implementation of dtmf encoder final document
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Implementation of DTMF
Encoder, Decoder
ABSTRACT
Dual Tone Multiple Frequency (DTMF) codec is used to encode and decode the key strokes in a
telephone. It can also be used to perform a basic data transfer operation. In this project a basic
DTMP encoder and decoder are implemented in Mat lab. The DTMF decoder is used to decode
unknown noisy versions of DTMF encoded data.
INTRODUCTION
Telecommunication
Telecommunication is the transmission of messages, over significant distances, for the purpose
of communication. In earlier times, telecommunications involved the use of visual signals, such
as beacons, smoke, semaphore telegraphs, signal flags, and optical heliographs, or audio
messages via coded drumbeats, lung-blown horns, or sent by loud whistles, for example. In the
modern age of electricity and electronics, telecommunications now also includes the use of
electrical devices such as telegraphs, telephones, and teletypes, the use of radio and microwave
communications, as well as fiber optics and their associated electronics, plus the use of
the orbiting satellites and the Internet.
The first breakthrough into modern electrical telecommunications came with the push to fully
develop the telegraph starting in the 1830s. The use of these electrical means of communications
exploded into use on all of the continents of the world during the 19th century, and these also
connected the continents via cables on the floors of the ocean. The use of the first three popular
systems of electrical telecommunications, the telegraph, telephone and teletype, all required the
use of conducting metal wires.
A revolution in wireless telecommunications began in the first decade of the 20th century,
with Guglielmo Marconi winning the Nobel Prize in Physics in 1909 for his pioneering
developments in wireless radio communications. Other highly notable pioneering inventors and
developers in the field of electrical and electronic telecommunications include Charles
Wheatstone and Samuel Morse (telegraph), Alexander Graham Bell(telephone), Nikola
Tesla, Edwin Armstrong, and Lee de Forest (radio), as well as John Logie Baird and Philo
Farnsworth (television).
Telecommunications play an important role in the world economy and the worldwide
telecommunication industry's revenue was estimated to be $3.85 trillion in 2008.[1] The service
revenue of the global telecommunications industry was estimated to be $1.7 trillion in 2008, and
is expected to touch $2.7 trillion by 2013.[1]
Key concepts
A number of key concepts reoccur throughout the literature on modern telecommunication
systems. Some of these concepts are discussed below.
Basic elements
A basic telecommunication system consists of three primary units that are always present in
some form:
A transmitter that takes information and converts it to a signal.
A transmission medium, also called the "physical channel" that carries the signal. An
example of this is the "free space channel".
A receiver that takes the signal from the channel and converts it back into usable information.
For example, in a radio broadcasting station the station's large power amplifier is the transmitter;
and the broadcasting antenna is the interface between the power amplifier and the "free space
channel". The free space channel is the transmission medium; and the receiver's antenna is the
interface between the free space channel and the receiver. Next, the radio receiver is the
destination of the radio signal, and this is where it is converted from electricity to sound for
people to listen to.
Sometimes, telecommunication systems are "duplex" (two-way systems) with a single box
of electronics working as both a transmitter and a receiver, or a transceiver. For example, a
cellular is a transceiver.[22] The transmission electronics and the receiver electronics in a
transceiver are actually quite independent of each other. This can be readily explained by the fact
that radio transmitters contain power amplifiers that operate with electrical powers measured in
the watts or kilowatts, but radio receivers deal with radio powers that are measured in
the microwatts or nanowatts. Hence, transceivers have to be carefully designed and built to
isolate their high-power circuitry and their low-power circuitry from each other.
Telecommunication over telephone lines is called point-to-point communication because it is
between one transmitter and one receiver. Telecommunication through radio broadcasts is
called broadcast communication because it is between one powerful transmitter and numerous
low-power but sensitive radio receiver. Telecommunications in which multiple transmitters and
multiple receivers have been designed to cooperate and to share the same physical channel are
called multiplex systems.
Analog or digital communications
Communications signals can be either by analog signals or digital signals. There are analog
communication systems and digital communication systems. For an analog signal, the signal is
varied continuously with respect to the information. In a digital signal, the information is
encoded as a set of discrete values (for example, a set of ones and zeros). During the propagation
and reception, the information contained in analog signals will inevitably be degraded
by undesirable physical noise.
(The output of a transmitter is noise-free for all practical purposes.) Commonly, the noise in a
communication system can be expressed as adding or subtracting from the desirable signal in a
completely random way. This form of noise is called "additive noise", with the understanding
that the noise can be negative or positive at different instants of time. Noise that is not additive
noise is a much more difficult situation to describe or analyze, and these other kinds of noise will
be omitted here. On the other hand, unless the additive noise disturbance exceeds a certain
threshold, the information contained in digital signals will remain intact. Their resistance to noise
represents a key advantage of digital signals over analog signals.
Communications networks
A communications network is a collection of transmitters, receivers, and communications
channels that send messages to one another. Some digital communications networks contain one
or more routers that work together to transmit information to the correct user. An analog
communications network consists of one or more switches that establish a connection between
two or more users. For both types of network, repeaters may be necessary to amplify or recreate
the signal when it is being transmitted over long distances. This is to combat attenuation that can
render the signal indistinguishable from the noise.
Communication channels
The term "channel" has two different meanings. In one meaning, a channel is the physical
medium that carries a signal between the transmitter and the receiver. Examples of this include
the atmosphere for sound communications, glass optical fibers for some kinds of optical
communications, coaxial cables for communications by way of the voltages and electric currents
in them, and free space for communications using visible light, infrared waves, ultraviolet light,
and radio waves. This last channel is called the "free space channel". The sending of radio waves
from one place to another has nothing to do with the presence or absence of an atmosphere
between the two. Radio waves travel through a perfect vacuum just as easily as they travel
through air, fog, clouds, or any other kind of gas besides air.
The other meaning of the term "channel" in telecommunications is seen in the
phrase communications channel, which is a subdivision of a transmission medium so that it can
be used to send multiple streams of information simultaneously. For example, one radio
station can broadcast radio waves into free space at frequencies in the neighborhood of
94.5 MHz(megahertz) while another radio station can simultaneously broadcast radio waves at
frequencies in the neighborhood of 96.1 MHz Each radio station would transmit radio waves
over a frequency bandwidth of about 180 kHz (kilohertz), centered at frequencies such as the
above, which are called the "carrier frequencies". Each station in this example is separated from
its adjacent stations by 200 kHz, and the difference between 200 kHz and 180 kHz (20 kHz) is
an engineering allowance for the imperfections in the communication system.
The example above, the "free space channel" has been divided into communications channels
according to frequencies, and each channel is assigned a separate frequency bandwidth in which
to broadcast radio waves. This system of dividing the medium into channels according to
frequency is called "frequency-division multiplexing" (FDM). Another way of dividing a
communications medium into channels is to allocate each sender a recurring segment of time (a
"time slot", for example, 20 milliseconds out of each second), and to allow each sender to send
messages only within its own time slot. This method of dividing the medium into communication
channels is called "time-division multiplexing" (TDM), and is used in optical fiber
communication. Some radio communication systems use TDM within an allocated FDM
channel. Hence, these systems use a hybrid of TDM and FDM.
Modulation
The shaping of a signal to convey information is known as modulation. Modulation can be used
to represent a digital message as an analog waveform. This is commonly called "keying" - a term
derived from the older use of Morse Code in telecommunications - and several keying techniques
exist (these include phase-shift keying, frequency-shift keying, and amplitude-shift keying). The
"Bluetooth" system, for example, uses phase-shift keying to exchange information between
various devices.[26][27] In addition, there are combinations of phase-shift keying and amplitude-
shift keying which is called (in the jargon of the field) "quadrature amplitude modulation"
(QAM) that are used in high-capacity digital radio communication systems.
Modulation can also be used to transmit the information of low-frequency analog signals at
higher frequencies. This is helpful because low-frequency analog signals cannot be effectively
transmitted over free space. Hence the information from a low-frequency analog signal must be
impressed into a higher-frequency signal (known as the "carrier wave") before transmission.
There are several different modulation schemes available to achieve this [two of the most basic
being amplitude modulation (AM) and frequency modulation (FM)]. An example of this process
is a disc jockey's voice being impressed into a 96 MHz carrier wave using frequency modulation
(the voice would then be received on a radio as the channel "96 FM").[28] In addition, modulation
has the advantage of being about to use frequency division multiplexing (FDM).
Signaling
In telecommunication, signaling (signaling in British spelling) has the following meanings:
the use of signals for controlling communications
the information exchange concerning the establishment and control of a telecommunication
circuit and the management of the network, in contrast to user information transfer
The sending of a signal from the transmitting end of a telecommunication circuit to inform a
user at the receiving end that a message is to be sent.
Signaling systems can be classified according to their principal properties, some of which are
described below:
In-band versus out-of-band signaling
In the public switched telephone network (PSTN), in-band signaling is the exchange of call
control information within the same channel that the telephone call itself is using. An example
is dual-tone multi-frequency signaling (DTMF), which is used on most telephone lines to
customer premises. Out-of-band signaling is telecommunication signaling on a channel that is
dedicated for the purpose and separate from the channels used for the telephone call. Out-of-
band signaling is used in Signaling System 7 (SS7), the standard for signaling among exchanges
that has controlled most of the world's phone calls for some twenty years.
Line versus register
Line signaling is concerned with conveying information on the state of the line or channel, such
as on-hook, off-hook (Answer supervision and Disconnect supervision, together referred to
as supervision), ringing current (alerting), and recall. In the middle 20th Century, supervision
signals on long distance trunks in North America were usually in band, for example at 2600 Hz,
necessitating a notch filter to prevent interference. Late in the century, all supervisory signals
were out of band. With the advent of digital trunks, supervision signals are carried by robbed or
other bits in the E1-carrier dedicated to signaling.
Register signaling is concerned with conveying addressing information, such as the calling
and/or called telephone number. In the early days of telephony, with operator handling calls, the
addressing information is by voice as "Operator, connect me to Mr. Smith please". In the first
half of the 20th century, addressing information is by using a rotary dial, which rapidly breaks
the line current into pulses, with the number of pulses conveying the address. Finally, starting in
the second half of the century, address signaling is by DTMF.
Channel-associated versus common-channel signaling
Channel Associated Signaling (CAS) employs a signaling channel which is dedicated to a
specific bearer channel. Common Channel Signaling (CCS) employs a signaling channel which
conveys signaling information relating to multiple bearer channels. These bearer channels
therefore have their signaling channel in common.
Compelled signaling
Compelled signaling is the case where receipt of each signal needs to be explicitly acknowledged
before the next signal is able to be sent. Most forms of R2 register signaling are compelled
(see R2 signaling), while R1 multi-frequency signaling is not. The term is only relevant in the
case of signaling systems that use discrete signals (e.g. a combination of tones to denote one
digit), as opposed to signaling systems which are message-oriented (such as SS7 and ISDN
Q.931) where each message is able to convey multiple items of information (e.g. multiple digits
of the called telephone number).
Subscriber versus trunk signaling
Subscriber signaling is between the telephone and the telephone exchange. Trunk signaling is
signaling between exchanges.
Classification examples
Note that every signaling system can be characterized along each of the above axes of
classification. A few examples:
DTMF is an in-band, channel-associated register signaling system. It is not compelled.
SS7 (e.g. TUP or ISUP) is an out-of-band, common-channel signaling system that
incorporates both line and register signaling.
Metering pulses (depending on the country, these are 50 Hz, 12 kHz or 16 kHz pulses
sent by the exchange to payphones or metering boxes) are out-of-band (because they do
not fall within the frequency range used by the telephony signal, which is 300 through
3400 Hz) and channel-associated. They are generally regarded as line signaling, although
this is open to debate.
E and M signaling (E&M) is an out-of-band channel-associated signaling system. The
base system is intended for line signaling, but if decade pulses are used it can also convey
register information. E&M line signaling is however usually paired with DTMF register
signaling.
By contrast, the L1 signaling system (which typically employs a 2280 Hz tone of various
durations) is an in-band channel-associated signaling system as was the SF hertz system
formerly used in the Bell System.
Loop start, Ground start, Reverse Battery and Reverie Pulse systems are all DC, thus out
of band, and all are channel-associated, since the DC currents are on the talking wires.
Whereas common-channel signaling systems are out-of-band by definition, and in-band
signaling systems are also necessarily channel-associated, the above metering pulse example
demonstrates that there exist channel-associated signaling systems which are out-of-band.
Modulation
Electronics, modulation is the process of varying one or more properties of a high frequency
periodic waveform, called the carrier signal, with respect to a modulating signal. This is done in
a similar fashion as a musician may modulate a tone (a periodic waveform) from a musical
instrument by varying its volume, timing and pitch. The three key parameters of a periodic
waveform are its amplitude ("volume"), its phase ("timing") and its frequency ("pitch"), all of
which can be modified in accordance with a low frequency signal to obtain the modulated signal.
Typically a high-frequency sinusoid waveform is used as carrier signal, but a square wave pulse
train may also occur.
In telecommunications, modulation is the process of conveying a message signal, for example a
digital bit stream or an analog audio signal, inside another signal that can be physically
transmitted. Modulation of a sine waveform is used to transform a baseband message signal to
a passband signal, for example a radio-frequency signal (RF signal). In radio communications,
cable TV systems or the public switched telephone network for instance, electrical signals can
only be transferred over a limited passband frequency spectrum, with specific (non-zero) lower
and upper cutoff frequencies. Modulating a sine wave carrier makes it possible to keep the
frequency content of the transferred signal as close as possible to the centre frequency (typically
the carrier frequency) of the passband. When coupled with demodulation, this technique can be
used to, among other things, transmit a signal through a channel which may be opaque to the
baseband frequency range (for instance, when sending a telephone signal through a fiber-
optic strand).
In music synthesizers, modulation may be used to synthesize waveforms with a desired overtone
spectrum. In this case the carrier frequency is typically in the same order or much lower than the
modulating waveform. See for example frequency modulation synthesis or ring A device that
performs modulation is known as a modulator and a device that performs the inverse operation
of modulation is known as a demodulator (sometimes detector or demod). A device that can do
both operations is a modem (short for "Modulator-Demodulator").
Aim
The aim of digital modulation is to transfer a digital bit stream over an
analog passband channel, for example over the public switched telephone network (where
a bandpass filter limits the frequency range to between 300 and 3400 Hz), or over a limited radio
frequency band.
The aim of analog modulation is to transfer an analog baseband (or lowpass) signal, for
example an audio signal or TV signal, over an analog passband channel, for example a limited
radio frequency band or a cable TV network channel. Analog and digital modulation
facilitate frequency division multiplexing (FDM), where several low pass information signals are
transferred simultaneously over the same shared physical medium, using separate passband
channels.
The aim of digital baseband modulation methods, also known as line coding, is to transfer a
digital bit stream over a baseband channel, typically a non-filtered copper wire such as a serial or
a wired local area network.
The aim of pulse modulation methods is to transfer a narrowband analog signal, for example a
phone call over a wideband baseband channel or, in some of the schemes, as a bit stream over
another digital transmission system.
Analog modulation methods
In analog modulation, the modulation is applied continuously in response to the analog
information signal.
A low-frequency message signal (top) may be carried by an AM or FM radio wave.
Common analog modulation techniques are:
Amplitude modulation (AM) (here the amplitude of the carrier signal is varied in
accordance to the instantaneous amplitude of the modulating signal)
Double-sideband modulation (DSB)
Double-sideband modulation with carrier (DSB-WC) (used on the AM
radio broadcasting band)
Double-sideband suppressed-carrier transmission (DSB-SC)
Double-sideband reduced carrier transmission (DSB-RC)
Single-sideband modulation (SSB, or SSB-AM),
SSB with carrier (SSB-WC)
SSB suppressed carrier modulation (SSB-SC)
Vestigial sideband modulation (VSB, or VSB-AM)
Quadrature amplitude modulation (QAM)
Angle modulation
Frequency modulation (FM) (here the frequency of the carrier signal is varied in
accordance to the instantaneous amplitude of the modulating signal)
Phase modulation (PM) (here the phase shift of the carrier signal is varied in
accordance to the instantaneous amplitude of the modulating signal)
The accompanying figure shows the results of (amplitude-) modulating a signal onto a carrier
(both of which are sine waves). At any point along the y-axis, the amplitude of the modulated
signal is equal to the sum of the carrier signal and the modulating signal amplitudes.
Simple example of amplitude modulation.
Digital modulation methods
In digital modulation, an analog carrier signal is modulated by a digital bit stream. Digital
modulation methods can be considered as digital-to-analog conversion, and the corresponding
demodulation or detection as analog-to-digital conversion. The changes in the carrier signal are
chosen from a finite number of M alternative symbols (the modulation alphabet).
Schematic of 4 baud (8 bps) data link.
A simple example: A telephone line is designed for transferring audible sounds, for example
tones, and not digital bits (zeros and ones). Computers may however communicate over a
telephone line by means of modems, which are representing the digital bits by tones, called
symbols. If there are four alternative symbols (corresponding to a musical instrument that can
generate four different tones, one at a time), the first symbol may represent the bit sequence 00,
the second 01, the third 10 and the fourth 11. If the modem plays a melody consisting of 1000
tons per second, the symbol rate is 1000 symbols/second, or baud. Since each tone (i.e., symbol)
represents a message consisting of two digital bits in this example, the bit rate is twice the
symbol rate, i.e. 2000 bits per second. This is similar to the technique used by dialup modems as
opposed to DSL modems.
.
According to one definition of digital signal, the modulated signal is a digital signal, and
according to another definition, the modulation is a form of digital. Most textbooks would
consider digital modulation schemes as a form of digital transmission, synonymous to data
transmission; very few would consider it as analog.
Fundamental digital modulation methods
The most fundamental digital modulation techniques are based on keying:
In the case of PSK (phase-shift keying), a finite number of phases are used.
In the case of FSK (frequency-shift keying), a finite number of frequencies are used.
In the case of ASK (amplitude-shift keying), a finite number of amplitudes are used.
In the case of QAM (quadrature amplitude modulation), a finite number of at least two
phases, and at least two amplitudes are used.
In QAM, an in phase signal (the I signal, for example a cosine waveform) and a quadrature phase
signal (the Q signal, for example a sine wave) are amplitude modulated with a finite number of
amplitudes, and summed. It can be seen as a two-channel system, each channel using ASK. The
resulting signal is equivalent to a combination of PSK and ASK.
In all of the above methods, each of these phases, frequencies or amplitudes are assigned a
unique pattern of binary bits. Usually, each phase, frequency or amplitude encodes an equal
number of bits. This number of bits comprises the symbol that is represented by the particular
phase, frequency or amplitude.
If the alphabet consists of M = 2N alternative symbols, each symbol represents a message
consisting of N bits. If the symbol rate (also known as the baud rate) is fS symbols/second
(or baud), the data rate is NfS bit/second. For example, with an alphabet consisting of 16
alternative symbols, each symbol represents 4 bits. Thus, the data rate is four times the baud rate.
In the case of PSK, ASK or QAM, where the carrier frequency of the modulated signal is
constant, the modulation alphabet is often conveniently represented on a constellation diagram,
showing the amplitude of the I signal at the x-axis, and the amplitude of the Q signal at the y-
axis, for each symbol.
Modulator and detector principles of operation
PSK and ASK, and sometimes also FSK, are often generated and detected using the principle of
QAM. The I and Q signals can be combined into a complex-valued signal I+jQ (where jis
the imaginary unit). The resulting so called equivalent low pass signal or equivalent baseband
signal is a complex-valued representation of the real-valued modulated physical signal (the so
called passband signal or RF signal).
These are the general steps used by the modulator to transmit data:
1. Group the incoming data bits into codewords, one for each symbol that will be
transmitted.
2. Map the codewords to attributes, for example amplitudes of the I and Q signals (the
equivalent low pass signal), or frequency or phase values.
3. Adapt pulse shaping or some other filtering to limit the bandwidth and form the spectrum
of the equivalent low pass signal, typically using digital signal processing.
4. Perform digital-to-analog conversion (DAC) of the I and Q signals (since today all of the
above is normally achieved using digital signal processing, DSP).
5. Generate a high-frequency sine wave carrier waveform, and perhaps also a cosine
quadrature component. Carry out the modulation, for example by multiplying the sine
and cosine wave form with the I and Q signals, resulting in that the equivalent low pass
signal is frequency shifted into a modulated passband signal or RF signal. Sometimes this
is achieved using DSP technology, for example direct digital synthesis using a waveform
table, instead of analog signal processing. In that case the above DAC step should be
done after this step.
6. Amplification and analog bandpass filtering to avoid harmonic distortion and periodic
spectrum
At the receiver side, the demodulator typically performs:
1. Bandpass filtering.
2. Automatic gain control, AGC (to compensate for attenuation, for example fading).
3. Frequency shifting of the RF signals to the equivalent baseband I and Q signals, or to an
intermediate frequency (IF) signal, by multiplying the RF signal with a local oscillator
sine wave and cosine wave frequency (see the super heterodyne receiver principle).
4. Sampling and analog-to-digital conversion (ADC) (Sometimes before or instead of the
above point, for example by means of under sampling).
5. Equalization filtering, for example a matched filter, compensation for multipath
propagation, time spreading, phase distortion and frequency selective fading, to
avoid intersymbol interference and symbol distortion.
6. Detection of the amplitudes of the I and Q signals, or the frequency or phase of the IF
signal.
7. Quantization of the amplitudes, frequencies or phases to the nearest allowed symbol
values.
8. Mapping of the quantized amplitudes, frequencies or phases to codewords (bit groups).
9. Parallel-to-serial conversion of the codewords into a bit stream.
10. Pass the resultant bit stream on for further processing such as removal of any error-
correcting codes.
As is common to all digital communication systems, the design of both the modulator and
demodulator must be done simultaneously. Digital modulation schemes are possible because the
transmitter-receiver pair have prior knowledge of how data is encoded and represented in the
communications system. In all digital communication systems, both the modulator at the
transmitter and the demodulator at the receiver are structured so that they perform inverse
operations.
Non-coherent modulation methods do not require a receiver reference clock signal that is phase
synchronized with the sender carrier wave. In this case, modulation symbols (rather than bits,
characters, or data packets) are asynchronously transferred. The opposite is coherent modulation.
List of common digital modulation techniques
The most common digital modulation techniques are:
Phase-shift keying (PSK):
Binary PSK (BPSK), using M=2 symbols
Quadrature PSK (QPSK), using M=4 symbols
8PSK, using M=8 symbols
16PSK, using M=16 symbols
Differential PSK (DPSK)
Differential QPSK (DQPSK)
Offset QPSK (OQPSK)
π/4–QPSK
Frequency-shift keying (FSK):
Audio frequency-shift keying (AFSK)
Multi-frequency shift keying (M-ary FSK or MFSK)
Dual-tone multi-frequency (DTMF)
Continuous-phase frequency-shift keying (CPFSK)
Amplitude-shift keying (ASK)
On-off keying (OOK), the most common ASK form
M-ary vestigial sideband modulation, for example 8VSB
Quadrature amplitude modulation (QAM) - a combination of PSK and ASK:
Polar modulation like QAM a combination of PSK and ASK.[citation needed]
Continuous phase modulation (CPM) methods:
Minimum-shift keying (MSK)
Gaussian minimum-shift keying (GMSK)
Orthogonal frequency-division multiplexing (OFDM) modulation:
Discrete multitone (DMT) - including adaptive modulation and bit-loading.
Wavelet modulation
Trellis coded modulation (TCM), also known as trellis modulation
Spread-spectrum techniques:
Direct-sequence spread spectrum (DSSS)
Chirp spread spectrum (CSS) according to IEEE 802.15.4a CSS uses pseudo-
stochastic coding
Frequency-hopping spread spectrum (FHSS) applies a special scheme for channel
release
MSK and GMSK are particular cases of continuous phase modulation. Indeed, MSK is a
particular case of the sub-family of CPM known as continuous-phase frequency-shift
keying(CPFSK) which is defined by a rectangular frequency pulse (i.e. a linearly increasing
phase pulse) of one symbol-time duration (total response signaling).
OFDM is based on the idea of frequency-division multiplexing (FDM), but is utilized as a digital
modulation scheme. The bit stream is split into several parallel data streams, each transferred
over its own sub-carrier using some conventional digital modulation scheme. The modulated
sub-carriers are summed to form an OFDM signal. OFDM is considered as a modulation
technique rather than a multiplex technique, since it transfers one bit stream over one
communication channel using one sequence of so-called OFDM symbols. OFDM can be
extended to multi-user channel access method in the orthogonal frequency-division multiple
access (OFDMA) and multi-carrier code division multiple access (MC-CDMA) schemes,
allowing several users to share the same physical medium by giving different sub-carriers
or spreading codes to different users.
Of the two kinds of RF power amplifier, switching amplifiers (Class C amplifiers) cost less and
use less battery power than linear amplifiers of the same output power. However, they only work
with relatively constant-amplitude-modulation signals such as angle modulation (FSK or PSK)
and CDMA, but not with QAM and OFDM. Nevertheless, even though switching amplifiers are
completely unsuitable for normal QAM constellations, often the QAM modulation principle are
used to drive switching amplifiers with these FM and other waveforms, and sometimes QAM
demodulators are used to receive the signals put out by these switching amplifiers.
Digital baseband modulation or line coding
Main article: Line code
The term digital baseband modulation (or digital baseband transmission) is synonymous to line
codes. These are methods to transfer a digital bit stream over an analog baseband channel
(a.k.a. low pass channel) using a pulse train, i.e. a discrete number of signal levels, by directly
modulating the voltage or current on a cable. Common examples are unipolar, non-return-to-
zero (NRZ), Manchester and alternate mark inversion (AMI) codlings.
Pulse modulation methods
Pulse modulation schemes aim at transferring a narrowband analog signal over an analog
baseband channel as a two-level signal by modulating a pulse wave. Some pulse modulation
schemes also allow the narrowband analog signal to be transferred as a digital signal (i.e. as
a quantized discrete-time signal) with a fixed bit rate, which can be transferred over an
underlying digital transmission system, for example some line code. These are not modulation
schemes in the conventional sense since they are not channel coding schemes, but should be
considered as source coding schemes, and in some cases analog-to-digital conversion techniques.
Analog-over-analog methods:
Pulse-amplitude modulation (PAM)
Pulse-width modulation (PWM)
Pulse-position modulation (PPM)
Analog-over-digital methods:
Pulse-code modulation (PCM)
Differential PCM (DPCM)
Adaptive DPCM (ADPCM)
Delta modulation (DM or Δ-modulation)
Sigma-delta modulation (∑Δ)
Continuously variable slope delta modulation (CVSDM), also called Adaptive-delta
modulation (ADM)
Pulse-density modulation (PDM)
Miscellaneous modulation techniques
The use of on-off keying to transmit Morse code at radio frequencies is known
as continuous wave (CW) operation.
Adaptive modulation
Space modulation A method whereby signals are modulated within airspace, such as that
used in Instrument landing systems.
Dual-tone multi-frequency signaling
Dual-tone multi-frequency signaling (DTMF) is used for telecommunication signaling over
analog telephone lines in the voice-frequency band between telephone handsets and other
communications devices and the switching center. The version of DTMF that is used in push-
button telephones for tone dialing is known as Touch-Tone, was first used by AT&T in
commerce as a registered trademark, and is standardized by ITU-T Recommendation Q.23. It is
also known in the UK as MF4.
Other multi-frequency systems are used for internal signaling within the telephone network. The
Touch-Tone system, using the telephone keypad, gradually replaced the use of rotary
dial starting in 1963, and since then DTMF or Touch-Tone became the industry standard for
both cell phones and landline service.[1]
Multi frequency signaling
Prior to the development of DTMF, automated telephone systems employed pulse dialing (Dial
Pulse or DP in the U.S.) or loop disconnect (LD) signaling to dial numbers. It functions by
rapidly disconnecting and re-connecting the calling party's telephone line, similar to flicking a
light switch on and off. The repeated interruptions of the line, as the dial spins, sounds like a
series of clicks. The exchange equipment interprets these dial pulses to determine the dialed
number. Loop disconnect range was restricted by telegraphic distortion and other technical
problems[which?] , and placing calls over longer distances required either operator assistance
(operators used an earlier kind of multi-frequency dial) or the provision of subscriber trunk
dialing equipment.
Multi-frequency signaling (see also MF) is a group of signaling methods, that use a mixture of
two pure tone (pure sine wave) sounds. Various MF signaling protocols were devised by the Bell
System and CCITT. The earliest of these were for in-band signaling between switching centers,
where long-distance telephone operators used a 16-digit keypad to input the next portion of the
destination telephone number in order to contact the next downstream long-distance telephone
operator. This semi-automated signaling and switching proved successful in both speed and cost
effectiveness. Based on this prior success with using MF by specialists to establish long-
distance telephone calls, Dual-tone multi-frequency (DTMF) signaling was developed for
the consumer to signal their own telephone-call's destination telephone number instead of talking
to a telephone operator.
AT&Ts Compatibility Bulletin No. 105 described the product as "a method for pushbutton
signaling from customer stations using the voice transmission path." In order to prevent using a
consumer telephone to interfere with the MF-based routing and switching between telephone
switching centers, DTMF's frequencies differ from all of the pre-existing MF signaling protocols
between switching centers: MF/R1, R2, CCS4, CCS5, and others that were later replaced
by SS7 digital signaling. DTMF, as used in push-button telephone tone dialing, was known
throughout the Bell System by the trademark Touch-Tone. This term was first used by AT&T in
commerce on July 5, 1960 and then was introduced to the public on November 18, 1963, when
the first push-button telephone was made available to the public. It was AT&T's registered
trademark from September 4, 1962 to March 13, 1984,[2] and is standardized byITU-
T Recommendation Q.23. It is also known in the UK as MF4.
Other vendors of compatible telephone equipment called the Touch-Tone feature Tone
dialing or DTMF, or used their own registered trade names such as the Digit one of Northern
Electric (now known as Nortel Networks). The DTMF system uses eight different frequency
signals transmitted in pairs to represent sixteen different numbers, symbols and letters - as
detailed below.
As a method of in-band signaling, DTMF tones were also used by cable
television broadcasters to indicate the start and stop times of local commercial insertion points
during station breaks for the benefit of cable companies. Until better out-of-band
signaling equipment was developed in the 1990s, fast, unacknowledged, and loud DTMF tone
sequences could be heard during the commercial breaks of cable channels in the United States
and elsewhere.
#, *, A, B, C, and D
DTMF keypad layout.
The engineers[who?] had envisioned[when?] phones being used to access computers, and surveyed a
number of companies to see what they would need for this role. This led to the addition of
the number sign (#, sometimes called 'octothorpe' or 'pound' in this context - 'hash' or 'gate' in the
UK) and asterisk or "star" (*) keys as well as a group of keys for menu selection: A, B, C and D.
In the end, the lettered keys were dropped from most phones, and it was many years before these
keys became widely used for vertical service codes such as *67 in the United States and Canada
to suppress caller ID. Public payphones that accept credit cards use these additional codes to
send the information from the magnetic strip.
The U.S. military also used the letters, relabeled, in their now defunct AutoVIN phone system.
Here they were used before dialing the phone in order to give some calls priority, cutting in over
existing calls if need be. The idea was to allow important traffic to get through every time. The
levels of priority available were Flash Override (A), Flash (B), Immediate (C), and Priority (D),
with Flash Override being the highest priority. Pressing one of these keys gave your call priority,
overriding other conversations on the network. Pressing C, Immediate, before dialing would
make the switch first look for any free lines, and if all lines were in use, it would disconnect any
non-priority calls, and then any priority calls. Flash Override will kick every other call off the
DTMF dialing
How DTMF dialing sounds.
Problems listening to this file? See media help.
trunks between the origin and destination. Consequently, it was limited to the White House
Communications Agency. Precedence dialing is still done on the military phone networks, but
using number combinations (Example: Entering 93 before a number is a priority call) rather than
the separate tones and the Government Emergency Telecommunications Service has
superseded Autovonfor any civilian priority Telco access.
Present-day uses of the A, B, C and D keys on telephone networks are few, and exclusive to
network control. For example, the A key is used on some networks to cycle through different
carriers at will (thereby listening in on calls). Their use is probably prohibited by most carriers.
The A, B, C and D tones are used in amateur radio phone patch and repeater operations to allow,
among other uses, control of the repeater while connected to an active phone line. DTMF tones
are also used by some cable television networks and radio networks to signal the local cable
company/network station to insert a local advertisement or station identification. These tones
were often heard during a station ID preceding a local ad insert. Previously, terrestrial television
stations also used DTMF tones to shut off and turn on remote transmitters.
DTMF signaling tones can also be heard at the start or end of some VHS (Video Home System)
cassette tapes. Information on the master version of the video tape is encoded in the DTMF tone.
The encoded tone provides information to automatic duplication machines, such as format,
duration and volume levels, in order to replicate the original video as closely as possible. DTMF
tones are sometimes used in caller ID systems to transfer the caller ID information, however in
the USA only Bell 202 modulated FSK signaling is used to transfer the data. A DTMF can be
heard on most Whelen Outdoor Warning systems.
Keypad
1209 Hz on 697 Hz to make the 1 tone
The DTMF keypad is laid out in a 4×4 matrix, with each row representing a low frequency, and
each column representing a high frequency. Pressing a single key (such as '1' ) will send
a sinusoidal tone for each of the two frequencies (697 and 1209 hertz (Hz)). The original
keypads had levers inside, so each button activated two contacts. The multiple tones are the
reason for calling the system multifrequency. These tones are then decoded by the switching
center to determine which key was pressed.
DTMF keypad frequencies (with sound clips)
1209 Hz 1336 Hz 1477 Hz 1633 Hz
697 Hz 1 2 3 A
770 Hz 4 5 6 B
852 Hz 7 8 9 C
941 Hz * 0 # D
Special tone frequencies
National telephone systems define additional tones to indicate the status of lines, equipment, or
the result of calls with special tones. Such tones are standardized in each country and may
consist of single or multiple frequencies. Most European countries use a single frequency, where
the United States uses a dual frequency system presented in the following table.
Event Low frequency High frequency
Busy signal 480 Hz 620 Hz
Ring back
tone (US)440 Hz 480 Hz
Dial tone 350 Hz 440 Hz
The tone frequencies, as defined by the Precise Tone Plan, are selected such
that harmonics and intermediation products will not cause an unreliable signal. No frequency is a
multiple of another, the difference between any two frequencies does not equal any of the
frequencies, and the sum of any two frequencies does not equal any of the frequencies.
The frequencies were initially designed with a ratio of 21/19, which is slightly less than a whole
tone. The frequencies may not vary more than ±1.8% from their nominal frequency, or the
switching center will ignore the signal. The high frequencies may be the same volume as – or
louder than – the low frequencies when sent across the line.
The loudness difference between the high and low frequencies can be as large as 3 decibels (dB)
and is referred to as "twist." The duration of the tone should be at least 70 ms, although in some
countries and applications DTMF receivers must be able to reliably detect DTMF tones as short
as 45m. As with other multi-frequency receivers, DTMF was originally decoded by tuned filter
banks. Late in the 20th century most were replaced with digital signal processors. DTMF can be
decoded using the Goertzel algorithm.
Encoder
An encoder is a device, circuit, transducer, software program, algorithm or person
that converts information from one format or code to another, for the purposes of
standardization, speed, secrecy, security, or saving space by shrinking size.
Examples
Media
Software for encoding audio, video, text into standardized formats:
A compressor encodes data (e.g., audio/video/images) into a smaller form (See codec.)
An audio encoder may be capable of capturing, compressing and converting audio
A video encoder may be capable of capturing, compressing and converting audio/video
An email encoder secures online email addresses from email harvesters
A PHTML encoder preserves script code logic in a secure format that is transparent to
visitors on a web site
A multiplexer combines multiple inputs into one output.
Job positions
A Data Entry Encoder may enter data from phone surveys in a coded format into a
database.
A Data Entry Encoder may enter payment amounts from legal tender documents from
financial institutions into a database.
A Manual Encoder may manually scan code tags on baggage that were missed by an
automated system.
Security
A device or person that encodes or encrypts military messages, such as the ADFGVX
Cipher in WWI or the Enigma device in WWII.
A Microchip hopping encoder integrated circuit for non-fixed-code secured entry.
Medical encoding software
EncoderPro searches ICD-9-CM, CPT, and HCPCS Level II medical codes, to increase
accuracy and allow ease of auditing for compliance.
Transducers
Transducers (such as optical or magnetic encoders) sense position or orientation for use as a
reference or active feedback to control position:
A rotary encoder converts rotary position to an analog (e.g., analog quadrature) or digital
(e.g., digital quadrature, 32-bit parallel, or USB) electronic signal.
A linear encoder similarly converts linear position to an electronic signal.
Such encoders can be either absolute or incremental. The signal from an absolute encoder gives
an unambiguous position within the travel range without requiring knowledge of any previous
position. The signal from an incremental encoder is cyclical, thus ambiguous, and requires
counting of cycles to maintain absolute position within the travel range. Both can provide the
same accuracy, but the absolute encoder is more robust to interruptions in transducer signal.
Telecommunications
A device used to change a signal (such as a bit stream) or data into a code.
Encoder circuits
A simple encoder assigns a binary code to an active input line.
Priority encoders establish the priority of competing inputs (such as interrupt requests) by
outputting a binary code representing the highest-priority active input.
Decoder
A decoder is a device which does the reverse of an encoder, undoing the encoding so that the
original information can be retrieved. The same method used to encode is usually just reversed in
order to decode. In digital electronics, a decoder can take the form of a multiple-input, multiple-
output logic circuit that converts coded inputs into coded outputs, where the input and output
codes are different. e.g. n-to-2n, binary-coded decimal decoders. Enable inputs must be on for the
decoder to function, otherwise its outputs assume a single "disabled" output code word.
Decoding is necessary in applications such as data multiplexing, 7 segment display
and memory address decoding.
The example decoder circuit would be an AND gate because the output of an AND gate is
"High" (1) only when all its inputs are "High." Such output is called as "active High output". If
instead of AND gate, the NAND gate is connected the output will be "Low" (0) only when all its
inputs are "High". Such output is called as "active low output".
Example: A 2-to-4 Line Single Bit Decoder
A slightly more complex decoder would be the n-to-2n type binary decoders. These type of
decoders are combinational circuits that convert binary information from 'n' coded inputs to a
maximum of 2n unique outputs. We say a maximum of 2n outputs because in case the 'n' bit coded
information has unused bit combinations, the decoder may have less than 2n outputs. We can
have 2-to-4 decoder, 3-to-8 decoder or 4-to-16 decoder. We can form a 3-to-8 decoder from two
2-to-4 decoders (with enable signals).
Similarly, we can also form a 4-to-16 decoder by combining two 3-to-8 decoders. In this type of
circuit design, the enable inputs of both 3-to-8 decoders originate from a 4th input, which acts as
a selector between the two 3-to-8 decoders. This allows the 4th input to enable either the top or
bottom decoder, which produces outputs of D(0) through D(7) for the first decoder, and D(8)
through D(15) for the second decoder.
A decoder that contains enable inputs is also known as a decoder-demultiplexer. Thus, we have a
4-to-16 decoder produced by adding a 4th input shared among both decoders, producing 16
outputs.
Row select
Most kinds of random-access memory use a n-to-2n decoder to convert the selected address on
the address bus to one of the row address select lines.
Instruction decoder
In CPU design, the instruction decoder is the part of the CPU that converts the bits stored in
the instruction register -- or, in CPUs that have microcode, the microinstruction -- into the
control signals that control the other parts of the CPU. A simple CPU with 8 registers may use 3-
to-8 logic decoders inside the instruction decoder to select two source registers of the register
file to feed into the ALU as well as the destination register to accept the output of the ALU. A
typical CPU instruction decoder also includes several other things.
Implementation of DTMF Encoder, Decoder
Introduction
Telephone signaling is based on encoding keypad digits using two sinusoids of different
frequencies, hence the name DTMF. Each digit is represented by a low frequency and a high
frequency sinusoid. The frequencies used were recommended by AT&T such that no two
frequencies are integral multiples of each other. This facilitates correct decoding even in the
presence of non linearity of filters which cause higher harmonics to be present.
1 2 3 A 697
4 5 6 B 770
7 8 9 C 852
* 0 # D 941
1209 1336 1477 1633 Hz
The above table gives the lower and higher frequencies associated with each frequency. Also
DTMF codec specifies that the pulse should be at least ms in duration and should be followed
by scielence for another ms duration.
Encoder Implementation
In DTMF each key is encoded using two sinusoids of different frequencies. So encoder can be
implemented in DSP as a look up table corresponding to each key pressed. This procedure
requires 16 lookup tables for a duration of at least 45ms (i.e. 360 samples long) double precision.
This makes its implementation costly. Instead of implementing encoder using lookup table we
can use digital sinusoid oscillators to generate the required frequency. Each oscillator requires
only two parameters.
A digital sinusoidal oscillator is a tw pole resonator for which the complex conjugate poles lie on
the unit circle. The system function is given by
(1)
where , , . When repreented as a difference equation we
get
(2)
The coefficients were calculated beforehand and stored in a table. Depending on the key pressed
and the duration of the signaling interval the oscillator is invoked with the required parameters.
Also the higher frequency component is made 2dB louder than the lower frequency component.
Decoder Implementation
The decoder has to make a decision by looking at the constituent frequency components. Also
only certain frequency components are required. So using fft on all the frames will be redundant.
So Goertzel algorithm which computes fft only for required frequencies is used. The length of
the frame to be considered is also critical because of the windowing effect. main lobe width and
the frame length are connected by . Also to distinguish between speech,
music and the keys pressed second harmonics were calculated. Speech and music will have
power even at the second harmonics so the relative power between the first and second
harmonics will allow us to distinguish between speech and actual keys.
Goertzel algorithm was used to find the strengt of the required frequency components.
Goertael algorithm operates on a frame of length 205. This corresponds to a main lobe of
width 39 Hz. This mainlobe is sufficiently narrow to reject the tones +/-2.3 percent of the
main frequency. Also 205 samples correspond to 26 ms . this makes taking care of the
time domain constraints easy. A tone is valid if it is present only for atleast two frames.
Overlap of the frames can be used to tune the effective duration.
Second harmonics are computed, so as to distinguish keys from speech and music which
have significant second harmonics present. These are computed only for the strongest
row and collum frequencies thus keeping the computation requirement low.
The maximum power each frame is calculated and if less tahn MIN_POW the frame is
rejected. This is to reject noise.
Also the relative powers between spectral row and column components are calculated.
The strongest component must stand out from its proximity tones.(THR_REL)
The ratio between the second harmonic and the first harmonic should be lower than
THR_2ND
Parameters Chosen
signalling time:
No signaling time :
Frame length :
MIN_POW :
THR_REL :
THR_2ND :
These parameters were chosen after decoding a known sequence with noise and speech.
Results
The encoded signals for the following numbers are as follows ( The program to be run
is telephone. This guy is made with mat lab 6.5)
831-4000
1-219-631-5480
610
555-3200
*71
The decoded numbers are
1.au 2264810
2.au 12196318308
3.au 12196315480
4.au 2342591
5.au 8621*#
6.au 005448757
7.au 110551212
8.au 8318314
9.au 197
10.au 8184
ENCODER
Main program with gui : telephone’s, Output is stored in 'dtmf.au' (If GUI does not work the
function is encoded=dtmf_encode([1 2 3 4],4,0.065 ,0.065) encodes 1234 and produces an
encoded stream encode . Also A B C D * # are mapped to 12 13 14 15 10 11
DECODER
Decoder : dtmf_decode('filename.au'); will decode the file.au
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