mirror of https://github.com/g4klx/MMDVM.git
174 lines
4.8 KiB
C++
174 lines
4.8 KiB
C++
/*
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* Copyright (C) 2020 by Jonathan Naylor G4KLX
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*
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* This program is free software; you can redistribute it and/or modify
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* it under the terms of the GNU General Public License as published by
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* the Free Software Foundation; either version 2 of the License, or
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* (at your option) any later version.
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*
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* This program is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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* GNU General Public License for more details.
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*
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* You should have received a copy of the GNU General Public License
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* along with this program; if not, write to the Free Software
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* Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
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*/
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#include "Config.h"
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#include "Globals.h"
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#include "AX25Demodulator.h"
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const float32_t DELAY = 0.000448F;
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const float32_t SAMPLE_RATE = 24000.0F;
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const float32_t SYMBOL_RATE = 1200.0F;
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const uint16_t DELAY_LEN = uint16_t((DELAY / (1.0F / SAMPLE_RATE)) + 0.5F);
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const float32_t SAMPLES_PER_SYMBOL = SAMPLE_RATE / SYMBOL_RATE;
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const float32_t PLL_LIMIT = SAMPLES_PER_SYMBOL / 2.0F;
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// XXX This is for the wrong sample rate
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const uint32_t LPF_FILTER_LEN = 96U;
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q15_t LPF_FILTER_COEFFS[] = {
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0, 1, 3, 5, 8, 11, 14, 17, 19, 20, 18, 14,
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7, -2, -16, -33, -53, -76, -101, -126, -151, -174, -194, -208,
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-215, -212, -199, -173, -133, -79, -10, 74, 173, 287, 413, 549,
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693, 842, 993, 1142, 1287, 1423, 1547, 1656, 1747, 1817, 1865, 1889,
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1889, 1865, 1817, 1747, 1656, 1547, 1423, 1287, 1142, 993, 842, 693,
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549, 413, 287, 173, 74, -10, -79, -133, -173, -199, -212, -215,
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-208, -194, -174, -151, -126, -101, -76, -53, -33, -16, -2, 7,
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14, 18, 20, 19, 17, 14, 11, 8, 5, 3, 1, 0,
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};
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// 64 Hz loop filter.
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// scipy.signal:
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// loop_coeffs = firwin(9, [64.0/(1200/2)], width = None,
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// pass_zero = True, scale = True, window='hann')
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//
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const uint32_t PLL_FILTER_LEN = 7U;
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float32_t PLL_FILTER_COEFFS[] = {3.196252e-02F, 1.204223e-01F, 2.176819e-01F, 2.598666e-01F, 2.176819e-01F, 1.204223e-01F, 3.196252e-02F};
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CAX25Demodulator::CAX25Demodulator(float32_t* coeffs, uint16_t length) :
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m_audioFilter(),
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m_audioState(),
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m_lpfFilter(),
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m_lpfState(),
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m_delayLine(NULL),
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m_delayPos(0U),
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m_nrziState(false),
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m_pllFilter(),
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m_pllState(),
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m_pllLast(false),
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m_pllBits(1U),
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m_pllCount(0.0F)
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{
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m_delayLine = new bool[2U * DELAY_LEN];
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m_audioFilter.numTaps = length;
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m_audioFilter.pState = m_audioState;
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m_audioFilter.pCoeffs = coeffs;
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m_lpfFilter.numTaps = LPF_FILTER_LEN;
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m_lpfFilter.pState = m_lpfState;
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m_lpfFilter.pCoeffs = LPF_FILTER_COEFFS;
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m_pllFilter.numTaps = PLL_FILTER_LEN;
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m_pllFilter.pState = m_pllState;
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m_pllFilter.pCoeffs = PLL_FILTER_COEFFS;
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}
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bool CAX25Demodulator::process(const q15_t* samples, uint8_t length, CAX25Frame& frame)
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{
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bool result = false;
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float32_t input[RX_BLOCK_SIZE];
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for (size_t i = 0; i < length; i++)
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input[i] = float(samples[i]);
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float32_t fa[RX_BLOCK_SIZE];
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::arm_fir_f32(&m_audioFilter, input, fa, RX_BLOCK_SIZE);
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int16_t buffer[RX_BLOCK_SIZE];
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for (uint8_t i = 0; i < length; i++) {
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int16_t sample = int16_t(fa[i]);
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bool level = (sample >= 0);
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bool delayed = delay(level);
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buffer[i] = (int16_t(level ^ delayed) << 1) - 1;
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}
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q15_t fc[RX_BLOCK_SIZE];
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::arm_fir_fast_q15(&m_lpfFilter, buffer, fc, RX_BLOCK_SIZE);
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for (uint8_t i = 0; i < length; i++) {
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bool bit = fc[i] >= 0;
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bool sample = PLL(bit);
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if (sample) {
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// We will only ever get one frame because there are
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// not enough bits in a block for more than one.
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if (result)
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hdlc_decoder_(NRZI(bit), true);
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else
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result = hdlc_decoder_(NRZI(bit), true);
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}
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}
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return result;
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}
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bool CAX25Demodulator::delay(bool b)
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{
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bool r = m_delayLine[m_delayPos];
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m_delayLine[m_delayPos++] = b;
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if (m_delayPos >= DELAY_LEN)
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m_delayPos = 0U;
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return r;
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}
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bool CAX25Demodulator::NRZI(bool b)
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{
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bool result = (b == m_nrziState);
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m_nrziState = b;
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return result;
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}
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bool CAX25Demodulator::PLL(bool input)
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{
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bool sample = false;
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if (input != m_pllLast or m_pllBits > 16U) {
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// Record transition.
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m_pllLast = input;
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if (m_pllCount > PLL_LIMIT)
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m_pllCount -= SAMPLES_PER_SYMBOL;
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float32_t offset = m_pllCount / float32_t(m_pllBits);
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float32_t jitter;
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::arm_fir_f32(&m_pllFilter, &offset, &jitter, 1U);
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m_pllCount -= jitter / 2.0F;
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m_pllBits = 1U;
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} else {
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if (m_pllCount > PLL_LIMIT) {
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sample = true;
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m_pllCount -= SAMPLES_PER_SYMBOL;
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m_pllBits++;
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}
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}
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m_pllCount += 1.0F;
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return sample;
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}
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