// // This file is part of Dire Wolf, an amateur radio packet TNC. // // Copyright (C) 2011, 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 . // /*------------------------------------------------------------------ * * 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 #include #include #include #include #include "audio.h" #include "ax25_pad.h" #include "hdlc_send.h" #include "gen_tone.h" #include "textcolor.h" #include "morse.h" 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) { 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 10000) { text_color_set(DW_COLOR_ERROR); dw_printf ("Use a more reasonable bit rate in range of 100 - 10000.\n"); exit (EXIT_FAILURE); } break; case 'B': /* -B for data Bit rate */ /* 300 implies 1600/1800 AFSK. */ /* 1200 implies 1200/2200 AFSK. */ /* 9600 implies scrambled. */ 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 > 10000) { text_color_set(DW_COLOR_ERROR); dw_printf ("Use a more reasonable bit rate in range of 100 - 10000.\n"); exit (EXIT_FAILURE); } switch (modem.achan[0].baud) { case 300: modem.achan[0].mark_freq = 1600; modem.achan[0].space_freq = 1800; break; case 1200: modem.achan[0].mark_freq = 1200; modem.achan[0].space_freq = 2200; break; case 9600: modem.achan[0].modem_type = MODEM_SCRAMBLE; text_color_set(DW_COLOR_INFO); dw_printf ("Using scrambled baseband signal rather than AFSK.\n"); break; } 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. 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 '?': /* 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 ouput 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); } gen_tone_init (&modem, amplitude/2); morse_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); /* * 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[80]; if (modem.achan[0].modem_type == MODEM_SCRAMBLE) { g_noise_level = 0.33 * (amplitude / 200.0) * ((float)i / packet_count); } else if (modem.achan[0].baud < 600) { /* About 2/3 should be decoded properly. */ g_noise_level = amplitude *.0048 * ((float)i / packet_count); } else { /* About 2/3 should be decoded properly. */ g_noise_level = amplitude *.0023 * ((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 Signal amplitude in range of 0 - 200%%. Default 50.\n"); dw_printf (" -b Bits / second for data. Default is %d.\n", DEFAULT_BAUD); dw_printf (" -B Bits / second for data. Proper modem selected for 300, 1200, 9600.\n"); dw_printf (" -g Scrambled baseband rather than AFSK.\n"); dw_printf (" -m Mark frequency. Default is %d.\n", DEFAULT_MARK_FREQ); dw_printf (" -s Space frequency. Default is %d.\n", DEFAULT_SPACE_FREQ); dw_printf (" -r Audio sample Rate. Default is %d.\n", DEFAULT_SAMPLES_PER_SEC); dw_printf (" -n Generate specified number of frames with increasing noise.\n"); dw_printf (" -o Send output to .wav file.\n"); // dw_printf (" -8 8 bit audio rather than 16.\n"); // dw_printf (" -2 2 channels of audio rather than 1.\n"); // dw_printf (" -z 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. * *----------------------------------------------------------------*/ #define MY_RAND_MAX 0x7fffffff static int seed = 1; static int my_rand (void) { seed = ((seed * 1103515245) + 12345) & MY_RAND_MAX; return (seed); } 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 */