kopia lustrzana https://github.com/threeme3/usdx
Added attenuator on ADC input. Prevent RX leaking into microphone (AREF) during TX.
guido
rodzic
8f5ba3ad7b
commit
e7f8bd8650
168
QCX-SSB.ino
168
QCX-SSB.ino
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@ -4,6 +4,22 @@
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#define VERSION "1.01l"
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/*
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code definitions and re-use for comb, integrator, dc decoupling, arctan
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in func_ptr for different mode types
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agc based on rms256, agc/smeter after filter
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vox by sampling mic port in sdr_rx?
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tx mox in menu
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noisefree integrator (rx audio out) in lower range
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raised cosine tx amp for cw
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auto paddle
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cw tx message/cw encoder
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32 bin fft
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menu back (right button), menu enter (rotary button)
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dynamic range cw
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att extended agc
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*/
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// QCX pin defintion
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#define LCD_D4 0
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#define LCD_D5 1
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@ -42,17 +58,19 @@ public: // QCXLiquidCrystal extends LiquidCrystal library for pull-up driven LCD
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#define FAST_RS 1
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#ifdef FAST_RS // RFI quiet transfer (LCD_RS pull-up is causing RFI). Assumes LCD_D4..LCD_D7, LCD_EN are connected to PD0..PD4 and LCD_RS to PC4
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virtual size_t write(uint8_t value){ // overwrites LiquidCrystal::write() and re-implements LCD data writes
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uint8_t rshi = DDRC & ~(0x10);
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uint8_t rslo = DDRC | 0x10;
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uint8_t hi = (PORTD & 0xe0) | 0x10 | (value >> 4);
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uint8_t lo = (PORTD & 0xe0) | 0x10 | (value & 0x0f);
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PORTD = hi; // LCD_D4..D7=value, LCD_EN=1 enable pulse must be >450ns
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DDRC = 0x00; // LCD_RS=1 pullup //PORTC = 0x10; // LCD_RS=1
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PORTD = hi; // LCD_D4..D7=value, LCD_EN=1 enable pulse must at least 100ns
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DDRC = rshi; //0x00; // LCD_RS=1 pullup //PORTC = 0x10; // LCD_RS=1
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PORTD &= ~0x10; // LCD_EN=0
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PORTD = lo; // LCD_D4..D7=value (connecting to PD0..PD4), LCD_EN=1 enable pulse must be >450ns
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DDRC = 0x10; // LCD_RS=0 pulldown //PORTC = 0x00; // LCD_RS=0
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DDRC = 0x00; // LCD_RS=1 pullup //PORTC = 0x10; // LCD_RS=1
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PORTD = lo; // LCD_D4..D7=value (connecting to PD0..PD4), LCD_EN=1 enable pulse must at least 100ns
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DDRC = rslo; //0x10; // LCD_RS=0 pulldown //PORTC = 0x00; // LCD_RS=0
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DDRC = rshi; //0x00; // LCD_RS=1 pullup //PORTC = 0x10; // LCD_RS=1
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PORTD &= ~0x10; // LCD_EN=0
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DDRC = 0x10; // LCD_RS=0 pulldown //PORTC = 0x00; // LCD_RS=0
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delayMicroseconds(60); // tcycE [https://www.sparkfun.com/datasheets/LCD/HD44780.pdf, figure 25]
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DDRC = rslo; //0x10; // LCD_RS=0 pulldown //PORTC = 0x00; // LCD_RS=0
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delayMicroseconds(60); // tcycE [https://www.sparkfun.com/datasheets/LCD/HD44780.pdf, figure 25]
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return 1;
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}
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#else
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@ -349,6 +367,7 @@ public:
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prev_divider = divider;
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SendRegister(SI_PLL_RESET, 0xA0);
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}
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//SendRegister(24, 0b00000000); // CLK3-0 Disable State: CLK2=0 (BE CAREFUL TO CHANGE THIS!!!), CLK1,0=00 -> IC4-X0 selected -> 2,5V on IC5A/3(+) -> 12V on IC5A/1, IC6A/2(-) -> 0V on IC6A/1, AUDIO2
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}
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void alt_clk2(uint32_t freq)
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{
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@ -382,6 +401,11 @@ public:
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};
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static SI5351 si5351;
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volatile uint16_t param_a = 0;
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volatile int16_t param_b = 0;
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volatile int16_t param_c = 0;
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enum mode_t { LSB, USB, CW, AM, FM };
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volatile int8_t mode = USB;
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const char* mode_label[] = { "LSB", "USB", "CW ", "AM ", "FM " };
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@ -394,7 +418,7 @@ enum dsp_cap_t { ANALOG, DSP, SDR };
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static uint8_t dsp_cap = 0;
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static uint8_t ssb_cap = 0;
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volatile uint8_t att = 0;
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const char* att_label[] = { "0dB", "-13dB", "-20dB", "-33dB" };
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const char* att_label[] = { "0dB", "-13dB", "-20dB", "-33dB", "-40dB", "-53dB", "-60dB", "-73dB" };
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//#define PROFILING 1
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#ifdef PROFILING
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volatile uint32_t numSamples = 0;
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@ -408,7 +432,11 @@ inline void txen(bool en)
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lcd.setCursor(15, 1); lcd.print("T");
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si5351.SendRegister(SI_CLK_OE, 0b11111011); // CLK2_EN=1, CLK1_EN,CLK0_EN=0
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digitalWrite(RX, LOW); // TX
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pinMode(AUDIO1, OUTPUT); // prevent RX leakage into AREF
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pinMode(AUDIO2, OUTPUT); // prevent RX leakage into AREF
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} else {
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pinMode(AUDIO1, INPUT); // prevent RX leakage into AREF
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pinMode(AUDIO2, INPUT); // prevent RX leakage into AREF
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digitalWrite(RX, !(att == 2)); // RX (enable RX when attenuator not on)
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si5351.SendRegister(SI_CLK_OE, 0b11111100); // CLK2_EN=0, CLK1_EN,CLK0_EN=1
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lcd.setCursor(15, 1); lcd.print((vox_enable) ? "V" : "R");
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@ -504,6 +532,7 @@ inline int16_t ssb(int16_t in)
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}
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#define MIC_ATTEN 0 // 0*6dB attenuation (note that the LSB bits are quite noisy)
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//volatile int8_t mon = 0;
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// This is the ADC ISR, issued with sample-rate via timer1 compb interrupt.
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// It performs in real-time the ADC sampling, calculation of SSB phase-differences, calculation of SI5351 frequency registers and send the registers to SI5351 over I2C.
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@ -517,8 +546,40 @@ void dsp_tx()
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int16_t df = ssb(adc >> MIC_ATTEN); // convert analog input into phase-shifts (carrier out by periodic frequency shifts)
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si5351.freq_calc_fast(df); // calculate SI5351 registers based on frequency shift and carrier frequency
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numSamples++;
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//if(mon) OCR1AL = (adc << (mon-1)) + 128; // TX audio monitoring
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}
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volatile int16_t p_sin = 0; // initialized with A*sin(t), where t=0
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volatile int16_t n_cos = 448; // initialized with A*cos(t), where t=0
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inline void process_minsky() // Minsky circle sample [source: https://www.cl.cam.ac.uk/~am21/hakmemc.html, ITEM 149]: p_sin+=n_cos*2*PI*f/fs; n_cos-=p_sin*2*PI*f/fs;
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{
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p_sin += (n_cos*64)/((uint8_t)param_c); // set param_c=79
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n_cos -= (p_sin*64)/((uint8_t)param_c);
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}
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volatile uint16_t acc;
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void dsp_tx_cw()
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{ // jitter dependent things first
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/*
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ADCSRA |= (1 << ADSC); // start next ADC conversion (trigger ADC interrupt if ADIE flag is set)
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OCR1BL = amp; // submit amplitude to PWM register (actually this is done in advance (about 140us) of phase-change, so that phase-delays in key-shaping circuit filter can settle)
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si5351.SendPLLBRegisterBulk(); // submit frequency registers to SI5351 over 731kbit/s I2C (transfer takes 64/731 = 88us, then PLL-loopfilter probably needs 50us to stabalize)
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//OCR1BL = amp; // submit amplitude to PWM register (takes about 1/32125 = 31us+/-31us to propagate) -> amplitude-phase-alignment error is about 30-50us
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int16_t adc = ADC - 512; // current ADC sample 10-bits analog input, NOTE: first ADCL, then ADCH
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int16_t df = ssb(adc >> MIC_ATTEN); // convert analog input into phase-shifts (carrier out by periodic frequency shifts)
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si5351.freq_calc_fast(df); // calculate SI5351 registers based on frequency shift and carrier frequency
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numSamples++;*/
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//if(mon) OCR1AL = (adc << (mon-1)) + 128; // TX audio monitoring
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//process_minsky();
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//OCR1AL = (p_sin >> (16 - param_b)) + 128;
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acc = acc + param_c; // param_c = 7570
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int8_t temp = acc >> 8;
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int8_t mask = temp >> 7;
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OCR1AL = temp ^ mask;
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}
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volatile int8_t cwdec = 0;
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@ -576,7 +637,12 @@ static char out[] = " ";
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volatile bool cw_event = false;
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//#define F_SAMP_RX 78125
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//#define DUPLEX 1
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#ifdef DUPLEX
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#define F_SAMP_RX (2*62500) //overrun; sample rate of 55500 can be obtained
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#else
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#define F_SAMP_RX 62500 //overrun; sample rate of 55500 can be obtained
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#endif
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//#define F_SAMP_RX 52083
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//#define F_SAMP_RX 44643
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//#define F_SAMP_RX 39062
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@ -587,7 +653,7 @@ volatile bool cw_event = false;
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volatile int8_t volume = 8;
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volatile bool agc = true;
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volatile bool nr = false;
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volatile uint8_t nr = 0;
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//static uint32_t gain = 1024;
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static int16_t gain = 1024;
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@ -627,16 +693,13 @@ inline int16_t process_nr_old2(int16_t ac)
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return ea1;
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}
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volatile uint16_t param_a = 0;
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volatile int16_t param_b = 0;
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volatile int16_t param_c = 0;
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inline int16_t process_nr(int16_t in)
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{
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{
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/*
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static int16_t avg;
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avg = EA(avg, abs(in), 64); // alpha=1/64=0.0156
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param_c = avg;
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*/
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/*
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int32_t _avg = 64 * avg;
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@ -651,7 +714,7 @@ param_c = avg;
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else inv_gain = 32768;*/
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static int16_t ea1;
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ea1 = EA(ea1, in, 1 << param_b );
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ea1 = EA(ea1, in, 1 << (nr-1) );
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//static int16_t ea2;
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//ea2 = EA(ea2, ea1, inv_gain);
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@ -715,18 +778,6 @@ inline int16_t filt_var(int16_t v)
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return zx0;
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}
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/*
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iterate through modes always from begin, skipping the current one
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code definitions and re-use for comb, integrator, dc decoupling, arctan
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add sdr_rx capability to skip ADCMUX changing and Q processing, (while oversampling I branch?)
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in func_ptr for different mode types
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agc based on rms256
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skip adc processing while smeter lcd update?
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vox by sampling mic port in sdr_rx?
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*/
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typedef void (*func_t)(void);
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#ifdef JUNK
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volatile func_t func_ptr = junk;
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@ -736,12 +787,31 @@ volatile func_t func_ptr;
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static uint32_t absavg256 = 0;
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volatile uint32_t _absavg256 = 0;
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volatile uint8_t admux[2];
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volatile uint8_t admux[3];
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volatile uint8_t _init;
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volatile int16_t ocomb, i, q, qh;
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#undef R // Decimating 2nd Order CIC filter
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#define R 4 // Rate change from 52.083/2 kSPS to 6510.4SPS, providing 12dB gain
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volatile uint16_t adc_a, adc_b, adc_c, adc_d;
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void sdr_rx_a()
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{
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//ADMUX = admux[2]; // set MUX for next conversion
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adc_a = ADC;
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//ADCSRA |= (1 << ADSC); // start next ADC conversion
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func_ptr = sdr_rx_2; // processing function for next conversion
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}
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void sdr_rx_b()
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{
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ADMUX = admux[2]; // set MUX for next conversion
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adc_b = ADC;
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//delayMicroseconds(4);
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ADCSRA |= (1 << ADSC); // start next ADC conversion
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func_ptr = sdr_rx; // processing function for next conversion
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}
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// Non-recursive CIC Filter (M=2, R=4) implementation, so two-stages of (followed by down-sampling with factor 2):
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// H1(z) = (1 + z^-1)^2 = 1 + 2*z^-1 + z^-2 = (1 + z^-2) + (2) * z^-1 = FA(z) + FB(z) * z^-1;
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// with down-sampling before stage translates into poly-phase components: FA(z) = 1 + z^-1, FB(z) = 2
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@ -749,12 +819,19 @@ volatile int16_t ocomb, i, q, qh;
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void sdr_rx()
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{
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static int16_t zi1, zi2, zd1, zd2;
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// process I for even samples [75% CPU@R=4;Fs=62.5k] (excluding the Comb branch and output stage)
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#ifdef DUPLEX
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adc_c = ADC;
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ADMUX = admux[1]; // set MUX for next conversion
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int16_t adc = adc_b - 512; // current ADC sample 10-bits analog input, NOTE: first ADCL, then ADCH
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ADCSRA |= (1 << ADSC); // start next ADC conversion
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func_ptr = sdr_rx_a; // processing function for next conversion
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#else
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ADMUX = admux[1]; // set MUX for next conversion
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int16_t adc = ADC - 512; // current ADC sample 10-bits analog input, NOTE: first ADCL, then ADCH
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ADCSRA |= (1 << ADSC); // start next ADC conversion
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func_ptr = sdr_rx_2; // processing function for next conversion
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int16_t adc = ADC - 512; // current ADC sample 10-bits analog input, NOTE: first ADCL, then ADCH
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#endif
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sdr_rx_common();
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// Only for I: correct I/Q sample delay by means of linear interpolation
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@ -850,10 +927,18 @@ void sdr_rx()
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void sdr_rx_2()
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{
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// process Q for odd samples [75% CPU@R=4;Fs=62.5k] (excluding the Comb branch and output stage)
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#ifdef DUPLEX
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adc_d = ADC;
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ADMUX = admux[0]; // set MUX for next conversion
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int16_t adc = adc_a - 512; // current ADC sample 10-bits analog input, NOTE: first ADCL, then ADCH
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ADCSRA |= (1 << ADSC); // start next ADC conversion
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func_ptr = sdr_rx_b; // processing function for next conversion
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#else
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ADMUX = admux[0]; // set MUX for next conversion
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int16_t adc = ADC - 512; // current ADC sample 10-bits analog input, NOTE: first ADCL, then ADCH
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ADCSRA |= (1 << ADSC); // start next ADC conversion
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func_ptr = sdr_rx; // processing function for next conversion
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int16_t adc = ADC - 512; // current ADC sample 10-bits analog input, NOTE: first ADCL, then ADCH
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#endif
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static int16_t dc;
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dc += (adc - dc) / 2;
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@ -896,12 +981,14 @@ inline void sdr_rx_common()
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ozi2 = ozi1 + ozi2; // Integrator section
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ozi1 = ocomb + ozi1;
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#ifndef PROFILING
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if(volume) OCR1AL = min(max((ozi2>>5) + ICR1L/2, 0), ICR1L); // center and clip wrt PWM working range
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if(volume) OCR1AL = min(max((ozi2>>5) + 128, 0), 255); //if(volume) OCR1AL = min(max((ozi2>>5) + ICR1L/2, 0), ICR1L); // center and clip wrt PWM working range
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//OCR1AL = adc_c;
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#endif
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}
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ISR(TIMER2_COMPA_vect) // Timer2 COMPA interrupt
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{
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//interrupts();
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func_ptr();
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}
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@ -1142,10 +1229,14 @@ public:
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pinMode(KEY_OUT, OUTPUT);
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pinMode(BUTTONS, INPUT); // L/R/rotary button
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pinMode(DIT, INPUT_PULLUP);
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pinMode(DAH, INPUT);
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pinMode(DAH, INPUT); //pinMode(DAH, INPUT_PULLUP); // Could this replace D4?
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digitalWrite(AUDIO1, LOW); // when used as output, help can mute RX leakage into AREF
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digitalWrite(AUDIO2, LOW);
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pinMode(AUDIO1, INPUT);
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pinMode(AUDIO2, INPUT);
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}
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int8_t smode = 1;
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@ -1376,6 +1467,7 @@ public:
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_init = 1;
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numSamples = 0;
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func_ptr = sdr_rx; //enable RX DSP/SDR
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adc_start(2, true, F_ADC_CONV); admux[2] = ADMUX;
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if(dsp_cap == SDR){
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adc_start(0, !(att == 1)/*true*/, F_ADC_CONV); admux[0] = ADMUX;
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adc_start(1, !(att == 1)/*true*/, F_ADC_CONV); admux[1] = ADMUX;
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@ -1393,7 +1485,7 @@ public:
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adc_stop(); //disable RX DSP
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tx = 0; // transit from RX to TX
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numSamples = 0;
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func_ptr = dsp_tx;
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func_ptr = (mode == CW) ? dsp_tx_cw : dsp_tx;
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amp = 0; // initialize
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adc_start(2, true, F_ADC_CONV);
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timer2_start(F_SAMP_TX);
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@ -1559,8 +1651,8 @@ void paramAction(uint8_t action, uint8_t id = ALL) // list of parameters
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case BAND: paramAction(action, qcx.bandval, F("A4"), F("Band"), qcx.band_label, 0, sizeof(qcx.band_label)/sizeof(char*) - 1, false, 1); break;
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case STEP: paramAction(action, qcx.stepsize, F("A5"), F("Tuning step"), /*qcx.*/stepsize_label, 0, sizeof(/*qcx.*/stepsize_label)/sizeof(char*) - 1, false, qcx.STEP_1k); break;
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case AGC: paramAction(action, agc, F("A6"), F("AGC"), offon_label, 0, 1, false, true); break;
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case NR: paramAction(action, nr, F("A7"), F("NR"), offon_label, 0, 1, false, false); break;
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case ATT: paramAction(action, att, F("A8"), F("ATT"), att_label, 0, 3, false, 0); break;
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case NR: paramAction(action, nr, F("A7"), F("NR"), NULL, 0, 8, false, 0); break;
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case ATT: paramAction(action, att, F("A8"), F("ATT"), att_label, 0, 7, false, 0); break;
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case SMETER: paramAction(action, qcx.smode, F("A9"), F("S-meter"), qcx.smode_label, 0, sizeof(qcx.smode_label)/sizeof(char*) - 1, false, 1); break;
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case CWDEC: paramAction(action, cwdec, F("B1"), F("CW Decoder"), offon_label, 0, 1, false, false); break;
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case VOX: paramAction(action, vox_enable, F("C1"), F("VOX"), offon_label, 0, 1, false, false); break;
|
||||
|
|
@ -1942,6 +2034,8 @@ void loop()
|
|||
adc_start(1, !(att & 0x01), F_ADC_CONV); admux[1] = ADMUX;
|
||||
}
|
||||
digitalWrite(RX, !(att & 0x02)); // RX (enable RX when attenuator not on), otherwise keep Q5 off (attenuate)
|
||||
pinMode(AUDIO1, (att & 0x04) ? OUTPUT : INPUT);
|
||||
pinMode(AUDIO2, (att & 0x04) ? OUTPUT : INPUT);
|
||||
}
|
||||
}
|
||||
}
|
||||
|
|
|
|||
|
|
@ -33,7 +33,7 @@ pe1nnz@amsat.org
|
|||
- Possibility to extend the QCX analog phasing stage with a **DSP stage**
|
||||
- Could replace the QCX analog phasing stage completely with a **digital SDR receiver stage**, taking away the need for the manual side-band rejection adjustment procedure and delivering DSP features such as the joy of having a **AGC, adjustable CW/SSB filters**.
|
||||
- A theoretical **digital receiver dynamic range of 83dB** at 2.4kHz BW. (1 dB) Compression point (at -126dBm sensitivity): -44dBm/1mV (for in-band signal); -4dBm/160mV (for signal at 15kHz offset); 19dBm/2V (for signal at 100kHz offset or more).
|
||||
- Switchable RF/IF attenuator steps: 0dB, -13dB, -20dB, -33dB
|
||||
- Digitally switchable RF front-end **attenuators (0dB, -13dB, -20dB, -33dB, -53dB, -60dB, -73dB)**.
|
||||
- SDR implementation **simplifies** the receiver heaviliy and **shaves off roughly 30% of the components** from the original QCX design while adding new and improving existing features. On a new QCX build: 46 components less to be installed, 8 component design changes, 9 additional wires.
|
||||
- Can be used as alternate firmware on an unmodified QCX.
|
||||
|
||||
|
|
|
|||
Ładowanie…
Reference in New Issue