[GreenKeys] Minimal DSP TU Working!

[William Wuttke]

Very cool Harold.  You are sparking my interest!

Bill
KE3BK

On Mon, Mar 14, 2022 at 7:03 PM Harold Hallikainen via GreenKeys <
greenkeys at mailman.qth.net> wrote:

> 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
>
>
>
>
>
> --
> FCC Rules Updated Daily at http://www.hallikainen.com
> Not sent from an iPhone.
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