ic-756proII Technical Report
ic-756proII Technical Report
CONTENTS
11. Profile . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3 16. Connection to option/peripheral units . . . . . . . . . . . . . . . . . 25
12. Features . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4 6-1 ACC Sockets
13. Front and rear panel . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 7 6-2 HF/50MHz, 1kW linear amplifier
3-1 Front panel 6-3 Interface for digital mode
3-2 Rear panel 6-4 External control unit for voice memory keyer
14. What is DSP in radio Communication? . . . . . . . . . . . . . . . . 10 6-5 Installation of UT-102 optional Voice Synthesizer Unit
15. Circuit description . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 12 17. CI-V control . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 28
5-1 Digital IF filter 7-1 Remote jack
5-2 Digital functions 7-2 Data format of CI-V
5-3 PSN modulation 7-3 List of commands
5-4 Manual notch 18. Inside Views . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 32
5-5 Speech compressor 19. Options . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 33
5-6 Microphone equalizer 10. Specifications . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 34
5-7 RTTY demodulator 10-1 General
5-8 Receiver 10-2 Transmitter
5-9 Transmitter 10-3 Receiver
5-10 Dual-watch function 10-4 Antenna tuner
5-11 Real-time spectrum scope 11. Block diagram . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . foldout
5-12 Voice record/playback function
5-13 PLL circuit
2
The IC-756PROII is the high performance HF transceiver of choice for today’s discriminating amateur
radio operator. Icom’s engineers took cutting-edge digital technology and paired it with Icom’s
extensive experience with analog technology. The result is a major advancement of Icom’s original
digital IF filter which, in the earlier IC-756PRO model, enjoyed a great reputation around the world.
The IC-756PROII uses the same 32-bit floating point DSP and a 24-bit A/D-D/A converter as the IC-
756PRO. It is now possible to execute the digital IF filter, noise reduction and the digital IF filter in the
AGC loop processing, and to select the soft/sharp filter shapes. The IC-756PROII employs exclusive
DSP/analog circuit matching to further improve receiver performance.
Icom’s engineers analyzed the influence of the AGC loop upon the received audio, matching it to an
analog circuit suitable for the dynamic range of the A/D converter and the other parts used, and also
re-examined the core stage of the receiver (ranging from RF top to mixer circuit), to distribute the
mixer levels properly. As a result, the matching of digital and analog technology has attained a level
never before achieved.
This technical report does not explain in depth all the digital engineering with its many calculations and
formulas. Instead, it focuses on the DSP engineering in an easy-to-understand manner. This report
also explains why the 32-bit floating point DSP and 24-bit A/D-D/A converter are included. The
dynamic range of the 32-bit floating point DSP and the 24-bit A/D-D/A converter may seem to be an
over specification for amateur radio. But this is not the case. This technical report helps clarify these
points.
It is Icom’s hope that in providing you with this report you will discover the IC-756PROII’s many digital
advantages. Enjoy!
3
2. Features
32-bit floating point DSP and 24-bit A/D-D/A ■ Manual notch with superior attenuation level
The IC-756PROII’s manual notch filter has extremely sharp
converter characteristics for processing in the DSP and provides tremendous
The adoption of a 32-bit floating point DSP and 24-bit A/D-D/A performance for attenuation levels >70dB. Analog notch circuits are
converter in the IF stage (36kHz) was originally developed by susceptible to fluctuations in attenuation or changes in temperature.
Icom. It enables various digital functions which amateur radio The DSP-based manual notch provides stable performance and is
operator’s desire. not susceptible to such changes. Also, the DSP signal processing
executed within the AGC loop completely shuts off undesirable
■ 51 types of digital IF filtering signals, even with the AGC set to high speed. An automatic notch is
The digital IF filter has superior filtering performance and a included to further enhance receiver performance.
distinguished shaping factor that demonstrates the power of 32-bit
floating point DSP. The digital filter is completely free from ■ Demodulator/decoder for RTTY
deterioration due to deviations in band characteristics, temperature This transceiver is equipped with a demodulator and a decoder for
change, or mechanical vibration, all of which have been observed in BAUDOT RTTY as a standard feature. On-air station calls may be
analog filters. It also provides excellent ripple characteristics that recognized instantaneously by reading the received RTTY message
have never been available with analog filters. The passband directly on the IC-756PROII’s LCD – no personal computer or
(50Hz–3.6kHz) of the digital IF filters used for IC-756PROII come in external components are required. The transceiver is fitted with an
51 types. This function allows 3 of these 51 types to be pre-set for on-screen tuning indicator that allows the RTTY to be fine tuned with
each mode and to be changed instantaneously by using the filter ease. A DSP based twin-peak audio
button, depending on the situation. filter further improves the S/N ratio.
This filter will reduce interference that
■ 2 types of filter shape (Sharp/Soft) appears between each tone (mark
Changing the IF filter shape is a feature that is not available with and space), which cannot be removed
analog radios. Select the filter shape from two types, Sharp and Soft, by conventional analog filters. This
depending on the purpose, operating band, band conditions, etc. It is twin-peak audio filter works to capture
possible to set CW and SSB filter characteristics independently and noise-level signals accurately and to
also select the filter shape while actually receiving a signal. significantly reduce the generation of
noise distortion. RTTY reception screen
■ Digital IF filter in AGC loop
The digital IF filter, manual notch filter, etc. are located in the AGC ■ New-generation speech compressor
loop, using DSP, which completely eliminates interference from The DSP based speech compressor enhances the readability of your
adjacent strong signals. This allows the AGC to be operated only on transmitted signal at a receiving station without any distortion, even
the selected frequency. It is also possible to pre-set the operation of when the compression is set to a high level. The gradation of voice
AGC in each mode in accordance with 13 types of time constants. processing is extremely close to the original sound. This assures
superior sound quality at all compression levels.
455 36
kHz kHz Speech compressor on Speech compressor off
ADC DET DAC AF
Manual IF
3Lo NOTCH Filter AGC
100W 100W
AGC
DAC DSP
The IC-756PROII is equipped with DSP based twin PBT filtering. It ■ Microphone equalizer (enables 121 different settings)
provides excellent performance on a completely different level than The IC-756PROII is equipped with a microphone equalizer that makes it
analog filters. Set the frequency, and then adjust the passband width possible to set the frequency characteristics of the transmitted signal in
of the received signal in steps of 50Hz using the dedicated two- 11 different stages for both the high-tone range and the low-tone range.
position knob. The passband width and direction of shift may be Considering all permutations, this provides for a total of 121 different
graphically displayed on the LCD, if the operator chooses. settings. With this flexibility of DSP based waveform shaping, it is
possible to adjust transmit audio quality depending on the application.
■ High-accuracy digital modulation and demodulation in all For example, it is possible to set the dynamic sound quality for “Pileup”
modes or to set pleasant sound for “Ragchewing”.
The DSP unit allows you to increase transmit/receive audio levels,
modulation, and demodulation — even while decoding RTTY. This ■ Variable level type noise reduction
makes it possible to set the passband width of the IF filter for SSB The 32-bit floating point DSP has excellent calculation performance,
transmit to 3 different stages. The DSP unit also provides a which processes complex and sophisticated algorithms. This allows
demodulation level suitable for high-grade HF performance and high- the DSP to attenuate noise without delay and extracts noise-level
fidelity sound. signals. It is possible to vary the suppression level in 16 stages.
4
Enhanced functions The best in operating convenience and features
■ Advanced receive functions ■ Dual-watch
The RF stage’s front-end receive mixer is designed in a 4-element Dual-watch enables simultaneous two-frequency receive in the same
configuration. In the IC-756PROII, this configuration is used in the band, providing identical band and filter configurations in both receive
BPF stage at the RF top. Each element is examined to tune the systems. This makes it possible to receive two signals simultaneously
circuit after RF stage to mixer, which makes it possible to enhance as if two separate receivers are being used. This greatly enhances
the receive performance. This significantly reduces 3rd and other split frequency operation; enjoy enhanced DX-operation by searching
order distortion and provides a wide dynamic range. This means the for pickup frequencies while watching the transmit frequency of a DX
IC-756PROII will accurately capture weak signals that analog type station experiencing pile-up. Or have a QSO while simultaneously
radios cannot hear, even in low bands with high noise levels. monitoring a DX net.
5
■ Quick split function ■ Digital meter simultaneously displays 4 transmit level indicators
When the split button is pressed and held, the frequency of the sub- With the digital meter (including peak-hold), it is possible to confirm
VFO is adjusted to the frequency in the main VFO. Using the split the output power, ALC, SWR, and COMP, all at the same time while
function, it is also possible to control the following: transmitting. The signal strength is also displayed while receiving.
1. Vary the transmit frequency via the main dial.
2. Direct entry of the designated frequency.
3. Direct entry of the shift frequency. Enhanced functions
You are now ready to “bag the DX” while other operators are still
■ Antenna system
tuning up.
· High-speed built-in auto antenna tuner covering up to the 50MHz
band
■ Preamplifier and attenuator
· 2-piece antenna terminal (incorporated with auto antenna selector)
The IC-756PROII incorporates two types of receive preamplifiers:
· Dedicated receive antenna connector
Preamplifier 1 (10dB) emphasizes modulation across all bands, and
preamplifier 2 (16dB) emphasizes sensitivity especially for high
■ Receive system
bands. The attenuator is selectable in three stages, 6, 12 and 18dB.
· General coverage receive (30kHz to 60MHz)
When there is a strong signal from a local commercial station it
· Control of RF gain and squelch with one knob
becomes possible to control the generation of distortion at the RF
stage of the receiver. It is also possible to retain the preamplifier and
■ Transmit system
attenuator settings for each band.
· IF Monitor function allows the transmissions of your station to be
listened to locally
■ Variable noise blanker
· Built-in 50 frequencies of tone encoder/decoder
The transceiver uses a new noise blanker design that provides
· VOX function allows the automatic selection of transmit and receive
significant reduction of pulse-type noise. The noise blanker also
for “hands free” operation
greatly enhances weak signal copy, allowing the operator to change
· All-mode power control function
the sensitivity in 100 stages in accordance with the noise level
without distorting the target signal.
■ CW system
· CW pitch control function allows the CW receive tone to be set to a
■ Frequency shift function for change from/to SSB to/from CW
desired frequency (300 to 900Hz continuously)
A frequency shift function automatically adjusts the CW carrier point
· Double key jack allows 2 types of keys to be connected
when selecting from SSB mode, or vice versa. You may select “Shift
simultaneously
function off” whereby the frequency remains the same (by moving the
· Full break-in function allows receive during a break while keying
carrier point), or “Shift function on” in which the frequency is shifted
without moving the carrier point. Using CW-R mode it is possible to
■ Operation system
set the carrier point to USB.
· 5-channel memo pad saves frequency and mode
(It is also possible to change the 5-channel memo pad to a 10-
channel type.)
Enhanced TFT color display · RIT and TX variable up to ± 9.999kHz
■ High visibility · 1Hz pitch tuning
A high visibility 5-inch TFT color display has been integrated into the · Optional frequency speech allows the S-meter level to be
IC-756PROII to provide ease of use and clear indication of the radio’s announced
many features. Various function settings such as frequency, memory · High visibility needle type white-tone analog meter
frequency, comment, filter setting status, RTTY tuning indicator, and · Various scanning functions (program, memory, select memory, F)
more are displayed in the upper portion of the display, The lower · Auto Tuning Step
portion of the display gives voice memory, characters of received · Dial-lock
RTTY, and the real-time spectrum scope information. The display · Split frequency lock
color may be selected from 8 types, from vivid color to muted grays. 7 · Torque adjustment mechanism for main dial
different font types may also be selected. These settings may be · Band edge beep function
made in any combination – customize your display to best suit your · CI-V terminal allows control from a personal computer
personality or favored operating set-up. · Clock/timer function
· AH-4 control circuit
■ Memory channel/memory list
The transceiver is equipped with 99 regular memory channels and 2
scan edges, totaling 101 channels. It is possible to enter text of up to
10 characters in each memory channel. It is also possible to display a
list of up to 13 memories at a time.
6
3. Front and rear panel
3-1 Front panel
A D E
B C F
C
LCD FUNCTION SWITCHES [F1]-[F5]
MODE SWITCHES
FILTER SWITCH [FILTER]
EXIT/SET SWITCH [EXIT/SET]
7
D
MULTI-FUNCTION SWITCH GUIDE
LCD FUNCTION DISPLAY
KEYPAD
E MEMORY UP/DOWN SWITCHES [ ][ ]
MEMORY WRITE SWITCH [MW]
MEMORY CLEAR SWITCH [M-CL]
QUICK TUNING SWITCH [TS]
TRANSMIT FREQUENCY CHECK SWITCH [XFC]
MEMO PAD-READ SWITCH [MP-R]
MEMO PAD-WRITE SWITCH [MP-W]
MAIN/SUB CHANGE SWITCH [CHANGE]
VFO/MEMORY SWITCH [VFO/MEMO]
MAIN/SUB CHANGE SWITCH [CHANGE]
DUALWATCH SWITCH [DUALWATCH]
SPLIT SWITCH [SPLIT]
8
G PASSBAND TUNING CONTROLS [TWIN PBT]
PBT CLEAR SWITCH [PBT CLR]
NOTCH SWITCH [NOTCH]
MANUAL NOTCH FILTER CONTROL [NOTCH]
CW PITCH CONTROL [CW PITCH]
TX SWITCH [ TX]
RIT SWITCH [RIT]
RIT/ TX CONTROL [RIT/ TX]
CLEAR SWITCH [CLEAR]
ANT2 ANT1
9
4. What is DSP in radio communication?
The term DSP stands for “digital signal processor”. When DSP is Background to development of the first-generation
used in a communication unit, the electrical signal processes IC-756PRO
(amplification, filtering mixer, modulation, demodulation, etc.) are From the early stages of research into DSP transceivers Icom has
handled by the DSP. Such signal processing, using numerical been on the forefront of shifting IF filter design from analog to digital
calculations, is called “digital signal processing”. type filters. To put the digital IF filter to practical use it was necessary
Digital signal processing assures the same results every time to incorporate the digital IF filter into the AGC loop. It was also
providing for the characteristics defined in the design phase. When necessary to provide AGC processing using the DSP. To achieve this
digital signal processing is utilized, it is not necessary to take the there remained a lot of technical problems to be resolved.
adjustment deviations of the conventional analog circuit into
consideration. These deviations are caused by variations in In the initial stage of research, it was not possible to complete the
component characteristics, temperature change, or deterioration over DSP, A/D, and D/A devices in a radio unit at a practical cost, as
time. It is also possible to perform complicated processing tasks such shortage in device capacity was a significant factor. Icom conducted
as Fourier transformation, adaptive control, special function research into digital PSN modulation, noise reduction, automatic
processing (*1), and more. Such complicated processing tasks are notch, and audio peak filter, while also proceeding with research into
very difficult and costly for a conventional analog circuit. digital IF filter processing and digital AGC processing. This research
*1 Special function processing: Trigonometric function, inverse function of trigonometric
includes the ultra-narrow filter for CW that allows the advantages of
function, square root, logarithmic function, exponential function, etc.
DSP to be fully utilized for commercialization of a DSP transceiver.
As the first devices were developed with improved capacity, Icom
Digital signal processing is also widely used in fields other than radio
started full-scale research into integrating the digital IF filter
communication units, such as:
processing (*2) and digital AGC processing in practical applications.
· Modems for telephone circuits *2 Advantages of digital IF filter processing:
· Surround-sound effects (stereo sets, stereo components) Since a digital IF filter is free from deterioration due to passband width deviations,
· Echo canceller (telephone) temperature changes, change in mechanical strength, etc., the changes seen in an
· Voice compression/coding (cellular phones) analog filter will not occur. It will not deteriorate through years of use and will provide
excellent ripple characteristics that are not possible with analog filters.
10
The A/D converter used for the IC-756PROII is a super-high If the receive frequency is substituted for “fRX”, the input frequency for
performance A/D converter that is also used in digital mixers for “f 1 ” and the input frequency for “f 2 ” respectively, the following
recording studios and provides an actual analog performance value relationship is established for 3rd order distortion component.
of 120dB. To bring this performance to an optimum level it is
necessary to execute calibration for 10 seconds after powering on. f1 × 2 ± f2 = fRX, or
The wait time when IC-756PROII is started is allotted to the f1 ± f2 × 2 = fRX
calibration operation.
If there are inputs of 14.2MHz and 14.3MHz while 14.1MHz is being
To execute the processing of data sampled by the 24-bit A/D received, the distortion component is heard at 14.1MHz. The relative
converter it is necessary to obtain 24-bit calculation accuracy. Since value of the input level when the signal can be heard at 14.1MHz and
the dynamic range is decreased substantially due to the scaling the level of the signal received at the essential receive frequency is
operation (*5) for the accumulation of calculation errors or digital filter called the “dynamic range”.
processing, Icom felt the 24-bit fixed decimal point DSP would
provide insufficient calculation accuracy. Figure 1 shows an example in which the following are plotted on the
*5 Scaling same axis.
For digital filter processing, a frequency which causes the gain to increase may exist at
the intermediate stage of processing even if the filter used provides a passband gain of
· Input/output characteristics at receive frequency, or the
0dB. For a fixed decimal point, DSP the calculation is executed with the gain decreased characteristic data (a) for a case when the receive frequency
in advance so as not to allow an overflow to occur due to a signal of that frequency. component input from the ANT is detected and output as a low
This gain adjustment operation is called “scaling”. frequency signal
· Input level of frequency component (generating 3rd order distortion
Since the level of scaling required is also increased to provide an IF from the receive frequency) and level (b) at which the distortion can
filter with a sharp shape factor the calculation accuracy is liable to be be heard at the receive frequency.
decreased, even if double-precision (32-bit fixed decimal point) The difference in level at which (a) and (b) above can be heard is
calculation is executed when using a high-speed 16-bit DSP. To the dynamic range.
provide both the digital IF filter processing and digital AGC The level at intersecting point between (a) and (b) above is called
processing using DSP, Icom determined it was necessary to use a IP3 (3rd order intercept point).
32-bit floating point DSP.
If these numerical values are large, it can be said that signal
For a 32-bit floating point DSP, the numerical data within the DSP is processing is executed without distortion. When the numerical values
adjusted automatically according to the size of the numerical value. are small, a frequency component that does not exist in the essential
Consequently, errors generated due to calculation are extremely receive frequency is heard and distortion will be generated.
limited and the influence of calculation errors is almost negligible.
Because it is not necessary to consider the overflow during Dynamic range for A/D converter
calculation, the dynamic range will not be decreased due to the Consider the dynamic range (used as an index for the performance of
scaling operation. an A/D converter) as the ratio between maximum value and minimum
value to be treated by the A/D converter. If the maximum resolution
The 32-bit floating point DSP and 24-bit A/D-D/A converter use a for one bit is “Vmax” in the case of a 16-bit A/D converter, the
signal processing algorithm (newly developed to demonstrate its following is given:
performance) in combination for the reasons above, which make it
possible to provide highly accurate digital IF filter processing and Vmax ÷ 216 = Vmax ÷ 65536
digital AGC processing. These new functions (FM demodulation, AM
modulation/demodulation, RTTY modulator, etc.) were incorporated In other words, the change in level for one bit is 1/65536 of Vmax.
in the IC-756PRO to make it an IF DSP radio. This value seems to be an extremely small value, in decibels it will be
as follows:
Two Dynamic ranges
20log (1/65536) = –96.33dB
Dynamic range as RF performance
Fig. 1 This means that an S/N ratio of over 96dB is never allowed for
[dB] 200 transmit. The minimum resolution of signal the A/D converter can
180 treat is affected by its specifications, which are 24 bits and 144dB
160
logical value. Some may say that a transceiver is not a high-grade
140
120
Output audio system and therefore does not require a specification of 144dB,
100 or that a specification of 96dB is sufficient; however this value is not
(a) 3rdIMD
80 an over specification. If there is no AGC in the DSP and the input
60 level of the A/D converter is properly controlled by the analog circuit
(b)
40
AGC, the specification of 96dB will be sufficient (the IC-775DSP uses
20
0
this system). When the A/D converter is in the AGC loop, the input
–140 –120 –100 –80 –60 –40 –20 0 10 20 30 40 50 [dBm] level of the A/D converter may fluctuate significantly. For this reason,
–130 –110 –90 –70 –50 –30 –10
the gain control by AGC within DSP requires at least the dynamic
Dynamic IP3 range of the A/D converter.
“Dynamic range” from the viewpoint of RF performance indicates to
what extent the distortion component (generated due to the
frequency of a signal) can be heard at the receive frequency when a
frequency component different from two receive frequencies is input.
“Dynamic range” generally means the value by 3rd order distortion
component.
11
5. Circuit description
When using a digital IF filter the beat frequency of an unwanted
5-1 Digital IF filter
adjacent signal moves out of the filter passband width, which will not
cause interference. (Fig. 1.2) This is the greatest difference between
For IC-756PROII the transmit/receive passband width in all modes is
an analog IF filter and a digital IF filter. During “pile-ups”, such as
determined by the digital IF filter using DSP. A filter of this type
those that occur in DX’peditions, contests, etc., it is possible to make
provides an ideal shape factor that cannot be achieved by an analog
a proper selection suitable to the application by selecting the broad
filter. If an attempt is made to increase the shape factor and band
filter shape (SOFT).
ripple characteristics of an analog filter, it is necessary to increase the
number of crystal components (or ceramic elements), which may
5-1-1 CW sharp filter
result in physical restrictions. A digital IF filter using DSP assures the
The digital IF filter offers an ideal shape factor which has never been
desired characteristics by overlaying multiple filters. This is governed
available with conventional analog filters. It enables a greater ability
only by the processing volume of the software and it is possible to
to receive weak stations that may lie behind radio interference. This
overlay such filters with any number of stages.
is the filter shape that Icom would suggest to the DX hunter due to its
756PROII SSB COLINS10 superior cut-off performance. The cut-off performance is of a level to
actually extend the CW band as explained above.
[dB] 10
–0
CW sharp filter characteristic
–10
–20 CW Filter (BW 50/100/150/200/250/300/350/400/450/500Hz)
Response
–30 [dB] 10
–40
0
–50
–60 –10
–70 –20
00 1500 1000 –500 0 500 1000 1500 2000 [Hz]
–20 – – –30
Input frequency
–40
The diagram shows a graph of receive selectivity when the IC-
–50
756PROII is set to the SSB BW mode of 2.4kHz as well as the
–60
selectivity characteristic of each Collins 10-pole mechanical filter. The
digital IF filter of the IC-756PROII is of a design equivalent to a 14- –70
pole filter. The filter serves to cut the undesired adjacent signals –80
sharply under any circumstances using the superior shape factor
–90
(sharp/soft) and 51 types of variable passband width provided by IF
–100
stage processing using the DSP. When viewing a received CW signal 0 50 100 150 200 250 300 350 400 450 500
the difference between the cut-off performance of this filter and that [Hz]
of an analog IF filter is evident.
In a transceiver equipped with a conventional analog IF filter the beat
5-1-2 CW soft filter
frequency of an adjacent signal is present when the CW signal is
The skirt characteristics of the soft filter are broadened so that the
received resulting in interference. The beat frequency is contained in
listening level of the filtered signal is the same level as that of a
the skirt of the filter even if it is out of the set band range. (Fig. 1.1)
conventional analog filter. When using the radio for DX’pedition the
[dB] 30
filter is recommended for “pile-up” operation and is most suitable for
Fig. 1.1 20 the CW DX’peditioner and CW contestant.
10
0
–10 CW soft filter characteristics
Conventional –20
analog filter –30 Beat frequency CW Filter (BW 50/100/150/200/250/300/350/400/450/500Hz)
–40
The signal out of setting –50 [dB] 10
band is also heard as –60
–70 0
CW reception sound.
–80
–90 –10
–100
–800 –600 –400 –200 0 200 400 600 800 –20
[Hz]
500Hz setting –30
[dB] 30 –40
Fig. 1.2 20
–50
10
–10 –60
Digital filter of
–20
IC-756PROII –30
–70
–40 Beat frequency –80
The signal out of setting –50
band is not heard as –60
–90
CW reception sound. –70
–80 –100
–90 0 50 100 150 200 250 300 350 400 450 500
–100
–800 –600 –400 –200 0 200 400 600 800 [Hz]
[Hz]
500Hz setting
12
5-1-3 SSB sharp filter AM Filter Characteristic
This filter creates an ideal shape factor and in-band flatness, and AM Filter (BW 3/6/9kHz)
makes it possible to cut out-of-band signals while reproducing the in- [dB] 20
band signal, without deteriorating sound quality. This filter shape is
most suited for situations which emphasize ragchewing and receive 0
sound quality.
–20
–100
–40
–60 –120
–10 –8 –6 –4 –2 0 2 4 6 8 10
–80 [kHz]
–100
FM Filter Characteristic
–120
FM Filter (BW 7/10/15kHz)
–140
[dB] 0
–160
–4000 –3000 –2000 –1000 0 1000 2000 3000 4000 5000 6000
[Hz]
–20
–20
Digital IF filter transmission band (51 types)
–40 Application mode FILTER Standard values Setting range (step width)
–60 FIL1 3.0kHz
50–500Hz (50Hz) /
–80 SSB FIL2 2.4kHz
600–3.6kHz (100Hz)
–100 FIL3 1.8kHz
–120 FIL1 1.2kHz
SSB•D 50–500Hz (50Hz) /
–140 FIL2 500Hz
CW 600–3.6kHz (100Hz)
–160 FIL3 250Hz
–4000 –3000 –2000 –1000 0 1000 2000 3000 4000 5000 6000
[Hz] FIL1 2.4kHz
50–500Hz (50Hz) /
5-1-5 Other digital filters RTTY FIL2 500Hz
600–2.7kHz (100Hz)
FIL3 250Hz
RTTY filter characteristics FIL1 9.0kHz
RTTY Filter (BW 250/300/350/500/1000Hz)
[dB] 10
AM FIL2 6.0kHz –
0 FIL3 3.0kHz
–10 FIL1 15kHz
–20
FM FIL2 10kHz –
–30
FIL3 7.0kHz
–40
–50
–60
–70
–80
–90
–100
–100 0 100 200 300 400 500 600 700 800 900 1000
[Hz]
13
remove interference correctly even when more than two tones occur.
5-2 Digital functions
As the tone frequency changes the interference is followed and
5-2-1 Noise reduction, automatic notch removed automatically. Since the characteristics are adjusted to
An adaptive filter made up of an FIR filter and LMS algorithm as minimize the influence upon a voice, it can be used in SSB mode
shown in Fig. 2 is used to provide the basic configurations of noise without any sense of incongruity even if automatic notch is turned on
reduction and automatic notch. This adaptive filter (*5) separates the all the time.
target signal and noise, the correlation of separation parameters, and
controls the coefficient of the FIR filter with the LMS adaptive
5-3 PSN modulation
algorithm to minimize the error between the output of the FIR filter
and the reference signal.
*5 Adaptive filter The IC-756PROII adopts a digital PSN modulation system for SSB
This type of filter is called an “adaptive filter” since the filter characteristics are changed by modulation processing to provide superior band characteristics and a
adapting to the characteristics of the input signal. high transmission S/N ratio. This means that unwanted sidebands
and carrier leaks are almost completely eliminated. This section
Adaptive Error signal
explains the principle of operation while comparing the PSN type
algorithm
SSB modulator with the analog filter type SSB modulator used in
conventional analog transceivers.
Coefficient
control
5-3-1 Analog filter type SSB demodulator
Correlation – The configuration of analog filter type SSB modulator is shown in Fig. 3.
Input signal separation FIR filter ∑
parameters Balanced mixer
+
MIC
IF filter
Fig. 2
f2
5-2-2 Noise reduction Modulation carrier
The adaptive filter allows the target signal to pass while the noise
Fig. 3
component (random signal) is attenuated. The voice signal has a
high short-time correlation and a low long-time correlation. (For If the tone signal of frequency (f1) is presented to a microphone, two
discrimination the signal correlation is called “short-time” or “long- spectra (f2–f1 and f2+f1) are generated against the mixer output as
time” for convenience, however it is a difference of only several shown in Fig. 4.
hundreds microseconds.) If the correlation separation parameters are
set to allow short-time correlation to be detectable, the voice signal is
detected as a low correlation component, and the noise as a high
correlation component. In this case the voice component must pass
as it is, and only the noise component is attenuated. The noise
reduction effect is random at the head of a word (the moment when Frequency
speech begins) and at points where intonation changes significantly.
(f2–f1) f2 (f2+f1)
If the noise reduction effect is increased too much, the voice
component may be attenuated together with the noise as described Fig. 4 Mixer output spectrum
above. In this case it may decrease clarity, even if the S/N ratio is
improved.
Passing
The transceiver is designed with the flexibility to set the noise characteristics
of IF filter
reduction level accurately (16 stages) in order to meet all
circumstances. This makes it possible to adjust the balance between
the S/N ratio and clarity quickly.
correlation component, and only the tone signal is detected as a high Fig. 5 SSB demodulated wave
correlation component. If the correlation separation parameters are
set to allow long-time correlation to be detectable the voice signal is This mixer output passes through the IF filter, passing only the
detected as a low correlation component, and only the tone signal is necessary band. Its unwanted sideband is attenuated, which assures
detected as a high correlation component. Since this setting makes it a modulated SSB signal. (Fig. 5)
possible to separate the tone signal component from the voice
component the output from the adaptive filter will be only the tone
signal. Since the phase and amplitude of the tone signal from this
adaptive filter become the same as those of the input signal, the
output of the error signal shown in Fig. 2 makes it possible to obtain a
voice signal from which the tone signal is removed. In other words,
this adaptive filter setting will operate as an automatic notch to
remove beat interference such as CW and RTTY signals, which may
interfere with SSB. Automatic notch makes it possible to detect and
14
Since the performance limits of the IF filter become the performance (f2–f1) (f2+f1)
limits of the modulator in an analog filter type SSB modulator (Fig. 3
shown on page 14) the problems below will exist: Point A2
1. The ripple characteristics within the passband of the IF filter is
reflected directly upon the entire frequency characteristic of the 180° phase difference same phase
modulator.
2. There is a limitation in the shape factor of an IF filter.
Point B2
If an attempt is made to execute the modulation output to be
excessively low-bandwidth it becomes unable to fully restrict the
unwanted sideband signal.
3. A crystal IF filter with a good shape factor may not provide the
satisfactory group delay characteristics in many cases and may be
inferior from the viewpoint of sound quality.
Modulated output
5-3-2 PSN type SSB modulator (basic type)
The PSN type SSB modulator uses phase shift operation to negate
(f2–f1) (f2+f1)
the unwanted sideband signal and to attain a modulated SSB signal.
If it is possible to reduce the phase difference at low-band of a 90°
Fig. 8 Phase relationship of modulated signal
phase shifter it will assure superior characteristics to the filter type
SSB modulator as it is possible to attain a higher unwanted sideband
signal suppression ratio compared with that at low-band. The PSN type SSB modulator provides an SSB modulated signal by
eliminating the unwanted side band component. To achieve this it is
A1 A2 necessary to keep the phase difference accurately and to set the
All-pass
filter-A amplitudes to precisely the same level.
MIC With a PSN type SSB modulator using analog circuit, such problems
f1 Modulated output
as changes in characteristics due to deviation in parts or temperature
90° phase
shifter may occur. Accordingly, it is very difficult to achieve the same
unwanted sideband signal restriction ratio with a filter type SSB
All-pass
filter-B
modulator.
B2 For these reasons few transceivers adopt the analog type PSN.
B1
Using the DSP it is possible to provide stabilized performance even if
the PSN method is used, as it has few of the fluctuations seen in the
f2 Modulation carrier analog circuit.
Fig. 6 Configuration of PSN type SSB modulator 5-3-3 Icom’s PSN type SSB modulator
The two filters (filter A and filter B) shown in Fig. 6 are combined to Figure 6 is a basic configuration drawing of a PSN type SSB
make a 90° phase shifter. This is an all-pass filter (*6) designed using modulator. The IC-756PROII adopts the PSN type SSB modulator
two filters in pairs so that the signal output from each filter appears to using Icom’s unique architecture shown in Fig. 9.
have a phase difference of exactly 90° when the same signal is input. This method makes it possible to obtain an effect equivalent to multi-
*6 All-pass filter: rate processing (*7) even if the sampling rate is not decreased during
An all-pass filter is used to change only the phase without changing the amplitude of the all-pass filter processing. This makes it possible to improve the DSP
signal sent from the all-pass filter.
calculation by more than two times which is required for accurate
SSB modulation processing. The part of modulated carrier
Point A1 multiplication in the conventional method is changed to the
multiplication of a constant leading to an improvement in efficiency.
90° phase difference *7 Multi-rate processing
A method of processing that uses the multiple sampling rates selectively, depending on the
Point B1 frequency of signal to be processed.
Even if the processing contents are the same, the processing of a lower sampling rate will
decrease the volume of calculations.
Fig. 7 Output signal of all-pass filter
a1
All-pass filter-B
frequency (f1) presented to the microphone, two spectra (f2–f1 and
Multiplexer
a3
f2+f1) are generated at two points, point A2 and point B2 respectively, Voice input Modulated output
each of whose phase relationship is as shown in Fig. 8. The signal at All-pass filter-A
point A2 is added to that at point B2 as indicated. The sideband a4
15
For the SSB modulator shown in Fig. 9, the signal is input to each [dB] 20
filter sequentially for each sampling cycle by using a multiplexer with
4 all-pass filters (filter A, filter B, each designed for a phase difference 0
of 90°) arranged alternately to multiply each filter output by the
constants (a1 to a4).
–20
Using the multiplexer the result of multiplication is output sequentially,
making it possible to gain the desired SSB-modulated output signal.
–40
· For USB: Constant {a1, a2, a3, a4} = {1, 1, –1, –1}
· For LSB: Constant {a1, a2, a3, a4} = {1, –1, –1, 1} –60
For PSN modulation processing using the 16-bit fixed decimal point
–80
DSP of conventional transceivers, the characteristics are adjusted to
decrease the influence of the rounding error (when the filter
–100
coefficient is quantized) as it occurs. For the 32-bit floating point DSP –800 –600 –400 –200 0 200 400 600 800
nearly ideal characteristics are assured as the influence of errors due [Hz]
to quantizing is extremely limited. The IC-756PROII was re-designed
with this point taken into consideration to further improve the low- Characteristics of manual notch
band characteristics as compared with conventional transceivers.
Figure 10 shows the restriction characteristics of an unwanted
sideband signal and the pass characteristics of the desired sideband
signal.
[dB] 0 [dB] –20
Desired sideband
–10 –30
–20
–40
–30
Unnecessary sideband
–50
–40
–50 –60
–60 –70
–70
–80
–80
–90
–90
–100 –100
–150 –100 –50 0 50 100 150
101 102 103 104
[Hz]
[Hz]
The IC-756PROII manual notch filter has extremely sharp The IC-756PROII is equipped with a newly developed RF type
characteristics which can be provided only by DSP processing. speech compressor. The configuration of the speech compressor is
Since this manual notch is processed within the AGC loop even shown in Fig. 11.
powerful beats are cut-off sharply without any influence upon the Amplitude control Band limiting
amplifier filter
AGC. The filter characteristics are sharp and the passband width is
PSN
held to approximately 50Hz with an attenuation level of over 70dB. modulation
processing
This makes it possible to adjust the notch point accurately. Only the
DSP provides the characteristics as shown above.
Data buffer
With an analog type notch filter (crystal or LC notch filter) it is not
possible to adjust the notch point characteristics accurately as shown
above, as the frequency characteristics are liable to deviate. The
Adjustment of
manual notch assures stable filter characteristics by DSP processing Analysis of Determination
compression gain
amplitude level of control level
because of its extremely sharp characteristics and the high-stability
reference oscillator provides superior frequency stability. Fig. 11 Configuration of speech compressor
Accordingly it provides stable operation such that it is not necessary
to re-adjust the notch point, provided the beat signal is not moved The operating principle of this compressor is that the SSB-modulated
once it is set. IF signal is saved in the data buffer for a fixed time at first, and then
the IF signal saved in the buffer is analyzed for amplitude level. The
control level of the amplitude control amplifier is determined in
accordance with the analysis, providing compression control such
that the signal peak does not exceed a certain level. In other words,
the amplitude of the current signal is controlled in accordance with the
change in amplitude over a certain previous period.
16
Unlike the RF compressor used widely in conventional analog The microphone equalizer of IC-756PROII allows smooth selection of
processing type transceivers little distortion will occur as the signal is characteristics and may be adjusted accurately over 11 stages for
not clipped. The speech compressor resembles an AGC type high band and low band. This makes the frequency characteristics
compressor in that the signal level is controlled, however the normal adjustable without any sense of incongruity.
AGC method has a lot of problems. It is usually considered that the
AGC type has an improved compression effect along with shortened
gain recovery time constant, compared to the grip type. Setting the
5-7 RTTY demodulator
time constant to a low level may bring about an inferior compression
effect as the adjustment range of the time constant is limited due to The IC-756PROII is equipped with a built-in demodulator/decoder
spoiled AGC loop stability. function (for BAUDOT RTTY) for the first time in an HF amateur
The Icom type compressor assures a high compression effect as transceiver. It is possible to decode RTTY signals using the
there are no problems due to the non-execution of feedback transceiver independently even if external units such as multi-function
processing with a proper follow-up performance against changes in TNC, and a RTTY terminal unit (compatible to RTTY) are not used.
amplitude of the IF signal. Even when the compression level is high When the RTTY signal is decoded the DSP unit executes the
only a slight distortion outside audible range may occur. To prevent demodulator processing and the binary signal (BAUDOT) obtained is
the transmit passband width from extending a wide-band limiting filter decoded by the main CPU, and its characters are displayed in the
is used. Since this filter was designed to prevent group delay lower portion of the display. Figure 12 shows the configuration of
degradation, it does not have an influence upon the modulated sound demodulator processing using DSP.
quality. TPF OFF AF amplifier
DAC
Distortion generated by compressor processing
TPF ON
For distortion generated by compressor processing, only the high
order distortion may be addressed in many cases. Also, mutual
modulation distortion may occur when the input signal is of 2 tones or Amplitude limit
more. The RF stage grip-processing compressor is better than the AF
stage grip-processing compressor from the viewpoint of high-order
distortion. The reason why it is not so highly rated from the sound Twin-peak Hysteresis
audio filter comparator Decoding processing
quality viewpoint is because there is a problem with mutual
Mark space
modulation distortion. The AGC type compressor provides a lower Main
signal
detection CPU
mutual modulation distortion level as compared to the grip-processing
compressor assuring better sound quality. The Icom type restricts Threshold level
BAUDOT
mutual modulation distortion similarly. (setting of hysteresis level)
demodulation
signal
Inside of DSP
5-6 Microphone equalizer Fig. 12 Configuration of RTTY demodulator
The microphone equalizer characteristics used for the IC-756PROII Most conventional RTTY terminal units or TNCs use either the PLL
are based on the frequency characteristics of the audio tone control type or filter type demodulator to detect the mark/space signal. When
circuit which has been re-designed to be dedicated to voice the communication conditions are undesirable due to interference,
frequency range. The transmit function of an analog filter is simulated fading, etc., the filter type is generally superior. The demodulator
and converted into that of a digital filter to provide the microphone processing of IC-756PROII uses the basic configuration of a filter type
equalizer function. In some microphone equalizers for transceivers demodulator.
the characteristics may change suddenly with a specific frequency as
a limit. Unnatural sounds may be generated by such equalizers For demodulator processing in DSP the amplification and amplitude
depending on their tone quality. Not in Icom’s. limitations are first executed against the audio signal demodulated
through product detection. This processing provides sufficient
[dB] 25 demodulation performance against even low level signals that do not
20 move the S-meter, so there is no influence due to deviations in
amplitude. The twin-peak audio filter then removes the radio
15
interference and improves the S/N ratio before detecting mark/space
10 signals. Two narrow-band pass filters are used in detection processing
5 to extract the components near the mark frequency and space
frequency. The output of each filter is detected and balanced, polarity
0
reversed, and then passed to comparator processing. The comparator
–5 processing has a hysteresis characteristic such that it is hardly
–10 affected by fluctuations in the noise component. The hysteresis width
is adjustable by changing the threshold level value on the RTTY
–15
decoding screen. The comparator determines the signal for polarity.
–20 The result is converted into a logic signal and then transmitted to the
–25 main CPU. The main CPU decodes the RTTY signal and displays the
101 102 103 104 105 characters on the display screen.
[Hz]
17
In filter type demodulators, the difference in filter characteristics IC-756PROII RTTY demodulator the effect of twin-peak audio filtering
appears to be a difference in decoding performance. The filter will has made a significant contribution to improving the decoding ratio.
enhance the decoding ratio provided a high performance filter is When the RTTY mode setting is selected, it becomes possible to
used. It is also influenced by the phase and time response change the speaker output and the audio output through the
characteristics. Twin-peak audio filters and mark/space signal accessory terminal to a signal filtered by the twin-peak audio filter.
detection filters are carefully tested to adjust their characteristics. Using this function it is possible to improve the decoding ratio of a
Final development of the Icom filter was conducted in part in TNC, terminal unit, etc. connected to the radio. Since the twin-peak
cooperation with veteran stations with a long RTTY history. A audio filter is connected at all times to the built-in demodulator, it is
decoding ratio equivalent to a dedicated RTTY unit such as TNC or not necessary to set the twin-peak audio filter output when using only
RTTY terminal units designed for existing RTTY is achieved. For the the built-in demodulator.
5-8 Receiver
18
The mixer circuit incorporates a double balanced mixer in which four The signal is further amplified by the 2nd IF amplifier and enters the
FETs are used to provide high IP and high dynamic range. This 2nd IF filter. This is a ceramic filter with a high shape factor and a
provides a significant improvement of the S/N ratio with limited center frequency of 455kHz to restrict the maximum passband width
distortion against large input signals, and provides superior 2-signal of the signal passed to the DSP. The 455kHz signal is then passed to
characteristics with no influence from the strong signals of an the 3rd mixer. The IC-756PROII uses a high-speed analog switch
adjacent frequency. instead of the conventional mixing IC to improve the adjacent
The 1st mixer and LO circuits are arranged in two sets to provide the dynamic range characteristics and to restrict distortion.
dual-watch function. The signal is converted to 64.455MHz by the 1st An active LPF (consisting mainly of an operating amplifier) is included
mixer and then passes through a variable type attenuator (using the to collect the necessary frequency component (36kHz) from the 3rd
PIN diode) to adjust the dual-watch balance where an attenuation of mixer output. The capacitor of this active LPF circuit is a film type
approximately 70dB (maximum) is assured. The receive level is capacitor with limited distortion and superior temperature
adjustable for main band and sub band by changing the balance. characteristics. The signal is then amplified and passed to the DSP
The GG amplifier (located in the succeeding stage) as well as the GG port.
amplifier (located at the input side of the mixer) isolates the main The 36kHz IF signal is differentially converted by the operating
mixer from the sub-mixer, improving the 2-signal characteristics, amplifier and is passed to the A/D converter. The signal is passed to
while maintaining the impedance (as viewed from the mixer) at a the DSP IC through the level converter. The DSP IC is operated as a
constant level. A combiner transformer determines the output for digital IF filter of 36kHz or as a demodulator under each mode. The
main mixer and sub mixer. The IF stage following the combiner demodulated signal is then passed to the D/A converter through a
transformer uses the circuit used for the main mixer and sub-mixer in level converter and converted into an analog signal to pass through
common. The received signal passes through the 1st filter to the low-pass filter via a differential input type active filter, buffer
eliminate unwanted signal components in the mixer stage. The 1st IF amplifier and analog switch to remove unwanted signals. The filtered
filter is a crystal filter selected taking 3rd order distortion into signal passes through the analog switch to absorb the demodulation
consideration. After passing through the 1st IF filter the signal is level difference between each mode with a demodulation level
controlled by the AGC. It then enters the 2nd mixer through the 1st IF equalizing circuit.
amplifier. This mixer is a diode double-balanced type with high IP
which is highly effective against in-band IMD and adjacent signal
interference. The element of the signal converted to 455kHz by the
2nd mixer enters the noise blanker circuit. The IF amplifier is
connected to the noise blanker circuit by 4 stages in series to assure
high gain. When the threshold level of the circuit used to control the
noise blanker gate is varied, it is possible to change the noise blanker
level in 100 stages.
19
5-9 Transmitter
The voice signal enters through the microphone and is amplified by The signal then leaves the analog switch and enters the Main-A unit
the VCA (voltage control amplifier). The voice signal is controlled in through the LPF as the DTIF signal to attenuate the out-of-band,
gain and passed to the DSP as the DTAF signal through the analog spurious, or image noise. The signal is converted to the 2nd IF of
switch. The VCA controls the gain of the microphone in accordance 455kHz by the 3rd mixer circuit built into the Main-A unit and passes
with a signal output from the main CPU. When SSB mode is through the ceramic filter and IF filter via IF amplifier, to enter the RF-
selected, the signal enters the amplifier through the analog switch A unit. The 2nd IF signal is mixed with a 64MHz signal sent from the
and passes through the low-pass filter entering the differential PLL circuit by the 2nd mixer, converted to an IF signal of 64.455MHz,
amplifier, to restrict the band of the A/D converter input signal. When stripped of unwanted components by the XTAL BPF, and enters the
FM/AM mode is selected, the signal passes through the limiter IF amplifier. The ALC is applied to the IF amplifier.
amplifier, low-pass filter and pre-emphasis circuit, and enters the The IF signal is converted to the desired frequency by the HSB88WS
differential amplifier in the same manner as SSB mode. The amplified diode mixer, stripped of unwanted frequency components by the
signal enters the A/D converter and enters the DSP IC through the 60MHz cut-off LPF, amplified by the RF YGR amplifier, and is then
level converter. After the signal has been demodulated by the DSP output to the PA unit. The transmit signal passes through a class A
IC, it is output as a 36kHz transmit IF signal. The demodulated signal type amplifier, is amplified by the class AB push-pull amplifier, and is
passes through the level converter and is converted to an analog then amplified to 100W by the final amplifier (2SC5125 × 2). In the
signal by the D/A converter. The analog signal passes through the output of the final amplifier the higher harmonic is attenuated by the
differential input type active filter and enters the analog switch transmit PF compatible with each band.
through the buffer amplifier.
The IC-756PROII uses a well-balanced push-pull amplifier and LPF be decreased resulting in an increased noise level when
to provide an enhanced harmonic level for all bands of approximately demodulation is not executed. This relationship will theoretically be
60dB (practical value). “number of bits × 5dB”. For a 16-bit D/A converter this is a S/N of
96dB. Comparing the rated output of 100W with the noise when
The demodulation input/output to/from DSP uses the 24-bit A/D-D/A. demodulation is not executed the S/N will be the value obtained by
The demodulation input/output to and from the DSP uses a 24-bit subtracting the gain controlled by ALC from 96dB. When the 20dB
A/D-D/A converter. The use of the high-bit A/D-D/A converter gain control is executed at 100W, using the ALC for instance, the
significantly reduces modulation distortion due to bit error. Note that value of 76dB (=96 – 20) will be the ratio between noise when
the limited number of bits causes the level deviation/bit to be demodulation is not executed and level at time of 100W transmit.
increased and consequently causes the non-linear movement and For 24 bits this is 124dB (=24 × 6 – 20). As a result the noise of the
demodulation distortion to be increased. The limited number of bits A/D converter is reduced to a level where it is not a problem.
may also cause the maximum output level/noise output level ratio to
CENTER 14.20150 MHz SPAN 50.00 kHz CENTER 7.05000 MHz SPAN 10.00 kHz
20
1st mixing circuit Low-pass Amplifier Band pass Low-pass Pre-amplifier Driving Power Low-pass
amplifier amplifier
HSB88WS filter µPC2708 filter filter 2SC1971 2SC1972×2 2SC5125×2 filter
Limiting Limiting D/A CONV. AF Band pass Phase Band pass RF POWER VR
amplifier amplifier amplifier
NJM2058 1/4 M5218 1/2 AD1855JRS µPC4570 1/2 filter comparison filter
MAIN/SUB DIAL
21
5-10 Dual-watch function
The dual-watch function allows the designated receive frequency and difference in the levels of the signals received is increased right after
another frequency or the transmit/receive frequency of a DX station the output from the 1st mixer. The transceiver is designed to receive
(used for split operation) to be watched at the same time. Fitting the these two signals properly by operating the balance adjustment knob
transceiver with two receive circuits is one way of obtaining this dual- located on the front panel. The transceiver uses a gate earth buffer
watch function. If a sub-receive circuit is used the performance is amplifier before and after the mixer, with the FET having significant
liable to decrease compared with the receive performance of the reverse isolation, so that the oscillation signal of the 1st stations (2
main receive circuit. The IC-756PROII has a dual-watch system sets) will not be mixed with each other. The receive characteristics of
which exceeds those in conventional equipment. Here two sets of a this system when the main receive circuit receives a signal is the
PLL circuit and 1st mixer are used for the dual-watch function. Both same as when the sub-receive circuit receives a signal. This makes it
sets are used when the first intermediate frequency is attained to possible to validate the noise reduction, noise blanker, etc. even
provide for dual watch. The ATT circuit has a PIN diode (used to when a signal is received by the sub-receive circuit.
adjust the balance so that the weak signal is not masked) when the
The receive system circuit is separate from the real-time spectrum Comparison of characteristics of spectrum scope
scope in the IC-756PROII. The circuit dedicated to the spectrum
IC-756PROII 14.1M,P.AMP1
scope is used to enhance its accuracy. This makes it possible to use
IC-756PROII 50.1M,P.AMP2
an attenuator dedicated to the spectrum scope, and to reflect the IC-756 14.1M P.AMP1
desired signal clearly without the band scope being saturated when
the attenuator is turned on, even while receiving a low-level signal 6
with a high noise floor.
5
Scale of scope
4
3
2
1
0
–40 –20 0 20 40 60 80 100
ANT input level (dBµV)
Description of circuit
The signal from the 1st mixer (while receiving) or the mixer in the RF- algorithmic output RSSI terminal and MIX, and is operated by
A unit (while transmitting) is passed through the PIN attenuator sweeping the LO input to this point. The FI842 filter determines the
(D801) and amplified at Q811 and Q812, and applied to the D831 resolution of the spectrum scope, using a ceramic filter in CW mode,
mixer. The D831 converts this signal to the 13MHz band using the to assure stable performance without need for adjustment. The RSSI
2nd mixer. The converted signal passes through the BPF, which is voltage output from IC841 is amplified by IC871 to provide both a
composed of two ceramic filters to suppress unwanted signals. The scope voltage and to apply the AGC to Q811 and Q812, to extend
signal then enters IC841. This IC is designed for FM IF and has an the dynamic range of the spectrum scope.
22
5-12 Voice record/playback function
This radio uses a dedicated IC (ISD4003-04) for voice The IC-756PROII does not require these devices as it uses an IC
recording/playback, and stores the analog signal as an analog value. dedicated to voice recording/playback which provides full quality
Usually an analog signal is digitized temporarily and stored as a audio reproduction. The previous IC-756PRO also used this method.
numerical value, to be converted back into an analog signal when it is The storage chip in the IC-756PROII was changed to one with a
reproduced. With this method it is necessary to use expensive A/D greater memory capacity to allow continuous recording capability.
and D/A converters and storage media (RAM).
Since the IC-756PROII is equipped with a Dual-watch function, two (0.5 × 10-6) when at temperatures between –30°C to +60°C. This
sets of PLL circuits with the same configuration are included. Unlike oscillator is a crystal oscillator called POC, in which the oscillator is
the PLL circuits of other HF transceivers, no mixer is used. The mixer thermally balanced due to the heat generated in the posistor in the
is intended to create a sum or difference for the two signals to pick oscillator, which does not allow the temperature change to occur in
out the desired frequency component. In fact the input signal as well the oscillator even if an external temperature change occurs. It is also
as the sum and difference are output for mixer output. For this possible to adjust the deviation generated due to deterioration over
reason, it is necessary to arrange a filter for the mixer output. The time.
unwanted components required by the mixer are also spurious for
transmit/receive. For a system configuration in which reference Configuration of LO for split operation
oscillation is controlled by the DDS (Direct Digital Synthesizer) to The 1st transmit LO for split operation is generated by changing the
output 10MHz, 10MHz is oscillated by the VCO of PLL on the basis of oscillation frequency of the PLL that generates the 1st receive LO.
the 10MHz generated by this DDS. The PLL oscillating the desired With this configuration no transmit or receive signals will leak at the
frequency at 10MHz created by the VCO is controlled to oscillate the receive frequency during split transmission.
VCO. In this case, it becomes unnecessary to arrange the mixer in
the transmit system. VCO of PLL for 1st LO
This successful design concept, using high-speed DDS, was adopted Four VCOs cover the receive frequency range from 0.03MHz to
for the first time by the IC-775DSP. 60MHz:
0.03–7.999MHz VCO1
Reference oscillator circuit 8–19.999MHz VCO2
The reference oscillator circuit generates the frequency used as the 20–44.999MHz VCO3
reference by all oscillation circuits concerned with transmit/receive 45–60MHz VCO4
frequency. Thus the frequency accuracy depends on the accuracy of This VCO uses a HI-Q coil to minimize the noise generation, thus
this oscillation circuit. The oscillator used for the reference oscillation assuring high C/N characteristics.
circuit of the IC-756PROII provides the high accuracy of < –0.5ppm (C/N characteristics diagram)
1st Lo C/N characteristics of IC-756PROII 1st Lo C/N characteristic of high-grade HF radio of a competitor
[dB] 0 [dB] 0
–10 –10
–20 –20
–30 –30
–40 –40
–50 –50
–60 –60
–70 –70
–80 –80
–90 –90
–100 –100
–10 –8 –6 –4 –2 0 2 4 6 8 10 –10 –8 –6 –4 –2 0 2 4 6 8 10
[kHz] [kHz]
The above graphs show the 1st LO C/N characteristics for IC- slight, the output of the 1st LO significantly affects the
756PROII and the PLL of a high grade HF unit from another transmit/receive performance. When the low-band signal in the HF
manufacturer. The graph to the left indicates the LO C/N band is received, the difference will be apparent.
characteristic for the IC-756PROII. While the difference seems to be
23
Other LOs
The 2nd LO works to double the output of the reference oscillator
circuit previously described, and the 3rd LO is obtained directly from
the DDS operating in accordance with the output of the 2nd LO
reference oscillator circuit. Since the PLL is not used for such
frequency components, high purity and stable operation is obtained.
24
6. Connection to option/peripheral units
ANT REMOTE
ANT 2
ANT 1 ACC (2) REMOTE
GND
GND
IC-PW1 Ground
IC-756PROII
AC outlet
Non Europe versions: 100–120/220–240V
European version: 230V
25
6-3 Interface for digital mode
To use a personal computer to operate the digital modes (SSTV, If a filter width of 500Hz or less is selected the receive passband filter
PSK31, BAUDOT RTTY, etc.), it is necessary to install the following is activated to avoid interference while the transceiver receives SSB-
interface. D (SSB data mode).
The IC-756PROII is equipped with a digital IF filter that may narrow Refer to the instruction manual or help file contained in the 3rd party
the receive passband range to 50Hz making it possible to select and software prior to use.
receive only one station, even when it is used in PSK31 mode.
E C B
4.7kΩ* 1S1588 TXD
Pin 2 Pin-3
This circuit connects the input and output through transformers to prevent RF feedback and to isolate the
transceiver from the computer.
Example of external control circuit When a properly constructed control circuit is connected to the
(Not provided by Icom.) microphone connector, it is possible to control the transmission of the
CW memory keyer (M1 to M4) and DVR (T1 to T4).
Connection diagram Microphone connector This also makes it possible to transmit the memory keyer and voice
Pine memory while displaying the scope.
MIC
U/D
Pin u
MIC
GND
1.5kΩ
S1 ±5%
1.5kΩ
S2 ±5%
2.2kΩ
S3 ±5%
4.7kΩ
S4 ±5%
External keypad
26
6-5 Installation of UT-102 optional Voice Synthesizer Unit
UT
-10
2
UT-102 J3502
27
7. CI-V control
Connection of computer A personal computer may be used to control the frequency, operating
mode, VFO/memory status, etc. via its serial port, using the Icom
IC-756PROII
Power cable Communication Interface V (CI-V).
(9-15 V DC)
When the optional CT-17 (CI-V level converter) is connected, it is
possible to control up to 4 Icom Transceivers Receivers with a
personal computer. A wide variety of 3rd party software applications
may be used to provide automated logging and control of your radio.
Personal computer
RS-232C cable
CI-V cable
frequency or memory
(see command table)
OK code
(fixed)
Command number
End of message
End of message
default address
default address
default address
default address
TransceiverÕs
TransceiverÕs
number entry
code (fixed)
code (fixed)
code (fixed)
code (fixed)
ControllerÕs
ControllerÕs
Preamble
Preamble
NG code
(fixed)
FE FE E0 64 Cn Sc Data area FD FE FE E0 64 FA FD
q w e r t y u
IC-756PROII to controller NG message to controller
28
7-3 List of commands
• Command table
29
• Command table (continued)
30
• Command table (continued) • Channel code for memory keyer
To send or read the desired memory keyer contents, the channel and
Command Sub command Description character codes as follows are used.
1A 0562 Send/read twin peak filter (0=OFF,
1=ON)
• Channel code
0563 Send/read timer functions (0=OFF, Code Channel number
1=ON) 01 M1
0564 Send/read DSP filter type 02 M2
(0=SSB: sharp; CW: sharp, 03 M3
1=SSB: sharp; CW: soft,
04 M4
2=SSB: soft CW: sharp,
3=SSB: soft CW: soft)
0565 Send/read quick RIT/ TX clear function • Character’s code
(0=OFF, 1=ON) Character ASCII code Description
0566 Send/read SSB/CW synchronous tuning 0–9 30–39 Numerals
function (0=OFF, 1=ON) A–Z 41–5A Alphabetical characters
0567 Send/read CW normal side set
a–z 61–7A Alphabetical characters
(0=LSB, 1=USB)
space 20 Word space
0568 Send/read external keypad type
/ 2F Symbol
(0=OFF, 1=Keyer send, 2=Voice play
(Tx), 3=Auto) ? 3F Symbol
0569 Send/read NB level (0=0% to , 2C Symbol
255=100%) . 2E Symbol
06 Send/read DATA mode (0=OFF, 1=ON) ∧ 5E e.g., to send BT, enter ∧4254
07 Send/read SSB transmit bandwidth * 2A Inserts contact number (can be used for 1
(0=Wide, 1=Middle, 2=Narrow) channel only)
1B 00 Set repeater tone frequency
01 Set tone squelch tone frequency
• Character’s code for my call
1C 00 Set the transceiver to receive or
transmit condition (0=Rx; 1=Tx) Character ASCII code Description
0–9 30–39 Numerals
04 10 19.900000–10.499999
100Hz digit: 0 (fixed)
05 14 13.900000–14.499999
100kHz digit: 0–9
06 18 17.900000–18.499999
00=+direction
01=–direction
07 21 20.900000–21.499999
Direction:
08 24 24.400000–25.099999
09 28 28.000000–29.999999
10 50 50.000000–54.000000
11 GENE Other than above
• Register code
Code Registered number
01 1 (latest)
02 2
03 3 (oldest)
31
8. Inside Views
Final Power amplifier
(2SC5125 × 2)
Drive amplifier
Fan
Filter unit
Common filter
Current transformer
(current, voltage, etc.)
Antenna tuner CPU
C-MOS IC
BPF board
YGR amplifier
Preamplifier
VCO-B circuit
FM IF IC
VCO-A circuit
Ceramic filter for
scope IF
Memory board
Voice synthesizer unit
UT-102 (option)
2nd IF filter
DSP-A board
3rd mixer
MAIN-A unit
32
9. Options
IC-PW1 HF/50MHz ALL BAND 1kW LINEAR AMPLIFIER PS-125 DC POWER SUPPLY
Full-duty 1kW linear amplifier including an automatic antenna tuner. Has automatic tuning Light weight power supply.
and band selection capability. Full break-in (QSK) operation is possible. The • Output voltage: 13.8V DC
amplifier/power supply unit and the remote control unit are separated. • Max. current drain: 25A
AH-4 HF/50 MHz AUTOMATIC AH-2b ANTENNA ELEMENT HM-36 HAND MICROPHONE
ANTENNA TUNER
For remote transceiver control using a PC. Announces the receive frequency, mode
You can change frequencies, operating and S-meter level in a clear, electronically-
mode, memory channels, etc. generated voice, in English (or Japanese).
33
10. Specifications
GENERAL RECEIVER
34
11. Block diagram
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