The most basic component of radio wave transmission is the oscillator. An oscillator is an amplifier circuit with a controlled feed back path. It can be said that an oscillator is a regenerative device though it will need a power source to continue oscillating due to power losses in ineffecient Xsistor and tube devices as well as resistors and other lossy components.
The oscillator has a resonant portion of it's circuit which is used to determine the frequency of the oscillator. These components can be what is known as an LC tank circuit or a crystal resonator which has properties of both a capacitor and inductor and a vary small resistance We will cover that topic later.

Once an oscillator begins to oscillate or feeds back on its self than it has become a transmitter. That's right it can be detected by a radio receiver. An oscillator will fit into the catagory of what is know as a CW (continuous wave) device. It Xmits a single frequency while it is on.
But what constitutes a Xmitter? I mean in actual practice you want to send a signal further than your neighbors house. Simple, a series of amplifiers in a string each multiplying the power from the previous stage.

There are some rules to adhere to in developing these amplifiers. First off, remember the oscillator? The oscillator is the single source of frequency generation, therefore it must be handled with care. The LC tank circuit discussed previously is subject to all sorts of conditions that can alter it's frequency. This is not so substantial in a Xtal oscillator but still exist none the less.
So how do we protect an oscillator from drifting or being pulled off frequency? We use buffer stages. Buffer stages are extremely important in radio design. What is a buffer? The best buffer amplifiers have less than unity voltage gain. Ah huh? We are talking about an emitter follower or common collector amplifier.
Why are they the best? Because they have a very high impeadance input therefore not loading the oscillator to heavily. The input impeadance is basically R1/R2. It also has a low output impeadance of roughly RE which is a good match to a CE amplifier which has a high voltage gain. Another advantage of an emmiter follower amplifier is that it is a current amplifier and therefore is a good current source. Now if you are familiar wth a darlington pair which is easily made using two Xsistors, we find that the current is beta X beta of the two Xsistors. The net result is you will obtain good isolation between the oscillator and amplifier stages. This also helps to keep the oscillator from feeling phase changes from the output stages which can most definately effect the oscillator frequency.
Below are some examples of buffer amplifiers

Let's step back a minute and discuss the differences between an LC tank circuit and a crystal which will be abbreviate Xtal. Assuming we understand that a C (capacitor) stores an electro-static charge and an L (inductor) stores an electromagnetic charge, without going into too much detail we find that these opposing fields transfer energy back and forth at a given rate detemined by the value of the components. These LC components can be notorious for drifting off frequency with temperature being the primary culprit. Now an Xtal is less subject to these conditions and has a much higher Q (quality factor). The Q is determined by X/R or the amount of reactance to resistance. I'm not going to delve into the subject of reactance because I want to discuss radio in general. You can look up the subject of reactance.

XL= 2 ||  f L                    Inductive reactance

XC=  ___________      Capacitive reactance
              2 ||  f C

fo =    ____________      Resonant frequency
            __    ______
          2 ||  \/  LC

Now to complete what is known as a CW Xmitter you will find that these buffer stages must be added at various parts of the amplifier string.
OK CW is fine but there is no intelligence on the signal.Well if the Cw transmitter is keyed on and off by applying power to the amplifier stages than we have what is known as Morse code Xmission. We will need a special type of receiver to interpret the CW transmissions though. This receiver will have a BFO (beat frequency oscillator) which applies a substitute carrier to beat against the in coming carrier and produce a beat note. We will discuss receivers later. But just so you know, a carrier is the CW signal being transmitted from the transmitter. It is the frequency or unmodulated RF signal.

Ah modulation... Modulation can come in various forms but it usually canotes vioce communication or intelligence. How do we produce modulation? I'm not going to go into high level and low level modulation. The most common method of producing modulation takes us back to the all important oscillator. We can amplitude modulate or frequency modulate the oscillator. We do this by applying an audio signal such as the audio that goes to the speakers of your stereo etc. Now of course we usually input modulation from a microphone. Then we amplify the signal from the microphone to the oscillator.
It is the way that the modulation is applied that distinguishes the type of modulation being used, either AM (amplitude modulation) or FM (frequency modulation). If we apply the amplified audio signal to say the bias of the oscillator circuit or the bias of a successive amplifier, the audio will be super-imposed onto the output by way of amplitude. In other words a loud audio voltage level will cause the amplifier to increase its output respectively.
In FM we input the same audio signal but, instead of changing the amplifier output level we will vary the tank circuit frequency.
This can done by various methods the most common being the use of a varactor diode. Because varactor diodes are used in virtually all PLL circuits it has become the dominate method today. Now some will argue that using a varactor to change the frequency is not true FM and is actually PM (phase modulation) and they will say that true FM sounds better. Well that maybe true but they are detected the same way in a receiver and in narrow band communication services designated by FCC to be 5KHz band width or less, you cannot tell the difference.

There are many techniques used to improve the quality of an FM/PM signal. One such method is know as pre-emphasis/de-emphasis. This has to do with the fact that a varactor tends to be more responsive to higher frequencies of the audio signal. This can make the audio at the receiver sound kind of tinny like talking through a tin can. A simple low pass filter at the Xmitter end and a high pass at the receiver, puts it all back in unison.
Another method commonly used method to increase the modulation level is known as a "compander" (compressor/expander) system. This is sort of an AGC (automatic gain control) circuit that prevents the audio from exceeding a certain band width by controlling the audio level. The receiver than expands the audio back to its original level giving it a fuller sound.

bare with me's a way from being done and just tryin to clear up a few things here for the group.
?  ?

edit: added some spacing to make it easier to read (nice post :) )
Posted on 2006-08-01 22:03:37 by mrgone
  I will have two break this up into sections because I can't keep redoing formulas and indentations etc.
    Even though there are really only two types of modulation, AM & FM, they can be broken down into sub catagories. Some of these catagories would be: Spread spectrum, Single Side Band (SSB), double side band and various sub carrier techniques.
    Since we have recently discussed AM I would like to continue with a form of AM known as single sideband. Before we can grasp the subject of SSB we must understand the basic components of an AM signal.

As you you now know that a carrier is simply the oscillator frequency being Xmitted through the amplifier string and is the basic rf signal with no modulation. So what happens when we apply audio AM modulation to the oscillator carrier. Well the basic components of an AM transmission are:
1. The carrier
2. An upper side band signal and
3. A lower side band.
The upper and lower side bands ride on the carrier and are the upper and lower frequencies outside the carrier. Typically a carrier will be approx. 5KHz wide with the upper sideband being another 5KHZ and the lower sideband being another 5KHz bringing the total bandwidth of an AM signal to 15 KHz.
These signals can actually be observed on a spectrum analyzer as a large peak in the middle with two smaller peaks on either side of the carrier. When 100% modulation is used than 50% of the energy will be in the carrier and the other 50% will be in the two sidebands. So on a spectrum analyzer the two sidebands will be half the amplitude of the carrier.

An interesting thing to note here is that the carrier its self still has no intelligence. The intellegence is in the upper and lower sidebands. One day a very smart ham radio operator said to himself, "Ya know, the same intelligence that is in the upper sideband is also contained in the lower sideband." What if we could shave off the upper or lower sideband and send only one sideband. (Today this is called SSB reduced carrier)

But let's not get distracted here. Well if we can eliminate one of the sidebands and the carrier which contains no intelligence, than why not get rid of it too. So we will now send only the upper or the lower sideband which contains all of the intelligence of the original audio signal. Presto! SSB.

What is the advantage of SSB? Well obviously we have reduced band width which allows room in the radio spectrum for more communication services. If the AM signal was 15KHz than it is now 5 KHz or less.
Another very interesting point to ponder is that it is extremely power efficient. Why is that you say? Because the carreir is always putting out RF energy, but the sidebands only exist when there is modulation. If the radio announcer stops speaking, than there are no sidebands. Only a dead carrier. So if we are sending only one sideband out of our transmitter, there will be no power out until voice modulation is applied.
This can be observed on a simple power output meter. You will actually see the needle bounce up and down according to the level of modulation. If the operator stops speaking than there will be zero output from the Xmitter. That's efficiency!

edit: added some spacing to make it easier to read
Posted on 2006-08-02 13:31:39 by mrgone
  Pat of the reason I wanted to do this tutorial is so we can group some of this info that  is spread around into one location. So I'm sure you have seen some of this recently.

    Antenna Basics:

Granted a "long wire" which is a term used for random length of wire will receive all freqs., an antenna will be resonant at some frequency depending on it's length. This is not crucial in receiving a radio signal but is crucial in a transmitter application. The antenna must be cut to resonance or you will have very high SWR (Standing Wave Ratio) which simply put you will have a bad impeadance match and the antenna will reject the RF frequency energy and will send it back to the transmitter amplifier which can quite easily burn up the final amplifier. A resonant antenna will also improve reception at a particular band of frequencies.
  Radio waves & light travel at a speed of 300 million meters per second. From this you can calculate the length of the antenna by dividing the frequency in megahertz in to 300 to abtain the length of a full wave antenna. Example 300/7 = approx. 41 which is why the ham radio guys refer to 7MHz as the 40 meter band. The transmissin line also refered to as coax cable "if using an industry standard cable" is always usually 50 ohms. It's been found that a half wave antenna is a very good resonator and if the antenna is fed at the center, this will be it's lowest impeadance point giving a close match to the 50 ohm transmission line. The voltage swing of the rf signal is lowest at center.
    A cheap and dirty method of calculating a half wave antenna length also known as a DIPOLE antenna, is to divide the frequeny in megahertz into the magic number of 468 to get the length in feet. The 468 number is the 1/2 wave length inside a wire. The actual 1/2 wave length in free space is obtained by the number 492 for foot calculation. A frequency wave length is actually shorter inside a wire.
    All of these concepts can be applied to HF,VHF & UHF antennas. Many times at these frequencies we are concerned with even more filtration as well as amplification in the antenna its self. These type antennas are called beam antennas. A beam antenna consists of a driven element which is the dipole that we have described and is the element which is fed directly by the Xmission line. It is always a half wave length though often times you might see loaded coils on the ends or near the ends which effectively simulate a full half wave antenna even though physically it is much shorter than a half wave length. An inductor will appear elecrically to add to the length, and a capacitor will effectively shorten the length. The capacitors are a little different than a conventional capacitor. They are usually air dielectric and can be easily made. I have taken beer cans and smashed them into a round flat disc. Then drill holes in the center of the discs and slide them over the ends of the wire and seperate them with some tape so they don't move. The loading coils can be some heavy gauge mabe no 12 or thiscker wire wound around a PVC pipe or a toilet paper roll and attached to the ends of the wire.
  Now why is it necessary to add loading coils you ask? Well if your antenna is 40 meters long it may be a little difficult to rotate it. Thi is the point of a beam antenna. A simple beam antenna wil be 2 or 3 elements. We already discussed the driven element. You will also need a reflector. The reflector is placed behind the driven element perpendicularly and is slightly larger the the driven element. It can be at some division dimention of distance behind. There are many theories to this but the key factor is that the energy either Xmitted or Rcvd will be reflected back toward the driven element. Now we have signal adding or aiding each other. Any other elements will go in front of the driven element and they are refered to as directors. So if we add more directors we will sharpen the radiation lobes coming from the antenna.
  Now at SHF and above, everyone is famaliar with the common dish antenna. The dish is designed as a mathematica portion of a parabola. That is why they are known as a parabolic dish. We find that at a gven distance fron the center of the partial parabola, we have an optimal focal point. That means that the energy bouncing off the inside of the dish will all converge at a single point. This in a sense is a form of amplification although a dish antenna usually has an active amplifier in the focal point. This is know as an LNA or Low Noise Amplifier. It is designed to have an extremely high signal-to-noise ratio to pump very week signals from outer space into the transmission line.

Posted on 2006-08-06 21:42:33 by mrgone
好bsp; 好bsp;Again you have seen some of this before:

好bsp; 好bsp; 好bsp; Receiver Basics:

好bsp; 好bsp;The way that virtually all radios today select a particular frequency is a process know as Superhetrodyne. What superhet is, is the mixing of two or more signals. So your radio and for simplicity purposes we will suppose a single conversion superhet and the stages will be like this. 1. You will usually have a front end filter for band selection. This is used to attenuate undesired frequency energy, kind of a focusing unit. It is made of capacitors & inductors. 2. You will usually have a small signal rf amplifier. 3. You will have an oscillator usually variable by VFO or PLL and this stage is known as the 1st Local Oscillator. 4. The mixer where you will feed the filtered/amplified signal & the variable oscillator signal which is your tuning mechanism. The output of the mixer will be (for the most part and is a complicate subject) the sum and difference frequencies of the two input signals and the original two signals. These four signals will have the highest signal strength.5. The crystal filter. Well we are only interested in one and it may be your favorite station you listen to. Here's how we obtain that frequency. OK if the desired signal is say 98.6 FM and my oscillator frequency is being read by a frequency counter and it is at 115.6 MHz the difference between these two frequencies is 17 MHz. I'm kinda working backwards here but this example shows you that we have discovered that the crystal filter is at 17 MHz. Crystal filters are used because they have a very high Q resonance and they are very selective on the order of 5 to 10 KHz. That means when we pass this mess of mixed signals from the mixer to the crystal filter it will cut out all the undesirable mixer products leaving you with only the desire frequency. The important thing to note here is that your tuning oscillator is not at the desire frequency you wish to receive but it will be offset by either the sum or difference of the crystal filter frequency. Also notice that the desired frequency has been converted to 17 MHz. This will be true for all frequencies as you tune through the dial. They will all end up being 17 Mhz, this is the job of the mixer. I tried to make this simple and don't think I did too good of a job but yes there are other more modern methods of filtering in place of a crystal filter known as DSP (digital signal processing) but is is simply a digital version of the same filtering stage.
好bsp; 好bsp;After the filter you will detect the signal which means to demodulate depending on the type of modulation sent by the transmitter. Either AM or FM. AM detection is nothing more than a diode and FM detector use eithe a slope detector or a discriminator circuit. From there you send it though your audio amplifiers and to the speaker.
好bsp; 好bsp; That's the basic stages for selecting a frequency but the overall system can get quite complicated. Look up dynamic range of a mixer and you will see what I mean. Normally you will require post mixer amplifier and IF amplifiers etc. The IF or intermediate frequency is the signal coming out of the crystal filter. It is known as first IF. If you are doing multiple conversions than you will have 2nd IF and 3rd etc.
好bsp; 好bsp; Now interestingly enough, if we refer back to our discussion about single sideband, we find that the same proceses used to receive a narrow band of frequencies is also the processes we use to generate a single sideband signal. Incidentally the section associated with the Xtal filter is known as receiver "Selectivety".
好bsp; 好bsp; Let's walk through the generation of a single sideband signal. Then we will put together the receiver processes for receiving SSB.
好bsp; 好bsp; If we have an oscillator and we have another stage called a speech amplifier and we mix the two together we will have the same sum,difference and original two frequencies except we use a variation of a mixer known as a balanced modulator. What this does is to help balance out the oscillator frequency by shifting the phase of the (now called carrier Oscillator) 180 degrees. The output of the balanced modulator is known as double sideband signal. So it is好bsp; a typical AM signal with the two sidebands but the carrier has been removed. Typical audio or speech will fall into a frequency range of 1.5 KHz to approx. 15 KHz. So by mixing the audio with the carrier in the ballance modulator we end up with the sum & difference of the carrier frequency, because remember the carrier its self has been removed through the phase cancellation process in the ballanced modulator. At this point we will send the output through a Xtal filter. The Xtal filter will be set to a mid-range audio frequency above or below the carrier oscillator frequency. Lets assume the audio frequency range we are after is 6 KHZ with a band pass of 3KHz and the carrier osc. is running at 10 MHz. If we wish only upper sideband to pass through the filter than we will have an Xtal resonant frequency of 10MHz + 6KHz = 10.006MHz. You will subtract for lower sideband. Usually there are two Xtals made switchable so that we can choose either upper or lower sideband. This signal is now mixed with your Xmitter tuner or a local oscillator that is made variable by some means and run through a high or low pass filter, then on to the linear amplifiers and out to the antenna. Notice I said linear amplifiers? We will discuss that later. For now it is sufficient to know that an SSB signal requires class A linier amplification.
?  ? Now why did we go through all this stuff about SSB when we already covered most of it earlier and what does this have to do with receivers? Well first of all we discussed the efficiency of SSB. A typical SSB transmitter will have a 9 DB gain over an AM transmitter with the same peak envelope power. So we want to make sure that we know how to receive this very important type of transmission.
?  ? So let's now go back to our receiver. If we stick with our most basic single conversion receiver, we remember there is a local oscillator and the RF signal coming in from the antenna being heterodyned in the mixer. Well remember the SSB Xmission has no carrier so we must insert it back in, inside the receiver. So one good place to do this would be in the mixer. We will choose a frequency of 1.5 KHz or higher (audio frequency) than the frequency of the Xtal filter for upper sideband and 1.5KHz or lower than the Xtal filter for lower sideband. With a sharp bandpass Xtal filter of say of say 3 MHz we will receive only the desired sideband which is now intelligeable do to the inserted carrier. The carrier oscillator is known as the BFO (Beat Frequency Oscillator).?  A BFO is also required for CW reception because if you remember, CW (Continuous Wave) is only a carrier so we need to ave the BFO to create a heterodyne beat note that we can hear. By varying either the 1st local oscillator or the BFO we can change the beat note for both CW and SSB. If you zero beat the BFO to the carrier frequency than you will hear neither.
Posted on 2006-08-07 00:35:54 by mrgone
好bsp; 好bsp; 好bsp; 好bsp; 好bsp; 好bsp; 好bsp; 好bsp; 好bsp; 好bsp; 好bsp; 好bsp; A little more on oscillators

好bsp; 好bsp; If anyone wants to join in here you are welcome. Nan had a page with a whole host of oscillator configurations. The only one not listed was the Hartley. The Hartley oscillator happens to be my favorite type of osc. for a number of reasons. Below is a Hartley with darlington buffer stage. The Hartley is a fairly stable oscillator as far as free running oscilators go. It's configuation is that of a source follower or common drain as is the case using a JFET in this illustration. This is good because we don't want an osc to run hot. It has a very high impeadance which also helps it to run low power. With good PC board layout this Hartley under test conditions will take up to a half hour to stabalize and of course it must have a solid enclosure to keep air currents from changing its temperature. After a half and hour, assuming we are using a well regulated power supply, we should see almost no drift in frequency for days on in. This also assumes a stable room temperature. The Hartley is easily recognized by its tapped coil. It is usually tapped about 1/3 turns from the ground side of the coil. The coil being used is a powdered iron toroid. These have excellent self sheilding characteristics. I recommend Amidon T-37-02 and similar Amidon products. The windings can be no. 30 gauge enameled copper wire. When connecting the tap on the coil you must make sure you scrape off the enamel at that part of the wire so that you can solder to it. Sand paper works very well for this.
好bsp; 好bsp; The buffer puts a light load on the oscillator and gives it isolation from succesive stages.
好bsp; 好bsp;So we now know that some of the enemies of a good stable oscillator are, temperature both external or self generated if oscillator runs too hot and stray capacitance which can be greatly reduced with good PC board layout techniques. We want to keep all leads as short as possible. Some of the other enemies of the oscillator stability are excessive loading from succeeding stages and standing waves which cause the impeadances from successive stages to reflect back to the oscillator. The latter will effect the frequency because it changes the LC tank circuit values which are the frequency determining components.
好bsp; 好bsp;Below the Hartley is a PLL (Phase Lock Loop) digital frequency synthesizer. The PLL is commonly used today for various reasons but the most important being it's frequency stability. Let's discuss the components of a synthesizer. First we have the VCO (voltage controlled oscillator). We also have a phase detector and a loop filter and a reference oscillator which is your basic loop but if you want to be able to change frequeny you will also need a couple of dividers (digital counters like four bit binary or hex). This is how it works and for simplicity sake we will start with the basic loop with no ablility to alter the frequency. Starting with the VCO if we look at the PLL circuit below we see another hartley oscillator only there is one difference. Unstead of having a tuning capacitor (or variable capacitor) we have a varactor diode in it's place. That's it! It's that simple, a VCO is just an oscillator with a varactor diode unstead of a hand variable capacitor. So what we do is, a varactor is inherently used in reverse bias and the way it functions is as you apply more voltage to the cathode side of the varactor it causes the P/N junction to physically spread due to the expansion of the depletion layer. So you can think of the P and the N materials as being the two plates of your variable capacitor. You will couple the capacitance of the varactor through a series capacitor for both DC isolation and frequency control. Remeber that series capacitors divide, if we want a broader range we will use a larger value capacitor. The VCO will be the rf source that will feed the whole radio or its tuning device. While we're discussing capacitors dividing let's consider a breif topic of controversy. Most of the industry seems to prefer a Colpits type oscillator because of claims that they produce less phase noise. If you look at a Colpitts oscillator configuration the thing that distinguishes a colpitts from other oscillators is the fact that its tuning section uses split stator capacitors which in effect is two series capacitors. These remember will divide so in effect if one of the capacitors is a varactor diode you will have less tuning range than the PLL circuit below which has only one tuning cap/varactor. Now we have a reference oscillator. It is simply a crystal controlled oscillator. The circuit below doesn't show the Xtal oscillator because what is done here is the PIC internal timer is being used to generate that signal, for now just consider the we do have a Xtal reference oscillator. So if the VCO is set up so that it's free running frequency is say 4MHz (remember it will drift with temperature and other changes) and we have a 4MHz Xtal oscillator, then we have one nifty drifty oscillator and one oscillator that is highly stable. So how do we make that VCO as stable as the crystal oscillator? Ah! Let's put that feed back loop together. If we take the output of the VCO and the output of the reference oscillator and run them both into a phase detector and for simplicity sake, an "exclusive OR" gate makes a very nice phase detector. Then we take the output of the exclusive OR we have what appears to be random swithing on & off at its output. Well obviously we do not want to feed this radical signal directly to the VCO because we are only concerned with the average DC level of this output. So we run it through a loop filter. A loop filter is a DC amplifier that has no coupling or bypass capacitors and an RC low pass filter. In other words we hard wire the output of the XOR gate or phase detector directly to the loop filter amplifier and the output of the loop filter is also hard wired directly to the VCO. Guess where that contact point is? Yes right to the cathode side of the varactor diode. We now have a self controlled frequency controlling feed back loop. If the loop is tuned correctly you will find that the VCO will be locked right at the 4MHz that the reference oscillator is at. Not only that but the Q of the VCO output will also be vary sharp. A high Q output would appear on a spectrum analyzer as a very narrow peak. In other words the bandwidth is very narrow. Basically the only phase noise from a circuit running at the same frequency as the reference oscillator would be only that which is inherent to the VCO oscillator components and stray capacitaces etc. The Xtal referece will actually greatly reduce these natural phase noise producers. There is another type of phase noise which is developed as a result of the divider circuits which we will discuss next.
好bsp; 好bsp;OK, how do we change frequency then? What we do is to insert a counter in between the VCO and the phase detector and another counter in between the reference oscillator and the phase detector. For example if we have a VCO with a free running frequency of approimately 10 MHz and a reference oscillator running at 5 MHz. We can use a digital counter to divide the VCO frequency down to say 10 KHz by dividing it by 1000. Then if we divide the reference by 500 we also end up with a frequency of 10 KHz. We now feed these two frequencies to the phase detector. The loop will now hold the VCO frequency at precisely 10 MHz plus or minus any tolerence in the Xtal of the reference oscillator. Because we have chosen to divide the two frequencied (reference & VCO) down to 10 KHz, this will be our frequency resolution of the PLL or the step ratio. For example if instead of dividing the VCO frequency by 1000, we divide it by 999 we get a frequency at the phase detector of 10.010 KHz (10 Mhz / 999 = 10.010 KHz). Now keep in mind the original divide ratio of 1000. So if we multiply 10.010 KHz by 1000 we get a new frequency of 10.010 MHz output from the VCO. Notice the frequency has moved 10 KHz which is our comparison frequency or step ratio more commonly refered to as the resolution. As a result of this resolution frequency we will also find that we have what are know as phase sidebands. These phase sidebands are a direct result of the resolution. This means that the frequency is adjusted every 10 KHz by the PLL loop. If we look at a spectrum analyzer we can see these phase sidebands. You will see one large peak in the middle which is the VCO center frequency, and two smaller peaks either side which are 10 Khz above center and 10 KHz below center. You will also see even smaller peaks at 20 KHz above and below and 30 and so on all droping off in amplitude the farther away they are from center frequency. The ones starting at 20 Khz & 30 etc. are harmonics of the orginal phase sidebands. If we reduce the original two phase sidebands through use of our loop filter you will see a proportional decrease in the phase sideband harmonics. These original phase sidebands are necessary because it is the energy that holds our PLL on frequency. Now if we are transmitting a 1000 watt signal then they are undesireable and are considered spurious emissions. So we need to reduce them down to an exceptable level. The FCC would like to see an attenuation of the phase sidebands of 60 DB down which is "full quite". Full quite is just a term used for 60 DB down, but in effect if you are transmitting at 1000 watts then the phase sidebands will be heard a couple of miles.
好bsp; 好bsp;If we look at the PLL schematic we see R5,C4 & R4. This is the loop filter. I realize I made a mistake in the schematic. This network should be connected to the collector of the DC amp and not the emitter. Anyway there are a couple of important issues to discuss about the loop filter before we move on to a full circuit description of the PLL schematic. There are many very highly complex mathematical equasions for you to calculate the values of the loop filter components. I find it is much easier to just work with it. First of all we know that the set parameters by the FCC are full quite or 60 DB down of the first two sidebands from the center frequency. There's one parameter. If you don't have a spectrum analyzer it is possible to do audio measurements with a receiver that has BFO. You can tune up & down through center frequency and move up 10KHz and check the audio level in comparison to the center frequency. What you don't want to do is over filter the signal or you will dramatically reduce the overall tuning range of the PLL. So it comes down to the usual engineering practices that state : "it will be the best possible compromize".
好bsp; 好bsp;Let's discuss the PLL circuit below. Keep in mind the mistake where the DC Amp collector is the actual connection point to the loop filter. The chip "U1" is a Motorola MC145170. Motorola no longer manufactures these special purpose IC and sol all rights to Lansdale. So you can obtain the data sheet from . Notice it has internal "ref in" and "ref out". This would be the reference oscillator but in order to reduce the number of oscillators on board the circuit is using the PIC's crystal through the timer. The program is alerted when the internal TMR0 has counted down then at the time the interrupt is triggered it flips the output port RA0 high & low. We feed this signal into "osc in" and this is used as the reference. This is an interesting approach because it has been found that we can achieve fairly fine incremental tuning by jam loading the timer with new values. On the chip we also see "f in". This is the VCO input. Now the chip has two internal counters I think one is a 14 stage and the other a 20 stage counter. Notice the chip has "PD out". This is an internal phase detector that is a bit more sofisticated that an XOR gate. It has J-K flip flops that also help steer the frequency. In other words, a phase/frequency detector. The other inputs are for serial data, notice the enable input. It also has a patented bit grabber technique which work very well but you will need the data sheets to understand it. I think the rest is self explanatory. No need to go back through the functional fow of a PLL again.
好bsp; 好bsp;There is another type of frequency control that has gained in popularity do to improved IC manufacturing techniques. It is called DDS or "Direct Digital Synthesis". DDS is simply a changing D to A converter where the output or the sine-wave signal is actually digitally produced. It also has an nherent phase noise known as aliasing. You will have to filter this out as much as possible.
Posted on 2006-08-10 21:47:12 by mrgone
Broad Band amplifiers & RF Power AMPS

好bsp; 好bsp;Broad band small signal amplifiers are comonly used today due to their ability to pass a wide range of frequencies. Some of their uses might be to amplify the local oscillator signal before it is applied to the mixer stage or in multi-band operation where we are Xmitting or receiving on different freqs. etc Broad band small signal amplifiers can be done a few different ways. Some poorer quality amplifiers used in radio might have a resistor in the collector of a CE amplifier. This is really what qualifies as a "wide band amplifier" and can have the tendancy to be lossy as well as performing undesireable phase shifting. It is better practice to use inductors in the collector. The only real difference between a broad band amp and a typical rf amp is one is tuned and one is not. Bare in mind there are two types of tuning that can be done and one of them is more critical in transmitter application where it is desireable to increase the rf signal strength. The tuned amp that is not broad band will have an LC tank circuit in the collector where a broadband will have no capacitor but only an inductor. Granted inductors can be had off the shelf in various different values but they are not optimal for quality design. One of the primary reasons is they tend to throw off a large field which can interfere or even feedback into other rf stages and will require quite a bit of sheilding. One way to dramatically reduce the amount of sheilding needed is through the use of toroid or dohnut shaped ciol forms known as toroidial cores. Due to the shape of these toroids, they tend to have self sheilding characteristics. In broad band application they are usually made of ferrite. Ferrite is a mixure of graphite and powdered iron and other secret ingrediants that manufacturers use for different frequency ranges. For instance, one of the best types for the hf (high frequency band/3 to 30 MHz) is Amidon's no. 43 mix ferrite. A good general purpose ferrite toroid for this band would be FT-50-43. The FT of course stands for "ferrite toroid" and the (43) is for the mix and (50) is the size. Amidon uses an "AL" value figure for calculating the inductance per turns of enameled wire. The formula is usually right on the pack when you buy from Amidon.

A word about Linear Power Amplifiers

好bsp; 好bsp;Some of you may be waiting with bated breath on this very important topic of RF Amplifiers in general and then again maybe If you are I would like to appologize because I really need to do some good diagrams here and have'nt had time to draw them up.
好bsp; 好bsp;This seems like a good place to continue the discussion of why and how to amplify a Single Sideband signal. Remeber that we discussed it needing to be amplified by linear amplification only. We cannot afford any distortion of an SSB signal. More times than not, when you obtain data sheets on power MOSFETs & Xsistors they will include a test fixture diagram. These are usually single frequency and not a broad band configuration. They are simply to guide you in the right direction in your PC board layout. As you go higher in frequency it beomes increasingly important to do a proper board layout. Let's discuss the electrical characteristics of both power amps and linear power amps for now. The input and output of power amplifiers is usually very very low impeadance. Usually on the order of less than 1 ohm (example: ".8-j38"). As you can see the phase angle becomes increasingly critical since we know from the example that a minus j operator means the impeadance is largely capacitive. We use Xformer techniques to shift the phase to match the phase angle. This is because of the high power produced by these power amp devices. They can quite easily break into oscillation from the output feeding back into the input. If the semi-conductor device starts to heat up it can go into thermal run away or also know as avalanch break down. This means that as the device is heated it will increase in amplification to a point where the barrier region will break down and self destruct. There are some commonly used tranformer techniques for broad band amplifiers and this would be where I need to put up some diagrams which I will get to.
好bsp; 好bsp;But what makes an amplifier linear? Well it must be biased class A or class AB. Many of you have studies load lines of a Xsistor device and so I don't want to go into saturation and cutoff right here, but I want to discuss real world construction techniques. Sufice it to say that a linear amplifier is always on through out the rf ac sign wave. In the data sheets you will see a term called the "IDC". The IDC is the suggested standing base current. As you know for a Xsistor to remain on all the time it must be biased to .7 volts atleast. To measure your IDC you must break the base circuit and complete the circuit with an Ammeter (Amp Meter). So if the IDC is recomended to be 200 mA then you must bias the Xsistor accordingly.
  Getting back to broad band ampifiers in general. If we look at the 2nd diagram below we see that the rf signal is capacitively coupled to the first stage. Notice the collector now has a step down transformer. This is what must done when we begin to develope significant power in a primarily transmitter amplifier string. When the power approaches values of 100 milliwatts or less you will need to couple using Xformers especially in broad band application. The Xformer here could be an Amidon ferrite toroid core for example at hf frequency you might use an FT-50-43. The turns ratio will depend upon the impeadance match between stages. Notice it is a step down Xformer. The windings will be say no. 30 guage enameled wire with for example 30 turns on the primary and the secondary is usually wrapped right over top of the primary and may be say 10 turns. We want to make sure that the polarity is the same for both primary & secondary. That means that the end of the primary that goes to the power source will be the same end that the secondary uses for the ground because in AC as we remember the positive and negetive are both effectively ground connections. We also notice in the two Xformers that there are no capacitors in the collector leg because we want to keep the amplifiers broad band. The ferrite toriods are capable of offereing the high impeadance needed without forming a frequency selective LC tank circuit. Once again the toriod by the nature of its shape offers very resonalble self shielding charateristics therefore giving lea way to non sheilded design if proper PC board techniques are applied.
Posted on 2006-08-17 09:24:57 by mrgone
To the Ineffable All,

    Speaking of oscillators, my favorite is the Wien-Bridge oscillator.  It is easy to build and needs no special components like split coils or ganged capacitors.  If fact it only needs capacitors and resistors for its frequency determining network.  And the frequency is easily variable.  Good distortion specs also.  There's lots of links and literature on this oscillator.  Ratch
Posted on 2006-08-19 08:23:36 by Ratch
  Well the link you provided shows that it used to produce audio frequency. You want to stick with coils in the RF spectrum. I personally would not use an OP Amp in any radio frequency oscillator applications. The key to a good stable oscillator is low power consumption. This also applies to VCOs, and the same priciples should be adhered to though a PLL is more forgiving.
Posted on 2006-08-20 20:24:55 by mrgone

    Yes, coils become more effective and smaller at higher frequencies.  I submit that the key to a stable oscillator in not power consumption, but a stable feedback loop that controls the oscillation frequency.  For instance, microwaves and radars consume and emit large amount of electrical energy in a short time, but they are stable.  Ratch
Posted on 2006-08-21 06:53:46 by Ratch
好bsp; Yeah that Upper UHF & SHF & EHF are a different animal unto its self. The reason they consume more power is due to the extremely high switching rate required by the devices that produce these frequencies. Not only that but it actually causes the devices to slowly deteriorate. I beleive that other than tubes the only semi-conductor devices capable of ths high rate of switching are gallium arsenide devices and they do deteriorate over time. Bare in mind that they usually are not used in the oscillator its self but the oscillator at those frequencies runs at a lower frequency and a frequency multiplier circuit is used to bring the frequency up. Alot of time if the frequency isn't too too offully high then we can use some kind of ECL (emitter coupled logic) to produce a frequency multiplier. Usually you will see ECL being used as a divider in a PLL circuit to take the output of a high frequency oscillator and divide down the frequency to a point where it can be divided down even further by TTL/HMOS/CMOS common digital logic. Even with "HCT" (example 74HCT191) "High Speed CMOS Technology" it still is only realiable at frequencies under 100 MHZ. So the ECL is used to break down the extremely high frequencies to a comfortable frequency for the logic. The ECL is what is known as the "Prescaler" in a PLL loop. The prescaler usually has a pin that can be toggled so that it can change its divide ratio from say divide by 20 to divide by 17. They call this dual modulus. So now you know what a dual modulus prescaler is. Actually as far as permanent errosion of a semi-conductor device, I beleive that only applies to gallium arsenide. Maybe someone has some input on that? Appreiate the chatter Ratch. It was getting lonely over hear trying to clarify some radio questions I've seen floatin :D
Posted on 2006-08-21 08:43:27 by mrgone