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.
(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.
