chiark / gitweb /
Merge from disorder.dev.
[disorder] / clients / playrtp.c
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1/*
2 * This file is part of DisOrder.
4778e044 3 * Copyright (C) 2007-2009 Richard Kettlewell
e83d0967 4 *
e7eb3a27 5 * This program is free software: you can redistribute it and/or modify
e83d0967 6 * it under the terms of the GNU General Public License as published by
e7eb3a27 7 * the Free Software Foundation, either version 3 of the License, or
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8 * (at your option) any later version.
9 *
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10 * This program is distributed in the hope that it will be useful,
11 * but WITHOUT ANY WARRANTY; without even the implied warranty of
12 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
13 * GNU General Public License for more details.
14 *
e83d0967 15 * You should have received a copy of the GNU General Public License
e7eb3a27 16 * along with this program. If not, see <http://www.gnu.org/licenses/>.
e83d0967 17 */
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18/** @file clients/playrtp.c
19 * @brief RTP player
20 *
b0fdc63d 21 * This player supports Linux (<a href="http://www.alsa-project.org/">ALSA</a>)
22 * and Apple Mac (<a
23 * href="http://developer.apple.com/audio/coreaudio.html">Core Audio</a>)
24 * systems. There is no support for Microsoft Windows yet, and that will in
25 * fact probably an entirely separate program.
26 *
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27 * The program runs (at least) three threads:
28 *
29 * listen_thread() is responsible for reading RTP packets off the wire and
30 * adding them to the linked list @ref received_packets, assuming they are
31 * basically sound.
32 *
33 * queue_thread() takes packets off this linked list and adds them to @ref
34 * packets (an operation which might be much slower due to contention for @ref
35 * lock).
36 *
37 * control_thread() accepts commands from Disobedience (or anything else).
38 *
39 * The main thread activates and deactivates audio playing via the @ref
40 * lib/uaudio.h API (which probably implies at least one further thread).
b0fdc63d 41 *
42 * Sometimes it happens that there is no audio available to play. This may
43 * because the server went away, or a packet was dropped, or the server
44 * deliberately did not send any sound because it encountered a silence.
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45 *
46 * Assumptions:
47 * - it is safe to read uint32_t values without a lock protecting them
28bacdc0 48 */
e83d0967 49
05b75f8d 50#include "common.h"
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51
52#include <getopt.h>
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53#include <sys/socket.h>
54#include <sys/types.h>
55#include <sys/socket.h>
56#include <netdb.h>
57#include <pthread.h>
0b75463f 58#include <locale.h>
2c7c9eae 59#include <sys/uio.h>
c593cf7c 60#include <errno.h>
e3426f7b 61#include <netinet/in.h>
2d2effe2 62#include <sys/time.h>
a99c4e9a 63#include <sys/un.h>
9fbe0996 64#include <unistd.h>
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65#include <sys/mman.h>
66#include <fcntl.h>
b0619501 67#include <math.h>
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68
69#include "log.h"
70#include "mem.h"
71#include "configuration.h"
72#include "addr.h"
73#include "syscalls.h"
74#include "rtp.h"
0b75463f 75#include "defs.h"
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76#include "vector.h"
77#include "heap.h"
189e9830 78#include "timeval.h"
a7e9570a 79#include "client.h"
8e3fe3d8 80#include "playrtp.h"
a99c4e9a 81#include "inputline.h"
3fbdc96d 82#include "version.h"
7a2c7068 83#include "uaudio.h"
e83d0967 84
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85/** @brief Obsolete synonym */
86#ifndef IPV6_JOIN_GROUP
87# define IPV6_JOIN_GROUP IPV6_ADD_MEMBERSHIP
88#endif
89
0b75463f 90/** @brief RTP socket */
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91static int rtpfd;
92
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93/** @brief Log output */
94static FILE *logfp;
95
0b75463f 96/** @brief Output device */
0b75463f 97
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98/** @brief Buffer low watermark in samples */
99unsigned minbuffer = 4 * (2 * 44100) / 10; /* 0.4 seconds */
0b75463f 100
ad535598 101/** @brief Maximum buffer size in samples
9086a105 102 *
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103 * We'll stop reading from the network if we have this many samples.
104 */
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105static unsigned maxbuffer;
106
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107/** @brief Received packets
108 * Protected by @ref receive_lock
109 *
110 * Received packets are added to this list, and queue_thread() picks them off
111 * it and adds them to @ref packets. Whenever a packet is added to it, @ref
112 * receive_cond is signalled.
113 */
8e3fe3d8 114struct packet *received_packets;
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115
116/** @brief Tail of @ref received_packets
117 * Protected by @ref receive_lock
118 */
8e3fe3d8 119struct packet **received_tail = &received_packets;
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120
121/** @brief Lock protecting @ref received_packets
122 *
123 * Only listen_thread() and queue_thread() ever hold this lock. It is vital
124 * that queue_thread() not hold it any longer than it strictly has to. */
8e3fe3d8 125pthread_mutex_t receive_lock = PTHREAD_MUTEX_INITIALIZER;
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126
127/** @brief Condition variable signalled when @ref received_packets is updated
128 *
129 * Used by listen_thread() to notify queue_thread() that it has added another
130 * packet to @ref received_packets. */
8e3fe3d8 131pthread_cond_t receive_cond = PTHREAD_COND_INITIALIZER;
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132
133/** @brief Length of @ref received_packets */
8e3fe3d8 134uint32_t nreceived;
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135
136/** @brief Binary heap of received packets */
8e3fe3d8 137struct pheap packets;
28bacdc0 138
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139/** @brief Total number of samples available
140 *
141 * We make this volatile because we inspect it without a protecting lock,
142 * so the usual pthread_* guarantees aren't available.
143 */
8e3fe3d8 144volatile uint32_t nsamples;
0b75463f 145
146/** @brief Timestamp of next packet to play.
147 *
148 * This is set to the timestamp of the last packet, plus the number of
09ee2f0d 149 * samples it contained. Only valid if @ref active is nonzero.
0b75463f 150 */
8e3fe3d8 151uint32_t next_timestamp;
e83d0967 152
09ee2f0d 153/** @brief True if actively playing
154 *
155 * This is true when playing and false when just buffering. */
8e3fe3d8 156int active;
09ee2f0d 157
189e9830 158/** @brief Lock protecting @ref packets */
8e3fe3d8 159pthread_mutex_t lock = PTHREAD_MUTEX_INITIALIZER;
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160
161/** @brief Condition variable signalled whenever @ref packets is changed */
8e3fe3d8 162pthread_cond_t cond = PTHREAD_COND_INITIALIZER;
2c7c9eae 163
c593cf7c 164/** @brief Backend to play with */
7a2c7068 165static const struct uaudio *backend;
c593cf7c 166
8e3fe3d8 167HEAP_DEFINE(pheap, struct packet *, lt_packet);
e83d0967 168
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169/** @brief Control socket or NULL */
170const char *control_socket;
171
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172/** @brief Buffer for debugging dump
173 *
174 * The debug dump is enabled by the @c --dump option. It records the last 20s
175 * of audio to the specified file (which will be about 3.5Mbytes). The file is
176 * written as as ring buffer, so the start point will progress through it.
177 *
178 * Use clients/dump2wav to convert this to a WAV file, which can then be loaded
179 * into (e.g.) Audacity for further inspection.
180 *
181 * All three backends (ALSA, OSS, Core Audio) now support this option.
182 *
183 * The idea is to allow the user a few seconds to react to an audible artefact.
184 */
e9b635a3 185int16_t *dump_buffer;
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186
187/** @brief Current index within debugging dump */
e9b635a3 188size_t dump_index;
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189
190/** @brief Size of debugging dump in samples */
191size_t dump_size = 44100/*Hz*/ * 2/*channels*/ * 20/*seconds*/;
e9b635a3 192
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193static const struct option options[] = {
194 { "help", no_argument, 0, 'h' },
195 { "version", no_argument, 0, 'V' },
196 { "debug", no_argument, 0, 'd' },
0b75463f 197 { "device", required_argument, 0, 'D' },
1153fd23 198 { "min", required_argument, 0, 'm' },
9086a105 199 { "max", required_argument, 0, 'x' },
1f10f780 200 { "rcvbuf", required_argument, 0, 'R' },
a9f0ad12 201#if HAVE_SYS_SOUNDCARD_H || EMPEG_HOST
c593cf7c 202 { "oss", no_argument, 0, 'o' },
203#endif
146e86fb 204#if HAVE_ALSA_ASOUNDLIB_H
c593cf7c 205 { "alsa", no_argument, 0, 'a' },
206#endif
207#if HAVE_COREAUDIO_AUDIOHARDWARE_H
208 { "core-audio", no_argument, 0, 'c' },
209#endif
e9b635a3 210 { "dump", required_argument, 0, 'r' },
e979b844 211 { "command", required_argument, 0, 'e' },
287ad384 212 { "pause-mode", required_argument, 0, 'P' },
a99c4e9a 213 { "socket", required_argument, 0, 's' },
a7e9570a 214 { "config", required_argument, 0, 'C' },
b0619501 215 { "monitor", no_argument, 0, 'M' },
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216 { 0, 0, 0, 0 }
217};
218
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219/** @brief Control thread
220 *
221 * This thread is responsible for accepting control commands from Disobedience
222 * (or other controllers) over an AF_UNIX stream socket with a path specified
223 * by the @c --socket option. The protocol uses simple string commands and
224 * replies:
225 *
226 * - @c stop will shut the player down
227 * - @c query will send back the reply @c running
228 * - anything else is ignored
229 *
230 * Commands and response strings terminated by shutting down the connection or
231 * by a newline. No attempt is made to multiplex multiple clients so it is
232 * important that the command be sent as soon as the connection is made - it is
233 * assumed that both parties to the protocol are entirely cooperating with one
234 * another.
235 */
236static void *control_thread(void attribute((unused)) *arg) {
237 struct sockaddr_un sa;
238 int sfd, cfd;
239 char *line;
240 socklen_t salen;
241 FILE *fp;
242
243 assert(control_socket);
244 unlink(control_socket);
245 memset(&sa, 0, sizeof sa);
246 sa.sun_family = AF_UNIX;
247 strcpy(sa.sun_path, control_socket);
248 sfd = xsocket(PF_UNIX, SOCK_STREAM, 0);
249 if(bind(sfd, (const struct sockaddr *)&sa, sizeof sa) < 0)
250 fatal(errno, "error binding to %s", control_socket);
251 if(listen(sfd, 128) < 0)
252 fatal(errno, "error calling listen on %s", control_socket);
253 info("listening on %s", control_socket);
254 for(;;) {
255 salen = sizeof sa;
256 cfd = accept(sfd, (struct sockaddr *)&sa, &salen);
257 if(cfd < 0) {
258 switch(errno) {
259 case EINTR:
260 case EAGAIN:
261 break;
262 default:
263 fatal(errno, "error calling accept on %s", control_socket);
264 }
265 }
266 if(!(fp = fdopen(cfd, "r+"))) {
267 error(errno, "error calling fdopen for %s connection", control_socket);
268 close(cfd);
269 continue;
270 }
271 if(!inputline(control_socket, fp, &line, '\n')) {
272 if(!strcmp(line, "stop")) {
273 info("stopped via %s", control_socket);
274 exit(0); /* terminate immediately */
275 }
276 if(!strcmp(line, "query"))
277 fprintf(fp, "running");
278 xfree(line);
279 }
280 if(fclose(fp) < 0)
281 error(errno, "error closing %s connection", control_socket);
282 }
283}
284
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285/** @brief Drop the first packet
286 *
287 * Assumes that @ref lock is held.
288 */
289static void drop_first_packet(void) {
290 if(pheap_count(&packets)) {
291 struct packet *const p = pheap_remove(&packets);
292 nsamples -= p->nsamples;
c593cf7c 293 playrtp_free_packet(p);
2c7c9eae 294 pthread_cond_broadcast(&cond);
2c7c9eae 295 }
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296}
297
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298/** @brief Background thread adding packets to heap
299 *
300 * This just transfers packets from @ref received_packets to @ref packets. It
301 * is important that it holds @ref receive_lock for as little time as possible,
302 * in order to minimize the interval between calls to read() in
303 * listen_thread().
304 */
305static void *queue_thread(void attribute((unused)) *arg) {
306 struct packet *p;
307
308 for(;;) {
309 /* Get the next packet */
310 pthread_mutex_lock(&receive_lock);
4dadf1a2 311 while(!received_packets) {
189e9830 312 pthread_cond_wait(&receive_cond, &receive_lock);
4dadf1a2 313 }
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314 p = received_packets;
315 received_packets = p->next;
316 if(!received_packets)
317 received_tail = &received_packets;
318 --nreceived;
319 pthread_mutex_unlock(&receive_lock);
320 /* Add it to the heap */
321 pthread_mutex_lock(&lock);
322 pheap_insert(&packets, p);
323 nsamples += p->nsamples;
324 pthread_cond_broadcast(&cond);
325 pthread_mutex_unlock(&lock);
326 }
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327#if HAVE_STUPID_GCC44
328 return NULL;
329#endif
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330}
331
09ee2f0d 332/** @brief Background thread collecting samples
0b75463f 333 *
334 * This function collects samples, perhaps converts them to the target format,
b0fdc63d 335 * and adds them to the packet list.
336 *
337 * It is crucial that the gap between successive calls to read() is as small as
338 * possible: otherwise packets will be dropped.
339 *
340 * We use a binary heap to ensure that the unavoidable effort is at worst
341 * logarithmic in the total number of packets - in fact if packets are mostly
342 * received in order then we will largely do constant work per packet since the
343 * newest packet will always be last.
344 *
345 * Of more concern is that we must acquire the lock on the heap to add a packet
346 * to it. If this proves a problem in practice then the answer would be
347 * (probably doubly) linked list with new packets added the end and a second
348 * thread which reads packets off the list and adds them to the heap.
349 *
350 * We keep memory allocation (mostly) very fast by keeping pre-allocated
c593cf7c 351 * packets around; see @ref playrtp_new_packet().
b0fdc63d 352 */
0b75463f 353static void *listen_thread(void attribute((unused)) *arg) {
2c7c9eae 354 struct packet *p = 0;
0b75463f 355 int n;
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356 struct rtp_header header;
357 uint16_t seq;
358 uint32_t timestamp;
359 struct iovec iov[2];
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360
361 for(;;) {
189e9830 362 if(!p)
c593cf7c 363 p = playrtp_new_packet();
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364 iov[0].iov_base = &header;
365 iov[0].iov_len = sizeof header;
366 iov[1].iov_base = p->samples_raw;
b64efe7e 367 iov[1].iov_len = sizeof p->samples_raw / sizeof *p->samples_raw;
2c7c9eae 368 n = readv(rtpfd, iov, 2);
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369 if(n < 0) {
370 switch(errno) {
371 case EINTR:
372 continue;
373 default:
374 fatal(errno, "error reading from socket");
375 }
376 }
0b75463f 377 /* Ignore too-short packets */
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378 if((size_t)n <= sizeof (struct rtp_header)) {
379 info("ignored a short packet");
0b75463f 380 continue;
345ebe66 381 }
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382 timestamp = htonl(header.timestamp);
383 seq = htons(header.seq);
09ee2f0d 384 /* Ignore packets in the past */
2c7c9eae 385 if(active && lt(timestamp, next_timestamp)) {
c0e41690 386 info("dropping old packet, timestamp=%"PRIx32" < %"PRIx32,
2c7c9eae 387 timestamp, next_timestamp);
09ee2f0d 388 continue;
c0e41690 389 }
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390 /* Ignore packets with the extension bit set. */
391 if(header.vpxcc & 0x10)
392 continue;
189e9830 393 p->next = 0;
58b5a68f 394 p->flags = 0;
2c7c9eae 395 p->timestamp = timestamp;
e83d0967 396 /* Convert to target format */
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397 if(header.mpt & 0x80)
398 p->flags |= IDLE;
2c7c9eae 399 switch(header.mpt & 0x7F) {
4fd38868 400 case 10: /* L16 */
2c7c9eae 401 p->nsamples = (n - sizeof header) / sizeof(uint16_t);
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402 break;
403 /* TODO support other RFC3551 media types (when the speaker does) */
404 default:
0b75463f 405 fatal(0, "unsupported RTP payload type %d",
2c7c9eae 406 header.mpt & 0x7F);
e83d0967 407 }
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408 /* See if packet is silent */
409 const uint16_t *s = p->samples_raw;
7bdf42d0 410 n = p->nsamples;
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411 for(; n > 0; --n)
412 if(*s++)
413 break;
414 if(!n)
415 p->flags |= SILENT;
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416 if(logfp)
417 fprintf(logfp, "sequence %u timestamp %"PRIx32" length %"PRIx32" end %"PRIx32"\n",
2c7c9eae 418 seq, timestamp, p->nsamples, timestamp + p->nsamples);
0b75463f 419 /* Stop reading if we've reached the maximum.
420 *
421 * This is rather unsatisfactory: it means that if packets get heavily
422 * out of order then we guarantee dropouts. But for now... */
345ebe66 423 if(nsamples >= maxbuffer) {
189e9830 424 pthread_mutex_lock(&lock);
4dadf1a2 425 while(nsamples >= maxbuffer) {
345ebe66 426 pthread_cond_wait(&cond, &lock);
4dadf1a2 427 }
189e9830 428 pthread_mutex_unlock(&lock);
345ebe66 429 }
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430 /* Add the packet to the receive queue */
431 pthread_mutex_lock(&receive_lock);
432 *received_tail = p;
433 received_tail = &p->next;
434 ++nreceived;
435 pthread_cond_signal(&receive_cond);
436 pthread_mutex_unlock(&receive_lock);
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437 /* We'll need a new packet */
438 p = 0;
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439 }
440}
441
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442/** @brief Wait until the buffer is adequately full
443 *
444 * Must be called with @ref lock held.
445 */
c593cf7c 446void playrtp_fill_buffer(void) {
0e72bf84 447 /* Discard current buffer contents */
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448 while(nsamples) {
449 //fprintf(stderr, "%8u/%u (%u) DROPPING\n", nsamples, maxbuffer, minbuffer);
bfd27c14 450 drop_first_packet();
ad535598 451 }
5626f6d2 452 info("Buffering...");
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453 /* Wait until there's at least minbuffer samples available */
454 while(nsamples < minbuffer) {
ad535598 455 //fprintf(stderr, "%8u/%u (%u) FILLING\n", nsamples, maxbuffer, minbuffer);
5626f6d2 456 pthread_cond_wait(&cond, &lock);
4dadf1a2 457 }
0e72bf84 458 /* Start from whatever is earliest */
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459 next_timestamp = pheap_first(&packets)->timestamp;
460 active = 1;
461}
462
463/** @brief Find next packet
464 * @return Packet to play or NULL if none found
465 *
466 * The return packet is merely guaranteed not to be in the past: it might be
467 * the first packet in the future rather than one that is actually suitable to
468 * play.
469 *
470 * Must be called with @ref lock held.
471 */
c593cf7c 472struct packet *playrtp_next_packet(void) {
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473 while(pheap_count(&packets)) {
474 struct packet *const p = pheap_first(&packets);
475 if(le(p->timestamp + p->nsamples, next_timestamp)) {
476 /* This packet is in the past. Drop it and try another one. */
477 drop_first_packet();
478 } else
479 /* This packet is NOT in the past. (It might be in the future
480 * however.) */
481 return p;
482 }
483 return 0;
484}
485
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486/* display usage message and terminate */
487static void help(void) {
488 xprintf("Usage:\n"
c897bb65 489 " disorder-playrtp [OPTIONS] [[ADDRESS] PORT]\n"
e83d0967 490 "Options:\n"
1153fd23 491 " --device, -D DEVICE Output device\n"
492 " --min, -m FRAMES Buffer low water mark\n"
9086a105 493 " --max, -x FRAMES Buffer maximum size\n"
1f10f780 494 " --rcvbuf, -R BYTES Socket receive buffer size\n"
a7e9570a 495 " --config, -C PATH Set configuration file\n"
146e86fb 496#if HAVE_ALSA_ASOUNDLIB_H
c593cf7c 497 " --alsa, -a Use ALSA to play audio\n"
498#endif
a9f0ad12 499#if HAVE_SYS_SOUNDCARD_H || EMPEG_HOST
c593cf7c 500 " --oss, -o Use OSS to play audio\n"
501#endif
502#if HAVE_COREAUDIO_AUDIOHARDWARE_H
503 " --core-audio, -c Use Core Audio to play audio\n"
504#endif
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505 " --command, -e COMMAND Pipe audio to command.\n"
506 " --pause-mode, -P silence For -e: pauses send silence (default)\n"
507 " --pause-mode, -P suspend For -e: pauses suspend writes\n"
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508 " --help, -h Display usage message\n"
509 " --version, -V Display version number\n"
510 );
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511 xfclose(stdout);
512 exit(0);
513}
514
4fd38868 515static size_t playrtp_callback(void *buffer,
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516 size_t max_samples,
517 void attribute((unused)) *userdata) {
518 size_t samples;
67d308e7 519 int silent = 0;
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520
521 pthread_mutex_lock(&lock);
522 /* Get the next packet, junking any that are now in the past */
523 const struct packet *p = playrtp_next_packet();
524 if(p && contains(p, next_timestamp)) {
525 /* This packet is ready to play; the desired next timestamp points
526 * somewhere into it. */
527
528 /* Timestamp of end of packet */
529 const uint32_t packet_end = p->timestamp + p->nsamples;
530
531 /* Offset of desired next timestamp into current packet */
532 const uint32_t offset = next_timestamp - p->timestamp;
533
534 /* Pointer to audio data */
535 const uint16_t *ptr = (void *)(p->samples_raw + offset);
536
537 /* Compute number of samples left in packet, limited to output buffer
538 * size */
539 samples = packet_end - next_timestamp;
540 if(samples > max_samples)
541 samples = max_samples;
542
543 /* Copy into buffer, converting to native endianness */
544 size_t i = samples;
545 int16_t *bufptr = buffer;
546 while(i > 0) {
547 *bufptr++ = (int16_t)ntohs(*ptr++);
548 --i;
549 }
67d308e7 550 silent = !!(p->flags & SILENT);
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551 } else {
552 /* There is no suitable packet. We introduce 0s up to the next packet, or
553 * to fill the buffer if there's no next packet or that's too many. The
554 * comparison with max_samples deals with the otherwise troubling overflow
555 * case. */
556 samples = p ? p->timestamp - next_timestamp : max_samples;
557 if(samples > max_samples)
558 samples = max_samples;
559 //info("infill by %zu", samples);
4fd38868 560 memset(buffer, 0, samples * uaudio_sample_size);
67d308e7 561 silent = 1;
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562 }
563 /* Debug dump */
564 if(dump_buffer) {
565 for(size_t i = 0; i < samples; ++i) {
4fd38868 566 dump_buffer[dump_index++] = ((int16_t *)buffer)[i];
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567 dump_index %= dump_size;
568 }
569 }
570 /* Advance timestamp */
571 next_timestamp += samples;
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572 /* If we're getting behind then try to drop just silent packets
573 *
574 * In theory this shouldn't be necessary. The server is supposed to send
575 * packets at the right rate and compares the number of samples sent with the
576 * time in order to ensure this.
577 *
578 * However, various things could throw this off:
579 *
580 * - the server's clock could advance at the wrong rate. This would cause it
581 * to mis-estimate the right number of samples to have sent and
582 * inappropriately throttle or speed up.
583 *
584 * - playback could happen at the wrong rate. If the playback host's sound
585 * card has a slightly incorrect clock then eventually it will get out
586 * of step.
587 *
588 * So if we play back slightly slower than the server sends for either of
589 * these reasons then eventually our buffer, and the socket's buffer, will
590 * fill, and the kernel will start dropping packets. The result is audible
591 * and not very nice.
592 *
593 * Therefore if we're getting behind, we pre-emptively drop silent packets,
594 * since a change in the duration of a silence is less noticeable than a
595 * dropped packet from the middle of continuous music.
596 *
597 * (If things go wrong the other way then eventually we run out of packets to
598 * play and are forced to play silence. This doesn't seem to happen in
599 * practice but if it does then in the same way we can artificially extend
600 * silent packets to compensate.)
601 *
602 * Dropped packets are always logged; use 'disorder-playrtp --monitor' to
603 * track how close to target buffer occupancy we are on a once-a-minute
604 * basis.
605 */
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606 if(nsamples > minbuffer && silent) {
607 info("dropping %zu samples (%"PRIu32" > %"PRIu32")",
608 samples, nsamples, minbuffer);
609 samples = 0;
610 }
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611 /* Junk obsolete packets */
612 playrtp_next_packet();
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613 pthread_mutex_unlock(&lock);
614 return samples;
615}
616
e83d0967 617int main(int argc, char **argv) {
a99c4e9a 618 int n, err;
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619 struct addrinfo *res;
620 struct stringlist sl;
0b75463f 621 char *sockname;
0e72bf84 622 int rcvbuf, target_rcvbuf = 0;
1f10f780 623 socklen_t len;
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624 struct ip_mreq mreq;
625 struct ipv6_mreq mreq6;
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626 disorder_client *c;
627 char *address, *port;
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628 int is_multicast;
629 union any_sockaddr {
630 struct sockaddr sa;
631 struct sockaddr_in in;
632 struct sockaddr_in6 in6;
633 };
634 union any_sockaddr mgroup;
e9b635a3 635 const char *dumpfile = 0;
7a2c7068 636 pthread_t ltid;
b0619501 637 int monitor = 0;
983c3357 638 static const int one = 1;
e83d0967 639
0b75463f 640 static const struct addrinfo prefs = {
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641 .ai_flags = AI_PASSIVE,
642 .ai_family = PF_INET,
643 .ai_socktype = SOCK_DGRAM,
644 .ai_protocol = IPPROTO_UDP
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645 };
646
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647 /* Timing information is often important to debugging playrtp, so we include
648 * timestamps in the logs */
649 logdate = 1;
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650 mem_init();
651 if(!setlocale(LC_CTYPE, "")) fatal(errno, "error calling setlocale");
902ccab0 652 backend = uaudio_apis[0];
b0619501 653 while((n = getopt_long(argc, argv, "hVdD:m:x:L:R:aocC:re:P:M", options, 0)) >= 0) {
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654 switch(n) {
655 case 'h': help();
3fbdc96d 656 case 'V': version("disorder-playrtp");
e83d0967 657 case 'd': debugging = 1; break;
e979b844 658 case 'D': uaudio_set("device", optarg); break;
1153fd23 659 case 'm': minbuffer = 2 * atol(optarg); break;
9086a105 660 case 'x': maxbuffer = 2 * atol(optarg); break;
345ebe66 661 case 'L': logfp = fopen(optarg, "w"); break;
1f10f780 662 case 'R': target_rcvbuf = atoi(optarg); break;
146e86fb 663#if HAVE_ALSA_ASOUNDLIB_H
7a2c7068 664 case 'a': backend = &uaudio_alsa; break;
c593cf7c 665#endif
a9f0ad12 666#if HAVE_SYS_SOUNDCARD_H || EMPEG_HOST
7a2c7068 667 case 'o': backend = &uaudio_oss; break;
c593cf7c 668#endif
669#if HAVE_COREAUDIO_AUDIOHARDWARE_H
7a2c7068 670 case 'c': backend = &uaudio_coreaudio; break;
c593cf7c 671#endif
a7e9570a 672 case 'C': configfile = optarg; break;
a99c4e9a 673 case 's': control_socket = optarg; break;
e9b635a3 674 case 'r': dumpfile = optarg; break;
e979b844 675 case 'e': backend = &uaudio_command; uaudio_set("command", optarg); break;
287ad384 676 case 'P': uaudio_set("pause-mode", optarg); break;
b0619501 677 case 'M': monitor = 1; break;
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678 default: fatal(0, "invalid option");
679 }
680 }
02ba7921 681 if(config_read(0, NULL)) fatal(0, "cannot read configuration");
9086a105 682 if(!maxbuffer)
0e72bf84 683 maxbuffer = 2 * minbuffer;
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684 argc -= optind;
685 argv += optind;
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686 switch(argc) {
687 case 0:
6fba990c 688 /* Get configuration from server */
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689 if(!(c = disorder_new(1))) exit(EXIT_FAILURE);
690 if(disorder_connect(c)) exit(EXIT_FAILURE);
691 if(disorder_rtp_address(c, &address, &port)) exit(EXIT_FAILURE);
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692 sl.n = 2;
693 sl.s = xcalloc(2, sizeof *sl.s);
694 sl.s[0] = address;
695 sl.s[1] = port;
a7e9570a 696 break;
6fba990c 697 case 1:
a7e9570a 698 case 2:
6fba990c 699 /* Use command-line ADDRESS+PORT or just PORT */
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700 sl.n = argc;
701 sl.s = argv;
702 break;
703 default:
6fba990c 704 fatal(0, "usage: disorder-playrtp [OPTIONS] [[ADDRESS] PORT]");
a7e9570a 705 }
6fba990c 706 /* Look up address and port */
0b75463f 707 if(!(res = get_address(&sl, &prefs, &sockname)))
e83d0967 708 exit(1);
6fba990c 709 /* Create the socket */
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710 if((rtpfd = socket(res->ai_family,
711 res->ai_socktype,
712 res->ai_protocol)) < 0)
713 fatal(errno, "error creating socket");
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714 /* Allow multiple listeners */
715 xsetsockopt(rtpfd, SOL_SOCKET, SO_REUSEADDR, &one, sizeof one);
716 is_multicast = multicast(res->ai_addr);
717 /* The multicast and unicast/broadcast cases are different enough that they
718 * are totally split. Trying to find commonality between them causes more
719 * trouble that it's worth. */
720 if(is_multicast) {
721 /* Stash the multicast group address */
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722 memcpy(&mgroup, res->ai_addr, res->ai_addrlen);
723 switch(res->ai_addr->sa_family) {
724 case AF_INET:
725 mgroup.in.sin_port = 0;
726 break;
727 case AF_INET6:
728 mgroup.in6.sin6_port = 0;
729 break;
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730 default:
731 fatal(0, "unsupported family %d", (int)res->ai_addr->sa_family);
6fba990c 732 }
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733 /* Bind to to the multicast group address */
734 if(bind(rtpfd, res->ai_addr, res->ai_addrlen) < 0)
735 fatal(errno, "error binding socket to %s", format_sockaddr(res->ai_addr));
736 /* Add multicast group membership */
6fba990c 737 switch(mgroup.sa.sa_family) {
23205f9c 738 case PF_INET:
6fba990c 739 mreq.imr_multiaddr = mgroup.in.sin_addr;
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740 mreq.imr_interface.s_addr = 0; /* use primary interface */
741 if(setsockopt(rtpfd, IPPROTO_IP, IP_ADD_MEMBERSHIP,
742 &mreq, sizeof mreq) < 0)
743 fatal(errno, "error calling setsockopt IP_ADD_MEMBERSHIP");
744 break;
745 case PF_INET6:
6fba990c 746 mreq6.ipv6mr_multiaddr = mgroup.in6.sin6_addr;
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747 memset(&mreq6.ipv6mr_interface, 0, sizeof mreq6.ipv6mr_interface);
748 if(setsockopt(rtpfd, IPPROTO_IPV6, IPV6_JOIN_GROUP,
749 &mreq6, sizeof mreq6) < 0)
750 fatal(errno, "error calling setsockopt IPV6_JOIN_GROUP");
751 break;
752 default:
753 fatal(0, "unsupported address family %d", res->ai_family);
754 }
983c3357 755 /* Report what we did */
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756 info("listening on %s multicast group %s",
757 format_sockaddr(res->ai_addr), format_sockaddr(&mgroup.sa));
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758 } else {
759 /* Bind to 0/port */
760 switch(res->ai_addr->sa_family) {
761 case AF_INET: {
762 struct sockaddr_in *in = (struct sockaddr_in *)res->ai_addr;
763
764 memset(&in->sin_addr, 0, sizeof (struct in_addr));
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765 break;
766 }
767 case AF_INET6: {
768 struct sockaddr_in6 *in6 = (struct sockaddr_in6 *)res->ai_addr;
769
770 memset(&in6->sin6_addr, 0, sizeof (struct in6_addr));
771 break;
772 }
773 default:
774 fatal(0, "unsupported family %d", (int)res->ai_addr->sa_family);
775 }
776 if(bind(rtpfd, res->ai_addr, res->ai_addrlen) < 0)
777 fatal(errno, "error binding socket to %s", format_sockaddr(res->ai_addr));
778 /* Report what we did */
6fba990c 779 info("listening on %s", format_sockaddr(res->ai_addr));
983c3357 780 }
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781 len = sizeof rcvbuf;
782 if(getsockopt(rtpfd, SOL_SOCKET, SO_RCVBUF, &rcvbuf, &len) < 0)
783 fatal(errno, "error calling getsockopt SO_RCVBUF");
f0bae611 784 if(target_rcvbuf > rcvbuf) {
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785 if(setsockopt(rtpfd, SOL_SOCKET, SO_RCVBUF,
786 &target_rcvbuf, sizeof target_rcvbuf) < 0)
787 error(errno, "error calling setsockopt SO_RCVBUF %d",
788 target_rcvbuf);
789 /* We try to carry on anyway */
790 else
791 info("changed socket receive buffer from %d to %d",
792 rcvbuf, target_rcvbuf);
793 } else
794 info("default socket receive buffer %d", rcvbuf);
ad535598 795 //info("minbuffer %u maxbuffer %u", minbuffer, maxbuffer);
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796 if(logfp)
797 info("WARNING: -L option can impact performance");
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798 if(control_socket) {
799 pthread_t tid;
800
801 if((err = pthread_create(&tid, 0, control_thread, 0)))
802 fatal(err, "pthread_create control_thread");
803 }
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804 if(dumpfile) {
805 int fd;
806 unsigned char buffer[65536];
807 size_t written;
808
809 if((fd = open(dumpfile, O_RDWR|O_TRUNC|O_CREAT, 0666)) < 0)
810 fatal(errno, "opening %s", dumpfile);
811 /* Fill with 0s to a suitable size */
812 memset(buffer, 0, sizeof buffer);
813 for(written = 0; written < dump_size * sizeof(int16_t);
814 written += sizeof buffer) {
815 if(write(fd, buffer, sizeof buffer) < 0)
816 fatal(errno, "clearing %s", dumpfile);
817 }
818 /* Map the buffer into memory for convenience */
819 dump_buffer = mmap(0, dump_size * sizeof(int16_t), PROT_READ|PROT_WRITE,
820 MAP_SHARED, fd, 0);
821 if(dump_buffer == (void *)-1)
822 fatal(errno, "mapping %s", dumpfile);
823 info("dumping to %s", dumpfile);
824 }
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825 /* Set up output. Currently we only support L16 so there's no harm setting
826 * the format before we know what it is! */
827 uaudio_set_format(44100/*Hz*/, 2/*channels*/,
828 16/*bits/channel*/, 1/*signed*/);
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829 backend->start(playrtp_callback, NULL);
830 /* We receive and convert audio data in a background thread */
831 if((err = pthread_create(&ltid, 0, listen_thread, 0)))
832 fatal(err, "pthread_create listen_thread");
833 /* We have a second thread to add received packets to the queue */
834 if((err = pthread_create(&ltid, 0, queue_thread, 0)))
835 fatal(err, "pthread_create queue_thread");
836 pthread_mutex_lock(&lock);
b0619501 837 time_t lastlog = 0;
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838 for(;;) {
839 /* Wait for the buffer to fill up a bit */
840 playrtp_fill_buffer();
841 /* Start playing now */
842 info("Playing...");
843 next_timestamp = pheap_first(&packets)->timestamp;
844 active = 1;
d4170ca7 845 pthread_mutex_unlock(&lock);
7a2c7068 846 backend->activate();
d4170ca7 847 pthread_mutex_lock(&lock);
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848 /* Wait until the buffer empties out
849 *
850 * If there's a packet that we can play right now then we definitely
851 * continue.
852 *
853 * Also if there's at least minbuffer samples we carry on regardless and
854 * insert silence. The assumption is there's been a pause but more data
855 * is now available.
856 */
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857 while(nsamples >= minbuffer
858 || (nsamples > 0
4fd38868 859 && contains(pheap_first(&packets), next_timestamp))) {
b0619501 860 if(monitor) {
4265e5d3 861 time_t now = xtime(0);
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862
863 if(now >= lastlog + 60) {
864 int offset = nsamples - minbuffer;
865 double offtime = (double)offset / (uaudio_rate * uaudio_channels);
866 info("%+d samples off (%d.%02ds, %d bytes)",
867 offset,
868 (int)fabs(offtime) * (offtime < 0 ? -1 : 1),
869 (int)(fabs(offtime) * 100) % 100,
870 offset * uaudio_bits / CHAR_BIT);
871 lastlog = now;
872 }
873 }
ad535598 874 //fprintf(stderr, "%8u/%u (%u) PLAYING\n", nsamples, maxbuffer, minbuffer);
7a2c7068 875 pthread_cond_wait(&cond, &lock);
4fd38868 876 }
ad535598
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877#if 0
878 if(nsamples) {
879 struct packet *p = pheap_first(&packets);
880 fprintf(stderr, "nsamples=%u (%u) next_timestamp=%"PRIx32", first packet is [%"PRIx32",%"PRIx32")\n",
881 nsamples, minbuffer, next_timestamp,p->timestamp,p->timestamp+p->nsamples);
882 }
883#endif
7a2c7068 884 /* Stop playing for a bit until the buffer re-fills */
d4170ca7 885 pthread_mutex_unlock(&lock);
7a2c7068 886 backend->deactivate();
d4170ca7 887 pthread_mutex_lock(&lock);
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888 active = 0;
889 /* Go back round */
890 }
e83d0967
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891 return 0;
892}
893
894/*
895Local Variables:
896c-basic-offset:2
897comment-column:40
898fill-column:79
899indent-tabs-mode:nil
900End:
901*/