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Overhaul PSK demodulator.

This commit is contained in:
wb2osz 2019-11-30 19:20:55 -05:00
parent b382e5fb0f
commit ff9eca682e
5 changed files with 332 additions and 344 deletions
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11
src/audio.h
View File

@ -145,16 +145,11 @@ struct audio_s {
// Original implementaion used alternative A for 2400 bbps PSK.
// Years later, we discover that MFJ-2400 used alternative B.
// It's likely the others did too.
// For release 1.6, default to original style but print warning.
// Later default to MFJ compatible and still print warning if
// if user did not pick one explicitly.
// It's likely the others did too. it also works a little better.
// Default to MFJ compatible and print warning if user did not
// pick one explicitly.
#if (MAJOR_VERSION > 1) || (MINOR_VERSION > 6)
#define V26_DEFAULT V26_B
#else
#define V26_DEFAULT V26_A
#endif
enum dtmf_decode_t { DTMF_DECODE_OFF, DTMF_DECODE_ON } dtmf_decode;

2
src/demod.c
View File

@ -494,7 +494,7 @@ int demod_init (struct audio_s *pa)
dw_printf ("Command line options -j and -J can be used for channel 0.\n");
dw_printf ("For more information, read the Dire Wolf User Guide and\n");
dw_printf ("2400-4800-PSK-for-APRS-Packet-Radio.pdf.\n");
dw_printf ("The default in this release could be different in a later release.\n");
dw_printf ("The default is now MFJ-2400 compatibility mode.\n");
save_audio_config_p->achan[chan].v26_alternative = V26_DEFAULT;
}

529
src/demod_psk.c
View File

@ -20,36 +20,17 @@
//#define DEBUG1 1 /* display debugging info */
//#define DEBUG3 1 /* print carrier detect changes. */
//#define DEBUG4 1 /* capture PSK demodulator output to log files */
//#define DEBUG5 1 /* Print bit stream */
/*------------------------------------------------------------------
*
* Module: demod_psk.c
*
* Purpose: Demodulator for Phase Shift Keying (PSK).
*
* This is my initial attempt at implementing a 2400 bps mode.
* The MFJ-2400 & AEA PK232-2400 used V.26 / Bell 201 so I will follow that precedent.
*
* Purpose: Demodulator for 2400 and 4800 bits per second Phase Shift Keying (PSK).
*
* Input: Audio samples from either a file or the "sound card."
*
* Outputs: Calls hdlc_rec_bit() for each bit demodulated.
*
* Current Status: New for Version 1.4.
*
* Don't know if this is correct and/or compatible with
* other implementations.
* There is a lot of stuff going on here with phase
* shifting, gray code, bit order for the dibit, NRZI and
* bit-stuffing for HDLC. Plenty of opportunity for
* misinterpreting a protocol spec or just stupid mistakes.
*
* References: MFJ-2400 Product description and manual:
*
* http://www.mfjenterprises.com/Product.php?productid=MFJ-2400
@ -82,16 +63,16 @@
* There are ealier references to an alternative B which uses other phase shifts offset
* by another 45 degrees.
*
* The XR-2123 does not perform the scrambling as specified in V.26 so I wonder if
* the vendors implemented it in software or just left it out.
* I left out scrambling for now. Eventually, I'd like to get my hands on an old
* 2400 bps TNC for compatibility testing.
*
* After getting QPSK working, it was not much more effort to add V.27 with 8 phases.
*
* https://www.itu.int/rec/dologin_pub.asp?lang=e&id=T-REC-V.27bis-198811-I!!PDF-E&type=items
* https://www.itu.int/rec/dologin_pub.asp?lang=e&id=T-REC-V.27ter-198811-I!!PDF-E&type=items
*
* Compatibility:
* V.26 has two variations, A and B. Initially I implemented the A alternative.
* It later turned out that the MFJ-2400 used the B alternative. In version 1.6 you have a
* choice between compatibility with MFJ (and probably the others) or the original implementation.
*
*---------------------------------------------------------------*/
#include "direwolf.h"
@ -115,6 +96,11 @@
#include "demod_psk.h"
#include "dsp.h"
static const int phase_to_gray_v26[4] = {0, 1, 3, 2};
static const int phase_to_gray_v27[8] = {1, 0, 2, 3, 7, 6, 4, 5};
static int phase_shift_to_symbol (float phase_shift, int bits_per_symbol, int *bit_quality);
/* Add sample to buffer and shift the rest down. */
@ -134,6 +120,7 @@ static inline float convolve (const float *__restrict__ data, const float *__res
float sum = 0.0;
int j;
#pragma GCC ivdep
for (j=0; j<filter_size; j++) {
sum += filter[j] * data[j];
}
@ -142,7 +129,7 @@ static inline float convolve (const float *__restrict__ data, const float *__res
}
/* Might replace this with faster, lower precision version someday. */
/* Might replace this with faster, lower precision, approximation someday if it does not harm results. */
static inline float my_atan2f (float y, float x)
{
@ -156,7 +143,7 @@ static inline float my_atan2f (float y, float x)
*
* Name: demod_psk_init
*
* Purpose: Initialization for an psk demodulator.
* Purpose: Initialization for a PSK demodulator.
* Select appropriate parameters and set up filters.
*
* Inputs: modem_type - MODEM_QPSK or MODEM_8PSK.
@ -166,8 +153,7 @@ static inline float my_atan2f (float y, float x)
* samples_per_sec - Audio sample rate.
*
* bps - Bits per second.
* Should be 2400 for V.26 but we don't enforce it.
* The carrier frequency will be proportional.
* Should be 2400 for V.26 or 4800 for V.27.
*
* profile - Select different variations. For QPSK:
*
@ -201,25 +187,20 @@ void demod_psk_init (enum modem_t modem_type, enum v26_e v26_alt, int samples_pe
memset (D, 0, sizeof(struct demodulator_state_s));
D->modem_type = modem_type;
D->v26_alt = v26_alt;
D->u.psk.v26_alt = v26_alt;
D->num_slicers = 1; // Haven't thought about this yet. Is it even applicable?
#ifdef TUNE_PROFILE
profile = TUNE_PROFILE;
#endif
if (modem_type == MODEM_QPSK) {
assert (D->v26_alt != V26_UNSPECIFIED);
assert (D->u.psk.v26_alt != V26_UNSPECIFIED);
correct_baud = bps / 2;
// Originally I thought of scaling it to the data rate,
// e.g. 2400 bps -> 1800 Hz, but decided to make it a
// constant since it is the same for V.26 and V.27.
carrier_freq = 1800;
#if DEBUG1
@ -231,11 +212,11 @@ void demod_psk_init (enum modem_t modem_type, enum v26_e v26_alt, int samples_pe
case 'P': /* Self correlation technique. */
D->use_prefilter = 0; /* No bandpass filter. */
D->u.psk.use_prefilter = 0; /* No bandpass filter. */
D->lpf_baud = 0.60;
D->lp_filter_len_bits = 39. * 1200. / 44100.;
D->lp_window = BP_WINDOW_COSINE;
D->u.psk.lpf_baud = 0.60;
D->u.psk.lp_filter_width_sym = 1.061; // 39. * 1200. / 44100.;
D->u.psk.lp_window = BP_WINDOW_COSINE;
D->pll_locked_inertia = 0.95;
D->pll_searching_inertia = 0.50;
@ -244,14 +225,14 @@ void demod_psk_init (enum modem_t modem_type, enum v26_e v26_alt, int samples_pe
case 'Q': /* Self correlation technique. */
D->use_prefilter = 1; /* Add a bandpass filter. */
D->prefilter_baud = 1.3;
D->pre_filter_len_bits = 55. * 1200. / 44100.;
D->pre_window = BP_WINDOW_COSINE;
D->u.psk.use_prefilter = 1; /* Add a bandpass filter. */
D->u.psk.prefilter_baud = 1.3;
D->u.psk.pre_filter_width_sym = 1.497; // 55. * 1200. / 44100.;
D->u.psk.pre_window = BP_WINDOW_COSINE;
D->lpf_baud = 0.60;
D->lp_filter_len_bits = 39. * 1200. / 44100.;
D->lp_window = BP_WINDOW_COSINE;
D->u.psk.lpf_baud = 0.60;
D->u.psk.lp_filter_width_sym = 1.061; // 39. * 1200. / 44100.;
D->u.psk.lp_window = BP_WINDOW_COSINE;
D->pll_locked_inertia = 0.87;
D->pll_searching_inertia = 0.50;
@ -265,13 +246,13 @@ void demod_psk_init (enum modem_t modem_type, enum v26_e v26_alt, int samples_pe
case 'R': /* Mix with local oscillator. */
D->psk_use_lo = 1;
D->u.psk.psk_use_lo = 1;
D->use_prefilter = 0; /* No bandpass filter. */
D->u.psk.use_prefilter = 0; /* No bandpass filter. */
D->lpf_baud = 0.70;
D->lp_filter_len_bits = 37. * 1200. / 44100.;
D->lp_window = BP_WINDOW_TRUNCATED;
D->u.psk.lpf_baud = 0.70;
D->u.psk.lp_filter_width_sym = 1.007; // 37. * 1200. / 44100.;
D->u.psk.lp_window = BP_WINDOW_TRUNCATED;
D->pll_locked_inertia = 0.925;
D->pll_searching_inertia = 0.50;
@ -280,16 +261,16 @@ void demod_psk_init (enum modem_t modem_type, enum v26_e v26_alt, int samples_pe
case 'S': /* Mix with local oscillator. */
D->psk_use_lo = 1;
D->u.psk.psk_use_lo = 1;
D->use_prefilter = 1; /* Add a bandpass filter. */
D->prefilter_baud = 0.55;
D->pre_filter_len_bits = 74. * 1200. / 44100.;
D->pre_window = BP_WINDOW_FLATTOP;
D->u.psk.use_prefilter = 1; /* Add a bandpass filter. */
D->u.psk.prefilter_baud = 0.55;
D->u.psk.pre_filter_width_sym = 2.014; // 74. * 1200. / 44100.;
D->u.psk.pre_window = BP_WINDOW_FLATTOP;
D->lpf_baud = 0.60;
D->lp_filter_len_bits = 39. * 1200. / 44100.;
D->lp_window = BP_WINDOW_COSINE;
D->u.psk.lpf_baud = 0.60;
D->u.psk.lp_filter_width_sym = 1.061; // 39. * 1200. / 44100.;
D->u.psk.lp_window = BP_WINDOW_COSINE;
D->pll_locked_inertia = 0.925;
D->pll_searching_inertia = 0.50;
@ -297,11 +278,11 @@ void demod_psk_init (enum modem_t modem_type, enum v26_e v26_alt, int samples_pe
break;
}
D->ms_filter_len_bits = 1.25; // Delay line > 13/12 * symbol period
D->u.psk.delay_line_width_sym = 1.25; // Delay line > 13/12 * symbol period
D->coffs = (int) round( (11.f / 12.f) * (float)samples_per_sec / (float)correct_baud );
D->boffs = (int) round( (float)samples_per_sec / (float)correct_baud );
D->soffs = (int) round( (13.f / 12.f) * (float)samples_per_sec / (float)correct_baud );
D->u.psk.coffs = (int) round( (11.f / 12.f) * (float)samples_per_sec / (float)correct_baud );
D->u.psk.boffs = (int) round( (float)samples_per_sec / (float)correct_baud );
D->u.psk.soffs = (int) round( (13.f / 12.f) * (float)samples_per_sec / (float)correct_baud );
}
else {
@ -318,11 +299,11 @@ void demod_psk_init (enum modem_t modem_type, enum v26_e v26_alt, int samples_pe
case 'T': /* Self correlation technique. */
D->use_prefilter = 0; /* No bandpass filter. */
D->u.psk.use_prefilter = 0; /* No bandpass filter. */
D->lpf_baud = 1.15;
D->lp_filter_len_bits = 32. * 1200. / 44100.;
D->lp_window = BP_WINDOW_COSINE;
D->u.psk.lpf_baud = 1.15;
D->u.psk.lp_filter_width_sym = 0.871; // 32. * 1200. / 44100.;
D->u.psk.lp_window = BP_WINDOW_COSINE;
D->pll_locked_inertia = 0.95;
D->pll_searching_inertia = 0.50;
@ -331,14 +312,14 @@ void demod_psk_init (enum modem_t modem_type, enum v26_e v26_alt, int samples_pe
case 'U': /* Self correlation technique. */
D->use_prefilter = 1; /* Add a bandpass filter. */
D->prefilter_baud = 0.9;
D->pre_filter_len_bits = 21. * 1200. / 44100.;
D->pre_window = BP_WINDOW_FLATTOP;
D->u.psk.use_prefilter = 1; /* Add a bandpass filter. */
D->u.psk.prefilter_baud = 0.9;
D->u.psk.pre_filter_width_sym = 0.571; // 21. * 1200. / 44100.;
D->u.psk.pre_window = BP_WINDOW_FLATTOP;
D->lpf_baud = 1.15;
D->lp_filter_len_bits = 32. * 1200. / 44100.;
D->lp_window = BP_WINDOW_COSINE;
D->u.psk.lpf_baud = 1.15;
D->u.psk.lp_filter_width_sym = 0.871; // 32. * 1200. / 44100.;
D->u.psk.lp_window = BP_WINDOW_COSINE;
D->pll_locked_inertia = 0.87;
D->pll_searching_inertia = 0.50;
@ -352,13 +333,13 @@ void demod_psk_init (enum modem_t modem_type, enum v26_e v26_alt, int samples_pe
case 'V': /* Mix with local oscillator. */
D->psk_use_lo = 1;
D->u.psk.psk_use_lo = 1;
D->use_prefilter = 0; /* No bandpass filter. */
D->u.psk.use_prefilter = 0; /* No bandpass filter. */
D->lpf_baud = 0.85;
D->lp_filter_len_bits = 31. * 1200. / 44100.;
D->lp_window = BP_WINDOW_COSINE;
D->u.psk.lpf_baud = 0.85;
D->u.psk.lp_filter_width_sym = 0.844; // 31. * 1200. / 44100.;
D->u.psk.lp_window = BP_WINDOW_COSINE;
D->pll_locked_inertia = 0.925;
D->pll_searching_inertia = 0.50;
@ -367,16 +348,16 @@ void demod_psk_init (enum modem_t modem_type, enum v26_e v26_alt, int samples_pe
case 'W': /* Mix with local oscillator. */
D->psk_use_lo = 1;
D->u.psk.psk_use_lo = 1;
D->use_prefilter = 1; /* Add a bandpass filter. */
D->prefilter_baud = 0.85;
D->pre_filter_len_bits = 31. * 1200. / 44100.;
D->pre_window = BP_WINDOW_COSINE;
D->u.psk.use_prefilter = 1; /* Add a bandpass filter. */
D->u.psk.prefilter_baud = 0.85;
D->u.psk.pre_filter_width_sym = 0.844; // 31. * 1200. / 44100.;
D->u.psk.pre_window = BP_WINDOW_COSINE;
D->lpf_baud = 0.85;
D->lp_filter_len_bits = 31. * 1200. / 44100.;
D->lp_window = BP_WINDOW_COSINE;
D->u.psk.lpf_baud = 0.85;
D->u.psk.lp_filter_width_sym = 0.844; // 31. * 1200. / 44100.;
D->u.psk.lp_window = BP_WINDOW_COSINE;
D->pll_locked_inertia = 0.925;
D->pll_searching_inertia = 0.50;
@ -384,42 +365,36 @@ void demod_psk_init (enum modem_t modem_type, enum v26_e v26_alt, int samples_pe
break;
}
D->ms_filter_len_bits = 1.25; // Delay line > 10/9 * symbol period
D->u.psk.delay_line_width_sym = 1.25; // Delay line > 10/9 * symbol period
D->coffs = (int) round( (8.f / 9.f) * (float)samples_per_sec / (float)correct_baud );
D->boffs = (int) round( (float)samples_per_sec / (float)correct_baud );
D->soffs = (int) round( (10.f / 9.f) * (float)samples_per_sec / (float)correct_baud );
D->u.psk.coffs = (int) round( (8.f / 9.f) * (float)samples_per_sec / (float)correct_baud );
D->u.psk.boffs = (int) round( (float)samples_per_sec / (float)correct_baud );
D->u.psk.soffs = (int) round( (10.f / 9.f) * (float)samples_per_sec / (float)correct_baud );
}
if (D->psk_use_lo) {
D->lo_step = (int) round( 256. * 256. * 256. * 256. * carrier_freq / (double)samples_per_sec);
if (D->u.psk.psk_use_lo) {
D->u.psk.lo_step = (int) round( 256. * 256. * 256. * 256. * carrier_freq / (double)samples_per_sec);
// Our own sin table for speed later.
assert (MAX_FILTER_SIZE >= 256);
for (j = 0; j < 256; j++) {
D->m_sin_table[j] = sinf(2.f * (float)M_PI * j / 256.f);
D->u.psk.sin_table256[j] = sinf(2.f * (float)M_PI * j / 256.f);
}
}
#ifdef TUNE_PRE_BAUD
D->prefilter_baud = TUNE_PRE_BAUD;
D->u.psk.prefilter_baud = TUNE_PRE_BAUD;
#endif
#ifdef TUNE_PRE_WINDOW
D->pre_window = TUNE_PRE_WINDOW;
D->u.psk.pre_window = TUNE_PRE_WINDOW;
#endif
#ifdef TUNE_LPF_BAUD
D->lpf_baud = TUNE_LPF_BAUD;
D->u.psk.lpf_baud = TUNE_LPF_BAUD;
#endif
#ifdef TUNE_LP_WINDOW
D->lp_window = TUNE_LP_WINDOW;
#endif
#ifdef TUNE_HYST
D->hysteresis = TUNE_HYST;
D->u.psk.lp_window = TUNE_LP_WINDOW;
#endif
#if defined(TUNE_PLL_SEARCHING)
@ -441,44 +416,41 @@ void demod_psk_init (enum modem_t modem_type, enum v26_e v26_alt, int samples_pe
* Convert number of symbol times to number of taps.
*/
D->pre_filter_size = (int) round( D->pre_filter_len_bits * (float)samples_per_sec / (float)correct_baud );
D->ms_filter_size = (int) round( D->ms_filter_len_bits * (float)samples_per_sec / (float)correct_baud );
D->lp_filter_size = (int) round( D->lp_filter_len_bits * (float)samples_per_sec / (float)correct_baud );
D->u.psk.pre_filter_taps = (int) round( D->u.psk.pre_filter_width_sym * (float)samples_per_sec / (float)correct_baud );
D->u.psk.delay_line_taps = (int) round( D->u.psk.delay_line_width_sym * (float)samples_per_sec / (float)correct_baud );
D->u.psk.lp_filter_taps = (int) round( D->u.psk.lp_filter_width_sym * (float)samples_per_sec / (float)correct_baud );
#ifdef TUNE_PRE_FILTER_SIZE
D->pre_filter_size = TUNE_PRE_FILTER_SIZE;
#ifdef TUNE_PRE_FILTER_TAPS
D->u.psk.pre_filter_taps = TUNE_PRE_FILTER_TAPS;
#endif
#ifdef TUNE_LP_FILTER_SIZE
D->lp_filter_size = TUNE_LP_FILTER_SIZE;
#ifdef TUNE_lp_filter_taps
D->u.psk.lp_filter_taps = TUNE_lp_filter_taps;
#endif
if (D->pre_filter_size > MAX_FILTER_SIZE)
{
if (D->u.psk.pre_filter_taps > 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 ("Calculated pre filter size of %d is too large.\n", D->u.psk.pre_filter_taps);
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)
{
if (D->u.psk.delay_line_taps > 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 ("Calculated delay line size of %d is too large.\n", D->u.psk.delay_line_taps);
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)
{
if (D->u.psk.lp_filter_taps > 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 ("Calculated low pass filter size of %d is too large.\n", D->u.psk.lp_filter_taps);
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);
@ -488,14 +460,17 @@ void demod_psk_init (enum modem_t modem_type, enum v26_e v26_alt, int samples_pe
/*
* Optionally apply a bandpass ("pre") filter to attenuate
* frequencies outside the range of interest.
* It's a tradeoff. Attenuate frequencies outside the the range of interest
* but also distort the signal. This demodulator is not compuationally
* intensive so we can usually run both in parallel.
*/
if (D->use_prefilter) {
if (D->u.psk.use_prefilter) {
float f1, f2;
f1 = carrier_freq - D->prefilter_baud * correct_baud;
f2 = carrier_freq + D->prefilter_baud * correct_baud;
#if 0
f1 = carrier_freq - D->u.psk.prefilter_baud * correct_baud;
f2 = carrier_freq + D->u.psk.prefilter_baud * correct_baud;
#if DEBUG1
text_color_set(DW_COLOR_DEBUG);
dw_printf ("Generating prefilter %.0f to %.0f Hz.\n", (double)f1, (double)f2);
#endif
@ -508,15 +483,15 @@ void demod_psk_init (enum modem_t modem_type, enum v26_e v26_alt, int samples_pe
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);
gen_bandpass (f1, f2, D->u.psk.pre_filter, D->u.psk.pre_filter_taps, D->u.psk.pre_window);
}
/*
* Now the lowpass filter.
*/
float fc = correct_baud * D->lpf_baud / (float)samples_per_sec;
(void)gen_lowpass (fc, D->lp_filter, D->lp_filter_size, D->lp_window, 0);
float fc = correct_baud * D->u.psk.lpf_baud / (float)samples_per_sec;
gen_lowpass (fc, D->u.psk.lp_filter, D->u.psk.lp_filter_taps, D->u.psk.lp_window, 0);
/*
* No point in having multiple numbers for signal level.
@ -525,10 +500,98 @@ void demod_psk_init (enum modem_t modem_type, enum v26_e v26_alt, int samples_pe
D->alevel_mark_peak = -1;
D->alevel_space_peak = -1;
#if 0
// QPSK - CSV format to make plot.
printf ("Phase shift degrees, bit 0, quality 0, bit 1, quality 1\n");
for (int degrees = 0; degrees <= 360; degrees++) {
float a = degrees * M_PI * 2./ 360.;
int bit_quality[3];
int new_gray = phase_shift_to_symbol (a, 2, bit_quality);
float offset = 3 * 1.5;
printf ("%d, ", degrees);
printf ("%.3f, ", offset + (new_gray & 1)); offset -= 1.5;
printf ("%.3f, ", offset + (bit_quality[0] / 100.)); offset -= 1.5;
printf ("%.3f, ", offset + ((new_gray >> 1) & 1)); offset -= 1.5;
printf ("%.3f\n", offset + (bit_quality[1] / 100.));
}
#endif
#if 0
// 8-PSK - CSV format to make plot.
printf ("Phase shift degrees, bit 0, quality 0, bit 1, quality 1, bit 2, quality 2\n");
for (int degrees = 0; degrees <= 360; degrees++) {
float a = degrees * M_PI * 2./ 360.;
int bit_quality[3];
int new_gray = phase_shift_to_symbol (a, 3, bit_quality);
float offset = 5 * 1.5;
printf ("%d, ", degrees);
printf ("%.3f, ", offset + (new_gray & 1)); offset -= 1.5;
printf ("%.3f, ", offset + (bit_quality[0] / 100.)); offset -= 1.5;
printf ("%.3f, ", offset + ((new_gray >> 1) & 1)); offset -= 1.5;
printf ("%.3f, ", offset + (bit_quality[1] / 100.)); offset -= 1.5;
printf ("%.3f, ", offset + ((new_gray >> 2) & 1)); offset -= 1.5;
printf ("%.3f\n", offset + (bit_quality[2] / 100.));
}
#endif
} /* demod_psk_init */
/*-------------------------------------------------------------------
*
* Name: phase_shift_to_symbol
*
* Purpose: Translate phase shift, between two symbols, into 2 or 3 bits.
*
* Inputs: phase_shift - in radians.
*
* bits_per_symbol - 2 for QPSK, 3 for 8PSK.
*
* Outputs: bit_quality[] - Value of 0 (at threshold) to 100 (perfect) for each bit.
*
* Returns: 2 or 3 bit symbol value in Gray code.
*
*--------------------------------------------------------------------*/
__attribute__((hot)) __attribute__((always_inline))
static inline int phase_shift_to_symbol (float phase_shift, int bits_per_symbol, int * __restrict__ bit_quality)
{
// Number of different symbol states.
assert (bits_per_symbol == 2 || bits_per_symbol == 3);
int N = 1 << bits_per_symbol;
assert (N == 4 || N == 8);
// Scale angle to 1 per symbol then separate into integer and fractional parts.
float a = phase_shift * (float)N / (M_PI * 2.0f);
while (a >= (float)N) a -= (float)N;
while (a < 0.) a += (float)N;
int i = (int)a;
if (i == N) i = N-1; // Should be < N. Watch out for possible roundoff errors.
float f = a - (float)i;
assert (i >= 0 && i < N);
assert (f >= -0.001f && f <= 1.001f);
// Interpolate between the ideal angles to get a level of certainty.
int result = 0;
for (int b = 0; b < bits_per_symbol; b++) {
float demod = bits_per_symbol == 2 ?
((phase_to_gray_v26[i] >> b) & 1) * (1.0f - f) + ((phase_to_gray_v26[(i+1)&3] >> b) & 1) * f :
((phase_to_gray_v27[i] >> b) & 1) * (1.0f - f) + ((phase_to_gray_v27[(i+1)&7] >> b) & 1) * f;
// Slice to get boolean value and quality measurement.
if (demod >= 0.5f) result |= 1<<b;
bit_quality[b] = lrintf (100.0f * 2.0f * fabsf(demod - 0.5f));
}
return (result);
} // end phase_shift_to_symbol
/*-------------------------------------------------------------------
*
@ -545,7 +608,7 @@ void demod_psk_init (enum modem_t modem_type, enum v26_e v26_alt, int samples_pe
*
* Outputs: For each recovered data bit, we call:
*
* hdlc_rec (channel, demodulated_bit);
* hdlc_rec (channel etc., demodulated_bit, quality);
*
* to decode HDLC frames from the stream of bits.
*
@ -585,203 +648,112 @@ void demod_psk_init (enum modem_t modem_type, enum v26_e v26_alt, int samples_pe
inline static void nudge_pll (int chan, int subchan, int slice, int demod_bits, struct demodulator_state_s *D);
inline static void nudge_pll (int chan, int subchan, int slice, int demod_bits, struct demodulator_state_s *D, int *bit_quality);
__attribute__((hot))
void demod_psk_process_sample (int chan, int subchan, int sam, struct demodulator_state_s *D)
{
float fsam;
float sam_x_cos, sam_x_sin;
float I, Q;
int demod_phase_shift; // Phase shift relative to previous symbol.
// range 0-3, 1 unit for each 90 degrees.
int slice = 0;
#if DEBUG4
static FILE *demod_log_fp = NULL;
static int log_file_seq = 0; /* Part of log file name */
#endif
int slice = 0; // Would it make sense to have more than one?
assert (chan >= 0 && chan < MAX_CHANS);
assert (subchan >= 0 && subchan < MAX_SUBCHANS);
/* Scale to nice number for plotting during debug. */
fsam = sam / 16384.0f;
float fsam = sam / 16384.0f;
/*
* Optional bandpass filter before the phase detector.
*/
if (D->use_prefilter) {
push_sample (fsam, D->raw_cb, D->pre_filter_size);
fsam = convolve (D->raw_cb, D->pre_filter, D->pre_filter_size);
if (D->u.psk.use_prefilter) {
push_sample (fsam, D->u.psk.audio_in, D->u.psk.pre_filter_taps);
fsam = convolve (D->u.psk.audio_in, D->u.psk.pre_filter, D->u.psk.pre_filter_taps);
}
if (D->psk_use_lo) {
float a, delta;
int id;
if (D->u.psk.psk_use_lo) {
/*
* Mix with local oscillator to obtain phase.
* The absolute phase doesn't matter.
* We are just concerned with the change since the previous symbol.
*/
sam_x_cos = fsam * D->m_sin_table[((D->lo_phase >> 24) + 64) & 0xff];
float sam_x_cos = fsam * D->u.psk.sin_table256[((D->u.psk.lo_phase >> 24) + 64) & 0xff];
push_sample (sam_x_cos, D->u.psk.I_raw, D->u.psk.lp_filter_taps);
float I = convolve (D->u.psk.I_raw, D->u.psk.lp_filter, D->u.psk.lp_filter_taps);
sam_x_sin = fsam * D->m_sin_table[(D->lo_phase >> 24) & 0xff];
float sam_x_sin = fsam * D->u.psk.sin_table256[(D->u.psk.lo_phase >> 24) & 0xff];
push_sample (sam_x_sin, D->u.psk.Q_raw, D->u.psk.lp_filter_taps);
float Q = convolve (D->u.psk.Q_raw, D->u.psk.lp_filter, D->u.psk.lp_filter_taps);
push_sample (sam_x_cos, D->m_amp_cb, D->lp_filter_size);
I = convolve (D->m_amp_cb, D->lp_filter, D->lp_filter_size);
float a = my_atan2f(I,Q);
push_sample (sam_x_sin, D->s_amp_cb, D->lp_filter_size);
Q = convolve (D->s_amp_cb, D->lp_filter, D->lp_filter_size);
// This is just a delay line of one symbol time.
a = my_atan2f(I,Q);
push_sample (a, D->ms_in_cb, D->ms_filter_size);
delta = a - D->ms_in_cb[D->boffs];
/* 256 units/cycle makes modulo processing easier. */
/* Make sure it is positive before truncating to integer. */
id = ((int)((delta / (2.f * (float)M_PI) + 1.f) * 256.f)) & 0xff;
push_sample (a, D->u.psk.delay_line, D->u.psk.delay_line_taps);
float delta = a - D->u.psk.delay_line[D->u.psk.boffs];
int gray;
int bit_quality[3];
if (D->modem_type == MODEM_QPSK) {
#ifdef TUNE_PSKOFFSET
demod_phase_shift = ((id + TUNE_PSKOFFSET) >> 6) & 0x3;
#else
if (D->v26_alt == V26_B) {
demod_phase_shift = ((id + 2) >> 6) & 0x3; // MFJ compatible
if (D->u.psk.v26_alt == V26_B) {
gray = phase_shift_to_symbol (delta + (float)(-M_PI/4), 2, bit_quality);; // MFJ compatible
}
else {
demod_phase_shift = ((id + 32) >> 6) & 0x3; // Classic
gray = phase_shift_to_symbol (delta, 2, bit_quality); // Classic
}
#endif
}
else {
demod_phase_shift = ((id + 16) >> 5) & 0x7;
gray = phase_shift_to_symbol (delta, 3, bit_quality);; // 8-PSK
}
nudge_pll (chan, subchan, slice, demod_phase_shift, D);
nudge_pll (chan, subchan, slice, gray, D, bit_quality);
D->lo_phase += D->lo_step;
D->u.psk.lo_phase += D->u.psk.lo_step;
}
else {
/*
* Correlate with previous symbol. We are looking for the phase shift.
*/
push_sample (fsam, D->ms_in_cb, D->ms_filter_size);
push_sample (fsam, D->u.psk.delay_line, D->u.psk.delay_line_taps);
sam_x_cos = fsam * D->ms_in_cb[D->coffs];
sam_x_sin = fsam * D->ms_in_cb[D->soffs];
float sam_x_cos = fsam * D->u.psk.delay_line[D->u.psk.coffs];
push_sample (sam_x_cos, D->u.psk.I_raw, D->u.psk.lp_filter_taps);
float I = convolve (D->u.psk.I_raw, D->u.psk.lp_filter, D->u.psk.lp_filter_taps);
push_sample (sam_x_cos, D->m_amp_cb, D->lp_filter_size);
I = convolve (D->m_amp_cb, D->lp_filter, D->lp_filter_size);
float sam_x_sin = fsam * D->u.psk.delay_line[D->u.psk.soffs];
push_sample (sam_x_sin, D->u.psk.Q_raw, D->u.psk.lp_filter_taps);
float Q = convolve (D->u.psk.Q_raw, D->u.psk.lp_filter, D->u.psk.lp_filter_taps);
push_sample (sam_x_sin, D->s_amp_cb, D->lp_filter_size);
Q = convolve (D->s_amp_cb, D->lp_filter, D->lp_filter_size);
int gray;
int bit_quality[3];
float delta = my_atan2f(I,Q);
if (D->modem_type == MODEM_QPSK) {
float a = my_atan2f(I,Q);
int id = ((int)((a / (2.f * (float)M_PI) + 1.f) * 256.f)) & 0xff;
// 128 compensates for 180 degree phase shift due
// to 1 1/2 carrier cycles per symbol period.
#ifdef TUNE_PSKOFFSET
demod_phase_shift = ((id + TUNE_PSKOFFSET) >> 6) & 0x3;
#else
if (D->v26_alt == V26_B) {
demod_phase_shift = ((id + 98) >> 6) & 0x3; // MFJ compatible
if (D->u.psk.v26_alt == V26_B) {
gray = phase_shift_to_symbol (delta + (float)(M_PI/2), 2, bit_quality); // MFJ compatible
}
else {
demod_phase_shift = ((id + 128) >> 6) & 0x3; // Classic
gray = phase_shift_to_symbol (delta + (float)(3*M_PI/4), 2, bit_quality); // Classic
}
#endif
}
else {
float a;
int idelta;
a = my_atan2f(I,Q);
idelta = ((int)((a / (2.f * (float)M_PI) + 1.f) * 256.f)) & 0xff;
// 32 (90 degrees) compensates for 1800 carrier vs. 1800 baud.
// 16 is to set threshold between constellation points.
demod_phase_shift = ((idelta - 32 - 16) >> 5) & 0x7;
gray = phase_shift_to_symbol (delta + (float)(3*M_PI/2), 3, bit_quality);
}
nudge_pll (chan, subchan, slice, demod_phase_shift, D);
nudge_pll (chan, subchan, slice, gray, D, bit_quality);
}
#if DEBUG4
if (chan == 0) {
if (1) {
//if (hdlc_rec_gathering (chan, subchan, slice)) {
char fname[30];
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, sin, cos, *cos, *sin, I, Q, phase, Clock\n");
}
fprintf (demod_log_fp, "%.3f, %.3f, %.3f, %.3f, %.3f, %.3f, %.2f, %.2f, %.2f\n",
fsam + 2,
- D->ms_in_cb[D->soffs] + 6,
- D->ms_in_cb[D->coffs] + 6,
sam_x_cos + 8,
sam_x_sin + 10,
2 * I + 12,
2 * Q + 12,
demod_phase_shift * 2. / 3. + 14.,
(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_psk_process_sample */
#ifdef TUNE_GRAY
TUNE_GRAY
#else
static const int phase_to_gray_v26[4] = {0, 1, 3, 2};
#endif
static const int phase_to_gray_v27[8] = {1, 0, 2, 3, 7, 6, 4, 5};
__attribute__((hot))
inline static void nudge_pll (int chan, int subchan, int slice, int demod_bits, struct demodulator_state_s *D)
static void nudge_pll (int chan, int subchan, int slice, int demod_bits, struct demodulator_state_s *D, int *bit_quality)
{
/*
* 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 points to a demodulator for a channel/subchannel pair.
*
* D->data_clock_pll is a SIGNED 32 bit variable.
* When it overflows from a large positive value to a negative value, we
@ -802,12 +774,8 @@ inline static void nudge_pll (int chan, int subchan, int slice, int demod_bits,
* 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.
@ -820,28 +788,17 @@ inline static void nudge_pll (int chan, int subchan, int slice, int demod_bits,
if (D->modem_type == MODEM_QPSK) {
int gray = phase_to_gray_v26[ demod_bits ];
int gray = demod_bits;
#if DEBUG4
text_color_set(DW_COLOR_DEBUG);
dw_printf ("a=%.2f deg, delta=%.2f deg, phaseshift=%d, bits= %d %d \n",
a * 360 / (2*M_PI), delta * 360 / (2*M_PI), demod_bits, (gray >> 1) & 1, gray & 1);
//dw_printf ("phaseshift=%d, bits= %d %d \n", demod_bits, (gray >> 1) & 1, gray & 1);
#endif
#if DEBUG5
dw_printf ("%d\n%d\n", (gray >> 1) & 1, gray & 1);
#endif
hdlc_rec_bit (chan, subchan, slice, (gray >> 1) & 1, 0, -1);
hdlc_rec_bit (chan, subchan, slice, gray & 1, 0, -1);
hdlc_rec_bit (chan, subchan, slice, (gray >> 1) & 1, 0, bit_quality[1]);
hdlc_rec_bit (chan, subchan, slice, gray & 1, 0, bit_quality[0]);
}
else {
int gray = phase_to_gray_v27[ demod_bits ];
int gray = demod_bits;
hdlc_rec_bit (chan, subchan, slice, (gray >> 2) & 1, 0, -1);
hdlc_rec_bit (chan, subchan, slice, (gray >> 1) & 1, 0, -1);
hdlc_rec_bit (chan, subchan, slice, gray & 1, 0, -1);
hdlc_rec_bit (chan, subchan, slice, (gray >> 2) & 1, 0, bit_quality[2]);
hdlc_rec_bit (chan, subchan, slice, (gray >> 1) & 1, 0, bit_quality[1]);
hdlc_rec_bit (chan, subchan, slice, gray & 1, 0, bit_quality[0]);
}
}

124
src/fsk_demod_state.h
View File

@ -28,7 +28,7 @@ struct demodulator_state_s
*/
enum modem_t modem_type; // MODEM_AFSK, MODEM_8PSK, etc.
enum v26_e v26_alt; // Which alternative when V.26.
// enum v26_e v26_alt; // Which alternative when V.26.
char profile; // 'A', 'B', etc. Upper case.
// Only needed to see if we are using 'F' to take fast path.
@ -147,21 +147,6 @@ struct demodulator_state_s
float s_cos_table[MAX_FILTER_SIZE] __attribute__((aligned(16)));
/*
* These are for PSK only.
* They are number of delay line taps into previous symbol.
* They are one symbol period and + or - 45 degrees of the carrier frequency.
*/
int boffs; /* symbol length based on sample rate and baud. */
int coffs; /* to get cos component of previous symbol. */
int soffs; /* to get sin component of previous symbol. */
unsigned int lo_step; /* How much to advance the local oscillator */
/* phase for each audio sample. */
int psk_use_lo; /* Use local oscillator rather than self correlation. */
/*
* The rest are continuously updated.
*/
@ -255,48 +240,99 @@ struct demodulator_state_s
} slicer [MAX_SLICERS]; // Actual number in use is num_slicers.
// Should be in range 1 .. MAX_SLICERS,
/*
* Special for Rino decoder only.
* One for each possible signal polarity.
* The project showed promise but fell by the wayside.
* Version 1.6:
*
* This has become quite disorganized and messy with different combinations of
* fields used for different demodulator types. Start to reorganize it into a common
* part (with things like the DPLL for clock recovery), and separate sections
* for each of the demodulator types.
* Still a lot to do here.
*/
#if 0
union {
struct gr_state_s {
//////////////////////////////////////////////////////////////////////////////////
// //
// PSK only. //
// //
//////////////////////////////////////////////////////////////////////////////////
signed int data_clock_pll; // PLL for data clock recovery.
// It is incremented by pll_step_per_sample
// for each audio sample.
signed int prev_d_c_pll; // Previous value of above, before
// incrementing, to detect overflows.
struct psk_only_s {
float gr_minus_peak; // For automatic gain control.
float gr_plus_peak;
enum v26_e v26_alt; // Which alternative when V.26.
int gr_sync; // Is sync pulse present?
int gr_prev_sync; // Previous state to detect leading edge.
float sin_table256[256]; // Precomputed sin table for speed.
int gr_first_sample; // Index of starting sample index for debugging.
int gr_dcd; // Data carrier detect. i.e. are we
// currently decoding a message.
// Optional band pass pre-filter before phase detector.
float gr_early_sum; // For averaging bit values in two regions.
int gr_early_count;
float gr_late_sum;
int gr_late_count;
float gr_sync_sum;
int gr_sync_count;
// TODO? put back into common section?
// TODO? Why was I thinking that?
int gr_bit_count; // Bit index into message.
int use_prefilter; // True to enable it.
struct rpack_s rpack; // Collection of bits.
float prefilter_baud; // Cutoff frequencies, as fraction of baud rate, beyond tones used.
// In the case of PSK, we use only a single tone of 1800 Hz.
// If we were using 2400 bps (= 1200 baud), this would be
// the fraction of 1200 for the cutoff below and above 1800.
} gr_state[2];
#endif
float pre_filter_width_sym; /* Length in number of symbol times. */
int pre_filter_taps; /* Size of pre filter, in audio samples. */
bp_window_t pre_window;
float audio_in[MAX_FILTER_SIZE] __attribute__((aligned(16)));
float pre_filter[MAX_FILTER_SIZE] __attribute__((aligned(16)));
// Use local oscillator or correlate with previous sample.
int psk_use_lo; /* Use local oscillator rather than self correlation. */
unsigned int lo_step; /* How much to advance the local oscillator */
/* phase for each audio sample. */
unsigned int lo_phase; /* Local oscillator phase accumulator for PSK. */
// After mixing with LO before low pass filter.
float I_raw[MAX_FILTER_SIZE] __attribute__((aligned(16))); // signal * LO cos.
float Q_raw[MAX_FILTER_SIZE] __attribute__((aligned(16))); // signal * LO sin.
// Number of delay line taps into previous symbol.
// They are one symbol period and + or - 45 degrees of the carrier frequency.
int boffs; /* symbol length based on sample rate and baud. */
int coffs; /* to get cos component of previous symbol. */
int soffs; /* to get sin component of previous symbol. */
float delay_line_width_sym;
int delay_line_taps; // In audio samples.
float delay_line[MAX_FILTER_SIZE] __attribute__((aligned(16)));
// Low pass filter Second is frequency as ratio to baud rate for FIR.
// TODO? put back into common section?
// TODO? What are the tradeoffs?
float lpf_baud; /* Cutoff frequency as fraction of baud. */
/* Intuitively we'd expect this to be somewhere */
/* in the range of 0.5 to 1. */
float lp_filter_width_sym; /* Length in number of symbol times. */
int lp_filter_taps; /* Size of Low Pass filter, in audio samples (i.e. filter taps). */
bp_window_t lp_window;
float lp_filter[MAX_FILTER_SIZE] __attribute__((aligned(16)));
} psk;
} u; // end of union for different demodulator types.
};

4
test/scripts/check-modem2400-b
View File

@ -1,5 +1,5 @@
@CUSTOM_SHELL_SHABANG@
@GEN_PACKETS_BIN@ -B2400 -J -n 100 -o test24-b.wav
@ATEST_BIN@ -B2400 -J -F0 -L79 -G83 test24-b.wav
@ATEST_BIN@ -B2400 -J -F1 -L87 -G91 test24-b.wav
@ATEST_BIN@ -B2400 -J -F0 -L84 -G88 test24-b.wav
@ATEST_BIN@ -B2400 -J -F1 -L86 -G90 test24-b.wav