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Implementation of DTMF Encoder Final Document

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Implementation of DTMF Encoder, Decoder
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Page 1: Implementation of DTMF Encoder Final Document

Implementation of DTMF

Encoder, Decoder

Page 2: Implementation of DTMF Encoder Final Document

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.

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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).

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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

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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.

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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

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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

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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.

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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

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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

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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.

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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.

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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)

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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.

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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).

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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

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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

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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).

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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

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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

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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

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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.

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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

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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

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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.

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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.

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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.

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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.

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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.

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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

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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.

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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".

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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

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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.

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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,

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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

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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

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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

Page 39: Implementation of DTMF Encoder Final Document

Recommended