//
// This file is part of Dire Wolf, an amateur radio packet TNC.
//
// Copyright (C) 2016 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 PSK demodulator output to log files */
/*------------------------------------------------------------------
*
* 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.
*
*
* 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
* http://www.mfjenterprises.com/Downloads/index.php?productid=MFJ-2400&filename=MFJ-2400.pdf&company=mfj
*
* AEA had a 2400 bps packet modem, PK232-2400.
*
* http://www.repeater-builder.com/aea/pk232/pk232-2400-baud-dpsk-modem.pdf
*
* There was also a Kantronics KPC-2400 that had 2400 bps.
*
* http://www.brazoriacountyares.org/winlink-collection/TNC%20manuals/Kantronics/2400_modem_operators_guide@rgf.pdf
*
*
* The MFJ and AEA both use the EXAR XR-2123 PSK modem chip.
* The Kantronics has a P423 ???
*
* Can't find the chip specs on the EXAR website so Google it.
*
* http://www.komponenten.es.aau.dk/fileadmin/komponenten/Data_Sheet/Linear/XR2123.pdf
*
* The XR-2123 implements the V.26 / Bell 201 standard:
*
* https://www.itu.int/rec/dologin_pub.asp?lang=e&id=T-REC-V.26-198811-I!!PDF-E&type=items
* https://www.itu.int/rec/dologin_pub.asp?lang=e&id=T-REC-V.26bis-198811-I!!PDF-E&type=items
* https://www.itu.int/rec/dologin_pub.asp?lang=e&id=T-REC-V.26ter-198811-I!!PDF-E&type=items
*
* "bis" and "ter" are from Latin for second and third.
* I used the "ter" version which has phase shifts of 0, 90, 180, and 270 degrees.
*
* There are other references to an alternative B which uses other multiples of 45.
* The XR-2123 data sheet mentions only multiples of 90. That's what I went with.
*
* 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
*
*---------------------------------------------------------------*/
#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_psk.h"
#include "dsp.h"
/* 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 = 0.0;
int j;
for (j=0; j<filter_size; j++) {
sum += filter[j] * data[j];
}
return (sum);
}
/* Might replace this with faster, lower precision version someday. */
static inline float my_atan2f (float y, float x)
{
if ( y == 0 && x == 0) return (0.0); // different atan2 implementations behave differently.
return (atan2f(y,x));
}
/*------------------------------------------------------------------
*
* Name: demod_psk_init
*
* Purpose: Initialization for an psk demodulator.
* Select appropriate parameters and set up filters.
*
* Inputs: modem_type - MODEM_QPSK or MODEM_8PSK.
*
* 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.
*
* profile - Select different variations. For QPSK:
*
* P - Using self-correlation technique.
* Q - Same preceded by bandpass filter.
* R - Using local oscillator to derive phase.
* S - Same with bandpass filter.
*
* For 8-PSK:
*
* T, U, V, W same as above.
*
* D - Pointer to demodulator state for given channel.
*
* Outputs: D->ms_filter_size
*
* Returns: None.
*
* Bugs: This doesn't do much error checking so don't give it
* anything crazy.
*
*----------------------------------------------------------------*/
void demod_psk_init (enum modem_t modem_type, int samples_per_sec, int bps, char profile, struct demodulator_state_s *D)
{
int correct_baud; // baud is not same as bits/sec here!
int carrier_freq;
int j;
memset (D, 0, sizeof(struct demodulator_state_s));
D->modem_type = modem_type;
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) {
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
dw_printf ("demod_psk_init QPSK (sample rate=%d, bps=%d, baud=%d, carrier=%d, profile=%c\n",
samples_per_sec, bps, correct_baud, carrier_freq, profile);
#endif
switch (toupper(profile)) {
case 'P': /* Self correlation technique. */
D->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->pll_locked_inertia = 0.95;
D->pll_searching_inertia = 0.50;
break;
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->lpf_baud = 0.60;
D->lp_filter_len_bits = 39. * 1200. / 44100.;
D->lp_window = BP_WINDOW_COSINE;
D->pll_locked_inertia = 0.87;
D->pll_searching_inertia = 0.50;
break;
default:
text_color_set (DW_COLOR_ERROR);
dw_printf ("Invalid demodulator profile %c for v.26 QPSK. Valid choices are P, Q, R, S. Using default.\n", profile);
// fall thru.
case 'R': /* Mix with local oscillator. */
D->psk_use_lo = 1;
D->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->pll_locked_inertia = 0.925;
D->pll_searching_inertia = 0.50;
break;
case 'S': /* Mix with local oscillator. */
D->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->lpf_baud = 0.60;
D->lp_filter_len_bits = 39. * 1200. / 44100.;
D->lp_window = BP_WINDOW_COSINE;
D->pll_locked_inertia = 0.925;
D->pll_searching_inertia = 0.50;
break;
}
D->ms_filter_len_bits = 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 );
}
else {
correct_baud = bps / 3;
carrier_freq = 1800;
#if DEBUG1
dw_printf ("demod_psk_init 8-PSK (sample rate=%d, bps=%d, baud=%d, carrier=%d, profile=%c\n",
samples_per_sec, bps, correct_baud, carrier_freq, profile);
#endif
switch (toupper(profile)) {
case 'T': /* Self correlation technique. */
D->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->pll_locked_inertia = 0.95;
D->pll_searching_inertia = 0.50;
break;
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->lpf_baud = 1.15;
D->lp_filter_len_bits = 32. * 1200. / 44100.;
D->lp_window = BP_WINDOW_COSINE;
D->pll_locked_inertia = 0.87;
D->pll_searching_inertia = 0.50;
break;
default:
text_color_set (DW_COLOR_ERROR);
dw_printf ("Invalid demodulator profile %c for v.27 8PSK. Valid choices are T, U, V, W. Using default.\n", profile);
// fall thru.
case 'V': /* Mix with local oscillator. */
D->psk_use_lo = 1;
D->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->pll_locked_inertia = 0.925;
D->pll_searching_inertia = 0.50;
break;
case 'W': /* Mix with local oscillator. */
D->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->lpf_baud = 0.85;
D->lp_filter_len_bits = 31. * 1200. / 44100.;
D->lp_window = BP_WINDOW_COSINE;
D->pll_locked_inertia = 0.925;
D->pll_searching_inertia = 0.50;
break;
}
D->ms_filter_len_bits = 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 );
}
if (D->psk_use_lo) {
D->lo_step = (int) round( 256. * 256. * 256. * 256. * carrier_freq / (double)samples_per_sec);
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);
}
}
#ifdef TUNE_PRE_BAUD
D->prefilter_baud = TUNE_PRE_BAUD;
#endif
#ifdef TUNE_PRE_WINDOW
D->pre_window = TUNE_PRE_WINDOW;
#endif
#ifdef TUNE_LPF_BAUD
D->lpf_baud = TUNE_LPF_BAUD;
#endif
#ifdef TUNE_LP_WINDOW
D->lp_window = TUNE_LP_WINDOW;
#endif
#ifdef TUNE_HYST
D->hysteresis = TUNE_HYST;
#endif
#if defined(TUNE_PLL_SEARCHING)
D->pll_searching_inertia = TUNE_PLL_SEARCHING;
#endif
#if defined(TUNE_PLL_LOCKED)
D->pll_locked_inertia = TUNE_PLL_LOCKED;
#endif
/*
* Calculate constants used for timing.
* The audio sample rate must be at least a few times the data rate.
*/
D->pll_step_per_sample = (int) round((TICKS_PER_PLL_CYCLE * (double)correct_baud) / ((double)samples_per_sec));
/*
* 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 );
#ifdef TUNE_PRE_FILTER_SIZE
D->pre_filter_size = TUNE_PRE_FILTER_SIZE;
#endif
#ifdef TUNE_LP_FILTER_SIZE
D->lp_filter_size = TUNE_LP_FILTER_SIZE;
#endif
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.
*/
if (D->use_prefilter) {
float f1, f2;
f1 = carrier_freq - D->prefilter_baud * correct_baud;
f2 = carrier_freq + D->prefilter_baud * correct_baud;
#if 0
text_color_set(DW_COLOR_DEBUG);
dw_printf ("Generating prefilter %.0f to %.0f Hz.\n", (double)f1, (double)f2);
#endif
if (f1 <= 0) {
text_color_set (DW_COLOR_ERROR);
dw_printf ("Prefilter of %.0f to %.0f Hz doesn't make sense.\n", (double)f1, (double)f2);
f1 = 10;
}
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);
}
/*
* Now the lowpass filter.
*/
float fc = correct_baud * D->lpf_baud / (float)samples_per_sec;
gen_lowpass (fc, D->lp_filter, D->lp_filter_size, D->lp_window);
/*
* No point in having multiple numbers for signal level.
*/
D->alevel_mark_peak = -1;
D->alevel_space_peak = -1;
} /* demod_psk_init */
/*-------------------------------------------------------------------
*
* Name: demod_psk_process_sample
*
* Purpose: (1) Demodulate the psk signal into I & Q components.
* (2) Recover clock and sample data at the right time.
* (3) Produce two bits per symbol based on phase change from previous.
*
* 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.
*
* Outputs: For each recovered data bit, we call:
*
* hdlc_rec (channel, demodulated_bit);
*
* to decode HDLC frames from the stream of bits.
*
* Returns: None
*
* Descripion: All the literature, that I could find, described mixing
* with a local oscillator. First we multiply the input by
* cos and sin then low pass filter each. This gives us
* correlation to the different phases. The signs of these two
* results produces two data bits per symbol period.
*
* An 1800 Hz local oscillator was derived from the 1200 Hz
* PLL used to sample the data.
* This worked wonderfully for the ideal condition where
* we start off with the proper phase and all the timing
* is perfect. However, when random delays were added
* before the frame, the PLL would lock on only about
* half the time.
*
* Late one night, it dawned on me that there is no
* need for a local oscillator (LO) at the carrier frequency.
* Simply correlate the signal with the previous symbol,
* phase shifted by + and - 45 degrees.
* The code is much simpler and very reliable.
*
* Later, I realized it was not necessary to synchronize the LO
* because we only care about the phase shift between symbols.
*
* This works better under noisy conditions because we are
* including the noise from only the current symbol and not
* the previous one.
*
* Finally, once we know how to distinguish 4 different phases,
* it is not much effort to use 8 phases to double the bit rate.
*
*--------------------------------------------------------------------*/
inline static void nudge_pll (int chan, int subchan, int slice, int demod_bits, struct demodulator_state_s *D);
__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
assert (chan >= 0 && chan < MAX_CHANS);
assert (subchan >= 0 && subchan < MAX_SUBCHANS);
/* Scale to nice number for plotting during debug. */
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->psk_use_lo) {
float a, delta;
int id;
/*
* 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];
sam_x_sin = fsam * D->m_sin_table[(D->lo_phase >> 24) & 0xff];
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);
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);
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;
if (D->modem_type == MODEM_QPSK) {
demod_phase_shift = ((id + 32) >> 6) & 0x3;
}
else {
demod_phase_shift = ((id + 16) >> 5) & 0x7;
}
nudge_pll (chan, subchan, slice, demod_phase_shift, D);
D->lo_phase += D->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);
sam_x_cos = fsam * D->ms_in_cb[D->coffs];
sam_x_sin = fsam * D->ms_in_cb[D->soffs];
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);
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);
if (D->modem_type == MODEM_QPSK) {
#if 1 // Speed up special case.
if (I > 0) {
if (Q > 0)
demod_phase_shift = 0; /* 0 to 90 degrees, etc. */
else
demod_phase_shift = 1;
}
else {
if (Q > 0)
demod_phase_shift = 3;
else
demod_phase_shift = 2;
}
#else
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.
demod_phase_shift = ((id + 128) >> 6) & 0x3;
#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;
}
nudge_pll (chan, subchan, slice, demod_phase_shift, D);
}
#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 */
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};
__attribute__((hot))
inline static void nudge_pll (int chan, int subchan, int slice, int demod_bits, 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;
D->slicer[slice].data_clock_pll += D->pll_step_per_sample;
if (D->slicer[slice].data_clock_pll < 0 && D->slicer[slice].prev_d_c_pll >= 0) {
/* Overflow of PLL counter. */
/* This is where we sample the data. */
if (D->modem_type == MODEM_QPSK) {
int gray = phase_to_gray_v26[ 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
hdlc_rec_bit (chan, subchan, slice, (gray >> 1) & 1, 0, -1);
hdlc_rec_bit (chan, subchan, slice, gray & 1, 0, -1);
}
else {
int gray = phase_to_gray_v27[ 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);
}
}
/*
* If demodulated data has changed,
* Pull the PLL phase closer to zero.
* Use "floor" instead of simply casting so the sign won't flip.
* For example if we had -0.7 we want to end up with -1 rather than 0.
*/
// TODO: demod_9600 has an improved technique. Would it help us here?
if (demod_bits != D->slicer[slice].prev_demod_data) {
if (hdlc_rec_gathering (chan, subchan, slice)) {
D->slicer[slice].data_clock_pll = (int)floorf((float)(D->slicer[slice].data_clock_pll) * D->pll_locked_inertia);
}
else {
D->slicer[slice].data_clock_pll = (int)floorf((float)(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_bits;
} /* end nudge_pll */
/* end demod_psk.c */