Thursday, 9 January 2020

A Simple Speech Processor
(For QRP/SSB Homebrew Transceivers  ) 

Over the last few weeks I had been thinking to build a small AF speech processor to add to my Bitx's microphone. I use a Baofeng microphone for my bitx, so the aim was to design a small PCB to fit inside the microphone enclouser itself. The schematic diagram of the circuit is given here:


The circuit is designed around a low noise OP-Amp. TL071 configured as a microphone pre-amplifier having 40dB gain. The output of it is sampled by an NPN transistor T1 and thus to control compression using the FET. The circuit is too simple but provide excellent results. OM M.V.S. Sarma has kindly produced and provided the following PCB layout for the Baofeng microphone.

Monday, 8 April 2019

FIREO©
A LOW POWER SSB/CW TRANSCEIVER WITH ITS UNIQUE SPEECH PROCESSING




“FIREO” (Pronounced FIRE-O), is basically a bit of non-conventional QRP (low power) SSB/CW transceiver design in which speech compression is implemented using FM limiter circuit. This unique approach of speech processing increases the effective average transmitted power and thus it improves on the signal strength reports at the receiving end. Consequently this technique also helps to cut through any man made or natural noise, very effectively.

Basics of RF speech processing:  Imagine a power amplifier designed for 10 W PEP driven by a mean SSB signal, which at least will be down by 6 db below the peak. This means a minimum output power of 2.5 watts and a resulting S-meter reading of one step down from the peak. Equalizing the dynamic range of the modulating signal will result in a better affectivity of the power amplifier as this will raise the "mean" output power. Even if this might not be directly visible at the receiver S-meter, the compression of the dynamic range will increase the readability and the SNR at the receiving side. In practice, it could be proven that a – moderate - clipping limit of 20 db virtually simulates a 10 watts transmitter to be an 80 watts transmitter while, in reality, the pep output is only 10 watts. Let's understand the root of this philosophy. (Download) 

The Felcher-Munson Philosophy:  Grokking this theory is a bit beyond my brain right now, but the Fletcher–Munson curves are one of many sets of equal-loudness contours for the human ear, determined experimentally by Harvey Fletcher and Wilden A. Munson, and reported in a 1933 paper entitled "Loudness, its definition, measurement and calculation". The first research on the topic of how the ear hears different frequencies at different levels was conducted by Fletcher and Munson in 1933. In 1937 they created the first equal-loudness curves. Until recently, it was common to see the term 'Fletcher–Munson' used to refer to equal-loudness contours generally, even though a re-determination was carried out by Robinson and Dadson in 1956, which became the basis for an ISO 226 standard.

It is now better to use the generic term "equal-loudness contours", especially as a recent survey by ISO redefined the curves in a new standard. According to the ISO report, the Robinson–Dadson results were the odd one out, differing more from the current standard than did the Fletcher Munson curves. The report states that it is fortunate that the 40-phon Fletcher–Munson curve on which the A-weighting standard was based turns out to have been in agreement with modern determinations. The article also comments on the large differences apparent in the low-frequency region, which remain unexplained. Possible explanations are:

1. The equipment used was not properly calibrated.

2. The criteria used for judging equal loudness at different frequencies had differed.

3. Subjects were not properly rested for days in advance, or were exposed to loud noise in traveling to the tests which tensed the tensor tympani and stapedius muscles controlling low-frequency mechanical coupling.

Thus equal-loudness curves derived using headphones are valid only for the special case of what is called side-presentation, which is not how we normally hear. Real-life sounds arrive as planar wave-fronts, if from a reasonably distant source. If the source of sound is directly in front of the listener, then both ears receive equal intensity, but at frequencies above about 1 kHz the sound that enters the ear canal is partially reduced by the masking effect of the head, and also highly dependent on reflection off the pinna (outer ear). Off-centre sounds result in increased head masking at one ear, and subtle changes in the effect of the pinna, especially at the other ear. This combined effect of head-masking and pinna reflection is quantified in a set of curves in three-dimensional space referred to as head-related transfer functions (HRTFs). Frontal presentation is now regarded as preferable when deriving equal-loudness contours and the latest ISO standard is specifically based on frontal and central presentation.

The A-weighting curve—in widespread use for noise measurement—is said to have been based on the 40-phon Fletcher–Munson curve. However, research in the 1960s demonstrated that determinations of equal-loudness made using pure tones are not directly relevant to our perception of noise. This is because the cochlea in our inner ear analyzes sounds in terms of spectral content, each "hair-cell" responding to a narrow band of frequencies known as a critical band. The high-frequency bands are wider in absolute terms than the low frequency bands, and therefore "collect" proportionately more power from a noise source. However, when more than one critical band is stimulated, the outputs of the brain sum the various bands to produce an impression of loudness. For these reasons Equal-loudness curves derived using noise bands show an upwards tilt above 1 kHz and a downward tilt below 1 kHz when compared to the curves derived using pure tones.

BBC Research conducted listening trials in an attempt to find the best weighting curve and rectifier combination for use when measuring noise in broadcast equipment, examining the various new weighting curves in the context of noise rather than tones, confirming that they were much more valid than A-weighting when attempting to measure the subjective loudness of noise. This work also investigated the response of human hearing to tone-bursts, clicks, pink noise and a variety of other sounds that, because of their brief impulsive nature, do not give the ear and brain sufficient time to respond. 

What does that actually mean: The way to read this graph is as follows: look at the blue curve at the 1 kHz / 40 dB point. Now follow the curve towards the left until you reach 50 Hz on the horizontal axis. You should now read about 70 dB on the vertical axis. In essence, this states that in order for a 50 Hz tone to be perceived as loud as a 1 kHz tone is at 40 dB, it needs to be played at 70 dB. That’s 30 dB difference! A similar thing happens when you move into the high frequencies. A 10 kHz tone needs to be played at about 55 dB to be perceived at the same loudness level. Notice that this difference in loudness evens out as the volume increases (the curves higher up in the figure), for example at 100 dB, the curves have flatten out considerably, meaning the perceived loudness difference between tones at different frequencies decreases. There are two important things to take away from these curves:

1. We are less sensitive to low and high frequencies, we hear mid frequencies more prominently (especially between 1-5 kHz)

2. As the volume increases, this perceived loudness difference between the frequencies diminishes.

However, this made the basis of one of the pioneering developments in low power DX voice communication in which the high amplitude vocals are compressed for an even distribution of power over the usable bandwidth. Based upon this research; in HF-SSB radio technology in the era of late sixties, became a dependable method of modifying the speech waveform in the transmitter to produce a marked improvement in the signal-to-noise ratio at the receiver without also causing any significant increase in distortion products, either in-band or out-of-band. Since RF speech processing was the key to the performance of low-power HF-SSB radio sets - and is now recognized almost as a sine-qua-non in SSB transmitters. Typically, unprocessed speech has a ratio of instantaneous peak to average power of about 16dB.

CIRCUIT DESCRIPTION:

RECEIVER: During the inception of the design of FIREO transceiver I zeroed upon my choice for the well known Motorola I.F. subsystem MC3362.  Though the chip is already common in amateur literature and has been used in a score of transceiver designs, both for H.F. and V.H.F/U.H.F as the chip contains most of the circuitry required for the job besides two Gilbert cell mixers, a limiter and a discriminator. So I found it to be well suited for the purpose of a portable H.F. transceiver design. During the development of design, my initial experiments revealed that the in-built mixers are quite vulnerable and are prone to easy overloading by strong signals on H.F. bands. Consequently I decided to use a home brewed double balanced diode mixer for the receiver front-end. The input signals from the antenna are filtered by a band pass filter wired around inductors L1 and L2 and are amplified by an amplifier made using transistor Q2. This amplified signal is then fed to a diode mixer made using diodes D1 to D4. Here it is mixed with L.O. signal to generate an I.F. signal. The I.F. signal is then fed to an I.F. amplifier using a FET Q9, through which AGC (Automatic Gain Control) function is also achieved. Much care is paid to the diode mixer port termination to achieve best IP3 and optimized performance. The I.F. signal thus passes through the diode switch D6 to the home made X-tal filter and after filtering is fed to the inbuilt mixer of IC MC3362 at its PIN 1. The I.F. signal is here mixed with the oscillator signal generated at PIN 3&4 of the IC and the demodulated audio thus generated is steered out from PIN 19 of the IC to the low pass filter constituted around R33 and its associated components. After filtration the recovered audio is then routed to the audio amplifier, through volume control for necessary amplification.

COMPLETE SCHEMATIC OF FIREO EXCITER:
(Last Updated On:25th March, 2020)

SPEECH PROCESSING IN FIREO:  As mentioned in the start of this article, An RF speech processor will give your qrp SSB signal valuable extra "punch" to cut through QRM. In FIREO transceiver a unique method of speech processing is used. After completing the basic transceiver design using both inbuilt mixers I took a detailed look on MC3362 datasheet. It contained an inbuilt limiter and a discriminator as well and I decided not to waste these circuit resources and to make full use of them.

The speech signal from the microphone is amplified by microphone amplifier made around transistor Q8 and is mixed with the 455 KHz signal from the DDS VFO to generate a DSB signal. This signal is then fed to the inbuilt limiter stage of IC MC3362 at its PIN 7. Where it is compressed and then it is demodulated using the inbuilt discriminator, using 455 KHz signal at PIN 12 of MC3362. The processed demodulated signal thus is available at PIN 13 of the IC which is buffered by transistor Q1 and is then routed to the balanced modulator at PIN 17 of the IC. R7 sets the required microphone gain. This increases the average output level of an audio signal from a microphone by clipping off the excessive signal peaks. By lowering the peaks in proportion to the average level, a higher average output level can be attained with an associated increase in intelligibility under difficult conditions. It is set up easily without special equipment because no RF filters are used. 

INTERNAL BLOCK DIAGRAM OF MC3362:


TRANSMITTER: The BFO signal is generated at the PIN 3, 4 of the IC. This signal is then modulated with the processed audio signal fed at PIN 17 of the IC, using the inbuilt double balanced modulator and the DSB signal thus generated is available at PIN 5 of the MC3362 which is then routed to the SSB X-tal filter through diode D7 and the SSB signal thus obtained is fed to the inbuilt second mixer at PIN 1 of the IC MC3362. The LO signal is applied at PIN 22 of the IC through the steering diode D15 and the transmit SSB signal is finally routed through diode D8, from PIN 19 of the MC3362 to the RF pre-amplifier Q3 which provides around 20dB of RF amplification.
 The RF amplifier constitutes three stage of amplification for the RF signal to reach required power level. Most of the circuit uses usual topology and is quite self explanatory. As it is becoming difficult to get some medium power, discrete RF devices I attempted to build the driver stage by wiring Q 14 and Q15 as a pseudo ballasted emitter transistor. RF final amplifier uses ubiquitous IRF 510. Q16 and Q17 are included as protection devices. In case RF output stage consumes more than a specified limit of current voltage developed across R83 causes the transistor Q16 to conduct. Thus a positive voltage flows through R85 to the base of Q17 pushes the Q17 into cut-off region and removes the gate bias. This way the final stage is protected against all odds. L3 is 9 turns wound of 26 SWG, self supporting coil wound on the body of a pencil. This air core coil tunes with the input, gate capacitance of the IRF and thus even on higher HF bands the device is made to perform with guts. Q19 along with diodes 21, 22 constitutes the antenna switching circuitry and diode D23, 24 protects the receiver front-end against RF spikes and thundering etc.



A discrete audio power amplifier is built around a low noise OP-Amp IC TL071 and a pair of complimentary power transistors. The amplifier has quite a high gain and can produce almost 2.5W of powerful audio. C68 and C69 are included to push the crossover distortion to the lowest possible mark. C67 shapes audio and can be increased to suit to your taste. Diodes D19 and D20 are included to generate an AGC signal in a simplest way. Transistors Q4 and Q5 generate switched RX and TX supply for different stages.





(Video: FIREO built by VU3VRL OM Ramesh.)

(Updated 5-5-2020: A revised version of Fireo transceiver with few refinements is under testing and will be posted soon.)

As for the requests, I am planning to arrange few kits of FIREO, those who need one, can drop me a line. My E-mail ID is:   kangkps@gmail.com.

Friday, 3 November 2017

BITX MODS AND THOUGHTS.

For the last  few weeks I have been busy twiddling  with my little QRP BITX rig. Of late, I have come across some problems and have devised some mods to these problems  and I feel these worth sharing  with  you.  


1. IMPROVED MIC AMPLIFIER: I felt that my BITX suffers from low microphone  sensitivity. You have to speak quite loud to get the signal properly modulated. A crass examination  of the microphone amplifier revealed that values biasing components seem to be little inappropriate and need to be recalculated.


Values of R126 (collector load of Q12), the mic amplifier  transistor and R 123 (emitter swamping resistor) are thus recalculated  and are replaced with a resistance of 4K7 for R126 (in place of its original  value of 1K) and R123 is replaced with  a value of 10 ohms ( in place of 100 ohms). You can spot these components near the left edge of PCB. 


The results of this simple modification were immidiately recognizable and were as expected. The rig now has required  microphone sensitivity and transmissions  are reportedly better.

2. CLICK FREE BITX MOD. : One annoying thing about BITX I felt is relay clicks and clicks caused by relay K2 in audio line.  Consequently  I decided to implement solid state circuitry to replace both relays used for TX/RX supply switching  and antenna changeover. The supply switching  circuit  is built on a pigmy  board as shown below :



During RX transistor conducts andRX stages get supply to work. During this condition Q3 grinds the pin 14 of K4 (Ex)  to ground the drive to pre-driver stages of the RF power amplifier. When PTT is pressed Q1 turns off and Q2 conducts to extend supply  to TX stages. During this period Q4 conducts and it's collector grounds the top (live) end of volume control causing the RX to mute. C1 and C2 ensure reliable switching  function even if a spurious RF radiation picked by the base of these transistors. It is necessary to remove capacitor C11 connected to the base of transistor Q1, for this MOD. as the antenna changeover function is performed by the following circuit :


I used an axial type moulded inductor readily available  in local market  for this MOD. However you can wind Your Own on a toroid available  to you.  The values of the components are not critical and are readily available. The results of this  MOD.  are really  satisfying. A professional  click free QSK experience.

3. AN R.F. BASED AGC MOD. :    BITX is a good  portable  rig  but its major handicap is that it lacks AGC. Though an AVC (automatic  volume control) circuit can be used to compensate it and is available online. I decided to include a more convincing RF AGC circuit to my BITX. Primitively it was thought to apply AGC through IF amplifier stages in a conventional design topology. So I thought  to put a transistor or a FET in the writer circuit of IF amplifiers as shown below.



The AGC voltage applied to the base of Q2 will cause to vary the current t in its collector circuit thus controlling the overall IF gain. But it was thought afterwards  that instead of course controlling the gain at IF level it would be better to control it before the mixer that would be an incentive considering the IMD performance of the mixer and all stages following it.  Consequently the following topology was initially thought to provide an effective AGC at RF stage. 


But in this case reducing the base voltage of RF amplifier can adversely affect the overall IMD performance of this stage. So after a detailed  analysis I reached the conclusion to use a FET to replace the Q1 based RF amplifier of the original design.  The resultant design is as under:


Audio signal is picked up from the hot end of the volume control and is amplified by transistor Q1. An AGC signal is then generated through  diodes D1 and D2, which is then used to control the gate bias of RF amplifier and thus its gain. The entire circuit is built on a small pigmy board and is installed close to the original BITX board. As the output of amplifier is about 440 ohms, you need to adjust primary turns to about the double of turns of other windings for a proper match. The results are as expected. Tuning through a crowded band full of 59+ signals is no more a bane. 



Wednesday, 23 August 2017

Two Essential Add-ons For Bitx and Other QRP Rigs.

In this post I am going to discuss two very simple but very essential add-ons for Bitx and other similar type of QRP sideband rigs. These two add-on modules are the recent addition to my XENA and BITX rigs that have already proved their worth during portable QRP operations. The first one is of course a simple speech processor. It can give more punch to your transmissions and more QSO's are certainly assured for you.

A SIMPLE SPEECH PROCESSOR:  A simple speech processor can enhance your readability on the receiving end by as much as two S-points and is an indispensable addition to any QRP voice rig. There are usually two types of speech processing techniques. One using limiting through compression and other through clipping. The latter is usually known to give better results on both AF and RF. The present circuit uses this technique. The signal from the single stage mic amplifier is routed through this module. The first stage gives it necessary amplification and the signal is clipped using a pair of back to back diodes. I used germanium diodes as these were available in my collection but BAT 54 or similar are likely to give better results. The signal is then filtered using an op- amp based sallen key filter for harmonic suppression and is then amplified and is routed to the diode DBM. The circuit can be use with almost any SSB/DSB rig. For use with bitx you should use 47K resistance for R1 or use a potentiometer at the output to get proper level of audio for modulation.  The circuit is very simple and is given below:


I have chosen an LM358 for being cheap, readily available and for its smaller footprint than discrete devices, to aid smaller design. You can play with the values of C1 and C4. Try a value between 0.0033uf to 0.0068uf for C1 to suit your voice quality. Generally the least bass input to the DBM is the key to best readability. Try 1nf for C4 for much less splatter across the transmitted bandwidth.

The alignment is simple. An oscilloscope can be indispensable but if you have none don't worry. Measure the average AF signal amplitude at DBM input using a sensitive AF meter. Now switch to compress mode and set the viper of R8 all the way to ground. Now adjust R4 for an equal and almost similar amplitude reading at the input of your DBM. Go on the air and call a friend. Adjust R8 now for best results......and you are all done.

A COMPACT ANTENNA TUNER: A simple and compact antenna tuner is made of just two components; C2 and L3. L3 is a half watt resistance sized moulded inductor of about 12 uH. For 30 and 20 meters an inductor of about 6.8 uH would be sufficient and for twenty through fifteen meters just 4.7uH suffice. C2 is a common BC type variable capacitor and is not hard to find. I usually use a half wave throw away wire for my portable QRP use as it is very convenient to carry.


This compact tuner can tune both high and low impedances very effectively and is small enough for portable use. I have included N7VE's SWR bridge for ease of tuning. The entire module is small enough to be built and accommodated within the bitx cabinet.




Sunday, 16 July 2017

VXOs-Simple Minimalist's Signal Sources.

With the recent advancement of technology, the availability and accessibility of stable signal sources has become within reach of an average radio home brewer. The newer versions of super stable and very dependable synthesizers based on both phase lock loop and direct digital synthesis are available for a moderate price. In addition to their accuracy, they usually come with an ornamental frequency display as well, to decorate the front panel of your rig.  

But for the minimalist QRP enthusiast X-tals are usually considered the cheapest stable alternatives. In practice though; they are undoubtedly cheap and stable but they do not offer the needed agility. Recently some good VXO designs have been published by some ham friends those offer the desired agility to work.  

In almost all of the projects I described, I use VXOs and I preferably advocate them as a low cost Scrooge's first choice. In this post I describe two ceramic resonator dual band VXOs, those can be adapted for X-tal operations as well.

1. BIPOLAR VXO: The bipolar VXO circuit uses three transistors, first of them wired as a colpitts oscillator. A common 3.58MHz ceramic resonator is used to produce oscillation. A varactor type tuning method is employed. I use a 5mm red LED as a varactor diode. Alternatively 1N5808 type diode can be used for this purpose as they exhibit linear tuning. A good quality ten turn potentiometer is recommended for the tuning control, preferably mounted with a calibrated rotary dial.


Q2 and Q3 are wired as a buffer amplifier that provides a reasonable output of about +17dBm. L2 should be wired on an eight mm dia slug tuned former with about 13 turns of primary and five turns of secondary. I used 28 SWG copper wire for the windings. If you use a toroid core for L2, add a trimmer capacitor parallel to C6 for fine tuning. C1 and C2 should be NP0 type or you can try styroflex type capacitors. L1 should be a moulded RF choke of about 10uH or more. In my version I just used the primary winding of an old 455KHz IF transformer with its internal capacitor removed. It measures 33uH. The VXO provides exceptionally stable signal even from the cold start. It provides about 80KHz tuning on 80 meters and about double on the forty.

2. DIGITAL VXO:  VXOs can also be built using digital invertor chips as these are cheaply available, these days. I have built many of them using 74HC04, 74HCU04 and CD4069 etc. The "U" suffix represents unbuffered version of the invertor and is usually preferred version for such applications. Though these chips are designed to be used at 5 Volts supply but they can work happily till 6 Volts and deliver a little more output; about 24mW. This level is sufficient to drive a diode ring type double balanced mixer. The schematic diagram of such a VXO is shown below:



C4 is an old ex-BC receiver component that can be replaced with a varactor arrangement for tuning, if required. L2 is wound using 32SWG, 11 turns close wound on a former made from the body of an old ball pen. The output low pass filter is included to filter out harmonic products that can cause erroneous harmonic mixing in the mixer, since being square wave the output of this VXO is rich in harmonics.

Both the VXO circuits illustrated above can be used in a variety of home brew direct conversion designs. Apart from ceramic resonators, three to four crystals can be used in parallel. I am using four 5MHz crystals cheaply available for use in microcontroller projects, in a bipolar VXO circuit mentioned above to provide coverage of QRP segment of 40 meter band on bitx. Mouser also lists 4.915 MHz ceramic resonators those can be used for a bit wider band coverage. 

An FLL can be used for more frequency precision. If you like to add a simple frequency counter, you can try Fredy's (DJ3KK) SPRAT counter. It uses an eight pin u-controller 12F675, thus it is very small and cheap and announces frequency in Morse, through a small piezo buzzer.


Friday, 26 May 2017

XENA- A Portable DSB/CW Transceiver

In my post of 24 th March, 2016 I shared a simple direct conversion receiver project built around a home-brewed mock I.C. This project grows out from that design. 'XENA' is a simple, portable DSB/CW transceiver for backpack use. As mentioned above this cute little project evolved around a single home made mock I.C. (made around components shown in a dotted square). It functions both as a transmitter and receiver mixer. For simplicity the design uses a ceramic resonator/crystal based dual band, super VXO for 80 and 40 meters (more about this later). The receiver is developed around ubiquitous parts generally available in one's collection and the complete project is developed in modular form with all modules made using ugly construction technique. The modular construction allows scope for future experimentation and further development of the project. The schematic of exciter cum receive module is given below:



Even to a crass view the circuit of exciter cum receiver module is too simple and hardly needs an explanation. I have added a multi-turn preset in the biasing arrangement of the mock mixer I.C. to aid precise mixer balance in order to achieve minimum carrier leakage during transmission. All broad band RF transformers are wound on pig-nose balun cores using 30 SWG enamelled copper wire.
The RF amplifier circuit is quite self explanatory. The schematic diagram of RF final is given below:


The three stage amplifier ensures almost seven watts of RF output. I used 2N4427 for the driver as it was available in my collection but many other suitable candidates like 2N3866, 2N5109 and 2SC1175 seem to work as well. The receiver band peak capacitor C21 is a 330pF type variable tuning capacitor. It eliminates the need of mechanical band switching arrangement in receiver front end using switches or relays etc. An attenuator ahead of it has been included for dire reception phases. It can be switched in during presence of receiver overloading.

The entire project is developed on 4"X6" PC clad pieces using ugly construction and each board is fixed to the sides of the enclosure body. The control circuit being in the middle. The control circuit provides the required QSK delay during CW operation, does all switching and provides side-tone during CW transmissions. I used a small piezoelectric buzzer, the type used in computer motherboards, UPS's and microwave ovens etc. to provide CW side tone, since the idea was simple to implement. The schematic for control board is given below:





Sunday, 2 April 2017

ZERON -A Super Simple QRP Dual Band Multimode Transceiver. -II

Just back after the hiatus, mainly due to a complicated leg fracture. However in this brief post I will share the direct conversion receiver part of "Zeron".

As I already told you about my obsession for digital chips for use in RF circuits and especially in switching mixers etc. I used an 'HC4053 as receiver mixer. This chip is an excellent choice over expensive diode mixers for many reasons. First as I already mentioned that it is cheap and easily available. The measured insertion loss is just about 0.5dB and the off state RF isolation is better than 45dB. The measured return loss is also better than 22dB for a variety of chips put under test, from different manufacturers. More than that you don't need to balance or match any devices! It is wonderfully simple to use.


The mixer is preceded by a pre-selector stage employing a common JFET in common source mode. For simplicity I used the air core coils but you can replace them by appropriate replacements in case compactness is more desirable. The variable gang capacitor is an ex- broadcast receiver type that facilitates the receiver peaking on the desired band of operation.

The AF component from the mixer is routed to a two stage AF preamplifier, through a duplexer circuit. The duplexer ensures that the mixer must see a fixed terminating impedance of fifty ohms on its output port, for an entire gamut of frequencies, literally from DC to daylight. The two stage preamplifier is modelled as constant impedance amplifiers, built around low noise transistors. 



A passive AF filter is placed after the first AF amplifier to shape the overall frequency response of the receiver. And as you can see it is the best place to place the filter in AF chain. The main reason of choosing a passive type of filter is that it has more dynamic range than its active counterparts. The inductors used are miniature encapsulated ferrite shielded units, generally available from Digi-Key. The two stage preamplifier is designed to eradicate any trace of fifty Hertz hum, generally considered a menace in most DC receiver designs. The two stage preamplifier provides about 60dB of gain. The following bode plot displays its frequency Vs gain characteristics:


I have built several receivers in past four decades ranging from re-gens, reflex, spontaflex, pentaflex, superjets and many others. I was delighted to build some and disappointed to hear others. One common last thing that can mar the overall performance of an otherwise good receiver design is; of course the final AF amplifier design. Consequently, I opted for a distortion free class-A amplifier employing a very low noise op-amplifier as its major gain block that supplies around 44dB of AF gain. The output of this amplifier is about 100mw and can drive a low impedance small speaker or a set of headphones.


The overall receiver performance is amazingly good. It sounded so well on crowded bands. There was no trace of overloading or broadcast breakthrough on forty meters but on eighty I had to switch in the attenuator on some occasions. The receiver performed as expected and was pleasingly sweet on ears.

Wednesday, 17 August 2016

ZERON -A Super Simple QRP Dual Band Multimode Transceiver. -1

I always admire the versatility of digital chips and use them in many analog designs as well. My curiosity for their implementation in analog circuits threads back to mid nineties, following an article in QEX magazine. Later I used them in many designs and their inexpensiveness and outstanding performance impressed me to the added advantage of compact designs. this resulted in many successful designs including "DIGIRIG" and "NANO" transceivers. After publication of "DIGIRIG" (SPRAT 84 Page 10), late Bill Currie VK3AWC wrote me mentioning that he actually liked this kind of H.F. designs and  he was doing similar sort of experiments.Later, between 1992 and 1999 we worked on many similar designs including receivers, multi-band mixer type VFOs and even weaver's exciter. This  inexpensive diminutive multi-mode, dual band (80/40M) rig is designed  on the similar lines, and is aimed  for the novice amateurs. The schematic diagram shows the transmitter side of the transceiver.


As I already mentioned in one of my previous post that QRP is synonymous with simplicity and minimalism, the present design illustrates it very well. The audio from the electret mic is amplified by a simple single stage amplifier wired around BC548B. This is coupled to a differential AF amplifier wired around a pair of LM 386N, via an audio transformer having 1K:1K+1K impedance taps. This sort of coupling transformer isn't  difficult to find and even digikey  stocks it.

A 74HC4053 functions as a mixer. I used it as it is very cheap, easier to find and requires fewer peripheral components. S1, a double pole three way rotary switch acts as a mode switch that chooses the desired mode of operation. The output of the mixer is amplified by a single IRF530 and is fed to the antenna through an LPF. Too simple to say anything about. Do you still feel the need to burn your bucks to get yourself on the air! In my next post I will discuss the RX and VFO units of this ultra simple project.

Friday, 15 July 2016

PIXET- AN H.F. CW/SSB TRANSMITTER-III

More time is spent on designing the RF power amplifier module than any other part of "PIXET" project. I started with a design based on single ended topology, but later gave up after cooking several IRF510s. Then I moved on to the present design topology for its inherent immunity to harmonics with added benefits of lesser heat and added stability.

Seperate biasing networks for both IRF510s are used. All tranformers are wound over common T50-43 toroids except the PA output transformer. This is wound with three turns each on BN61-202 twin hole large balun core. Mind phasing for the primary winding only. Winding is with 24 SWG copper enamelled wire. 


L2 is wound on a small pignose balun core having 10 turns of 30 SWG. I made it as a choke and a fuse as well but if you wish to use an extra fuse than you can do so. 

Though HEXFETS of IRF series are commonly used in HF power amplifiers but their gate capacitance is always to large and shunts down a considerable drive, especially on 20 meter and above. Thus the performance of HEXFET power amplifiers is usually, noticeably poorer on the 20 meter and above. L3 and L4 serves the purpose to tune this capacitance on desired band of operation. I used about seventeen turns of 26 SWG wound over a lead pencil and then removed as a self standing air core coil. You can either spread or compress the turns to make refinement in performance, or may add or remove turns as required on the band of operation. This just serves a starting point. These coils tunes the gate capacitance on the operting frequency of the band and the effect of input capacitance is thus nulls out, making the amplifier perform better on higher frequency bands.

Friday, 10 June 2016

PIXET- AN H.F. CW/SSB TRANSMITTER-II

In my last post I presented the circuit of the exciter unit of the "PIXET" transmitter, a companion unit for "PIXER" receiver described earlier. In this second part of it I am describing the RF speech processor unit, the second mixer and pre driver circuitry.

Basics of RF speech processing: Imagine a power amplifier designed for 10 W pep driven by a mean ssb signal, which at least will be down by 6 db below the peak. This means a minimum output power of 2.5 watts and a resulting S-meter reading of one step down from the peak. Equalizing the dynamic range of the modulating signal will result in a better effectivity of the power amplifier as this will raise the "mean" output power. Even if this might not be directly visible at the receiver S meter, the compression of the dynamic range will increase the readability and the SNR at the receiving side. In practice, it could be proven that a – moderate - clipping limit of 20 db virtually simulates an 80 watts transmitter while, in reality, the pep output is only 10 watts. Let's understand the root of this philosophy.

The Felcher-Munson Philosophy:  Grokking this theory is a bit beyond my brain right now, but the Fletcher–Munson curves are one of many sets of equal-loudness contours for the human ear, determined experimentally by Harvey Fletcher and Wilden A. Munson, and reported in a 1933 paper entitled "Loudness, its definition, measurement and calculation". The first research on the topic of how the ear hears different frequencies at different levels was conducted by Fletcher and Munson in 1933. In 1937 they created the first equal-loudness curves. Until recently, it was common to see the term 'Fletcher–Munson' used to refer to equal-loudness contours generally, even though a re-determination was carried out by Robinson and Dadson in 1956, which became the basis for an ISO 226 standard.

                              
It is now better to use the generic term "equal-loudness contours", especially as a recent survey by ISO redefined the curves in a new standard. According to the ISO report, the Robinson–Dadson results were the odd one out, differing more from the current standard than did the Fletcher Munson curves. The report states that it is fortunate that the 40-phon Fletcher–Munson curve on which the A-weighting standard was based turns out to have been in agreement with modern determinations. The article also comments on the large differences apparent in the low-frequency region, which remain unexplained. Possible explanations are:

1. The equipment used was not properly calibrated.

2. The criteria used for judging equal loudness at different frequencies had differed.

3. Subjects were not properly rested for days in advance, or were exposed to loud noise in traveling to the tests which tensed the tensor tympani and stapedius muscles controlling low-frequency mechanical coupling.

Thus equal-loudness curves derived using headphones are valid only for the special case of what is called side-presentation, which is not how we normally hear. Real-life sounds arrive as planar wavefronts, if from a reasonably distant source. If the source of sound is directly in front of the listener, then both ears receive equal intensity, but at frequencies above about 1 kHz the sound that enters the ear canal is partially reduced by the masking effect of the head, and also highly dependent on reflection off the pinna (outer ear). Off-centre sounds result in increased head masking at one ear, and subtle changes in the effect of the pinna, especially at the other ear. This combined effect of head-masking and pinna reflection is quantified in a set of curves in three-dimensional space referred to as head-related transfer functions (HRTFs). Frontal presentation is now regarded as preferable when deriving equal-loudness contours, and the latest ISO standard is specifically based on frontal and central presentation.

The A-weighting curve—in widespread use for noise measurement—is said to have been based on the 40-phon Fletcher–Munson curve. However, research in the 1960s demonstrated that determinations of equal-loudness made using pure tones are not directly relevant to our perception of noise. This is because the cochlea in our inner ear analyzes sounds in terms of spectral content, each "hair-cell" responding to a narrow band of frequencies known as a critical band. The high-frequency bands are wider in absolute terms than the low frequency bands, and therefore "collect" proportionately more power from a noise source. However, when more than one critical band is stimulated, the outputs of the brain sum the various bands to produce an impression of loudness. For these reasons Equal-loudness curves derived using noise bands show an upwards tilt above 1 kHz and a downward tilt below 1 kHz when compared to the curves derived using pure tones.

BBC Research conducted listening trials in an attempt to find the best weighting curve and rectifier combination for use when measuring noise in broadcast equipment, examining the various new weighting curves in the context of noise rather than tones, confirming that they were much more valid than A-weighting when attempting to measure the subjective loudness of noise. This work also investigated the response of human hearing to tone-bursts, clicks, pink noise and a variety of other sounds that, because of their brief impulsive nature, do not give the ear and brain sufficient time to respond. 

What does that actually mean: The way to read this graph is as follows: look at the blue curve at the 1 kHz / 40 dB point. Now follow the curve towards the left until you reach 50 Hz on the horizontal axis. You should now read about 70 dB on the vertical axis. In essence, this states that in order for a 50 Hz tone to be perceived as loud as a 1 kHz tone is at 40 dB, it needs to be played at 70 dB. That’s 30 dB difference! A similar thing happens when you move into the high frequencies. A 10 kHz tone needs to be played at about 55 dB to be perceived at the same loudness level. Notice that this difference in loudness evens out as the volume increases (the curves higher up in the figure), for example at 100 dB, the curves have flatten out considerably, meaning the perceived loudness difference between tones at different frequencies decreases. There are two important things to take away from these curves:

1. We are less sensitive to low and high frequencies, we hear mid frequencies more prominently (especially between 1-5 kHz)

2. As the volume increases, this perceived loudness difference between the frequencies diminishes.

However, this made the basis of one of the pioneering developments in low power DX voice communication in which the high amplitude vocals are compressed for an even distribution of power over the usable bandwidth. Based upon this research; in HF-SSB radio technology in the era of late sixties, became a dependable method of modifying the speech waveform in the transmitter to produce a marked improvement in the signal-to-noise ratio at the receiver without also causing any significant increase in distortion products, either in-band or out-of-band. Since RF speech processing was the key to the performance of low-power HF-SSB radio sets - and is now recognized almost as a sine qua non in SSB transmitters - the principles involved will be described briefly. Typically, unprocessed speech has a ratio of instantaneous peak to average power of about 16dB (see Pictures below:

Unprocessed Signal

Processed Signal
In a peak-power-limited system, such as an SSB transmitter, this represents a considerable loss of potential output power, so some means of compressing the dynamic range of the speech signal is required before transmission. It is now well known that clipping (or hard limiting) the peaks of an SSB waveform, and then filtering by a second bandpass filter similar to that in a filter-type SSB generator to remove the resulting harmonic and high-order products, can markedly improve the articulation index of the transmitted signal. Methods of doing this were just being developed around mid sixties. Though modern day transmitters implement compression through DSP techniques using digital algorithms but this project describes a very elegant alternative compressor design like one of the yore.

PIXET Speech processor: The following schematic illustrates the complete circuit diagram of the speech processor and the second X-tal filter:


The SSB signal from the exciter unit is compressed using   diode D1 and the base collector junction of transistor Q1. While developing the clipper circuit I initially employed two back to back diodes. But at such a low level signals adequate level of clipping demands for special hot carrier diodes like HP 5082-2811; which are both expensive and hard to find for an average experimenter. Consequently I zeroed my choice for this simple and effective alternative circuit. Both threshold and gain controls are required to be adjusted carefully. An oscilloscope is quite invaluable tool to do this precious adjustment but in case of its non- availability on the air adjustment also provides convincing results. You can feed the output signal of the transmitter into a dummy load and the adjustments of required gain and proper compression can be done by hearing the signal in a nearby receiver.

Second mixer and pre-driver: There is nothing special to explain in this section. An SBL-1 mixer is used but a home brew variety of double balanced diode mixer will perform equally well. Attempts are made to terminate all mixer ports towel defined 50 ohms impedance to ensure optimum IMD performance. The band pass filter is W7ZOI design which can be scaled to other bands of interest if desired. I tested the transmitter on 14MHz band by feeding VFO signal from 18.43 to 18.78 Mhz. The higher side VFO injection automatically puts you on the right side of sideband i.e. USB. For lower sideband operation as on 80 and 40 meters, lower side VFO injection could be used. Or you can just put up a two banded with single IF just by band switching VFO signal. In that case sideband selection would be automatic for both bands, but the VFO tuning will be in opposite direction. This will help a newbie to assemble a two bander with minimum effort and cost.



In the next post I will describe the R.F. linear  amplifier for the "PIXET" transmitter.

Monday, 23 May 2016

PIXET- AN H.F. CW/SSB TRANSMITTER-I

It all began with the nonavailability of expensive mixer ICs. I homebrewed my own mixer IC (parts shown in dotted square), and as I had already described it in my blog post of 9th February. Keen  to having made the "PIXER" superhet receiver and other projects based on it...........finally I started to build a DSB rig around it but over the last couple of weeks, design evolved with a bit of tinkring here and there and ultimately it took shape of a very nice companian H.F. CW/SSB transmitter to "PIXER"; having a clean spectral purity and a a side band suppression of around 58 dB. Indeed great for a thing developed around common off the shelf components and right on your kitchen table......and all that with a home brewed X-tal filter!!! The schematic of the exciter unit is as under:

The circuit of PIXET exciter is quite simple in itself and requires no description. I have used components that were available on hand. Especially the transistors. You can substitute them with any general purpose medium power ones having adequate gain bandwidth product. Though matched pair of transistors has been used in the balanced mixer but R6 has been included for refining the circuit balance. It can be carefully adjusted for minimum carrier during no audio signal present at the microphone input. A simple diode gate provides the CW keying. Could there be anything simpler than that....? And that's the real essence of QRP home brewing.....!!!

Sunday, 24 April 2016

PIXER- An empirical H.F. Superhet Receiver.-II

In the last post I described the IF and product detector of an empirical superhet receiver developed around home made ICs as described in my post of 9th February (parts shown in dotted squares). In this post I am going to describe the front end design of this receiver.

Initially, I decided to wire the front end using another home brewed mixer IC. But out of an amateur's true spirit I decided to do experiment a bit more. After a receiver's sensitivity, the next requirement is its ability to discern weak signals in the presence of strong signals in its pass-band. This is known as dynamic range of the receiver.

There are several types of dynamic range. The first one, and probably the easiest to understand-"AGC range"-concerns whether a receiver is capable of maintaining a constant audio output level over a large input-signal amplitude range. The traditional school of thought requires AGC action to commence at about 3µV, leading to a condition where signals that produce an excellent signal-to-noise ratio may show absolutely no S-meter indication-a most undesirable effect. The reason for this is inappropriate receiver gain distribution-generally, a lack of gain at the IF. Maintaining constant audio output must involve gain control at the receiver's IF, and possibly even at its input.

IMD Dynamic Range:  The output of a linear stage tracks the input signal decibel by decibel, with every 1-dB change in its input signal corresponding to an identical 1-dB output change. This is the stage's first-order response. Because no device is perfectly linear, however, two or more signals applied to it intermodulate to some degree, generating sum and difference frequencies. These intermodulation distortion (IMD) products occur at frequencies and amplitudes that depend on the order of the IMD response as follows:

•Second-order IMD products change 2 dB for every decibel of input-signal change, and appear at frequencies that result from the simple addition and subtraction of input-signal frequencies. For example, assuming that its input bandwidth is sufficient to pass them, an amplifier subjected to signals at 6 and 8 MHz will produce second-order IMD products at 2 MHz (8 - 6) and 14 MHz (8 + 6).

•Third-order IMD products change 3 dB for every decibel of input-signal change, and appear at frequencies corresponding to the sums and differences of twice one signal's frequency plus or minus the frequency of another. Assuming that its input bandwidth is sufficient to pass them, an amplifier subjected to signals at 14.02 MHz (f1) and 14.04 MHz (f2) produces third-order IMD products at 14.00 (2f1 - f2), 14.06 (2f2 - f1), 42.08 (2f1 + f2) and 42.10 (2f2 + f1) MHz. The subtractive products (the 14.00 and 14.06-MHz products in this example) are close to the desired signal and can cause significant interference. This is why our receivers' third-order IMD performance is so important. It can be seen that the IMD order determines how rapidly IMD products change level per unit change of input level. Nth-order IMD products therefore change by n dB for every decibel of input-level change. IMD products at orders higher than three can and do occur in communication systems, but the second- and third-order products are most important in receiver front ends.

Intercept Point: The second type of dynamic range concerns the receiver's intercept point, sometimes simply referred to as input intercept. Intercept point is typically measured by applying two or three closely spaced signals to the antenna input, tuning the receiver to count the number of resulting spurious responses, and measuring their level relative to the input signal.

Because a device's IMD products increase more rapidly than its desired output as the input level rises, it might seem that steadily increasing the level of multiple signals applied to an amplifier would eventually result in equal desired-signal and IMD levels at the amplifier output. Real devices are incapable of doing this, however. At some point, every device overloads, and changes in its output level no longer equally track changes at its input. The device is then said to be operating in compression. Pushing the process to its limit ultimately leads to saturation, at which point input-signal increases no longer increase the output level.

The power level at which a device's second-order IMD products equal its first-order output (a point that must be extrapolated because the device is in compression by this point) is its second-order intercept point. Likewise, its third-order intercept point is the power level at which third-order responses equal the desired signal. The following figure represents these relationships:

A linear stage's output tracks its input decibel by decibel on a 1:1 slope-its first-order response. Second-order intermodulation distortion (IMD) products produced by two equal-level input signals ("tones") rise on a 2:1 slope-2 dB for every 1 dB of input increase. Third-order IMD products likewise increase 3 dB for every 1 dB of increase in two equal tones. For each IMD order n, there is a corresponding intercept point IPn at which the stage's first-order and nth order products are equal in amplitude. The first order output of real amplifiers and mixers falls off (the device overloads and goes into compression) before IMD products can intercept it, but intercept point is nonetheless a useful, valid concept for comparing radio system performance. The higher an amplifier or mixer's intercept point, the stronger the input signals it can handle without overloading. The input and output powers shown are for purposes of example; every receiver exhibits its own particular IMD profile.

Input filtering can improve second-order intercept point; device non-linearities determine the third, fifth and higher-odd number intercept points. In pre-amplifiers, third-order intercept point is directly related to dc input power; in mixers, to the local-oscillator power applied.

Intercept point can be confusing because it can be specified in terms of input or output power. Intercept point should be referred to device output because that's where the trouble occurs, but input intercept is commonly given. Therefore, if an amplifier or a mixer has a particular intercept point-let's say +30 dBm at 10 dB gain-and then its gain is increased by an additional 10 dB, its dynamic range decreases by the amount of the gain.

Thus the first requirement for a receiver's front-end to have a good dynamic range, is a good mixer. My choice thus zeroed on the simple diode ring mixer which already has gained popularity among amateur fraternity. Double balanced mixers are a form of what is termed a "reversing switch mixer." Reversing switch mixers operate by using electronic switches in a bridge formation to reverse the input RF signal under the action of the local oscillator used as a square wave switching signal. They normally offer significant advantages over analogue mixers for radio communications and general RF design applications as they are able to offer better levels of dynamic range and noise. In view of this fact, they are normally used in high performance applications where noise and dynamic range are of importance - e.g. in the front end of a radio receiver or spectrum analyzer.

Although there are comparatively few components in a double balanced mixer, their individual performance is crucial to the performance of the RF mixer as a whole. Normally Schottky barrier diodes are used for the diode ring. They offer a low on resistance and they also have a good high frequency response. Ordinary signal diodes may be used for low performance applications, although the cost difference is small. It is found that the diode forward voltage drop for the diodes determines the optimum local oscillator drive level. RF mixers requiring to handle a high RF input level will need a correspondingly high LO input level. As a rule of thumb the LO signal level should be a minimum of 20dB higher than either the RF or IF signals. This ensures that the LO signal rather than the RF or IF signals switch the RF mixer, and this is a key element in reducing intermodulation distortion, IMD, and also maximising the dynamic range.

To increase the required drive level, it is possible to place multiple diodes in each leg. The most common LO drive level for a double balanced mixer is probably +7dBm. However they can be obtained with a variety of drive levels. Values of 0, +3, +7, +10, +13, +17, +23, and +27 dBm are normally used.

I decided to use a home brew mixer using common inexpensive diodes in the front-end, the schematic is as under:

The input signals is passed through a double tuned band pass circuit made around slug tuned, self wound inductors. I included a low pass circuit owing to ring mixers ability to respond to strong, harmonic signals. The given values are chosen for forty meter band, but they can easily be scaled for other frequency bands, too. Or even a multi band operation is also possible using suitable band switching. The home made mixer uses four matched diodes and two RF transformers wound on pig-nose balun cores. Transformers have thirteen trifilars turns. I used inexpensive diodes in the mixer but they behaved extremly well. Any 2.5MHz VFO capable of delivering reasonable power can be used. Amateur literature is already full of several circuits. For a good input intercept to be maintained it is important to properly terminate all mixer ports. IF port of the mixer is terminated in a post mixer amplifier that terminates the mixer output to an appropriate impedance, required to maintain a good IP3. The IF signal is then routed to a crystal filter through a post mixer amplifier, as shown below:


The post mixer amplifier uses noiseless inductive feedback. I used 2N3866 as it was available, but 2N4427, BFW16A etc. will seem to work equally well. Keep the transistor leads as short as possible and try to use ferrite beads in the collector lead to avoid spurs. I used a variable bandwidth X-Tal filter whose bandwidth can be controlled with R12. Cheap color burst X-Tals of 4.43 MHz are used for the x-tal filter. On twenty meters and above a low noise amplifier ahead of mixer is recommended to achive optimal noise figure.

The overall performance is amazingly good and despite designed around common off the shelf type inexpensive components it performs really well, far better than many commercial receivers.