chiark / gitweb /
Further ALSA underrun fiddling.
[disorder] / clients / playrtp.c
CommitLineData
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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 */
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20/** @file clients/playrtp.c
21 * @brief RTP player
22 *
b0fdc63d 23 * This player supports Linux (<a href="http://www.alsa-project.org/">ALSA</a>)
24 * and Apple Mac (<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.
28 *
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29 * The program runs (at least) three threads. listen_thread() is responsible
30 * for reading RTP packets off the wire and adding them to the linked list @ref
31 * received_packets, assuming they are basically sound. queue_thread() takes
32 * packets off this linked list and adds them to @ref packets (an operation
33 * which might be much slower due to contention for @ref lock).
b0fdc63d 34 *
35 * The main thread is responsible for actually playing audio. In ALSA this
36 * means it waits until ALSA says it's ready for more audio which it then
c593cf7c 37 * plays. See @ref clients/playrtp-alsa.c.
b0fdc63d 38 *
8e3fe3d8 39 * In Core Audio the main thread is only responsible for starting and stopping
b0fdc63d 40 * play: the system does the actual playback in its own private thread, and
c593cf7c 41 * calls adioproc() to fetch the audio data. See @ref
42 * clients/playrtp-coreaudio.c.
b0fdc63d 43 *
44 * Sometimes it happens that there is no audio available to play. This may
45 * because the server went away, or a packet was dropped, or the server
46 * deliberately did not send any sound because it encountered a silence.
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47 *
48 * Assumptions:
49 * - it is safe to read uint32_t values without a lock protecting them
28bacdc0 50 */
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51
52#include <config.h>
53#include "types.h"
54
55#include <getopt.h>
56#include <stdio.h>
57#include <stdlib.h>
58#include <sys/socket.h>
59#include <sys/types.h>
60#include <sys/socket.h>
61#include <netdb.h>
62#include <pthread.h>
0b75463f 63#include <locale.h>
2c7c9eae 64#include <sys/uio.h>
28bacdc0 65#include <string.h>
8e3fe3d8 66#include <assert.h>
c593cf7c 67#include <errno.h>
e3426f7b 68#include <netinet/in.h>
2d2effe2 69#include <sys/time.h>
a99c4e9a 70#include <sys/un.h>
9fbe0996 71#include <unistd.h>
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72#include <sys/mman.h>
73#include <fcntl.h>
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74
75#include "log.h"
76#include "mem.h"
77#include "configuration.h"
78#include "addr.h"
79#include "syscalls.h"
80#include "rtp.h"
0b75463f 81#include "defs.h"
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82#include "vector.h"
83#include "heap.h"
189e9830 84#include "timeval.h"
a7e9570a 85#include "client.h"
8e3fe3d8 86#include "playrtp.h"
a99c4e9a 87#include "inputline.h"
3fbdc96d 88#include "version.h"
e83d0967 89
1153fd23 90#define readahead linux_headers_are_borked
91
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92/** @brief Obsolete synonym */
93#ifndef IPV6_JOIN_GROUP
94# define IPV6_JOIN_GROUP IPV6_ADD_MEMBERSHIP
95#endif
96
0b75463f 97/** @brief RTP socket */
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98static int rtpfd;
99
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100/** @brief Log output */
101static FILE *logfp;
102
0b75463f 103/** @brief Output device */
8e3fe3d8 104const char *device;
0b75463f 105
9086a105 106/** @brief Minimum low watermark
0b75463f 107 *
108 * We'll stop playing if there's only this many samples in the buffer. */
c593cf7c 109unsigned minbuffer = 2 * 44100 / 10; /* 0.2 seconds */
0b75463f 110
9086a105 111/** @brief Buffer high watermark
1153fd23 112 *
113 * We'll only start playing when this many samples are available. */
8d0c14d7 114static unsigned readahead = 2 * 2 * 44100;
0b75463f 115
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116/** @brief Maximum buffer size
117 *
118 * We'll stop reading from the network if we have this many samples. */
119static unsigned maxbuffer;
120
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121/** @brief Received packets
122 * Protected by @ref receive_lock
123 *
124 * Received packets are added to this list, and queue_thread() picks them off
125 * it and adds them to @ref packets. Whenever a packet is added to it, @ref
126 * receive_cond is signalled.
127 */
8e3fe3d8 128struct packet *received_packets;
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129
130/** @brief Tail of @ref received_packets
131 * Protected by @ref receive_lock
132 */
8e3fe3d8 133struct packet **received_tail = &received_packets;
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134
135/** @brief Lock protecting @ref received_packets
136 *
137 * Only listen_thread() and queue_thread() ever hold this lock. It is vital
138 * that queue_thread() not hold it any longer than it strictly has to. */
8e3fe3d8 139pthread_mutex_t receive_lock = PTHREAD_MUTEX_INITIALIZER;
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140
141/** @brief Condition variable signalled when @ref received_packets is updated
142 *
143 * Used by listen_thread() to notify queue_thread() that it has added another
144 * packet to @ref received_packets. */
8e3fe3d8 145pthread_cond_t receive_cond = PTHREAD_COND_INITIALIZER;
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146
147/** @brief Length of @ref received_packets */
8e3fe3d8 148uint32_t nreceived;
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149
150/** @brief Binary heap of received packets */
8e3fe3d8 151struct pheap packets;
28bacdc0 152
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153/** @brief Total number of samples available
154 *
155 * We make this volatile because we inspect it without a protecting lock,
156 * so the usual pthread_* guarantees aren't available.
157 */
8e3fe3d8 158volatile uint32_t nsamples;
0b75463f 159
160/** @brief Timestamp of next packet to play.
161 *
162 * This is set to the timestamp of the last packet, plus the number of
09ee2f0d 163 * samples it contained. Only valid if @ref active is nonzero.
0b75463f 164 */
8e3fe3d8 165uint32_t next_timestamp;
e83d0967 166
09ee2f0d 167/** @brief True if actively playing
168 *
169 * This is true when playing and false when just buffering. */
8e3fe3d8 170int active;
09ee2f0d 171
189e9830 172/** @brief Lock protecting @ref packets */
8e3fe3d8 173pthread_mutex_t lock = PTHREAD_MUTEX_INITIALIZER;
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174
175/** @brief Condition variable signalled whenever @ref packets is changed */
8e3fe3d8 176pthread_cond_t cond = PTHREAD_COND_INITIALIZER;
2c7c9eae 177
146e86fb 178#if HAVE_ALSA_ASOUNDLIB_H
c593cf7c 179# define DEFAULT_BACKEND playrtp_alsa
a9f0ad12 180#elif HAVE_SYS_SOUNDCARD_H || EMPEG_HOST
c593cf7c 181# define DEFAULT_BACKEND playrtp_oss
182#elif HAVE_COREAUDIO_AUDIOHARDWARE_H
183# define DEFAULT_BACKEND playrtp_coreaudio
184#else
185# error No known backend
186#endif
187
188/** @brief Backend to play with */
189static void (*backend)(void) = &DEFAULT_BACKEND;
190
8e3fe3d8 191HEAP_DEFINE(pheap, struct packet *, lt_packet);
e83d0967 192
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193/** @brief Control socket or NULL */
194const char *control_socket;
195
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196/** @brief Buffer for debugging dump
197 *
198 * The debug dump is enabled by the @c --dump option. It records the last 20s
199 * of audio to the specified file (which will be about 3.5Mbytes). The file is
200 * written as as ring buffer, so the start point will progress through it.
201 *
202 * Use clients/dump2wav to convert this to a WAV file, which can then be loaded
203 * into (e.g.) Audacity for further inspection.
204 *
205 * All three backends (ALSA, OSS, Core Audio) now support this option.
206 *
207 * The idea is to allow the user a few seconds to react to an audible artefact.
208 */
e9b635a3 209int16_t *dump_buffer;
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210
211/** @brief Current index within debugging dump */
e9b635a3 212size_t dump_index;
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213
214/** @brief Size of debugging dump in samples */
215size_t dump_size = 44100/*Hz*/ * 2/*channels*/ * 20/*seconds*/;
e9b635a3 216
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217static const struct option options[] = {
218 { "help", no_argument, 0, 'h' },
219 { "version", no_argument, 0, 'V' },
220 { "debug", no_argument, 0, 'd' },
0b75463f 221 { "device", required_argument, 0, 'D' },
1153fd23 222 { "min", required_argument, 0, 'm' },
9086a105 223 { "max", required_argument, 0, 'x' },
1153fd23 224 { "buffer", required_argument, 0, 'b' },
1f10f780 225 { "rcvbuf", required_argument, 0, 'R' },
a9f0ad12 226#if HAVE_SYS_SOUNDCARD_H || EMPEG_HOST
c593cf7c 227 { "oss", no_argument, 0, 'o' },
228#endif
146e86fb 229#if HAVE_ALSA_ASOUNDLIB_H
c593cf7c 230 { "alsa", no_argument, 0, 'a' },
231#endif
232#if HAVE_COREAUDIO_AUDIOHARDWARE_H
233 { "core-audio", no_argument, 0, 'c' },
234#endif
e9b635a3 235 { "dump", required_argument, 0, 'r' },
a99c4e9a 236 { "socket", required_argument, 0, 's' },
a7e9570a 237 { "config", required_argument, 0, 'C' },
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238 { 0, 0, 0, 0 }
239};
240
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241/** @brief Control thread
242 *
243 * This thread is responsible for accepting control commands from Disobedience
244 * (or other controllers) over an AF_UNIX stream socket with a path specified
245 * by the @c --socket option. The protocol uses simple string commands and
246 * replies:
247 *
248 * - @c stop will shut the player down
249 * - @c query will send back the reply @c running
250 * - anything else is ignored
251 *
252 * Commands and response strings terminated by shutting down the connection or
253 * by a newline. No attempt is made to multiplex multiple clients so it is
254 * important that the command be sent as soon as the connection is made - it is
255 * assumed that both parties to the protocol are entirely cooperating with one
256 * another.
257 */
258static void *control_thread(void attribute((unused)) *arg) {
259 struct sockaddr_un sa;
260 int sfd, cfd;
261 char *line;
262 socklen_t salen;
263 FILE *fp;
264
265 assert(control_socket);
266 unlink(control_socket);
267 memset(&sa, 0, sizeof sa);
268 sa.sun_family = AF_UNIX;
269 strcpy(sa.sun_path, control_socket);
270 sfd = xsocket(PF_UNIX, SOCK_STREAM, 0);
271 if(bind(sfd, (const struct sockaddr *)&sa, sizeof sa) < 0)
272 fatal(errno, "error binding to %s", control_socket);
273 if(listen(sfd, 128) < 0)
274 fatal(errno, "error calling listen on %s", control_socket);
275 info("listening on %s", control_socket);
276 for(;;) {
277 salen = sizeof sa;
278 cfd = accept(sfd, (struct sockaddr *)&sa, &salen);
279 if(cfd < 0) {
280 switch(errno) {
281 case EINTR:
282 case EAGAIN:
283 break;
284 default:
285 fatal(errno, "error calling accept on %s", control_socket);
286 }
287 }
288 if(!(fp = fdopen(cfd, "r+"))) {
289 error(errno, "error calling fdopen for %s connection", control_socket);
290 close(cfd);
291 continue;
292 }
293 if(!inputline(control_socket, fp, &line, '\n')) {
294 if(!strcmp(line, "stop")) {
295 info("stopped via %s", control_socket);
296 exit(0); /* terminate immediately */
297 }
298 if(!strcmp(line, "query"))
299 fprintf(fp, "running");
300 xfree(line);
301 }
302 if(fclose(fp) < 0)
303 error(errno, "error closing %s connection", control_socket);
304 }
305}
306
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307/** @brief Drop the first packet
308 *
309 * Assumes that @ref lock is held.
310 */
311static void drop_first_packet(void) {
312 if(pheap_count(&packets)) {
313 struct packet *const p = pheap_remove(&packets);
314 nsamples -= p->nsamples;
c593cf7c 315 playrtp_free_packet(p);
2c7c9eae 316 pthread_cond_broadcast(&cond);
2c7c9eae 317 }
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318}
319
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320/** @brief Background thread adding packets to heap
321 *
322 * This just transfers packets from @ref received_packets to @ref packets. It
323 * is important that it holds @ref receive_lock for as little time as possible,
324 * in order to minimize the interval between calls to read() in
325 * listen_thread().
326 */
327static void *queue_thread(void attribute((unused)) *arg) {
328 struct packet *p;
329
330 for(;;) {
331 /* Get the next packet */
332 pthread_mutex_lock(&receive_lock);
333 while(!received_packets)
334 pthread_cond_wait(&receive_cond, &receive_lock);
335 p = received_packets;
336 received_packets = p->next;
337 if(!received_packets)
338 received_tail = &received_packets;
339 --nreceived;
340 pthread_mutex_unlock(&receive_lock);
341 /* Add it to the heap */
342 pthread_mutex_lock(&lock);
343 pheap_insert(&packets, p);
344 nsamples += p->nsamples;
345 pthread_cond_broadcast(&cond);
346 pthread_mutex_unlock(&lock);
347 }
348}
349
09ee2f0d 350/** @brief Background thread collecting samples
0b75463f 351 *
352 * This function collects samples, perhaps converts them to the target format,
b0fdc63d 353 * and adds them to the packet list.
354 *
355 * It is crucial that the gap between successive calls to read() is as small as
356 * possible: otherwise packets will be dropped.
357 *
358 * We use a binary heap to ensure that the unavoidable effort is at worst
359 * logarithmic in the total number of packets - in fact if packets are mostly
360 * received in order then we will largely do constant work per packet since the
361 * newest packet will always be last.
362 *
363 * Of more concern is that we must acquire the lock on the heap to add a packet
364 * to it. If this proves a problem in practice then the answer would be
365 * (probably doubly) linked list with new packets added the end and a second
366 * thread which reads packets off the list and adds them to the heap.
367 *
368 * We keep memory allocation (mostly) very fast by keeping pre-allocated
c593cf7c 369 * packets around; see @ref playrtp_new_packet().
b0fdc63d 370 */
0b75463f 371static void *listen_thread(void attribute((unused)) *arg) {
2c7c9eae 372 struct packet *p = 0;
0b75463f 373 int n;
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374 struct rtp_header header;
375 uint16_t seq;
376 uint32_t timestamp;
377 struct iovec iov[2];
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378
379 for(;;) {
189e9830 380 if(!p)
c593cf7c 381 p = playrtp_new_packet();
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382 iov[0].iov_base = &header;
383 iov[0].iov_len = sizeof header;
384 iov[1].iov_base = p->samples_raw;
b64efe7e 385 iov[1].iov_len = sizeof p->samples_raw / sizeof *p->samples_raw;
2c7c9eae 386 n = readv(rtpfd, iov, 2);
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387 if(n < 0) {
388 switch(errno) {
389 case EINTR:
390 continue;
391 default:
392 fatal(errno, "error reading from socket");
393 }
394 }
0b75463f 395 /* Ignore too-short packets */
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396 if((size_t)n <= sizeof (struct rtp_header)) {
397 info("ignored a short packet");
0b75463f 398 continue;
345ebe66 399 }
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400 timestamp = htonl(header.timestamp);
401 seq = htons(header.seq);
09ee2f0d 402 /* Ignore packets in the past */
2c7c9eae 403 if(active && lt(timestamp, next_timestamp)) {
c0e41690 404 info("dropping old packet, timestamp=%"PRIx32" < %"PRIx32,
2c7c9eae 405 timestamp, next_timestamp);
09ee2f0d 406 continue;
c0e41690 407 }
189e9830 408 p->next = 0;
58b5a68f 409 p->flags = 0;
2c7c9eae 410 p->timestamp = timestamp;
e83d0967 411 /* Convert to target format */
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412 if(header.mpt & 0x80)
413 p->flags |= IDLE;
2c7c9eae 414 switch(header.mpt & 0x7F) {
e83d0967 415 case 10:
2c7c9eae 416 p->nsamples = (n - sizeof header) / sizeof(uint16_t);
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417 break;
418 /* TODO support other RFC3551 media types (when the speaker does) */
419 default:
0b75463f 420 fatal(0, "unsupported RTP payload type %d",
2c7c9eae 421 header.mpt & 0x7F);
e83d0967 422 }
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423 if(logfp)
424 fprintf(logfp, "sequence %u timestamp %"PRIx32" length %"PRIx32" end %"PRIx32"\n",
2c7c9eae 425 seq, timestamp, p->nsamples, timestamp + p->nsamples);
0b75463f 426 /* Stop reading if we've reached the maximum.
427 *
428 * This is rather unsatisfactory: it means that if packets get heavily
429 * out of order then we guarantee dropouts. But for now... */
345ebe66 430 if(nsamples >= maxbuffer) {
189e9830 431 pthread_mutex_lock(&lock);
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432 while(nsamples >= maxbuffer)
433 pthread_cond_wait(&cond, &lock);
189e9830 434 pthread_mutex_unlock(&lock);
345ebe66 435 }
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436 /* Add the packet to the receive queue */
437 pthread_mutex_lock(&receive_lock);
438 *received_tail = p;
439 received_tail = &p->next;
440 ++nreceived;
441 pthread_cond_signal(&receive_cond);
442 pthread_mutex_unlock(&receive_lock);
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443 /* We'll need a new packet */
444 p = 0;
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445 }
446}
447
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448/** @brief Wait until the buffer is adequately full
449 *
450 * Must be called with @ref lock held.
451 */
c593cf7c 452void playrtp_fill_buffer(void) {
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453 while(nsamples)
454 drop_first_packet();
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455 info("Buffering...");
456 while(nsamples < readahead)
457 pthread_cond_wait(&cond, &lock);
458 next_timestamp = pheap_first(&packets)->timestamp;
459 active = 1;
460}
461
462/** @brief Find next packet
463 * @return Packet to play or NULL if none found
464 *
465 * The return packet is merely guaranteed not to be in the past: it might be
466 * the first packet in the future rather than one that is actually suitable to
467 * play.
468 *
469 * Must be called with @ref lock held.
470 */
c593cf7c 471struct packet *playrtp_next_packet(void) {
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472 while(pheap_count(&packets)) {
473 struct packet *const p = pheap_first(&packets);
474 if(le(p->timestamp + p->nsamples, next_timestamp)) {
475 /* This packet is in the past. Drop it and try another one. */
476 drop_first_packet();
477 } else
478 /* This packet is NOT in the past. (It might be in the future
479 * however.) */
480 return p;
481 }
482 return 0;
483}
484
09ee2f0d 485/** @brief Play an RTP stream
486 *
487 * This is the guts of the program. It is responsible for:
488 * - starting the listening thread
489 * - opening the audio device
490 * - reading ahead to build up a buffer
491 * - arranging for audio to be played
492 * - detecting when the buffer has got too small and re-buffering
493 */
0b75463f 494static void play_rtp(void) {
495 pthread_t ltid;
a99c4e9a 496 int err;
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497
498 /* We receive and convert audio data in a background thread */
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499 if((err = pthread_create(&ltid, 0, listen_thread, 0)))
500 fatal(err, "pthread_create listen_thread");
189e9830 501 /* We have a second thread to add received packets to the queue */
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502 if((err = pthread_create(&ltid, 0, queue_thread, 0)))
503 fatal(err, "pthread_create queue_thread");
c593cf7c 504 /* The rest of the work is backend-specific */
505 backend();
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506}
507
508/* display usage message and terminate */
509static void help(void) {
510 xprintf("Usage:\n"
511 " disorder-playrtp [OPTIONS] ADDRESS [PORT]\n"
512 "Options:\n"
1153fd23 513 " --device, -D DEVICE Output device\n"
514 " --min, -m FRAMES Buffer low water mark\n"
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515 " --buffer, -b FRAMES Buffer high water mark\n"
516 " --max, -x FRAMES Buffer maximum size\n"
1f10f780 517 " --rcvbuf, -R BYTES Socket receive buffer size\n"
a7e9570a 518 " --config, -C PATH Set configuration file\n"
146e86fb 519#if HAVE_ALSA_ASOUNDLIB_H
c593cf7c 520 " --alsa, -a Use ALSA to play audio\n"
521#endif
a9f0ad12 522#if HAVE_SYS_SOUNDCARD_H || EMPEG_HOST
c593cf7c 523 " --oss, -o Use OSS to play audio\n"
524#endif
525#if HAVE_COREAUDIO_AUDIOHARDWARE_H
526 " --core-audio, -c Use Core Audio to play audio\n"
527#endif
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528 " --help, -h Display usage message\n"
529 " --version, -V Display version number\n"
530 );
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531 xfclose(stdout);
532 exit(0);
533}
534
e83d0967 535int main(int argc, char **argv) {
a99c4e9a 536 int n, err;
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537 struct addrinfo *res;
538 struct stringlist sl;
0b75463f 539 char *sockname;
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540 int rcvbuf, target_rcvbuf = 131072;
541 socklen_t len;
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542 struct ip_mreq mreq;
543 struct ipv6_mreq mreq6;
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544 disorder_client *c;
545 char *address, *port;
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546 int is_multicast;
547 union any_sockaddr {
548 struct sockaddr sa;
549 struct sockaddr_in in;
550 struct sockaddr_in6 in6;
551 };
552 union any_sockaddr mgroup;
e9b635a3 553 const char *dumpfile = 0;
e83d0967 554
0b75463f 555 static const struct addrinfo prefs = {
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556 AI_PASSIVE,
557 PF_INET,
558 SOCK_DGRAM,
559 IPPROTO_UDP,
560 0,
561 0,
562 0,
563 0
564 };
565
566 mem_init();
567 if(!setlocale(LC_CTYPE, "")) fatal(errno, "error calling setlocale");
e9b635a3 568 while((n = getopt_long(argc, argv, "hVdD:m:b:x:L:R:M:aocC:r", options, 0)) >= 0) {
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569 switch(n) {
570 case 'h': help();
3fbdc96d 571 case 'V': version("disorder-playrtp");
e83d0967 572 case 'd': debugging = 1; break;
0b75463f 573 case 'D': device = optarg; break;
1153fd23 574 case 'm': minbuffer = 2 * atol(optarg); break;
575 case 'b': readahead = 2 * atol(optarg); break;
9086a105 576 case 'x': maxbuffer = 2 * atol(optarg); break;
345ebe66 577 case 'L': logfp = fopen(optarg, "w"); break;
1f10f780 578 case 'R': target_rcvbuf = atoi(optarg); break;
146e86fb 579#if HAVE_ALSA_ASOUNDLIB_H
c593cf7c 580 case 'a': backend = playrtp_alsa; break;
581#endif
a9f0ad12 582#if HAVE_SYS_SOUNDCARD_H || EMPEG_HOST
c593cf7c 583 case 'o': backend = playrtp_oss; break;
584#endif
585#if HAVE_COREAUDIO_AUDIOHARDWARE_H
586 case 'c': backend = playrtp_coreaudio; break;
587#endif
a7e9570a 588 case 'C': configfile = optarg; break;
a99c4e9a 589 case 's': control_socket = optarg; break;
e9b635a3 590 case 'r': dumpfile = optarg; break;
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591 default: fatal(0, "invalid option");
592 }
593 }
a7e9570a 594 if(config_read(0)) fatal(0, "cannot read configuration");
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595 if(!maxbuffer)
596 maxbuffer = 4 * readahead;
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597 argc -= optind;
598 argv += optind;
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599 switch(argc) {
600 case 0:
6fba990c 601 /* Get configuration from server */
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602 if(!(c = disorder_new(1))) exit(EXIT_FAILURE);
603 if(disorder_connect(c)) exit(EXIT_FAILURE);
604 if(disorder_rtp_address(c, &address, &port)) exit(EXIT_FAILURE);
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605 sl.n = 2;
606 sl.s = xcalloc(2, sizeof *sl.s);
607 sl.s[0] = address;
608 sl.s[1] = port;
a7e9570a 609 break;
6fba990c 610 case 1:
a7e9570a 611 case 2:
6fba990c 612 /* Use command-line ADDRESS+PORT or just PORT */
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613 sl.n = argc;
614 sl.s = argv;
615 break;
616 default:
6fba990c 617 fatal(0, "usage: disorder-playrtp [OPTIONS] [[ADDRESS] PORT]");
a7e9570a 618 }
6fba990c 619 /* Look up address and port */
0b75463f 620 if(!(res = get_address(&sl, &prefs, &sockname)))
e83d0967 621 exit(1);
6fba990c 622 /* Create the socket */
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623 if((rtpfd = socket(res->ai_family,
624 res->ai_socktype,
625 res->ai_protocol)) < 0)
626 fatal(errno, "error creating socket");
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627 /* Stash the multicast group address */
628 if((is_multicast = multicast(res->ai_addr))) {
629 memcpy(&mgroup, res->ai_addr, res->ai_addrlen);
630 switch(res->ai_addr->sa_family) {
631 case AF_INET:
632 mgroup.in.sin_port = 0;
633 break;
634 case AF_INET6:
635 mgroup.in6.sin6_port = 0;
636 break;
637 }
638 }
639 /* Bind to 0/port */
640 switch(res->ai_addr->sa_family) {
641 case AF_INET:
642 memset(&((struct sockaddr_in *)res->ai_addr)->sin_addr, 0,
643 sizeof (struct in_addr));
644 break;
645 case AF_INET6:
646 memset(&((struct sockaddr_in6 *)res->ai_addr)->sin6_addr, 0,
647 sizeof (struct in6_addr));
648 break;
649 default:
650 fatal(0, "unsupported family %d", (int)res->ai_addr->sa_family);
651 }
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652 if(bind(rtpfd, res->ai_addr, res->ai_addrlen) < 0)
653 fatal(errno, "error binding socket to %s", sockname);
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654 if(is_multicast) {
655 switch(mgroup.sa.sa_family) {
23205f9c 656 case PF_INET:
6fba990c 657 mreq.imr_multiaddr = mgroup.in.sin_addr;
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658 mreq.imr_interface.s_addr = 0; /* use primary interface */
659 if(setsockopt(rtpfd, IPPROTO_IP, IP_ADD_MEMBERSHIP,
660 &mreq, sizeof mreq) < 0)
661 fatal(errno, "error calling setsockopt IP_ADD_MEMBERSHIP");
662 break;
663 case PF_INET6:
6fba990c 664 mreq6.ipv6mr_multiaddr = mgroup.in6.sin6_addr;
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665 memset(&mreq6.ipv6mr_interface, 0, sizeof mreq6.ipv6mr_interface);
666 if(setsockopt(rtpfd, IPPROTO_IPV6, IPV6_JOIN_GROUP,
667 &mreq6, sizeof mreq6) < 0)
668 fatal(errno, "error calling setsockopt IPV6_JOIN_GROUP");
669 break;
670 default:
671 fatal(0, "unsupported address family %d", res->ai_family);
672 }
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673 info("listening on %s multicast group %s",
674 format_sockaddr(res->ai_addr), format_sockaddr(&mgroup.sa));
675 } else
676 info("listening on %s", format_sockaddr(res->ai_addr));
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677 len = sizeof rcvbuf;
678 if(getsockopt(rtpfd, SOL_SOCKET, SO_RCVBUF, &rcvbuf, &len) < 0)
679 fatal(errno, "error calling getsockopt SO_RCVBUF");
f0bae611 680 if(target_rcvbuf > rcvbuf) {
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681 if(setsockopt(rtpfd, SOL_SOCKET, SO_RCVBUF,
682 &target_rcvbuf, sizeof target_rcvbuf) < 0)
683 error(errno, "error calling setsockopt SO_RCVBUF %d",
684 target_rcvbuf);
685 /* We try to carry on anyway */
686 else
687 info("changed socket receive buffer from %d to %d",
688 rcvbuf, target_rcvbuf);
689 } else
690 info("default socket receive buffer %d", rcvbuf);
691 if(logfp)
692 info("WARNING: -L option can impact performance");
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693 if(control_socket) {
694 pthread_t tid;
695
696 if((err = pthread_create(&tid, 0, control_thread, 0)))
697 fatal(err, "pthread_create control_thread");
698 }
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699 if(dumpfile) {
700 int fd;
701 unsigned char buffer[65536];
702 size_t written;
703
704 if((fd = open(dumpfile, O_RDWR|O_TRUNC|O_CREAT, 0666)) < 0)
705 fatal(errno, "opening %s", dumpfile);
706 /* Fill with 0s to a suitable size */
707 memset(buffer, 0, sizeof buffer);
708 for(written = 0; written < dump_size * sizeof(int16_t);
709 written += sizeof buffer) {
710 if(write(fd, buffer, sizeof buffer) < 0)
711 fatal(errno, "clearing %s", dumpfile);
712 }
713 /* Map the buffer into memory for convenience */
714 dump_buffer = mmap(0, dump_size * sizeof(int16_t), PROT_READ|PROT_WRITE,
715 MAP_SHARED, fd, 0);
716 if(dump_buffer == (void *)-1)
717 fatal(errno, "mapping %s", dumpfile);
718 info("dumping to %s", dumpfile);
719 }
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720 play_rtp();
721 return 0;
722}
723
724/*
725Local Variables:
726c-basic-offset:2
727comment-column:40
728fill-column:79
729indent-tabs-mode:nil
730End:
731*/