The story of antenna power


  I think that many station managers have a misunderstanding about peak power and average power when listening to radio QSOs. The rated transmission power of a radio is displayed as peak power, but the average power meter fluctuates by 50W during live voice transmission on SSB, so I think the PEP is 3 to 4 times that, or 150 to 200W. I have heard such stories, and it seems that some people are misunderstanding PEP as PP, and are misunderstanding that peak power = 2√2 times average power. To clarify, let's clarify the definition of peak power and average power. Both are effective values.
 
  Peak power (PEP) = Peak Envelope Peak. The point of maximum amplitude on the envelope, which is the effective value of one cycle of the carrier wave.
  Average power (Average) = The effective value over a period (generally 1/10 second, 100 ms) that is sufficiently lower than the lowest frequency on the envelope.

① In the case of 2T (two-tone) signal
  Peak power = Effective value of the maximum point area on the envelope for one period
       

Average power = effective value on the envelope for 1/10th of a second, which means

for the above two signals, it is 1/2 the peak power.

(The envelope is not at the same level as in CW.)


② For CW signals

  Peak power = CW signal, the envelope

level is the same for any period, and is the same value as in the case of two signals.


Average power = envelope level over a 100 ms period is also flat and is the same value as the peak power. In the case of two signals, the envelope repeats maximum and minimum (zero) during the 100 ms period, so it is 1/2.

In other words, if it is a CW signal, whether the power meter is set to average or PEP, the needle deflection will be displayed in PEP. Therefore, if you output a CW signal, you can check the PEP. However, if the level is different when you output your voice and when you output CW, it's pointless. You can't compare it unless you measure whether the ALC of the device is fluctuating or the ALC is set by the AF system.


 Analog and digital


I was in a completely QRT state from age 22 to 40, and I resumed amateur radio at age 40, and it has been exactly 30 years since then. When I resumed radio, I was using a Kenwood TS-950. Kenwood was an SSB modulation/demodulation type using DSP, while other manufacturers were analog machines. Now amateur radio equipment has become digital processing equipment, but the signal itself has basically not changed from the analog signal of over 50 years ago. Recently, I have heard about digital mode, but it is similar to TTY mode.

As the signal itself of consumer audio equipment and TV receivers has changed from analog to digital, the inside of the equipment has also been digitized and become compact. In the case of audio equipment, the noise in the signal is zero, and in TV receivers, snow noise and ghost signals are completely eliminated by digitalization, and logically the transmission density of the signal quality has increased, so it can be said to be a step revolution. However, in amateur radio, the signal itself has not changed, only the inside of the equipment has been digitized, so there is no change for the radio operator who uses it. Of course, although the operation control system has changed, the noise and distortion in the signal will not disappear. As always, SSB signals are analog radio waves in the 3KHz band. The reason the inside of the device has been digitized is for the convenience of the manufacturer, who is simply using digital technology because it makes it easier to produce cost-effective and environmentally resistant devices, but on the other hand, it is a natural progression because cost-competitive products cannot be produced without digitalization. For users, it is also good to see that it is compact and beautiful, does not use rotary switches (mechanical switches) like in the past, and has increased reliability. However, as I have said many times before, there has been no change in the signal itself, and the quality of the signal being sent and received has not improved significantly.
  Benefits of digitalization
   ① Adjustment is no longer necessary, improving productivity.
   ② There is no drift due to environmental changes.
   ③ Excellent selectivity is easily achieved.
   ④ No device noise is generated except for the input/output blocks.
  Disadvantages
   ① Artificial noise is generated in the A/D and D/A blocks.
   ② Limited dynamic range (the more bits processed, the larger the system becomes)
   ③ The required amplitude (dynamic range) can be achieved by increasing the number of bits, but there is a problem with the sampling frequency.
      The question is how much baseband reproduction frequency band is required.

  Since the source of an audio/video signal is analog, the principle of analog → digital → analog is, for example, to convert the analog signal of one cycle on the left into digital, the analog signal is sampled at a certain interval (called the sampling frequency). At this time, how many bits of sampled data are saved is determined by the required resolution (dynamic range) of the vertical axis (voltage) amplitude. In the case of a CD, it is 16-bit data with 44.1KHz sampling, and has a dynamic range of 96dB. The black dots are sampled data. A sampling frequency of more than twice the baseband is required in the horizontal axis direction rather than the vertical axis direction, and since CDs are 44.1KHz, basebands of 22KHz or more cannot be reproduced. If the baseband is 20KHz, only two samples can be sampled per cycle.
The signal on the left is a digital audio signal with a baseband of 2KHz, and about 20 samples are taken. There is no data between the black dots. Since the analog signal is a red line, naturally there is no data missing at all. This is a theory, and there is nothing wrong with it when listening to it, but it is not interesting from a theoretical point of view. This may be the reason why people say that digital is no good and that analog is the only good, or is it just a preconceived notion?

  
As mentioned above, it is true that with recent equipment, frequency stability and performance are no longer affected, but I still feel that there is a problem with the reproduction of details (how fine details can be reproduced), which is often talked about in the analog world. What do you think, station managers?The reason why I have an attachment to analog PSN equipment since restarting broadcasting is because manufacturers could never produce it. Technically they could produce it, but they would never produce equipment with such low productivity. Even if they produced such equipment with many adjustment points, it would not be competitive in this day and age.That is why I continue to be attracted to it, because there is nothing more wonderful than being able to make things that manufacturers cannot produce. This is a big advantage of those who make their own equipment, because they can use first-class parts and make adjustments in many places, and I think this is the best part of making things yourself.

It is often said that homemade analog equipment is too old these days, but even with analog, there are no problems from the ultra-low base band to 3KHz, and the stability is absolutely fine for practical use. I have recently been doing a lot of QSOs with station leaders using digital equipment (IC-7300 and IC-7610), and there are no problems with this either, of course. However, when comparing stations with strong signals, I do feel a difference in the sense of detail. Maybe this is just a preconceived notion?



 The story of three terminals

 
 

When building your own power supply using three terminals, have you ever had the experience of failing to connect the terminals properly and damaging a peripheral device? 78/79/78L/79L each have different input/GND/output terminals. It's a hassle to open the specifications and check them every time, so here's a tip on how to remember them. The basic rule is to remember them from the left as high (KOH), low (TEE), and middle (CHU).


         

                
From the left, the order is: high (K) = highest potential, low (T) = lowest potential, medium (Chu) = medium potential. For the 78L type only, the order from the right is: high (K), low (T), medium (Chu).


 The story of ICs


IC has made rapid progress, but we amateur radio operators cannot say that we have benefited from it. Digital ICs have made great strides in terms of low consumption, high speed, and high integration, but analog ICs have moved in the direction of custom ICs for systems, and new single-function general-purpose ICs are rarely produced, except for operational amplifiers. Semiconductor manufacturers cannot produce tens of thousands of units per month. Although there are ICs that can be produced in small quantities and at high unit prices, if the IC manufacturer plans and produces them themselves, it is not possible to produce them in small quantities. In consumer devices, IC manufacturers have developed and produced ICs for each category of use, such as TV, radio, stereo, and video, and mass-produced them. However, in recent years, with the digitalization of systems, systems have become more complex, and dedicated ICs have been developed, and the number of ICs that can be applied to radio equipment has decreased. Although the signals of radio equipment are analog, the equipment has become digitalized, and although it has become stable through the use of CPUs and DSPs, there are only a few ICs developed for radio equipment. DIYers have no choice but to use ICs developed for old consumer devices, but these ICs are not commercially available and are difficult to collect. Also, both digital ICs (CPU/DSP/DDS) and current analog ICs are surface-mounted packages, which makes soldering difficult. I wish they would develop components for DIYers (all lead-type components), but that's not going to happen.

Here are some of my favorite ICs from throughout history.

    
  This IC is a Mitsubishi electronic volume IC that was developed when AV (Audio-Visual) equipment first came onto the market. It has a D-range of about 90 dB and excellent distortion characteristics at any point. It is convenient for making your own ISB mode radio with two channels on one chip. It is ideal for limiting amplifiers and AGC amplifiers at the audio level. It is easy to use because it is DC voltage controlled. Since then, ICs of this type have been developed that incorporate peripheral preamplifiers, and many electronic volumes have appeared that are digitally controlled rather than DC voltage controlled, but I still use this IC.
  Like the M5283P, this IC was developed when AV equipment first appeared on the market. It was developed for function selectors and can switch between audio and video signals. It can be used at ±15V (30V), providing a wide D-range. There are only a few ICs that can switch at ±15V, and a wide D-range is required up to the limiting amplifier, including the microphone amplifier, making it ideal for switching input signal sources. Distortion can also be kept below -80dB. However, because the switching control is digital, a CPU, etc. is required.
  This is a high-level mixer from Pressey, and is a much talked about IC with IP=30dBm. Its features include a small local OSC of about 100mVpp, and a gain of about 18dB. With an output signal amplitude of 700-800mVpp, it is possible to obtain an intermodulation (IMD) of -80dB. Recently, instead of using a PLL or similar to generate the local OSC, it is often the case that a signal (pulse output, TTL level) that directly outputs a signal connected to the reference clock is used, and mixer ICs that can be driven directly include Fairchild's FST3125 (4-bit bus switch). This is also a FB mixer circuit, and one of my favorites.


 
The story of opeamp

 

 Recent op-amps have an extremely wide range of applications, with more and more devices capable of handling extremely high frequencies. Current feedback op-amps were developed for use in wideband op-amps, but now many voltage feedback op-amps for wideband use are also available. Basically, the bandwidth of voltage feedback op-amps changes depending on the gain setting used, but the bandwidth of current feedback op-amps hardly changes at all. However, current feedback op-amps have a narrow range of peripheral constants that can be used, and deviations from this range can lead to sudden unstable operation. There is little to mention here, as there is plenty of information about op-amps available on the Internet. Here we will discuss the following four items.

① Choosing between inverting and non-inverting input amplification

    

The choice of which to select and configure will be fixed in cases related to phase (polarity), but otherwise either one is fine. The only differences are as follows:

   ● With the inverting type, the load of the previous stage is determined by R1 and a high input impedance configuration is not possible, whereas with the non-inverting type, a high input impedance configuration is possible and is determined by the bias resistor value.
   ● The inverting type has a gain of 1 (0 dB) when R1 = R2, negative dB when R1 > R2, and positive dB when R1 < R2. It is used for applications ranging from -dB to +dB. It is best to give R1/R2 characteristics (C/R configuration), i.e., filter circuits and servo circuit loop filters. The non-inverting type has a gain of 1 (0 dB) when R1 = ∞, and cannot be configured for negative dB gain.
② Op Amp Offset Voltage
The offset value specific to the device is determined by the IC specifications, but when configured as an amplifier circuit, the offset value that appears in the output is greatly influenced by the settings of peripheral constants and gain values. This is not a problem with FET inputs, but if the device is bipolar, the bias current will cause the offset value.

            

The bias current of an op-amp flows from each input terminal to GND (outside), and the bias current value is specified in the IC specifications. This current flows through R1 and R3, which means that a voltage due to this current value is generated at each input terminal. Since the two input terminals of an op-amp are differential inputs, even if a voltage is generated at the input terminal, if the voltage value is the same at the inverting terminal and the non-inverting terminal, it will be differentiated and will not appear in the output, so there is no problem. However, if this value is different, the difference will be amplified and will appear in the output. (Especially the greater the amplification) As you may already understand from this, in order to eliminate the influence of offset values ​​due to peripheral constants, it is assumed that R1 = R3 is configured. Also, the larger the values ​​of R1 and R3, the greater the influence. In reality, it is often difficult to maintain this condition. If the output offset value is a problem in reality, it is necessary to install an offset cancellation circuit in the input circuit.
③ Amplification circuit bandwidth
In amplifier circuits using op-amps, the GB product (gain bandwidth) is listed in the IC specifications. However, a common misconception is that if the GB product is listed as 150 MHz, the IC can be used up to 150 MHz. However, the GB product is specified when the gain is 1x, and the bandwidth changes depending on the gain. For example, if you create the inverting amplifier above using a voltage feedback IC, with R1 = 1KΩ, R2 = 2KΩ and a gain of 2x, the bandwidth will be GB/(1+G) = 150MHz/(1+2) = 50MHz.
If you configure R2 = 3KΩ and a gain of 3x, the bandwidth will be 150MHz/(1+3) = 37.5MHz. Conversely, there are also cases where the bandwidth is taken into consideration when designing.


The story Coupling C

Regarding the value of the coupling capacitor C, I often hear the following: 1uF or 2uF coupling is no good, it needs to be made bigger. Some people talk with the understanding that 1uF or 2uF coupling is no good, it needs to be made bigger, ignoring the input impedance of the next stage, but I feel that the capacitance is a bit unnecessary.

There are few cases where audio pre-circuits are configured with an input impedance of several hundred ohms to several kilohms, and most are designed with tens of kilohms, and in the case of 10Kohms, even 1uF has a cutoff of 15Hz. If you are driving a speaker (8Ω), you will need to make it bigger than 1uF or 10uF, because even 1000uF has a cutoff of 20Hz. 



The story of limiting amplifiers


 This is also something I often hear, but if you put limiting on it, it's no good, so you should take it off right away, or if there's a limit on the low range, the high range will be suppressed and the sound will not come through well. I don't know what kind of limiting is used, but I think it must be really worn out or used incorrectly, but the music on a CD is transmitted over the air, and CDs are already limited, so why add even more limiting? When a signal is moved from one object to another, if the original D-range can accept it as it is, there is no problem, but if it is moved to a place with a narrow D-range, it needs to be limited. In the case of CDs, it is limited to within 96 dB, and when this is transmitted from the radio to the radio wave propagation to the other receiver, it cannot be reproduced with the original D-range. I remembered an experiment I did before where I divided the frequency into three and limited amplification, and in such cases, depending on how the limiting is applied, it may have a three-division frequency characteristic, which can be a similar phenomenon to graphic EQ correction, but the limiting amplifiers used recently are full-bandwidth. It is probably a time constant issue, and if you set the optimal time constant, there will be no discomfort. The D-range reproduced by the air monitor is completely different from the D-range reproduced by the receiver of the other party, and I think there is some misunderstanding about this. I think there are people who use limiting amplifiers without understanding the purpose (reason) of installing them in the first place. There was a time when they were popular, and some people thought that installing them would make everything sound better, interpreting them as a magic box. Broadly speaking, there are two purposes.

① Set a limit on the upper level.
For example, you need to know how linear your radio waves emitted from the antenna are, and where they enter the non-linear region; the ALC is a system guideline. For example, a 100W machine can emit clean radio waves up to 70W, but what if it goes above that? In that case, you need to set the microphone so that it will not emit more than 70W no matter how you talk, in other words, set the operating point of the limiting amplifier to 70W. Doing this should eliminate the intense crackling radio waves.

② Increase the talk power.
The operating point is set at ①, but the talk power can be increased by increasing the input signal to the limiting amplifier. To take an extreme example, if there is no talking and only background noise in the radio room, and the input and output levels of the limiting amplifier are the same, the constant power will remain at 70W and will not change. What the talk power should be set to will depend on the usage environment. Sometimes the settings are inappropriate and the background noise is too noisy, but if both are set properly, you can operate comfortably.




The story of  SSB


In SSB communications, you often hear things like this: For example, when a station is communicating at 7.100MHz, if a station is communicating at 7.100+3KHz=7.103MHz, stations 3KHz lower at 7.1MHz can be heard sharply on the receiver, which is a nuisance, and the lower station's bandwidth is too wide; or, conversely, if an LSB signal transmitted at 7.100MHz is received at 7.097MHz, it causes interference, so when you ask the 7.100MHz station to narrow the bandwidth, they'll say, "Your receiver's bandwidth is too wide, so just narrow the receiver's bandwidth." It sometimes seems like some stations don't understand the theory behind SSB (LSB/USB) radio waves and reception (demodulation). There is a story like this, in the case of LSB radio waves, for example, if there is a station communicating at 7.100MHz, and you want to transmit radio waves, which would you transmit at, 7.1M-3KHz or 7.1M+3KHz? Some people say that some stations don't like hearing the sharp noise of lower stations, so the other party probably feels the same, and so they transmit at 7.1M+3KHz. Generally, a rasping sound is easier on the ear than a sharp sound, but in principle, the sharp noise of lower stations can be solved by narrowing the band in the receiver settings. However, the rasping sound of higher stations cannot be solved by the receiver band, and cannot be avoided because the higher high frequency components overlap with the receiving carrier point. Therefore, in principle, I think you should transmit at 7.1M-3KHz.

   Apart from recent digital processing radios, it is natural that analog radios and homemade radios will interfere with adjacent stations at 3KHz. For commercial use, adjacent frequencies are not allocated, so there is no problem, but in the amateur world, even if you try to narrow the band, there are cases where it cannot be avoided depending on the strength of the radio waves. We talked about demodulated audio, but in the case of SSB reception, even if the audio components of adjacent stations are not demodulated, the AGC becomes a problem. In this case, when 7.097MHz is received as LSB and 7.1MHz, 3KHz above, is received, it would be fine if the receiver filter only takes in 7.097MHz or less and does not take in the upper frequency, but in reality, in order to increase the fidelity of the sound, it also takes in the upper frequency slightly. This upper frequency takes in the AGC control, and the voice of the other station you are communicating with at 7.097MHz becomes small, which can sometimes feel strange. The only solution is to shift the filter, but in any case, apart from extreme leakage to the opposite side or splatter signals, isn't it inevitable that adjacent signals will overlap?