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FULL-DUPLEX SPEECH FOR HF RADIO SYSTEMS
N Serinken, B Gagnon, and 0 Erogul
Communications Research Centre, Canada
INTRODUCTION
Conventional telephone systems operate in full-duplex
mode where both parties can speak at the same time. HF
radio systems operate in a mode known as half-duplex
mode where only one person can talk at a given time. The
direction of the communication link is controlled by a
“push to talk” (PTT) button which is activated by the party
wishing to speak. In order to avoid confusion, the speaking
party must indicate verbally that the channel is being,
relinquished by saying “over”. Despite its inconvenience,
half-duplex has retained its popularity because it uses only
one radio channel, which is an important consideration1
when radio spectrum is at a premium. One disadvantage of
a manually or automatically controlled half-duplex
communications system is that it is difficult to interface to
a full-duplex conventional telephone system. In addition,
a ”normal” conversation in which one party may intermpi
the other is impeded by the nature of the radio link.
A full-duplex link can be achieved over a single-channel
radio system if both ends of the link operate in what is
known as time division duplex (TDD) mode. In this mode
the channel is allocated half of the time to transmit
information in each of the two directions. That is, the radio
channel is divided into time slots of Ti2 seconds, with each
end transmitting in alternate time slots. Stated simply,
when one end is transmitting, the opposite end is receiving
the information and vice versa. In this way a single radio
channel can support information flow in both directions
resulting in a virtual full-duplex link.
Full-duplex voice systems are used on radio links that can
support high speed digital data. For the TDD system to
work, the channel must be capable of carrying more than
twice the data rate needed for a single voice codec. For HE;
radio systems the maximum rate that can be reliably carried
within the assigned band is not sufficient for TDD voice
transmission given the present state of speech coding and
data transmission technology.
To overcome these problems the system described in thii3
paper uses a technique called “time-scale modification“
(TSM) which can compress a T second segment of analog
voice by a factor of two. An advantage of this approach is
that the compressed voice signal occupies the same
bandwidth and dynamic range and consequently no radio
modifications are required. Additionally since the
frequency occupation remains unchanged, all existing
radio regulations remain satisfied and the type approval
process for the radio need not be repeated. As the time-
‘HF Radio Systems and Techniques’, 7-10 July 1997, Conference Publication No. 41 1, 0 IEE, 1997
scale modification process is reversible the time
compressed speech segments can be restored to their
original form with minimal distortion. By using analog
time compression and expansion full-duplex audio can be
achieved when the available bandwidth of the system does
not allow for digital TDD operation. The real-time
implementation and test results of an analog time division
duplex system (ATDD) are presented in this paper.
TIME-SCALE MODIFICATION
Three different TSM algorithms have been evaluated for
the purpose of determining the best technique for the real-
time implementation of the ATDD system, Seneff (1),
McAulay and Quatieri ( 2 ) , and Verhelst and Roelands (3).
These algorithms have been applied to speech signals from
both genders as well as musical passages and their
performance has been evaluated through a series of
subjective listening tests, such as the mean opinion score
test (MOS), the degradation mean opinion score test
(DMOS) and the diagnostic rhyme test (DRT),
Papamichalis (4). The Waveform Similarity Overlap-and-
Add Algorithm (WSOLA) (3) was chosen as the reference
algorithm for the performance comparison of the other
algorithms in the DMOS test. Table 1 gives the subjective
evaluation results of these tests for sequential TSM
compression followed by expansion of speech signals by
factors of two.
TABLE 1- Subiective evaluation test results
( l=poor to 5=excellent)
A: WSOLA, Algorithm in reference (3)
B: Algorithm in reference (1)
C: Algorithm in reference (2).
Based on these tests, the WSOLA algorithm was selected
for the real-time implementation of the system because it
was computationally inexpensive, and the resulting speech
quality was found to be very high with respect to the
others.
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For a uniform change in the time scale, the time to
corresponding to the original articulation rate is mapped to
the transformed time t’, through the mapping t’, = p to.The
case p > 1 corresponds to slowing down the articulation
rate by means of time-scale expansion, while the case p <
1 corresponds to speeding up the articulation rate by means
of time-scale compression. Speech events which take place
at a time t‘, according to new time scale will have occurred
at p ’ t 6 in the original time scale.The WSOLA algorithm
is a time domain process. It seeks to find a segment of the
input signal that will be overlapped with and added to the
previous segment which lies within a prescribed tolerance
interval around the synthesis instant. The position of the
best segment is determined by finding the value A=Am
lying within a tolerance region [-&m...A,ax] around the
analysis instant and which maximizes the cross-correlation
coefficients between the previous segment and the segment
under consideration. The basic synthesis equation used by
the WSOLA procedure is:
where v(n) is the square of a windowing function, w(n), Ak
are the shift factors within the range [-A,,...&,,], x(n) and
y(n) are the input and output signal samples respectively,
L, represents the consecutive window positions, i.e., the
synthesis instants, and ?(LA represents analysis instants.
w(n) is a Hann window’ with 50% overlap. The operation
of the WSOLA technique is illustrated in Figure 1 and
explained below.
In the implementation of this algorithm, regularly spaced
synthesis instants L, = kL are chosen. Proceeding in a left-
to-right fashion (in Figure 1 ) and assuming the segment
(A) was the last segment excised from the input and added
to the output at time instant L,-/=(k-I)L, i.e., segment (a) =
segment (A). WSOLA seeks to find a segment (b) that will
overlap-add with (a) in a synchronized way and can be
excised from the input around time instant z’(LJ. As (A’)
will overlap-add with (A) = (a) in a natural way to form a
portion of the original speech, WSOLA can select (b) such
that it resembles (A‘) as closely as possible and is located
within the prescribed tolerance interval around z’(LJ in
the input wave. The position of this best segment (B) is
found by maximizing a normalized cross-correlation
similarity measure between the sample sequence
underlying (A’) and the input speech. After overlap-adding
@) with (a), WSOLA proceeds to the next output segment,
where (B’) now plays the same role as (A’) in the previous
step. In the overlap and adding process each overlapping
signal segment is weighted with one half of the windowing
function for smooth transition from one segment to the
‘The term Hann window is conventionally
known as Hanning window, after Julius Von Hann.
next segment. The old signal segment is weighted with the
falling portion of the windowing function while: the new
segment is weighted with the rising portion of the
windowing function.
Time-scale expansion is achieved by repeating the speech
segments excised from the previous ones within the
tolerance interval in the same manner explained above.
The time alignment between the successive windows with
respect to the signal similarity removes the phase
discontinuities. Therefore, time-scale modified waveform
by the WSOLA algorithm can maintain maximal similarity
to the original waveform across its segment joints.
SYSTEM IMPLEMENTATION
An ATDD system was implemented using a SO MHz
TMS320C3 1 floating-point digital signal processor with
512 kilobytes each of RAM and ROM. The computer
system and external inputloutput audio were interfaced
using a 14-bit linear AID and DIA with a 8 kHz audio
sampling frequency. The processor board has digital ports
for connection to the radio Wrx control lines and external
timing reference signals. The digitized audio was stored in
a 200 ms long buffer and compressed into bursts of 90 ms
in real-time. The remaining 10 ms reserved for in band
timing reference signals. The radio channel was
multiplexed into 100 ms transmission and reception slots.
DSP software was written in “C” language for the
implementation of the WSOLA algorithm. Figure 2 shows
the ATDD system and HF radio equipment connections
for one end of a radio link. The other end of the ATDD
link has the same configuration of radio and ATDD
processor. Audio signals from the ATDD system were
interfaced to the line-in and out ports of the radio
equipment and the control port of the ATDD was
connected to the PTT line of the radio system. Figure 3
shows the compressed signals to and from the audio ports
of the HF transmitter and receiver. The top trace shows the
compressed 100 ms bursts of audio fed to the transmitter
while the bottom trace shows the audio output from the
receiver as observed at points A and B of Figure 2
respectively.
Experimental Setup
The experiments employed a transmit site near Ottawa and
a receive site 210 km southwest of this location near Lake
Ontario, for a period of four days during October 1996.
Test signals were transmitted at a frequency of 4 MHz
using a horizontal multi-band fan dipole antenna at a
power level of 100 Watts and were received using a
horizontal dipole antenna. Since the ATDD systlem relies
on the accurate synchronization of transmission and
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reception time slots, both transrnitter and receiver were
synchronized using global positioning system (GPS)
receivers. To compensate for the end-to-end time delay of
the system the receiver timing reference was delayed to
coincide with the start of the timing reference signal
received from the transmitter. The experimental system
used the transmitted timing reference signal as a marker for
alignment of the external timing reference delay for the
receiver. At the receiver output the compressed speech
signal was separated from the timing reference and
expended into normal speech by the ATDD processor. To
simulate part of a full-duplex radio link, ATDD
transmission was tested for a one-way link with a 50%
channel occupation.
It was observed that the gain of the communications
receiver was unable to track signal transitions from
transmission time slot to reception time slot due to the
receiver automatic gain control (AGC) attack time-
constant. The onset of the burst of compressed speech
occasionally caused the ATDD time scale expansion to
distort the speech signal due to over-amplification of the
received audio. Overall, ATDD system was observed to
perform satisfactorily during various forms of HF radio
channel impairments such as noise, interference and
selective fading encountered during the tests.
ATDD system performance was evaluated by comparison
of ATDD processed HF radio link to an unprocessed HF
radio channel. The transmitter audio input was controlled
by a selector switch with two inputs which was repeatedly
fed with 10 minutes of ATDD processed signal followed
by 10 minutes of conventional speech signal. The receiver
audio and ATDD processed signals were recorded with an
audio tape recorder for informal subjective evaluation of
the ATDD system performance.
At the receiver, the processed and unprocessed systems
were evaluated using the DRT intelligibility test. This
intelligibility test is based on the ability of listeners to
distinguish phonemes with common attributes. The listener
is presented with one word from a pair and asked to
determine which word was spoken. The alternate choice is
a word differing from the presented word in one phoneme,
usually a consonant. This test contains 464 words in 232
rhyming pairs and a list of words spoken by both male and
female speakers. The intelligibility score is calculated as a
percentage of correct responses.
Both systems were evaluated under similar conditions and.
it was found that the scores for the female speaker were:
consistently lower than the male speaker. The average:
score for the ATDD and conventional radio links were:
80% and 90% respectively, and average intelligibility
score for the WSOLA algorithm was found to be 92%)
without the ATDD processing.
CONCLUSIONS
One disadvantage of ATDD systems is the delay
introduced to the speech path, which is in the order of
twice the input buffer size. In the test version of the
ATDD system the buffer size was chosen to be 200 ms,
resulting in a delay of 400 ms which is less then the delays
observed from geostationary communication satellites.
ATDD system feasibility has been demonstrated with a
real-time system and the quality of speech was found to be
satisfactory. The effect of channel impairments was
similar to those of a conventional half-duplex radio link.
The next phase of ATDD research will concentrate on the
management of transmiit/receive antenna switching,
receiver AGC matched to the ATDD processing and the
synchronization of multiplexing time slots through a
reference derived from transmitted reference signals.
REFERENCES
1. Seneff S., 1982, ”System to Independently Modify
Excitation andor Spectrum of Speech Waveform Without
Explicit Pitch Extraction”, IEEE Trans. on Acoustics,
Saeech, and Signal P r o c e s s h Vol. ASSP-30, NO. 4, pp.
566-578
2. McAulay R. J. and Quatieri, T. F., 1986, “Speech
Analysis Synthesis ]Based on a Sinusoidal
Representation”, IEEE Trans. Acoustics. Saeech. and
Siena1 Processing, Vol. ASSP-34, No. 4, pp. 744-754
3. Verhelst W. and Roelands M., 1993, “An Overlap-add
Technique Based on Waveform Similarity (WSOLA) For
High Quality Time Scale Modification of Speech”, Int.
Conf. Acoustics. Saeech. and Signal Processing;. ICASSP-
-
93, pp. 554-557
4. Papamichalis P.E., 1987, “Practical Approaches to
Speech Coding”, Prentice-Hall Inc., Englewood Cliffs,
N.J., pp. 177-198.
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Figure 1 WSOLA input and output waveforms

Figure 2 ATDD and radio interfaces

I , , A- Tx input

I R- Rx O u t D u t

Figure 3 Radio input/output audio signals