocoders operating 
at 2.4 and 4.8 kb/s lacked the noise immunity or the 
robustness to operate reliably in a cabin noise envi- 
ronment. One very attractive alternative approach was 
Spectrally Encoded Residual Excited LPC (SE-RELP) 
which is used in a multi-rate voice processor (MRP) 
developed at the Naval Research Lab (NRL). The MRP 
uses SE-RELP at rates of 9.6 and 16 kb/s. The 9.6 
kb/s rate can be lowered to 8.0 kb/s without loss of 
information by modifying the frame. An 8.0 kb/s 
vocoder was developed using SE-RELP as a demonstrator 
and testbed. This demonstrator is implemented in 
real-time using two Compaq II portable computers, each 
equipped with an ARIEL DSP-16 Data Acquisition Pro- 
cessor . 


INTRODUCTION 

The Airlines Electronic Engineering Committee (AEEC) has undertaken 
development of a "characteristic" for satellite communications equipment 
— in effect a guideline for equipment and airframe manufacturers, 
aviation users, and service providers. Designated Project 741, it is 
being developed in parallel with a wide variety of intense activities in 
the mobile satellite communications area. 

The primary schedule constraint for the AEEC has been introduction 
of a new generation of aircraft to be delivered "satcom ready" starting 
in 1989. As a consequence, the AEEC formulated a three-phase schedule 
for Project 741. By late 1986, the Satellite Subcommittee of the AEEC 
formed a working group to recommend a voice coding standard. Although 
there has been a hiatus in the activity of that working group, its 


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efforts both in fulfilling its mission and in assuring that all aspects 
of the voice communication system be considered together are continuing. 

This paper reports on the specific effort to develop a voice coding 
standard using the transmission rate of 8.0 kilobits per second (kb/s) 
and focuses on the similar performance, as shown in testing, of the 8.0 
kb/s and 9.6 kb/s codecs available today. 


ADAPTING TO THE USER COMMUNITY 

The aviation conmunity is loosely divided between "general avia- 
tion” and commercial passenger aviation. The aircraft used vary in size 
from the very large to the very small. The length of flights varies 
from less than one hour to more than eight hours. Although data commu- 
nications increasingly is the mechanism for reporting aircraft status 
and for routine operational exchanges, one common denominator is a 
requirement for cockpit voice — a communication path for the flight 
crew to talk with ground crews. 

While the primary volume application of voice appears to be passen- 
ger telephony, the cockpit voice quality requirements may be the primary 
determinant of the overall requirements because of the ties to safety. 

While only experimental equipment has been installed to date, it is 
clear that aviation will use both high-gain and low-gain antennas. The 
service providers will be using a mix of global beams and spot beams 
from the satellite. As a result, three classes of service can be pro- 
vided: low-rate data; medium— rate data and low— rate voice; and the 

nominal, which is medium-rate data and "near-toll-quality" voice. 

We suggest that a single algorithm for both low— rate and medium- 
rate voice would be the preferred solution. This approach was developed 
by the Naval Research Laboratory (NRL) 1 and is in operation today. The 
method for providing voice communication through a low-rate data service 
is to buffer short messages coded at the nominal voice rate, and trans- 
mit them as extended messages at the actual channel data rate. 


PERFORMANCE REQUIREMENTS 

So far, no one has defined well enough the measurable parameters 
for repeatable measurements of "near toll quality." However, the 
achievable performance of codecs is well known. Figure 1 depicts the 
achievable performance of a wide variety of codecs as determined in 
tests. But in order to specify a required performance, it is necessary 
to determine minimum acceptable performance in terms of parameters that 
can be measured in a consistent way. Unfortunately, voice quality is by 
its very nature subjective. It can only be quantified as a statistic 
derived from inherently variable data. 

The performance requirements endorsed by the AEEC Voice Working 
Group^ are simple: 

Subjective evaluation measured by Mean Opinion Score - midway 
between Fair and Good (3.25 on a scale of 1-5). 

Intelligibility measured by Rhyme Test - 87 or better. 

- Conversational evaluation - 75 or better. 

Delay measured with codecs back to back - no more than 
65 milliseconds. 


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Performance measured with a channel bit error rate (BER) of 1 in 

1000 bits. 

In addition, the group recognized that the operational environment 
would include low-rate data transmissions, rather severe outages and 
interfaces with established terrestrial telephone networks. It was 
decided that data should be routed around the codecs. Not only does the 
requirement for coding data as well as voice put significant demands on 
the coding algorithm, it implies significant additional acceptance test- 
ing. 

The new CCITT coding standard for digital transmission of wideband 
audio signals, G.722, provides a similar bypass mechanism for user data. 
The separation of voice and data allows a better system design 3 . 



APPROACHES 

A CCITT standard ADPCM would simplify the task, but there is more 
complexity in the transmission channel. Limitations of the medium, and 
power and bandwidth constraints drive the choice of rates to the lowest 
possible. On the other hand, the realizable quality at a given rate 
drives the choice upwards. The result then, is a compromise on an 
intermediate rate that approximates the desired performance without 
devouring all available capacity. 


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The rate issue has resolved to a choice between 8.0 kb/s and 9.6 
kb/s. The 8.0 kb/s rate is based on compatibility with expected 
subchannelling of the ISDN basic 64 kb/s communication rate. 

Intuitively, there should not be much difference in performance at these 
rates relative to, say, performance at 16 kb/s or at 4.8 kb/s. This 
intuition is supported by the evaluations received so far. (See Figures 
2 and 3.) 

Furthermore, performance differences seem to be related more to 
implementation than to algorithm. This result is quite reasonable since 
nost of the algorithms are variations on the idea of using residual 
excitation with a linear prediction filter. 

The conclusion then is that the AEEC's voice working group could 
not recommend a specific algorithm, or even a data rate. Only 
implementations of voice systems can be evaluated and recommended. 

From an overall systems point of view, there is a significant 
difference between these rates. Choosing 9.6 kb/s without gaining a 
significant performance advantage is wasteful. In the best of proposed 
implementations, 9.6 kb/s uses $1.20 worth of resources for every $1.00 
used by 8.0 kb/s. For SC PC voice channels as proposed in Project Paper 
741, the 9.6 kb/s implementation requires almost twice as much bandwidth 
as the 8.0 kb/s implementation. (See Table 1.) 

* 

Table 1. Voice Channel Frame Parameters 


Voice 

Rate 

(bits) 

FEC 

Rate 

Channel 

Rate 

(bit/s) 

Channel 

Spacing 

KHz 

Bits in Frame 

Voice/25 

(v) 

Data/96 

(SU) 

(n) 

Dummy 

(d2) 

9600** 

8000** 

0.50 

0.75 

21000 

12600 

17.5 

10.0 

192 

160 

3 

6 

84 

8 


♦Extracted from Project Paper 741, Part 2. 
♦♦Preferred Voice Channels 


Hie AEEC voice working group with the assistance of the Boeing Air- 
plane Corporation developed a standard test tape. This tape was 
distributed to approximately twenty— five organizations which had expres- 
sed an interest either in developing codecs for the aviation industry or 
in evaluating voice codecs. The tapes were fed in and out of candidate 
codecs, and the resulting tapes analyzed in listening tests conducted by 
Dynastat. 


CNE 8.0 KB/S SOLUTION 

There were two major concerns in dealing with established manufac- 
turers of voice codecs. First, a satisfactory licensing agreement for 
the algorithm and implementation must be available to all avionics manu- 
facturers. Second, manufacturers might continue to focus on products 
designed only for their terrestrial markets, i.e., codecs designed 


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Fig. 2. Intelligibility: Cockpit 

Noise Environment 



Fig. 3. Probability of User Acceptance Based 
Based on Cockpit Noise Environment 

for 2.4 kb/s, 4.8 kb/s, and 9.6 kb/s even though 8.0 kb/s is a 
demonstrably better choice for the emerging aviation industry character 
istic . 

In view of these concerns, ARINC contracted with Techno-Sciences, 
Incorporated (TSI) of Greenbelt, Maryland, to develop an 8.0 kb/s voice 
codec from information published in the public domain. The premise was 
if a credible voice codec could be developed by one or two key people 
working with limited time and budget, then avionics manufacturers could 
derive non-proprietary sources for voice codecs, and on a realistic 
schedule with confidence. 


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TSI, with ARINC assistance, developed a pair of 8 kb/s voice codecs 
operating in real time. The codecs are connected through the serial 
(RS232) ports of two Compaq Portable II computers. Speech is sampled 
8000 times a second and blocked into groups of 180 samples. Ten 
reflection and filter coefficients are calculated using autocorrelation. 
The quantized reflection coefficients are used to form the filter which 
then is used to generate the prediction residual. 

A 96-component spectral representation of the prediction residual 
is computed using the Winograd algorithm. These components are 
quantized and coded into a 180-bit block with the reflection 
coefficients. A new block is generated and transmitted 44.4444 times 
per second. The key factors in the codec performance are the reflection 
coefficient calculation and the number of transmitted spectral 
components. The current codec transmits 126 spectral components, the 
same number transmitted in the NRL 9.6 kb/s rate codec. 

After 6 months of development, a codec emerged that is credible but 
admittedly is incomplete and needs improvement. Several areas of the 
basic LPC algorithms have been tagged for further work in a second phase 
of the project. In the second phase of the codec development, vector 
coding of the spectral components will be introduced. There is good 
confidence in the expected results. 


CONCLUSION 

The AEEC Voice working Group, with cooperation of the Boeing 
Airplane Company, the Federal Aviation Administration, the Rome Air 
Development Center and others, has developed a set of standards for 
voice codecs in the air mobile environment, collected a representative 
set of samples for a variety of oodecs using a uniform test tape, and 
received an unbiased formal evaluation of the sample tapes. The results 
of the evaluation indicate that at least 7.2 kb/s speech coding will be 
required to reliably meet requirements in the aviation satellite 
environment. These results (see Figures 2 and 3) also indicate that 
performance at 8.0 kb/s is in the same range as performance at 9.6 kb/s 
— making 8.0 kb/s the better overall choice for a power and bandwidth- 
limited satellite environment to be globally interconnected with ISO- 
conforming networks. 


REFERENCES 

1. Kang, G. S. and Fransen, L. J., Second Report of the Multirate Pro- 

cessor for Digital Voice Communications, NRL Report 8614, 
September 30, 1982. 

2. AEEC Letter 87-014/SAT-33, p. 11, Aeronautical Radio, Inc., 2551 

Riva Road, Annapolis, MD 21401, February 13, 1987. 

3. Mermelstein, Paul, G.722, a New CCITT Coding Standard for Digital 

Transmission of Wideband Audio Signals, IEEE Communications 
Magazine, Vol. 26, No.l, January 1988. 

4. AEEC 700 Quick Check #50, November 4, 1987. 

5. Techno-Sciences, Inc., Report No. 870909, 8 kbps Speech Demo Pro- 

ject, prepared for Aeronautical Radio, Inc., September 9, 

1987. 


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