2 * This file is part of DisOrder.
3 * Copyright (C) 2007 Richard Kettlewell
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.
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.
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
20 /** @file clients/playrtp.c
23 * This player supports Linux (<a href="http://www.alsa-project.org/">ALSA</a>)
25 * href="http://developer.apple.com/audio/coreaudio.html">Core Audio</a>)
26 * systems. There is no support for Microsoft Windows yet, and that will in
27 * fact probably an entirely separate program.
29 * The program runs (at least) two threads. listen_thread() is responsible for
30 * reading RTP packets off the wire and adding them to the binary heap @ref
31 * packets, assuming they are basically sound.
33 * The main thread is responsible for actually playing audio. In ALSA this
34 * means it waits until ALSA says it's ready for more audio which it then
37 * InCore Audio the main thread is only responsible for starting and stopping
38 * play: the system does the actual playback in its own private thread, and
39 * calls adioproc() to fetch the audio data.
41 * Sometimes it happens that there is no audio available to play. This may
42 * because the server went away, or a packet was dropped, or the server
43 * deliberately did not send any sound because it encountered a silence.
52 #include <sys/socket.h>
53 #include <sys/types.h>
54 #include <sys/socket.h>
63 #include "configuration.h"
71 #if HAVE_COREAUDIO_AUDIOHARDWARE_H
72 # include <CoreAudio/AudioHardware.h>
75 #include <alsa/asoundlib.h>
78 #define readahead linux_headers_are_borked
80 /** @brief RTP socket */
83 /** @brief Log output */
86 /** @brief Output device */
87 static const char *device;
89 /** @brief Maximum samples per packet we'll support
91 * NB that two channels = two samples in this program.
93 #define MAXSAMPLES 2048
95 /** @brief Minimum low watermark
97 * We'll stop playing if there's only this many samples in the buffer. */
98 static unsigned minbuffer = 2 * 44100 / 10; /* 0.2 seconds */
100 /** @brief Buffer high watermark
102 * We'll only start playing when this many samples are available. */
103 static unsigned readahead = 2 * 2 * 44100;
105 /** @brief Maximum buffer size
107 * We'll stop reading from the network if we have this many samples. */
108 static unsigned maxbuffer;
110 /** @brief Number of samples to infill by in one go
112 * This is an upper bound - in practice we expect the underlying audio API to
113 * only ask for a much smaller number of samples in any one go.
115 #define INFILL_SAMPLES (44100 * 2) /* 1s */
117 /** @brief Received packet
119 * Received packets are kept in a binary heap (see @ref pheap) ordered by
123 /** @brief Number of samples in this packet */
126 /** @brief Timestamp from RTP packet
128 * NB that "timestamps" are really sample counters. Use lt() or lt_packet()
129 * to compare timestamps.
136 * - @ref IDLE - the idle bit was set in the RTP packet
139 /** @brief idle bit set in RTP packet*/
142 /** @brief Raw sample data
144 * Only the first @p nsamples samples are defined; the rest is uninitialized
147 uint16_t samples_raw[MAXSAMPLES];
150 /** @brief Return true iff \f$a < b\f$ in sequence-space arithmetic
152 * Specifically it returns true if \f$(a-b) mod 2^{32} < 2^{31}\f$.
154 * See also lt_packet().
156 static inline int lt(uint32_t a, uint32_t b) {
157 return (uint32_t)(a - b) & 0x80000000;
160 /** @brief Return true iff a >= b in sequence-space arithmetic */
161 static inline int ge(uint32_t a, uint32_t b) {
165 /** @brief Return true iff a > b in sequence-space arithmetic */
166 static inline int gt(uint32_t a, uint32_t b) {
170 /** @brief Return true iff a <= b in sequence-space arithmetic */
171 static inline int le(uint32_t a, uint32_t b) {
175 /** @brief Ordering for packets, used by @ref pheap */
176 static inline int lt_packet(const struct packet *a, const struct packet *b) {
177 return lt(a->timestamp, b->timestamp);
181 * @brief Binary heap of packets ordered by timestamp */
182 HEAP_TYPE(pheap, struct packet *, lt_packet);
184 /** @brief Binary heap of received packets */
185 static struct pheap packets;
187 /** @brief Total number of samples available */
188 static unsigned long nsamples;
190 /** @brief Timestamp of next packet to play.
192 * This is set to the timestamp of the last packet, plus the number of
193 * samples it contained. Only valid if @ref active is nonzero.
195 static uint32_t next_timestamp;
197 /** @brief True if actively playing
199 * This is true when playing and false when just buffering. */
202 /** @brief Structure of free packet list */
205 union free_packet *next;
208 /** @brief Linked list of free packets
210 * This is a linked list of formerly used packets. For preference we re-use
211 * packets that have already been used rather than unused ones, to limit the
212 * size of the program's working set. If there are no free packets in the list
213 * we try @ref next_free_packet instead.
215 * Must hold @ref lock when accessing this.
217 static union free_packet *free_packets;
219 /** @brief Array of new free packets
221 * There are @ref count_free_packets ready to use at this address. If there
222 * are none left we allocate more memory.
224 * Must hold @ref lock when accessing this.
226 static union free_packet *next_free_packet;
228 /** @brief Count of new free packets at @ref next_free_packet
230 * Must hold @ref lock when accessing this.
232 static size_t count_free_packets;
234 /** @brief Lock protecting @ref packets
236 * This also protects the packet memory allocation infrastructure, @ref
237 * free_packets and @ref next_free_packet. */
238 static pthread_mutex_t lock = PTHREAD_MUTEX_INITIALIZER;
240 /** @brief Condition variable signalled whenever @ref packets is changed */
241 static pthread_cond_t cond = PTHREAD_COND_INITIALIZER;
243 static const struct option options[] = {
244 { "help", no_argument, 0, 'h' },
245 { "version", no_argument, 0, 'V' },
246 { "debug", no_argument, 0, 'd' },
247 { "device", required_argument, 0, 'D' },
248 { "min", required_argument, 0, 'm' },
249 { "max", required_argument, 0, 'x' },
250 { "buffer", required_argument, 0, 'b' },
254 /** @brief Return a new packet
256 * Assumes that @ref lock is held. */
257 static struct packet *new_packet(void) {
261 p = &free_packets->p;
262 free_packets = free_packets->next;
264 if(!count_free_packets) {
265 next_free_packet = xcalloc(1024, sizeof (union free_packet));
266 count_free_packets = 1024;
268 p = &(next_free_packet++)->p;
269 --count_free_packets;
274 /** @brief Free a packet
276 * Assumes that @ref lock is held. */
277 static void free_packet(struct packet *p) {
278 union free_packet *u = (union free_packet *)p;
279 u->next = free_packets;
283 /** @brief Drop the first packet
285 * Assumes that @ref lock is held.
287 static void drop_first_packet(void) {
288 if(pheap_count(&packets)) {
289 struct packet *const p = pheap_remove(&packets);
290 nsamples -= p->nsamples;
292 pthread_cond_broadcast(&cond);
296 /** @brief Background thread collecting samples
298 * This function collects samples, perhaps converts them to the target format,
299 * and adds them to the packet list.
301 * It is crucial that the gap between successive calls to read() is as small as
302 * possible: otherwise packets will be dropped.
304 * We use a binary heap to ensure that the unavoidable effort is at worst
305 * logarithmic in the total number of packets - in fact if packets are mostly
306 * received in order then we will largely do constant work per packet since the
307 * newest packet will always be last.
309 * Of more concern is that we must acquire the lock on the heap to add a packet
310 * to it. If this proves a problem in practice then the answer would be
311 * (probably doubly) linked list with new packets added the end and a second
312 * thread which reads packets off the list and adds them to the heap.
314 * We keep memory allocation (mostly) very fast by keeping pre-allocated
315 * packets around; see @ref new_packet().
317 static void *listen_thread(void attribute((unused)) *arg) {
318 struct packet *p = 0;
320 struct rtp_header header;
327 pthread_mutex_lock(&lock);
329 pthread_mutex_unlock(&lock);
331 iov[0].iov_base = &header;
332 iov[0].iov_len = sizeof header;
333 iov[1].iov_base = p->samples_raw;
334 iov[1].iov_len = sizeof p->samples_raw / sizeof *p->samples_raw;
335 n = readv(rtpfd, iov, 2);
341 fatal(errno, "error reading from socket");
344 /* Ignore too-short packets */
345 if((size_t)n <= sizeof (struct rtp_header)) {
346 info("ignored a short packet");
349 timestamp = htonl(header.timestamp);
350 seq = htons(header.seq);
351 /* Ignore packets in the past */
352 if(active && lt(timestamp, next_timestamp)) {
353 info("dropping old packet, timestamp=%"PRIx32" < %"PRIx32,
354 timestamp, next_timestamp);
357 pthread_mutex_lock(&lock);
359 p->timestamp = timestamp;
360 /* Convert to target format */
361 if(header.mpt & 0x80)
363 switch(header.mpt & 0x7F) {
365 p->nsamples = (n - sizeof header) / sizeof(uint16_t);
367 /* TODO support other RFC3551 media types (when the speaker does) */
369 fatal(0, "unsupported RTP payload type %d",
373 fprintf(logfp, "sequence %u timestamp %"PRIx32" length %"PRIx32" end %"PRIx32"\n",
374 seq, timestamp, p->nsamples, timestamp + p->nsamples);
375 /* Stop reading if we've reached the maximum.
377 * This is rather unsatisfactory: it means that if packets get heavily
378 * out of order then we guarantee dropouts. But for now... */
379 if(nsamples >= maxbuffer) {
380 //info("Buffer full");
382 while(nsamples >= maxbuffer)
383 pthread_cond_wait(&cond, &lock);
385 /* Add the packet to the heap */
386 pheap_insert(&packets, p);
387 nsamples += p->nsamples;
388 /* We'll need a new packet */
390 pthread_cond_broadcast(&cond);
391 pthread_mutex_unlock(&lock);
395 /** @brief Return true if @p p contains @p timestamp
397 * Containment implies that a sample @p timestamp exists within the packet.
399 static inline int contains(const struct packet *p, uint32_t timestamp) {
400 const uint32_t packet_start = p->timestamp;
401 const uint32_t packet_end = p->timestamp + p->nsamples;
403 return (ge(timestamp, packet_start)
404 && lt(timestamp, packet_end));
407 /** @brief Wait until the buffer is adequately full
409 * Must be called with @ref lock held.
411 static void fill_buffer(void) {
412 info("Buffering...");
413 while(nsamples < readahead)
414 pthread_cond_wait(&cond, &lock);
415 next_timestamp = pheap_first(&packets)->timestamp;
419 /** @brief Find next packet
420 * @return Packet to play or NULL if none found
422 * The return packet is merely guaranteed not to be in the past: it might be
423 * the first packet in the future rather than one that is actually suitable to
426 * Must be called with @ref lock held.
428 static struct packet *next_packet(void) {
429 while(pheap_count(&packets)) {
430 struct packet *const p = pheap_first(&packets);
431 if(le(p->timestamp + p->nsamples, next_timestamp)) {
432 /* This packet is in the past. Drop it and try another one. */
435 /* This packet is NOT in the past. (It might be in the future
442 #if HAVE_COREAUDIO_AUDIOHARDWARE_H
443 /** @brief Callback from Core Audio */
444 static OSStatus adioproc
445 (AudioDeviceID attribute((unused)) inDevice,
446 const AudioTimeStamp attribute((unused)) *inNow,
447 const AudioBufferList attribute((unused)) *inInputData,
448 const AudioTimeStamp attribute((unused)) *inInputTime,
449 AudioBufferList *outOutputData,
450 const AudioTimeStamp attribute((unused)) *inOutputTime,
451 void attribute((unused)) *inClientData) {
452 UInt32 nbuffers = outOutputData->mNumberBuffers;
453 AudioBuffer *ab = outOutputData->mBuffers;
454 uint32_t samples_available;
456 pthread_mutex_lock(&lock);
457 while(nbuffers > 0) {
458 float *samplesOut = ab->mData;
459 size_t samplesOutLeft = ab->mDataByteSize / sizeof (float);
461 while(samplesOutLeft > 0) {
462 const struct packet *p = next_packet();
463 if(p && contains(p, next_timestamp)) {
466 /* This packet is ready to play */
467 const uint32_t packet_end = p->timestamp + p->nsamples;
468 const uint32_t offset = next_timestamp - p->timestamp;
469 const uint16_t *ptr = (void *)(p->samples_raw + offset);
471 samples_available = packet_end - next_timestamp;
472 if(samples_available > samplesOutLeft)
473 samples_available = samplesOutLeft;
474 next_timestamp += samples_available;
475 samplesOutLeft -= samples_available;
476 while(samples_available-- > 0)
477 *samplesOut++ = (int16_t)ntohs(*ptr++) * (0.5 / 32767);
478 /* We don't bother junking the packet - that'll be dealt with next time
482 /* No packet is ready to play (and there might be no packet at all) */
483 samples_available = p ? p->timestamp - next_timestamp
485 if(samples_available > samplesOutLeft)
486 samples_available = samplesOutLeft;
487 //info("infill by %"PRIu32, samples_available);
488 /* Conveniently the buffer is 0 to start with */
489 next_timestamp += samples_available;
490 samplesOut += samples_available;
491 samplesOutLeft -= samples_available;
498 pthread_mutex_unlock(&lock);
505 /** @brief PCM handle */
506 static snd_pcm_t *pcm;
508 /** @brief True when @ref pcm is up and running */
509 static int alsa_prepared = 1;
511 /** @brief Initialize @ref pcm */
512 static void setup_alsa(void) {
513 snd_pcm_hw_params_t *hwparams;
514 snd_pcm_sw_params_t *swparams;
515 /* Only support one format for now */
516 const int sample_format = SND_PCM_FORMAT_S16_BE;
517 unsigned rate = 44100;
518 const int channels = 2;
519 const int samplesize = channels * sizeof(uint16_t);
520 snd_pcm_uframes_t pcm_bufsize = MAXSAMPLES * samplesize * 3;
521 /* If we can write more than this many samples we'll get a wakeup */
522 const int avail_min = 256;
526 if((err = snd_pcm_open(&pcm,
527 device ? device : "default",
528 SND_PCM_STREAM_PLAYBACK,
530 fatal(0, "error from snd_pcm_open: %d", err);
531 /* Set up 'hardware' parameters */
532 snd_pcm_hw_params_alloca(&hwparams);
533 if((err = snd_pcm_hw_params_any(pcm, hwparams)) < 0)
534 fatal(0, "error from snd_pcm_hw_params_any: %d", err);
535 if((err = snd_pcm_hw_params_set_access(pcm, hwparams,
536 SND_PCM_ACCESS_RW_INTERLEAVED)) < 0)
537 fatal(0, "error from snd_pcm_hw_params_set_access: %d", err);
538 if((err = snd_pcm_hw_params_set_format(pcm, hwparams,
541 fatal(0, "error from snd_pcm_hw_params_set_format (%d): %d",
543 if((err = snd_pcm_hw_params_set_rate_near(pcm, hwparams, &rate, 0)) < 0)
544 fatal(0, "error from snd_pcm_hw_params_set_rate (%d): %d",
546 if((err = snd_pcm_hw_params_set_channels(pcm, hwparams,
548 fatal(0, "error from snd_pcm_hw_params_set_channels (%d): %d",
550 if((err = snd_pcm_hw_params_set_buffer_size_near(pcm, hwparams,
552 fatal(0, "error from snd_pcm_hw_params_set_buffer_size (%d): %d",
553 MAXSAMPLES * samplesize * 3, err);
554 if((err = snd_pcm_hw_params(pcm, hwparams)) < 0)
555 fatal(0, "error calling snd_pcm_hw_params: %d", err);
556 /* Set up 'software' parameters */
557 snd_pcm_sw_params_alloca(&swparams);
558 if((err = snd_pcm_sw_params_current(pcm, swparams)) < 0)
559 fatal(0, "error calling snd_pcm_sw_params_current: %d", err);
560 if((err = snd_pcm_sw_params_set_avail_min(pcm, swparams, avail_min)) < 0)
561 fatal(0, "error calling snd_pcm_sw_params_set_avail_min %d: %d",
563 if((err = snd_pcm_sw_params(pcm, swparams)) < 0)
564 fatal(0, "error calling snd_pcm_sw_params: %d", err);
567 /** @brief Wait until ALSA wants some audio */
568 static void wait_alsa(void) {
569 struct pollfd fds[64];
571 unsigned short events;
575 if((nfds = snd_pcm_poll_descriptors(pcm,
576 fds, sizeof fds / sizeof *fds)) < 0)
577 fatal(0, "error calling snd_pcm_poll_descriptors: %d", nfds);
578 } while(poll(fds, nfds, -1) < 0 && errno == EINTR);
579 if((err = snd_pcm_poll_descriptors_revents(pcm, fds, nfds, &events)))
580 fatal(0, "error calling snd_pcm_poll_descriptors_revents: %d", err);
586 /** @brief Play some sound via ALSA
587 * @param s Pointer to sample data
588 * @param n Number of samples
589 * @return 0 on success, -1 on non-fatal error
591 static int alsa_writei(const void *s, size_t n) {
593 const snd_pcm_sframes_t frames_written = snd_pcm_writei(pcm, s, n / 2);
594 if(frames_written < 0) {
595 /* Something went wrong */
596 switch(frames_written) {
601 error(0, "error calling snd_pcm_writei: %ld",
602 (long)frames_written);
605 fatal(0, "error calling snd_pcm_writei: %ld",
606 (long)frames_written);
610 next_timestamp += frames_written * 2;
615 /** @brief Play the relevant part of a packet
616 * @param p Packet to play
617 * @return 0 on success, -1 on non-fatal error
619 static int alsa_play(const struct packet *p) {
623 return alsa_writei(p->samples_raw + next_timestamp - p->timestamp,
624 (p->timestamp + p->nsamples) - next_timestamp);
627 /** @brief Play some silence
628 * @param p Next packet or NULL
629 * @return 0 on success, -1 on non-fatal error
631 static int alsa_infill(const struct packet *p) {
632 static const uint16_t zeros[INFILL_SAMPLES];
633 size_t samples_available = INFILL_SAMPLES;
635 if(p && samples_available > p->timestamp - next_timestamp)
636 samples_available = p->timestamp - next_timestamp;
638 return alsa_writei(zeros, samples_available);
641 /** @brief Reset ALSA state after we lost synchronization */
642 static void alsa_reset(int hard_reset) {
645 if((err = snd_pcm_nonblock(pcm, 0)))
646 fatal(0, "error calling snd_pcm_nonblock: %d", err);
648 if((err = snd_pcm_drop(pcm)))
649 fatal(0, "error calling snd_pcm_drop: %d", err);
651 if((err = snd_pcm_drain(pcm)))
652 fatal(0, "error calling snd_pcm_drain: %d", err);
653 if((err = snd_pcm_nonblock(pcm, 1)))
654 fatal(0, "error calling snd_pcm_nonblock: %d", err);
659 /** @brief Play an RTP stream
661 * This is the guts of the program. It is responsible for:
662 * - starting the listening thread
663 * - opening the audio device
664 * - reading ahead to build up a buffer
665 * - arranging for audio to be played
666 * - detecting when the buffer has got too small and re-buffering
668 static void play_rtp(void) {
671 /* We receive and convert audio data in a background thread */
672 pthread_create(<id, 0, listen_thread, 0);
678 /* Open the sound device */
680 pthread_mutex_lock(&lock);
682 /* Wait for the buffer to fill up a bit */
685 if((err = snd_pcm_prepare(pcm)))
686 fatal(0, "error calling snd_pcm_prepare: %d", err);
691 /* Keep playing until the buffer empties out, or ALSA tells us to get
693 while(nsamples >= minbuffer && !escape) {
694 /* Wait for ALSA to ask us for more data */
695 pthread_mutex_unlock(&lock);
697 pthread_mutex_lock(&lock);
698 /* ALSA is ready for more data, find something to play */
700 /* Play it or play some silence */
701 if(contains(p, next_timestamp))
702 escape = alsa_play(p);
704 escape = alsa_infill(p);
707 /* We stop playing for a bit until the buffer re-fills */
708 pthread_mutex_unlock(&lock);
710 pthread_mutex_lock(&lock);
714 #elif HAVE_COREAUDIO_AUDIOHARDWARE_H
719 AudioStreamBasicDescription asbd;
721 /* If this looks suspiciously like libao's macosx driver there's an
722 * excellent reason for that... */
724 /* TODO report errors as strings not numbers */
725 propertySize = sizeof adid;
726 status = AudioHardwareGetProperty(kAudioHardwarePropertyDefaultOutputDevice,
727 &propertySize, &adid);
729 fatal(0, "AudioHardwareGetProperty: %d", (int)status);
730 if(adid == kAudioDeviceUnknown)
731 fatal(0, "no output device");
732 propertySize = sizeof asbd;
733 status = AudioDeviceGetProperty(adid, 0, false,
734 kAudioDevicePropertyStreamFormat,
735 &propertySize, &asbd);
737 fatal(0, "AudioHardwareGetProperty: %d", (int)status);
738 D(("mSampleRate %f", asbd.mSampleRate));
739 D(("mFormatID %08lx", asbd.mFormatID));
740 D(("mFormatFlags %08lx", asbd.mFormatFlags));
741 D(("mBytesPerPacket %08lx", asbd.mBytesPerPacket));
742 D(("mFramesPerPacket %08lx", asbd.mFramesPerPacket));
743 D(("mBytesPerFrame %08lx", asbd.mBytesPerFrame));
744 D(("mChannelsPerFrame %08lx", asbd.mChannelsPerFrame));
745 D(("mBitsPerChannel %08lx", asbd.mBitsPerChannel));
746 D(("mReserved %08lx", asbd.mReserved));
747 if(asbd.mFormatID != kAudioFormatLinearPCM)
748 fatal(0, "audio device does not support kAudioFormatLinearPCM");
749 status = AudioDeviceAddIOProc(adid, adioproc, 0);
751 fatal(0, "AudioDeviceAddIOProc: %d", (int)status);
752 pthread_mutex_lock(&lock);
754 /* Wait for the buffer to fill up a bit */
756 /* Start playing now */
758 next_timestamp = pheap_first(&packets)->timestamp;
760 status = AudioDeviceStart(adid, adioproc);
762 fatal(0, "AudioDeviceStart: %d", (int)status);
763 /* Wait until the buffer empties out */
764 while(nsamples >= minbuffer)
765 pthread_cond_wait(&cond, &lock);
766 /* Stop playing for a bit until the buffer re-fills */
767 status = AudioDeviceStop(adid, adioproc);
769 fatal(0, "AudioDeviceStop: %d", (int)status);
775 # error No known audio API
779 /* display usage message and terminate */
780 static void help(void) {
782 " disorder-playrtp [OPTIONS] ADDRESS [PORT]\n"
784 " --device, -D DEVICE Output device\n"
785 " --min, -m FRAMES Buffer low water mark\n"
786 " --buffer, -b FRAMES Buffer high water mark\n"
787 " --max, -x FRAMES Buffer maximum size\n"
788 " --help, -h Display usage message\n"
789 " --version, -V Display version number\n"
795 /* display version number and terminate */
796 static void version(void) {
797 xprintf("disorder-playrtp version %s\n", disorder_version_string);
802 int main(int argc, char **argv) {
804 struct addrinfo *res;
805 struct stringlist sl;
808 static const struct addrinfo prefs = {
820 if(!setlocale(LC_CTYPE, "")) fatal(errno, "error calling setlocale");
821 while((n = getopt_long(argc, argv, "hVdD:m:b:x:L:", options, 0)) >= 0) {
825 case 'd': debugging = 1; break;
826 case 'D': device = optarg; break;
827 case 'm': minbuffer = 2 * atol(optarg); break;
828 case 'b': readahead = 2 * atol(optarg); break;
829 case 'x': maxbuffer = 2 * atol(optarg); break;
830 case 'L': logfp = fopen(optarg, "w"); break;
831 default: fatal(0, "invalid option");
835 maxbuffer = 4 * readahead;
838 if(argc < 1 || argc > 2)
839 fatal(0, "usage: disorder-playrtp [OPTIONS] ADDRESS [PORT]");
842 /* Listen for inbound audio data */
843 if(!(res = get_address(&sl, &prefs, &sockname)))
845 if((rtpfd = socket(res->ai_family,
847 res->ai_protocol)) < 0)
848 fatal(errno, "error creating socket");
849 if(bind(rtpfd, res->ai_addr, res->ai_addrlen) < 0)
850 fatal(errno, "error binding socket to %s", sockname);