mirror of
https://github.com/lwvmobile/dsd-fme.git
synced 2026-11-12 16:57:23 -05:00
554 lines
14 KiB
C
554 lines
14 KiB
C
/*-------------------------------------------------------------------------------
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* 3.c
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* Placeholder File 3 -- This file is for temporary code storage
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*
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* Multiple 1/2 Rate Convolutional Decoders for NXDN/M17/YSF
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* Used as a secondary decoder in case convolutional decoder fails (second opinion)
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*-----------------------------------------------------------------------------*/
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#include "dsd.h"
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static const int PARITY[] = {0, 1, 1, 0, 1, 0, 0, 1, 1, 0, 0, 1, 0, 1, 1, 0, 1, 0, 0, 1, 0, 1, 1, 0, 0, 1, 1, 0, 1, 0, 0, 1};
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// trellis_1_2 encode: source is in bits, result in bits
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void trellis_encode(uint8_t result[], const uint8_t source[], int result_len, int reg)
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{
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for (int i=0; i<result_len; i+=2) {
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reg = (reg << 1) | source[i>>1];
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result[i] = PARITY[reg & 0x19];
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result[i+1] = PARITY[reg & 0x17];
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}
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}
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// simplified trellis 2:1 decode; source and result in bits
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// assumes that encoding was done with NTEST trailing zero bits
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// result_len should be set to the actual number of data bits
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// in the original unencoded message (excl. these trailing bits)
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void trellis_decode(uint8_t result[], const uint8_t source[], int result_len)
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{
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int reg = 0;
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int min_d;
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int min_bt;
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static const int NTEST = 4;
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static const int NTESTC = 1 << NTEST;
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uint8_t bt[NTEST];
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uint8_t tt[NTEST*2];
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int dstats[4];
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int sum;
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for (int p=0; p < 4; p++)
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dstats[p] = 0;
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for (int p=0; p < result_len; p++) {
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for (int i=0; i<NTESTC; i++) {
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bt[0] = (i&8)>>3;
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bt[1] = (i&4)>>2;
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bt[2] = (i&2)>>1;
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bt[3] = (i&1);
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trellis_encode(tt, bt, NTEST*2, reg);
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sum=0;
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for (int j=0; j<NTEST*2; j++) {
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sum += tt[j] ^ source[p*2+j];
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}
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if (i == 0 || sum < min_d) {
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min_d = sum;
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min_bt = bt[0];
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}
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}
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result[p] = min_bt;
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reg = (reg << 1) | min_bt;
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dstats[(min_d > 3) ? 3 : min_d] += 1;
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}
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//debug output
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// fprintf (stderr, "\n stats\t%d %d %d %d\n", dstats[0], dstats[1], dstats[2], dstats[3]);
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}
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//Original Copyright/License
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/* -*- c++ -*- */
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/*
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* NXDN Encoder/Decoder (C) Copyright 2019 Max H. Parke KA1RBI
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*
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* This file is part of OP25
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*
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* This 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 3, or (at your option)
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* any later version.
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*
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* This software 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 software; see the file COPYING. If not, write to
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* the Free Software Foundation, Inc., 51 Franklin Street,
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* Boston, MA 02110-1301, USA.
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*/
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//Ripped from libM17
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#define K 5 //constraint length
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#define NUM_STATES (1 << (K - 1)) //number of states
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static uint32_t prevMetrics[NUM_STATES];
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static uint32_t currMetrics[NUM_STATES];
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static uint32_t prevMetricsData[NUM_STATES];
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static uint32_t currMetricsData[NUM_STATES];
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static uint16_t viterbi_history[244];
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/**
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* @brief Decode unpunctured convolutionally encoded data.
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*
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* @param out Destination array where decoded data is written.
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* @param in Input data.
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* @param len Input length in bits.
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* @return Number of bit errors corrected.
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*/
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uint32_t viterbi_decode(uint8_t* out, const uint16_t* in, const uint16_t len)
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{
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if(len > 244*2)
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fprintf(stderr, "Input size exceeds max history\n");
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viterbi_reset();
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size_t pos = 0;
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for(size_t i = 0; i < len; i += 2)
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{
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uint16_t s0 = in[i];
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uint16_t s1 = in[i + 1];
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viterbi_decode_bit(s0, s1, pos);
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pos++;
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}
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uint32_t err = viterbi_chainback(out, pos, len/2);
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//debug
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// fprintf (stderr, "\n vcb: \n");
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// for (size_t i = 0; i < len; i++)
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// fprintf (stderr, "%02X", out[i]);
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return err;
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}
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/**
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* @brief Decode punctured convolutionally encoded data.
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*
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* @param out Destination array where decoded data is written.
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* @param in Input data.
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* @param punct Puncturing matrix.
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* @param in_len Input data length.
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* @param p_len Puncturing matrix length (entries).
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* @return Number of bit errors corrected.
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*/
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uint32_t viterbi_decode_punctured(uint8_t* out, const uint16_t* in, const uint8_t* punct, const uint16_t in_len, const uint16_t p_len)
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{
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if(in_len > 244*2)
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fprintf(stderr, "Input size exceeds max history\n");
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uint16_t umsg[244*2]; //unpunctured message
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uint8_t p=0; //puncturer matrix entry
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uint16_t u=0; //bits count - unpunctured message
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uint16_t i=0; //bits read from the input message
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while(i<in_len)
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{
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if(punct[p])
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{
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umsg[u]=in[i];
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i++;
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}
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else
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{
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umsg[u]=0x7FFF;
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}
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u++;
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p++;
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p%=p_len;
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}
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//debug
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// fprintf (stderr, " p: %d, u: %d; p_len: %d; len: %d;", p, u, p_len, (u-in_len)*0x7FFF);
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return viterbi_decode(out, umsg, u) - (u-in_len)*0x7FFF;
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}
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/**
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* @brief Decode one bit and update trellis.
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*
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* @param s0 Cost of the first symbol.
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* @param s1 Cost of the second symbol.
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* @param pos Bit position in history.
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*/
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void viterbi_decode_bit(uint16_t s0, uint16_t s1, const size_t pos)
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{
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static const uint16_t COST_TABLE_0[] = {0, 0, 0, 0, 0xFFFF, 0xFFFF, 0xFFFF, 0xFFFF};
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static const uint16_t COST_TABLE_1[] = {0, 0xFFFF, 0xFFFF, 0, 0, 0xFFFF, 0xFFFF, 0};
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for(uint8_t i = 0; i < NUM_STATES/2; i++)
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{
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uint32_t metric = q_abs_diff(COST_TABLE_0[i], s0)
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+ q_abs_diff(COST_TABLE_1[i], s1);
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uint32_t m0 = prevMetrics[i] + metric;
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uint32_t m1 = prevMetrics[i + NUM_STATES/2] + (0x1FFFE - metric);
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uint32_t m2 = prevMetrics[i] + (0x1FFFE - metric);
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uint32_t m3 = prevMetrics[i + NUM_STATES/2] + metric;
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uint8_t i0 = 2 * i;
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uint8_t i1 = i0 + 1;
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if(m0 >= m1)
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{
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viterbi_history[pos]|=(1<<i0);
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currMetrics[i0] = m1;
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}
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else
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{
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viterbi_history[pos]&=~(1<<i0);
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currMetrics[i0] = m0;
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}
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if(m2 >= m3)
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{
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viterbi_history[pos]|=(1<<i1);
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currMetrics[i1] = m3;
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}
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else
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{
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viterbi_history[pos]&=~(1<<i1);
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currMetrics[i1] = m2;
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}
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}
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//swap
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uint32_t tmp[NUM_STATES];
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for(uint8_t i=0; i<NUM_STATES; i++)
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{
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tmp[i]=currMetrics[i];
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}
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for(uint8_t i=0; i<NUM_STATES; i++)
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{
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currMetrics[i]=prevMetrics[i];
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prevMetrics[i]=tmp[i];
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}
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}
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/**
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* @brief History chainback to obtain final byte array.
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*
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* @param out Destination byte array for decoded data.
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* @param pos Starting position for the chainback.
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* @param len Length of the output in bits.
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* @return Minimum Viterbi cost at the end of the decode sequence.
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*/
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uint32_t viterbi_chainback(uint8_t* out, size_t pos, uint16_t len)
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{
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uint8_t state = 0;
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size_t bitPos = len+4;
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memset(out, 0, (len-1)/8+1);
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while(pos > 0)
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{
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bitPos--;
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pos--;
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uint16_t bit = viterbi_history[pos]&((1<<(state>>4)));
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state >>= 1;
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if(bit)
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{
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state |= 0x80;
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out[bitPos/8]|=1<<(7-(bitPos%8));
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}
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}
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uint32_t cost = prevMetrics[0];
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for(size_t i = 0; i < NUM_STATES; i++)
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{
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uint32_t m = prevMetrics[i];
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if(m < cost) cost = m;
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}
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//debug
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// fprintf (stderr, "\n cb: \n");
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// for (size_t i = 0; i < len; i++)
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// fprintf (stderr, "%02X", out[i]);
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return cost;
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}
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/**
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* @brief Reset the decoder state. No args.
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*
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*/
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void viterbi_reset(void)
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{
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memset((uint8_t*)viterbi_history, 0, 2*244);
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memset((uint8_t*)currMetrics, 0, 4*NUM_STATES);
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memset((uint8_t*)prevMetrics, 0, 4*NUM_STATES);
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memset((uint8_t*)currMetricsData, 0, 4*NUM_STATES);
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memset((uint8_t*)prevMetricsData, 0, 4*NUM_STATES);
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}
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uint16_t q_abs_diff(const uint16_t v1, const uint16_t v2)
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{
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if(v2 > v1) return v2 - v1;
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return v1 - v2;
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}
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//original sources
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//--------------------------------------------------------------------
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// M17 C library - decode/viterbi.c
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//
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// This file contains:
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// - the Viterbi decoder
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//
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// Wojciech Kaczmarski, SP5WWP
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// M17 Project, 29 December 2023
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//--------------------------------------------------------------------
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//--------------------------------------------------------------------
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// M17 C library - math/math.c
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//
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// This file contains:
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// - absolute difference value
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// - Euclidean norm (L2) calculation for n-dimensional vectors (float)
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// - soft-valued arrays to integer conversion (and vice-versa)
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// - fixed-valued multiplication and division
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//
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// Wojciech Kaczmarski, SP5WWP
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// M17 Project, 29 December 2023
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//--------------------------------------------------------------------
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//audio filter stuff sourced from: https://github.com/NedSimao/FilteringLibrary
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//no license / information provided in source code
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#define PI 3.141592653
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//do we need this for test input sthit?
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// #define PLOT_POINTS 100
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// #define SAMPLE_TIME_S 0.001
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// #define CUTOFFFREQ_HZ 50
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// #define sine_freq 40
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// #define sine_freq_1 500
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// #define HFC 50
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// #define LFC 100
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void LPFilter_Init(LPFilter *filter, float cutoffFreqHz, float sampleTimeS){
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float RC=0.0;
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RC=1.0/(2*PI*cutoffFreqHz);
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filter->coef[0]=sampleTimeS/(sampleTimeS+RC);
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filter->coef[1]=RC/(sampleTimeS+RC);
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filter->v_out[0]=0.0;
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filter->v_out[1]=0.0;
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}
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float LPFilter_Update(LPFilter *filter, float v_in){
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filter->v_out[1]=filter->v_out[0];
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filter->v_out[0]=(filter->coef[0]*v_in) + (filter->coef[1]*filter->v_out[1]);
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return (filter->v_out[0]);
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}
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/********************************************************************************************************
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* HIGH PASS FILTER
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********************************************************************************************************/
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void HPFilter_Init(HPFilter *filter, float cutoffFreqHz, float sampleTimeS){
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float RC=0.0;
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RC=1.0/(2*PI*cutoffFreqHz);
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filter->coef=RC/(sampleTimeS+RC);
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filter->v_in[0]=0.0;
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filter->v_in[1]=0.0;
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filter->v_out[0]=0.0;
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filter->v_out[1]=0.0;
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}
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float HPFilter_Update(HPFilter *filter, float v_in){
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filter->v_in[1]=filter->v_in[0];
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filter->v_in[0]=v_in;
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filter->v_out[1]=filter->v_out[0];
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filter->v_out[0]=filter->coef * (filter->v_in[0] - filter->v_in[1]+filter->v_out[1]);
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return (filter->v_out[0]);
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}
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/********************************************************************************************************
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* BAND PASS FILTER
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********************************************************************************************************/
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void PBFilter_Init(PBFilter *filter, float HPF_cutoffFreqHz, float LPF_cutoffFreqHz, float sampleTimeS){
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LPFilter_Init(&filter->lpf, LPF_cutoffFreqHz, sampleTimeS);
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HPFilter_Init(&filter->hpf, HPF_cutoffFreqHz, sampleTimeS);
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filter->out_in=0.0;
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}
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float PBFilter_Update(PBFilter *filter, float v_in){
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filter->out_in=HPFilter_Update(&filter->hpf, v_in);
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filter->out_in=LPFilter_Update(&filter->lpf, filter->out_in);
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return (filter->out_in);
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}
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/********************************************************************************************************
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* NOTCH FILTER
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********************************************************************************************************/
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void NOTCHFilter_Init(NOTCHFilter *filter, float centerFreqHz, float notchWidthHz, float sampleTimeS){
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//filter frequency to angular (rad/s)
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float w0_rps=2.0 * PI *centerFreqHz;
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float ww_rps=2.0 * PI *notchWidthHz;
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//pre warp center frequency
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float w0_pw_rps=(2.0/sampleTimeS) * tanf(0.5 * w0_rps * sampleTimeS);
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//computing filter coefficients
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filter->alpha=4.0 + w0_rps*w0_pw_rps*sampleTimeS*sampleTimeS;
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filter->beta=2.0*ww_rps*sampleTimeS;
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//clearing input and output buffers
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for (uint8_t n=0; n<3; n++){
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filter->vin[n]=0;
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filter->vout[n]=0;
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}
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}
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float NOTCHFilter_Update(NOTCHFilter *filter, float vin){
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//shifting samples
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filter->vin[2]=filter->vin[1];
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filter->vin[1]=filter->vin[0];
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filter->vout[2]=filter->vout[1];
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filter->vout[1]=filter->vout[0];
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filter->vin[0]=vin;
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//compute new output
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filter->vout[0]=(filter->alpha*filter->vin[0] + 2.0 *(filter->alpha -8.0)*filter->vin[1] + filter->alpha*filter->vin[2]
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-(2.0f*(filter->alpha-8.0)*filter->vout[1]+(filter->alpha-filter->beta)*filter->vout[2]))/(filter->alpha+filter->beta);
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return (filter->vout[0]);
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}
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void init_audio_filters (dsd_state * state)
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{
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//still not sure if this is even correct or not, but 48k sounds good now
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LPFilter_Init(&state->RCFilter, 960, (float)1/(float)48000);
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HPFilter_Init(&state->HRCFilter, 960, (float)1/(float)48000);
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//left and right variants for stereo output testing on digital voice samples
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LPFilter_Init(&state->RCFilterL, 960, (float)1/(float)48000);
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HPFilter_Init(&state->HRCFilterL, 960, (float)1/(float)48000);
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LPFilter_Init(&state->RCFilterR, 960, (float)1/(float)48000);
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HPFilter_Init(&state->HRCFilterR, 960, (float)1/(float)48000);
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//PBFilter_Init(PBFilter *filter, float HPF_cutoffFreqHz, float LPF_cutoffFreqHz, float sampleTimeS);
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// void NOTCHFilter_Init(NOTCHFilter *filter, float centerFreqHz, float notchWidthHz, float sampleTimeS);
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//unclear what we should set up for these, output is super quiet after using them
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PBFilter_Init(&state->PBF, 48000, 12000, (float)1/(float)48000);
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NOTCHFilter_Init(&state->NF, 48000, 12000, (float)1/(float)48000);
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}
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//FUNCTIONS for handing use of above filters
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//lpf
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void lpf(dsd_state * state, short * input, int len)
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{
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int i;
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for (i = 0; i < len; i++)
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{
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// fprintf (stderr, "\n in: %05d", input[i]);
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input[i] = LPFilter_Update(&state->RCFilter, input[i]);
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// fprintf (stderr, "\n out: %05d", input[i]);
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}
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}
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//hpf
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void hpf(dsd_state * state, short * input, int len)
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{
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int i;
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for (i = 0; i < len; i++)
|
|
{
|
|
// fprintf (stderr, "\n in: %05d", input[i]);
|
|
input[i] = HPFilter_Update(&state->HRCFilter, input[i]);
|
|
// fprintf (stderr, "\n out: %05d", input[i]);
|
|
}
|
|
}
|
|
|
|
//nf
|
|
void nf(dsd_state * state, short * input, int len)
|
|
{
|
|
int i;
|
|
for (i = 0; i < len; i++)
|
|
{
|
|
// fprintf (stderr, "\n in: %05d", input[i]);
|
|
input[i] = NOTCHFilter_Update(&state->NF, input[i]);
|
|
// fprintf (stderr, "\n out: %05d", input[i]);
|
|
}
|
|
}
|
|
|
|
//pbf
|
|
void pbf(dsd_state * state, short * input, int len)
|
|
{
|
|
int i;
|
|
for (i = 0; i < len; i++)
|
|
{
|
|
// fprintf (stderr, "\n in: %05d", input[i]);
|
|
input[i] = PBFilter_Update(&state->PBF, input[i]);
|
|
// fprintf (stderr, "\n out: %05d", input[i]);
|
|
}
|
|
}
|
|
|
|
//Generic RMS function derived from RTL_FM (RTL_SDR) RMS code (doesnt' really work correctly outside of RTL)
|
|
long int raw_rms(int16_t *samples, int len, int step) //use samplespersymbol as len
|
|
{
|
|
|
|
int i;
|
|
long int rms;
|
|
long p, t, s;
|
|
double dc, err;
|
|
|
|
p = t = 0L;
|
|
for (i=0; i<len; i+=step) {
|
|
s = (long)samples[i];
|
|
t += s;
|
|
p += s * s;
|
|
}
|
|
/* correct for dc offset in squares */
|
|
dc = (double)(t*step) / (double)len;
|
|
err = t * 2 * dc - dc * dc * len;
|
|
|
|
rms = (long int)sqrt((p-err) / len);
|
|
if (rms < 0) rms = 150;
|
|
return rms;
|
|
} |