//
// 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.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 <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 "direwolf.h"
#include "audio.h"
#include "demod.h"
#include "tune.h"
#include "fsk_demod_state.h"
#include "fsk_gen_filter.h"
#include "fsk_fast_filter.h"
#include "hdlc_rec.h"
#include "textcolor.h"
#include "demod_9600.h"
#include "demod_afsk.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];
#define UPSAMPLE 2
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 j;
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->achan[chan].valid) {
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).
* New interleaved decoders.
*
* 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:
/*
* 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 == strlen(just_letters));
/*
* Pick a good default demodulator if none specified.
*/
if (num_letters == 0) {
if (save_audio_config_p->achan[chan].baud < 600) {
/* This has been optimized for 300 baud. */
strlcpy (just_letters, "D", sizeof(just_letters));
}
else {
#if __arm__
/* We probably don't have a lot of CPU power available. */
/* Previously we would use F if possible otherwise fall back to A. */
/* In version 1.2, new default is E+ /3. */
strlcpy (just_letters, "E", sizeof(just_letters)); // version 1.2 now E.
if (have_plus != -1) have_plus = 1; // Add as default for version 1.2
// If not explicitly turned off.
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;
}
}
#else
strlcpy (just_letters, "E", sizeof(just_letters)); // version 1.2 changed C to E.
if (have_plus != -1) have_plus = 1; // Add as default for version 1.2
// If not explicitly turned off.
#endif
}
num_letters = 1;
}
assert (num_letters == strlen(just_letters));
/*
* Put it back together again.
*/
/* 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, 'D') != 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;
/*
* Quick hack with special case for another experiment.
* Do this in a more general way if it turns out to be useful.
*/
save_audio_config_p->achan[chan].interleave = 1;
if (strcasecmp(save_audio_config_p->achan[chan].profiles, "EE") == 0) {
save_audio_config_p->achan[chan].interleave = 2;
save_audio_config_p->achan[chan].decimate = 1;
}
else if (strcasecmp(save_audio_config_p->achan[chan].profiles, "EEE") == 0) {
save_audio_config_p->achan[chan].interleave = 3;
save_audio_config_p->achan[chan].decimate = 1;
}
else if (strcasecmp(save_audio_config_p->achan[chan].profiles, "EEEE") == 0) {
save_audio_config_p->achan[chan].interleave = 4;
save_audio_config_p->achan[chan].decimate = 1;
}
else if (strcasecmp(save_audio_config_p->achan[chan].profiles, "EEEEE") == 0) {
save_audio_config_p->achan[chan].interleave = 5;
save_audio_config_p->achan[chan].decimate = 1;
}
else if (strcasecmp(save_audio_config_p->achan[chan].profiles, "GG") == 0) {
save_audio_config_p->achan[chan].interleave = 2;
save_audio_config_p->achan[chan].decimate = 1;
}
else if (strcasecmp(save_audio_config_p->achan[chan].profiles, "GGG") == 0) {
save_audio_config_p->achan[chan].interleave = 3;
save_audio_config_p->achan[chan].decimate = 1;
}
else if (strcasecmp(save_audio_config_p->achan[chan].profiles, "GGG+") == 0) {
save_audio_config_p->achan[chan].interleave = 3;
save_audio_config_p->achan[chan].decimate = 1;
}
else if (strcasecmp(save_audio_config_p->achan[chan].profiles, "GGGG") == 0) {
save_audio_config_p->achan[chan].interleave = 4;
save_audio_config_p->achan[chan].decimate = 1;
}
else if (strcasecmp(save_audio_config_p->achan[chan].profiles, "GGGGG") == 0) {
save_audio_config_p->achan[chan].interleave = 5;
save_audio_config_p->achan[chan].decimate = 1;
}
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].interleave),
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 siginal level reporting, we want a longer term view. */
// TODO: Should probably move this into the init functions.
D->quick_attack = D->agc_fast_attack * 0.2;
D->sluggish_decay = D->agc_slow_decay * 0.2;
}
}
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 siginal level reporting, we want a longer term view. */
D->quick_attack = D->agc_fast_attack * 0.2;
D->sluggish_decay = D->agc_slow_decay * 0.2;
}
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 siginal level reporting, we want a longer term view. */
D->quick_attack = D->agc_fast_attack * 0.2;
D->sluggish_decay = D->agc_slow_decay * 0.2;
} /* for each freq pair */
}
break;
//TODO: how about MODEM_OFF case?
case MODEM_BASEBAND:
case MODEM_SCRAMBLE:
default: /* Not AFSK */
{
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. */
#ifndef __arm__
strlcpy (save_audio_config_p->achan[chan].profiles, "+", sizeof(save_audio_config_p->achan[chan].profiles));
#endif
}
text_color_set(DW_COLOR_DEBUG);
dw_printf ("Channel %d: %d baud, K9NG/G3RUH, %s, %d sample rate x %d",
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, 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;
}
demod_9600_init (UPSAMPLE * save_audio_config_p->adev[ACHAN2ADEV(chan)].samples_per_sec, 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 siginal level reporting, we want a longer term view. */
D->quick_attack = D->agc_fast_attack * 0.2;
D->sluggish_decay = D->agc_slow_decay * 0.2;
}
break;
} /* switch on modulation type. */
} /* if channel number is valid */
} /* for 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 btyes 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 extention. */
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;
//float abs_fsam;
int k;
#if DEBUG4
static FILE *demod_log_fp = NULL;
static int seq = 0; /* for log file name */
#endif
//int j;
//int demod_data;
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:
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;
sample_count[chan][subchan] = 0;
}
}
else {
demod_afsk_process_sample (chan, subchan, sam, D);
}
break;
case MODEM_BASEBAND:
case MODEM_SCRAMBLE:
default:
#define ZEROSTUFF 1
#if ZEROSTUFF
/* Literature says this is better if followed */
/* by appropriate low pass filter. */
/* So far, both are same in tests with different */
/* optimal low pass filter parameters. */
for (k=1; k<UPSAMPLE; k++) {
demod_9600_process_sample (chan, 0, D);
}
demod_9600_process_sample (chan, sam*UPSAMPLE, D);
#else
/* Linear interpolation. */
static int prev_sam;
switch (UPSAMPLE) {
case 1:
demod_9600_process_sample (chan, sam);
break;
case 2:
demod_9600_process_sample (chan, (prev_sam + sam) / 2, D);
demod_9600_process_sample (chan, sam, D);
break;
case 3:
demod_9600_process_sample (chan, (2 * prev_sam + sam) / 3, D);
demod_9600_process_sample (chan, (prev_sam + 2 * sam) / 3, D);
demod_9600_process_sample (chan, sam, D);
break;
case 4:
demod_9600_process_sample (chan, (3 * prev_sam + sam) / 4, D);
demod_9600_process_sample (chan, (prev_sam + sam) / 2, D);
demod_9600_process_sample (chan, (prev_sam + 3 * sam) / 4, D);
demod_9600_process_sample (chan, sam, D);
break;
default:
assert (0);
break;
}
prev_sam = sam;
#endif
break;
}
return;
} /* end demod_process_sample */
/* Doesn't seem right. Need to revisit this. */
/* Resulting scale is 0 to almost 100. */
/* Cranking up the input level produces no more than 97 or 98. */
/* We currently produce a message when this goes over 90. */
alevel_t demod_get_audio_level (int chan, int subchan)
{
struct demodulator_state_s *D;
alevel_t alevel;
assert (chan >= 0 && chan < MAX_CHANS);
assert (subchan >= 0 && subchan < MAX_SUBCHANS);
/* We have to consider two different cases here. */
/* N demodulators, each with own slicer and HDLC decoder. */
/* Single demodulator, multiple slicers each with own HDLC decoder. */
if (demodulator_state[chan][0].num_slicers > 1) {
subchan = 0;
}
D = &demodulator_state[chan][subchan];
// Take half of peak-to-peak for received audio level.
alevel.rec = (int) (( D->alevel_rec_peak - D->alevel_rec_valley ) * 50.0f + 0.5f);
if (save_audio_config_p->achan[chan].modem_type == MODEM_AFSK) {
/* For AFSK, we have mark and space amplitudes. */
alevel.mark = (int) ((D->alevel_mark_peak ) * 100.0f + 0.5f);
alevel.space = (int) ((D->alevel_space_peak ) * 100.0f + 0.5f);
//alevel.ms_ratio = D->alevel_mark_peak / D->alevel_space_peak; // TODO: remove after temp test
}
else {
#if 1
/* Display the + and - peaks. */
/* Normally we'd expect them to be about the same. */
/* However, with SDR, or other DC coupling, we could have an offset. */
alevel.mark = (int) ((D->alevel_mark_peak) * 200.0f + 0.5f);
alevel.space = (int) ((D->alevel_space_peak) * 200.0f - 0.5f);
#else
/* Here we have + and - peaks after filtering. */
/* Take half of the peak to peak. */
/* The "5/6" factor worked out right for the current low pass filter. */
/* Will it need to be different if the filter is tweaked? */
alevel.mark = (int) ((D->alevel_mark_peak - D->alevel_space_peak) * 100.0f * 5.0f/6.0f + 0.5f);
alevel.space = -1; /* to print one number inside of ( ) */
#endif
}
return (alevel);
}
/* end demod.c */