Monday, 8 April 2019


“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.


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.

(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. 


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:


  1. Fantastic, Mr Kang! Looking forward to further posts and the kit!
    I have always enjoyed the posts on your website. We met for a few moments at LARC 19.

  2. Hi OM Sunil, thanks for your interest and liking my work. For reserving a kit you can drop me a line on my mail ID provided above. 73s de Kang.

  3. Congrats..!
    An AMAZING use of an otherwise overlooked I.C. for this task. I´m beginning to build an equipment for SSB at these levels of power and complexity, and your nice job is really inspiring to me.
    Kind regards and 73 de LU4EYW.-

  4. Is there a PCB available for this design?

    1. Yes PCB and IC is available. Please message me on my email ID given at the end of article.

  5. Fantastic, Mr Kang!One year back I tried with TBA120s IC but Ihave to failed.Input signal diminished at the output due to large number of limitter section in TBA120s.Now with the MC3362 has moderate numbers of limitter sections. Another ic similar to mc3362 is mc13135 which has one opamp section which will be used as for mic amplifier. For simplifying the most of circuits By using two ic of mc13135.