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demod.c
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//
// This file is part of Dire Wolf, an amateur radio packet TNC.
//
// Copyright (C) 2011, 2012, 2013, 2014, 2015, 2016, 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/>.
//
/*------------------------------------------------------------------
*
* Module: demod.c
*
* Purpose: Common entry point for multiple types of demodulators.
*
* 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 "demod.h"
#include "tune.h"
#include "fsk_demod_state.h"
#include "fsk_gen_filter.h"
#include "hdlc_rec.h"
#include "textcolor.h"
#include "demod_9600.h"
#include "demod_afsk.h"
#include "demod_psk.h"
// Properties of the radio channels.
static struct audio_s *save_audio_config_p;
// Current state of all the decoders.
static struct demodulator_state_s demodulator_state[MAX_CHANS][MAX_SUBCHANS];
static int sample_sum[MAX_CHANS][MAX_SUBCHANS];
static int sample_count[MAX_CHANS][MAX_SUBCHANS];
/*------------------------------------------------------------------
*
* Name: demod_init
*
* Purpose: Initialize the demodulator(s) used for reception.
*
* Inputs: pa - Pointer to audio_s structure with
* various parameters for the modem(s).
*
* Returns: 0 for success, -1 for failure.
*
*
* Bugs: This doesn't do much error checking so don't give it
* anything crazy.
*
*----------------------------------------------------------------*/
int demod_init (struct audio_s *pa)
{
int chan; /* Loop index over number of radio channels. */
char profile;
/*
* Save audio configuration for later use.
*/
save_audio_config_p = pa;
for (chan = 0; chan < MAX_CHANS; chan++) {
if (save_audio_config_p->chan_medium[chan] == MEDIUM_RADIO) {
char *p;
char just_letters[16];
int num_letters;
int have_plus;
/*
* These are derived from config file parameters.
*
* num_subchan is number of demodulators.
* This can be increased by:
* Multiple frequencies.
* Multiple letters (not sure if I will continue this).
*
* num_slicers is set to max by the "+" option.
*/
save_audio_config_p->achan[chan].num_subchan = 1;
save_audio_config_p->achan[chan].num_slicers = 1;
switch (save_audio_config_p->achan[chan].modem_type) {
case MODEM_OFF:
break;
case MODEM_AFSK:
case MODEM_EAS:
if (save_audio_config_p->achan[chan].modem_type == MODEM_EAS) {
if (save_audio_config_p->achan[chan].fix_bits != RETRY_NONE) {
text_color_set(DW_COLOR_INFO);
dw_printf ("Channel %d: FIX_BITS option has been turned off for EAS.\n", chan);
save_audio_config_p->achan[chan].fix_bits = RETRY_NONE;
}
if (save_audio_config_p->achan[chan].passall != 0) {
text_color_set(DW_COLOR_INFO);
dw_printf ("Channel %d: PASSALL option has been turned off for EAS.\n", chan);
save_audio_config_p->achan[chan].passall = 0;
}
}
/*
* Tear apart the profile and put it back together in a normalized form:
* - At least one letter, supply suitable default if necessary.
* - Upper case only.
* - Any plus will be at the end.
*/
num_letters = 0;
just_letters[num_letters] = '\0';
have_plus = 0;
for (p = save_audio_config_p->achan[chan].profiles; *p != '\0'; p++) {
if (islower(*p)) {
just_letters[num_letters] = toupper(*p);
num_letters++;
just_letters[num_letters] = '\0';
}
else if (isupper(*p)) {
just_letters[num_letters] = *p;
num_letters++;
just_letters[num_letters] = '\0';
}
else if (*p == '+') {
have_plus = 1;
if (p[1] != '\0') {
text_color_set(DW_COLOR_ERROR);
dw_printf ("Channel %d: + option must appear at end of demodulator types \"%s\" \n",
chan, save_audio_config_p->achan[chan].profiles);
}
}
else if (*p == '-') {
have_plus = -1;
if (p[1] != '\0') {
text_color_set(DW_COLOR_ERROR);
dw_printf ("Channel %d: - option must appear at end of demodulator types \"%s\" \n",
chan, save_audio_config_p->achan[chan].profiles);
}
} else {
text_color_set(DW_COLOR_ERROR);
dw_printf ("Channel %d: Demodulator types \"%s\" can contain only letters and + - characters.\n",
chan, save_audio_config_p->achan[chan].profiles);
}
}
assert (num_letters == (int)(strlen(just_letters)));
/*
* Pick a good default demodulator if none specified.
* Previously, we had "D" optimized for 300 bps.
* Gone in 1.7 so it is always "A+".
*/
if (num_letters == 0) {
strlcpy (just_letters, "A", sizeof(just_letters));
num_letters = strlen(just_letters);
if (have_plus != -1) have_plus = 1; // Add as default for version 1.2
// If not explicitly turned off.
}
/*
* Special case for ARM.
* The higher end ARM chips have loads of power but many people
* are using a single core Pi Zero or similar.
* (I'm still using a model 1 for my digipeater/IGate!)
* Decreasing CPU requirement has a negligible impact on decoding performance.
*
* atest -PA- 01_Track_1.wav --> 1002 packets decoded.
* atest -PA- -D3 01_Track_1.wav --> 997 packets decoded.
*
* Someone concerned about 1/2 of one percent difference can add "-D 1"
*/
#if __arm__
if (save_audio_config_p->achan[chan].decimate == 0) {
if (save_audio_config_p->adev[ACHAN2ADEV(chan)].samples_per_sec > 40000) {
save_audio_config_p->achan[chan].decimate = 3;
}
}
#endif
/*
* Number of filter taps is proportional to number of audio samples in a "symbol" duration.
* These can get extremely large for low speeds, e.g. 300 baud.
* In this case, increase the decimation ration. Crude approximation. Could be improved.
*/
if (save_audio_config_p->achan[chan].decimate == 0 &&
save_audio_config_p->adev[ACHAN2ADEV(chan)].samples_per_sec > 40000 &&
save_audio_config_p->achan[chan].baud < 600) {
// Avoid enormous number of filter taps.
save_audio_config_p->achan[chan].decimate = 3;
}
/*
* Put it back together again.
*/
assert (num_letters == (int)(strlen(just_letters)));
/* At this point, have_plus can have 3 values: */
/* 1 = turned on, either explicitly or by applied default */
/* -1 = explicitly turned off. change to 0 here so it is false. */
/* 0 = off by default. */
if (have_plus == -1) have_plus = 0;
strlcpy (save_audio_config_p->achan[chan].profiles, just_letters, sizeof(save_audio_config_p->achan[chan].profiles));
assert (strlen(save_audio_config_p->achan[chan].profiles) >= 1);
if (have_plus) {
strlcat (save_audio_config_p->achan[chan].profiles, "+", sizeof(save_audio_config_p->achan[chan].profiles));
}
/* These can be increased later for the multi-frequency case. */
save_audio_config_p->achan[chan].num_subchan = num_letters;
save_audio_config_p->achan[chan].num_slicers = 1;
/*
* Some error checking - Can use only one of these:
*
* - Multiple letters.
* - New + multi-slicer.
* - Multiple frequencies.
*/
if (have_plus && save_audio_config_p->achan[chan].num_freq > 1) {
text_color_set(DW_COLOR_ERROR);
dw_printf ("Channel %d: Demodulator + option can't be combined with multiple frequencies.\n", chan);
save_audio_config_p->achan[chan].num_subchan = 1; // Will be set higher later.
save_audio_config_p->achan[chan].num_freq = 1;
}
if (num_letters > 1 && save_audio_config_p->achan[chan].num_freq > 1) {
text_color_set(DW_COLOR_ERROR);
dw_printf ("Channel %d: Multiple demodulator types can't be combined with multiple frequencies.\n", chan);
save_audio_config_p->achan[chan].profiles[1] = '\0';
num_letters = 1;
}
if (save_audio_config_p->achan[chan].decimate == 0) {
save_audio_config_p->achan[chan].decimate = 1;
if (strchr (just_letters, 'B') != NULL && save_audio_config_p->adev[ACHAN2ADEV(chan)].samples_per_sec > 40000) {
save_audio_config_p->achan[chan].decimate = 3;
}
}
text_color_set(DW_COLOR_DEBUG);
dw_printf ("Channel %d: %d baud, AFSK %d & %d Hz, %s, %d sample rate",
chan, save_audio_config_p->achan[chan].baud,
save_audio_config_p->achan[chan].mark_freq, save_audio_config_p->achan[chan].space_freq,
save_audio_config_p->achan[chan].profiles,
save_audio_config_p->adev[ACHAN2ADEV(chan)].samples_per_sec);
if (save_audio_config_p->achan[chan].decimate != 1)
dw_printf (" / %d", save_audio_config_p->achan[chan].decimate);
if (save_audio_config_p->achan[chan].dtmf_decode != DTMF_DECODE_OFF)
dw_printf (", DTMF decoder enabled");
dw_printf (".\n");
/*
* Initialize the demodulator(s).
*
* We have 3 cases to consider.
*/
// TODO1.3: revisit this logic now that it is less restrictive.
if (num_letters > 1) {
int d;
/*
* Multiple letters, usually for 1200 baud.
* Each one corresponds to a demodulator and subchannel.
*
* An interesting experiment but probably not too useful.
* Can't have multiple frequency pairs.
* In version 1.3 this can be combined with the + option.
*/
save_audio_config_p->achan[chan].num_subchan = num_letters;
if (save_audio_config_p->achan[chan].num_subchan != num_letters) {
text_color_set(DW_COLOR_ERROR);
dw_printf ("INTERNAL ERROR, %s:%d, chan=%d, num_subchan(%d) != strlen(\"%s\")\n",
__FILE__, __LINE__, chan, save_audio_config_p->achan[chan].num_subchan, save_audio_config_p->achan[chan].profiles);
}
if (save_audio_config_p->achan[chan].num_freq != 1) {
text_color_set(DW_COLOR_ERROR);
dw_printf ("INTERNAL ERROR, %s:%d, chan=%d, num_freq(%d) != 1\n",
__FILE__, __LINE__, chan, save_audio_config_p->achan[chan].num_freq);
}
for (d = 0; d < save_audio_config_p->achan[chan].num_subchan; d++) {
int mark, space;
assert (d >= 0 && d < MAX_SUBCHANS);
struct demodulator_state_s *D;
D = &demodulator_state[chan][d];
profile = save_audio_config_p->achan[chan].profiles[d];
mark = save_audio_config_p->achan[chan].mark_freq;
space = save_audio_config_p->achan[chan].space_freq;
if (save_audio_config_p->achan[chan].num_subchan != 1) {
text_color_set(DW_COLOR_DEBUG);
dw_printf (" %d.%d: %c %d & %d\n", chan, d, profile, mark, space);
}
demod_afsk_init (save_audio_config_p->adev[ACHAN2ADEV(chan)].samples_per_sec / save_audio_config_p->achan[chan].decimate,
save_audio_config_p->achan[chan].baud,
mark,
space,
profile,
D);
if (have_plus) {
/* I'm not happy about putting this hack here. */
/* should pass in as a parameter rather than adding on later. */
save_audio_config_p->achan[chan].num_slicers = MAX_SLICERS;
D->num_slicers = MAX_SLICERS;
}
/* For signal level reporting, we want a longer term view. */
// TODO: Should probably move this into the init functions.
D->quick_attack = D->agc_fast_attack * 0.2f;
D->sluggish_decay = D->agc_slow_decay * 0.2f;
}
}
else if (have_plus) {
/*
* PLUS - which (formerly) implies we have only one letter and one frequency pair.
*
* One demodulator feeds multiple slicers, each a subchannel.
*/
if (num_letters != 1) {
text_color_set(DW_COLOR_ERROR);
dw_printf ("INTERNAL ERROR, %s:%d, chan=%d, strlen(\"%s\") != 1\n",
__FILE__, __LINE__, chan, just_letters);
}
if (save_audio_config_p->achan[chan].num_freq != 1) {
text_color_set(DW_COLOR_ERROR);
dw_printf ("INTERNAL ERROR, %s:%d, chan=%d, num_freq(%d) != 1\n",
__FILE__, __LINE__, chan, save_audio_config_p->achan[chan].num_freq);
}
if (save_audio_config_p->achan[chan].num_freq != save_audio_config_p->achan[chan].num_subchan) {
text_color_set(DW_COLOR_ERROR);
dw_printf ("INTERNAL ERROR, %s:%d, chan=%d, num_freq(%d) != num_subchan(%d)\n",
__FILE__, __LINE__, chan, save_audio_config_p->achan[chan].num_freq, save_audio_config_p->achan[chan].num_subchan);
}
struct demodulator_state_s *D;
D = &demodulator_state[chan][0];
/* I'm not happy about putting this hack here. */
/* This belongs in demod_afsk_init but it doesn't have access to the audio config. */
save_audio_config_p->achan[chan].num_slicers = MAX_SLICERS;
demod_afsk_init (save_audio_config_p->adev[ACHAN2ADEV(chan)].samples_per_sec / save_audio_config_p->achan[chan].decimate,
save_audio_config_p->achan[chan].baud,
save_audio_config_p->achan[chan].mark_freq,
save_audio_config_p->achan[chan].space_freq,
save_audio_config_p->achan[chan].profiles[0],
D);
if (have_plus) {
/* I'm not happy about putting this hack here. */
/* should pass in as a parameter rather than adding on later. */
save_audio_config_p->achan[chan].num_slicers = MAX_SLICERS;
D->num_slicers = MAX_SLICERS;
}
/* For signal level reporting, we want a longer term view. */
D->quick_attack = D->agc_fast_attack * 0.2f;
D->sluggish_decay = D->agc_slow_decay * 0.2f;
}
else {
int d;
/*
* One letter.
* Can be combined with multiple frequencies.
*/
if (num_letters != 1) {
text_color_set(DW_COLOR_ERROR);
dw_printf ("INTERNAL ERROR, %s:%d, chan=%d, strlen(\"%s\") != 1\n",
__FILE__, __LINE__, chan, save_audio_config_p->achan[chan].profiles);
}
save_audio_config_p->achan[chan].num_subchan = save_audio_config_p->achan[chan].num_freq;
for (d = 0; d < save_audio_config_p->achan[chan].num_freq; d++) {
int mark, space, k;
assert (d >= 0 && d < MAX_SUBCHANS);
struct demodulator_state_s *D;
D = &demodulator_state[chan][d];
profile = save_audio_config_p->achan[chan].profiles[0];
k = d * save_audio_config_p->achan[chan].offset - ((save_audio_config_p->achan[chan].num_freq - 1) * save_audio_config_p->achan[chan].offset) / 2;
mark = save_audio_config_p->achan[chan].mark_freq + k;
space = save_audio_config_p->achan[chan].space_freq + k;
if (save_audio_config_p->achan[chan].num_freq != 1) {
text_color_set(DW_COLOR_DEBUG);
dw_printf (" %d.%d: %c %d & %d\n", chan, d, profile, mark, space);
}
demod_afsk_init (save_audio_config_p->adev[ACHAN2ADEV(chan)].samples_per_sec / save_audio_config_p->achan[chan].decimate,
save_audio_config_p->achan[chan].baud,
mark, space,
profile,
D);
if (have_plus) {
/* I'm not happy about putting this hack here. */
/* should pass in as a parameter rather than adding on later. */
save_audio_config_p->achan[chan].num_slicers = MAX_SLICERS;
D->num_slicers = MAX_SLICERS;
}
/* For signal level reporting, we want a longer term view. */
D->quick_attack = D->agc_fast_attack * 0.2f;
D->sluggish_decay = D->agc_slow_decay * 0.2f;
} /* for each freq pair */
}
break;
case MODEM_QPSK: // New for 1.4
// In versions 1.4 and 1.5, V.26 "Alternative A" was used.
// years later, I discover that the MFJ-2400 used "Alternative B."
// It looks like the other two manufacturers use the same but we
// can't be sure until we find one for compatibility testing.
// In version 1.6 we add a choice for the user.
// If neither one was explicitly specified, print a message and take
// a default. My current thinking is that we default to direwolf <= 1.5
// compatible for version 1.6 and MFJ compatible after that.
if (save_audio_config_p->achan[chan].v26_alternative == V26_UNSPECIFIED) {
text_color_set(DW_COLOR_ERROR);
dw_printf ("Two incompatible versions of 2400 bps QPSK are now available.\n");
dw_printf ("For compatibility with direwolf <= 1.5, use 'V26A' modem option in config file.\n");
dw_printf ("For compatibility MFJ-2400 use 'V26B' modem option in config file.\n");
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 is now MFJ-2400 compatibility mode.\n");
save_audio_config_p->achan[chan].v26_alternative = V26_DEFAULT;
}
// TODO: See how much CPU this takes on ARM and decide if we should have different defaults.
if (strlen(save_audio_config_p->achan[chan].profiles) == 0) {
//#if __arm__
// strlcpy (save_audio_config_p->achan[chan].profiles, "R", sizeof(save_audio_config_p->achan[chan].profiles));
//#else
strlcpy (save_audio_config_p->achan[chan].profiles, "PQRS", sizeof(save_audio_config_p->achan[chan].profiles));
//#endif
}
save_audio_config_p->achan[chan].num_subchan = strlen(save_audio_config_p->achan[chan].profiles);
save_audio_config_p->achan[chan].decimate = 1; // think about this later.
text_color_set(DW_COLOR_DEBUG);
dw_printf ("Channel %d: %d bps, QPSK, %s, %d sample rate",
chan, save_audio_config_p->achan[chan].baud,
save_audio_config_p->achan[chan].profiles,
save_audio_config_p->adev[ACHAN2ADEV(chan)].samples_per_sec);
if (save_audio_config_p->achan[chan].decimate != 1)
dw_printf (" / %d", save_audio_config_p->achan[chan].decimate);
if (save_audio_config_p->achan[chan].v26_alternative == V26_B)
dw_printf (", compatible with MFJ-2400");
else
dw_printf (", compatible with earlier direwolf");
if (save_audio_config_p->achan[chan].dtmf_decode != DTMF_DECODE_OFF)
dw_printf (", DTMF decoder enabled");
dw_printf (".\n");
int d;
for (d = 0; d < save_audio_config_p->achan[chan].num_subchan; d++) {
assert (d >= 0 && d < MAX_SUBCHANS);
struct demodulator_state_s *D;
D = &demodulator_state[chan][d];
profile = save_audio_config_p->achan[chan].profiles[d];
//text_color_set(DW_COLOR_DEBUG);
//dw_printf ("About to call demod_psk_init for Q-PSK case, modem_type=%d, profile='%c'\n",
// save_audio_config_p->achan[chan].modem_type, profile);
demod_psk_init (save_audio_config_p->achan[chan].modem_type,
save_audio_config_p->achan[chan].v26_alternative,
save_audio_config_p->adev[ACHAN2ADEV(chan)].samples_per_sec / save_audio_config_p->achan[chan].decimate,
save_audio_config_p->achan[chan].baud,
profile,
D);
//text_color_set(DW_COLOR_DEBUG);
//dw_printf ("Returned from demod_psk_init\n");
/* For signal level reporting, we want a longer term view. */
/* Guesses based on 9600. Maybe revisit someday. */
D->quick_attack = 0.080 * 0.2;
D->sluggish_decay = 0.00012 * 0.2;
}
break;
case MODEM_8PSK: // New for 1.4
// TODO: See how much CPU this takes on ARM and decide if we should have different defaults.
if (strlen(save_audio_config_p->achan[chan].profiles) == 0) {
//#if __arm__
// strlcpy (save_audio_config_p->achan[chan].profiles, "V", sizeof(save_audio_config_p->achan[chan].profiles));
//#else
strlcpy (save_audio_config_p->achan[chan].profiles, "TUVW", sizeof(save_audio_config_p->achan[chan].profiles));
//#endif
}
save_audio_config_p->achan[chan].num_subchan = strlen(save_audio_config_p->achan[chan].profiles);
save_audio_config_p->achan[chan].decimate = 1; // think about this later
text_color_set(DW_COLOR_DEBUG);
dw_printf ("Channel %d: %d bps, 8PSK, %s, %d sample rate",
chan, save_audio_config_p->achan[chan].baud,
save_audio_config_p->achan[chan].profiles,
save_audio_config_p->adev[ACHAN2ADEV(chan)].samples_per_sec);
if (save_audio_config_p->achan[chan].decimate != 1)
dw_printf (" / %d", save_audio_config_p->achan[chan].decimate);
if (save_audio_config_p->achan[chan].dtmf_decode != DTMF_DECODE_OFF)
dw_printf (", DTMF decoder enabled");
dw_printf (".\n");
//int d;
for (d = 0; d < save_audio_config_p->achan[chan].num_subchan; d++) {
assert (d >= 0 && d < MAX_SUBCHANS);
struct demodulator_state_s *D;
D = &demodulator_state[chan][d];
profile = save_audio_config_p->achan[chan].profiles[d];
//text_color_set(DW_COLOR_DEBUG);
//dw_printf ("About to call demod_psk_init for 8-PSK case, modem_type=%d, profile='%c'\n",
// save_audio_config_p->achan[chan].modem_type, profile);
demod_psk_init (save_audio_config_p->achan[chan].modem_type,
save_audio_config_p->achan[chan].v26_alternative,
save_audio_config_p->adev[ACHAN2ADEV(chan)].samples_per_sec / save_audio_config_p->achan[chan].decimate,
save_audio_config_p->achan[chan].baud,
profile,
D);
//text_color_set(DW_COLOR_DEBUG);
//dw_printf ("Returned from demod_psk_init\n");
/* For signal level reporting, we want a longer term view. */
/* Guesses based on 9600. Maybe revisit someday. */
D->quick_attack = 0.080 * 0.2;
D->sluggish_decay = 0.00012 * 0.2;
}
break;
//TODO: how about MODEM_OFF case?
case MODEM_BASEBAND:
case MODEM_SCRAMBLE:
case MODEM_AIS:
default: /* Not AFSK */
{
// For AIS we will accept only a good CRC without any fixup attempts.
// Even with that, there are still a lot of CRC false matches with random noise.
if (save_audio_config_p->achan[chan].modem_type == MODEM_AIS) {
if (save_audio_config_p->achan[chan].fix_bits != RETRY_NONE) {
text_color_set(DW_COLOR_INFO);
dw_printf ("Channel %d: FIX_BITS option has been turned off for AIS.\n", chan);
save_audio_config_p->achan[chan].fix_bits = RETRY_NONE;
}
if (save_audio_config_p->achan[chan].passall != 0) {
text_color_set(DW_COLOR_INFO);
dw_printf ("Channel %d: PASSALL option has been turned off for AIS.\n", chan);
save_audio_config_p->achan[chan].passall = 0;
}
}
if (strcmp(save_audio_config_p->achan[chan].profiles, "") == 0) {
/* Apply default if not set earlier. */
/* Not sure if it should be on for ARM too. */
/* Need to take a look at CPU usage and performance difference. */
/* Version 1.5: Remove special case for ARM. */
/* We want higher performance to be the default. */
/* "MODEM 9600 -" can be used on very slow CPU if necessary. */
strlcpy (save_audio_config_p->achan[chan].profiles, "+", sizeof(save_audio_config_p->achan[chan].profiles));
}
/*
* We need a minimum number of audio samples per bit time for good performance.
* Easier to check here because demod_9600_init might have an adjusted sample rate.
*/
float ratio = (float)(save_audio_config_p->adev[ACHAN2ADEV(chan)].samples_per_sec)
/ (float)(save_audio_config_p->achan[chan].baud);
/*
* Set reasonable upsample ratio if user did not override.
*/
if (save_audio_config_p->achan[chan].upsample == 0) {
if (ratio < 4) {
// This is extreme.
// No one should be using a sample rate this low but
// amazingly a recording with 22050 rate can be decoded.
// 3 and 4 are the same. Need more tests.
save_audio_config_p->achan[chan].upsample = 4;
}
else if (ratio < 5) {
// example: 44100 / 9600 is 4.59
// 3 is slightly better than 2 or 4.
save_audio_config_p->achan[chan].upsample = 3;
}
else if (ratio < 10) {
// example: 48000 / 9600 = 5
// 3 is slightly better than 2 or 4.
save_audio_config_p->achan[chan].upsample = 3;
}
else if (ratio < 15) {
// ... guessing
save_audio_config_p->achan[chan].upsample = 2;
}
else { // >= 15
//
// An example of this might be .....
// Probably no benefit.
save_audio_config_p->achan[chan].upsample = 1;
}
}
#ifdef TUNE_UPSAMPLE
save_audio_config_p->achan[chan].upsample = TUNE_UPSAMPLE;
#endif
text_color_set(DW_COLOR_DEBUG);
dw_printf ("Channel %d: %d baud, %s, %s, %d sample rate x %d",
chan,
save_audio_config_p->achan[chan].baud,
save_audio_config_p->achan[chan].modem_type == MODEM_AIS ? "AIS" : "K9NG/G3RUH",
save_audio_config_p->achan[chan].profiles,
save_audio_config_p->adev[ACHAN2ADEV(chan)].samples_per_sec,
save_audio_config_p->achan[chan].upsample);
if (save_audio_config_p->achan[chan].dtmf_decode != DTMF_DECODE_OFF)
dw_printf (", DTMF decoder enabled");
dw_printf (".\n");
struct demodulator_state_s *D;
D = &demodulator_state[chan][0]; // first subchannel
save_audio_config_p->achan[chan].num_subchan = 1;
save_audio_config_p->achan[chan].num_slicers = 1;
if (strchr(save_audio_config_p->achan[chan].profiles, '+') != NULL) {
/* I'm not happy about putting this hack here. */
/* This belongs in demod_9600_init but it doesn't have access to the audio config. */
save_audio_config_p->achan[chan].num_slicers = MAX_SLICERS;
}
text_color_set(DW_COLOR_INFO);
dw_printf ("The ratio of audio samples per sec (%d) to data rate in baud (%d) is %.1f\n",
save_audio_config_p->adev[ACHAN2ADEV(chan)].samples_per_sec,
save_audio_config_p->achan[chan].baud,
(double)ratio);
if (ratio < 3) {
text_color_set(DW_COLOR_ERROR);
dw_printf ("There is little hope of success with such a low ratio. Use a higher sample rate.\n");
}
else if (ratio < 5) {
dw_printf ("This is on the low side for best performance. Can you use a higher sample rate?\n");
if (save_audio_config_p->adev[ACHAN2ADEV(chan)].samples_per_sec == 44100) {
dw_printf ("For example, can you use 48000 rather than 44100?\n");
}
}
else if (ratio < 6) {
dw_printf ("Increasing the sample rate should improve decoder performance.\n");
}
else if (ratio > 15) {
dw_printf ("Sample rate is more than adequate. You might lower it if CPU load is a concern.\n");
}
else {
dw_printf ("This is a suitable ratio for good performance.\n");
}
demod_9600_init (save_audio_config_p->achan[chan].modem_type,
save_audio_config_p->adev[ACHAN2ADEV(chan)].samples_per_sec,
save_audio_config_p->achan[chan].upsample,
save_audio_config_p->achan[chan].baud, D);
if (strchr(save_audio_config_p->achan[chan].profiles, '+') != NULL) {
/* I'm not happy about putting this hack here. */
/* should pass in as a parameter rather than adding on later. */
save_audio_config_p->achan[chan].num_slicers = MAX_SLICERS;
D->num_slicers = MAX_SLICERS;
}
/* For signal level reporting, we want a longer term view. */
D->quick_attack = D->agc_fast_attack * 0.2f;
D->sluggish_decay = D->agc_slow_decay * 0.2f;
}
break;
} /* switch on modulation type. */
} /* if channel medium is radio */
// FIXME dw_printf ("-------- end of loop for chn %d \n", chan);
} /* for chan ... */
// Now the virtual channels. FIXME: could be single loop.
for (chan = MAX_CHANS; chan < MAX_TOTAL_CHANS; chan++) {
// FIXME dw_printf ("-------- virtual channel loop %d \n", chan);
if (chan == save_audio_config_p->igate_vchannel) {
text_color_set(DW_COLOR_DEBUG);
dw_printf ("Channel %d: IGate virtual channel.\n", chan);
}
}
return (0);
} /* end demod_init */
/*------------------------------------------------------------------
*
* Name: demod_get_sample
*
* Purpose: Get one audio sample fromt the specified sound input source.
*
* Inputs: a - Index for audio device. 0 = first.
*
* Returns: -32768 .. 32767 for a valid audio sample.
* 256*256 for end of file or other error.
*
* Global In: save_audio_config_p->adev[ACHAN2ADEV(chan)].bits_per_sample - So we know whether to
* read 1 or 2 bytes from audio stream.
*
* Description: Grab 1 or two bytes depending on data source.
*
* When processing stereo, the caller will call this
* at twice the normal rate to obtain alternating left
* and right samples.
*
*----------------------------------------------------------------*/
#define FSK_READ_ERR (256*256)
__attribute__((hot))
int demod_get_sample (int a)
{
int x1, x2;
signed short sam; /* short to force sign extension. */
assert (save_audio_config_p->adev[a].bits_per_sample == 8 || save_audio_config_p->adev[a].bits_per_sample == 16);
if (save_audio_config_p->adev[a].bits_per_sample == 8) {
x1 = audio_get(a);
if (x1 < 0) return(FSK_READ_ERR);
assert (x1 >= 0 && x1 <= 255);
/* Scale 0..255 into -32k..+32k */
sam = (x1 - 128) * 256;
}
else {
x1 = audio_get(a); /* lower byte first */
if (x1 < 0) return(FSK_READ_ERR);
x2 = audio_get(a);
if (x2 < 0) return(FSK_READ_ERR);
assert (x1 >= 0 && x1 <= 255);
assert (x2 >= 0 && x2 <= 255);
sam = ( x2 << 8 ) | x1;
}
return (sam);
}
/*-------------------------------------------------------------------
*
* Name: demod_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.
*
*--------------------------------------------------------------------*/
__attribute__((hot))
void demod_process_sample (int chan, int subchan, int sam)
{
float fsam;
//int k;
struct demodulator_state_s *D;
assert (chan >= 0 && chan < MAX_CHANS);
assert (subchan >= 0 && subchan < MAX_SUBCHANS);
D = &demodulator_state[chan][subchan];
/* Scale to nice number, actually -2.0 to +2.0 for extra headroom */
fsam = sam / 16384.0f;
/*
* Accumulate measure of the input signal level.
*/
/*
* Version 1.2: Try new approach to capturing the amplitude.
* This is same as the later AGC without the normalization step.
* We want decay to be substantially slower to get a longer
* range idea of the received audio.
*/
if (fsam >= D->alevel_rec_peak) {
D->alevel_rec_peak = fsam * D->quick_attack + D->alevel_rec_peak * (1.0f - D->quick_attack);
}
else {
D->alevel_rec_peak = fsam * D->sluggish_decay + D->alevel_rec_peak * (1.0f - D->sluggish_decay);
}
if (fsam <= D->alevel_rec_valley) {
D->alevel_rec_valley = fsam * D->quick_attack + D->alevel_rec_valley * (1.0f - D->quick_attack);
}
else {
D->alevel_rec_valley = fsam * D->sluggish_decay + D->alevel_rec_valley * (1.0f - D->sluggish_decay);
}
/*
* Select decoder based on modulation type.
*/
switch (save_audio_config_p->achan[chan].modem_type) {
case MODEM_OFF:
// Might have channel only listening to DTMF for APRStt gateway.
// Don't waste CPU time running a demodulator here.
break;
case MODEM_AFSK:
case MODEM_EAS:
if (save_audio_config_p->achan[chan].decimate > 1) {
sample_sum[chan][subchan] += sam;
sample_count[chan][subchan]++;
if (sample_count[chan][subchan] >= save_audio_config_p->achan[chan].decimate) {
demod_afsk_process_sample (chan, subchan, sample_sum[chan][subchan] / save_audio_config_p->achan[chan].decimate, D);
sample_sum[chan][subchan] = 0;