TEEL 4204 – Communications Lab II
DTMF Receiver– Part I
Name:
Date:
Introduction
The circuit diagram of a telephone circuit is shown in Fig. 1. The telephone is generally
connected to the central office using a pair of twisted wires. However, new installations are using
a coaxial-fiber system for wide bandwidth communications. Unlike audio systems with
bandwidths of 20 Hz – 20 kHz for high fidelity sound, a telephone operates over the 300 Hz –
3.3 kHz bandwidth. The reason is that most of the energy of human voice is within this
frequency bandwidth and a 3 kHz bandwidth is enough for reliable conversation (Reliable: yes,
but not excellent!). The bandwidth limitation is the main reason why we have trouble
distinguishing “b” from “p” from “d” over a telephone. The audio voltage swing is 5-500 mV
peak, leading to a dynamic range of 40 dB, which is much lower than hi-fi system (dynamic
range of 70-90 dB). The telephone operates on a 48 V DC system supplied over a pair of lines
from the central telephone office. This is historical since the telephone was invented before the
AC 60 Hz power distribution system and could not be changed anymore. To grab your attention,
the control office sends bursts of a 20 Hz sinusoidal signal with a 75 V rms voltage to activate
the ringer. The bursts are on for 2 sec and off for 4 seconds. When a party answers the phone, the
telephone switch closes, the central office detects a DC current in the circuit and stops the
ringing signal. You could ask why 75 V? It is huge! The answer is that this signal was needed to
activate inefficient ringers on old telephones. In newer phones with electronic ringers, a TTL (5
V) digital signal is enough to activate the ringer. However, this telephone system will not be
compatible with old phones!
Fig. 1. The home telephone system.
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The lines between the central telephone office and your home therefore carry:
In order to dial a phone number, you need to transmit specific frequencies which are within the
300 Hz – 3,300 Hz range. However, if you assign a simple frequency to each number, then
somebody whistling when you are dialing (or a large clean sound) can actually cause you to
misdial! A very nice way to solve this interference problem is to send two frequencies for each
number. The probability that two specific frequencies with a ratio equal to a rational number are
present in the background noise when you are dialing is really very low!
The dial pad of a telephone is shown in Fig. 2. When a button is pushed, the two tones
corresponding to the intersection of the vertical and horizontal axes are sent. Notice that no
frequency is the harmonic of any other frequency thereby avoiding problems due to distortion
and harmonic generation. Also, no frequency can be synthesized from the sum or difference of
any two frequencies, thereby avoiding misdialing problems due to intermodulation products.
Fig. 2: A telephone dial-pad. (All frequencies are in Hz.)
Background
You will be asked to observe the power spectrum of a given signal (or combination of signals)
and design a series of digital filters that will allow you to isolate and analyze specific signal
frequencies. You will be designing the filters in a DTMF receiver to detect the numbers dialed
by the user. You have the option of designing both a Finite Impulse Response (FIR) and an
Infinite Impulse Response (IIR) filters. You will need to determine a sampling frequency, filter
specifications, and design method.
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4.2.1 Finite Impulse Response Filters
A FIR or non-recursive filter has no feedback. The filter’s output is a function of the input signal
only. The difference equation is given by
where, y[n] is the filter output, x[n] is the signal being filtered, bk are the filter coefficients, and
where N is the filter order. The output signal of the filter in response to an impulse is limited
only the last values of x[n], so after N + 1 samples the response returns to zero. For example, the
response of a fifth order filter (Fig. 21) consists of a finite sequence of 6(N + 1) samples.
4.2.2 Infinite Impulse Response filters
Infinite Impulse Response (or recursive) filters are a more complex type of filter than a FIR
filter, with an output at time n, given by:
where y[n] is the filter output, x[n] is the signal being filtered, and bk and ai are the filter
coefficients, and where N previous inputs and M outputs. The output signal of the filter can be
non-zero infinitely, even when the input signal has a value of zero. In theory, when a recursive
filter is excited by an impulse, the output will persist forever.
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The corresponding transfer function for an IIR filter is given by equation.
This equation, in addition to having N zeros (as the FIR, the roots of N(Z)), it also has poles M
poles, (the roots of D[Z]), which for a stable filter, are required to be inside the unit circle in the
z plane.
Sampling frequency
The symbol fs denotes the sampling frequency, which is the expected rate at which you sample
the input signal to the filter. In the LabVIEW Digital Filter Design Toolkit, the default sampling
frequency is 1, which is the normalized sampling frequency.
Filter design specifications
In this lab, you will be asked to design low-pass, high-pass, band-pass, and band-stop filters, the
characteristics of which are outlined below.
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As illustrated in Fig. 23 and Fig. 24 (low-pass and high-pass), the frequency range from the passband edge frequency (cut-off frequency) to the stop-band edge frequency is the transition band,
which has an unspecified frequency response. The filter pass-band and stop-band may contain
oscillations, which are known as ripples. A typical example of a ripple appears in the circle
(zoomed view) of the previous in Fig. 23. In the figure, δP indicates the magnitude of the passband ripple (or the maximum deviation from the unity) and δS indicates the magnitude of the
response of the stop-band ripple (or the maximum deviation from zero).
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For band-pass filters (Fig. 25), stop-band edge frequency 1 indicates the maximum frequency of
the lower frequency range to be attenuated, and the stop-band edge frequency 2 indicates the
minimum frequency of the higher frequency range to be attenuated. The frequency range
between pass-band edge frequencies 1 and 2 indicates the range of frequencies that can pass
through the filter.
For band-stop filters, (Fig. 26), pass-band edge frequency 1 indicates the maximum frequency of
the lower frequency range that can pass through the filter, and pass-band edge frequency 2
indicates the minimum frequency of the higher frequency range that can pass through the filter.
The frequency range between stop-band edge frequencies 1 and 2 indicates the range of
frequencies to be attenuated.
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Design methods
The LabVIEW Digital Filter Design Toolkit[1] provides the following finite impulse response
(FIR) filter design methods:



Kaiser Window
Dolph-Chebyshev Window
Equiripple FIR
The Kaiser Window method and the Dolph-Chebyshev Window method allow you to obtain the
filter coefficients directly from the analytical equations, making them easier to use than the
Equiripple FIR method. The Equiripple FIR method is more complicated because it uses a least
square optimization to produce an optimal filter and is often the best solution for most FIR filter
design problems.
In addition to the FIR-based methods, the Digital Filter Design Toolkit supports the following
infinite impulse response (IIR) filter design methods:




Butterworth
Chebyshev
Inverse Chebyshev
Elliptic
The following table summarizes the main features of the four IIR-based design methods so you
can determine the best IIR filter design method to use.
Pre-Lab
In this lab, you will be designing four digital filters: low pass, high-pass, band-pass, and bandstop. For each filter type, you will analyze the power spectrum of a chosen signal and frame a set
of specifications for a desired output. You will be using the Dual-Tone Multi-Frequency
(DTMF) coding scheme. You have probably heard them when you dial a telephone number.
Each key on the phone is assigned a pair of frequencies. Thus each key will two peaks in the
power spectrum of the signal (Fig. 27).
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Given a desired output, you can easily determine the frequency specifications of your filter
design by using tones of the DTMF coding.
a) A Low-pass filter that isolates the lower tone in the DTMF for a key press of 4 (Fig. 28).
Record your specifications in Table IV for future reference.
b) A High-pass filter that isolates the higher frequency in the DTMF for a key press of 4
(Fig. 28). Record your specifications in Table IV for future reference.
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Lab Procedure
Description
First, you will download to your PC’s hard disk the file DTMF_demo.vi which is available at
www.aprende-pr.com. As shown in the next Figure, you will be required to design the two
marked filters and the RMS voltage threshold necessary to detect the number 4 pulse. That is,
the filter parameters and the RMS voltage threshold should be selected so that only when you
generate the dual tones (770 Hz and 1209 Hz) the threshold will be exceeded and the receiver
will acknowledge, with a green light, that a “4” was received. Of course, the signal detection
becomes harder under the presence of noise, and if the noise power is high enough, the receiver
will occasionally fail in detecting the right signal.
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Double click on the “Classical Filter” VI and a pop-up window (Fig. 31) will appear. This popup window is your design tool screen where you enter the respective filter design specifications
(Table IV).
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Use either one, or both, of the following tables to report the parameters selected for each one of
the two designed filters.
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Your design will be judged on the following criteria:
1. Filter order: The smaller it is, the better. The higher the filter order, the higher is the
computational burden.
2. Ability to detect the signal when there is no noise present.
3. The highest noise power under which the receiver is able to successfully (i.e. without
errors) detect the signal.
When done, e-mail this report to ricardo.mediavilla@upr.edu. Do not forget to include your filter
specifications (Table 4 and/or Table 5). Also, e-mail your VI. It will be tested by your professor.
The best group design will earn full points, plus a 5 points bonus transferable to the mid-term
exam.
In the second part of this lab/project, you will be required to modify the current program so that
any key pressed, not just the number 4, is successfully detected by the receiver.
Start thinking how to do it!