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Filters reworked for new samplerate
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Mark Qvist
parent
74e0b0d1db
commit
56bed68143
5
device.h
5
device.h
@ -14,8 +14,6 @@
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// Sampling & timer setup
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#define CONFIG_SAMPLERATE 19200UL
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//#define CONFIG_SAMPLERATE 19200UL
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//#define CONFIG_SAMPLERATE 9600
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// Serial settings
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#define BAUD 115200
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@ -25,6 +23,9 @@
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// CSMA Settings
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#define CONFIG_CSMA_P 255
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// Packet settings
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#define CONFIG_PASSALL false
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// Port settings
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#if TARGET_CPU == m1284p
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#define ADC_PORT PORTA
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@ -375,54 +375,53 @@ void AFSK_adc_isr(Afsk *afsk, int8_t currentSample) {
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afsk->iirX[0] = afsk->iirX[1];
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#if CONFIG_SAMPLERATE == 9600
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#if FILTER_CUTOFF == 600
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afsk->iirX[1] = ((int8_t)fifo_pop(&afsk->delayFifo) * currentSample) >> 2;
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// The above is a simplification of:
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// afsk->iirX[1] = ((int8_t)fifo_pop(&afsk->delayFifo) * currentSample) / 3.558147322;
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#elif FILTER_CUTOFF == 800
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afsk->iirX[1] = ((int8_t)fifo_pop(&afsk->delayFifo) * currentSample) >> 2;
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// The above is a simplification of:
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// afsk->iirX[1] = ((int8_t)fifo_pop(&afsk->delayFifo) * currentSample) / 2.899043379;
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#elif FILTER_CUTOFF == 1200
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afsk->iirX[1] = ((int8_t)fifo_pop(&afsk->delayFifo) * currentSample) >> 1;
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// The above is a simplification of:
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// afsk->iirX[1] = ((int8_t)fifo_pop(&afsk->delayFifo) * currentSample) / 2.228465666;
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#elif FILTER_CUTOFF == 1600
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afsk->iirX[1] = ((int8_t)fifo_pop(&afsk->delayFifo) * currentSample) >> 1;
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// The above is a simplification of:
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// afsk->iirX[1] = ((int8_t)fifo_pop(&afsk->delayFifo) * currentSample) / 1.881349100;
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#else
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#error Unsupported filter cutoff!
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#endif
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#elif CONFIG_SAMPLERATE == 19200
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#if FILTER_CUTOFF == 600
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afsk->iirX[1] = ((int8_t)fifo_pop(&afsk->delayFifo) * currentSample) / 6;
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#else
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#error Unsupported filter cutoff!
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#endif
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#else
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#error Unsupported samplerate!
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#endif
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afsk->iirY[0] = afsk->iirY[1];
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#if CONFIG_SAMPLERATE == 9600
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#if FILTER_CUTOFF == 600
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afsk->iirY[1] = afsk->iirX[0] + afsk->iirX[1] + (afsk->iirY[0] >> 1);
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// The above is a simplification of a first-order 600Hz chebyshev filter:
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// The above is a simplification of:
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// afsk->iirY[1] = afsk->iirX[0] + afsk->iirX[1] + (afsk->iirY[0] * 0.4379097269);
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#elif FILTER_CUTOFF == 800
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afsk->iirY[1] = afsk->iirX[0] + afsk->iirX[1] + (afsk->iirY[0] / 3);
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// The above is a simplification of a first-order 800Hz chebyshev filter:
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// afsk->iirY[1] = afsk->iirX[0] + afsk->iirX[1] + (afsk->iirY[0] * 0.3101172565);
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#elif FILTER_CUTOFF == 1200
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afsk->iirY[1] = afsk->iirX[0] + afsk->iirX[1] + (afsk->iirY[0] / 10);
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// The above is a simplification of a first-order 1200Hz chebyshev filter:
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// afsk->iirY[1] = afsk->iirX[0] + afsk->iirX[1] + (afsk->iirY[0] * 0.1025215106);
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#elif FILTER_CUTOFF == 1600
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afsk->iirY[1] = afsk->iirX[0] + afsk->iirX[1] + -1*(afsk->iirY[0] / 17);
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// The above is a simplification of a first-order 1600Hz chebyshev filter:
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// afsk->iirY[1] = afsk->iirX[0] + afsk->iirX[1] + (afsk->iirY[0] * -0.0630669239);
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#else
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#error Unsupported filter cutoff!
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#endif
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#elif CONFIG_SAMPLERATE == 19200
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#if FILTER_CUTOFF == 600
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afsk->iirY[1] = afsk->iirX[0] + afsk->iirX[1] + (afsk->iirY[0] / 2);
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#else
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#error Unsupported filter cutoff!
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#endif
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#else
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#error Unsupported samplerate!
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#endif
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//int8_t freq_disc = (int8_t)fifo_pop(&afsk->delayFifo) * currentSample;
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// We put the sampled bit in a delay-line:
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// First we bitshift everything 1 left
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afsk->sampledBits <<= 1;
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// And then add the sampled bit to our delay line
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afsk->sampledBits |= (afsk->iirY[1] > 0) ? 0 : 1;
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//afsk->sampledBits |= (freq_disc > 0) ? 0 : 1;
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// Put the current raw sample in the delay FIFO
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fifo_push(&afsk->delayFifo, currentSample);
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@ -489,6 +488,7 @@ void AFSK_adc_isr(Afsk *afsk, int8_t currentSample) {
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// the last 3 sampled bits. If there is two or
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// more 1's, we will assume that the transmitter
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// sent us a one, otherwise we assume a zero
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uint8_t bits = afsk->sampledBits & 0x07;
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if (bits == 0x07 || // 111
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bits == 0x06 || // 110
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@ -498,14 +498,16 @@ void AFSK_adc_isr(Afsk *afsk, int8_t currentSample) {
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afsk->actualBits |= 1;
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}
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//// Alternative using five bits ////////////////
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// uint8_t bits = afsk->sampledBits & 0x0f;
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//// Alternative using six bits ////////////////
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// uint8_t bits = afsk->sampledBits & 0x3F;
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// uint8_t c = 0;
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// c += bits & BV(1);
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// c += bits & BV(2);
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// c += bits & BV(3);
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// c += bits & BV(4);
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// c += bits & BV(5);
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// c += bits & _BV(0);
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// c += bits & _BV(1);
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// c += bits & _BV(2);
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// c += bits & _BV(3);
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// c += bits & _BV(4);
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// c += bits & _BV(5);
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// if (c >= 3) afsk->actualBits |= 1;
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/////////////////////////////////////////////////
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@ -35,9 +35,11 @@ inline static uint8_t sinSample(uint16_t i) {
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#define BITS_DIFFER(bits1, bits2) (((bits1)^(bits2)) & 0x01)
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#define TRANSITION_FOUND(bits) BITS_DIFFER((bits), (bits) >> 1)
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// TODO: Maybe expand number of bits looked at here:
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// TODO: Maybe revert to only looking at two samples
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#define DUAL_XOR(bits1, bits2) ((((bits1)^(bits2)) & 0x03) == 0x03)
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#define SIGNAL_TRANSITIONED(bits) DUAL_XOR((bits), (bits) >> 2)
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#define QUAD_XOR(bits1, bits2) ((((bits1)^(bits2)) & 0x0F) == 0x0F)
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#define SIGNAL_TRANSITIONED(bits) QUAD_XOR((bits), (bits) >> 4)
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// #define SIGNAL_TRANSITIONED(bits) DUAL_XOR((bits), (bits) >> 2)
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#define CPU_FREQ F_CPU
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@ -53,16 +55,19 @@ inline static uint8_t sinSample(uint16_t i) {
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#define SAMPLESPERBIT (CONFIG_SAMPLERATE / BITRATE)
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#define TICKS_BETWEEN_SAMPLES ((((CPU_FREQ+FREQUENCY_CORRECTION)) / CONFIG_SAMPLERATE) - 1)
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// TODO: Calculate based on sample rate
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#define PHASE_INC SAMPLESPERBIT/8 // Nudge by an eigth of a sample each adjustment
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#define PHASE_BITS 8 // How much to increment phase counter each sample
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// TODO: Calculate based on sample rate [Done?]
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#define PHASE_BITS 8 // 8 // Sub-sample phase counter resolution
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#define PHASE_INC 1 // 1 // Nudge by above resolution for each adjustment
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#define PHASE_MAX (SAMPLESPERBIT * PHASE_BITS) // Resolution of our phase counter
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#define PHASE_THRESHOLD (PHASE_MAX / 2) // Target transition point of our phase window
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#define PHASE_MAX (SAMPLESPERBIT * PHASE_BITS) // 128 // Size of our phase counter
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// TODO: Test which target is best in real world
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#define PHASE_THRESHOLD (PHASE_MAX / 2)+3*PHASE_BITS // Target transition point of our phase window
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//#define PHASE_THRESHOLD (PHASE_MAX / 2) // 64 // Target transition point of our phase window
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#define DCD_TIMEOUT_SAMPLES CONFIG_SAMPLERATE/100
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#define DCD_MIN_COUNT CONFIG_SAMPLERATE/1600
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// TODO: Revamp filtering
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#if BITRATE == 1200
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#define FILTER_CUTOFF 600
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#define MARK_FREQ 1200
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@ -120,8 +125,12 @@ typedef struct Afsk
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int16_t iirX[2]; // IIR Filter X cells
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int16_t iirY[2]; // IIR Filter Y cells
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uint8_t sampledBits; // Bits sampled by the demodulator (at ADC speed)
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int8_t currentPhase; // Current phase of the demodulator
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#if SAMPLESPERBIT < 17
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uint16_t sampledBits; // Bits sampled by the demodulator (at ADC speed)
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#else
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#error Not enough space in sampledBits variable!
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#endif
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int16_t currentPhase; // Current phase of the demodulator
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uint8_t actualBits; // Actual found bits at correct bitrate
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volatile int status; // Status of the modem, 0 means OK
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@ -31,7 +31,7 @@ void ax25_poll(AX25Ctx *ctx) {
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while ((c = fgetc(ctx->ch)) != EOF) {
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if (!ctx->escape && c == HDLC_FLAG) {
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if (ctx->frame_len >= AX25_MIN_FRAME_LEN) {
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if (ctx->crc_in == AX25_CRC_CORRECT) {
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if (ctx->crc_in == AX25_CRC_CORRECT || CONFIG_PASSALL) {
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#if OPEN_SQUELCH == true
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LED_RX_ON();
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#endif
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