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
) {
381 while(nsamples
>= maxbuffer
)
382 pthread_cond_wait(&cond
, &lock
);
384 /* Add the packet to the heap */
385 pheap_insert(&packets
, p
);
386 nsamples
+= p
->nsamples
;
387 /* We'll need a new packet */
389 pthread_cond_broadcast(&cond
);
390 pthread_mutex_unlock(&lock
);
394 /** @brief Return true if @p p contains @p timestamp
396 * Containment implies that a sample @p timestamp exists within the packet.
398 static inline int contains(const struct packet
*p
, uint32_t timestamp
) {
399 const uint32_t packet_start
= p
->timestamp
;
400 const uint32_t packet_end
= p
->timestamp
+ p
->nsamples
;
402 return (ge(timestamp
, packet_start
)
403 && lt(timestamp
, packet_end
));
406 /** @brief Wait until the buffer is adequately full
408 * Must be called with @ref lock held.
410 static void fill_buffer(void) {
411 info("Buffering...");
412 while(nsamples
< readahead
)
413 pthread_cond_wait(&cond
, &lock
);
414 next_timestamp
= pheap_first(&packets
)->timestamp
;
418 /** @brief Find next packet
419 * @return Packet to play or NULL if none found
421 * The return packet is merely guaranteed not to be in the past: it might be
422 * the first packet in the future rather than one that is actually suitable to
425 * Must be called with @ref lock held.
427 static struct packet
*next_packet(void) {
428 while(pheap_count(&packets
)) {
429 struct packet
*const p
= pheap_first(&packets
);
430 if(le(p
->timestamp
+ p
->nsamples
, next_timestamp
)) {
431 /* This packet is in the past. Drop it and try another one. */
434 /* This packet is NOT in the past. (It might be in the future
441 #if HAVE_COREAUDIO_AUDIOHARDWARE_H
442 /** @brief Callback from Core Audio */
443 static OSStatus adioproc
444 (AudioDeviceID
attribute((unused
)) inDevice
,
445 const AudioTimeStamp
attribute((unused
)) *inNow
,
446 const AudioBufferList
attribute((unused
)) *inInputData
,
447 const AudioTimeStamp
attribute((unused
)) *inInputTime
,
448 AudioBufferList
*outOutputData
,
449 const AudioTimeStamp
attribute((unused
)) *inOutputTime
,
450 void attribute((unused
)) *inClientData
) {
451 UInt32 nbuffers
= outOutputData
->mNumberBuffers
;
452 AudioBuffer
*ab
= outOutputData
->mBuffers
;
453 uint32_t samples_available
;
455 pthread_mutex_lock(&lock
);
456 while(nbuffers
> 0) {
457 float *samplesOut
= ab
->mData
;
458 size_t samplesOutLeft
= ab
->mDataByteSize
/ sizeof (float);
460 while(samplesOutLeft
> 0) {
461 const struct packet
*p
= next_packet();
462 if(p
&& contains(p
, next_timestamp
)) {
465 /* This packet is ready to play */
466 const uint32_t packet_end
= p
->timestamp
+ p
->nsamples
;
467 const uint32_t offset
= next_timestamp
- p
->timestamp
;
468 const uint16_t *ptr
= (void *)(p
->samples_raw
+ offset
);
470 samples_available
= packet_end
- next_timestamp
;
471 if(samples_available
> samplesOutLeft
)
472 samples_available
= samplesOutLeft
;
473 next_timestamp
+= samples_available
;
474 samplesOutLeft
-= samples_available
;
475 while(samples_available
-- > 0)
476 *samplesOut
++ = (int16_t)ntohs(*ptr
++) * (0.5 / 32767);
477 /* We don't bother junking the packet - that'll be dealt with next time
481 /* No packet is ready to play (and there might be no packet at all) */
482 samples_available
= p ? p
->timestamp
- next_timestamp
484 if(samples_available
> samplesOutLeft
)
485 samples_available
= samplesOutLeft
;
486 //info("infill by %"PRIu32, samples_available);
487 /* Conveniently the buffer is 0 to start with */
488 next_timestamp
+= samples_available
;
489 samplesOut
+= samples_available
;
490 samplesOutLeft
-= samples_available
;
497 pthread_mutex_unlock(&lock
);
504 /** @brief PCM handle */
505 static snd_pcm_t
*pcm
;
507 /** @brief True when @ref pcm is up and running */
508 static int alsa_prepared
= 1;
510 /** @brief Initialize @ref pcm */
511 static void setup_alsa(void) {
512 snd_pcm_hw_params_t
*hwparams
;
513 snd_pcm_sw_params_t
*swparams
;
514 /* Only support one format for now */
515 const int sample_format
= SND_PCM_FORMAT_S16_BE
;
516 unsigned rate
= 44100;
517 const int channels
= 2;
518 const int samplesize
= channels
* sizeof(uint16_t);
519 snd_pcm_uframes_t pcm_bufsize
= MAXSAMPLES
* samplesize
* 3;
520 /* If we can write more than this many samples we'll get a wakeup */
521 const int avail_min
= 256;
525 if((err
= snd_pcm_open(&pcm
,
526 device ? device
: "default",
527 SND_PCM_STREAM_PLAYBACK
,
529 fatal(0, "error from snd_pcm_open: %d", err
);
530 /* Set up 'hardware' parameters */
531 snd_pcm_hw_params_alloca(&hwparams
);
532 if((err
= snd_pcm_hw_params_any(pcm
, hwparams
)) < 0)
533 fatal(0, "error from snd_pcm_hw_params_any: %d", err
);
534 if((err
= snd_pcm_hw_params_set_access(pcm
, hwparams
,
535 SND_PCM_ACCESS_RW_INTERLEAVED
)) < 0)
536 fatal(0, "error from snd_pcm_hw_params_set_access: %d", err
);
537 if((err
= snd_pcm_hw_params_set_format(pcm
, hwparams
,
540 fatal(0, "error from snd_pcm_hw_params_set_format (%d): %d",
542 if((err
= snd_pcm_hw_params_set_rate_near(pcm
, hwparams
, &rate
, 0)) < 0)
543 fatal(0, "error from snd_pcm_hw_params_set_rate (%d): %d",
545 if((err
= snd_pcm_hw_params_set_channels(pcm
, hwparams
,
547 fatal(0, "error from snd_pcm_hw_params_set_channels (%d): %d",
549 if((err
= snd_pcm_hw_params_set_buffer_size_near(pcm
, hwparams
,
551 fatal(0, "error from snd_pcm_hw_params_set_buffer_size (%d): %d",
552 MAXSAMPLES
* samplesize
* 3, err
);
553 if((err
= snd_pcm_hw_params(pcm
, hwparams
)) < 0)
554 fatal(0, "error calling snd_pcm_hw_params: %d", err
);
555 /* Set up 'software' parameters */
556 snd_pcm_sw_params_alloca(&swparams
);
557 if((err
= snd_pcm_sw_params_current(pcm
, swparams
)) < 0)
558 fatal(0, "error calling snd_pcm_sw_params_current: %d", err
);
559 if((err
= snd_pcm_sw_params_set_avail_min(pcm
, swparams
, avail_min
)) < 0)
560 fatal(0, "error calling snd_pcm_sw_params_set_avail_min %d: %d",
562 if((err
= snd_pcm_sw_params(pcm
, swparams
)) < 0)
563 fatal(0, "error calling snd_pcm_sw_params: %d", err
);
566 /** @brief Wait until ALSA wants some audio */
567 static void wait_alsa(void) {
568 struct pollfd fds
[64];
570 unsigned short events
;
574 if((nfds
= snd_pcm_poll_descriptors(pcm
,
575 fds
, sizeof fds
/ sizeof *fds
)) < 0)
576 fatal(0, "error calling snd_pcm_poll_descriptors: %d", nfds
);
577 } while(poll(fds
, nfds
, -1) < 0 && errno
== EINTR
);
578 if((err
= snd_pcm_poll_descriptors_revents(pcm
, fds
, nfds
, &events
)))
579 fatal(0, "error calling snd_pcm_poll_descriptors_revents: %d", err
);
585 /** @brief Play some sound via ALSA
586 * @param s Pointer to sample data
587 * @param n Number of samples
588 * @return 0 on success, -1 on non-fatal error
590 static int alsa_writei(const void *s
, size_t n
) {
592 const snd_pcm_sframes_t frames_written
= snd_pcm_writei(pcm
, s
, n
/ 2);
593 if(frames_written
< 0) {
594 /* Something went wrong */
595 switch(frames_written
) {
600 error(0, "error calling snd_pcm_writei: %ld",
601 (long)frames_written
);
604 fatal(0, "error calling snd_pcm_writei: %ld",
605 (long)frames_written
);
609 next_timestamp
+= frames_written
* 2;
614 /** @brief Play the relevant part of a packet
615 * @param p Packet to play
616 * @return 0 on success, -1 on non-fatal error
618 static int alsa_play(const struct packet
*p
) {
622 return alsa_writei(p
->samples_raw
+ next_timestamp
- p
->timestamp
,
623 (p
->timestamp
+ p
->nsamples
) - next_timestamp
);
626 /** @brief Play some silence
627 * @param p Next packet or NULL
628 * @return 0 on success, -1 on non-fatal error
630 static int alsa_infill(const struct packet
*p
) {
631 static const uint16_t zeros
[INFILL_SAMPLES
];
632 size_t samples_available
= INFILL_SAMPLES
;
634 if(p
&& samples_available
> p
->timestamp
- next_timestamp
)
635 samples_available
= p
->timestamp
- next_timestamp
;
637 return alsa_writei(zeros
, samples_available
);
640 /** @brief Reset ALSA state after we lost synchronization */
641 static void alsa_reset(int hard_reset
) {
644 if((err
= snd_pcm_nonblock(pcm
, 0)))
645 fatal(0, "error calling snd_pcm_nonblock: %d", err
);
647 if((err
= snd_pcm_drop(pcm
)))
648 fatal(0, "error calling snd_pcm_drop: %d", err
);
650 if((err
= snd_pcm_drain(pcm
)))
651 fatal(0, "error calling snd_pcm_drain: %d", err
);
652 if((err
= snd_pcm_nonblock(pcm
, 1)))
653 fatal(0, "error calling snd_pcm_nonblock: %d", err
);
658 /** @brief Play an RTP stream
660 * This is the guts of the program. It is responsible for:
661 * - starting the listening thread
662 * - opening the audio device
663 * - reading ahead to build up a buffer
664 * - arranging for audio to be played
665 * - detecting when the buffer has got too small and re-buffering
667 static void play_rtp(void) {
670 /* We receive and convert audio data in a background thread */
671 pthread_create(<id
, 0, listen_thread
, 0);
677 /* Open the sound device */
679 pthread_mutex_lock(&lock
);
681 /* Wait for the buffer to fill up a bit */
684 if((err
= snd_pcm_prepare(pcm
)))
685 fatal(0, "error calling snd_pcm_prepare: %d", err
);
690 /* Keep playing until the buffer empties out, or ALSA tells us to get
692 while(nsamples
>= minbuffer
&& !escape
) {
693 /* Wait for ALSA to ask us for more data */
694 pthread_mutex_unlock(&lock
);
696 pthread_mutex_lock(&lock
);
697 /* ALSA is ready for more data, find something to play */
699 /* Play it or play some silence */
700 if(contains(p
, next_timestamp
))
701 escape
= alsa_play(p
);
703 escape
= alsa_infill(p
);
706 /* We stop playing for a bit until the buffer re-fills */
707 pthread_mutex_unlock(&lock
);
709 pthread_mutex_lock(&lock
);
713 #elif HAVE_COREAUDIO_AUDIOHARDWARE_H
718 AudioStreamBasicDescription asbd
;
720 /* If this looks suspiciously like libao's macosx driver there's an
721 * excellent reason for that... */
723 /* TODO report errors as strings not numbers */
724 propertySize
= sizeof adid
;
725 status
= AudioHardwareGetProperty(kAudioHardwarePropertyDefaultOutputDevice
,
726 &propertySize
, &adid
);
728 fatal(0, "AudioHardwareGetProperty: %d", (int)status
);
729 if(adid
== kAudioDeviceUnknown
)
730 fatal(0, "no output device");
731 propertySize
= sizeof asbd
;
732 status
= AudioDeviceGetProperty(adid
, 0, false,
733 kAudioDevicePropertyStreamFormat
,
734 &propertySize
, &asbd
);
736 fatal(0, "AudioHardwareGetProperty: %d", (int)status
);
737 D(("mSampleRate %f", asbd
.mSampleRate
));
738 D(("mFormatID %08lx", asbd
.mFormatID
));
739 D(("mFormatFlags %08lx", asbd
.mFormatFlags
));
740 D(("mBytesPerPacket %08lx", asbd
.mBytesPerPacket
));
741 D(("mFramesPerPacket %08lx", asbd
.mFramesPerPacket
));
742 D(("mBytesPerFrame %08lx", asbd
.mBytesPerFrame
));
743 D(("mChannelsPerFrame %08lx", asbd
.mChannelsPerFrame
));
744 D(("mBitsPerChannel %08lx", asbd
.mBitsPerChannel
));
745 D(("mReserved %08lx", asbd
.mReserved
));
746 if(asbd
.mFormatID
!= kAudioFormatLinearPCM
)
747 fatal(0, "audio device does not support kAudioFormatLinearPCM");
748 status
= AudioDeviceAddIOProc(adid
, adioproc
, 0);
750 fatal(0, "AudioDeviceAddIOProc: %d", (int)status
);
751 pthread_mutex_lock(&lock
);
753 /* Wait for the buffer to fill up a bit */
755 /* Start playing now */
757 next_timestamp
= pheap_first(&packets
)->timestamp
;
759 status
= AudioDeviceStart(adid
, adioproc
);
761 fatal(0, "AudioDeviceStart: %d", (int)status
);
762 /* Wait until the buffer empties out */
763 while(nsamples
>= minbuffer
)
764 pthread_cond_wait(&cond
, &lock
);
765 /* Stop playing for a bit until the buffer re-fills */
766 status
= AudioDeviceStop(adid
, adioproc
);
768 fatal(0, "AudioDeviceStop: %d", (int)status
);
774 # error No known audio API
778 /* display usage message and terminate */
779 static void help(void) {
781 " disorder-playrtp [OPTIONS] ADDRESS [PORT]\n"
783 " --device, -D DEVICE Output device\n"
784 " --min, -m FRAMES Buffer low water mark\n"
785 " --buffer, -b FRAMES Buffer high water mark\n"
786 " --max, -x FRAMES Buffer maximum size\n"
787 " --help, -h Display usage message\n"
788 " --version, -V Display version number\n"
794 /* display version number and terminate */
795 static void version(void) {
796 xprintf("disorder-playrtp version %s\n", disorder_version_string
);
801 int main(int argc
, char **argv
) {
803 struct addrinfo
*res
;
804 struct stringlist sl
;
807 static const struct addrinfo prefs
= {
819 if(!setlocale(LC_CTYPE
, "")) fatal(errno
, "error calling setlocale");
820 while((n
= getopt_long(argc
, argv
, "hVdD:m:b:x:L:", options
, 0)) >= 0) {
824 case 'd': debugging
= 1; break;
825 case 'D': device
= optarg
; break;
826 case 'm': minbuffer
= 2 * atol(optarg
); break;
827 case 'b': readahead
= 2 * atol(optarg
); break;
828 case 'x': maxbuffer
= 2 * atol(optarg
); break;
829 case 'L': logfp
= fopen(optarg
, "w"); break;
830 default: fatal(0, "invalid option");
834 maxbuffer
= 4 * readahead
;
837 if(argc
< 1 || argc
> 2)
838 fatal(0, "usage: disorder-playrtp [OPTIONS] ADDRESS [PORT]");
841 /* Listen for inbound audio data */
842 if(!(res
= get_address(&sl
, &prefs
, &sockname
)))
844 if((rtpfd
= socket(res
->ai_family
,
846 res
->ai_protocol
)) < 0)
847 fatal(errno
, "error creating socket");
848 if(bind(rtpfd
, res
->ai_addr
, res
->ai_addrlen
) < 0)
849 fatal(errno
, "error binding socket to %s", sockname
);