[GreenKeys] Minimal DSP TU Working!

[W2HX]

Fascinating. What a cool project. Keep up the great work. Bringing TTY machines into the 21st century!


73 Eugene W2HX
Subscribe to my Youtube Channel: https://www.youtube.com/c/w2hx-channel/videos

-----Original Message-----
From: greenkeys-bounces at mailman.qth.net <greenkeys-bounces at mailman.qth.net> On Behalf Of Harold Hallikainen via GreenKeys
Sent: Monday, March 14, 2022 10:03 PM
To: Green Keys <greenkeys at mailman.qth.net>
Subject: [GreenKeys] Minimal DSP TU Working!

I've wanted to learn a bit about DSP, so a TU seemed like the perfect project. It is now driving my model 15 with ITTY!

https://drive.google.com/file/d/1VSPhojioKn0XA0mof5SyRESE7UNYKEyh/view?usp=sharing

This is running on a PIC32MZ. I plan on checking how much time it is spending doing stuff and perhaps it would run on the PIC32MX.

Here's my code as of right now:

int main ( void ){
    biquad *MarkFilter; // Audio BPF for mark
    biquad *SpaceFilter; // Audio BPF for space
    biquad *MarkDataFilter; // Mark LPF after filtered data squaring
    biquad *SpaceDataFilter;
    uint16_t AdcSample;
    double DdsFreq=100.0;
    /* Initialize all modules */
    SYS_Initialize ( NULL );
    TMR2_Start();
    OCMP1_Enable();
    AudioPwmSet(0.0);
    ADCHS_ChannelConversionStart(4);
    MarkFilter = BiQuad_new(BPF, 0.0, 2125.0, 8000.0, 50.0);    // Q-50
    SpaceFilter = BiQuad_new(BPF, 0.0, 2295.0, 8000.0, 50.0);
    // Create biquad BPF for mark and space, 0 dB gain, 2.125 or 2.296 kHz, 8 kHz sample freq,, 50 Jz BW
    MarkDataFilter=BiQuad_new(LPF, 0.0, 136.0, 8000.0, 0.707 );
    SpaceDataFilter=BiQuad_new(LPF, 0.0, 136.0, 8000.0, 0.707 );

    while ( true )
    {
        if(Timer2TimeoutCounter==0){        // We have timed out 10 times,
so it has been 125 us
            if(ADCHS_ChannelResultIsReady(4)){
                AdcSample=ADCHS_ChannelResultGet(4);
                AdcSamplef=(double)(AdcSample-2048)/2048.0;
                ADCHS_ChannelConversionStart(4);
            }
            // Tone Sweep
            DdsFreq*=1.00001;
            if(DdsFreq>2295.0) DdsFreq=2125.0;
            DdsFreqSet(DdsFreq);
            // End tone sweep
            //sample=DdsNextSample();
            MarkSample=BiQuad(AdcSamplef,MarkFilter);
            SpaceSample=BiQuad(AdcSamplef,SpaceFilter);
            DiscrimOut=BiQuad(fabs(MarkSample),
MarkDataFilter)-BiQuad(fabs(SpaceSample), SpaceDataFilter);
            if(DiscrimOut>=0){      // Mark
                LATDbits.LATD4=1;
                LATBbits.LATB1=0;   // light green LED
                LATBbits.LATB5=1;   // red LED off
            }else{
                LATDbits.LATD4=0;
                LATBbits.LATB1=1;       // green LED off
                LATBbits.LATB5=0;       // red LED on
            }
            AudioPwmSet(AdcSamplef);
            Timer2TimeoutCounter=10;  // come back in 125 us. PWM frequency is 80 kHz, so change every 10 cycles
        }
        /* Maintain state machines of all polled MPLAB Harmony modules. */
        SYS_Tasks ( );
    }


Timer 2 is generating 80 kHz PWM. Every 10 times around, the ADC is read and started again. The ADC is converted to a double precision float called AdcSamplef. This runs through a bandpass filter for mark and another for space. The absolute values of the outputs of the filters are computed (same as full wave rectification). The results go into a low pass filter for mark and another for space. If the absolute value of the mark LPF is greater than that of the space LPF, the SSR is turned on causing loop current (mark). Otherwise, the loop current is interrupted (space).

There is also some Direct Digital Synthesis code in there for test. When a frequency is set, DdsRadiansPerSample is set telling the DDS code how many radians to add to the last value, then do a sine lookup. The result of the DDS can be sent to the audio output (for transmit audio).

The audio output is 80 kHz PWM generated by timer 2 and an output compare module. The duty cycle is updated at 8 kHz. The PWM goes to a single pole RC LPF with a cutoff of 4 kHz. This is followed by a 2 pole active LPF with the same cutoff. For testing right now, the AdcSamplef is being fed to the audio output. I can hear it on speakers and see it on the scope.

Next steps:

1. I plan to do a long transmission from punched tape and record the audio on an SDR (probably KFS). I will then vary filter parameters and see what gives me the lowest error rate with noise and QSB.

2. I'll get the transmit side running. That will include an opto detecting loop current which will then FSK the DDS. I'll put an LPF between the data and the DDS to minimize keyclicks.

Fun stuff!

Harold
https://w6iwi.org





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