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Data Mapping onto Speech-like Signal to Transmission over the GSM Voice Channel

Conference Paper · April 2008

DOI: 10.1109/SSST.2008.4480189 · Source: IEEE Xplore

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40th Southeastern Symposium on System Theory MA2.5
University of New Orleans
New Orleans, LA, USA, March 16-18, 2008

Data Mapping onto Speech-like Signal to

Transmission over the GSM Voice Channel

Mahsa Rashidi Abolghasem Sayadiyan Pejman Mowlaee

Msc student at Electrical Engineering Associate professor at Electrical PhD student at Electrical Engineering
Department Engineering Department Department
Amirkabir University of Amirkabir University of Amirkabir University of
m rashidi(gau.tac.ir eea35gaut.ac.ir P MowlaeeLajeee.org

Abstract- One of the most important objectives in mobile
communication systems is secure voice and data
communication (including text, picture, video and voice) esp.
in high bit rates. In this paper, a new procedure is proposed
in which the intended data or voice is encrypted and
modulated onto speech-like waveforms. The modulated
waveforms are transmitted over the global system for mobile
communications (GSM) voice channel and then demodulated
and decrypted at the receiver. We propose an appropriate
modelfor the GSMAFull Rate (FR) speech codec by mapping
data onto the fundamental parameters related to formants in
a speech-like waveform including phases, frequencies and
pitch frequencies. The proposed model has been evaluated
for a GSM-to-GSM connection. Conducting different
simulations we observed that the proposed approach results
in a bit error rate (BER) of 0.020o when Signal-to-Noise
Ratio (SNR) is 15 dB in a 1.5kbps channeL As a result,
proposed method can be considered as afavorable choice for
robustness to additive noise.
Keywords - formants, GSM, LSF, speech-like waveform,
formants.
1. INTRODUCTION
Hardware and protocol deficiencies are two drawbacks
for the 2 generaLion communications systems
makes them only capable for data transmission at low
bit rates (e.g. 1120 bits per page) for Short Message
Service (SMS) in G.7 signaling channel. However, as
the data channel is available for a limited number of
subscribers, data transmissions are still possible to a
maximum rate of 9.6 kbps.
In contrast to data channel, using a voice channel
can result in negligible time delays as reported in [1].
In addition, one of the most important problems in data
transmission over the GSM voice channel is to make
sure whether the transmitted data is highly secure. To
cope this problem, the resulting bit stream from a low
bit rate speech coder implemented for voice channel
adaptation, usually enters into data encrypting block
[2]. Data will be modulated on speech-like waveforms
prior entering the global system for mobile
communications (GSM) network. The resulting
waveform then enters the first GSM handset,
communications channel and then reaches the second
GSM handset. The received waveform from second
handset is demodulated, decrypted and finally decoded
[3]
The ov ll syste ock diagrm lss nsucin
Fg1.uecto lw raeqof spee chan ing
communications, we require modems having data
transfer capability with low bit rates. As a result, based
on the proposed method in this paper, an appropriate
modem is presented. However, in some recent works
by Katugampala reported in [3], codebooks including
the values of speech-like waveform parameters are
defined including pitch frequency, Line Spectral
Frequencies (LSF) coefficients and frame energy in
modulator side. Next these parameters are used for
waveforms synthesis. Finally, the encrypted data are
mapped onto these waveforms. These parameters will
be derived from the received speech-like waveforms in
demodulator side and compared to codebook and
finally the best one is chosen [3]. Meanwhile, this
approach has been adopted for GSM Enhanced Full
Rate (EFR) speech codes 12.2 kbps whereas the
proposed approach in this paper is considered for 13
kbps GSM FR speech codes as reported in ETSI GSM
06.10 [4].
This paper is organized in the following sections. In
section 2, the complete procedure of speech-like signal
production, data mapping are presented. Section 3 is
dedicated to synchronization. In section 4 simulation
results are reported. Section 5 concludes.
2. SPEECH-LIKE SIGNAL PRODUCTION
PROCEDURE
We require mapping data bit stream on speech-like
waveforms of2G ms length (equal to what is usually
available in GSM speech coder). Therefore in this
paper, we produce speech-like waveforms based on
Auto-regressive (AR) model. Waveforms should be
produced with four formants so that they can be

978-1-4244-1807-7/08/$25.OO ©2008 IEEE. 54

 MA2.5

adapted to GSM coder. As formants are sensitive to
changes and for simply extraction of signal
characteristics in the demodulation, we would prefer to
parallelized the resulting transfer function according to
work in [5]. Finally, by applying excitation signal to
resulting transfer function, appropriate speech-like
signal will be produced. As a result, related the transfer
function of ith formant is a second-order difference
equation as follows:
where HTotal (z) is the same paralleling transfer
function, Nana is the analysis window length, and Nfr is
the frame shift length. Note that, vectors a, f and T in
(4) are obtained from M prominent peaks found trough
peak picking procedure as reported in [6]. Fig.2 shows
a prototype of speech-like waveforms with 20ms
length produced by harmonic modeling approach
discussed above.
Speech-like

where A.=1+B.+C. (1)

4lCpare Bit Strem Data
Speech-like
waveform Speech Channel
B1 = 2 exp(-7w Af fs). Cos(27f Ifs)
I 1j Ot Demodulator | CodC Decod ing

Ci =-exp(-27 Af1/ fs) Figure 1: overview of the complete system.

1.2
wherefs is sampling frequency, Af andf are formants'
frequencies and bandwidths, respectively. In order to 1
have a logic compare between the received envelope 0.8
spectrum and the one in the transmitter paralleling of
Hi (z) is done under conditions as follows: E 06

* Firstly, we normnalize the transfer function related 0.4

to each formant to its central frequency: z 0

Hi (ee) 2w ki =1 i 1,2,3,4 (2) 0

0 20 40 60 80 100 120 140 160
Time(sample,8KHz)
* Secondly, another normalization should be Figure 2: Synthesized speech-like signal
employed in a parallel resultant format transfer
function which can be written as below: 2.2. Data mapping on speech-like signal

a, H (ejn) +/3 IH2(eicon +P H(e One of the key points is to correctly select the
H4(e'X)+/I=1HI(e'I)
+2n H4(ej-" ~+t...~(e') (3) formants' frequencies within telephone voice band
n= 1,...,4 (300-3400 Hz). During experiments and investigations
we concluded that among the mentioned parameters in
where, an Pn Pn and An are the normalized equation formants, their related frequencies and phases can only
coefficients. Finally, the speech-like waveform will be be detected as speech-coded passing signals in a voice
resulted from the overall spectral envelope by channel. As a result, we explain in detail how to select
employing the harmonic model synthesis method based parameters and to allocate data bits to frequencies and
on [6]. As a result, the complete process for waveforms phases parameters. We should select the 1 st and 2nd
production can be demonstrated as follows: frequency formants among the frequencies ranges in
300 to 1000 Hz. These frequencies are encoded by 3
bits. Note that, the frequency range of the third formant
a HTtl(f) (4) ranges in 1400 to 2500 Hz and coded by 3 bits and
v=2a (Naa
N 2 -Nf) fourth format range between 2900 to 3400 Hz coded in
2 bits. Note that since harmonic model is used in the
Al
(27f proposed method, the formant frequencies discussed
s
=ncosATnp
a n=I
-above should be selected as a multiple of the pitch
frequency which results in a negligible error in

55

extracting information. The selection criteria are as
follows:
1. Received signal amplitude should be more than
70 percent of transmitted amplitude.
2. Frequency displacement of received formants
should not be more than a default frequency steps
for each formant. Otherwise, it causes incorrect
extraction of the mapped information in the
resulting frequencies. As a result, selecting the
frequency steps as a multiple of the given pitch
frequency fulfills such a condition. However, a
larger frequency step is selected for the 4th
formant due to its high sensitivity to
displacement.
3 nother important point is the lack of proximity
Another impotacntformants.
in two adjacent poimantsisAsthresulack
a result, pheroxicauses
there are
unusable regions iin bounary
band regons
unusale
bad boundary btwee
between that the overlappingg samples in each frame should not
formants. This is due to the fact that minimum
distance for two adjacent formants is twice the
bandwidth considered while their bandwidths are
the same. Due to the lack of fidelity in GSM
coder/decoder to formants bandwidths, we only
consider constant and similar bandwidth
ofAf =160 Hz in whole synthesis process.
As the phase fidelity only holds for frequencies under
1 kHz, some information should be preserved in phases
related to first and second formants. As a result, the
difference between the extracted phase from the
received signal envelope and the mapped phase in that
particular frequency phase should be coded within 3
bits. Another important parameter is pitch-frequency
selection problem which is proportionate to the choice
of the synthesize window employed in harmonic
analysis procedure discussed earlier in Section 2.1. As
a result, we observed that using pitch frequencies
fp=123 Hz and fp=125 Hz result in acceptable
performance. Therefore we coded the mapped data on
pitch frequencies while employing 1 bit. Finally, the
whole speech-like waveform procedure can be
modulated by 12 data bits in a 20ms frame length. In
addition, we demonstrate in the simulation results that
using the proposed technique we achieved at a bitrate
of 600bps.
2.3. Intra-frame Interpolation
In order to achieve phase continuity which is an
important characteristic in speech signals and some
56
MA2.5
jumps occurring in frame boundaries, it is necessary to
overlap the produced speech-like waveforms with
above approach. Also it should be considered that data
bit streams on speech-like signal remain undamaged.
To this end, it is so important to select pitch period i.e.
1/fp that has direct relation to data mapped on each
frame. GSM codec does a linear interpolation between
Log Area Ratio (LAR) coefficients of two adjacent
frames (each frame consisting of 160 samples). To
avoid spurious transients as well as interpolating LAR
coefficients of the last frame's the primary 40 samples
with LAR coefficients of the current frame's the
primary 40 samples [4]. This motivates us to the idea
that adjacent frames should have the minimum overlap.
This is due to the fact when a PCM waveform signal
starts GSM tandem connection; high overlapping of
intra-frames does not cause tremendous changes in
reflection coefficients of each frame. This, as a result,
incorrect detection of transmitted data. Note
ththeorainsmesnechfmehulnt
be chosen in order to prevent inter-modulation effects.
As a result, (5) presents the linear interpolation for
proposed modulator:
Y (1+a+160-n)
2a +1
Y2 = (n + a-160) n (160- a), ...160
2a+I (5)
(m + a)
-Y3 2a +1 m n=1, ,(1+a)
l (+ a-m)
Y4 = a-n
2a +1
Where a equals the overlapping samples in each frame.
Note that, Y1, Y2 are multiplied by samples of
(160-a) to 160 in the last frame, s(i 1), and y3 y4 are
multiplied by samples of Ito (1+ a) in the current
frame, sI, presented in (6), respectvely.
L1 = Y1 x S(j1) (¼)
L2= Y2 x S(i_1) (k1)
L3 = Y3 x S1 (½)
L4 = x Si (k2) (6)
Interpolat ion1 = L + L2
Interpolat ion 2 =L3 + L4
Interpolat ion = Interpolat ion 1 + Interpolat ion 2
where k1 k2 are the numbers of samples interpolated,
Interpolation1, Interpolation2 are the overlapped
 MA2.5

samples of each frame and Interpolation is the whole 0.8

interpolated samples of two adjacent frames and note 0.7
that in demodulator this region shouldn't be chosen. 0.6
Finally, appropriate PCM waveform signal has been 0.5
prepared to enter into the speech coded voice channel. 0.4
E
< 0.3
3. Synchronization N 0.2
° 0.1
One of the important things in simulation is 0A
synchronization of system elements so in order to
simulate the synchronization of the speech codec Received signal
frames in two base-stations, we considered this effect 0.2 Synthesized signal
0 20 40 60 80 100 120 140
inserted into system by a random number of samples Time(samples,8KHz)
before the signal passed to the second codec. Then to Figure 3: Synthesized and Received signals
simulate the synchronization of modulator and
demodulator, at the start of any communication a
predefined synchronization sequence is sent from the 0.9
modulator to the demodulator. This sequence of 0.8
samples is known to both. Since in the simulation it is 0.7
known that there will be a synchronization sequence in _ 0.6
the input signal, the synchronization module cross-
correlate a fixed predefined number of input samples at _
the beginning of the transmission with the predefined 0.4
synchronization sequence. The sample sequence that 03
best matched is used for synchronization in the 0.2
demodulator. 0.1 Received spectral envelope
Synchronized spectral envelope
0 500 1000 1500 2000 2500 3000 3500 4000
4. Simulation Results frequency(Hz)
Figure 4: Synchronized and Received envelopes
We tested our system on the GSM-to-GSM connection. We achieved a throughput of 2kbps with 0.3°0 Bit
In our simulation we generate speech-like signal by the error rate (BER). Using a Punctured l2-rate
proposed method with the length of 2.5s consisting of convolutional code with constraint length of 7, on the
about 120 different waveforms (with 20ms length) and 2kbps, achieved a 1.15 kbps channel with 0.02% BER
The best interpolation occur in a=7. To synchronize for SNR=15dB. As a result, the proposed method can
modulator and demodulator, before starting of any be considered as a favorable choice due to its
transmission a predefined synchronization sequence is robustness to additive noise as depicted in Fig. 5.
sent, that the synchronization process occurs in the
22nd sample of this sequence. Next, generated signal is 10-3
transmitted from modulator to the coder, channel, Empirical
decoder and demodulator. Fig.3 evaluates 3rd frame of
synthesized and received signals.
In order to extract the important speech-like
parameters, we need to have the envelope spectrum of
the received signal. Hence, fig.4 illustrates the 4

envelope spectra of signals. Note fig. 3, 4 have been

generated by pitch-frequency corresponding to 125 Hz 2
and signals are not selected from interpolated samples
as explained in section 2.3. Four peaks with maximum
amplitudes show displacement of central frequencies
of 3rd frame's formants in synthesized and received 1o-5 10 15 20 25 30
signals as depicted in Fig.4. SNR per bit,Eb/No (dB)
Fig.5. BER over Fading channel with additive Noise for
BT=0.3

57
 MA2.5

5. Conclusion
A robust method is proposed for secure data
transmission over a GSM voice channel. The method
was based on transmitting of the mapped data on the
fundamental parameters related to formants in a
speech-like waveform including phases, frequencies
and pitch resulting in transferring 12 bits data on a
speech-like waveform using frame size of 20ms.

Reference
[1] M. Street, "Interoperability and international operation:
An introduction to end to end mobile security", IEE Secure
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"A 2.4/1.2 kb/s speech coder with noise pre-processor",
proceedings EUSIPCO 2000, Tampere, Finland, pp. 4-8,
Sept., 2000.
[3] N. Katugampala, S. Villette, and A. Kondoz, "Secure
voice over GSM and other low bit rate systems," IEE Secure
GSM and Beyond: End to End Security for Mobile
Communications, London, Feb., 2003.
[4] J. Degener and C. Bormann." Gsm 06.10 lossy speech
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/gsm-1.0. 10.tar.gz.
[5] D. H. Klatt, "Software of cascade/parallel formant
synthesizer", J. Acoustic. Soc. Am. 67 (3), Massachusett, pp
971-996, Mar., 1980
[6] R.J. McAulay and T.F. Quatieri, "Speech
analysis/synthesis based on a sinusoidal representation,"
IEEE Trans. ASSP, vol.34, pp. 744-754, Aug., 1986.

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