This work demonstrates the recording and modulation of human voice through MATLAB IDE. Because the human voice cannot travel a far distance, because of its low frequency, due to attenuation, is multiplied with another signal that has higher frequency. The human voice used in this work has a frequency of 340Hz with amplitude 0.11V and it modulates a carrier frequency of 9800Hz with amplitude 0.15V. The modulated signal is demodulated to reproduce the human voice. The human voice reproduced has a lot of ripples which are removed by FIR low pass filter and the final output signal has an amplitude of 0.014V and frequency 338Hz.
Audio band is within the frequency range of 20Hz and 20KHz. This means that any signal between this range can be heard by humans [1].
However, audio wave cannot travel a far distance because of attenuation. Attenuation is the loss in signal strength as it travels form one location to another. Attenuation is represented in decibels (dB), which is ten times the logarithm of the signal power at a particular input divided by the signal at an output of a specified medium.(CAS DATA LOGGERS,n.d.).
The higher the frequency of a signal, the farther it can travel and vice versa. Thus, for an audio signal to be able to travel far, it must be modulated.
This work will be viewed under the following objectives:
Objectives
To determine the frequency at the centre of my voice in the MATLAB IDE
To produce a modulated frequency with my voice using a chosen a carrier frequency in the MATLAB IDE
To demodulate the modulated signal in order to recover back a sine wave( i.e the frequency of my voice) in the MATLAB IDE
Theory
Modulation: Modulation is the process of modifying some parameters of a CARRIER SIGNAL with the aid of a lower frequency-MODULATING SIGNAL. The message signal to be transmitted is the modulating signal. When a signal is modulated, its frequency characteristics are modified to match a certain purpose. Types of modulation is basically categorised into two depending on whether the technology employed is analogue or digital. The two categories are:
Analogue modulation
Digital modulation
Analogue Modulation Types
Amplitude Modulation (AM): is a process where the amplitude (voltage) applied to the carrier is varied over time
Frequency Modulation (FM): is a process where the frequency of the carrier waveform is made to change in a meaningful amount depending on the frequency of the modulating signal
Phase Modulation (PM): is a process where the natural flow of the periodic sinusoidal signal is delayed temporarily
The above analogue modulation can also be called continuous modulation [2].
Digital Modulation Types
Phase Shift Keying (PSK): this describes the modulation technique used to alter the phase of a carrier
Frequency Shift Keying (FSK): this describes the technique used in the modulation of a carrier which results in a frequency for a logic 1 and a different frequency for a logic 0
Amplitude Shift Keying (ASK): it involves the modulation technique used to multiply the carrier wave by the digital signal
Reasons for Modulation
It enhances easy radiation and reception of signals
It affords the use of practically realisable antenna sizes
It allows for channel assignment. That is, each message signal can be assigned a unique frequency band within the area of operation
It reduces noise and interference
It allows multiplexing of several messages over a single channel [2]
Am Modulation
Because AM modulation is used in this literature for audio signal to be transmitted, more emphasis will be laid on it.
In AM modulation, the information signal which is the modulating frequency varies the amplitude of the carrier sine wave. Thus, the amplitude of the carrier frequency changes in accordance with the variation in amplitude and frequency of the modulating signal. The frequency of the carrier waveform remains constant, but its amplitude changes in accordance with the amplitude of the modulating signal. Increase in the amplitude of the modulating signal causes a corresponding increase in the peak of the carrier frequency and vice versa.
AM is used in sound broadcasting(medium wave and short wave), in television(picture transmission), in telephony, etc.
The following is a list of some types of amplitude modulation:
DSB (double sideband full- carrier) AM
DSB (double sideband suppressed- carrier) AM
SSB-RC (single sideband reduced-carrier) AM
SSB-SC(single side band suppressed-carrier) AM
Vestigial sideband TV
Independent-sideband emission etc [2]
During modulation of AM signal, up to two-third of the transmitted power is in the carrier which does not carry any information. To improve the efficiency of amplitude modulation, the carrier wave must be suppressed.
If the carrier is suppressed the two side bands remaining, we have: Double side band suppressed carrier ( popularly called DSSC or DSB).
To better improve the efficiency of modulation, one sideband can be eliminated after the carrier is suppressed. This is called single side band full carrier ( SSB-FC).
In this assignment double side band (DSB-FC) was employed.
Construction of AM wave
An amplitude-modulated wave is obtained by superimposing modulating signal on the carrier signal as diagrammatically shown in Figure 1 below:
v2 = Vcsin 2
fct+ (Vm sin 2
fmt)(sin 2
fct)
Carrier wave can be expressed with the following mathematical expression:
vc = Vc sin 2
fct
(1)
vc= instantaneous value of the carrier sine wave voltage at some specific time in the circle
Vc= peak value of the constant unmodulatedcarrier sine wave as measured between zero and the maximum amplitude of eitherthe positive-going or the negative-going alternation

Figure 1: Modulation of Signal
Where
Vc = V(p-p)
fc = frequency of carrier sine wave
t = a point in time during the carrier cycle
Also, the modulating signal can be expressed with the following mathematical
Expression:
vm = Vm sin 2
fmt
(2)
vm= instantaneous value of information signal.
Vm= peak amplitude of information signal, where

Fm = frequency of carrier sine wave
t = a point in time during the information signal cycle
The degree of superimposition of modulating signal on the carrier signal is measured by the MODULATION DEPTH, m, of the AM wave. Modulation depth can be defined as the ratio of amplitude of modulating signal to the carrier signal amplitude. It can be expressed in percentage. Then modulation depth is Modulation Percentage(m%).

If m > 1 we’ve over modulation
If m = 1 we’ve critical modulation
If m < 1 we’ve under modulation
In percentage, we’ve:

In practice, the modulating percentage of an AM signal is never allowed to reach 100% or above. If this happens recovering the information will be almost impossible with a simple circuit. If modulation percentage exceeds 100%, this is called OVER-MODULATION [2].
To avoid distortion of modulated signal, it is important that the peak value of the modulating signal be less than the peak value of the carrier. Mathematically,
Vm<Vc
From the knowledge that the peak value of the carrier is the reference point for the modulating signal (i.e. the zero reference line of the modulating signal coincides with the peak value of unmodulated carrier), Formula to express the complete modulated wave is as derived below:
As shown in Figure 2-3 below, the instantaneous value of either the top or the bottom voltage envelope v1 can be computed by using the following equation:
v1 =Vc + vm = Vc + Vm sin 2
fm
(6)
The above equation expresses the fact that the instantaneous value of the modulating signal algebraically adds to the peak carrier. The instantaneous value of the complete modulated wave v2 can be written like as given below:
v2 = v1 sin 2
fct
(7)
v2 = (Vc+ Vm sin 2
fmt)sin 2
fct
(7.1)
v2 = Vcsin 2
fct+ (Vm sin 2
fmt)(sin 2
fct),
Where
v2 is the instantaneous value of the AM wave
Vcsin 2
fct is the carrier waveform
(Vm sin 2
fmt)(sin2
fct) is the product of carrier frequency and modulating frequency
Amplitude modulation showing input and output
Where


Figure 2: AM modulation of signals (Data Communications, n.d.)

Figure 3: Illustrating Over-Modulation

Figure 4: AM Spectrum

AM spectrum and Bandwidth
A simple AM signal contains three frequency components:
The carrier frequency, fm
The lower side-frequency, fc - fm
the upperside-frequency, fc +fm
Frequencies fc and fm are carrier carrier and modulating frequencies respectively.
As shown above, the two side frequencies contain the original message information. The two side-frequency-components have equal amplitude which is much smaller than the amplitude of the carrier frequency. This requires more power for transmission even though it has no message information.
For the transmission of AM signal whose spectrum is shown above, the two side frequencies must be given because if not, message will be lost. TRANSMISSION BANDWIDTH, B, is
B = (fc +fm) – (fc - fm) = 2 fm
Power Distribution in the AM Waveform
The total effective (rms) power, PTin the AM wave is the addition of effective carrier power level, Pc and the effective sideband power levels, PLSB and PUSB. i.e.
PT=Pc +PLSB+PUSB
(8)
= V2c(rms)/R + V2LSB(rms)/R + V2USB(rms)/R
(8.1)
= (0.707 Vc)2/R + ( 0.707VLSB)2/R + (0.707 VUSB)2/R
(8.2)
= Vc2/2R + VLSB2/2R + VUSB2/2R
(8.3)
Since power in the upper sideband is equal to power in the lower sideband And putting Vm = mVc into equation 8.2
We’ve:

Because total power Pcin the carrier is, Vc2/2R, we’ve:
We’ve: PLSB= PUSB = m2Pc/4
(8.4)
FromPT = PLSB+ PUSB+ Pc
We’ve: PT = m2Pc/4 + m2Pc/4 + Pc
PT = Pc (m2/4 + m2/4 + 1)
= Pc (m2/2 +1)
(8.5)
Power Distribution in AM Transmission for 100% Modulation
PT = Pc (12/2 + 1)
PT = 1.5 Pc
(8.6)
Conclusion
Following what we have in equation 8-4 above, if one is transmitting only one sideband of the 100% modulation, one-quarter of the effective power consumed by the carrier is used in transmission
If it is two sidebands of the 100% modulation is being transmitted, one and a half of the effective power consumed by the carrier is used in transmission
If carrier is suppressed, total effective power used in transmission of the 100% modulation AM signal is a lot reduced(i.e 1.5/Pc )
To Determine the Frequency at the Centre of my Voice in the MATLAB IDE
Matlab was loaded from the ‘start menu’ of the system and the window below came up:
The Simulink icon in MATLAB task bar was clicked upon and ‘Simulink Library Browser’ came up as shown below:

Figure 5: Matlab IDE

Figure 6: Simulink Library Browser

Figure 7: Model Window

Figure 8: Model Window Showing Procedure to Save File

Figure 9: Simulation of my Voice
From the Simulink library browser, ‘File’ followed by ‘New’ followed by ‘Model’ were clicked upon and a GUI work space came up as shown (Figure 6).
From the Simulink Model Window, ‘File’ was clicked upon followed by ‘Save As’ as shown above and this work was saved as ‘my voice.mdl’ on the desktop. From the ‘Simulink Library Browser’, ‘DSP System Tool Box’ was clicked upon followed by the ‘Sources.’ Then ‘From Audio Device’ block was dragged and dropped in ‘model window.’
From the ‘Simulink Library Browser’ again, ‘DSP System Tool Box’ was clicked upon again then after the ‘Sinks’ and ‘Time Scope’ was dragged and dropped in the Model Window. Both ‘From Audio Device’and’ Time Scope’ blocks were connected together as shown in the Figure 8.
Then earpiece was connected to the PC in which there was the setup shown in Figure 9 and the simulation was run. The result shown in Figure 10 below came up.
From the Time Scope shown above, amplitude of man’s voice is 0.11V Figure 11.
Conclusion
From the frequency spectrum of a man’s voice as shown in figure6 above and peak finder shown in figure7 above, the fundamental frequency of the man’s voice is found to be 411Hz and its highest harmonic 269Hz
Frequency at the centre of my voice range = average of the two frequencies given above


Figure 10: Showing Frequency Spectrum of Man’s Voice

Figure 11: Peak Finder Showing the Peak Frequencies of a Man’s Voice
To produce a modulated frequency with my voice using a chosen a carrier frequency in the
Matlab Ide
A new ‘Model Environment’ in the MPLAB IDE was opened. The procedure followed in the production of a man’s voice as described above was repeated but this time ‘Sine Wave’ block, ‘product ’ block and ‘Time Scope’ block were connected together as shown in figure8 below. The parameters of the message and carrier blocks were set as follows:
Message
Frequency 340Hz (fundamental frequency of my voice as produced in figure6 above)
Amplitude modulated signal was0.11V
F = 1/T, i.e. Sampling frequency Fs = 40000Hz
Fs> 2(Fc + BW), where BW is the original modulated signal bandwidth.
BW = 2fm = 2 x 340 = 680Hz
Fc = 9800Hz
Fs> 2( 9800 + 680)
Fs> 2 (10480)
Fs> 20960(meaning Fs must be greater than 20,928Hz)
So, choosing Fs = 40000Hz is justified (Mathworks, n.d.)
Carrier
Frequency chosen carrier frequency is 9800Hz because it must be higher than the information signal due to due Shannon’s Theorem as stated in point iii below
Amplitude was set at 0.15 ( because Vc must be greater than Vmto avoid distortion)
Sample Time was set at 1/40000 for the following reason
F = 1/T, i.e. Sampling frequency Fs = 40000Hz
Fs> 2(Fc + BW), where BW is the bandwidth of original modulated signal
BW = 2fm = 2 x 332 = 664Hz
Fc = 9800Hz
Fs> 2( 9800 + 680)
Fs> 2 (10480)
Fs> 20960 (meaning Fs must be greater than 20,960Hz)
So, choosing Fs = 40000Hz is justified
The type of Modulation Employed in this Assignment is DSB-FC
Then this setup was simulated, the outcome is as shown in Figure 13 below:

Figure 13: Modulated Signal
Calculations
Transmission bandwidth WB = 2 fm

Conclusion
My voice (with amplitude 0.11V and frequency 340Hz) modulated a carrier ( with amplitude 0.15V and frequency 9800Hz) to produce a modulated signal with the following parameters:
Bandwidth of transmission is 680kHz
Modulation index, m is 73.33%. This is under modulation. The ratio of modulating signal amplitude to carrier amplitude was carefully chosen to avoid distortion. Any value of m above 100%(called over modulation) causes distortion
The instantaneous value of my modulating signal at a given time is as given below:
vm = 0.11sin 122400t Volt
The instantaneous value of my carrier signal at a given time is as given below:
vc = 0.15sin 352800t Volt
The instantaneous voltage of the modulated waveform at a given timeis as given below:
v2= (0.5sin3528000t) + ((0.055cos3405600t) - (0.055cos3650400t))
Effective power distribution in the AM waveform is as given below:
PT = Pc (1.267) Watt
To demodulate the modulated signal in order to recover back a sine wave (i.e the frequency of my voice) in the MATLAB IDE
A modulated signal cannot be heard by human ear. This is because it is transmitted at the frequency of carrier signal and this is not audible by human ear. For the transmitted information to be heard by ordinary ear, it must be demodulated.
Thus, demodulation can be defined as a means of recovering information from modulated signal.
There are basically two types of modulation:
Coherent
Incoherent
Coherent Demodulation
This involves multiplying the AM signal by a sinusoid of exactly the same frequency and phase
Incoherent Demodulation
This does not involve multiplying the AM signal by a sinusoid
Importance of Demodulation
Demodulation is important because if it is not done, after filtering, the positive and negative envelopes will cancel each other so the original information cannot be recovered. This is because the modulation enveloped on the positive and negative sides are 180o out of phase. (TutorVista.Com,n.d.).

Figure 14: Showing Demodulation Action (TutorVista.Com,n.d.)
Important Characteristics of a Demodulator
Linearity
Signal handling capacity
Sensitivity
Linearity
This is the ability of a demodulator to produce an output with an amplitude of very insignificant variation to the amplitude of its input.
Signal handling capacity
This is a measure of signal amplitudes that a demodulator can accept without distortion.
Sensitivity
This is a measure of how much useful output is recovered when input is fed in. (TutorVista.Com,n.d.).
Demodulation in MATLAB IDE
Coherent Demodulation: which involves multiplying the modulated AM signal by a sinusoid of exactly the same frequency and phase of carrier signal was adopted in this work.
The procedure is as followed in modulation described above method but this time the same carrier frequency which was used for modulation was now multiplied with the modulated signal and the result was filtered (using digital low pass filter)as shown in Figure16 below.

Figure 15: Showing Chosen Parameters of Low Pass Filter

Figure16: Setup Displaying Demodulation Building Blocks
The parameters of the digital low pass filter were chosen as seen in figure15 above and enumerated below:
Uni Hz
Frequency pass(Fpass) 0.340kHz(my voice fundamental frequency)
Frequency stop(Fstop) 0.40kHz(frequency above my voice fundamental frequency)
F = 1/T, i.e Sampling frequency Fs = 40000Hz
Fs> 2(Fc + BW), where BW is the bandwidth of the original modulated signal
Generalized equiripple FIR was chosen as shown in Figure 15 above as my filter because it gave the maximum stopband attenuation. The higher the stopband attenuation the better the filtering ability of the filter
Transition band for the filter was chosen to be very small (i.e. 0.4 – 0.34 = 0.06Hz) because the thinner it is the more nearly ideal the filter become

Figure 17: Demodulated Voice

Figure 18: Showing Spectrum Analyser with The Display of the Frequency of Demodulated Signal

Figure 19: Clear Picture of Demodulated Signal Frequency from Spectrum Analyser
From Figure16 given above, it is shown through the Time Scope labelled ‘C’ that after the demodulation of modulated signal, some ripples were still present in the signal. The ripples were removed through the use of FIR filter. The final output is as shown in the Time Scope labelled ‘D’ in the same Figure16. And the signal is a replica of my voice but has suffered a lot of attenuation because its new amplitude has reduced to 0.014V.
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