Fourier Analysis

 

Overview

Fourier analysis is a fascinating activity. It deals with the essential properties of periodic waveforms of all kinds, and it can be used to find signals lost in apparently overwhelming noise. As just one example of its usefulness, if SETI (the Search for Extraterrestrial Intelligence) should ever detect an alien signal, that discovery will be made using Fourier analysis.
It's important to say that, even though this article deals with simple waveforms, Fourier analysis is by no means limited to these classic examples — it can analyze and process images, it can efficiently compress images and video streams, and it can assist in visual pattern recognition, where a complex pattern may be efficiently and concisely described using a set of Fourier terms.
One of the principles of Fourier analysis is that any imaginable waveform can be constructed out of a carefully chosen set of sinewave components, assembled in a particular way (the frequency -> time task). And conversely, any complex periodic signal can be broken down into a series of sinewave components for analysis (the time -> frequency task). And most important, the described tasks are reciprocal operations — in the same way that integration and differentiation are reciprocal operations in Calculus, encoding and decoding signals are reciprocal operations in Fourier analysis.
Using that notion as a guide, this page has a section in which we will construct complex signals using sinewave components, and another in which we will decode complex signals into their components.

The Frequency -> Time Task

This section deals with the task of creating time-domain signals out of individual sinewave components in the frequency domain. First, download this Fourier example file and load it into Maxima. It contains a set of example functions that show aspects of Fourier analysis.
In this section we'll be using Maxima to build complex waveforms out of simple components. To reiterate, any signal can be built out of individual sinewave components, arranged in particular ways. Remember this the next time you're listening to your favorite music — in principle, it can be created out of a mathematical description consisting only of sinewave frequencies, amplitudes and phases.
Let's begin with a table of classic wave equations and corresponding graphs:

Name	Equation	Graph
Sine
Square
Triangle
Sawtooth
Rectified Sine

Obviously these graphs weren't created by summing terms from 1 to ∞ as indicated by the equations in the left column. In practice one chooses a summation number that balances display quality with a reasonable running time. But by choosing different upper bounds for these summations, we can learn something important about signal generation.
Make sure you have loaded this Fourier examples file into Maxima, then enter this at the keyboard:

fpsq(1);

This will accumulate one element from the square-wave equation, which, if you think about it, must be a sine wave (because all these generators use individual sine waves as their building blocks). Now change your entry to this:

fpsq(2);

Notice the difference. Now change the number, go as high as you like, considering the issue of running time, and see how the plotted wave changes. You will begin to see a pronounced "ringing" transient at the switching points in the waveform (like the graph on this page), and the ringing will become more severe as the number of accumulations increases. There is a longstanding debate among mathematicians as to whether this ringing is an artifact of our imperfect generation methods or a real property of square waves.
In case it isn't obvious from the equation listed above, the square wave is created by accumulating the odd multiples of a specified frequency — in this way (where x = 2 π f t):

y = (4/π) (sin(x) + 1/3 sin(3x) + 1/5 sin(5x) + 1/7 sin(7x) ...)

These sine terms are combined to create the time-varying waveform in the display. And if the opposite operation were to be performed on the result (something called a Fourier transform), these individual harmonic elements would reappear. It is important to reemphasize that waveform generation, and the Fourier transform, are reciprocal operations. You can use frequency components to generate a waveform in the time domain, then transform the result back to the frequency domain and recover what you started with. This reciprocal relationship is to Fourier analysis what the Fundamental Theorem of Calculus (the idea that integration and derivation are reciprocal operations) is to Calculus.
The Fourier examples file contains a set of waveform generating functions, one for each kind of waveform in the above table. Here are their names:

• fpsq(n); (square)
• fptri(n); (triangle)
• fpsaw(n); (sawtooth)
• fprec(n); (rectified sine)

Play with these functions — each will create a graph that results from n summations of the generating function, and you choose the value for n. The Fourier examples file also has functions to create multiple plots for each choice, with results that look like this:

Function	Graph
fsqmult(8);
fsawmult(8);

Here is a list of the multiple-plot functions:

• fsqmult(n); (square)
• ftrimult(n); (triangle)
• fsawmult(n); (sawtooth)
• frecmult(n); (rectified sine)

These functions take much more time to generate their plots (because they must perform a summation for each individual trace in the plot), so don't enter large numbers unless you are a very patient person.

The Time -> Frequency Task

Converting from the time domain to the frequency domain is a bit more labor-intensive than the task of generating complex waveforms, the topic covered above. The general name for this conversion is "Fourier Transform", and because of its usefulness, much thought and ingenuity have been expended on this task.
In the 1960s, an apparently new and much more efficient method for this conversion, now called the "Fast Fourier Transform" (FFT), was devised by J. W. Cooley and John Tukey. I say "apparently" because it turns out the basic idea behind the FFT was developed by Carl Friedrich Gauss in 1805 and was rediscovered by others in various limited forms in the intervening years.
Maxima has an FFT package available, made accessible by loading a library named "fft". Here is an outline of the steps needed to perform an FFT in Maxima:

Code	Explanation
load(fft);	Load the required library.
n:2048;	Establish an array size, which must be: A power of 2 (512,1024,2048, etc.). Twice the size of the highest frequency of interest in Hertz, e.g. to resolve frequencies between 0 and N Hertz, the array size must be the next higher power of 2 above N * 2.
array(ra,float,n-1);	Declare a real-value array.
array(ia,float,n-1);	Declare an imaginary-value array.
for i:0 thru n-1 do (	Fill the array with the desired data.
ra[i]:my_function(i),	This is a user-defined function that is able to create suitable data.
ia[i]:0.0	Must set the imaginary array to zero if there are no data.
);
fft(ra,ia);	Perform the FFT conversion in-place (the original array data are overwritten).
recttopolar(ra,ia);	Optional: convert complex a+bi form to polar magnitude and angle form.
lira:makelist(ra[i],i,0,(n/2));	For plotting, create a list out of the first half of the data array.
xIndexList:makelist(i,i,0,length(lira));	Create an index list required by the plotting routine.
wxplot2d([discrete, xIndexList,lira], [gnuplot_preamble, "unset key;set xrange [0:1024];set xlabel 'Frequency';set ylabel 'Amplitude';set title 'Amplitude Modulation — Frequency Domain';"]);	Create an inline plot (one that is integrated into wxMaxima's display).

There are example functions in the Fourier examples file that show a typical conversion:

Function	Graph	Comment
fpt();		This time-domain plot shows a waveform that was once very familiar to electrical engineers: an amplitude-modulated (AM) radio signal. In the AM scheme, information is added to a radio carrier wave by changing its amplitude. Speaking mathematically, this modulation method creates sidebands whose distance from the carrier frequency is equal to the modulation frequency. Typically the modulation of a real-world AM radio signal is complex and broad. In this example, we're modulating the carrier with a simple sinewave. This plot is the input data for the FFT. If you have loaded the Fourier examples file into Maxima, and if you enter the function name at the left, you will get this plot.
fpf();		Here is the frequency-domain equivalent of the above time-domain plot (with some random noise added for a touch of realism). The central peak is the carrier frequency, and there are two sidebands, one above and one below the carrier frequency. The sideband frequencies are located at c+m and c-m, where c = carrier frequency and m = modulation frequency. This plot is the output data from the FFT. If you have loaded the Fourier examples file into Maxima, and if you enter the function name at the left, you will get this plot.

There is one more experiment in the Fourier examples file. It is meant to clearly show the equivalence of time-domain and frequency-domain representations of spectral data. In this experiment, we will:

• Create a square wave signal in the time domain using the methods established above,
• Convert the time-domain data back to the frequency domain using the FFT,
• Compare the resulting harmonic components to the individual terms in the square wave generator equation.

Actually, this process is automated for you by two functions that are part of the the Fourier examples file. They are:

Function	Comment
fph();	Predicts (and prints a list of) frequencies and amplitudes for square-wave harmonics based on the mathematical identity: y = (4/π) (sin(x) + 1/3 sin(3x) + 1/5 sin(5x) + 1/7 sin(7x) ...)
fch();	Generates a square-wave data table in the time domain, converts it to the frequency domain using the FFT, then prints out the frequencies and amplitudes of the resulting harmonic components.

Make sure you have loaded the Fourier examples, then enter the two function names listed at the left above. One ("fph()") will predict the expected harmonic frequencies and amplitudes based on the mathematical identity of a square wave, the other ("fch()") will use the FFT to convert a time-domain square-wave data list into its harmonic components, then search the results and print out the harmonics it finds.
If everything is in order on your system and there is nothing wrong with your Maxima installation, the predicted and result data lists should be nearly identical. Because the FFT is a numerical procedure performed on machines with finite floating-pint resolution and a finite array size, all FFT results are approximate.
I want to emphasize that the numerical, approximate nature of the FFT doesn't imply that all Fourier transforms are approximate. The FFT is a specific, very fast, embodiment of a well-understood mathematical transformation that can be carried out analytically in some circumstances. What I am saying here is ... don't confuse the FFT with the field itself.

VHF/FM Radio Transistor

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VHF/FM Transistor Radios

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Because of the additional complexity of VHF/FM circuits, I would recommend that newcomers to transistor radio repair should repair a couple of MW/LW sets first. The information in this section assumes you have already read and understood the preceding section on AM (MW/LW) RF and IF circuits.

There are two main differences between MW/LW (AM) broadcasts and VHF (FM) broadcasts - both of which are indicated by the names. VHF radio signals are transmitted at a much higher frequency - around 100MHz for VHF compared to 1.6MHz (1600kHz) at the top end of MW. VHF radio uses Frequency Modulation (FM) whereas MW/LW uses Amplitude Modulation (AM). It therefore follows that the two main areas of the set that are changed are the mixer-oscillator and the detector.

This block diagram shows the makeup of a typical AM/FM receiver. The component functions on AM are shown along the bottom, and follow the arrangement we have already discussed. On VHF/FM, two additional transistors are included in the circuit, acting as an RF amplifier and mixer-oscillator. The transistor that was used as the mixer-oscillator on AM becomes the first IF amplifier on FM, giving a total of three IF stages on FM. The IF on VHF is a higher frequency - 10.7MHz compared to 470kHz on AM. Because of the different type of modulation, a different detector circuit is used on FM. The remaining audio and output stages are the same.

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VHF mixer-oscillator

The RF amplifier and mixer-oscillator are normally assembled in a screened can, due to the high frequencies involved. A typical (simplified) circuit is shown here.

The first transistor is the RF amplifier and is configured in common-base mode to achieve the highest gain. The RF amplifier serves two purposes. Firstly it amplifies the signal before it reaches the mixer-oscillator, as its name suggests. Secondly it provides some isolation between the mixer-oscillator and the aerial, preventing the oscillator frequency being radiated by the aerial and causing interference to other equipment.

The second transistor is the mixer-oscillator and is also working in common-base mode. Because the transistor is running near the limits of its frequency range, the phase shift between emitter and collector is about 90 degrees. Because of this, the oscillator tuned circuit only needs to provide another 90 degrees of phase shift to achieve oscillation.

L3 and L5 are the tuning control. Variable inductors are used on most early VHF sets, whereas later sets used variable capacitors and fixed inductors. L3 and its parallel capacitor adjust the tuning of the RF amplifier to suit the received signal. L5 and its capacitor set the frequency of the local oscillator to 10.7 MHz above the received signal. A damping diode may be connected across L3 to prevent the amplified RF signals from more powerful transmitters from overloading the mixer-oscillator stage.

The received and oscillator signals are mixed as described for the AM circuit and appear as a 10.7MHz If signal on the first IF transformer (L6/L7).

The VHF RF amplifier and mixer-oscillator assembly is often referred to as the "VHF front-end". The entire circuit is only powered when the set is switched to VHF, the power being switched by a section of the waveband switch. In this circuit, OC171 transistors are used in both positions. In many sets the RF amplifier will be an AF114 and the mixer-oscillator will be an AF115.

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VHF IF amplifier

This circuit shows the typical IF amplifier arrangement (Ekco/Pye/Invicta diagram). VT4 and VT5 are the AM IF amplifiers. The circuit is very similar to that used for AM only sets, however there are two IF transformers, one for 470kHz and the other for 10.7MHz, with their primaries and secondaries connected in series.

In order to use the AM mixer-oscillator (VT3) as an IF amplifier on VHF, the oscillator needs to be disabled. This is usually achieved by bypassing the emitter resistor with a capacitor (C21), so that the stage cannot work in common base mode, using contacts on the waveband switch (SW1F). The input to the stage is switched to the output of the VHF mixer-oscillator instead of the MW/LW aerial circuit (SW1E). In addition the VHF IF primary must be bypassed when the set is working on AM otherwise the oscillator will not run reliably. In this circuit this is done by SW2A (MW) and SW3A (LW).

On AM the IF bandwidth needs to be quite tightly peaked to give good selectivity and avoid interference from adjacent channels. However, since FM broadcasting relies on frequency variations, the IF performance for this band must be flat over the range of frequency variations expected for normal broadcasts (normally +/-75KHz from the nominal frequency) the gain of each IF amplifier stage will be lower. Three stages of IF amplification are therefore needed for FM, compared to two for AM.

On AM the gain of the first IF amplifier is controlled by AGC as usual. On FM, no AGC is applied since it does not matter if the signal is clipped.

The transistors used are all AF116 types. These have a better response at 10.7MHz than the AF117 types used in AM only sets. VHF sets cannot use the earlier OC45 transistors. These only operate reliably at 470kHz with some fiddling (neutralisation), and will not work at all at 10.7MHz. This is why transistor sets with the VHF band were not produced until a few years after MW/LW models appeared - manufacturers were waiting for suitable transistors and circuits to be developed.

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Frequency Modulation Detector

This diagram shows the difference between Amplitude Modulation (a) and Frequency Modulation (b).

Since the amplitude of an FM signal is constant, the single diode type detector used for AM would not work for FM. For FM detection we need to convert the frequency variations into a voltage variation. This is a little more complicated than AM detection.

There are a few different types of FM detector circuit. The most commonly used type is the "ratio detector" which is shown here.

Whereas the IF transformers between the IF amplifier stages needed to have a flat response across the frequency variation of the modulated signal, the final IF transformer has a tightly tuned peak at the unmodulated IF. Therefore when an unmodulated carrier is received (silence is broadcast), this is exactly at the peak or resonant frequency of the IF transformer. When the carrier is modulated by the transmitted audio, its frequency varies either side of the resonant frequency of the transformer.

The actual operation of the ratio detector depends on the phase relationships between voltage and current in a tuned circuit at and near resonance. To explain this fully would involve covering a significant amount of AC electrical theory, which is not appropriate here.

When an unmodulated carrier is received, the IF transformer is at resonance. At resonance the voltage and current are in phase, so the output power is at its maximum. The output is rectified by the diodes and C1 (typically 8uF) charges to the peak voltage. These act rather like an AM detector, but the value of C1 is such that rapid variations are filtered and only slow changes in level are responded to.

When the carrier is modulated, the frequency varies away from the resonant frequency of the IF transformer. Then the frequency is above resonance the phase of current lags (is behind) the voltage and below resonance the current phase leads (is before) the voltage. The amount of lead or lag depends on how far away from the resonance the signal is.

These current/voltage phase variations cause current variations through the diodes and C1. Since C1 is fairly large, the charge stored in it does not vary with the current/voltage variations caused by the modulation. The variations appear as a varying voltage on the centre tap of the secondary. This is combined with the voltage/current variations induced in winding "L" (an untuned winding close to the primary), to form the demodulated signal at the bottom of "L".

The remaining IF signal is removed from the audio by C2.

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De-emphasis

One problem with FM broadcasting is that noise and hiss tends to be more noticeable, especially when receiving weaker stations. To reduce this effect, the treble response of the audio signal is artificially boosted prior to transmission. This is known as pre-emphasis.

In the set a corresponding filter or "de-emphasis" circuit is required to reduce the treble response to the correct level. Since most noise and hiss tends to be at the higher frequencies, the de-emphasis removes a lot of this. Pre-emphasis and de-emphasis thus allow an improved signal-to-noise ration to be achieved while maintaining the frequency response of the original audio signal.

In the circuit above, R2 and C3 are the de-emphasis components.

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Practical ratio detector circuit

This diagram shows the FM ratio detector and AM diode detector from the same set as the IF stages above.

The basic arrangement is similar to that shown above, but with a couple of refinements. Because the performance of early germanium diodes at these frequencies varies, some means of balancing the two diodes is needed. On this set this is done by RV1. If the diodes were identical this would be set to 2.2k-ohms to match R25. In practice it is set to give minimum distortion once the IF alignment is completed. Three IF filter capacitors are included, one (C47) to ground as usual and the other two (C44 and C45) to either side of the ratio detector. This improves the balance and performance of the circuit.

It will be noted that there is no separate de-emphasis in this early FM set. A note in the service data states that the value of C47 was increased to 0.02uF in later models. Even then the combined parallel capacitance of C44, C45 and C47 with R23 at 100 ohms would not provide sufficient de-emphasis, so this set (which has no tone control) may sound rather bright on FM. There is some fixed tone correction in the output stage, and speaker may not have a particularly good high frequency response, so in practice it probably sounds OK.

SW1B selects whether the output of the AM detector or FM detector reaches the audio stages.

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Foster-Seeley detectors

The main advantage of the ratio detector described above is that it is largely immune to amplitude variations, due to the effect of C1. Since most interference tends to affect the amplitude of an RF signal rather than the frequency, such interference is rejected as an inherent part of the detector operation.

The disadvantage of this circuit is that quite accurate alignment is needed to give good results. If the alignment is slightly out, such that the resonant frequency of the transformer is not exactly the unmodulated IF, then distortion can result.

In a few sets an alternative detector circuit, known as the "Foster-Seeley discriminator" (shown here), is used. At first glance the circuit appears very similar to the ratio detector, except the two diodes point in the same direction and an additional inductor is used. This circuit also relies on voltage/current phase shifts in the transformer, however the alignment is less critical.

The drawback is that it is not immune to amplitude variations so an additional circuit, known as a limiter, is needed. A limiter is basically an IF amplifier driven into clipping, so the amplitude of the signal is fixed. The distortion caused to the IF waveform is irrelevant since it does not affect the frequency variations that carry the audio information we want.

The additional cost of a limiter stage cannot normally be justified in a portable transistor radio. However the Foster-Seeley circuit can give better performance and audio quality than the ratio detector, so it is sometimes used together with a limiter in high quality equipment where the additional cost can be justified.

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Slope detector

The most basic form of FM detector is the slope detector. The circuit is the same as an AM detector, and relies on the IF transformer being detuned.

This graph shows the frequency response of the final IF transformer. It is adjusted so that its resonant frequency (4) is higher than the IF frequency. The IF frequency (2) is therefore part way up the response slope, and the modulation variations move it closer to (3) and further away (1) from the resonant frequency. The output voltage from the transformer thus varies depending on the frequency input, and a simple single-diode AM detector is sufficient to resolve the audio.

This circuit has a couple of disadvantages. It has no immunity to amplitude variations, so a limiter would be needed to clamp the IF level. It also requires fairly careful alignment to get the frequency variations in an area where the slope is reasonably linear. In practice this circuit is rarely used in radio receivers. However it is important to be aware of the effect because you can have a set with a detector working this way, unintentionally.

If one diode or the electrolytic capacitor in a ratio detector circuit failed, the output would be distorted. If a previous repairer has tried adjusting the final IF transformer, this would cause an apparent improvement, and the repairer might think the problem was solved. However, rather than fixing the fault, the repairer has simply changed the operation to a slope detector. The set might work fairly well on strong signals, but on weaker signals it would be very prone to noise and interference because there is no limiter.

If an FM set is distorted, resist the temptation to twiddle with the IF transformers unless they have clearly already been disturbed. It is more likely that there is an electronic fault causing the distortion. By fiddling with the IF alignment you will make the diagnosis of the real fault more difficult because the set will sound worse when the fault is fixed, until the alignment is corrected.

Modulasi Amplitudo silahkan di download

http://www.total.or.id/info.php?kk=amplitude%20modulation

Phase MODULATION

PHASE MODULATION

A technique used in telecommunications transmission systems whereby the phase of a periodic carrier signal is changed in accordance with the characteristics of an information signal, called the modulating signal. Phase modulation (PM) is a form of angle modulation. For systems in which the modulating signal is digital, the term “phase-shift keying” (PSK) is usually employed. See also Angle modulation.
In typical applications of phase modulation or phase-shift keying, the carrier signal is a pure sine wave of constant amplitude, represented mathematically as Eq. (1),
1.
where the constant A is its amplitude, θ(t) = ωt is its phase, which increases linearly with time, and ω = 2πf and f are constants that represent the carrier signal's radian and linear frequency, respectively.
Phase modulation varies the phase of the carrier signal in direct relation to the modulating signal m(t), resulting in
2.
Eq. (2), where k is a constant of proportionality. The resulting transmitted signal s(t) is therefore given by Eq. (3).
3.
At the receiver, m(t) is reconstructed by measuring the variations in the phase of the received modulated carrier.
Phase modulation is intimately related to frequency modulation (FM) in that changing the phase of c(t) in accordance with m(t) is equivalent to changing the instantaneous frequency of c(t) in accordance with the time derivative of m(t). See also Frequency modulation.
Among the advantages of phase modulation are superior noise and interference rejection, enhanced immunity to signal fading, and reduced susceptibility to nonlinearities in the transmission and receiving systems. See also Distortion (electronic circuits); Electrical interference; Electrical noise.
When the modulating signal m(t) is digital, so that its amplitude assumes a discrete set of values, the phase of the carrier signal is “shifted” by m(t) at the points in time where m(t) changes its amplitude. The amount of the shift in phase is usually determined by the number of different possible amplitudes of m(t). In binary phase-shift keying (BPSK), where m(t) assumes only two amplitudes, the phase of the carrier differs by 180°. An example of a higher-order system is quadrature phase-shift keying (QPSK), in which four amplitudes of m(t) are represented by four different phases of the carrier signal, usually at 90° intervals. See also Modulation.

modulation (PM) is a form of modulation that represents information as variations in the instantaneous phase of a carrier wave.
Unlike its more popular counterpart, frequency modulation (FM), PM is not very widely used for radio transmissions. This is because it tends to require more complex receiving hardware and there can be ambiguity problems in determining whether, for example, the signal has changed phase by +180° or -180°. PM is used, however, in digital music synthesizers such as the Yamaha DX7, even though these instruments are usually referred to as "FM" synthesizers (both modulation types sound very similar, but PM is usually easier to implement in this area).

Theory

An example of phase modulation. The top diagram shows the modulating signal superimposed on the carrier wave. The bottom diagram shows the resulting phase-modulated signal.

PM changes the phase angle of the complex envelope in direct proportion to the message signal.
Suppose that the signal to be sent (called the modulating or message signal) is m(t) and the carrier onto which the signal is to be modulated is

Annotated:

carrier(time) = (carrier amplitude)*sin(carrier frequency*time + phase shift)

This makes the modulated signal

This shows how m(t) modulates the phase - the greater m(t) is at a point in time, the greater the phase shift of the modulated signal at that point. It can also be viewed as a change of the frequency of the carrier signal, and phase modulation can thus be considered a special case of FM in which the carrier frequency modulation is given by the time derivative of the phase modulation.
The spectral behaviour of phase modulation is difficult to derive, but the mathematics reveals that there are two regions of particular interest:

• For small amplitude signals, PM is similar to amplitude modulation (AM) and exhibits its unfortunate doubling of baseband bandwidth and poor efficiency.
• For a single large sinusoidal signal, PM is similar to FM, and its bandwidth is approximately

,

where f_M = ω_m / 2π and h is the modulation index defined below. This is also known as Carson's Rule for PM.

Modulation index

As with other modulation indices, this quantity indicates by how much the modulated variable varies around its unmodulated level. It relates to the variations in the phase of the carrier signal:

,

where Δθ is the peak phase deviation. Compare to the modulation index for frequency modulation.

Senin, 27 September 2010

modulasi

Modulasi adalah proses perubahan (varying) suatu gelombang periodik sehingga menjadikan suatu sinyal mampu membawa suatu informasi. Dengan proses modulasi, suatu informasi (biasanya berfrekeunsi rendah) bisa dimasukkan ke dalam suatu gelombang pembawa, biasanya berupa gelombang sinus berfrekuensi tinggi. Terdapat tiga parameter kunci pada suatu gelombang sinusiuodal yaitu : amplitudo, fase dan frekuensi. Ketiga parameter tersebut dapat dimodifikasi sesuai dengan sinyal informasi (berfrekuensi rendah) untuk membentuk sinyal yang termodulasi.
Peralatan untuk melaksanakan proses modulasi disebut modulator, sedangkan peralatan untuk memperoleh informasi informasi awal (kebalikan dari dari proses modulasi) disebut demodulator dan peralatan yang melaksanakan kedua proses tersebut disebut modem.
Informasi yang dikirim bisa berupa data analog maupun digital sehingga terdapat dua jenis modulasi yaitu

• modulasi analaog
• modulasi digital

Modulasi Analog

Dalam modulasi analog, proses modulasi merupakan respon atas informasi sinyal analog.
Teknik umum yang dipakai dalam modulasi analog :

• Modulasi berdasarkan sudut
  • Modulasi Fase (Phase Modulation - PM)
  • Modulasi Frekuensi (Frequency Modulatio - FM)
• Modulasi Amplitudo (Amplitudo Modulation - AM)
  • Double-sideband modulation with unsuppressed carrier (used on the radio AM band)
  • Double-sideband suppressed-carrier transmission (DSB-SC)
  • Double-sideband reduced carrier transmission (DSB-RC)
  • Single-sideband modulation (SSB, or SSB-AM), very similar to single-sideband suppressed carrier modulation (SSB-SC)
  • Vestigial-sideband modulation (VSB, or VSB-AM)
  • Quadrature amplitude modulation (QAM)

Modulasi Digital

Dalam modulasi digital, suatu sinyal analog di-modulasi berdasarkan aliran data digital.
Perubahan sinyal pembawa dipilih dari jumlah terbatas simbol alternatif. Teknik yang umum dipakai adalah :

• Phase Shift Keying (PSK), digunakan suatu jumlah terbatas berdasarkan fase.
• Frekeunsi Shift Keying (FSK), digunakan suatu jumlah terbatas berdasarkan frekuensi.
• Amplitudo Shift Keying (ASK), digunakan suatu jumlah terbatas amplitudo.