| 1 | /* |
| 2 | * This file is part of DisOrder. |
| 3 | * Copyright (C) 2007 Richard Kettlewell |
| 4 | * |
| 5 | * This program is free software; you can redistribute it and/or modify |
| 6 | * it under the terms of the GNU General Public License as published by |
| 7 | * the Free Software Foundation; either version 2 of the License, or |
| 8 | * (at your option) any later version. |
| 9 | * |
| 10 | * This program is distributed in the hope that it will be useful, but |
| 11 | * WITHOUT ANY WARRANTY; without even the implied warranty of |
| 12 | * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU |
| 13 | * General Public License for more details. |
| 14 | * |
| 15 | * You should have received a copy of the GNU General Public License |
| 16 | * along with this program; if not, write to the Free Software |
| 17 | * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 |
| 18 | * USA |
| 19 | */ |
| 20 | |
| 21 | #include <config.h> |
| 22 | #include "types.h" |
| 23 | |
| 24 | #include <getopt.h> |
| 25 | #include <stdio.h> |
| 26 | #include <stdlib.h> |
| 27 | #include <sys/socket.h> |
| 28 | #include <sys/types.h> |
| 29 | #include <sys/socket.h> |
| 30 | #include <netdb.h> |
| 31 | #include <pthread.h> |
| 32 | #include <locale.h> |
| 33 | |
| 34 | #include "log.h" |
| 35 | #include "mem.h" |
| 36 | #include "configuration.h" |
| 37 | #include "addr.h" |
| 38 | #include "syscalls.h" |
| 39 | #include "rtp.h" |
| 40 | #include "defs.h" |
| 41 | |
| 42 | #if HAVE_COREAUDIO_AUDIOHARDWARE_H |
| 43 | # include <CoreAudio/AudioHardware.h> |
| 44 | #endif |
| 45 | #if API_ALSA |
| 46 | #include <alsa/asoundlib.h> |
| 47 | #endif |
| 48 | |
| 49 | /** @brief RTP socket */ |
| 50 | static int rtpfd; |
| 51 | |
| 52 | /** @brief Output device */ |
| 53 | static const char *device; |
| 54 | |
| 55 | /** @brief Maximum samples per packet we'll support |
| 56 | * |
| 57 | * NB that two channels = two samples in this program. |
| 58 | */ |
| 59 | #define MAXSAMPLES 2048 |
| 60 | |
| 61 | /** @brief Minimum buffer size |
| 62 | * |
| 63 | * We'll stop playing if there's only this many samples in the buffer. */ |
| 64 | #define MINBUFFER 8820 |
| 65 | |
| 66 | /** @brief Maximum sample size |
| 67 | * |
| 68 | * The maximum supported size (in bytes) of one sample. */ |
| 69 | #define MAXSAMPLESIZE 2 |
| 70 | |
| 71 | #define READAHEAD 88200 /* how far to read ahead */ |
| 72 | |
| 73 | #define MAXBUFFER (3 * 88200) /* maximum buffer contents */ |
| 74 | |
| 75 | /** @brief Received packet |
| 76 | * |
| 77 | * Packets are recorded in an ordered linked list. */ |
| 78 | struct packet { |
| 79 | /** @brief Pointer to next packet |
| 80 | * The next packet might not be immediately next: if packets are dropped |
| 81 | * or mis-ordered there may be gaps at any given moment. */ |
| 82 | struct packet *next; |
| 83 | /** @brief Number of samples in this packet */ |
| 84 | int nsamples; |
| 85 | /** @brief Number of samples used from this packet */ |
| 86 | int nused; |
| 87 | /** @brief Timestamp from RTP packet |
| 88 | * |
| 89 | * NB that "timestamps" are really sample counters.*/ |
| 90 | uint32_t timestamp; |
| 91 | #if HAVE_COREAUDIO_AUDIOHARDWARE_H |
| 92 | /** @brief Converted sample data */ |
| 93 | float samples_float[MAXSAMPLES]; |
| 94 | #else |
| 95 | /** @brief Raw sample data */ |
| 96 | unsigned char samples_raw[MAXSAMPLES * MAXSAMPLESIZE]; |
| 97 | #endif |
| 98 | }; |
| 99 | |
| 100 | /** @brief Total number of samples available */ |
| 101 | static unsigned long nsamples; |
| 102 | |
| 103 | /** @brief Linked list of packets |
| 104 | * |
| 105 | * In ascending order of timestamp. */ |
| 106 | static struct packet *packets; |
| 107 | |
| 108 | /** @brief Timestamp of next packet to play. |
| 109 | * |
| 110 | * This is set to the timestamp of the last packet, plus the number of |
| 111 | * samples it contained. |
| 112 | */ |
| 113 | static uint32_t next_timestamp; |
| 114 | |
| 115 | /** @brief Lock protecting @ref packets */ |
| 116 | static pthread_mutex_t lock = PTHREAD_MUTEX_INITIALIZER; |
| 117 | |
| 118 | /** @brief Condition variable signalled whenever @ref packets is changed */ |
| 119 | static pthread_cond_t cond = PTHREAD_COND_INITIALIZER; |
| 120 | |
| 121 | static const struct option options[] = { |
| 122 | { "help", no_argument, 0, 'h' }, |
| 123 | { "version", no_argument, 0, 'V' }, |
| 124 | { "debug", no_argument, 0, 'd' }, |
| 125 | { "device", required_argument, 0, 'D' }, |
| 126 | { 0, 0, 0, 0 } |
| 127 | }; |
| 128 | |
| 129 | /** @brief Return true iff a < b in sequence-space arithmetic */ |
| 130 | static inline int lt(const struct packet *a, const struct packet *b) { |
| 131 | return (uint32_t)(a->timestamp - b->timestamp) & 0x80000000; |
| 132 | } |
| 133 | |
| 134 | /** Background thread collecting samples |
| 135 | * |
| 136 | * This function collects samples, perhaps converts them to the target format, |
| 137 | * and adds them to the packet list. */ |
| 138 | static void *listen_thread(void attribute((unused)) *arg) { |
| 139 | struct packet *f = 0, **ff; |
| 140 | int n; |
| 141 | union { |
| 142 | struct rtp_header header; |
| 143 | uint8_t bytes[sizeof(uint16_t) * MAXSAMPLES + sizeof (struct rtp_header)]; |
| 144 | } packet; |
| 145 | const uint16_t *const samples = (uint16_t *)(packet.bytes |
| 146 | + sizeof (struct rtp_header)); |
| 147 | |
| 148 | for(;;) { |
| 149 | if(!f) |
| 150 | f = xmalloc(sizeof *f); |
| 151 | n = read(rtpfd, packet.bytes, sizeof packet.bytes); |
| 152 | if(n < 0) { |
| 153 | switch(errno) { |
| 154 | case EINTR: |
| 155 | continue; |
| 156 | default: |
| 157 | fatal(errno, "error reading from socket"); |
| 158 | } |
| 159 | } |
| 160 | /* Ignore too-short packets */ |
| 161 | if((size_t)n <= sizeof (struct rtp_header)) |
| 162 | continue; |
| 163 | /* Convert to target format */ |
| 164 | switch(packet.header.mpt & 0x7F) { |
| 165 | case 10: |
| 166 | f->nsamples = (n - sizeof (struct rtp_header)) / sizeof(uint16_t); |
| 167 | #if HAVE_COREAUDIO_AUDIOHARDWARE_H |
| 168 | /* Convert to what Core Audio expects */ |
| 169 | for(n = 0; n < f->nsamples; ++n) |
| 170 | f->samples_float[n] = (int16_t)ntohs(samples[n]) * (0.5f / 32767); |
| 171 | #else |
| 172 | /* ALSA can do any necessary conversion itself (though it might be better |
| 173 | * to do any necessary conversion in the background) */ |
| 174 | memcpy(f->samples_raw, samples, n - sizeof (struct rtp_header)); |
| 175 | #endif |
| 176 | break; |
| 177 | /* TODO support other RFC3551 media types (when the speaker does) */ |
| 178 | default: |
| 179 | fatal(0, "unsupported RTP payload type %d", |
| 180 | packet.header.mpt & 0x7F); |
| 181 | } |
| 182 | f->nused = 0; |
| 183 | f->timestamp = ntohl(packet.header.timestamp); |
| 184 | pthread_mutex_lock(&lock); |
| 185 | /* Stop reading if we've reached the maximum. |
| 186 | * |
| 187 | * This is rather unsatisfactory: it means that if packets get heavily |
| 188 | * out of order then we guarantee dropouts. But for now... */ |
| 189 | while(nsamples >= MAXBUFFER) |
| 190 | pthread_cond_wait(&cond, &lock); |
| 191 | for(ff = &packets; *ff && lt(*ff, f); ff = &(*ff)->next) |
| 192 | ; |
| 193 | /* So now either !*ff or *ff >= f */ |
| 194 | if(*ff && f->timestamp == (*ff)->timestamp) { |
| 195 | /* *ff == f; a duplicate. Ideally we avoid the translation step here, |
| 196 | * but we'll worry about that another time. */ |
| 197 | free(f); |
| 198 | } else { |
| 199 | f->next = *ff; |
| 200 | *ff = f; |
| 201 | nsamples += f->nsamples; |
| 202 | pthread_cond_broadcast(&cond); |
| 203 | } |
| 204 | pthread_mutex_unlock(&lock); |
| 205 | f = 0; |
| 206 | } |
| 207 | } |
| 208 | |
| 209 | #if HAVE_COREAUDIO_AUDIOHARDWARE_H |
| 210 | static OSStatus adioproc(AudioDeviceID inDevice, |
| 211 | const AudioTimeStamp *inNow, |
| 212 | const AudioBufferList *inInputData, |
| 213 | const AudioTimeStamp *inInputTime, |
| 214 | AudioBufferList *outOutputData, |
| 215 | const AudioTimeStamp *inOutputTime, |
| 216 | void *inClientData) { |
| 217 | UInt32 nbuffers = outOutputData->mNumberBuffers; |
| 218 | AudioBuffer *ab = outOutputData->mBuffers; |
| 219 | float *samplesOut; /* where to write samples to */ |
| 220 | size_t samplesOutLeft; /* space left */ |
| 221 | size_t samplesInLeft; |
| 222 | size_t samplesToCopy; |
| 223 | |
| 224 | pthread_mutex_lock(&lock); |
| 225 | samplesOut = ab->data; |
| 226 | samplesOutLeft = ab->mDataByteSize / sizeof (float); |
| 227 | while(packets && nbuffers > 0) { |
| 228 | if(packets->used == packets->nsamples) { |
| 229 | /* TODO if we dropped a packet then we should introduce a gap here */ |
| 230 | struct packet *const f = packets; |
| 231 | packets = f->next; |
| 232 | free(f); |
| 233 | pthread_cond_broadcast(&cond); |
| 234 | continue; |
| 235 | } |
| 236 | if(samplesOutLeft == 0) { |
| 237 | --nbuffers; |
| 238 | ++ab; |
| 239 | samplesOut = ab->data; |
| 240 | samplesOutLeft = ab->mDataByteSize / sizeof (float); |
| 241 | continue; |
| 242 | } |
| 243 | /* Now: (1) there is some data left to read |
| 244 | * (2) there is some space to put it */ |
| 245 | samplesInLeft = packets->nsamples - packets->used; |
| 246 | samplesToCopy = (samplesInLeft < samplesOutLeft |
| 247 | ? samplesInLeft : samplesOutLeft); |
| 248 | memcpy(samplesOut, packet->samples + packets->used, samplesToCopy); |
| 249 | packets->used += samplesToCopy; |
| 250 | samplesOut += samplesToCopy; |
| 251 | samesOutLeft -= samplesToCopy; |
| 252 | } |
| 253 | pthread_mutex_unlock(&lock); |
| 254 | return 0; |
| 255 | } |
| 256 | #endif |
| 257 | |
| 258 | static void play_rtp(void) { |
| 259 | pthread_t ltid; |
| 260 | |
| 261 | /* We receive and convert audio data in a background thread */ |
| 262 | pthread_create(<id, 0, listen_thread, 0); |
| 263 | #if API_ALSA |
| 264 | { |
| 265 | snd_pcm_t *pcm; |
| 266 | snd_pcm_hw_params_t *hwparams; |
| 267 | snd_pcm_sw_params_t *swparams; |
| 268 | /* Only support one format for now */ |
| 269 | const int sample_format = SND_PCM_FORMAT_S16_BE; |
| 270 | unsigned rate = 44100; |
| 271 | const int channels = 2; |
| 272 | const int samplesize = channels * sizeof(uint16_t); |
| 273 | snd_pcm_uframes_t pcm_bufsize = MAXSAMPLES * samplesize * 3; |
| 274 | /* If we can write more than this many samples we'll get a wakeup */ |
| 275 | const int avail_min = 256; |
| 276 | snd_pcm_sframes_t frames_written; |
| 277 | size_t samples_written; |
| 278 | int prepared = 1; |
| 279 | int err; |
| 280 | |
| 281 | /* Open ALSA */ |
| 282 | if((err = snd_pcm_open(&pcm, |
| 283 | device ? device : "default", |
| 284 | SND_PCM_STREAM_PLAYBACK, |
| 285 | SND_PCM_NONBLOCK))) |
| 286 | fatal(0, "error from snd_pcm_open: %d", err); |
| 287 | /* Set up 'hardware' parameters */ |
| 288 | snd_pcm_hw_params_alloca(&hwparams); |
| 289 | if((err = snd_pcm_hw_params_any(pcm, hwparams)) < 0) |
| 290 | fatal(0, "error from snd_pcm_hw_params_any: %d", err); |
| 291 | if((err = snd_pcm_hw_params_set_access(pcm, hwparams, |
| 292 | SND_PCM_ACCESS_RW_INTERLEAVED)) < 0) |
| 293 | fatal(0, "error from snd_pcm_hw_params_set_access: %d", err); |
| 294 | if((err = snd_pcm_hw_params_set_format(pcm, hwparams, |
| 295 | sample_format)) < 0) |
| 296 | fatal(0, "error from snd_pcm_hw_params_set_format (%d): %d", |
| 297 | sample_format, err); |
| 298 | if((err = snd_pcm_hw_params_set_rate_near(pcm, hwparams, &rate, 0)) < 0) |
| 299 | fatal(0, "error from snd_pcm_hw_params_set_rate (%d): %d", |
| 300 | rate, err); |
| 301 | if((err = snd_pcm_hw_params_set_channels(pcm, hwparams, |
| 302 | channels)) < 0) |
| 303 | fatal(0, "error from snd_pcm_hw_params_set_channels (%d): %d", |
| 304 | channels, err); |
| 305 | if((err = snd_pcm_hw_params_set_buffer_size_near(pcm, hwparams, |
| 306 | &pcm_bufsize)) < 0) |
| 307 | fatal(0, "error from snd_pcm_hw_params_set_buffer_size (%d): %d", |
| 308 | MAXSAMPLES * samplesize * 3, err); |
| 309 | if((err = snd_pcm_hw_params(pcm, hwparams)) < 0) |
| 310 | fatal(0, "error calling snd_pcm_hw_params: %d", err); |
| 311 | /* Set up 'software' parameters */ |
| 312 | snd_pcm_sw_params_alloca(&swparams); |
| 313 | if((err = snd_pcm_sw_params_current(pcm, swparams)) < 0) |
| 314 | fatal(0, "error calling snd_pcm_sw_params_current: %d", err); |
| 315 | if((err = snd_pcm_sw_params_set_avail_min(pcm, swparams, avail_min)) < 0) |
| 316 | fatal(0, "error calling snd_pcm_sw_params_set_avail_min %d: %d", |
| 317 | avail_min, err); |
| 318 | if((err = snd_pcm_sw_params(pcm, swparams)) < 0) |
| 319 | fatal(0, "error calling snd_pcm_sw_params: %d", err); |
| 320 | |
| 321 | /* Ready to go */ |
| 322 | |
| 323 | pthread_mutex_lock(&lock); |
| 324 | for(;;) { |
| 325 | /* Wait for the buffer to fill up a bit */ |
| 326 | while(nsamples < READAHEAD) |
| 327 | pthread_cond_wait(&cond, &lock); |
| 328 | if(!prepared) { |
| 329 | if((err = snd_pcm_prepare(pcm))) |
| 330 | fatal(0, "error calling snd_pcm_prepare: %d", err); |
| 331 | prepared = 1; |
| 332 | } |
| 333 | /* Wait until the buffer empties out */ |
| 334 | while(nsamples >= MINBUFFER) { |
| 335 | /* Wait for ALSA to ask us for more data */ |
| 336 | pthread_mutex_unlock(&lock); |
| 337 | snd_pcm_wait(pcm, -1); |
| 338 | pthread_mutex_lock(&lock); |
| 339 | /* ALSA wants more data */ |
| 340 | if(packets && packets->timestamp + packets->nused == next_timestamp) { |
| 341 | /* Hooray, we have a packet we can play */ |
| 342 | const size_t samples_available = packets->nsamples - packets->nused; |
| 343 | const size_t frames_available = samples_available / 2; |
| 344 | |
| 345 | frames_written = snd_pcm_writei(pcm, |
| 346 | packets->samples_raw + packets->nused, |
| 347 | frames_available); |
| 348 | if(frames_written < 0) |
| 349 | fatal(0, "error calling snd_pcm_writei: %d", err); |
| 350 | samples_written = frames_written * 2; |
| 351 | packets->nused += samples_written; |
| 352 | next_timestamp += samples_written; |
| 353 | if(packets->nused == packets->nsamples) { |
| 354 | struct packet *f = packets; |
| 355 | |
| 356 | packets = f->next; |
| 357 | nsamples -= f->nsamples; |
| 358 | free(f); |
| 359 | pthread_cond_broadcast(&cond); |
| 360 | } |
| 361 | } else { |
| 362 | /* We don't have anything to play! We'd better play some 0s. */ |
| 363 | static const uint16_t zeros[1024]; |
| 364 | size_t samples_available = 1024, frames_available; |
| 365 | if(packets && next_timestamp + samples_available > packets->timestamp) |
| 366 | samples_available = packets->timestamp - next_timestamp; |
| 367 | frames_available = samples_available / 2; |
| 368 | frames_written = snd_pcm_writei(pcm, |
| 369 | zeros, |
| 370 | frames_available); |
| 371 | if(frames_written < 0) |
| 372 | fatal(0, "error calling snd_pcm_writei: %d", err); |
| 373 | next_timestamp += samples_written; |
| 374 | } |
| 375 | } |
| 376 | /* We stop playing for a bit until the buffer re-fills */ |
| 377 | pthread_mutex_unlock(&lock); |
| 378 | if((err = snd_pcm_drain(pcm))) |
| 379 | fatal(0, "error calling snd_pcm_drain: %d", err); |
| 380 | prepared = 0; |
| 381 | pthread_mutex_lock(&lock); |
| 382 | } |
| 383 | |
| 384 | } |
| 385 | #elif HAVE_COREAUDIO_AUDIOHARDWARE_H |
| 386 | { |
| 387 | OSStatus status; |
| 388 | UInt32 propertySize; |
| 389 | AudioDeviceID adid; |
| 390 | AudioStreamBasicDescription asbd; |
| 391 | |
| 392 | /* If this looks suspiciously like libao's macosx driver there's an |
| 393 | * excellent reason for that... */ |
| 394 | |
| 395 | /* TODO report errors as strings not numbers */ |
| 396 | propertySize = sizeof adid; |
| 397 | status = AudioHardwareGetProperty(kAudioHardwarePropertyDefaultOutputDevice, |
| 398 | &propertySize, &adid); |
| 399 | if(status) |
| 400 | fatal(0, "AudioHardwareGetProperty: %d", (int)status); |
| 401 | if(adid == kAudioDeviceUnknown) |
| 402 | fatal(0, "no output device"); |
| 403 | propertySize = sizeof asbd; |
| 404 | status = AudioDeviceGetProperty(adid, 0, false, |
| 405 | kAudioDevicePropertyStreamFormat, |
| 406 | &propertySize, &asbd); |
| 407 | if(status) |
| 408 | fatal(0, "AudioHardwareGetProperty: %d", (int)status); |
| 409 | D(("mSampleRate %f", asbd.mSampleRate)); |
| 410 | D(("mFormatID %08"PRIx32, asbd.mFormatID)); |
| 411 | D(("mFormatFlags %08"PRIx32, asbd.mFormatFlags)); |
| 412 | D(("mBytesPerPacket %08"PRIx32, asbd.mBytesPerPacket)); |
| 413 | D(("mFramesPerPacket %08"PRIx32, asbd.mFramesPerPacket)); |
| 414 | D(("mBytesPerFrame %08"PRIx32, asbd.mBytesPerFrame)); |
| 415 | D(("mChannelsPerFrame %08"PRIx32, asbd.mChannelsPerFrame)); |
| 416 | D(("mBitsPerChannel %08"PRIx32, asbd.mBitsPerChannel)); |
| 417 | D(("mReserved %08"PRIx32, asbd.mReserved)); |
| 418 | if(asbd.mFormatID != kAudioFormatLinearPCM) |
| 419 | fatal(0, "audio device does not support kAudioFormatLinearPCM"); |
| 420 | status = AudioDeviceAddIOProc(adid, adioproc, 0); |
| 421 | if(status) |
| 422 | fatal(0, "AudioDeviceAddIOProc: %d", (int)status); |
| 423 | pthread_mutex_lock(&lock); |
| 424 | for(;;) { |
| 425 | /* Wait for the buffer to fill up a bit */ |
| 426 | while(nsamples < READAHEAD) |
| 427 | pthread_cond_wait(&cond, &lock); |
| 428 | /* Start playing now */ |
| 429 | status = AudioDeviceStart(adid, adioproc); |
| 430 | if(status) |
| 431 | fatal(0, "AudioDeviceStart: %d", (int)status); |
| 432 | /* Wait until the buffer empties out */ |
| 433 | while(nsamples >= MINBUFFER) |
| 434 | pthread_cond_wait(&cond, &lock); |
| 435 | /* Stop playing for a bit until the buffer re-fills */ |
| 436 | status = AudioDeviceStop(adid, adioproc); |
| 437 | if(status) |
| 438 | fatal(0, "AudioDeviceStop: %d", (int)status); |
| 439 | /* Go back round */ |
| 440 | } |
| 441 | } |
| 442 | #else |
| 443 | # error No known audio API |
| 444 | #endif |
| 445 | } |
| 446 | |
| 447 | /* display usage message and terminate */ |
| 448 | static void help(void) { |
| 449 | xprintf("Usage:\n" |
| 450 | " disorder-playrtp [OPTIONS] ADDRESS [PORT]\n" |
| 451 | "Options:\n" |
| 452 | " --help, -h Display usage message\n" |
| 453 | " --version, -V Display version number\n" |
| 454 | " --debug, -d Turn on debugging\n" |
| 455 | " --device, -D DEVICE Output device\n"); |
| 456 | xfclose(stdout); |
| 457 | exit(0); |
| 458 | } |
| 459 | |
| 460 | /* display version number and terminate */ |
| 461 | static void version(void) { |
| 462 | xprintf("disorder-playrtp version %s\n", disorder_version_string); |
| 463 | xfclose(stdout); |
| 464 | exit(0); |
| 465 | } |
| 466 | |
| 467 | int main(int argc, char **argv) { |
| 468 | int n; |
| 469 | struct addrinfo *res; |
| 470 | struct stringlist sl; |
| 471 | char *sockname; |
| 472 | |
| 473 | static const struct addrinfo prefs = { |
| 474 | AI_PASSIVE, |
| 475 | PF_INET, |
| 476 | SOCK_DGRAM, |
| 477 | IPPROTO_UDP, |
| 478 | 0, |
| 479 | 0, |
| 480 | 0, |
| 481 | 0 |
| 482 | }; |
| 483 | |
| 484 | mem_init(); |
| 485 | if(!setlocale(LC_CTYPE, "")) fatal(errno, "error calling setlocale"); |
| 486 | while((n = getopt_long(argc, argv, "hVdD", options, 0)) >= 0) { |
| 487 | switch(n) { |
| 488 | case 'h': help(); |
| 489 | case 'V': version(); |
| 490 | case 'd': debugging = 1; break; |
| 491 | case 'D': device = optarg; break; |
| 492 | default: fatal(0, "invalid option"); |
| 493 | } |
| 494 | } |
| 495 | argc -= optind; |
| 496 | argv += optind; |
| 497 | if(argc < 1 || argc > 2) |
| 498 | fatal(0, "usage: disorder-playrtp [OPTIONS] ADDRESS [PORT]"); |
| 499 | sl.n = argc; |
| 500 | sl.s = argv; |
| 501 | /* Listen for inbound audio data */ |
| 502 | if(!(res = get_address(&sl, &prefs, &sockname))) |
| 503 | exit(1); |
| 504 | if((rtpfd = socket(res->ai_family, |
| 505 | res->ai_socktype, |
| 506 | res->ai_protocol)) < 0) |
| 507 | fatal(errno, "error creating socket"); |
| 508 | if(bind(rtpfd, res->ai_addr, res->ai_addrlen) < 0) |
| 509 | fatal(errno, "error binding socket to %s", sockname); |
| 510 | play_rtp(); |
| 511 | return 0; |
| 512 | } |
| 513 | |
| 514 | /* |
| 515 | Local Variables: |
| 516 | c-basic-offset:2 |
| 517 | comment-column:40 |
| 518 | fill-column:79 |
| 519 | indent-tabs-mode:nil |
| 520 | End: |
| 521 | */ |