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direwolf/gen_packets.c

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//
// This file is part of Dire Wolf, an amateur radio packet TNC.
//
// Copyright (C) 2011, 2013, 2014, 2015, 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/>.
//
/*------------------------------------------------------------------
*
* Name: gen_packets.c
*
* Purpose: Test program for generating AX.25 frames.
*
* Description: Given messages are converted to audio and written
* to a .WAV type audio file.
*
* Bugs: Most options are implemented for only one audio channel.
*
* Examples: Different speeds:
*
* gen_packets -o z1.wav
* atest z1.wav
*
* gen_packets -B 300 -o z3.wav
* atest -B 300 z3.wav
*
* gen_packets -B 9600 -o z9.wav
* atest -B 300 z9.wav
*
* User-defined content:
*
* echo "WB2OSZ>APDW12:This is a test" | gen_packets -o z.wav -
* atest z.wav
*
* echo "WB2OSZ>APDW12:Test line 1" > z.txt
* echo "WB2OSZ>APDW12:Test line 2" >> z.txt
* echo "WB2OSZ>APDW12:Test line 3" >> z.txt
* gen_packets -o z.wav z.txt
* atest z.wav
*
* With artificial noise added:
*
* gen_packets -n 100 -o z2.wav
* atest z2.wav
*
*
*------------------------------------------------------------------*/
#include "direwolf.h"
#include <stdio.h>
#include <stdlib.h>
#include <getopt.h>
#include <string.h>
#include <assert.h>
#include "audio.h"
#include "ax25_pad.h"
#include "hdlc_send.h"
#include "gen_tone.h"
#include "textcolor.h"
#include "morse.h"
#include "dtmf.h"
/* Own random number generator so we can get */
/* same results on Windows and Linux. */
#define MY_RAND_MAX 0x7fffffff
static int seed = 1;
static int my_rand (void) {
// Perform the calculation as unsigned to avoid signed overflow error.
seed = (int)(((unsigned)seed * 1103515245) + 12345) & MY_RAND_MAX;
return (seed);
}
static void usage (char **argv);
static int audio_file_open (char *fname, struct audio_s *pa);
static int audio_file_close (void);
static int g_add_noise = 0;
static float g_noise_level = 0;
static int g_morse_wpm = 0; /* Send morse code at this speed. */
static struct audio_s modem;
static void send_packet (char *str)
{
packet_t pp;
unsigned char fbuf[AX25_MAX_PACKET_LEN+2];
int flen;
int c;
if (g_morse_wpm > 0) {
// TODO: Why not use the destination field instead of command line option?
morse_send (0, str, g_morse_wpm, 100, 100);
}
else {
pp = ax25_from_text (str, 1);
flen = ax25_pack (pp, fbuf);
for (c=0; c<modem.adev[0].num_channels; c++)
{
#if 1
int samples_per_symbol, n, j;
/* Insert random amount of quiet time, approx. 0 to 10 symbol times, to test */
/* how well the clock recovery PLL can regain lock after random phase shifts. */
if (modem.achan[c].modem_type == MODEM_QPSK) {
samples_per_symbol = modem.adev[0].samples_per_sec / (modem.achan[c].baud / 2);
}
else if (modem.achan[c].modem_type == MODEM_8PSK) {
samples_per_symbol = modem.adev[0].samples_per_sec / (modem.achan[c].baud / 3);
}
else {
samples_per_symbol = modem.adev[0].samples_per_sec / modem.achan[c].baud;
}
// for 1200 baud, 44100/sec, this should be 0 to 360.
n = samples_per_symbol * 10 * (float)my_rand() / (float)MY_RAND_MAX;
//dw_printf ("Random 0-360 = %d\n", n);
for (j=0; j<n; j++) {
gen_tone_put_sample (c, 0, 0);
}
#endif
hdlc_send_flags (c, 8, 0);
hdlc_send_frame (c, fbuf, flen, 0);
hdlc_send_flags (c, 2, 1);
}
ax25_delete (pp);
}
}
int main(int argc, char **argv)
{
int c;
//int digit_optind = 0;
int err;
int packet_count = 0;
int i;
int chan;
int experiment = 0;
/*
* Set up default values for the modem.
*/
memset (&modem, 0, sizeof(modem));
modem.adev[0].defined = 1;
modem.adev[0].num_channels = DEFAULT_NUM_CHANNELS; /* -2 stereo */
modem.adev[0].samples_per_sec = DEFAULT_SAMPLES_PER_SEC; /* -r option */
modem.adev[0].bits_per_sample = DEFAULT_BITS_PER_SAMPLE; /* -8 for 8 instead of 16 bits */
for (chan = 0; chan < MAX_CHANS; chan++) {
modem.achan[chan].modem_type = MODEM_AFSK; /* change with -g */
modem.achan[chan].mark_freq = DEFAULT_MARK_FREQ; /* -m option */
modem.achan[chan].space_freq = DEFAULT_SPACE_FREQ; /* -s option */
modem.achan[chan].baud = DEFAULT_BAUD; /* -b option */
}
modem.achan[0].valid = 1;
/*
* Set up other default values.
*/
int amplitude = 50; /* -a option */
/* 100% is actually half of the digital signal range so */
/* we have some headroom for adding noise, etc. */
int leading_zeros = 12; /* -z option TODO: not implemented, should replace with txdelay frames. */
char output_file[256]; /* -o option */
FILE *input_fp = NULL; /* File or NULL for built-in message */
strlcpy (output_file, "", sizeof(output_file));
/*
* Parse the command line options.
*/
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, "gm:s:a:b:B:r:n:o:z:82M:X",
long_options, &option_index);
if (c == -1)
break;
switch (c) {
case 0: /* possible future use */
text_color_set(DW_COLOR_INFO);
dw_printf("option %s", long_options[option_index].name);
if (optarg) {
dw_printf(" with arg %s", optarg);
}
dw_printf("\n");
break;
case 'b': /* -b for data Bit rate */
modem.achan[0].baud = atoi(optarg);
text_color_set(DW_COLOR_INFO);
dw_printf ("Data rate set to %d bits / second.\n", modem.achan[0].baud);
if (modem.achan[0].baud < MIN_BAUD || modem.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);
}
break;
case 'B': /* -B for data Bit rate */
/* 300 implies 1600/1800 AFSK. */
/* 1200 implies 1200/2200 AFSK. */
/* 9600 implies scrambled. */
/* If you want something else, specify -B first */
/* then anything to override these defaults. */
modem.achan[0].baud = atoi(optarg);
text_color_set(DW_COLOR_INFO);
dw_printf ("Data rate set to %d bits / second.\n", modem.achan[0].baud);
if (modem.achan[0].baud != 100 && (modem.achan[0].baud < MIN_BAUD || modem.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 (modem.achan[0].baud == 100) {
modem.achan[0].modem_type = MODEM_AFSK;
modem.achan[0].mark_freq = 1615;
modem.achan[0].space_freq = 1785;
}
else if (modem.achan[0].baud < 600) {
modem.achan[0].modem_type = MODEM_AFSK;
modem.achan[0].mark_freq = 1600; // Typical for HF SSB
modem.achan[0].space_freq = 1800;
}
else if (modem.achan[0].baud < 1800) {
modem.achan[0].modem_type = MODEM_AFSK;
modem.achan[0].mark_freq = DEFAULT_MARK_FREQ;
modem.achan[0].space_freq = DEFAULT_SPACE_FREQ;
}
else if (modem.achan[0].baud < 3600) {
modem.achan[0].modem_type = MODEM_QPSK;
modem.achan[0].mark_freq = 0;
modem.achan[0].space_freq = 0;
dw_printf ("Using V.26 QPSK rather than AFSK.\n");
if (modem.achan[0].baud != 2400) {
text_color_set(DW_COLOR_ERROR);
dw_printf ("Bit rate should be standard 2400 rather than specified %d.\n", modem.achan[0].baud);
}
}
else if (modem.achan[0].baud < 7200) {
modem.achan[0].modem_type = MODEM_8PSK;
modem.achan[0].mark_freq = 0;
modem.achan[0].space_freq = 0;
dw_printf ("Using V.27 8PSK rather than AFSK.\n");
if (modem.achan[0].baud != 4800) {
text_color_set(DW_COLOR_ERROR);
dw_printf ("Bit rate should be standard 4800 rather than specified %d.\n", modem.achan[0].baud);
}
}
else {
modem.achan[0].modem_type = MODEM_SCRAMBLE;
text_color_set(DW_COLOR_INFO);
dw_printf ("Using scrambled baseband signal rather than AFSK.\n");
}
break;
case 'g': /* -g for g3ruh scrambling */
modem.achan[0].modem_type = MODEM_SCRAMBLE;
text_color_set(DW_COLOR_INFO);
dw_printf ("Using scrambled baseband signal rather than AFSK.\n");
break;
case 'm': /* -m for Mark freq */
modem.achan[0].mark_freq = atoi(optarg);
text_color_set(DW_COLOR_INFO);
dw_printf ("Mark frequency set to %d Hz.\n", modem.achan[0].mark_freq);
if (modem.achan[0].mark_freq < 300 || modem.achan[0].mark_freq > 3000) {
text_color_set(DW_COLOR_ERROR);
dw_printf ("Use a more reasonable value in range of 300 - 3000.\n");
exit (EXIT_FAILURE);
}
break;
case 's': /* -s for Space freq */
modem.achan[0].space_freq = atoi(optarg);
text_color_set(DW_COLOR_INFO);
dw_printf ("Space frequency set to %d Hz.\n", modem.achan[0].space_freq);
if (modem.achan[0].space_freq < 300 || modem.achan[0].space_freq > 3000) {
text_color_set(DW_COLOR_ERROR);
dw_printf ("Use a more reasonable value in range of 300 - 3000.\n");
exit (EXIT_FAILURE);
}
break;
case 'n': /* -n number of packets with increasing noise. */
packet_count = atoi(optarg);
g_add_noise = 1;
break;
case 'a': /* -a for amplitude */
amplitude = atoi(optarg);
text_color_set(DW_COLOR_INFO);
dw_printf ("Amplitude set to %d%%.\n", amplitude);
if (amplitude < 0 || amplitude > 200) {
text_color_set(DW_COLOR_ERROR);
dw_printf ("Amplitude must be in range of 0 to 200.\n");
exit (EXIT_FAILURE);
}
break;
case 'r': /* -r for audio sample Rate */
modem.adev[0].samples_per_sec = atoi(optarg);
text_color_set(DW_COLOR_INFO);
dw_printf ("Audio sample rate set to %d samples / second.\n", modem.adev[0].samples_per_sec);
if (modem.adev[0].samples_per_sec < MIN_SAMPLES_PER_SEC || modem.adev[0].samples_per_sec > MAX_SAMPLES_PER_SEC) {
text_color_set(DW_COLOR_ERROR);
dw_printf ("Use a more reasonable audio sample rate in range of %d - %d.\n",
MIN_SAMPLES_PER_SEC, MAX_SAMPLES_PER_SEC);
exit (EXIT_FAILURE);
}
break;
case 'z': /* -z leading zeros before frame flag */
leading_zeros = atoi(optarg);
text_color_set(DW_COLOR_INFO);
dw_printf ("Send %d zero bits before frame flag.\n", leading_zeros);
/* The demodulator needs a few for the clock recovery PLL. */
/* We don't want to be here all day either. */
/* We can't translate to time yet because the data bit rate */
/* could be changed later. */
if (leading_zeros < 8 || leading_zeros > 12000) {
text_color_set(DW_COLOR_ERROR);
dw_printf ("Use a more reasonable value.\n");
exit (EXIT_FAILURE);
}
break;
case '8': /* -8 for 8 bit samples */
modem.adev[0].bits_per_sample = 8;
text_color_set(DW_COLOR_INFO);
dw_printf("8 bits per audio sample rather than 16.\n");
break;
case '2': /* -2 for 2 channels of sound */
modem.adev[0].num_channels = 2;
modem.achan[1].valid = 1;
text_color_set(DW_COLOR_INFO);
dw_printf("2 channels of sound rather than 1.\n");
break;
case 'o': /* -o for Output file */
strlcpy (output_file, optarg, sizeof(output_file));
text_color_set(DW_COLOR_INFO);
dw_printf ("Output file set to %s\n", output_file);
break;
case 'M': /* -M for morse code speed */
//TODO: document this.
// Why not base it on the destination field instead?
g_morse_wpm = atoi(optarg);
text_color_set(DW_COLOR_INFO);
dw_printf ("Morse code speed set to %d WPM.\n", g_morse_wpm);
if (g_morse_wpm < 5 || g_morse_wpm > 50) {
text_color_set(DW_COLOR_ERROR);
dw_printf ("Morse code speed must be in range of 5 to 50 WPM.\n");
exit (EXIT_FAILURE);
}
break;
case 'X':
experiment = 1;
break;
case '?':
/* Unknown option message was already printed. */
usage (argv);
break;
default:
/* Should not be here. */
text_color_set(DW_COLOR_ERROR);
dw_printf("?? getopt returned character code 0%o ??\n", c);
usage (argv);
}
}
/*
* Open the output file.
*/
if (strlen(output_file) == 0) {
text_color_set(DW_COLOR_ERROR);
dw_printf ("ERROR: The -o output file option must be specified.\n");
usage (argv);
exit (1);
}
err = audio_file_open (output_file, &modem);
if (err < 0) {
text_color_set(DW_COLOR_ERROR);
dw_printf ("ERROR - Can't open output file.\n");
exit (1);
}
if (experiment) {
modem.achan[0].modem_type = MODEM_QPSK;
modem.achan[0].baud = 2400; // really bps not baud.
amplitude = 100;
}
gen_tone_init (&modem, amplitude/2, 1);
morse_init (&modem, amplitude/2);
dtmf_init (&modem, amplitude/2);
assert (modem.adev[0].bits_per_sample == 8 || modem.adev[0].bits_per_sample == 16);
assert (modem.adev[0].num_channels == 1 || modem.adev[0].num_channels == 2);
assert (modem.adev[0].samples_per_sec >= MIN_SAMPLES_PER_SEC && modem.adev[0].samples_per_sec <= MAX_SAMPLES_PER_SEC);
if (experiment) {
int chan = 0;
int n;
// 6 cycles of 1800 Hz.
for (n=0; n<8; n++) {
tone_gen_put_bit (chan, 0);
}
// Shift 90
tone_gen_put_bit (chan, 0);
tone_gen_put_bit (chan, 1);
// Shift 90
tone_gen_put_bit (chan, 0);
tone_gen_put_bit (chan, 1);
// Shift 90
tone_gen_put_bit (chan, 0);
tone_gen_put_bit (chan, 1);
// Shift 90
tone_gen_put_bit (chan, 0);
tone_gen_put_bit (chan, 1);
// Shift 180
tone_gen_put_bit (chan, 1);
tone_gen_put_bit (chan, 1);
// Shift 270
tone_gen_put_bit (chan, 1);
tone_gen_put_bit (chan, 0);
// Shift 0
tone_gen_put_bit (chan, 0);
tone_gen_put_bit (chan, 0);
// Shift 0
tone_gen_put_bit (chan, 0);
tone_gen_put_bit (chan, 0);
// HDLC flag - six 1 in a row.
tone_gen_put_bit (chan, 0);
tone_gen_put_bit (chan, 1);
tone_gen_put_bit (chan, 1);
tone_gen_put_bit (chan, 1);
tone_gen_put_bit (chan, 1);
tone_gen_put_bit (chan, 1);
tone_gen_put_bit (chan, 1);
tone_gen_put_bit (chan, 0);
tone_gen_put_bit (chan, 0); // reverse even/odd position
tone_gen_put_bit (chan, 0);
tone_gen_put_bit (chan, 1);
tone_gen_put_bit (chan, 1);
tone_gen_put_bit (chan, 1);
tone_gen_put_bit (chan, 1);
tone_gen_put_bit (chan, 1);
tone_gen_put_bit (chan, 1);
tone_gen_put_bit (chan, 0);
tone_gen_put_bit (chan, 0);
// Shift 0
tone_gen_put_bit (chan, 0);
tone_gen_put_bit (chan, 0);
// Shift 0
tone_gen_put_bit (chan, 0);
tone_gen_put_bit (chan, 0);
audio_file_close ();
return (EXIT_SUCCESS);
}
/*
* Get user packets(s) from file or stdin if specified.
* "-n" option is ignored in this case.
*/
if (optind < argc) {
char str[400];
// dw_printf("non-option ARGV-elements: ");
// while (optind < argc)
// dw_printf("%s ", argv[optind++]);
//dw_printf("\n");
if (optind < argc - 1) {
text_color_set(DW_COLOR_ERROR);
dw_printf ("Warning: File(s) beyond the first are ignored.\n");
}
if (strcmp(argv[optind], "-") == 0) {
text_color_set(DW_COLOR_INFO);
dw_printf ("Reading from stdin ...\n");
input_fp = stdin;
}
else {
input_fp = fopen(argv[optind], "r");
if (input_fp == NULL) {
text_color_set(DW_COLOR_ERROR);
dw_printf ("Can't open %s for read.\n", argv[optind]);
exit (EXIT_FAILURE);
}
text_color_set(DW_COLOR_INFO);
dw_printf ("Reading from %s ...\n", argv[optind]);
}
while (fgets (str, sizeof(str), input_fp) != NULL) {
text_color_set(DW_COLOR_REC);
dw_printf ("%s", str);
send_packet (str);
}
if (input_fp != stdin) {
fclose (input_fp);
}
audio_file_close();
return EXIT_SUCCESS;
}
/*
* Otherwise, use the built in packets.
*/
text_color_set(DW_COLOR_INFO);
dw_printf ("built in message...\n");
if (packet_count > 0) {
/*
* Generate packets with increasing noise level.
* Would probably be better to record real noise from a radio but
* for now just use a random number generator.
*/
for (i = 1; i <= packet_count; i++) {
char stemp[88];
if (modem.achan[0].baud < 600) {
/* e.g. 300 bps AFSK - About 2/3 should be decoded properly. */
g_noise_level = amplitude *.0048 * ((float)i / packet_count);
}
else if (modem.achan[0].baud < 1800) {
/* e.g. 1200 bps AFSK - About 2/3 should be decoded properly. */
g_noise_level = amplitude *.0023 * ((float)i / packet_count);
}
else if (modem.achan[0].baud < 3600) {
/* e.g. 2400 bps QPSK - T.B.D. */
g_noise_level = amplitude *.0015 * ((float)i / packet_count);
}
else if (modem.achan[0].baud < 7200) {
/* e.g. 4800 bps - T.B.D. */
g_noise_level = amplitude *.0007 * ((float)i / packet_count);
}
else {
/* e.g. 9600 */
g_noise_level = 0.33 * (amplitude / 200.0) * ((float)i / packet_count);
// temp test
//g_noise_level = 0.20 * (amplitude / 200.0) * ((float)i / packet_count);
}
snprintf (stemp, sizeof(stemp), "WB2OSZ-15>TEST:,The quick brown fox jumps over the lazy dog! %04d of %04d", i, packet_count);
send_packet (stemp);
}
}
else {
/*
* Builtin default 4 packets.
*/
send_packet ("WB2OSZ-15>TEST:,The quick brown fox jumps over the lazy dog! 1 of 4");
send_packet ("WB2OSZ-15>TEST:,The quick brown fox jumps over the lazy dog! 2 of 4");
send_packet ("WB2OSZ-15>TEST:,The quick brown fox jumps over the lazy dog! 3 of 4");
send_packet ("WB2OSZ-15>TEST:,The quick brown fox jumps over the lazy dog! 4 of 4");
}
audio_file_close();
return EXIT_SUCCESS;
}
static void usage (char **argv)
{
text_color_set(DW_COLOR_ERROR);
dw_printf ("\n");
dw_printf ("Usage: gen_packets [options] [file]\n");
dw_printf ("Options:\n");
dw_printf (" -a <number> Signal amplitude in range of 0 - 200%%. Default 50.\n");
dw_printf (" -b <number> Bits / second for data. Default is %d.\n", DEFAULT_BAUD);
dw_printf (" -B <number> Bits / second for data. Proper modem selected for 300, 1200, 2400, 4800, 9600.\n");
dw_printf (" -g Scrambled baseband rather than AFSK.\n");
dw_printf (" -m <number> Mark frequency. Default is %d.\n", DEFAULT_MARK_FREQ);
dw_printf (" -s <number> Space frequency. Default is %d.\n", DEFAULT_SPACE_FREQ);
dw_printf (" -r <number> Audio sample Rate. Default is %d.\n", DEFAULT_SAMPLES_PER_SEC);
dw_printf (" -n <number> Generate specified number of frames with increasing noise.\n");
dw_printf (" -o <file> Send output to .wav file.\n");
dw_printf (" -8 8 bit audio rather than 16.\n");
dw_printf (" -2 2 channels (stereo) audio rather than one channel.\n");
// dw_printf (" -z <number> Number of leading zero bits before frame.\n");
// dw_printf (" Default is 12 which is .01 seconds at 1200 bits/sec.\n");
dw_printf ("\n");
dw_printf ("An optional file may be specified to provide messages other than\n");
dw_printf ("the default built-in message. The format should correspond to\n");
dw_printf ("the standard packet monitoring representation such as,\n\n");
dw_printf (" WB2OSZ-1>APDW12,WIDE2-2:!4237.14NS07120.83W#\n");
dw_printf ("\n");
dw_printf ("Example: gen_packets -o x.wav \n");
dw_printf ("\n");
dw_printf (" With all defaults, a built-in test message is generated\n");
dw_printf (" with standard Bell 202 tones used for packet radio on ordinary\n");
dw_printf (" VHF FM transceivers.\n");
dw_printf ("\n");
dw_printf ("Example: gen_packets -o x.wav -g -b 9600\n");
dw_printf ("Shortcut: gen_packets -o x.wav -B 9600\n");
dw_printf ("\n");
dw_printf (" 9600 baud mode.\n");
dw_printf ("\n");
dw_printf ("Example: gen_packets -o x.wav -m 1600 -s 1800 -b 300\n");
dw_printf ("Shortcut: gen_packets -o x.wav -B 300\n");
dw_printf ("\n");
dw_printf (" 200 Hz shift, 300 baud, suitable for HF SSB transceiver.\n");
dw_printf ("\n");
dw_printf ("Example: echo -n \"WB2OSZ>WORLD:Hello, world!\" | gen_packets -a 25 -o x.wav -\n");
dw_printf ("\n");
dw_printf (" Read message from stdin and put quarter volume sound into the file x.wav.\n");
exit (EXIT_FAILURE);
}
/*------------------------------------------------------------------
*
* Name: audio_file_open
*
* Purpose: Open a .WAV format file for output.
*
* Inputs: fname - Name of .WAV file to create.
*
* pa - Address of structure of type audio_s.
*
* The fields that we care about are:
* num_channels
* samples_per_sec
* bits_per_sample
* If zero, reasonable defaults will be provided.
*
* Returns: 0 for success, -1 for failure.
*
*----------------------------------------------------------------*/
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. */
} ;
/* 8 bit samples are unsigned bytes */
/* in range of 0 .. 255. */
/* 16 bit samples are signed short */
/* in range of -32768 .. +32767. */
static FILE *out_fp = NULL;
static struct wav_header header;
static int byte_count; /* Number of data bytes written to file. */
/* Will be written to header when file is closed. */
static int audio_file_open (char *fname, struct audio_s *pa)
{
int n;
/*
* Fill in defaults for any missing values.
*/
if (pa -> adev[0].num_channels == 0)
pa -> adev[0].num_channels = DEFAULT_NUM_CHANNELS;
if (pa -> adev[0].samples_per_sec == 0)
pa -> adev[0].samples_per_sec = DEFAULT_SAMPLES_PER_SEC;
if (pa -> adev[0].bits_per_sample == 0)
pa -> adev[0].bits_per_sample = DEFAULT_BITS_PER_SAMPLE;
/*
* Write the file header. Don't know length yet.
*/
out_fp = fopen (fname, "wb");
if (out_fp == NULL) {
text_color_set(DW_COLOR_ERROR); dw_printf ("Couldn't open file for write: %s\n", fname);
perror ("");
return (-1);
}
memset (&header, 0, sizeof(header));
memcpy (header.riff, "RIFF", (size_t)4);
header.filesize = 0;
memcpy (header.wave, "WAVE", (size_t)4);
memcpy (header.fmt, "fmt ", (size_t)4);
header.fmtsize = 16; // Always 16.
header.wformattag = 1; // 1 for PCM.
header.nchannels = pa -> adev[0].num_channels;
header.nsamplespersec = pa -> adev[0].samples_per_sec;
header.wbitspersample = pa -> adev[0].bits_per_sample;
header.nblockalign = header.wbitspersample / 8 * header.nchannels;
header.navgbytespersec = header.nblockalign * header.nsamplespersec;
memcpy (header.data, "data", (size_t)4);
header.datasize = 0;
assert (header.nchannels == 1 || header.nchannels == 2);
n = fwrite (&header, sizeof(header), (size_t)1, out_fp);
if (n != 1) {
text_color_set(DW_COLOR_ERROR);
dw_printf ("Couldn't write header to: %s\n", fname);
perror ("");
fclose (out_fp);
out_fp = NULL;
return (-1);
}
/*
* Number of bytes written will be filled in later.
*/
byte_count = 0;
return (0);
} /* end audio_open */
/*------------------------------------------------------------------
*
* Name: audio_put
*
* Purpose: Send one byte to the audio output file.
*
* Inputs: c - One byte in range of 0 - 255.
*
* Returns: Normally non-negative.
* -1 for any type of error.
*
* Description: The caller must deal with the details of mono/stereo
* and number of bytes per sample.
*
*----------------------------------------------------------------*/
int audio_put (int a, int c)
{
static short sample16;
int s;
if (g_add_noise) {
if ((byte_count & 1) == 0) {
sample16 = c & 0xff; /* save lower byte. */
byte_count++;
return c;
}
else {
float r;
sample16 |= (c << 8) & 0xff00; /* insert upper byte. */
byte_count++;
s = sample16; // sign extend.
/* Add random noise to the signal. */
/* r should be in range of -1 .. +1. */
/* Use own function instead of rand() from the C library. */
/* Windows and Linux have different results, messing up my self test procedure. */
/* No idea what Mac OSX and BSD might do. */
r = (my_rand() - MY_RAND_MAX/2.0) / (MY_RAND_MAX/2.0);
s += 5 * r * g_noise_level * 32767;
if (s > 32767) s = 32767;
if (s < -32767) s = -32767;
putc(s & 0xff, out_fp);
return (putc((s >> 8) & 0xff, out_fp));
}
}
else {
byte_count++;
return (putc(c, out_fp));
}
} /* end audio_put */
int audio_flush (int a)
{
return 0;
}
/*------------------------------------------------------------------
*
* Name: audio_file_close
*
* Purpose: Close the audio output file.
*
* Returns: Normally non-negative.
* -1 for any type of error.
*
*
* Description: Must go back to beginning of file and fill in the
* size of the data.
*
*----------------------------------------------------------------*/
static int audio_file_close (void)
{
int n;
//text_color_set(DW_COLOR_DEBUG);
//dw_printf ("audio_close()\n");
/*
* Go back and fix up lengths in header.
*/
header.filesize = byte_count + sizeof(header) - 8;
header.datasize = byte_count;
if (out_fp == NULL) {
return (-1);
}
fflush (out_fp);
fseek (out_fp, 0L, SEEK_SET);
n = fwrite (&header, sizeof(header), (size_t)1, out_fp);
if (n != 1) {
text_color_set(DW_COLOR_ERROR);
dw_printf ("Couldn't write header to audio file.\n");
perror (""); // TODO: remove perror.
fclose (out_fp);
out_fp = NULL;
return (-1);
}
fclose (out_fp);
out_fp = NULL;
return (0);
} /* end audio_close */
// To keep dtmf.c happy.
#include "hdlc_rec.h" // for dcd_change
void dcd_change (int chan, int subchan, int slice, int state)
{
}
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