RTTY seems like a great opportunity to learn a bit about digital signal processing. I decided to use the PIC32MZ series of chips since I was most familiar with them. The last 15 years of my career involved work with the PIC32MX.
The first image shows the red CURIOSITY PIC32MZEF DEVELOPMENT BOARD plugged into a piece of Vector board that has the I/O circuitry (audio input and output analog circuitry, loop keyer, etc.) wire wrapped. The transformer on the left of the Vector board is the audio input transformer. It is being driven with audio from here from the tablet computer seen at the top left. The scope probe is connected to the audio output which the firmware is directing the discriminator output to. The audio output uses a PWM output from the PIC and passes it through a low pass filter to recover the audio. The white chip on the right is an SSR keying the 60 mA Teletype loop.
The second photo shows more of the development platform. A surplus television is used as a computer monitor. The MPLAB X software is on the screen with some of the code visible.The scope is showing an eye pattern from the discriminator. Below the scope is an audio mixer used to increase the audio level from the tablet computer to drive the demodulator.
The Terminal Unit consists of a modulator and a demodulator. The modulator seems like the simplest part while the demoduator is more complex and needs to deal with noise, interference, and selective fading.
All DSP functions are done on a sample by sample basis (instead of using ping pong buffers and acting on the full buffer at once). As long as all operations are completed before the next sample shows up, this should work. The sample rate is 8 kHz. All calculations are double precision.
Incoming 12 bit unsigned integer samples are converted to a double precision floating point with a value between -1.0 and +1.0 where -1.0 represents an analog input of 0V, and +1.0 represents full scale. Hardware centers the audio between Vcc and GND.
Audio is output using PWM. The PWM frequency is 80 kHz. The PWM duty cycle is updated 8,000 times per second, corresponding with the system audio sample rate. A sample value of -1.0 yields a duty cycle of 0%. A sample value of 0.0 yields a duty cycle of 50%. A sample value of +1.0 yields a duty cycle of 100%.
As shown in the sketch, the audio input is transformer isolated, then centered between ground and +3.3V, then applied to analog input 4 on the PIC32MZ. Also shown here is a relay to drive the transformer with the AFSK output. The SEA245 transceiver uses a "tri-state" audio bus for audio in to and out of the transceiver. I currently have a Flesher TU-170 driving this bus. I'll probably separate this out to two transformers, one input and the other output, to allow easy interface to other transceivers. When using the SEA245, the input and output would just be placed in parallel.
The AFSK audio output is shown in this sketch. Since the PICMZ does not have a DAC, a PWM output is used instead. The PWM frequency is 80 kHz, and the duty cycle is updated 8,000 times per second. The high PWM frequency makes it easier for the output LPF to remove the PWM signal while passing the audio. Due to the high frequency and the relatively low GBW of the op amp, an RC filter is before the active filter to remove much of the PWM signal. All calculations are done as double precision floats. When driving the PWM generator, the double precision float is converted to duty cycle such that -1.0 outputs a 0% duty cycle, +1.0 outputs a 100% duty cycle, and 0.0% outputs a 50% duty cycle. At the AFSK output, this represents a range of 0V, +3.3V, and 1.65V.
As shown in the sketch, the look is keyed by a G3VM-601BY solid state relay. The SSR is wired to "connection ADC with the two output FETs in series. This allows the connection to the loop independent of polarity. The SSR supports a maximum load current of 100 mA, so it is suitable for the 60 mA Teletype loop. It can also handle 480 V, so it does allows the magnetic field in the selector magnet to collapse very quickly on a mark to space transition. The BZW04-342B transient suppressor diode limits the flyback voltage to 380 V (1 mA).
The loop current detection is not yet built, but will probably use an H11AA1 optical coupler. It has inverse paralell LEDs on the input, so it will not care about loop polarity. The LEDs will be placed in parallel with about 42 ohm resistor. This should result in 30 mA through the resistor and 30 mA through the LED (which is rated at a maximum of 60 mA). The loop current sense will be in series with the loop keyer so a single 1/4 inch TRS plug can insert the TU into the loop.
The sketch shows the interface to the CURIOSITY PIC32MZEF DEVELOPMENT BOARD.
Display Wiring shows the interface between the tuning display and the development board. The display uses SPI1.
Data from the loop current detect opto coupler will cause the AFSK frequency to switch between 2125 (mark, loop current present) and 2295 (space, no loop current). These values will be passed through a low pass filter to limit the keying harmonics. The output of the LPF will drive the DDS tone generator. The code for the DDS tone generator accepts a frequency in Hz. The frequency will swing between the mark and space frequencies with a limited slew rate.
The DDS code calculates how many radians need to be added to the current value on the next sample (125 us) to result in the correct frequency. On each call of DdsNextSample(), the code adds that value, corrects for overflowed past 2*pi, then calculates the sine of resulting angle. The sine value is returned. This can drive the audio output (described above) to drive the transmitter audio input.
Code under development is available on GitHub.
Code is being developed using Microchip's MPLAB X software. MPBABX includes a code generator called Harmony. It generates code for initializing ports and peripherals. Since most pins include multiplexers that allow a pin to serve one of several uses, the Pin Configurator makes configuring the pins much easier than writing directly to the corresponding registers. Harmony also includes a system clock configurator to make sense of all the options for the system clocks. Harmony also generates initialization code for peripherals along with interrupt code (such as for timer 2 and the UART), and code to communicate with the peripherals. The next several images show the Harmony configuration for the DSP TU.
|The image at the right shows the clock configuration as set in Harmony. Note that we are using the 8 MHz FRC (fast RC) oscillator to drive the System PLL to generate the 200 MHz system clock. The peripheral clocks (such as timers and UARTS) are driven with 100 MHz.|
|The image at the right shows the configuration of timer 2. It counts a 100 MHz peripheral clock. The configurator generates code that sets the timer period register so it times out in 12,500 ns resulting in a frequency of 80 kHz. The timer generates an interrupt that decrements a software counter to count 10 timeouts resulting in the audio sampling frequency of 8 kHz. Timer 2 drives OC1 (Output Compare 1) that is configured to generate PWM for the audio output. The PWM is 80 kHz, and the duty cycle is updated 8,000 times per second by the DSP code.|
|The image at the right shows the configuration of OC1 (Output Compare 1). It is configured to generate PWM based on Timer 2. It does not have PWM fault inputs. Timer 2 is operating as a 16 bit timer (not combined with timer 3 for 32 bits), so OC1 is also configured for 16 bits. OC1R is the output compare register, and OC1RS is the slave output compare register. OCRS is transferred to OCR at the start of each timer 2 cycle (when reset by the Period Register). This prevents PWM glitches that would occur if the OCR register were written to directly ( for example, if timer 2 has a value of 2,000, OCR has a value of 3,000 but is changed to 1,000, the OCR would not match timer 2 on this cycle resulting in a 100% duty cycle for this one pulse.)|
|The image at the right shows the configuation of UART2, which is used for debug output. It is set to use a 512 byte ring buffer on the transmit side so code is not slowed down waiting for each character to be transmitted. Harmony generates the initialization code, the interrupt code to drive the UART, and the code to put data into the ring buffer. The UART is configured to use only the TX and RX pins. Handshake pins (CTS, RTS) are not used and are free to be used for GPIO. The UART bit rate generator is driven by the 100 MHz peripheral clock and transmits at 2 Mbps.|
|The image at the right shows the configuration of the analog to digital converter used for the AFSK input. Most settings are their default values. Note that the power supply is used for the ADC reference, the output format is unsigned integer, and the resolution is 12 bits. The software starts an ADC conversion every 125 us, immediately after it has retrieved the previous conversion so the next conversion happens "in the background" while code is running instead of having to wait for a conversion to complete.|
|The image at the right shows the configuration of SPI1 to drive the display.|
|This image shows the configuration of several pins in the
|The image at the right shows the pins assigned to the PWM output (discussed above) and CPU_IDLEn. CPU_IDLEn is driven high at the start of the DSP code and driven low at the end. The CPU load can be determined from the duty cycle of this signal (8 kHz).|
|The image at the right shows CPU_IDLEn, discussed above, and DEBUG_UART2_TX. This is the transmit data output of UART2 which is configured to transmit at 2 Mbps. Debug information is sent out this port when needed.|
|The image at the right shows LOOP_KEY_OUT. This pin is driven high to put the 60 mA current loop in the mark condition. It drives a solid state relay that keys the loop.|
|The images at the right show the addition of the pins to drive the display over SPI.|
|The biquad filters store data on the heap. When the number of cascaded filters in each of the tone filters was increased from one to two, the heap overflowed. The image at the right shows the linker properties where the heap was increased from 512 bytes to 1024 bytes.|
Some tests were run with different filters. In the images below, the demodulator was driven with random 22 ms per bit audio with space 20.8 dB below mark to simulate selective fading. The audio is here. The audio output of the DSP TU was "connected" to the discriminator output. The eye diagrams below are the results. The audio output is 0V for a sample value of -1.0 and +3.3V for a sample value of +1.0V. Therefore, +1.65V corresponds to a sample value of 0.0. The scope is adjusted to place +1.65V on the center graticule.
|Software duplicate of Flesher
TU-170. The data crossover points are on the 0.0 point, so we have no bias
|The tone filters here are narrower than those suggested in
Filters for RTTY. The eye is open, but alternating bits do not
reach the full discriminator output. If successive bits are the same,
the full output is reached. The data crossover points are on the 0.0 point, so we have no bias distortion.
|Same as above, tone filters are single biquads so bandwidth is closer
to ideal. The wider bandwidth lets single bits reach a higher voltage.
|Same as above, but a bandpass filter is added in front of the limiter.
TU Notes showed no bias distortion
due to pre-limiter filtering and selective fading with Q=10, but substantial
bias distortion with Q=20.
All the below tests were done with this audio, random data with 22 ms per bit and space 20.8 dB below mark to simulate selective fading.
The slow rise time resulting in less than full excursion on single bits could be due to the pre-limiter filter, the tone filters, or the data low pass filter. It appears that the major contributor is the data low pass filter. Disabling the pre-limiter filter made no difference. The image to the right is with the following settings.
The tone filter Q and data LPF cutoff frequency were adjusted so the peak excursion was about the same for single bits and sequential bits. The last two images (single sweep and eye diagram) were with the following parameters.
Having each bit result in the same excursion should allow us to capture the maximum excursion during each bit to determine a dynamic threshold.
|To get an idea of how heavily loaded the CPU is, RE7 is driven high when the DSP code is entered (every 125 us), and set low when the code is exited. The code then loops until the next 125 us time is up (timer 2 running at 80 kHz and a software counter counting 10 timer 2 timeouts for a sampling frequency of 8 kHz. The 80 kHz is the raw PWM frequency on the audio output. The duty cycle is updated 8,000 times a second. The image at the right shows that the CPU is spending about 3.7 percent of the time executing the DSP code. There is some interrupt code not included in this, but it is minor (timer 2 timeout decrementing a software counter). The code running is that described above plus the DDS tone generator.|
Lots of changes of the last few days! First off, when the project was previously renamed to DSP_TU, some files and directories changed, but not all. After spending a couple days trying to fix it, I created a new project named DSP_TU, went through all the configuration (see Harmony at the top of the software section) and copied the source code files into the new project. It finally looks good!
Added an AGC after the input filter. This is a simple AGC that compares the full wave rectified and filtered level to a target level and applies a gain. To keep it from running away, a maximum gain is specified.
Added Dynamic Threshold Control. similar to "slideback" compensation as discussed in A New Approach to TU Design Using a Limiterless Two-Tone Method. The analog circuitry discussed there provided a threshold (actually biased the bipolar envelope) such that the mark/space switch was half way between the peak mark and space voltages. The "slideback" voltages were held in capacitors that quickly charged to the peak mark or space envelope voltage and slowly discharged to follow the varying peak envelope voltage. This approach had a problem with hand keyed RTTY since there was a long delay between space signals which would adjust the threshold as though the space envelope was low. In this software version, as each bit is received, a maximum level is determined. The threshold is (MarkMax - SpaceMax)/2 passed through a low pass filter. The low pass filter keeps the threshold from changing instantly on each bit but, ideally fast enough to deal with selective fading. Originally, the code compared the current MarkLevel-SpaceLevel to the LastThreshold (that which was last calculated) to determine if the received signal was mark or space so the maximum level calculation ended up properly in a max for mark or space. However, if the LastThreshold gets messed up (so that MarkLevel-SpaceLevel) never crosses the threshold, the threshold updating stops. For that reason, the mark/space decision here is done just based on which level is higher.
The various parameters were adjusted. With a maximum AGC gain of 50 and a dynamic threshold low pass filter cutoff of 50 Hz, the test described here resulted in about 84 lines printing correctly. This is quite a bit better than DSP TU 220615 on that page (67 lines correct, the same as the Flesher TU-170).
The overall design at this point is:
Analog to digital converter output.
Output of the input bandpass filter.
Output of the automatic gain control.
Output of the mark tone filter.
Output of the space tone filter.
Output of the mark demodulator (full wave rectifier followed by data LPF).
Output of the space demodulator (full wave rectifier followed by data LPF).
Output of the discriminator (mark demod minus space demod) at tone start. Initial clipping due to high AGC gain.
Output of the discriminator (mark demod minus space demod) once the AGC adjusts to the input level.
Output of the dynamic threshold detector. The TU was fed with real RTTY received over an SDR so there would be some selective fading.
Discriminator less dynamic threshold. Above 0.0 (center scale) should be mark and below space. The TU was fed with real RTTY received over an SDR so there would be some selective fading.
Discriminator less threshold fed with random 22 ms data to generate eye pattern.
With the added code, the CPU load has increased to about 5%.
This commit adds an XY tuning scope to the terminal unit. The output of the mark BPF causes deflection in the X direction, while the output of the space BPF causes deflection in the Y direction.
The display is a 128x128 OLED display with an SPI interface. The display comes with no doucmentation other than it uses the SSD1351 controller. The SSD1351 supports a wide variety of displays and has registers that are set up during initialization for that particular display. buyDisplay.com offers a similar display, also a 150x150 OLED display with the SSD1351 controller. There is also little documentation for that display, but they DO provide 8051 demo code. That code bit bangs the SPI and drives CSn high after each transmitted byte, which conflicts with the datasheet where CSn is held low through the whole transaction (command and data).
The display can operate as either a 3 wire or 4 wire SPI device. The display is write only, so the MISO input of the SPI port on the PIC is not used. In 4 wire mode, the wires are CSn (active low chip select), clock, MOSI, and Data/Commandn. Data/Commandn is low for the first byte of a command and high for the following data bytes.
The display can also operate in a 3 wire mode. In this mode, each byte is transmitted as 9 bits with the first bit being the DataCommandn bit. It is low for the first byte of the command and high for the remaining data bits. Since the PIC does not support 9 bit SPI, the 3 wire mode was not used.
The hardware SPI1 module in the PIC32 is used to drive the display. The SPI clock runs at 25 MHz. To avoid having the system wait for the transmission of each byte over SPI, a large 16 bit wide FIFO is used. The lower 8 bits contain the byte to be transmitted over SPI. The upper 8 bits hold the state of the Data/Commandn line for this byte along with with the state of the CSn line. The buffer holds all the information to properly transmit a byte. The main loop polls the SPI in a loop pass where there was no audio sample to process. The SPI poll checks to see if the SPI port is busy. If not, it checks to see if there is data in the FIFO to transmit. If so, it sets CSn and Data/Commandn as indicated in the high half of the FIFO word, then puts the low half in the SPI transmit buffer. If the buffer entry has CSn false, the low byte is not transmitted. See DisplayPoll() in display.c for details.
To allow the display to be blanked, the display FIFO has to be relatively deep. The display address pointer is set to the first position on the display, then black is written to 128 lines and columns (16,384 pixels). Writing each pixel takes three bytes (the command followed by the 16 bit color as shown below). This requires the FIFO to be about 49,000 words deep (recall that each 16 bit word holds the 8 bits of data to be sent along with the state of the chip select and Data/Commandn pin). The FIFO depth is set to 70,000 in display.c. Since the FIFO is on the heap, the heap size had to be increased to 200,000 bytes.
The format of the color data sent to a pixel is also not documented. Through experimentation, it was determined that each pixel color is a 16 bit values with the bits assigned as:
The addition of the display code increased the CPU utilization from the previous 5% to about 12%.
The video to the right shows the tuning display in operation. The system is receiving ITTY. For each audio sample (8,000 per second), the sample value out of the mark BPF filter determines the X position of the pixel to be lit, and the sample value out of the space BPF determines the Y position of the pixel to be lit. This is the same approach used with an analog CRT-based tuning scope. Lighting a pixel for 1/8000 of a second (125 microseconds) would result in a very dim display. A CRT display has persistence where the phosphor continues to glow after being hit by the electron beam. The light output from the phosphor dims over time. Here, to simulate persistence, a pixel is not turned off (rewritten with the code for black) until 1024 samples (128ms) later. This approximates CRT persistence but does not duplicate it since the CRT phosphors dim over the persistence period while here the pixel stays full brightness and then turns off.
Notice also that the pattern is a cross of lines instead of elipses. This is due to the narrow bandwidth of the mark and space bandpass filters. Elipses on other terminal units are due to "leakage" where the mark signal gets through the space filter (and vice versa) but is attenuated. Here, the opposite tone is attenuated more due to the narrow bandwidth of the filters. As described above, the filter bandwidth was made just wide enough for a mark to space transition (or vice versa) to reach full amplitude by the middle of the bit time. Alternating mark and space signals reach full amplitude, while they would not with a narrower filter.
Note also the random flickering pixel pattern in the four quadrants. This is due to the rise time and decay time of the mark and space bandpass filters. On a mark to space transition, the signal transitions from all out of the mark filter to all out of the space filter. During the transition, it comes out of both filters, thus causing the flickering pixels in the four quadrants defined by the mark and space lines.
Various hardware and software updates are included in this commit. For the latest hardware sketch, see DspTuHw221023.pdf. The latest software changes are shown here.
Here's a brief explanation of the latest hardware drawings.
The latest software updates, as shown here, are:
PTTn has been added to drive the transceiver PTT line. It is driven with a 5V tolerant open drain output. This will be driven by a "Keyboard Operated Relay" in the code. The KOR samples the loop current and if it finds the loop is in space and the loop is not being keyed by the TU, PTTn will be pulled low. A timer will hold PTTn low for several seconds after the last space. In addition, PTTn will drive the AFSK output when low. Since PTTn is open drain, this can be driven by either KOR or an external switch to ground.
I intend to add a command interpreter to the Baudot Uart. With an escape sequence, the TU will change from normal transmit/receive operation to command entry. Various options can be set, such as shift, KOR enable and time, mark hold threshold, etc.