W6IWI DSP TU Part Two

This project has many firsts for me. These firsts include:

• First DSP project
• First MPLAB X project
• First KiCAD project
• First SMT project

SMT soldering is a challenge, but I AM getting better. I'm using a YIHUA 959D hot air rework station with Essmetuim ZB628 solder paste. The air temperature was set to 300 C. The process works very well, though it was a challenge to solder down the 64 pin microcontroller. On the first try, the chip was tilted a bit so the pins on one side were between the pads instead of on them. I finally removed the chip, placed it again, and soldered it down. It is amazing that the chip still worked after being heated for so long. The soldering was viewed under a microscope to ensure there were no shorts. Any shorts were removed by wiping a hot soldering iron on the leads moving away from the chip body.

The only problem found with the board so far is the orientation of the display connector. The display board was supposed to plug onto the right angle header and then extend above and perpindicular to the board. However, the pins are reversed so that when plugged in, the display extends below the board instead of above. For now, the display is wired to the display connector using individual lead wires.

A fair amount of code remains to be done, especially dealing with the front panel switches, display, etc.

The video at the right shows the board receiving ITTY.

Preliminary Tests

Some preliminary tests were run by setting AudioOut to DISCRIM and watching the signal on the audio output, U2-1. Note that U2 is disabled unless the TU is in transmit, so it was put in transmit by pressing the TX button. The eye diagrams show minimal jitter in the zero crossing and full amplitude at the middle of the bit position. While full amplitude is not required for proper demodulation, the mark and space amplitudes are used to set the dynamic threshold control level, so full amplitude for each bit is desired.

The 170 Hz eye diagram was generated with this audio. The 850 Hz eye diagram was generated with this audio. Both use random data with each bit being 22 ms. The audio was generated using LTSPICE RandomAfskGenerator170.asc and RandomAfskGenerator850.asc.

The following filter settings are in UserConfig.c.

  const UserConfig_t UserConfigDefault={
    .NarrowShiftCenterFreq=2210.0,    // Center frequency for 170 Hz shift
    .NarrowShiftHz=170.0,
    .WideShiftCenterFreq=2000.0,      // Center freq for 850 Hz shift
    .WideShiftHz=850.0,
    .BaudRate=45.45,                  // Used to determine filter bandwidth and in software uart
    .ToneFilterBwBrMult=2.0,          // Tone filter bandwidth is the baud rate
                                      // times this number.
                                      // Make wide enough for minimal attenuation
                                      // of BR/2 sideband
    .AutostartShutdownSeconds=30.0,   // Keep motor running this may seconds after signal drop
    .KosDropSeconds=5.0,              // Drop transmitter 5 seconds after last character
    .AgcTargetLevel=0.5,              // AGC adjusts to this level
    .AgcLpfF=0.1,                   // Cutoff frequency of the LPF in the AGC gain control 
    .UseInputBpf=0,
    .UseLimiter=0,
    .UseAgc=1
  };
		  

The DSP TU does not have any high voltage on the PCB. It provides a 5V logic output to drive an external autostart motor control relay. The relay box at the right was constructed by adding a solid state relay to a Home Depot 1007 546 208 four outlet "power block." The "power block" sells for less than $15. It includes four outlets, a circuit breaker, and a heavy duty line cord. It was a tight fit, but a TE SSRMP-240D10R solid state relay was stuffed into the box and screwed down with a couple 6-32 screws. The black wire with the heat shrink on it originally went to the center connection on the circuit breaker. It was moved to one of the SSR AC connections. The added red wire connects between the circuit breaker center connection and the other SSR AC connection. An incoming two conductor cable was connected to the DC control terminals on the SSR. When the DSP TU puts 5V on the autostart output (either by autorstart or hitting the motor button), the SSR runs the printer motor.

The menu system allows continuous adjustment of various parameters. In addition, the menu allows the audio output (PWM through LPF) to reflect various points in the DSP "circuit." The images at the right show the discriminator output (mark LPF - space LPF) with different tone filter bandwidths (multiples of the baud rate). The eye diagrams were generated with this audio. The audio uses random data with each bit being 22 ms long. The audio was generated with LTSPICE RandomAfskGenerator170.asc. An eye diagram is difficult to generate with real RRTY data, including from punched tape, since the stop bit is 31 ms long. This causes a transition between bit times in the eye diagram.

As discussed some previously, it is interesting to think of each tone of the FSK signal as being an on-off keyed signal. A 45.45 baud signal (22 ms per bit) consists of a square wave at 22.72 Hz and subharmonics of this frequency (when there are successive mark or space bits).

Each tone is 100% modulated with a square wave with a maximum frequency of 22.72 Hz. This places sidebands at the tone frequency ±22.7 Hz. If the tone filters pass these sidebands without attenuation, we would get a sine wave with an amplitude that is 1.274 times the amplitude of the square wave with the peak at the middle of the bit time (half way between the mark-space transition). A filter that passes these two sidebands would have a bandwidth equal to the baud rate since the square wave frequency is 1/2 the baud rate, and there are two sidebands: one above, and one below the tone frequency. However, the filter bandwidth is where the gain is 3 dB down from the maximum, so the tone is only modulated about 71% after going through a bandpass filter where the bandwidth is equal to the baud rate. As noted above, the fundamental frequency component is 1.274 times the amplitude of the final square wave, but 0.71 * 1.274 = 0.90454, so the sine wave does not reach the full amplitude of the square wave if the tone filter bandwidth is equal to the baud rate. A somewhat higher bandwidth is required.

What is the harm in not having the sine wave amplitude reach the full square wave amplitude? Since our random data includes square waves at BaudRate/2 PLUS its subharmonics (when there are successive mark or space bits instead of alternating mark and space), these subarmonics WILL reach the full square wave amplitude (or even 1.274 times that amplitude), while the peaks of alternate mark and space bits would not. Note in the first image, where the bandwidth is 70% of the baud rate, that some bits reach a fairly high amplitude while others do not. If there are , for example, successive mark bits followed by a space bit, the space bit has to start at full amplitude and does not have time to reach the full space amplitude. This introduces bias distortion, though it is variable bias distortion instead of a fixed shift of the mark-space threshold. Note also that the peak to peak amplitude of the signal is about 1.4 volts.

The second image shows the eye pattern with the tone filter bandwidth set to 100% of the baud rate. As discussed above, this results in 3 dB attenuation of the fundamental of the BaudRate/2 square wave. It can be seen that bits are not reaching the full amplitude. The peak to peak amplitude has increased to about 2 volts.

The third image shows the eye pattern with the tone filter bandwidth set to 150% of the baud rate. The sine wave peaks are still a little short of maximim amplitude. The peak to peak amplitude is about 2.8 volts.

The final eye pattern is with the tone filter bandwidth set to 170% of the baud rate. All the bit peaks appear to be reaching full amplitude. The peak to peak amplitude is still about 2.8 volts.

Filters for RTTY suggests use of third order filters with the tone filters being 1.2 times the baud rate. The DSP TU uses fourth order filters (two biquads), so attenuation outside the passband is a bit more than that of a third order filter. Wider filters will have output due to off-frequency signals, such as noise, so filters should be as narrow as possible. Based on these experiments, it appears we should use tone filters that are 170% of the baud rate.

850 Hz shift was tried on the DSP TU. The center frequency is 2 kHz with mark 426 Hz below that, and space 425 Hz above. It was found that there is considerable amplitude skew with wide shift. The table below demonstrates the observed transmitter output power and TU output (on the primary of the output transformer) at various frequencies.

Frequency	TU Output		Transmitter Output
(Shift, M/S)	VoltsRMS	dB	Watts	dB
1 kHz	1.11	0	105	0
1.575 kHz 850 Mark	1.05 V	-0.5	95	-0.4
2.125 kHz 170 Mark	946 mV	-1.4	65	-2.1
2.295 kHz 170 Space	707 mV	-3.9	60	-2.4
2.425 kHz 850 Space	880 mV	-3.2	50	-3.2

The first image at the right shows the transmit frequency response of the SEA245. As noted, each peak is 100 Hz from the next peak. The first peak is at 300 Hz and down about 6 dB from the mid-band level. On the right, a peak at 2.7 kHz is down about 1 dB. The peak at 2.8 kHz is down about 12 dB.

The second image at the right shows the predicted frequency response of the 4 kHz low pass filter on the DSP TU AFSK audio output. This filter smooths the 80 kHz PWM which has a sample rate of 8 kHz (the duty cycle is changed 8,000 times per second).

Based on these two graphs, there should be minimal attenuation of the 850 Hz shift space frequency (2.425 kHz), though the RF power drops 3.2 dB from the power with 1 kHz input. Further, the audio level out of the LPF also drops 3.2 dB indicating the loss is in the DSP TU instead of the SEA245.