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direwolf/src/demod_9600.c

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//
// This file is part of Dire Wolf, an amateur radio packet TNC.
//
// Copyright (C) 2011, 2012, 2013, 2015, 2019, 2021 John Langner, WB2OSZ
//
// This program is free software: you can redistribute it and/or modify
// it under the terms of the GNU General Public License as published by
// the Free Software Foundation, either version 2 of the License, or
// (at your option) any later version.
//
// This program is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
// GNU General Public License for more details.
//
// You should have received a copy of the GNU General Public License
// along with this program. If not, see <http://www.gnu.org/licenses/>.
//
//#define DEBUG4 1 /* capture 9600 output to log files */
/*------------------------------------------------------------------
*
* Module: demod_9600.c
*
* Purpose: Demodulator for baseband signal.
* This is used for AX.25 (with scrambling) and IL2P without.
*
* Input: Audio samples from either a file or the "sound card."
*
* Outputs: Calls hdlc_rec_bit() for each bit demodulated.
*
*---------------------------------------------------------------*/
#include "direwolf.h"
#include <stdlib.h>
#include <stdio.h>
#include <math.h>
#include <unistd.h>
#include <sys/stat.h>
#include <string.h>
#include <assert.h>
#include <ctype.h>
// Fine tuning for different demodulator types.
// Don't remove this section. It is here for a reason.
#define DCD_THRESH_ON 32 // Hysteresis: Can miss 0 out of 32 for detecting lock.
// This is best for actual on-the-air signals.
// Still too many brief false matches.
#define DCD_THRESH_OFF 8 // Might want a little more fine tuning.
#define DCD_GOOD_WIDTH 1024 // No more than 1024!!!
#include "fsk_demod_state.h" // Values above override defaults.
#include "tune.h"
#include "hdlc_rec.h"
#include "demod_9600.h"
#include "textcolor.h"
#include "dsp.h"
static float slice_point[MAX_SUBCHANS];
/* Add sample to buffer and shift the rest down. */
__attribute__((hot)) __attribute__((always_inline))
static inline void push_sample (float val, float *buff, int size)
{
memmove(buff+1,buff,(size-1)*sizeof(float));
buff[0] = val;
}
/* FIR filter kernel. */
__attribute__((hot)) __attribute__((always_inline))
static inline float convolve (const float *__restrict__ data, const float *__restrict__ filter, int filter_size)
{
float sum = 0.0f;
int j;
//#pragma GCC ivdep // ignored until gcc 4.9
for (j=0; j<filter_size; j++) {
sum += filter[j] * data[j];
}
return (sum);
}
/* Automatic gain control. */
/* Result should settle down to 1 unit peak to peak. i.e. -0.5 to +0.5 */
__attribute__((hot)) __attribute__((always_inline))
static inline float agc (float in, float fast_attack, float slow_decay, float *ppeak, float *pvalley)
{
if (in >= *ppeak) {
*ppeak = in * fast_attack + *ppeak * (1.0f - fast_attack);
}
else {
*ppeak = in * slow_decay + *ppeak * (1.0f - slow_decay);
}
if (in <= *pvalley) {
*pvalley = in * fast_attack + *pvalley * (1.0f - fast_attack);
}
else {
*pvalley = in * slow_decay + *pvalley * (1.0f - slow_decay);
}
if (*ppeak > *pvalley) {
return ((in - 0.5f * (*ppeak + *pvalley)) / (*ppeak - *pvalley));
}
return (0.0);
}
/*------------------------------------------------------------------
*
* Name: demod_9600_init
*
* Purpose: Initialize the 9600 (or higher) baud demodulator.
*
* Inputs: modem_type - Determines whether scrambling is used.
*
* samples_per_sec - Number of samples per second for audio.
*
* upsample - Factor to upsample the incoming stream.
* After a lot of experimentation, I discovered that
* it works better if the data is upsampled.
* This reduces the jitter for PLL synchronization.
*
* baud - Data rate in bits per second.
*
* D - Address of demodulator state.
*
* Returns: None
*
*----------------------------------------------------------------*/
void demod_9600_init (enum modem_t modem_type, int original_sample_rate, int upsample, int baud, struct demodulator_state_s *D)
{
float fc;
int j;
if (upsample < 1) upsample = 1;
if (upsample > 4) upsample = 4;
memset (D, 0, sizeof(struct demodulator_state_s));
D->modem_type = modem_type;
D->num_slicers = 1;
// Multiple profiles in future?
// switch (profile) {
// case 'J': // upsample x2 with filtering.
// case 'K': // upsample x3 with filtering.
// case 'L': // upsample x4 with filtering.
D->lp_filter_len_bits = 1.0; // -U4 = 61 4.59 samples/symbol
// Works best with odd number in some tests. Even is better in others.
//D->lp_filter_size = ((int) (0.5f * ( D->lp_filter_len_bits * (float)original_sample_rate / (float)baud ))) * 2 + 1;
// Just round to nearest integer.
D->lp_filter_size = (int) (( D->lp_filter_len_bits * (float)original_sample_rate / baud) + 0.5f);
D->lp_window = BP_WINDOW_COSINE;
D->lpf_baud = 1.00;
D->agc_fast_attack = 0.080;
D->agc_slow_decay = 0.00012;
D->pll_locked_inertia = 0.89;
D->pll_searching_inertia = 0.67;
// break;
// }
#if 0
text_color_set(DW_COLOR_DEBUG);
dw_printf ("---------- %s (%d, %d) -----------\n", __func__, samples_per_sec, baud);
dw_printf ("filter_len_bits = %.2f\n", D->lp_filter_len_bits);
dw_printf ("lp_filter_size = %d\n", D->lp_filter_size);
dw_printf ("lp_window = %d\n", D->lp_window);
dw_printf ("lpf_baud = %.2f\n", D->lpf_baud);
dw_printf ("samples per bit = %.1f\n", (double)samples_per_sec / baud);
#endif
// PLL needs to use the upsampled rate.
D->pll_step_per_sample =
(int) round(TICKS_PER_PLL_CYCLE * (double) baud / (double)(original_sample_rate * upsample));
#ifdef TUNE_LP_WINDOW
D->lp_window = TUNE_LP_WINDOW;
#endif
#if TUNE_LP_FILTER_SIZE
D->lp_filter_size = TUNE_LP_FILTER_SIZE;
#endif
#ifdef TUNE_LPF_BAUD
D->lpf_baud = TUNE_LPF_BAUD;
#endif
#ifdef TUNE_AGC_FAST
D->agc_fast_attack = TUNE_AGC_FAST;
#endif
#ifdef TUNE_AGC_SLOW
D->agc_slow_decay = TUNE_AGC_SLOW;
#endif
#if defined(TUNE_PLL_LOCKED)
D->pll_locked_inertia = TUNE_PLL_LOCKED;
#endif
#if defined(TUNE_PLL_SEARCHING)
D->pll_searching_inertia = TUNE_PLL_SEARCHING;
#endif
// Initial filter (before scattering) is based on upsampled rate.
fc = (float)baud * D->lpf_baud / (float)(original_sample_rate * upsample);
//dw_printf ("demod_9600_init: call gen_lowpass(fc=%.2f, , size=%d, )\n", fc, D->lp_filter_size);
gen_lowpass (fc, D->u.bb.lp_filter, D->lp_filter_size * upsample, D->lp_window);
// New in 1.7 -
// Use a polyphase filter to reduce the CPU load.
// Originally I used zero stuffing to upsample.
// Here is the general idea.
//
// Suppose the input samples are 1 2 3 4 5 6 7 8 9 ...
// Filter coefficients are a b c d e f g h i ...
//
// With original sampling rate, the filtering would involve multiplying and adding:
//
// 1a 2b 3c 4d 5e 6f ...
//
// When upsampling by 3, each of these would need to be evaluated
// for each audio sample:
//
// 1a 0b 0c 2d 0e 0f 3g 0h 0i ...
// 0a 1b 0c 0d 2e 0f 0g 3h 0i ...
// 0a 0b 1c 0d 0e 2f 0g 0h 3i ...
//
// 2/3 of the multiplies are always by a stuffed zero.
// We can do this more efficiently by removing them.
//
// 1a 2d 3g ...
// 1b 2e 3h ...
// 1c 2f 3i ...
//
// We scatter the original filter across multiple shorter filters.
// Each input sample cycles around them to produce the upsampled rate.
//
// a d g ...
// b e h ...
// c f i ...
//
// There are countless sources of information DSP but this one is unique
// in that it is a college course that mentions APRS.
// https://www2.eecs.berkeley.edu/Courses/EE123
//
// Was the effort worthwhile? Times on an RPi 3.
//
// command: atest -B9600 ~/walkabout9600[abc]-compressed*.wav
//
// These are 3 recordings of a portable system being carried out of
// range and back in again. It is a real world test for weak signals.
//
// options num decoded seconds x realtime
// 1.6 1.7 1.6 1.7 1.6 1.7
// --- --- --- --- --- ---
// -P- 171 172 23.928 17.967 14.9 19.9
// -P+ 180 180 54.688 48.772 6.5 7.3
// -P- -F1 177 178 32.686 26.517 10.9 13.5
//
// So, it turns out that -P+ doesn't have a dramatic improvement, only
// around 4%, for drastically increased CPU requirements.
// Maybe we should turn that off by default, especially for ARM.
//
int k = 0;
for (int i = 0; i < D->lp_filter_size; i++) {
D->u.bb.lp_polyphase_1[i] = D->u.bb.lp_filter[k++];
if (upsample >= 2) {
D->u.bb.lp_polyphase_2[i] = D->u.bb.lp_filter[k++];
if (upsample >= 3) {
D->u.bb.lp_polyphase_3[i] = D->u.bb.lp_filter[k++];
if (upsample >= 4) {
D->u.bb.lp_polyphase_4[i] = D->u.bb.lp_filter[k++];
}
}
}
}
/* Version 1.2: Experiment with different slicing levels. */
// Really didn't help that much because we should have a symmetrical signal.
for (j = 0; j < MAX_SUBCHANS; j++) {
slice_point[j] = 0.02f * (j - 0.5f * (MAX_SUBCHANS-1));
//dw_printf ("slice_point[%d] = %+5.2f\n", j, slice_point[j]);
}
} /* end fsk_demod_init */
/*-------------------------------------------------------------------
*
* Name: demod_9600_process_sample
*
* Purpose: (1) Filter & slice the signal.
* (2) Descramble it.
* (2) Recover clock and data.
*
* Inputs: chan - Audio channel. 0 for left, 1 for right.
*
* sam - One sample of audio.
* Should be in range of -32768 .. 32767.
*
* Returns: None
*
* Descripion: "9600 baud" packet is FSK for an FM voice transceiver.
* By the time it gets here, it's really a baseband signal.
* At one extreme, we could have a 4800 Hz square wave.
* A the other extreme, we could go a considerable number
* of bit times without any transitions.
*
* The trick is to extract the digital data which has
* been distorted by going thru voice transceivers not
* intended to pass this sort of "audio" signal.
*
* For G3RUH mode, data is "scrambled" to reduce the amount of DC bias.
* The data stream must be unscrambled at the receiving end.
*
* We also have a digital phase locked loop (PLL)
* to recover the clock and pick out data bits at
* the proper rate.
*
* For each recovered data bit, we call:
*
* hdlc_rec (channel, demodulated_bit);
*
* to decode HDLC frames from the stream of bits.
*
* Future: This could be generalized by passing in the name
* of the function to be called for each bit recovered
* from the demodulator. For now, it's simply hard-coded.
*
* After experimentation, I found that this works better if
* the original signal is upsampled by 2x or even 4x.
*
* References: 9600 Baud Packet Radio Modem Design
* http://www.amsat.org/amsat/articles/g3ruh/109.html
*
* The KD2BD 9600 Baud Modem
* http://www.amsat.org/amsat/articles/kd2bd/9k6modem/
*
* 9600 Baud Packet Handbook
* ftp://ftp.tapr.org/general/9600baud/96man2x0.txt
*
*
*--------------------------------------------------------------------*/
inline static void nudge_pll (int chan, int subchan, int slice, float demod_out, struct demodulator_state_s *D);
static void process_filtered_sample (int chan, float fsam, struct demodulator_state_s *D);
__attribute__((hot))
void demod_9600_process_sample (int chan, int sam, int upsample, struct demodulator_state_s *D)
{
float fsam;
#if DEBUG4
static FILE *demod_log_fp = NULL;
static int log_file_seq = 0; /* Part of log file name */
#endif
int subchan = 0;
assert (chan >= 0 && chan < MAX_CHANS);
assert (subchan >= 0 && subchan < MAX_SUBCHANS);
/* Scale to nice number for convenience. */
/* Consistent with the AFSK demodulator, we'd like to use */
/* only half of the dynamic range to have some headroom. */
/* i.e. input range +-16k becomes +-1 here and is */
/* displayed in the heard line as audio level 100. */
fsam = (float)sam / 16384.0f;
// Low pass filter
push_sample (fsam, D->u.bb.audio_in, D->lp_filter_size);
fsam = convolve (D->u.bb.audio_in, D->u.bb.lp_polyphase_1, D->lp_filter_size);
process_filtered_sample (chan, fsam, D);
if (upsample >= 2) {
fsam = convolve (D->u.bb.audio_in, D->u.bb.lp_polyphase_2, D->lp_filter_size);
process_filtered_sample (chan, fsam, D);
if (upsample >= 3) {
fsam = convolve (D->u.bb.audio_in, D->u.bb.lp_polyphase_3, D->lp_filter_size);
process_filtered_sample (chan, fsam, D);
if (upsample >= 4) {
fsam = convolve (D->u.bb.audio_in, D->u.bb.lp_polyphase_4, D->lp_filter_size);
process_filtered_sample (chan, fsam, D);
}
}
}
}
__attribute__((hot))
static void process_filtered_sample (int chan, float fsam, struct demodulator_state_s *D)
{
int subchan = 0;
/*
* Version 1.2: Capture the post-filtering amplitude for display.
* This is similar to the AGC without the normalization step.
* We want decay to be substantially slower to get a longer
* range idea of the received audio.
* For AFSK, we keep mark and space amplitudes.
* Here we keep + and - peaks because there could be a DC bias.
*/
// TODO: probably no need for this. Just use D->m_peak, D->m_valley
if (fsam >= D->alevel_mark_peak) {
D->alevel_mark_peak = fsam * D->quick_attack + D->alevel_mark_peak * (1.0f - D->quick_attack);
}
else {
D->alevel_mark_peak = fsam * D->sluggish_decay + D->alevel_mark_peak * (1.0f - D->sluggish_decay);
}
if (fsam <= D->alevel_space_peak) {
D->alevel_space_peak = fsam * D->quick_attack + D->alevel_space_peak * (1.0f - D->quick_attack);
}
else {
D->alevel_space_peak = fsam * D->sluggish_decay + D->alevel_space_peak * (1.0f - D->sluggish_decay);
}
/*
* The input level can vary greatly.
* More importantly, there could be a DC bias which we need to remove.
*
* Normalize the signal with automatic gain control (AGC).
* This works by looking at the minimum and maximum signal peaks
* and scaling the results to be roughly in the -1.0 to +1.0 range.
*/
float demod_out;
int demod_data; /* Still scrambled. */
demod_out = agc (fsam, D->agc_fast_attack, D->agc_slow_decay, &(D->m_peak), &(D->m_valley));
// TODO: There is potential for multiple decoders with one filter.
//dw_printf ("peak=%.2f valley=%.2f fsam=%.2f norm=%.2f\n", D->m_peak, D->m_valley, fsam, norm);
if (D->num_slicers <= 1) {
/* Normal case of one demodulator to one HDLC decoder. */
/* Demodulator output is difference between response from two filters. */
/* AGC should generally keep this around -1 to +1 range. */
demod_data = demod_out > 0;
nudge_pll (chan, subchan, 0, demod_out, D);
}
else {
int slice;
/* Multiple slicers each feeding its own HDLC decoder. */
for (slice=0; slice<D->num_slicers; slice++) {
demod_data = demod_out - slice_point[slice] > 0;
nudge_pll (chan, subchan, slice, demod_out - slice_point[slice], D);
}
}
// demod_data is used only for debug out.
// suppress compiler warning about it not being used.
(void) demod_data;
#if DEBUG4
if (chan == 0) {
if (1) {
//if (D->slicer[slice].data_detect) {
char fname[30];
int slice = 0;
if (demod_log_fp == NULL) {
log_file_seq++;
snprintf (fname, sizeof(fname), "demod/%04d.csv", log_file_seq);
//if (log_file_seq == 1) mkdir ("demod", 0777);
if (log_file_seq == 1) mkdir ("demod");
demod_log_fp = fopen (fname, "w");
text_color_set(DW_COLOR_DEBUG);
dw_printf ("Starting demodulator log file %s\n", fname);
fprintf (demod_log_fp, "Audio, Filtered, Max, Min, Normalized, Sliced, Clock\n");
}
fprintf (demod_log_fp, "%.3f, %.3f, %.3f, %.3f, %.3f, %d, %.2f\n",
fsam + 6,
fsam + 4,
D->m_peak + 4,
D->m_valley + 4,
demod_out + 2,
demod_data + 2,
(D->slicer[slice].data_clock_pll & 0x80000000) ? .5 : .0);
fflush (demod_log_fp);
}
else {
if (demod_log_fp != NULL) {
fclose (demod_log_fp);
demod_log_fp = NULL;
}
}
}
#endif
} /* end demod_9600_process_sample */
/*-------------------------------------------------------------------
*
* Name: nudge_pll
*
* Purpose: Update the PLL state for each audio sample.
*
* (2) Descramble it.
* (2) Recover clock and data.
*
* Inputs: chan - Audio channel. 0 for left, 1 for right.
*
* subchan - Which demodulator. We could have several running in parallel.
*
* slice - Determines which Slicing level & HDLC decoder to use.
*
* demod_out_f - Demodulator output, possibly shifted by slicing level
* It will be compared with 0.0 to bit binary value out.
*
* D - Demodulator state for this channel / subchannel.
*
* Returns: None
*
* Description: A PLL is used to sample near the centers of the data bits.
*
* D->data_clock_pll is a SIGNED 32 bit variable.
* When it overflows from a large positive value to a negative value, we
* sample a data bit from the demodulated signal.
*
* Ideally, the the demodulated signal transitions should be near
* zero we we sample mid way between the transitions.
*
* Nudge the PLL by removing some small fraction from the value of
* data_clock_pll, pushing it closer to zero.
*
* This adjustment will never change the sign so it won't cause
* any erratic data bit sampling.
*
* If we adjust it too quickly, the clock will have too much jitter.
* If we adjust it too slowly, it will take too long to lock on to a new signal.
*
* I don't think the optimal value will depend on the audio sample rate
* because this happens for each transition from the demodulator.
*
* Version 1.4: Previously, we would always pull the PLL phase toward 0 after
* after a zero crossing was detetected. This adds extra jitter,
* especially when the ratio of audio sample rate to baud is low.
* Now, we interpolate between the two samples to get an estimate
* on when the zero crossing happened. The PLL is pulled toward
* this point.
*
* Results??? TBD
*
* Version 1.6: New experiment where filter size to extract clock is not the same
* as filter to extract the data bit value.
*
*--------------------------------------------------------------------*/
__attribute__((hot))
inline static void nudge_pll (int chan, int subchan, int slice, float demod_out_f, struct demodulator_state_s *D)
{
D->slicer[slice].prev_d_c_pll = D->slicer[slice].data_clock_pll;
// Perform the add as unsigned to avoid signed overflow error.
D->slicer[slice].data_clock_pll = (signed)((unsigned)(D->slicer[slice].data_clock_pll) + (unsigned)(D->pll_step_per_sample));
if ( D->slicer[slice].prev_d_c_pll > 1000000000 && D->slicer[slice].data_clock_pll < -1000000000) {
/* Overflow. Was large positive, wrapped around, now large negative. */
hdlc_rec_bit (chan, subchan, slice, demod_out_f > 0, D->modem_type == MODEM_SCRAMBLE, D->slicer[slice].lfsr);
pll_dcd_each_symbol2 (D, chan, subchan, slice);
}
/*
* Zero crossing?
*/
if ((D->slicer[slice].prev_demod_out_f < 0 && demod_out_f > 0) ||
(D->slicer[slice].prev_demod_out_f > 0 && demod_out_f < 0)) {
// Note: Test for this demodulator, not overall for channel.
pll_dcd_signal_transition2 (D, slice, D->slicer[slice].data_clock_pll);
float target = D->pll_step_per_sample * demod_out_f / (demod_out_f - D->slicer[slice].prev_demod_out_f);
if (D->slicer[slice].data_detect) {
D->slicer[slice].data_clock_pll = (int)(D->slicer[slice].data_clock_pll * D->pll_locked_inertia + target * (1.0f - D->pll_locked_inertia) );
}
else {
D->slicer[slice].data_clock_pll = (int)(D->slicer[slice].data_clock_pll * D->pll_searching_inertia + target * (1.0f - D->pll_searching_inertia) );
}
}
#if DEBUG5
//if (chan == 0) {
if (D->slicer[slice].data_detect) {
char fname[30];
if (demod_log_fp == NULL) {
seq++;
snprintf (fname, sizeof(fname), "demod96/%04d.csv", seq);
if (seq == 1) mkdir ("demod96"
#ifndef __WIN32__
, 0777
#endif
);
demod_log_fp = fopen (fname, "w");
text_color_set(DW_COLOR_DEBUG);
dw_printf ("Starting 9600 decoder log file %s\n", fname);
fprintf (demod_log_fp, "Audio, Peak, Valley, Demod, SData, Descram, Clock\n");
}
fprintf (demod_log_fp, "%.3f, %.3f, %.3f, %.3f, %.2f, %.2f, %.2f\n",
0.5f * fsam + 3.5,
0.5f * D->m_peak + 3.5,
0.5f * D->m_valley + 3.5,
0.5f * demod_out + 2.0,
demod_data ? 1.35 : 1.0,
descram ? .9 : .55,
(D->data_clock_pll & 0x80000000) ? .1 : .45);
}
else {
if (demod_log_fp != NULL) {
fclose (demod_log_fp);
demod_log_fp = NULL;
}
}
//}
#endif
/*
* Remember demodulator output (pre-descrambling) so we can compare next time
* for the DPLL sync.
*/
D->slicer[slice].prev_demod_out_f = demod_out_f;
} /* end nudge_pll */
/* end demod_9600.c */
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