//
// This file is part of Dire Wolf, an amateur radio packet TNC.
//
// Copyright (C) 2011, 2012, 2013, 2014, 2015, 2016, 2019 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/>.
//
/*-------------------------------------------------------------------
*
* Name: atest.c
*
* Purpose: Test fixture for the AFSK demodulator.
*
* Inputs: Takes audio from a .WAV file insted of the audio device.
*
* Description: This can be used to test the AFSK demodulator under
* controlled and reproducable conditions for tweaking.
*
* For example
*
* (1) Download WA8LMF's TNC Test CD image file from
* http://wa8lmf.net/TNCtest/index.htm
*
* (2) Burn a physical CD.
*
* (3) "Rip" the desired tracks with Windows Media Player.
* Select .WAV file format.
*
* "Track 2" is used for most tests because that is more
* realistic for most people using the speaker output.
*
*
* Without ONE_CHAN defined:
*
* Notice that the number of packets decoded, as reported by
* this test program, will be twice the number expected because
* we are decoding the left and right audio channels separately.
*
*
* With ONE_CHAN defined:
*
* Only process one channel.
*
*--------------------------------------------------------------------*/
// #define X 1
#include "direwolf.h"
#include <stdio.h>
#include <unistd.h>
#include <stdlib.h>
#include <assert.h>
#include <string.h>
#include <time.h>
#include <getopt.h>
#define ATEST_C 1
#include "audio.h"
#include "demod.h"
#include "multi_modem.h"
#include "textcolor.h"
#include "ax25_pad.h"
#include "hdlc_rec2.h"
#include "dlq.h"
#include "ptt.h"
#include "dtime_now.h"
#if 0 /* Typical but not flexible enough. */
struct wav_header { /* .WAV file header. */
char riff[4]; /* "RIFF" */
int filesize; /* file length - 8 */
char wave[4]; /* "WAVE" */
char fmt[4]; /* "fmt " */
int fmtsize; /* 16. */
short wformattag; /* 1 for PCM. */
short nchannels; /* 1 for mono, 2 for stereo. */
int nsamplespersec; /* sampling freq, Hz. */
int navgbytespersec; /* = nblockalign*nsamplespersec. */
short nblockalign; /* = wbitspersample/8 * nchannels. */
short wbitspersample; /* 16 or 8. */
char data[4]; /* "data" */
int datasize; /* number of bytes following. */
} ;
#endif
/* 8 bit samples are unsigned bytes */
/* in range of 0 .. 255. */
/* 16 bit samples are signed short */
/* in range of -32768 .. +32767. */
static struct {
char riff[4]; /* "RIFF" */
int filesize; /* file length - 8 */
char wave[4]; /* "WAVE" */
} header;
static struct {
char id[4]; /* "LIST" or "fmt " */
int datasize;
} chunk;
static struct {
short wformattag; /* 1 for PCM. */
short nchannels; /* 1 for mono, 2 for stereo. */
int nsamplespersec; /* sampling freq, Hz. */
int navgbytespersec; /* = nblockalign*nsamplespersec. */
short nblockalign; /* = wbitspersample/8 * nchannels. */
short wbitspersample; /* 16 or 8. */
char extras[4];
} format;
static struct {
char data[4]; /* "data" */
int datasize;
} wav_data;
static FILE *fp;
static int e_o_f;
static int packets_decoded_one = 0;
static int packets_decoded_total = 0;
static int decimate = 0; /* Reduce that sampling rate if set. */
/* 1 = normal, 2 = half, 3 = 1/3, etc. */
static int upsample = 0; /* Upsample for G3RUH decoder. */
/* Non-zero will override the default. */
static struct audio_s my_audio_config;
static int error_if_less_than = -1; /* Exit with error status if this minimum not reached. */
/* Can be used to check that performance has not decreased. */
static int error_if_greater_than = -1; /* Exit with error status if this maximum exceeded. */
/* Can be used to check that duplicate removal is not broken. */
//#define EXPERIMENT_G 1
//#define EXPERIMENT_H 1
#if defined(EXPERIMENT_G) || defined(EXPERIMENT_H)
static int count[MAX_SUBCHANS];
#if EXPERIMENT_H
extern float space_gain[MAX_SUBCHANS];
#endif
#endif
static void usage (void);
static int decode_only = 0; /* Set to 0 or 1 to decode only one channel. 2 for both. */
static int sample_number = -1; /* Sample number from the file. */
/* Incremented only for channel 0. */
/* Use to print timestamp, relative to beginning */
/* of file, when frame was decoded. */
// command line options.
static int B_opt = DEFAULT_BAUD; // Bits per second. Need to change all baud references to bps.
static int g_opt = 0; // G3RUH modem regardless of speed.
static int j_opt = 0; /* 2400 bps PSK compatible with direwolf <= 1.5 */
static int J_opt = 0; /* 2400 bps PSK compatible MFJ-2400 and maybe others. */
static int h_opt = 0; // Hexadecimal display of received packet.
static char P_opt[16] = ""; // Demodulator profiles.
int main (int argc, char *argv[])
{
int err;
int c;
int channel;
double start_time; // Time when we started so we can measure elapsed time.
double one_filetime = 0; // Length of one audio file in seconds.
double total_filetime = 0; // Length of all audio files in seconds.
double elapsed; // Time it took us to process it.
#if defined(EXPERIMENT_G) || defined(EXPERIMENT_H)
int j;
for (j=0; j<MAX_SUBCHANS; j++) {
count[j] = 0;
}
#endif
text_color_init(1);
text_color_set(DW_COLOR_INFO);
/*
* First apply defaults.
*/
memset (&my_audio_config, 0, sizeof(my_audio_config));
my_audio_config.adev[0].num_channels = DEFAULT_NUM_CHANNELS;
my_audio_config.adev[0].samples_per_sec = DEFAULT_SAMPLES_PER_SEC;
my_audio_config.adev[0].bits_per_sample = DEFAULT_BITS_PER_SAMPLE;
for (channel=0; channel<MAX_CHANS; channel++) {
my_audio_config.achan[channel].modem_type = MODEM_AFSK;
my_audio_config.achan[channel].mark_freq = DEFAULT_MARK_FREQ;
my_audio_config.achan[channel].space_freq = DEFAULT_SPACE_FREQ;
my_audio_config.achan[channel].baud = DEFAULT_BAUD;
strlcpy (my_audio_config.achan[channel].profiles, "E", sizeof(my_audio_config.achan[channel].profiles));
my_audio_config.achan[channel].num_freq = 1;
my_audio_config.achan[channel].offset = 0;
my_audio_config.achan[channel].fix_bits = RETRY_NONE;
my_audio_config.achan[channel].sanity_test = SANITY_APRS;
//my_audio_config.achan[channel].sanity_test = SANITY_AX25;
//my_audio_config.achan[channel].sanity_test = SANITY_NONE;
my_audio_config.achan[channel].passall = 0;
//my_audio_config.achan[channel].passall = 1;
}
while (1) {
//int this_option_optind = optind ? optind : 1;
int option_index = 0;
static struct option long_options[] = {
{"future1", 1, 0, 0},
{"future2", 0, 0, 0},
{"future3", 1, 0, 'c'},
{0, 0, 0, 0}
};
/* ':' following option character means arg is required. */
c = getopt_long(argc, argv, "B:P:D:U:gjJF:L:G:012h",
long_options, &option_index);
if (c == -1)
break;
switch (c) {
case 'B': /* -B for data Bit rate */
/* Also implies modem type based on speed. */
B_opt = atoi(optarg);
break;
case 'g': /* -G Force G3RUH regardless of speed. */
g_opt = 1;
break;
case 'j': /* -j V.26 compatible with earlier direwolf. */
j_opt = 1;
break;
case 'J': /* -J V.26 compatible with MFJ-2400. */
J_opt = 1;
break;
case 'P': /* -P for modem profile. */
// Wait until after other options processed.
strlcpy (P_opt, optarg, sizeof(P_opt));
break;
case 'D': /* -D reduce sampling rate for lower CPU usage. */
decimate = atoi(optarg);
dw_printf ("Divide audio sample rate by %d\n", decimate);
if (decimate < 1 || decimate > 8) {
text_color_set(DW_COLOR_ERROR);
dw_printf ("Unreasonable value for -D.\n");
exit (EXIT_FAILURE);
}
dw_printf ("Divide audio sample rate by %d\n", decimate);
my_audio_config.achan[0].decimate = decimate;
break;
case 'U': /* -U upsample for G3RUH to improve performance */
/* when the sample rate to baud ratio is low. */
/* Actually it is set automatically and this will */
/* override the normal calculation. */
upsample = atoi(optarg);
dw_printf ("Multiply audio sample rate by %d\n", upsample);
if (upsample < 1 || upsample > 4) {
text_color_set(DW_COLOR_ERROR);
dw_printf ("Unreasonable value for -U.\n");
exit (EXIT_FAILURE);
}
dw_printf ("Multiply audio sample rate by %d\n", upsample);
my_audio_config.achan[0].upsample = upsample;
break;
case 'F': /* -F set "fix bits" level. */
my_audio_config.achan[0].fix_bits = atoi(optarg);
if (my_audio_config.achan[0].fix_bits < RETRY_NONE || my_audio_config.achan[0].fix_bits >= RETRY_MAX) {
text_color_set(DW_COLOR_ERROR);
dw_printf ("Invalid Fix Bits level.\n");
exit (EXIT_FAILURE);
}
break;
case 'L': /* -L error if less than this number decoded. */
error_if_less_than = atoi(optarg);
break;
case 'G': /* -G error if greater than this number decoded. */
error_if_greater_than = atoi(optarg);
break;
case '0': /* channel 0, left from stereo */
decode_only = 0;
break;
case '1': /* channel 1, right from stereo */
decode_only = 1;
break;
case '2': /* decode both from stereo */
decode_only = 2;
break;
case 'h': /* Hexadecimal display. */
h_opt = 1;
break;
case '?':
/* Unknown option message was already printed. */
usage ();
break;
default:
/* Should not be here. */
text_color_set(DW_COLOR_ERROR);
dw_printf("?? getopt returned character code 0%o ??\n", c);
usage ();
}
}
/*
* Set modem type based on data rate.
* (Could be overridden by -g, -j, or -J later.)
*/
/* 300 implies 1600/1800 AFSK. */
/* 1200 implies 1200/2200 AFSK. */
/* 2400 implies V.26 QPSK. */
/* 4800 implies V.27 8PSK. */
/* 9600 implies G3RUH baseband scrambled. */
my_audio_config.achan[0].baud = B_opt;
if (my_audio_config.achan[0].baud < MIN_BAUD || my_audio_config.achan[0].baud > MAX_BAUD) {
text_color_set(DW_COLOR_ERROR);
dw_printf ("Use a more reasonable bit rate in range of %d - %d.\n", MIN_BAUD, MAX_BAUD);
exit (EXIT_FAILURE);
}
/* We have similar logic in direwolf.c, config.c, gen_packets.c, and atest.c, */
/* that need to be kept in sync. Maybe it could be a common function someday. */
if (my_audio_config.achan[0].baud == 100) {
my_audio_config.achan[0].modem_type = MODEM_AFSK;
my_audio_config.achan[0].mark_freq = 1615;
my_audio_config.achan[0].space_freq = 1785;
strlcpy (my_audio_config.achan[0].profiles, "D", sizeof(my_audio_config.achan[0].profiles));
}
else if (my_audio_config.achan[0].baud < 600) {
my_audio_config.achan[0].modem_type = MODEM_AFSK;
my_audio_config.achan[0].mark_freq = 1600;
my_audio_config.achan[0].space_freq = 1800;
strlcpy (my_audio_config.achan[0].profiles, "D", sizeof(my_audio_config.achan[0].profiles));
}
else if (my_audio_config.achan[0].baud < 1800) {
my_audio_config.achan[0].modem_type = MODEM_AFSK;
my_audio_config.achan[0].mark_freq = DEFAULT_MARK_FREQ;
my_audio_config.achan[0].space_freq = DEFAULT_SPACE_FREQ;
// Should default to E+ or something similar later.
}
else if (my_audio_config.achan[0].baud < 3600) {
my_audio_config.achan[0].modem_type = MODEM_QPSK;
my_audio_config.achan[0].mark_freq = 0;
my_audio_config.achan[0].space_freq = 0;
strlcpy (my_audio_config.achan[0].profiles, "", sizeof(my_audio_config.achan[0].profiles));
}
else if (my_audio_config.achan[0].baud < 7200) {
my_audio_config.achan[0].modem_type = MODEM_8PSK;
my_audio_config.achan[0].mark_freq = 0;
my_audio_config.achan[0].space_freq = 0;
strlcpy (my_audio_config.achan[0].profiles, "", sizeof(my_audio_config.achan[0].profiles));
}
else {
my_audio_config.achan[0].modem_type = MODEM_SCRAMBLE;
my_audio_config.achan[0].mark_freq = 0;
my_audio_config.achan[0].space_freq = 0;
strlcpy (my_audio_config.achan[0].profiles, " ", sizeof(my_audio_config.achan[0].profiles)); // avoid getting default later.
}
/*
* -g option means force g3RUH regardless of speed.
*/
if (g_opt) {
my_audio_config.achan[0].modem_type = MODEM_SCRAMBLE;
my_audio_config.achan[0].mark_freq = 0;
my_audio_config.achan[0].space_freq = 0;
strlcpy (my_audio_config.achan[0].profiles, " ", sizeof(my_audio_config.achan[0].profiles)); // avoid getting default later.
}
/*
* We have two different incompatible flavors of V.26.
*/
if (j_opt) {
// V.26 compatible with earlier versions of direwolf.
// Example: -B 2400 -j or simply -j
my_audio_config.achan[0].v26_alternative = V26_A;
my_audio_config.achan[0].modem_type = MODEM_QPSK;
my_audio_config.achan[0].mark_freq = 0;
my_audio_config.achan[0].space_freq = 0;
my_audio_config.achan[0].baud = 2400;
strlcpy (my_audio_config.achan[0].profiles, "", sizeof(my_audio_config.achan[0].profiles));
}
if (J_opt) {
// V.26 compatible with MFJ and maybe others.
// Example: -B 2400 -J or simply -J
my_audio_config.achan[0].v26_alternative = V26_B;
my_audio_config.achan[0].modem_type = MODEM_QPSK;
my_audio_config.achan[0].mark_freq = 0;
my_audio_config.achan[0].space_freq = 0;
my_audio_config.achan[0].baud = 2400;
strlcpy (my_audio_config.achan[0].profiles, "", sizeof(my_audio_config.achan[0].profiles));
}
// Needs to be after -B, -j, -J.
if (strlen(P_opt) > 0) {
dw_printf ("Demodulator profile set to \"%s\"\n", P_opt);
strlcpy (my_audio_config.achan[0].profiles, P_opt, sizeof(my_audio_config.achan[0].profiles));
}
memcpy (&my_audio_config.achan[1], &my_audio_config.achan[0], sizeof(my_audio_config.achan[0]));
if (optind >= argc) {
text_color_set(DW_COLOR_ERROR);
dw_printf ("Specify .WAV file name on command line.\n");
usage ();
}
start_time = dtime_now();
while (optind < argc) {
fp = fopen(argv[optind], "rb");
if (fp == NULL) {
text_color_set(DW_COLOR_ERROR);
dw_printf ("Couldn't open file for read: %s\n", argv[optind]);
//perror ("more info?");
exit (EXIT_FAILURE);
}
/*
* Read the file header.
* Doesn't handle all possible cases but good enough for our purposes.
*/
err= fread (&header, (size_t)12, (size_t)1, fp);
(void)(err);
if (strncmp(header.riff, "RIFF", 4) != 0 || strncmp(header.wave, "WAVE", 4) != 0) {
text_color_set(DW_COLOR_ERROR);
dw_printf ("This is not a .WAV format file.\n");
exit (EXIT_FAILURE);
}
err = fread (&chunk, (size_t)8, (size_t)1, fp);
if (strncmp(chunk.id, "LIST", 4) == 0) {
err = fseek (fp, (long)chunk.datasize, SEEK_CUR);
err = fread (&chunk, (size_t)8, (size_t)1, fp);
}
if (strncmp(chunk.id, "fmt ", 4) != 0) {
text_color_set(DW_COLOR_ERROR);
dw_printf ("WAV file error: Found \"%4.4s\" where \"fmt \" was expected.\n", chunk.id);
exit(EXIT_FAILURE);
}
if (chunk.datasize != 16 && chunk.datasize != 18) {
text_color_set(DW_COLOR_ERROR);
dw_printf ("WAV file error: Need fmt chunk datasize of 16 or 18. Found %d.\n", chunk.datasize);
exit(EXIT_FAILURE);
}
err = fread (&format, (size_t)chunk.datasize, (size_t)1, fp);
err = fread (&wav_data, (size_t)8, (size_t)1, fp);
if (strncmp(wav_data.data, "data", 4) != 0) {
text_color_set(DW_COLOR_ERROR);
dw_printf ("WAV file error: Found \"%4.4s\" where \"data\" was expected.\n", wav_data.data);
exit(EXIT_FAILURE);
}
if (format.wformattag != 1) {
text_color_set(DW_COLOR_ERROR);
dw_printf ("Sorry, I only understand audio format 1 (PCM). This file has %d.\n", format.wformattag);
exit (EXIT_FAILURE);
}
if (format.nchannels != 1 && format.nchannels != 2) {
text_color_set(DW_COLOR_ERROR);
dw_printf ("Sorry, I only understand 1 or 2 channels. This file has %d.\n", format.nchannels);
exit (EXIT_FAILURE);
}
if (format.wbitspersample != 8 && format.wbitspersample != 16) {
text_color_set(DW_COLOR_ERROR);
dw_printf ("Sorry, I only understand 8 or 16 bits per sample. This file has %d.\n", format.wbitspersample);
exit (EXIT_FAILURE);
}
my_audio_config.adev[0].samples_per_sec = format.nsamplespersec;
my_audio_config.adev[0].bits_per_sample = format.wbitspersample;
my_audio_config.adev[0].num_channels = format.nchannels;
my_audio_config.achan[0].medium = MEDIUM_RADIO;
if (format.nchannels == 2) {
my_audio_config.achan[1].medium = MEDIUM_RADIO;
}
text_color_set(DW_COLOR_INFO);
dw_printf ("%d samples per second. %d bits per sample. %d audio channels.\n",
my_audio_config.adev[0].samples_per_sec,
my_audio_config.adev[0].bits_per_sample,
my_audio_config.adev[0].num_channels);
one_filetime = (double) wav_data.datasize /
((my_audio_config.adev[0].bits_per_sample / 8) * my_audio_config.adev[0].num_channels * my_audio_config.adev[0].samples_per_sec);
total_filetime += one_filetime;
dw_printf ("%d audio bytes in file. Duration = %.1f seconds.\n",
(int)(wav_data.datasize),
one_filetime);
dw_printf ("Fix Bits level = %d\n", my_audio_config.achan[0].fix_bits);
/*
* Initialize the AFSK demodulator and HDLC decoder.
*/
multi_modem_init (&my_audio_config);
packets_decoded_one = 0;
e_o_f = 0;
while ( ! e_o_f)
{
int audio_sample;
int c;
for (c=0; c<my_audio_config.adev[0].num_channels; c++)
{
/* This reads either 1 or 2 bytes depending on */
/* bits per sample. */
audio_sample = demod_get_sample (ACHAN2ADEV(c));
if (audio_sample >= 256 * 256) {
e_o_f = 1;
continue;
}
if (c == 0) sample_number++;
if (decode_only == 0 && c != 0) continue;
if (decode_only == 1 && c != 1) continue;
multi_modem_process_sample(c,audio_sample);
}
/* When a complete frame is accumulated, */
/* process_rec_frame, below, is called. */
}
text_color_set(DW_COLOR_INFO);
dw_printf ("\n\n");
#if EXPERIMENT_G
for (j=0; j<MAX_SUBCHANS; j++) {
float db = 20.0 * log10f(space_gain[j]);
dw_printf ("%+.1f dB, %d\n", db, count[j]);
}
#endif
#if EXPERIMENT_H
for (j=0; j<MAX_SUBCHANS; j++) {
dw_printf ("%d\n", count[j]);
}
#endif
dw_printf ("%d from %s\n", packets_decoded_one, argv[optind]);
packets_decoded_total += packets_decoded_one;
fclose (fp);
optind++;
}
elapsed = dtime_now() - start_time;
dw_printf ("%d packets decoded in %.3f seconds. %.1f x realtime\n", packets_decoded_total, elapsed, total_filetime/elapsed);
if (error_if_less_than != -1 && packets_decoded_total < error_if_less_than) {
text_color_set(DW_COLOR_ERROR);
dw_printf ("\n * * * TEST FAILED: number decoded is less than %d * * * \n", error_if_less_than);
exit (EXIT_FAILURE);
}
if (error_if_greater_than != -1 && packets_decoded_total > error_if_greater_than) {
text_color_set(DW_COLOR_ERROR);
dw_printf ("\n * * * TEST FAILED: number decoded is greater than %d * * * \n", error_if_greater_than);
exit (EXIT_FAILURE);
}
exit (EXIT_SUCCESS);
}
/*
* Simulate sample from the audio device.
*/
int audio_get (int a)
{
int ch;
if (wav_data.datasize <= 0) {
e_o_f = 1;
return (-1);
}
ch = getc(fp);
wav_data.datasize--;
if (ch < 0) {
text_color_set(DW_COLOR_ERROR);
dw_printf ("Unexpected end of file.\n");
e_o_f = 1;
}
return (ch);
}
/*
* Rather than queuing up frames with bad FCS,
* try to fix them immediately.
*/
void rdq_append (rrbb_t rrbb)
{
int chan, subchan, slice;
alevel_t alevel;
chan = rrbb_get_chan(rrbb);
subchan = rrbb_get_subchan(rrbb);
slice = rrbb_get_slice(rrbb);
alevel = rrbb_get_audio_level(rrbb);
hdlc_rec2_try_to_fix_later (rrbb, chan, subchan, slice, alevel);
rrbb_delete (rrbb);
}
/*
* This is called when we have a good frame.
*/
void dlq_rec_frame (int chan, int subchan, int slice, packet_t pp, alevel_t alevel, retry_t retries, char *spectrum)
{
char stemp[500];
unsigned char *pinfo;
int info_len;
int h;
char heard[AX25_MAX_ADDR_LEN];
char alevel_text[AX25_ALEVEL_TO_TEXT_SIZE];
packets_decoded_one++;
ax25_format_addrs (pp, stemp);
info_len = ax25_get_info (pp, &pinfo);
/* Print so we can see what is going on. */
//TODO: quiet option - suppress packet printing, only the count at the end.
#if 1
/* Display audio input level. */
/* Who are we hearing? Original station or digipeater? */
if (ax25_get_num_addr(pp) == 0) {
/* Not AX.25. No station to display below. */
h = -1;
strlcpy (heard, "", sizeof(heard));
}
else {
h = ax25_get_heard(pp);
ax25_get_addr_with_ssid(pp, h, heard);
}
text_color_set(DW_COLOR_DEBUG);
dw_printf ("\n");
dw_printf("DECODED[%d] ", packets_decoded_one );
/* Insert time stamp relative to start of file. */
double sec = (double)sample_number / my_audio_config.adev[0].samples_per_sec;
int min = (int)(sec / 60.);
sec -= min * 60;
dw_printf ("%d:%07.4f ", min, sec);
if (h != AX25_SOURCE) {
dw_printf ("Digipeater ");
}
ax25_alevel_to_text (alevel, alevel_text);
if (my_audio_config.achan[chan].fix_bits == RETRY_NONE && my_audio_config.achan[chan].passall == 0) {
dw_printf ("%s audio level = %s %s\n", heard, alevel_text, spectrum);
}
else {
dw_printf ("%s audio level = %s [%s] %s\n", heard, alevel_text, retry_text[(int)retries], spectrum);
}
#endif
//#if defined(EXPERIMENT_G) || defined(EXPERIMENT_H)
// int j;
//
// for (j=0; j<MAX_SUBCHANS; j++) {
// if (spectrum[j] == '|') {
// count[j]++;
// }
// }
//#endif
// Display non-APRS packets in a different color.
// Display channel with subchannel/slice if applicable.
if (ax25_is_aprs(pp)) {
text_color_set(DW_COLOR_REC);
}
else {
text_color_set(DW_COLOR_DEBUG);
}
if (my_audio_config.achan[chan].num_subchan > 1 && my_audio_config.achan[chan].num_slicers == 1) {
dw_printf ("[%d.%d] ", chan, subchan);
}
else if (my_audio_config.achan[chan].num_subchan == 1 && my_audio_config.achan[chan].num_slicers > 1) {
dw_printf ("[%d.%d] ", chan, slice);
}
else if (my_audio_config.achan[chan].num_subchan > 1 && my_audio_config.achan[chan].num_slicers > 1) {
dw_printf ("[%d.%d.%d] ", chan, subchan, slice);
}
else {
dw_printf ("[%d] ", chan);
}
dw_printf ("%s", stemp); /* stations followed by : */
ax25_safe_print ((char *)pinfo, info_len, 0);
dw_printf ("\n");
/*
* -h option for hexadecimal display. (new in 1.6)
*/
if (h_opt) {
text_color_set(DW_COLOR_DEBUG);
dw_printf ("------\n");
ax25_hex_dump (pp);
dw_printf ("------\n");
}
#if 1 // temp experiment TODO: remove this.
#include "decode_aprs.h"
#include "log.h"
if (ax25_is_aprs(pp)) {
decode_aprs_t A;
decode_aprs (&A, pp, 0);
// Temp experiment to see how different systems set the RR bits in the source and destination.
// log_rr_bits (&A, pp);
}
#endif
ax25_delete (pp);
} /* end fake dlq_append */
void ptt_set (int ot, int chan, int ptt_signal)
{
return;
}
int get_input (int it, int chan)
{
return -1;
}
static void usage (void) {
text_color_set(DW_COLOR_ERROR);
dw_printf ("\n");
dw_printf ("atest is a test application which decodes AX.25 frames from an audio\n");
dw_printf ("recording. This provides an easy way to test Dire Wolf decoding\n");
dw_printf ("performance much quicker than normal real-time. \n");
dw_printf ("\n");
dw_printf ("usage:\n");
dw_printf ("\n");
dw_printf (" atest [ options ] wav-file-in\n");
dw_printf ("\n");
dw_printf (" -B n Bits/second for data. Proper modem automatically selected for speed.\n");
dw_printf (" 300 bps defaults to AFSK tones of 1600 & 1800.\n");
dw_printf (" 1200 bps uses AFSK tones of 1200 & 2200.\n");
dw_printf (" 2400 bps uses QPSK based on V.26 standard.\n");
dw_printf (" 4800 bps uses 8PSK based on V.27 standard.\n");
dw_printf (" 9600 bps and up uses K9NG/G3RUH standard.\n");
dw_printf ("\n");
dw_printf (" -g Use G3RUH modem rather rather than default for data rate.\n");
dw_printf (" -j 2400 bps QPSK compatible with direwolf <= 1.5.\n");
dw_printf (" -J 2400 bps QPSK compatible with MFJ-2400.\n");
dw_printf ("\n");
dw_printf (" -D n Divide audio sample rate by n.\n");
dw_printf ("\n");
dw_printf (" -h Print frame contents as hexadecimal bytes.\n");
dw_printf ("\n");
dw_printf (" -F n Amount of effort to try fixing frames with an invalid CRC. \n");
dw_printf (" 0 (default) = consider only correct frames. \n");
dw_printf (" 1 = Try to fix only a single bit. \n");
dw_printf (" more = Try modifying more bits to get a good CRC.\n");
dw_printf ("\n");
dw_printf (" -L Error if less than this number decoded.\n");
dw_printf ("\n");
dw_printf (" -G Error if greater than this number decoded.\n");
dw_printf ("\n");
dw_printf (" -P m Select the demodulator type such as D (default for 300 bps),\n");
dw_printf (" E+ (default for 1200 bps), PQRS for 2400 bps, etc.\n");
dw_printf ("\n");
dw_printf (" -0 Use channel 0 (left) of stereo audio (default).\n");
dw_printf (" -1 use channel 1 (right) of stereo audio.\n");
dw_printf (" -2 decode both channels of stereo audio.\n");
dw_printf ("\n");
dw_printf (" wav-file-in is a WAV format audio file.\n");
dw_printf ("\n");
dw_printf ("Examples:\n");
dw_printf ("\n");
dw_printf (" gen_packets -o test1.wav\n");
dw_printf (" atest test1.wav\n");
dw_printf ("\n");
dw_printf (" gen_packets -B 300 -o test3.wav\n");
dw_printf (" atest -B 300 test3.wav\n");
dw_printf ("\n");
dw_printf (" gen_packets -B 9600 -o test9.wav\n");
dw_printf (" atest -B 9600 test9.wav\n");
dw_printf ("\n");
dw_printf (" This generates and decodes 3 test files with 1200, 300, and 9600\n");
dw_printf (" bits per second.\n");
dw_printf ("\n");
dw_printf (" atest 02_Track_2.wav\n");
dw_printf (" atest -P E- 02_Track_2.wav\n");
dw_printf (" atest -F 1 02_Track_2.wav\n");
dw_printf (" atest -P E- -F 1 02_Track_2.wav\n");
dw_printf ("\n");
dw_printf (" Try different combinations of options to compare decoding\n");
dw_printf (" performance.\n");
exit (1);
}
/* end atest.c */