/* * osmo-fl2k, turns FL2000-based USB 3.0 to VGA adapters into * low cost DACs * * Copyright (C) 2016-2018 by Steve Markgraf * * based on FM modulator code from VGASIG: * Copyright (C) 2009 by Bartek Kania * * 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 . */ #include #include #include #include #include #ifndef _WIN32 #include #include #else #include #include #include #include "getopt/getopt.h" #endif #include #include #include "osmo-fl2k.h" #define BUFFER_SAMPLES_SHIFT 16 #define BUFFER_SAMPLES (1 << BUFFER_SAMPLES_SHIFT) #define BUFFER_SAMPLES_MASK ((1 << BUFFER_SAMPLES_SHIFT)-1) #define AUDIO_BUF_SIZE 1024 fl2k_dev_t *dev = NULL; int do_exit = 0; pthread_t fm_thread; pthread_mutex_t cb_mutex; pthread_mutex_t fm_mutex; pthread_cond_t cb_cond; pthread_cond_t fm_cond; FILE *file; int8_t *txbuf = NULL; int8_t *fmbuf = NULL; int8_t *buf1 = NULL; int8_t *buf2 = NULL; uint32_t samp_rate = 100000000; /* default signal parameters */ int delta_freq = 75000; int carrier_freq = 97000000; int carrier_per_signal; int input_freq = 44100; double *freqbuf; double *slopebuf; int writepos, readpos; void usage(void) { fprintf(stderr, "fl2k_fm, an FM modulator for FL2K VGA dongles\n\n" "Usage:" "\t[-d device index (default: 0)]\n" "\t[-c carrier frequency (default: 9.7 MHz)]\n" "\t[-f FM deviation (default: 75000 Hz, WBFM)]\n" "\t[-i input audio sample rate (default: 44100 Hz)]\n" "\t[-s samplerate in Hz (default: 100 MS/s)]\n" "\tfilename (use '-' to read from stdin)\n\n" ); exit(1); } #ifdef _WIN32 BOOL WINAPI sighandler(int signum) { if (CTRL_C_EVENT == signum) { fprintf(stderr, "Signal caught, exiting!\n"); fl2k_stop_tx(dev); do_exit = 1; pthread_cond_signal(&fm_cond); return TRUE; } return FALSE; } #else static void sighandler(int signum) { fprintf(stderr, "Signal caught, exiting!\n"); fl2k_stop_tx(dev); do_exit = 1; pthread_cond_signal(&fm_cond); } #endif /* DDS Functions */ #ifndef M_PI # define M_PI 3.14159265358979323846 /* pi */ # define M_PI_2 1.57079632679489661923 /* pi/2 */ # define M_PI_4 0.78539816339744830962 /* pi/4 */ # define M_1_PI 0.31830988618379067154 /* 1/pi */ # define M_2_PI 0.63661977236758134308 /* 2/pi */ #endif #define DDS_2PI (M_PI * 2) /* 2 * Pi */ #define DDS_3PI2 (M_PI_2 * 3) /* 3/2 * pi */ #define SIN_TABLE_ORDER 8 #define SIN_TABLE_SHIFT (32 - SIN_TABLE_ORDER) #define SIN_TABLE_LEN (1 << SIN_TABLE_ORDER) #define ANG_INCR (0xffffffff / DDS_2PI) int8_t sine_table[SIN_TABLE_LEN]; int sine_table_init = 0; typedef struct { double sample_freq; double freq; double fslope; unsigned long int phase; unsigned long int phase_step; unsigned long int phase_slope; } dds_t; inline void dds_setphase(dds_t *dds, double phase) { dds->phase = phase * ANG_INCR; } inline double dds_getphase(dds_t *dds) { return dds->phase / ANG_INCR; } inline void dds_set_freq(dds_t *dds, double freq, double fslope) { dds->fslope = fslope; dds->phase_step = (freq / dds->sample_freq) * 2 * M_PI * ANG_INCR; /* The slope parameter is used with the FM modulator to create * a simple but very fast and effective interpolation filter. * See the fm modulator for details */ dds->freq = freq; dds->phase_slope = (fslope / dds->sample_freq) * 2 * M_PI * ANG_INCR; } dds_t dds_init(double sample_freq, double freq, double phase) { dds_t dds; int i; dds.sample_freq = sample_freq; dds.phase = phase * ANG_INCR; dds_set_freq(&dds, freq, 0); /* Initialize sine table, prescaled for 8 bit signed integer */ if (!sine_table_init) { double incr = 1.0 / (double)SIN_TABLE_LEN; for (i = 0; i < SIN_TABLE_LEN; i++) sine_table[i] = sin(incr * i * DDS_2PI) * 127; sine_table_init = 1; } return dds; } inline int8_t dds_real(dds_t *dds) { int tmp; tmp = dds->phase >> SIN_TABLE_SHIFT; dds->phase += dds->phase_step; dds->phase &= 0xffffffff; dds->phase_step += dds->phase_slope; return sine_table[tmp]; } inline void dds_real_buf(dds_t *dds, int8_t *buf, int count) { int i; for (i = 0; i < count; i++) buf[i] = dds_real(dds); } /* Signal generation and some helpers */ /* Generate the radio signal using the pre-calculated frequency information * in the freq buffer */ static void *fm_worker(void *arg) { register double freq; register double tmp; dds_t carrier; int8_t *tmp_ptr; uint32_t len = 0; uint32_t readlen, remaining; int buf_prefilled = 0; /* Prepare the oscillators */ carrier = dds_init(samp_rate, carrier_freq, 0); while (!do_exit) { dds_set_freq(&carrier, freqbuf[readpos], slopebuf[readpos]); readpos++; readpos &= BUFFER_SAMPLES_MASK; /* check if we reach the end of the buffer */ if ((len + carrier_per_signal) > FL2K_BUF_LEN) { readlen = FL2K_BUF_LEN - len; remaining = carrier_per_signal - readlen; dds_real_buf(&carrier, &fmbuf[len], readlen); if (buf_prefilled) { /* swap buffers */ tmp_ptr = fmbuf; fmbuf = txbuf; txbuf = tmp_ptr; pthread_cond_wait(&cb_cond, &cb_mutex); } dds_real_buf(&carrier, fmbuf, remaining); len = remaining; buf_prefilled = 1; } else { dds_real_buf(&carrier, &fmbuf[len], carrier_per_signal); len += carrier_per_signal; } pthread_cond_signal(&fm_cond); } pthread_exit(NULL); } inline int writelen(int maxlen) { int rp = readpos; int len; int r; if (rp < writepos) rp += BUFFER_SAMPLES; len = rp - writepos; r = len > maxlen ? maxlen : len; return r; } inline double modulate_sample(int lastwritepos, double lastfreq, double sample) { double freq, slope; /* Calculate modulator frequency at this point to lessen * the calculations needed in the signal generator */ freq = sample * delta_freq; freq += carrier_freq; /* What we do here is calculate a linear "slope" from the previous sample to this one. This is then used by the modulator to gently increase/decrease the frequency with each sample without the need to recalculate the dds parameters. In fact this gives us a very efficient and pretty good interpolation filter. */ slope = freq - lastfreq; slope /= carrier_per_signal; slopebuf[lastwritepos] = slope; freqbuf[writepos] = freq; return freq; } void modulator(void) { unsigned int i; size_t len; double freq; double lastfreq = carrier_freq; double slope; int16_t audio_buf[AUDIO_BUF_SIZE]; uint32_t lastwritepos = writepos; while (!do_exit) { len = writelen(AUDIO_BUF_SIZE); if (len > 1) { len = fread(audio_buf, 2, len, file); if (len == 0) do_exit = 1; for (i = 0; i < len; i++) { /* Modulate and buffer the sample */ lastfreq = modulate_sample(lastwritepos, lastfreq, audio_buf[i]/32767.0); lastwritepos = writepos++; writepos %= BUFFER_SAMPLES; } } else { pthread_cond_wait(&fm_cond, &fm_mutex); } } } void fl2k_callback(fl2k_data_info_t *data_info) { if (data_info->device_error) { do_exit = 1; pthread_cond_signal(&fm_cond); } pthread_cond_signal(&cb_cond); data_info->sampletype_signed = 1; data_info->r_buf = (char *)txbuf; } int main(int argc, char **argv) { int r, opt; uint32_t buf_num = 0; int dev_index = 0; pthread_attr_t attr; char *filename = NULL; #ifndef _WIN32 struct sigaction sigact, sigign; #endif while ((opt = getopt(argc, argv, "d:c:f:i:s:")) != -1) { switch (opt) { case 'd': dev_index = (uint32_t)atoi(optarg); break; case 'c': carrier_freq = (uint32_t)atof(optarg); break; case 'f': delta_freq = (uint32_t)atof(optarg); break; case 'i': input_freq = (uint32_t)atof(optarg); break; case 's': samp_rate = (uint32_t)atof(optarg); break; default: usage(); break; } } if (argc <= optind) { usage(); } else { filename = argv[optind]; } if (dev_index < 0) { exit(1); } if(strcmp(filename, "-") == 0) { /* Read samples from stdin */ file = stdin; #ifdef _WIN32 _setmode(_fileno(stdin), _O_BINARY); #endif } else { file = fopen(filename, "rb"); if (!file) { fprintf(stderr, "Failed to open %s\n", filename); return -ENOENT; } } /* allocate buffer */ buf1 = malloc(FL2K_BUF_LEN); buf2 = malloc(FL2K_BUF_LEN); if (!buf1 || !buf2) { fprintf(stderr, "malloc error!\n"); exit(1); } fmbuf = buf1; txbuf = buf2; /* Decoded audio */ freqbuf = malloc(BUFFER_SAMPLES * sizeof(double)); slopebuf = malloc(BUFFER_SAMPLES * sizeof(double)); readpos = 0; writepos = 1; fprintf(stderr, "Samplerate:\t%3.2f MHz\n", (double)samp_rate/1000000); fprintf(stderr, "Carrier:\t%3.2f MHz\n", (double)carrier_freq/1000000); fprintf(stderr, "Frequencies:\t%3.2f MHz, %3.2f MHz\n", (double)((samp_rate - carrier_freq) / 1000000.0), (double)((samp_rate + carrier_freq) / 1000000.0)); pthread_mutex_init(&cb_mutex, NULL); pthread_mutex_init(&fm_mutex, NULL); pthread_cond_init(&cb_cond, NULL); pthread_cond_init(&fm_cond, NULL); pthread_attr_init(&attr); fl2k_open(&dev, (uint32_t)dev_index); if (NULL == dev) { fprintf(stderr, "Failed to open fl2k device #%d.\n", dev_index); goto out; } r = pthread_create(&fm_thread, &attr, fm_worker, NULL); if (r < 0) { fprintf(stderr, "Error spawning FM worker thread!\n"); goto out; } pthread_attr_destroy(&attr); r = fl2k_start_tx(dev, fl2k_callback, NULL, 0); /* Set the sample rate */ r = fl2k_set_sample_rate(dev, samp_rate); if (r < 0) fprintf(stderr, "WARNING: Failed to set sample rate. %d\n", r); /* read back actual frequency */ samp_rate = fl2k_get_sample_rate(dev); /* Calculate needed constants */ carrier_per_signal = samp_rate / input_freq; carrier_freq = samp_rate - carrier_freq; #ifndef _WIN32 sigact.sa_handler = sighandler; sigemptyset(&sigact.sa_mask); sigact.sa_flags = 0; sigign.sa_handler = SIG_IGN; sigaction(SIGINT, &sigact, NULL); sigaction(SIGTERM, &sigact, NULL); sigaction(SIGQUIT, &sigact, NULL); sigaction(SIGPIPE, &sigign, NULL); #else SetConsoleCtrlHandler( (PHANDLER_ROUTINE) sighandler, TRUE ); #endif modulator(); out: fl2k_close(dev); if (file != stdin) fclose(file); free(freqbuf); free(slopebuf); free(buf1); free(buf2); return 0; }