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direwolf/src/demod_afsk.c

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//
// This file is part of Dire Wolf, an amateur radio packet TNC.
//
// Copyright (C) 2011, 2012, 2013, 2014, 2015 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 DEBUG1 1 /* display debugging info */
// #define DEBUG3 1 /* print carrier detect changes. */
// #define DEBUG4 1 /* capture AFSK demodulator output to log files */
// #define DEBUG5 1 /* capture 9600 output to log files */
/*------------------------------------------------------------------
*
* Module: demod_afsk.c
*
* Purpose: Demodulator for Audio Frequency Shift Keying (AFSK).
*
* 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>
#include "audio.h"
#include "tune.h"
#include "fsk_demod_state.h"
#include "fsk_gen_filter.h"
#include "hdlc_rec.h"
#include "textcolor.h"
#include "demod_afsk.h"
#include "dsp.h"
#define MIN(a,b) ((a)<(b)?(a):(b))
#define MAX(a,b) ((a)>(b)?(a):(b))
/* Quick approximation to sqrt(x*x+y*y) */
/* No benefit for regular PC. */
/* Should help with microcomputer platform. */
#if 0 // not using anymore
__attribute__((hot)) __attribute__((always_inline))
static inline float z (float x, float y)
{
x = fabsf(x);
y = fabsf(y);
if (x > y) {
return (x * .941246f + y * .41f);
}
else {
return (y * .941246f + x * .41f);
}
}
#endif
/* 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 (0.0f);
}
/*
* for multi-slicer experiment.
*/
#define MIN_G 0.5f
#define MAX_G 4.0f
/* TODO: static */ float space_gain[MAX_SUBCHANS];
/*------------------------------------------------------------------
*
* Name: demod_afsk_init
*
* Purpose: Initialization for an AFSK demodulator.
* Select appropriate parameters and set up filters.
*
* Inputs: samples_per_sec
* baud
* mark_freq
* space_freq
*
* D - Pointer to demodulator state for given channel.
*
* Outputs: D->ms_filter_size
* D->m_sin_table[]
* D->m_cos_table[]
* D->s_sin_table[]
* D->s_cos_table[]
*
* Returns: None.
*
* Bugs: This doesn't do much error checking so don't give it
* anything crazy.
*
*----------------------------------------------------------------*/
void demod_afsk_init (int samples_per_sec, int baud, int mark_freq,
int space_freq, char profile, struct demodulator_state_s *D)
{
int j;
memset (D, 0, sizeof(struct demodulator_state_s));
D->num_slicers = 1;
#if DEBUG1
dw_printf ("demod_afsk_init (rate=%d, baud=%d, mark=%d, space=%d, profile=%c\n",
samples_per_sec, baud, mark_freq, space_freq, profile);
#endif
#ifdef TUNE_PROFILE
profile = TUNE_PROFILE;
#endif
D->profile = profile; // so we know whether to take fast path later.
switch (profile) {
case 'D':
/* Prefilter, Cosine window, FIR lowpass. Tweeked for 300 baud. */
D->use_prefilter = 1; /* first, a bandpass filter. */
D->prefilter_baud = 0.87;
D->pre_filter_len_bits = 1.857;
D->pre_window = BP_WINDOW_COSINE;
D->ms_filter_len_bits = 1.857; /* 91 @ 44100/3, 300 */
D->ms_window = BP_WINDOW_COSINE;
//D->bp_window = BP_WINDOW_COSINE;
D->lpf_use_fir = 1;
D->lpf_baud = 1.10;
D->lp_filter_len_bits = D->ms_filter_len_bits;
D->lp_window = BP_WINDOW_TRUNCATED;
D->agc_fast_attack = 0.495;
D->agc_slow_decay = 0.00022;
D->hysteresis = 0.027;
D->pll_locked_inertia = 0.620;
D->pll_searching_inertia = 0.350;
break;
case 'F': // removed obsolete. treat as E for now.
case 'E':
/* 1200 baud - Started out similar to C but add prefilter. */
/* Version 1.2 */
/* Enhancements: */
/* + Add prefilter. Previously used for 300 baud D, but not 1200. */
/* + Prefilter length now independent of M/S filters. */
/* + Lowpass filter length now independent of M/S filters. */
/* + Allow mixed window types. */
//D->bp_window = BP_WINDOW_COSINE; /* The name says BP but it is used for all of them. */
D->use_prefilter = 1; /* first, a bandpass filter. */
D->prefilter_baud = 0.23;
D->pre_filter_len_bits = 156 * 1200. / 44100.;
D->pre_window = BP_WINDOW_TRUNCATED;
D->ms_filter_len_bits = 74 * 1200. / 44100.;
D->ms_window = BP_WINDOW_COSINE;
D->lpf_use_fir = 1;
D->lpf_baud = 1.18;
D->lp_filter_len_bits = 63 * 1200. / 44100.;
D->lp_window = BP_WINDOW_TRUNCATED;
//D->agc_fast_attack = 0.300;
//D->agc_slow_decay = 0.000185;
D->agc_fast_attack = 0.820;
D->agc_slow_decay = 0.000214;
D->hysteresis = 0.01;
//D->pll_locked_inertia = 0.57;
//D->pll_searching_inertia = 0.33;
D->pll_locked_inertia = 0.74;
D->pll_searching_inertia = 0.50;
break;
default:
text_color_set(DW_COLOR_ERROR);
dw_printf ("Invalid filter profile = %c\n", profile);
exit (1);
}
#ifdef TUNE_PRE_WINDOW
D->pre_window = TUNE_PRE_WINDOW;
#endif
#ifdef TUNE_MS_WINDOW
D->ms_window = TUNE_MS_WINDOW;
#endif
#ifdef TUNE_MS2_WINDOW
D->ms2_window = TUNE_MS2_WINDOW;
#endif
#ifdef TUNE_LP_WINDOW
D->lp_window = TUNE_LP_WINDOW;
#endif
#if defined(TUNE_AGC_FAST) && defined(TUNE_AGC_SLOW)
D->agc_fast_attack = TUNE_AGC_FAST;
D->agc_slow_decay = TUNE_AGC_SLOW;
#endif
#ifdef TUNE_HYST
D->hysteresis = TUNE_HYST;
#endif
#if defined(TUNE_PLL_LOCKED) && defined(TUNE_PLL_SEARCHING)
D->pll_locked_inertia = TUNE_PLL_LOCKED;
D->pll_searching_inertia = TUNE_PLL_SEARCHING;
#endif
#ifdef TUNE_LPF_BAUD
D->lpf_baud = TUNE_LPF_BAUD;
#endif
#ifdef TUNE_PRE_BAUD
D->prefilter_baud = TUNE_PRE_BAUD;
#endif
#ifdef TUNE_LP_DELAY_FRACT
D->lp_delay_fract = TUNE_LP_DELAY_FRACT;
#endif
/*
* Calculate constants used for timing.
* The audio sample rate must be at least a few times the data rate.
*
* Baud is an integer so we hack in a fine ajustment for EAS.
* Probably makes no difference because the DPLL keeps it in sync.
*
* A fraction if a Hz would make no difference for the filters.
*/
if (baud == 521) {
D->pll_step_per_sample = (int) round((TICKS_PER_PLL_CYCLE * (double)520.83) / ((double)samples_per_sec));
}
else {
D->pll_step_per_sample = (int) round((TICKS_PER_PLL_CYCLE * (double)baud) / ((double)samples_per_sec));
}
/*
* Convert number of bit times to number of taps.
*/
D->pre_filter_size = (int) round( D->pre_filter_len_bits * (float)samples_per_sec / (float)baud );
D->ms_filter_size = (int) round( D->ms_filter_len_bits * (float)samples_per_sec / (float)baud );
D->lp_filter_size = (int) round( D->lp_filter_len_bits * (float)samples_per_sec / (float)baud );
/* Experiment with other sizes. */
#ifdef TUNE_PRE_FILTER_SIZE
D->pre_filter_size = TUNE_PRE_FILTER_SIZE;
#endif
#ifdef TUNE_MS_FILTER_SIZE
D->ms_filter_size = TUNE_MS_FILTER_SIZE;
#endif
#ifdef TUNE_LP_FILTER_SIZE
D->lp_filter_size = TUNE_LP_FILTER_SIZE;
#endif
//assert (D->pre_filter_size >= 4);
assert (D->ms_filter_size >= 4);
//assert (D->lp_filter_size >= 4);
if (D->pre_filter_size > MAX_FILTER_SIZE)
{
text_color_set (DW_COLOR_ERROR);
dw_printf ("Calculated filter size of %d is too large.\n", D->pre_filter_size);
dw_printf ("Decrease the audio sample rate or increase the baud rate or\n");
dw_printf ("recompile the application with MAX_FILTER_SIZE larger than %d.\n",
MAX_FILTER_SIZE);
exit (1);
}
if (D->ms_filter_size > MAX_FILTER_SIZE)
{
text_color_set (DW_COLOR_ERROR);
dw_printf ("Calculated filter size of %d is too large.\n", D->ms_filter_size);
dw_printf ("Decrease the audio sample rate or increase the baud rate or\n");
dw_printf ("recompile the application with MAX_FILTER_SIZE larger than %d.\n",
MAX_FILTER_SIZE);
exit (1);
}
if (D->lp_filter_size > MAX_FILTER_SIZE)
{
text_color_set (DW_COLOR_ERROR);
dw_printf ("Calculated filter size of %d is too large.\n", D->pre_filter_size);
dw_printf ("Decrease the audio sample rate or increase the baud rate or\n");
dw_printf ("recompile the application with MAX_FILTER_SIZE larger than %d.\n",
MAX_FILTER_SIZE);
exit (1);
}
/*
* Optionally apply a bandpass ("pre") filter to attenuate
* frequencies outside the range of interest.
* This was first used for the "D" profile for 300 baud
* which uses narrow shift. We expect it to have significant
* benefit for a narrow shift.
* In version 1.2, we will also try it with 1200 baud "E" as
* an experiment to see how much it actually helps.
*/
if (D->use_prefilter) {
float f1, f2;
f1 = MIN(mark_freq,space_freq) - D->prefilter_baud * baud;
f2 = MAX(mark_freq,space_freq) + D->prefilter_baud * baud;
#if 0
text_color_set(DW_COLOR_DEBUG);
dw_printf ("Generating prefilter %.0f to %.0f Hz.\n", f1, f2);
#endif
f1 = f1 / (float)samples_per_sec;
f2 = f2 / (float)samples_per_sec;
gen_bandpass (f1, f2, D->pre_filter, D->pre_filter_size, D->pre_window);
}
/*
* Filters for detecting mark and space tones.
*/
#if DEBUG1
text_color_set(DW_COLOR_DEBUG);
dw_printf ("%s: \n", __FILE__);
dw_printf ("%d baud, %d samples_per_sec\n", baud, samples_per_sec);
dw_printf ("AFSK %d & %d Hz\n", mark_freq, space_freq);
dw_printf ("spll_step_per_sample = %d = 0x%08x\n", D->pll_step_per_sample, D->pll_step_per_sample);
dw_printf ("D->ms_filter_size = %d = 0x%08x\n", D->ms_filter_size, D->ms_filter_size);
dw_printf ("\n");
dw_printf ("Mark\n");
dw_printf (" j shape M sin M cos \n");
#endif
gen_ms (mark_freq, samples_per_sec, D->m_sin_table, D->m_cos_table, D->ms_filter_size, D->ms_window);
#if DEBUG1
text_color_set(DW_COLOR_DEBUG);
dw_printf ("Space\n");
dw_printf (" j shape S sin S cos\n");
#endif
gen_ms (space_freq, samples_per_sec, D->s_sin_table, D->s_cos_table, D->ms_filter_size, D->ms_window);
/*
* Now the lowpass filter.
* I thought we'd want a cutoff of about 0.5 the baud rate
* but it turns out about 1.1x is better. Still investigating...
*/
if (D->lpf_use_fir) {
float fc;
fc = baud * D->lpf_baud / (float)samples_per_sec;
D->lp_filter_delay = gen_lowpass (fc, D->lp_filter, D->lp_filter_size, D->lp_window, D->lp_delay_fract);
}
else {
// D->lp_filter_delay =
// Only needed for looking back and I don't expect to use IIR in that case.
}
/*
* A non-whole number of cycles results in a DC bias.
* Let's see if it helps to take it out.
* Actually makes things worse: 20 fewer decoded.
* Might want to try again after EXPERIMENTC.
*/
#if 0
#ifndef AVOID_FLOATING_POINT
failed experiment
dc_bias = 0;
for (j=0; j<D->ms_filter_size; j++) {
dc_bias += D->m_sin_table[j];
}
for (j=0; j<D->ms_filter_size; j++) {
D->m_sin_table[j] -= dc_bias / D->ms_filter_size;
}
dc_bias = 0;
for (j=0; j<D->ms_filter_size; j++) {
dc_bias += D->m_cos_table[j];
}
for (j=0; j<D->ms_filter_size; j++) {
D->m_cos_table[j] -= dc_bias / D->ms_filter_size;
}
dc_bias = 0;
for (j=0; j<D->ms_filter_size; j++) {
dc_bias += D->s_sin_table[j];
}
for (j=0; j<D->ms_filter_size; j++) {
D->s_sin_table[j] -= dc_bias / D->ms_filter_size;
}
dc_bias = 0;
for (j=0; j<D->ms_filter_size; j++) {
dc_bias += D->s_cos_table[j];
}
for (j=0; j<D->ms_filter_size; j++) {
D->s_cos_table[j] -= dc_bias / D->ms_filter_size;
}
#endif
#endif
/*
* In version 1.2 we try another experiment.
* Try using multiple slicing points instead of the traditional AGC.
*/
space_gain[0] = MIN_G;
float step = powf(10.0, log10f(MAX_G/MIN_G) / (MAX_SUBCHANS-1));
for (j=1; j<MAX_SUBCHANS; j++) {
space_gain[j] = space_gain[j-1] * step;
}
#if 0
text_color_set(DW_COLOR_DEBUG);
for (j=0; j<MAX_SUBCHANS; j++) {
float db = 20.0 * log10f(space_gain[j]);
dw_printf ("G = %.3f, %+.1f dB\n", space_gain[j], db);
}
#endif
} /* fsk_gen_filter */
/*-------------------------------------------------------------------
*
* Name: demod_afsk_process_sample
*
* Purpose: (1) Demodulate the AFSK signal.
* (2) Recover clock and data.
*
* Inputs: chan - Audio channel. 0 for left, 1 for right.
* subchan - modem of the channel.
* sam - One sample of audio.
* Should be in range of -32768 .. 32767.
*
* Returns: None
*
* Descripion: We start off with two bandpass filters tuned to
* the given frequencies. In the case of VHF packet
* radio, this would be 1200 and 2200 Hz.
*
* The bandpass filter amplitudes are compared to
* obtain the demodulated signal.
*
* 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.
*
*--------------------------------------------------------------------*/
inline static void nudge_pll (int chan, int subchan, int slice, int demod_data, struct demodulator_state_s *D);
__attribute__((hot))
void demod_afsk_process_sample (int chan, int subchan, int sam, struct demodulator_state_s *D)
{
float fsam;
//float abs_fsam;
float m_sum1, m_sum2, s_sum1, s_sum2;
float m_amp, s_amp;
float m_norm, s_norm;
float demod_out;
#if DEBUG4
static FILE *demod_log_fp = NULL;
static int seq = 0; /* for log file name */
#endif
//int j;
int demod_data;
assert (chan >= 0 && chan < MAX_CHANS);
assert (subchan >= 0 && subchan < MAX_SUBCHANS);
/*
* Filters use last 'filter_size' samples.
*
* First push the older samples down.
*
* Finally, put the most recent at the beginning.
*
* Future project? Can we do better than shifting each time?
*/
/* Scale to nice number, TODO: range -1.0 to +1.0, not 2. */
fsam = sam / 16384.0f;
//abs_fsam = fsam >= 0.0f ? fsam : -fsam;
/*
* Optional bandpass filter before the mark/space discriminator.
*/
// FIXME: calculate how much we really need.
int extra = 0;
if (D->use_prefilter) {
float cleaner;
push_sample (fsam, D->raw_cb, D->pre_filter_size);
cleaner = convolve (D->raw_cb, D->pre_filter, D->pre_filter_size);
push_sample (cleaner, D->ms_in_cb, D->ms_filter_size + extra);
}
else {
push_sample (fsam, D->ms_in_cb, D->ms_filter_size + extra);
}
/*
* Next we have bandpass filters for the mark and space tones.
*/
/*
* find amplitude of "Mark" tone.
*/
m_sum1 = convolve (D->ms_in_cb, D->m_sin_table, D->ms_filter_size);
m_sum2 = convolve (D->ms_in_cb, D->m_cos_table, D->ms_filter_size);
m_amp = sqrtf(m_sum1 * m_sum1 + m_sum2 * m_sum2);
/*
* Find amplitude of "Space" tone.
*/
s_sum1 = convolve (D->ms_in_cb, D->s_sin_table, D->ms_filter_size);
s_sum2 = convolve (D->ms_in_cb, D->s_cos_table, D->ms_filter_size);
s_amp = sqrtf(s_sum1 * s_sum1 + s_sum2 * s_sum2);
/*
* Apply some low pass filtering BEFORE the AGC to remove
* overshoot, ringing, and other bad stuff.
*
* A simple IIR filter is faster but FIR produces better results.
*
* It is a balancing act between removing high frequency components
* from the tone dectection while letting the data thru.
*/
if (D->lpf_use_fir) {
push_sample (m_amp, D->m_amp_cb, D->lp_filter_size);
m_amp = convolve (D->m_amp_cb, D->lp_filter, D->lp_filter_size);
push_sample (s_amp, D->s_amp_cb, D->lp_filter_size);
s_amp = convolve (D->s_amp_cb, D->lp_filter, D->lp_filter_size);
}
else {
/* Original, but faster, IIR. */
m_amp = D->lpf_iir * m_amp + (1.0f - D->lpf_iir) * D->m_amp_prev;
D->m_amp_prev = m_amp;
s_amp = D->lpf_iir * s_amp + (1.0f - D->lpf_iir) * D->s_amp_prev;
D->s_amp_prev = s_amp;
}
/*
* Version 1.2: Try new approach to capturing the amplitude for display.
* This is same as the AGC above without the normalization step.
* We want decay to be substantially slower to get a longer
* range idea of the received audio.
*/
if (m_amp >= D->alevel_mark_peak) {
D->alevel_mark_peak = m_amp * D->quick_attack + D->alevel_mark_peak * (1.0f - D->quick_attack);
}
else {
D->alevel_mark_peak = m_amp * D->sluggish_decay + D->alevel_mark_peak * (1.0f - D->sluggish_decay);
}
if (s_amp >= D->alevel_space_peak) {
D->alevel_space_peak = s_amp * D->quick_attack + D->alevel_space_peak * (1.0f - D->quick_attack);
}
else {
D->alevel_space_peak = s_amp * D->sluggish_decay + D->alevel_space_peak * (1.0f - D->sluggish_decay);
}
/*
* Which tone is stronger?
*
* In an ideal world, simply compare. In my first naive attempt, that
* worked perfectly with perfect signals. In the real world, we don't
* have too many perfect signals.
*
* Here is an excellent explanation:
* http://www.febo.com/packet/layer-one/transmit.html
*
* Under real conditions, we find that the higher tone usually has a
* considerably smaller amplitude due to the passband characteristics
* of the transmitter and receiver. To make matters worse, it
* varies considerably from one station to another.
*
* The two filters also have different amounts of DC bias.
*
* My solution was to apply automatic gain control (AGC) to the mark and space
* levels. This works by looking at the minimum and maximum outputs
* for each filter and scaling the results to be roughly in the -0.5 to +0.5 range.
* Results were excellent after tweaking the attack and decay times.
*
* 4X6IZ took a different approach. See QEX Jul-Aug 2012.
*
* He ran two different demodulators in parallel. One of them boosted the higher
* frequency tone by 6 dB. Any duplicates were removed. This produced similar results.
* He also used a bandpass filter before the mark/space filters.
* I haven't tried this combination yet for 1200 baud.
*
* First, let's take a look at Track 1 of the TNC test CD. Here the receiver
* has a flat response. We find the mark/space strength ratios very from 0.53 to 1.38
* with a median of 0.81. This in in line with expections because most
* transmitters add pre-emphasis to boost the higher audio frequencies.
* Track 2 should more closely resemble what comes out of the speaker on a typical
* transceiver. Here we see a ratio from 1.73 to 3.81 with a median of 2.48.
*
* This is similar to my observations of local signals, from the speaker.
* The amplitude ratio varies from 1.48 to 3.41 with a median of 2.70.
*
* Rather than only two filters, let's try slicing the data in more places.
*/
/* Fast attack and slow decay. */
/* Numbers were obtained by trial and error from actual */
/* recorded less-than-optimal signals. */
/* See fsk_demod_agc.h for more information. */
m_norm = agc (m_amp, D->agc_fast_attack, D->agc_slow_decay, &(D->m_peak), &(D->m_valley));
s_norm = agc (s_amp, D->agc_fast_attack, D->agc_slow_decay, &(D->s_peak), &(D->s_valley));
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_out = m_norm - s_norm;
/* Try adding some Hysteresis. */
/* (Not to be confused with Hysteria.) */
if (demod_out > D->hysteresis) {
demod_data = 1;
}
else if (demod_out < (- (D->hysteresis))) {
demod_data = 0;
}
else {
demod_data = D->slicer[subchan].prev_demod_data;
}
nudge_pll (chan, subchan, 0, demod_data, D);
}
else {
int slice;
for (slice=0; slice<D->num_slicers; slice++) {
demod_data = m_amp > s_amp * space_gain[slice];
nudge_pll (chan, subchan, slice, demod_data, D);
}
}
#if DEBUG4
if (chan == 0) {
if (D->slicer[slice].data_detect) {
char fname[30];
if (demod_log_fp == NULL) {
seq++;
snprintf (fname, sizeof(fname), "demod/%04d.csv", seq);
if (seq == 1) mkdir ("demod", 0777);
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, Mark, Space, Demod, Data, Clock\n");
}
fprintf (demod_log_fp, "%.3f, %.3f, %.3f, %.3f, %.2f, %.2f\n", fsam + 3.5, m_norm + 2, s_norm + 2,
(m_norm - s_norm) / 2 + 1.5,
demod_data ? .9 : .55,
(D->data_clock_pll & 0x80000000) ? .1 : .45);
}
else {
if (demod_log_fp != NULL) {
fclose (demod_log_fp);
demod_log_fp = NULL;
}
}
}
#endif
} /* end demod_afsk_process_sample */
__attribute__((hot))
inline static void nudge_pll (int chan, int subchan, int slice, int demod_data, struct demodulator_state_s *D)
{
/*
* Finally, a PLL is used to sample near the centers of the data bits.
*
* D points to a demodulator for a channel/subchannel pair so we don't
* have to keep recalculating it.
*
* 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.
*
* Be a little more agressive about adjusting the PLL
* phase when searching for a signal. Don't change it as much when
* locked on to a signal.
*
* I don't think the optimal value will depend on the audio sample rate
* because this happens for each transition from the demodulator.
*/
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));
//text_color_set(DW_COLOR_DEBUG);
// dw_printf ("prev = %lx, new data clock pll = %lx\n" D->prev_d_c_pll, D->data_clock_pll);
if (D->slicer[slice].data_clock_pll < 0 && D->slicer[slice].prev_d_c_pll > 0) {
/* Overflow - this is where we sample. */
hdlc_rec_bit (chan, subchan, slice, demod_data, 0, -1);
pll_dcd_each_symbol2 (D, chan, subchan, slice);
}
// Transitions nudge the DPLL phase toward the incoming signal.
if (demod_data != D->slicer[slice].prev_demod_data) {
pll_dcd_signal_transition2 (D, slice, D->slicer[slice].data_clock_pll);
if (D->slicer[slice].data_detect) {
D->slicer[slice].data_clock_pll = (int)(D->slicer[slice].data_clock_pll * D->pll_locked_inertia);
}
else {
D->slicer[slice].data_clock_pll = (int)(D->slicer[slice].data_clock_pll * D->pll_searching_inertia);
}
}
/*
* Remember demodulator output so we can compare next time.
*/
D->slicer[slice].prev_demod_data = demod_data;
} /* end nudge_pll */
/* end demod_afsk.c */
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