kopia lustrzana https://github.com/ArjanteMarvelde/uSDR-pico
243 wiersze
6.9 KiB
C
243 wiersze
6.9 KiB
C
/*
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* dsp_tim.c
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* ==>TO BE INCLUDED IN dsp.c
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*
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* Created: May 2022
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* Author: Arjan te Marvelde
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*
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* Signal processing of RX and TX branch, to be run on the second processor core.
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* Each branch has a dedicated routine that must run on set times.
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* The period is determined by reads from the inter-core fifo, by the dsp_loop() routine.
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* This fifo is written from core0 from a 16us timer callback routine (i.e. 62.5kHz)
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*
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* The RX branch:
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* - Sample I and Q QSD channels, and shift into I and Q delay line (62.5 kHz per channel)
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* - Low pass filter: Fc=4kHz
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* - Quarter rate (15.625 kHz) to improve low freq behavior of Hilbert transform
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* - Calculate 15 tap Hilbert transform on Q
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* - Demodulate, taking proper delays into account
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* - Push to Audio output DAC
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*
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* Always perform audio sampling (62.5kHz) and level detections, in case of VOX active
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*
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* The TX branch (if VOX or PTT):
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* - Low pass filter: Fc=3kHz
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* - Eight rate (7.8125 kHz) to improve low F behavior of Hilbert transform
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* - Generate Q samples by doing a Hilbert transform
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* - Push I and Q to QSE output DACs
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*
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*/
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#include "uSDR.h"
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volatile int32_t q_sample, i_sample, a_sample; // Latest processed sample values
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/*
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* Low pass FIR filters Fc=3, 7 and 15 kHz (see http://t-filter.engineerjs.com/)
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* Settings: sample rates 62500, 31250 or 15625 Hz, stopband -40dB, passband ripple 5dB
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* Note: 8 bit precision, so divide sum by 256 (this could be improved when 32bit accumulator)
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*/
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int16_t lpf3_62[15] = { 3, 3, 5, 7, 9, 10, 11, 11, 11, 10, 9, 7, 5, 3, 3}; // Pass: 0-3000, Stop: 6000-31250
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int16_t lpf3_31[15] = { -2, -3, -3, 1, 10, 21, 31, 35, 31, 21, 10, 1, -3, -3, -2}; // Pass: 0-3000, Stop: 6000-15625
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int16_t lpf3_15[15] = { 3, 4, -3,-14,-15, 6, 38, 53, 38, 6,-15,-14, -3, 4, 3}; // Pass: 0-3000, Stop: 4500-7812
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int16_t lpf7_62[15] = { -2, -1, 1, 7, 16, 26, 33, 36, 33, 26, 16, 7, 1, -1, -2}; // Pass: 0-7000, Stop: 10000-31250
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int16_t lpf7_31[15] = { -1, 4, 9, 2,-12, -2, 40, 66, 40, -2,-12, 2, 9, 4, -1}; // Pass: 0-7000, Stop: 10000-15625
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int16_t lpf15_62[15] = { -1, 3, 12, 6,-12, -4, 40, 69, 40, -4,-12, 6, 12, 3, -1}; // Pass: 0-15000, Stop: 20000-31250
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/** CORE1: RX Branch **/
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/*
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* Execute RX branch signal processing
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* max time to spend is <64us (TIM_US)
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* The pre-processed I/Q samples are passed in i_sample and q_sample
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* The calculated A sample is passed in a_sample
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*/
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volatile int32_t i_s_raw[15], q_s_raw[15]; // Raw I/Q samples minus DC bias
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volatile int32_t i_s[15], q_s[15]; // Filtered I/Q samples
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bool __not_in_flash_func(rx)(void)
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{
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int32_t q_accu, i_accu;
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int32_t qh;
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uint16_t i;
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/*** SAMPLING ***/
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/*
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* Shift-in I and Q raw samples
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*/
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for (i=0; i<14; i++) // Store preprocessed samples in shift registers
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{
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q_s_raw[i] = q_s_raw[i+1];
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i_s_raw[i] = i_s_raw[i+1];
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}
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q_s_raw[14] = q_sample;
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i_s_raw[14] = i_sample;
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/*
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* Low pass FIR filter
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*/
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q_accu = 0; // Initialize accumulators
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i_accu = 0;
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for (i=0; i<15; i++) // Low pass FIR filter
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{
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q_accu += (int32_t)q_s_raw[i]*lpf3_15[i]; // Fc=3kHz, at 15625 Hz sampling
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i_accu += (int32_t)i_s_raw[i]*lpf3_15[i]; // Fc=3kHz, at 15625 Hz sampling
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}
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q_accu = q_accu/256;
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i_accu = i_accu/256;
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for (i=0; i<14; i++) // Store filtered samples in shift registers
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{
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q_s[i] = q_s[i+1];
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i_s[i] = i_s[i+1];
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}
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q_s[14] = q_accu;
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i_s[14] = i_accu;
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/*** DEMODULATION ***/
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switch (dsp_mode)
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{
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case MODE_USB:
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/*
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* USB demodulate: I[7] - Qh,
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* Qh is Classic Hilbert transform 15 taps, 12 bits
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* (see Iowa Hills calculator)
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*/
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q_accu = (q_s[0]-q_s[14])*315L + (q_s[2]-q_s[12])*440L +
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(q_s[4]-q_s[10])*734L + (q_s[6]-q_s[ 8])*2202L;
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qh = q_accu / 4096L;
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a_sample = i_s[7] - qh;
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break;
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case MODE_LSB:
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/*
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* LSB demodulate: I[7] + Qh,
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* Qh is Classic Hilbert transform 15 taps, 12 bits
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* (see Iowa Hills calculator)
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*/
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q_accu = (q_s[0]-q_s[14])*315L + (q_s[2]-q_s[12])*440L +
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(q_s[4]-q_s[10])*734L + (q_s[6]-q_s[ 8])*2202L;
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qh = q_accu / 4096L;
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a_sample = i_s[7] + qh;
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break;
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case MODE_AM:
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/*
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* AM demodulate: sqrt(sqr(i)+sqr(q))
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* Approximated with mag(i,q)
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*/
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a_sample = mag(i_s[7], q_s[7]);
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break;
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default:
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break;
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}
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/*** AUDIO GENERATION ***/
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/*
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* Scale and clip output,
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* Send to audio DAC output
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*/
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a_sample = (a_sample/64) + DAC_BIAS; // -18dB and add bias level
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if (a_sample > DAC_RANGE) // Clip to DAC range
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a_sample = DAC_RANGE;
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else if (a_sample<0)
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a_sample = 0;
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return true;
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}
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/** CORE1: TX branch **/
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/*
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* Execute TX branch signal processing,
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* max time to spend is <64us (TIM_US)
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* The pre-processed audio sample is passed in a_sample
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* The calculated I and Q samples are passed in i_sample and q_sample
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*/
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volatile int16_t a_s_raw[15]; // Raw samples, minus DC bias
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volatile int16_t a_s[15]; // Filtered and decimated samplesvolatile int16_t
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bool __not_in_flash_func(tx)(void)
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{
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int32_t a_accu, q_accu;
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int16_t qh;
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int i;
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uint16_t i_dac, q_dac;
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/*** RAW Audio ***/
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for (i=0; i<14; i++) // and store in shift register
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a_s_raw[i] = a_s_raw[i+1];
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a_s_raw[14] = a_sample;
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/*** Low pass filter ***/
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a_accu = 0; // Initialize accumulator
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for (i=0; i<15; i++) // Low pass FIR filter, using raw samples
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a_accu += (int32_t)a_s_raw[i]*lpf3_15[i]; // Fc=3kHz, at 15.625 kHz sampling
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for (i=0; i<14; i++) // Shift decimated samples
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a_s[i] = a_s[i+1];
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a_s[14] = a_accu / 256; // Store rescaled accumulator
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/*** MODULATION ***/
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switch (dsp_mode)
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{
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case 0: // USB
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/*
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* qh is Classic Hilbert transform 15 taps, 12 bits
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* (see Iowa Hills calculator)
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*/
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q_accu = (a_s[0]-a_s[14])*315L + (a_s[2]-a_s[12])*440L +
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(a_s[4]-a_s[10])*734L + (a_s[6]-a_s[ 8])*2202L;
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qh = -(q_accu / 4096L); // USB: sign is negative
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break;
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case 1: // LSB
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/*
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* qh is Classic Hilbert transform 15 taps, 12 bits
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* (see Iowa Hills calculator)
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*/
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q_accu = (a_s[0]-a_s[14])*315L + (a_s[2]-a_s[12])*440L +
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(a_s[4]-a_s[10])*734L + (a_s[6]-a_s[ 8])*2202L;
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qh = q_accu / 4096L; // LSB: sign is positive
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break;
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case 2: // AM
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/*
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* I and Q values are identical
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*/
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qh = a_s[7];
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break;
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default:
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break;
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}
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/*
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* Write I and Q to QSE DACs, phase is 7 samples back.
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* Need to multiply AC with DAC_RANGE/ADC_RANGE (appr 1/8)
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* Any case: clip to range
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*/
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a_accu = DAC_BIAS - (qh/8);
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if (a_accu<0)
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q_sample = 0;
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else if (a_accu>(DAC_RANGE-1))
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q_sample = DAC_RANGE-1;
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else
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q_sample = a_accu;
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a_accu = DAC_BIAS + (a_s[7]/8);
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if (a_accu<0)
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i_sample = 0;
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else if (a_accu>(DAC_RANGE-1))
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i_sample = DAC_RANGE-1;
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else
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i_sample = a_accu;
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return true;
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}
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