| 1 | /* |
| 2 | * This file is part of DisOrder. |
| 3 | * Copyright (C) 2007 Richard Kettlewell |
| 4 | * |
| 5 | * This program is free software; you can redistribute it and/or modify |
| 6 | * it under the terms of the GNU General Public License as published by |
| 7 | * the Free Software Foundation; either version 2 of the License, or |
| 8 | * (at your option) any later version. |
| 9 | * |
| 10 | * This program is distributed in the hope that it will be useful, but |
| 11 | * WITHOUT ANY WARRANTY; without even the implied warranty of |
| 12 | * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU |
| 13 | * General Public License for more details. |
| 14 | * |
| 15 | * You should have received a copy of the GNU General Public License |
| 16 | * along with this program; if not, write to the Free Software |
| 17 | * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 |
| 18 | * USA |
| 19 | */ |
| 20 | /** @file clients/playrtp.c |
| 21 | * @brief RTP player |
| 22 | * |
| 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 | * |
| 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). |
| 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 |
| 37 | * plays. |
| 38 | * |
| 39 | * InCore Audio the main thread is only responsible for starting and stopping |
| 40 | * play: the system does the actual playback in its own private thread, and |
| 41 | * calls adioproc() to fetch the audio data. |
| 42 | * |
| 43 | * Sometimes it happens that there is no audio available to play. This may |
| 44 | * because the server went away, or a packet was dropped, or the server |
| 45 | * deliberately did not send any sound because it encountered a silence. |
| 46 | * |
| 47 | * Assumptions: |
| 48 | * - it is safe to read uint32_t values without a lock protecting them |
| 49 | */ |
| 50 | |
| 51 | #include <config.h> |
| 52 | #include "types.h" |
| 53 | |
| 54 | #include <getopt.h> |
| 55 | #include <stdio.h> |
| 56 | #include <stdlib.h> |
| 57 | #include <sys/socket.h> |
| 58 | #include <sys/types.h> |
| 59 | #include <sys/socket.h> |
| 60 | #include <netdb.h> |
| 61 | #include <pthread.h> |
| 62 | #include <locale.h> |
| 63 | #include <sys/uio.h> |
| 64 | #include <string.h> |
| 65 | |
| 66 | #include "log.h" |
| 67 | #include "mem.h" |
| 68 | #include "configuration.h" |
| 69 | #include "addr.h" |
| 70 | #include "syscalls.h" |
| 71 | #include "rtp.h" |
| 72 | #include "defs.h" |
| 73 | #include "vector.h" |
| 74 | #include "heap.h" |
| 75 | #include "timeval.h" |
| 76 | |
| 77 | #if HAVE_COREAUDIO_AUDIOHARDWARE_H |
| 78 | # include <CoreAudio/AudioHardware.h> |
| 79 | #endif |
| 80 | #if API_ALSA |
| 81 | #include <alsa/asoundlib.h> |
| 82 | #endif |
| 83 | |
| 84 | #define readahead linux_headers_are_borked |
| 85 | |
| 86 | /** @brief RTP socket */ |
| 87 | static int rtpfd; |
| 88 | |
| 89 | /** @brief Log output */ |
| 90 | static FILE *logfp; |
| 91 | |
| 92 | /** @brief Output device */ |
| 93 | static const char *device; |
| 94 | |
| 95 | /** @brief Maximum samples per packet we'll support |
| 96 | * |
| 97 | * NB that two channels = two samples in this program. |
| 98 | */ |
| 99 | #define MAXSAMPLES 2048 |
| 100 | |
| 101 | /** @brief Minimum low watermark |
| 102 | * |
| 103 | * We'll stop playing if there's only this many samples in the buffer. */ |
| 104 | static unsigned minbuffer = 2 * 44100 / 10; /* 0.2 seconds */ |
| 105 | |
| 106 | /** @brief Buffer high watermark |
| 107 | * |
| 108 | * We'll only start playing when this many samples are available. */ |
| 109 | static unsigned readahead = 2 * 2 * 44100; |
| 110 | |
| 111 | /** @brief Maximum buffer size |
| 112 | * |
| 113 | * We'll stop reading from the network if we have this many samples. */ |
| 114 | static unsigned maxbuffer; |
| 115 | |
| 116 | /** @brief Number of samples to infill by in one go |
| 117 | * |
| 118 | * This is an upper bound - in practice we expect the underlying audio API to |
| 119 | * only ask for a much smaller number of samples in any one go. |
| 120 | */ |
| 121 | #define INFILL_SAMPLES (44100 * 2) /* 1s */ |
| 122 | |
| 123 | /** @brief Received packet |
| 124 | * |
| 125 | * Received packets are kept in a binary heap (see @ref pheap) ordered by |
| 126 | * timestamp. |
| 127 | */ |
| 128 | struct packet { |
| 129 | /** @brief Next packet in @ref next_free_packet or @ref received_packets */ |
| 130 | struct packet *next; |
| 131 | |
| 132 | /** @brief Number of samples in this packet */ |
| 133 | uint32_t nsamples; |
| 134 | |
| 135 | /** @brief Timestamp from RTP packet |
| 136 | * |
| 137 | * NB that "timestamps" are really sample counters. Use lt() or lt_packet() |
| 138 | * to compare timestamps. |
| 139 | */ |
| 140 | uint32_t timestamp; |
| 141 | |
| 142 | /** @brief Flags |
| 143 | * |
| 144 | * Valid values are: |
| 145 | * - @ref IDLE - the idle bit was set in the RTP packet |
| 146 | */ |
| 147 | unsigned flags; |
| 148 | /** @brief idle bit set in RTP packet*/ |
| 149 | #define IDLE 0x0001 |
| 150 | |
| 151 | /** @brief Raw sample data |
| 152 | * |
| 153 | * Only the first @p nsamples samples are defined; the rest is uninitialized |
| 154 | * data. |
| 155 | */ |
| 156 | uint16_t samples_raw[MAXSAMPLES]; |
| 157 | }; |
| 158 | |
| 159 | /** @brief Return true iff \f$a < b\f$ in sequence-space arithmetic |
| 160 | * |
| 161 | * Specifically it returns true if \f$(a-b) mod 2^{32} < 2^{31}\f$. |
| 162 | * |
| 163 | * See also lt_packet(). |
| 164 | */ |
| 165 | static inline int lt(uint32_t a, uint32_t b) { |
| 166 | return (uint32_t)(a - b) & 0x80000000; |
| 167 | } |
| 168 | |
| 169 | /** @brief Return true iff a >= b in sequence-space arithmetic */ |
| 170 | static inline int ge(uint32_t a, uint32_t b) { |
| 171 | return !lt(a, b); |
| 172 | } |
| 173 | |
| 174 | /** @brief Return true iff a > b in sequence-space arithmetic */ |
| 175 | static inline int gt(uint32_t a, uint32_t b) { |
| 176 | return lt(b, a); |
| 177 | } |
| 178 | |
| 179 | /** @brief Return true iff a <= b in sequence-space arithmetic */ |
| 180 | static inline int le(uint32_t a, uint32_t b) { |
| 181 | return !lt(b, a); |
| 182 | } |
| 183 | |
| 184 | /** @brief Ordering for packets, used by @ref pheap */ |
| 185 | static inline int lt_packet(const struct packet *a, const struct packet *b) { |
| 186 | return lt(a->timestamp, b->timestamp); |
| 187 | } |
| 188 | |
| 189 | /** @brief Received packets |
| 190 | * Protected by @ref receive_lock |
| 191 | * |
| 192 | * Received packets are added to this list, and queue_thread() picks them off |
| 193 | * it and adds them to @ref packets. Whenever a packet is added to it, @ref |
| 194 | * receive_cond is signalled. |
| 195 | */ |
| 196 | static struct packet *received_packets; |
| 197 | |
| 198 | /** @brief Tail of @ref received_packets |
| 199 | * Protected by @ref receive_lock |
| 200 | */ |
| 201 | static struct packet **received_tail = &received_packets; |
| 202 | |
| 203 | /** @brief Lock protecting @ref received_packets |
| 204 | * |
| 205 | * Only listen_thread() and queue_thread() ever hold this lock. It is vital |
| 206 | * that queue_thread() not hold it any longer than it strictly has to. */ |
| 207 | static pthread_mutex_t receive_lock = PTHREAD_MUTEX_INITIALIZER; |
| 208 | |
| 209 | /** @brief Condition variable signalled when @ref received_packets is updated |
| 210 | * |
| 211 | * Used by listen_thread() to notify queue_thread() that it has added another |
| 212 | * packet to @ref received_packets. */ |
| 213 | static pthread_cond_t receive_cond = PTHREAD_COND_INITIALIZER; |
| 214 | |
| 215 | /** @brief Length of @ref received_packets */ |
| 216 | static uint32_t nreceived; |
| 217 | |
| 218 | /** @struct pheap |
| 219 | * @brief Binary heap of packets ordered by timestamp */ |
| 220 | HEAP_TYPE(pheap, struct packet *, lt_packet); |
| 221 | |
| 222 | /** @brief Binary heap of received packets */ |
| 223 | static struct pheap packets; |
| 224 | |
| 225 | /** @brief Total number of samples available |
| 226 | * |
| 227 | * We make this volatile because we inspect it without a protecting lock, |
| 228 | * so the usual pthread_* guarantees aren't available. |
| 229 | */ |
| 230 | static volatile uint32_t nsamples; |
| 231 | |
| 232 | /** @brief Timestamp of next packet to play. |
| 233 | * |
| 234 | * This is set to the timestamp of the last packet, plus the number of |
| 235 | * samples it contained. Only valid if @ref active is nonzero. |
| 236 | */ |
| 237 | static uint32_t next_timestamp; |
| 238 | |
| 239 | /** @brief True if actively playing |
| 240 | * |
| 241 | * This is true when playing and false when just buffering. */ |
| 242 | static int active; |
| 243 | |
| 244 | /** @brief Lock protecting @ref packets */ |
| 245 | static pthread_mutex_t lock = PTHREAD_MUTEX_INITIALIZER; |
| 246 | |
| 247 | /** @brief Condition variable signalled whenever @ref packets is changed */ |
| 248 | static pthread_cond_t cond = PTHREAD_COND_INITIALIZER; |
| 249 | |
| 250 | /** @brief Structure of free packet list */ |
| 251 | union free_packet { |
| 252 | struct packet p; |
| 253 | union free_packet *next; |
| 254 | }; |
| 255 | |
| 256 | /** @brief Linked list of free packets |
| 257 | * |
| 258 | * This is a linked list of formerly used packets. For preference we re-use |
| 259 | * packets that have already been used rather than unused ones, to limit the |
| 260 | * size of the program's working set. If there are no free packets in the list |
| 261 | * we try @ref next_free_packet instead. |
| 262 | * |
| 263 | * Must hold @ref lock when accessing this. |
| 264 | */ |
| 265 | static union free_packet *free_packets; |
| 266 | |
| 267 | /** @brief Array of new free packets |
| 268 | * |
| 269 | * There are @ref count_free_packets ready to use at this address. If there |
| 270 | * are none left we allocate more memory. |
| 271 | * |
| 272 | * Must hold @ref lock when accessing this. |
| 273 | */ |
| 274 | static union free_packet *next_free_packet; |
| 275 | |
| 276 | /** @brief Count of new free packets at @ref next_free_packet |
| 277 | * |
| 278 | * Must hold @ref lock when accessing this. |
| 279 | */ |
| 280 | static size_t count_free_packets; |
| 281 | |
| 282 | /** @brief Lock protecting packet allocator */ |
| 283 | static pthread_mutex_t mem_lock = PTHREAD_MUTEX_INITIALIZER; |
| 284 | |
| 285 | static const struct option options[] = { |
| 286 | { "help", no_argument, 0, 'h' }, |
| 287 | { "version", no_argument, 0, 'V' }, |
| 288 | { "debug", no_argument, 0, 'd' }, |
| 289 | { "device", required_argument, 0, 'D' }, |
| 290 | { "min", required_argument, 0, 'm' }, |
| 291 | { "max", required_argument, 0, 'x' }, |
| 292 | { "buffer", required_argument, 0, 'b' }, |
| 293 | { 0, 0, 0, 0 } |
| 294 | }; |
| 295 | |
| 296 | /** @Brief Return a new packet */ |
| 297 | static struct packet *new_packet(void) { |
| 298 | struct packet *p; |
| 299 | |
| 300 | pthread_mutex_lock(&mem_lock); |
| 301 | if(free_packets) { |
| 302 | p = &free_packets->p; |
| 303 | free_packets = free_packets->next; |
| 304 | } else { |
| 305 | if(!count_free_packets) { |
| 306 | next_free_packet = xcalloc(1024, sizeof (union free_packet)); |
| 307 | count_free_packets = 1024; |
| 308 | } |
| 309 | p = &(next_free_packet++)->p; |
| 310 | --count_free_packets; |
| 311 | } |
| 312 | pthread_mutex_unlock(&mem_lock); |
| 313 | return p; |
| 314 | } |
| 315 | |
| 316 | /** @brief Free a packet */ |
| 317 | static void free_packet(struct packet *p) { |
| 318 | union free_packet *u = (union free_packet *)p; |
| 319 | pthread_mutex_lock(&mem_lock); |
| 320 | u->next = free_packets; |
| 321 | free_packets = u; |
| 322 | pthread_mutex_unlock(&mem_lock); |
| 323 | } |
| 324 | |
| 325 | /** @brief Drop the first packet |
| 326 | * |
| 327 | * Assumes that @ref lock is held. |
| 328 | */ |
| 329 | static void drop_first_packet(void) { |
| 330 | if(pheap_count(&packets)) { |
| 331 | struct packet *const p = pheap_remove(&packets); |
| 332 | nsamples -= p->nsamples; |
| 333 | free_packet(p); |
| 334 | pthread_cond_broadcast(&cond); |
| 335 | } |
| 336 | } |
| 337 | |
| 338 | /** @brief Background thread adding packets to heap |
| 339 | * |
| 340 | * This just transfers packets from @ref received_packets to @ref packets. It |
| 341 | * is important that it holds @ref receive_lock for as little time as possible, |
| 342 | * in order to minimize the interval between calls to read() in |
| 343 | * listen_thread(). |
| 344 | */ |
| 345 | static void *queue_thread(void attribute((unused)) *arg) { |
| 346 | struct packet *p; |
| 347 | |
| 348 | for(;;) { |
| 349 | /* Get the next packet */ |
| 350 | pthread_mutex_lock(&receive_lock); |
| 351 | while(!received_packets) |
| 352 | pthread_cond_wait(&receive_cond, &receive_lock); |
| 353 | p = received_packets; |
| 354 | received_packets = p->next; |
| 355 | if(!received_packets) |
| 356 | received_tail = &received_packets; |
| 357 | --nreceived; |
| 358 | pthread_mutex_unlock(&receive_lock); |
| 359 | /* Add it to the heap */ |
| 360 | pthread_mutex_lock(&lock); |
| 361 | pheap_insert(&packets, p); |
| 362 | nsamples += p->nsamples; |
| 363 | pthread_cond_broadcast(&cond); |
| 364 | pthread_mutex_unlock(&lock); |
| 365 | } |
| 366 | } |
| 367 | |
| 368 | /** @brief Background thread collecting samples |
| 369 | * |
| 370 | * This function collects samples, perhaps converts them to the target format, |
| 371 | * and adds them to the packet list. |
| 372 | * |
| 373 | * It is crucial that the gap between successive calls to read() is as small as |
| 374 | * possible: otherwise packets will be dropped. |
| 375 | * |
| 376 | * We use a binary heap to ensure that the unavoidable effort is at worst |
| 377 | * logarithmic in the total number of packets - in fact if packets are mostly |
| 378 | * received in order then we will largely do constant work per packet since the |
| 379 | * newest packet will always be last. |
| 380 | * |
| 381 | * Of more concern is that we must acquire the lock on the heap to add a packet |
| 382 | * to it. If this proves a problem in practice then the answer would be |
| 383 | * (probably doubly) linked list with new packets added the end and a second |
| 384 | * thread which reads packets off the list and adds them to the heap. |
| 385 | * |
| 386 | * We keep memory allocation (mostly) very fast by keeping pre-allocated |
| 387 | * packets around; see @ref new_packet(). |
| 388 | */ |
| 389 | static void *listen_thread(void attribute((unused)) *arg) { |
| 390 | struct packet *p = 0; |
| 391 | int n; |
| 392 | struct rtp_header header; |
| 393 | uint16_t seq; |
| 394 | uint32_t timestamp; |
| 395 | struct iovec iov[2]; |
| 396 | |
| 397 | for(;;) { |
| 398 | if(!p) |
| 399 | p = new_packet(); |
| 400 | iov[0].iov_base = &header; |
| 401 | iov[0].iov_len = sizeof header; |
| 402 | iov[1].iov_base = p->samples_raw; |
| 403 | iov[1].iov_len = sizeof p->samples_raw / sizeof *p->samples_raw; |
| 404 | n = readv(rtpfd, iov, 2); |
| 405 | if(n < 0) { |
| 406 | switch(errno) { |
| 407 | case EINTR: |
| 408 | continue; |
| 409 | default: |
| 410 | fatal(errno, "error reading from socket"); |
| 411 | } |
| 412 | } |
| 413 | /* Ignore too-short packets */ |
| 414 | if((size_t)n <= sizeof (struct rtp_header)) { |
| 415 | info("ignored a short packet"); |
| 416 | continue; |
| 417 | } |
| 418 | timestamp = htonl(header.timestamp); |
| 419 | seq = htons(header.seq); |
| 420 | /* Ignore packets in the past */ |
| 421 | if(active && lt(timestamp, next_timestamp)) { |
| 422 | info("dropping old packet, timestamp=%"PRIx32" < %"PRIx32, |
| 423 | timestamp, next_timestamp); |
| 424 | continue; |
| 425 | } |
| 426 | p->next = 0; |
| 427 | p->flags = 0; |
| 428 | p->timestamp = timestamp; |
| 429 | /* Convert to target format */ |
| 430 | if(header.mpt & 0x80) |
| 431 | p->flags |= IDLE; |
| 432 | switch(header.mpt & 0x7F) { |
| 433 | case 10: |
| 434 | p->nsamples = (n - sizeof header) / sizeof(uint16_t); |
| 435 | break; |
| 436 | /* TODO support other RFC3551 media types (when the speaker does) */ |
| 437 | default: |
| 438 | fatal(0, "unsupported RTP payload type %d", |
| 439 | header.mpt & 0x7F); |
| 440 | } |
| 441 | if(logfp) |
| 442 | fprintf(logfp, "sequence %u timestamp %"PRIx32" length %"PRIx32" end %"PRIx32"\n", |
| 443 | seq, timestamp, p->nsamples, timestamp + p->nsamples); |
| 444 | /* Stop reading if we've reached the maximum. |
| 445 | * |
| 446 | * This is rather unsatisfactory: it means that if packets get heavily |
| 447 | * out of order then we guarantee dropouts. But for now... */ |
| 448 | if(nsamples >= maxbuffer) { |
| 449 | pthread_mutex_lock(&lock); |
| 450 | while(nsamples >= maxbuffer) |
| 451 | pthread_cond_wait(&cond, &lock); |
| 452 | pthread_mutex_unlock(&lock); |
| 453 | } |
| 454 | /* Add the packet to the receive queue */ |
| 455 | pthread_mutex_lock(&receive_lock); |
| 456 | *received_tail = p; |
| 457 | received_tail = &p->next; |
| 458 | ++nreceived; |
| 459 | pthread_cond_signal(&receive_cond); |
| 460 | pthread_mutex_unlock(&receive_lock); |
| 461 | /* We'll need a new packet */ |
| 462 | p = 0; |
| 463 | } |
| 464 | } |
| 465 | |
| 466 | /** @brief Return true if @p p contains @p timestamp |
| 467 | * |
| 468 | * Containment implies that a sample @p timestamp exists within the packet. |
| 469 | */ |
| 470 | static inline int contains(const struct packet *p, uint32_t timestamp) { |
| 471 | const uint32_t packet_start = p->timestamp; |
| 472 | const uint32_t packet_end = p->timestamp + p->nsamples; |
| 473 | |
| 474 | return (ge(timestamp, packet_start) |
| 475 | && lt(timestamp, packet_end)); |
| 476 | } |
| 477 | |
| 478 | /** @brief Wait until the buffer is adequately full |
| 479 | * |
| 480 | * Must be called with @ref lock held. |
| 481 | */ |
| 482 | static void fill_buffer(void) { |
| 483 | info("Buffering..."); |
| 484 | while(nsamples < readahead) |
| 485 | pthread_cond_wait(&cond, &lock); |
| 486 | next_timestamp = pheap_first(&packets)->timestamp; |
| 487 | active = 1; |
| 488 | } |
| 489 | |
| 490 | /** @brief Find next packet |
| 491 | * @return Packet to play or NULL if none found |
| 492 | * |
| 493 | * The return packet is merely guaranteed not to be in the past: it might be |
| 494 | * the first packet in the future rather than one that is actually suitable to |
| 495 | * play. |
| 496 | * |
| 497 | * Must be called with @ref lock held. |
| 498 | */ |
| 499 | static struct packet *next_packet(void) { |
| 500 | while(pheap_count(&packets)) { |
| 501 | struct packet *const p = pheap_first(&packets); |
| 502 | if(le(p->timestamp + p->nsamples, next_timestamp)) { |
| 503 | /* This packet is in the past. Drop it and try another one. */ |
| 504 | drop_first_packet(); |
| 505 | } else |
| 506 | /* This packet is NOT in the past. (It might be in the future |
| 507 | * however.) */ |
| 508 | return p; |
| 509 | } |
| 510 | return 0; |
| 511 | } |
| 512 | |
| 513 | #if HAVE_COREAUDIO_AUDIOHARDWARE_H |
| 514 | /** @brief Callback from Core Audio */ |
| 515 | static OSStatus adioproc |
| 516 | (AudioDeviceID attribute((unused)) inDevice, |
| 517 | const AudioTimeStamp attribute((unused)) *inNow, |
| 518 | const AudioBufferList attribute((unused)) *inInputData, |
| 519 | const AudioTimeStamp attribute((unused)) *inInputTime, |
| 520 | AudioBufferList *outOutputData, |
| 521 | const AudioTimeStamp attribute((unused)) *inOutputTime, |
| 522 | void attribute((unused)) *inClientData) { |
| 523 | UInt32 nbuffers = outOutputData->mNumberBuffers; |
| 524 | AudioBuffer *ab = outOutputData->mBuffers; |
| 525 | uint32_t samples_available; |
| 526 | |
| 527 | pthread_mutex_lock(&lock); |
| 528 | while(nbuffers > 0) { |
| 529 | float *samplesOut = ab->mData; |
| 530 | size_t samplesOutLeft = ab->mDataByteSize / sizeof (float); |
| 531 | |
| 532 | while(samplesOutLeft > 0) { |
| 533 | const struct packet *p = next_packet(); |
| 534 | if(p && contains(p, next_timestamp)) { |
| 535 | /* This packet is ready to play */ |
| 536 | const uint32_t packet_end = p->timestamp + p->nsamples; |
| 537 | const uint32_t offset = next_timestamp - p->timestamp; |
| 538 | const uint16_t *ptr = (void *)(p->samples_raw + offset); |
| 539 | |
| 540 | samples_available = packet_end - next_timestamp; |
| 541 | if(samples_available > samplesOutLeft) |
| 542 | samples_available = samplesOutLeft; |
| 543 | next_timestamp += samples_available; |
| 544 | samplesOutLeft -= samples_available; |
| 545 | while(samples_available-- > 0) |
| 546 | *samplesOut++ = (int16_t)ntohs(*ptr++) * (0.5 / 32767); |
| 547 | /* We don't bother junking the packet - that'll be dealt with next time |
| 548 | * round */ |
| 549 | } else { |
| 550 | /* No packet is ready to play (and there might be no packet at all) */ |
| 551 | samples_available = p ? p->timestamp - next_timestamp |
| 552 | : samplesOutLeft; |
| 553 | if(samples_available > samplesOutLeft) |
| 554 | samples_available = samplesOutLeft; |
| 555 | //info("infill by %"PRIu32, samples_available); |
| 556 | /* Conveniently the buffer is 0 to start with */ |
| 557 | next_timestamp += samples_available; |
| 558 | samplesOut += samples_available; |
| 559 | samplesOutLeft -= samples_available; |
| 560 | } |
| 561 | } |
| 562 | ++ab; |
| 563 | --nbuffers; |
| 564 | } |
| 565 | pthread_mutex_unlock(&lock); |
| 566 | return 0; |
| 567 | } |
| 568 | #endif |
| 569 | |
| 570 | |
| 571 | #if API_ALSA |
| 572 | /** @brief PCM handle */ |
| 573 | static snd_pcm_t *pcm; |
| 574 | |
| 575 | /** @brief True when @ref pcm is up and running */ |
| 576 | static int alsa_prepared = 1; |
| 577 | |
| 578 | /** @brief Initialize @ref pcm */ |
| 579 | static void setup_alsa(void) { |
| 580 | snd_pcm_hw_params_t *hwparams; |
| 581 | snd_pcm_sw_params_t *swparams; |
| 582 | /* Only support one format for now */ |
| 583 | const int sample_format = SND_PCM_FORMAT_S16_BE; |
| 584 | unsigned rate = 44100; |
| 585 | const int channels = 2; |
| 586 | const int samplesize = channels * sizeof(uint16_t); |
| 587 | snd_pcm_uframes_t pcm_bufsize = MAXSAMPLES * samplesize * 3; |
| 588 | /* If we can write more than this many samples we'll get a wakeup */ |
| 589 | const int avail_min = 256; |
| 590 | int err; |
| 591 | |
| 592 | /* Open ALSA */ |
| 593 | if((err = snd_pcm_open(&pcm, |
| 594 | device ? device : "default", |
| 595 | SND_PCM_STREAM_PLAYBACK, |
| 596 | SND_PCM_NONBLOCK))) |
| 597 | fatal(0, "error from snd_pcm_open: %d", err); |
| 598 | /* Set up 'hardware' parameters */ |
| 599 | snd_pcm_hw_params_alloca(&hwparams); |
| 600 | if((err = snd_pcm_hw_params_any(pcm, hwparams)) < 0) |
| 601 | fatal(0, "error from snd_pcm_hw_params_any: %d", err); |
| 602 | if((err = snd_pcm_hw_params_set_access(pcm, hwparams, |
| 603 | SND_PCM_ACCESS_RW_INTERLEAVED)) < 0) |
| 604 | fatal(0, "error from snd_pcm_hw_params_set_access: %d", err); |
| 605 | if((err = snd_pcm_hw_params_set_format(pcm, hwparams, |
| 606 | sample_format)) < 0) |
| 607 | |
| 608 | fatal(0, "error from snd_pcm_hw_params_set_format (%d): %d", |
| 609 | sample_format, err); |
| 610 | if((err = snd_pcm_hw_params_set_rate_near(pcm, hwparams, &rate, 0)) < 0) |
| 611 | fatal(0, "error from snd_pcm_hw_params_set_rate (%d): %d", |
| 612 | rate, err); |
| 613 | if((err = snd_pcm_hw_params_set_channels(pcm, hwparams, |
| 614 | channels)) < 0) |
| 615 | fatal(0, "error from snd_pcm_hw_params_set_channels (%d): %d", |
| 616 | channels, err); |
| 617 | if((err = snd_pcm_hw_params_set_buffer_size_near(pcm, hwparams, |
| 618 | &pcm_bufsize)) < 0) |
| 619 | fatal(0, "error from snd_pcm_hw_params_set_buffer_size (%d): %d", |
| 620 | MAXSAMPLES * samplesize * 3, err); |
| 621 | if((err = snd_pcm_hw_params(pcm, hwparams)) < 0) |
| 622 | fatal(0, "error calling snd_pcm_hw_params: %d", err); |
| 623 | /* Set up 'software' parameters */ |
| 624 | snd_pcm_sw_params_alloca(&swparams); |
| 625 | if((err = snd_pcm_sw_params_current(pcm, swparams)) < 0) |
| 626 | fatal(0, "error calling snd_pcm_sw_params_current: %d", err); |
| 627 | if((err = snd_pcm_sw_params_set_avail_min(pcm, swparams, avail_min)) < 0) |
| 628 | fatal(0, "error calling snd_pcm_sw_params_set_avail_min %d: %d", |
| 629 | avail_min, err); |
| 630 | if((err = snd_pcm_sw_params(pcm, swparams)) < 0) |
| 631 | fatal(0, "error calling snd_pcm_sw_params: %d", err); |
| 632 | } |
| 633 | |
| 634 | /** @brief Wait until ALSA wants some audio */ |
| 635 | static void wait_alsa(void) { |
| 636 | struct pollfd fds[64]; |
| 637 | int nfds, err; |
| 638 | unsigned short events; |
| 639 | |
| 640 | for(;;) { |
| 641 | do { |
| 642 | if((nfds = snd_pcm_poll_descriptors(pcm, |
| 643 | fds, sizeof fds / sizeof *fds)) < 0) |
| 644 | fatal(0, "error calling snd_pcm_poll_descriptors: %d", nfds); |
| 645 | } while(poll(fds, nfds, -1) < 0 && errno == EINTR); |
| 646 | if((err = snd_pcm_poll_descriptors_revents(pcm, fds, nfds, &events))) |
| 647 | fatal(0, "error calling snd_pcm_poll_descriptors_revents: %d", err); |
| 648 | if(events & POLLOUT) |
| 649 | return; |
| 650 | } |
| 651 | } |
| 652 | |
| 653 | /** @brief Play some sound via ALSA |
| 654 | * @param s Pointer to sample data |
| 655 | * @param n Number of samples |
| 656 | * @return 0 on success, -1 on non-fatal error |
| 657 | */ |
| 658 | static int alsa_writei(const void *s, size_t n) { |
| 659 | /* Do the write */ |
| 660 | const snd_pcm_sframes_t frames_written = snd_pcm_writei(pcm, s, n / 2); |
| 661 | if(frames_written < 0) { |
| 662 | /* Something went wrong */ |
| 663 | switch(frames_written) { |
| 664 | case -EAGAIN: |
| 665 | return 0; |
| 666 | case -EPIPE: |
| 667 | error(0, "error calling snd_pcm_writei: %ld", |
| 668 | (long)frames_written); |
| 669 | return -1; |
| 670 | default: |
| 671 | fatal(0, "error calling snd_pcm_writei: %ld", |
| 672 | (long)frames_written); |
| 673 | } |
| 674 | } else { |
| 675 | /* Success */ |
| 676 | next_timestamp += frames_written * 2; |
| 677 | return 0; |
| 678 | } |
| 679 | } |
| 680 | |
| 681 | /** @brief Play the relevant part of a packet |
| 682 | * @param p Packet to play |
| 683 | * @return 0 on success, -1 on non-fatal error |
| 684 | */ |
| 685 | static int alsa_play(const struct packet *p) { |
| 686 | return alsa_writei(p->samples_raw + next_timestamp - p->timestamp, |
| 687 | (p->timestamp + p->nsamples) - next_timestamp); |
| 688 | } |
| 689 | |
| 690 | /** @brief Play some silence |
| 691 | * @param p Next packet or NULL |
| 692 | * @return 0 on success, -1 on non-fatal error |
| 693 | */ |
| 694 | static int alsa_infill(const struct packet *p) { |
| 695 | static const uint16_t zeros[INFILL_SAMPLES]; |
| 696 | size_t samples_available = INFILL_SAMPLES; |
| 697 | |
| 698 | if(p && samples_available > p->timestamp - next_timestamp) |
| 699 | samples_available = p->timestamp - next_timestamp; |
| 700 | return alsa_writei(zeros, samples_available); |
| 701 | } |
| 702 | |
| 703 | /** @brief Reset ALSA state after we lost synchronization */ |
| 704 | static void alsa_reset(int hard_reset) { |
| 705 | int err; |
| 706 | |
| 707 | if((err = snd_pcm_nonblock(pcm, 0))) |
| 708 | fatal(0, "error calling snd_pcm_nonblock: %d", err); |
| 709 | if(hard_reset) { |
| 710 | if((err = snd_pcm_drop(pcm))) |
| 711 | fatal(0, "error calling snd_pcm_drop: %d", err); |
| 712 | } else |
| 713 | if((err = snd_pcm_drain(pcm))) |
| 714 | fatal(0, "error calling snd_pcm_drain: %d", err); |
| 715 | if((err = snd_pcm_nonblock(pcm, 1))) |
| 716 | fatal(0, "error calling snd_pcm_nonblock: %d", err); |
| 717 | alsa_prepared = 0; |
| 718 | } |
| 719 | #endif |
| 720 | |
| 721 | /** @brief Play an RTP stream |
| 722 | * |
| 723 | * This is the guts of the program. It is responsible for: |
| 724 | * - starting the listening thread |
| 725 | * - opening the audio device |
| 726 | * - reading ahead to build up a buffer |
| 727 | * - arranging for audio to be played |
| 728 | * - detecting when the buffer has got too small and re-buffering |
| 729 | */ |
| 730 | static void play_rtp(void) { |
| 731 | pthread_t ltid; |
| 732 | |
| 733 | /* We receive and convert audio data in a background thread */ |
| 734 | pthread_create(<id, 0, listen_thread, 0); |
| 735 | /* We have a second thread to add received packets to the queue */ |
| 736 | pthread_create(<id, 0, queue_thread, 0); |
| 737 | #if API_ALSA |
| 738 | { |
| 739 | struct packet *p; |
| 740 | int escape, err; |
| 741 | |
| 742 | /* Open the sound device */ |
| 743 | setup_alsa(); |
| 744 | pthread_mutex_lock(&lock); |
| 745 | for(;;) { |
| 746 | /* Wait for the buffer to fill up a bit */ |
| 747 | fill_buffer(); |
| 748 | if(!alsa_prepared) { |
| 749 | if((err = snd_pcm_prepare(pcm))) |
| 750 | fatal(0, "error calling snd_pcm_prepare: %d", err); |
| 751 | alsa_prepared = 1; |
| 752 | } |
| 753 | escape = 0; |
| 754 | info("Playing..."); |
| 755 | /* Keep playing until the buffer empties out, or ALSA tells us to get |
| 756 | * lost */ |
| 757 | while(nsamples >= minbuffer && !escape) { |
| 758 | /* Wait for ALSA to ask us for more data */ |
| 759 | pthread_mutex_unlock(&lock); |
| 760 | wait_alsa(); |
| 761 | pthread_mutex_lock(&lock); |
| 762 | /* ALSA is ready for more data, find something to play */ |
| 763 | p = next_packet(); |
| 764 | /* Play it or play some silence */ |
| 765 | if(contains(p, next_timestamp)) |
| 766 | escape = alsa_play(p); |
| 767 | else |
| 768 | escape = alsa_infill(p); |
| 769 | } |
| 770 | active = 0; |
| 771 | /* We stop playing for a bit until the buffer re-fills */ |
| 772 | pthread_mutex_unlock(&lock); |
| 773 | alsa_reset(escape); |
| 774 | pthread_mutex_lock(&lock); |
| 775 | } |
| 776 | |
| 777 | } |
| 778 | #elif HAVE_COREAUDIO_AUDIOHARDWARE_H |
| 779 | { |
| 780 | OSStatus status; |
| 781 | UInt32 propertySize; |
| 782 | AudioDeviceID adid; |
| 783 | AudioStreamBasicDescription asbd; |
| 784 | |
| 785 | /* If this looks suspiciously like libao's macosx driver there's an |
| 786 | * excellent reason for that... */ |
| 787 | |
| 788 | /* TODO report errors as strings not numbers */ |
| 789 | propertySize = sizeof adid; |
| 790 | status = AudioHardwareGetProperty(kAudioHardwarePropertyDefaultOutputDevice, |
| 791 | &propertySize, &adid); |
| 792 | if(status) |
| 793 | fatal(0, "AudioHardwareGetProperty: %d", (int)status); |
| 794 | if(adid == kAudioDeviceUnknown) |
| 795 | fatal(0, "no output device"); |
| 796 | propertySize = sizeof asbd; |
| 797 | status = AudioDeviceGetProperty(adid, 0, false, |
| 798 | kAudioDevicePropertyStreamFormat, |
| 799 | &propertySize, &asbd); |
| 800 | if(status) |
| 801 | fatal(0, "AudioHardwareGetProperty: %d", (int)status); |
| 802 | D(("mSampleRate %f", asbd.mSampleRate)); |
| 803 | D(("mFormatID %08lx", asbd.mFormatID)); |
| 804 | D(("mFormatFlags %08lx", asbd.mFormatFlags)); |
| 805 | D(("mBytesPerPacket %08lx", asbd.mBytesPerPacket)); |
| 806 | D(("mFramesPerPacket %08lx", asbd.mFramesPerPacket)); |
| 807 | D(("mBytesPerFrame %08lx", asbd.mBytesPerFrame)); |
| 808 | D(("mChannelsPerFrame %08lx", asbd.mChannelsPerFrame)); |
| 809 | D(("mBitsPerChannel %08lx", asbd.mBitsPerChannel)); |
| 810 | D(("mReserved %08lx", asbd.mReserved)); |
| 811 | if(asbd.mFormatID != kAudioFormatLinearPCM) |
| 812 | fatal(0, "audio device does not support kAudioFormatLinearPCM"); |
| 813 | status = AudioDeviceAddIOProc(adid, adioproc, 0); |
| 814 | if(status) |
| 815 | fatal(0, "AudioDeviceAddIOProc: %d", (int)status); |
| 816 | pthread_mutex_lock(&lock); |
| 817 | for(;;) { |
| 818 | /* Wait for the buffer to fill up a bit */ |
| 819 | fill_buffer(); |
| 820 | /* Start playing now */ |
| 821 | info("Playing..."); |
| 822 | next_timestamp = pheap_first(&packets)->timestamp; |
| 823 | active = 1; |
| 824 | status = AudioDeviceStart(adid, adioproc); |
| 825 | if(status) |
| 826 | fatal(0, "AudioDeviceStart: %d", (int)status); |
| 827 | /* Wait until the buffer empties out */ |
| 828 | while(nsamples >= minbuffer) |
| 829 | pthread_cond_wait(&cond, &lock); |
| 830 | /* Stop playing for a bit until the buffer re-fills */ |
| 831 | status = AudioDeviceStop(adid, adioproc); |
| 832 | if(status) |
| 833 | fatal(0, "AudioDeviceStop: %d", (int)status); |
| 834 | active = 0; |
| 835 | /* Go back round */ |
| 836 | } |
| 837 | } |
| 838 | #else |
| 839 | # error No known audio API |
| 840 | #endif |
| 841 | } |
| 842 | |
| 843 | /* display usage message and terminate */ |
| 844 | static void help(void) { |
| 845 | xprintf("Usage:\n" |
| 846 | " disorder-playrtp [OPTIONS] ADDRESS [PORT]\n" |
| 847 | "Options:\n" |
| 848 | " --device, -D DEVICE Output device\n" |
| 849 | " --min, -m FRAMES Buffer low water mark\n" |
| 850 | " --buffer, -b FRAMES Buffer high water mark\n" |
| 851 | " --max, -x FRAMES Buffer maximum size\n" |
| 852 | " --help, -h Display usage message\n" |
| 853 | " --version, -V Display version number\n" |
| 854 | ); |
| 855 | xfclose(stdout); |
| 856 | exit(0); |
| 857 | } |
| 858 | |
| 859 | /* display version number and terminate */ |
| 860 | static void version(void) { |
| 861 | xprintf("disorder-playrtp version %s\n", disorder_version_string); |
| 862 | xfclose(stdout); |
| 863 | exit(0); |
| 864 | } |
| 865 | |
| 866 | int main(int argc, char **argv) { |
| 867 | int n; |
| 868 | struct addrinfo *res; |
| 869 | struct stringlist sl; |
| 870 | char *sockname; |
| 871 | |
| 872 | static const struct addrinfo prefs = { |
| 873 | AI_PASSIVE, |
| 874 | PF_INET, |
| 875 | SOCK_DGRAM, |
| 876 | IPPROTO_UDP, |
| 877 | 0, |
| 878 | 0, |
| 879 | 0, |
| 880 | 0 |
| 881 | }; |
| 882 | |
| 883 | mem_init(); |
| 884 | if(!setlocale(LC_CTYPE, "")) fatal(errno, "error calling setlocale"); |
| 885 | while((n = getopt_long(argc, argv, "hVdD:m:b:x:L:", options, 0)) >= 0) { |
| 886 | switch(n) { |
| 887 | case 'h': help(); |
| 888 | case 'V': version(); |
| 889 | case 'd': debugging = 1; break; |
| 890 | case 'D': device = optarg; break; |
| 891 | case 'm': minbuffer = 2 * atol(optarg); break; |
| 892 | case 'b': readahead = 2 * atol(optarg); break; |
| 893 | case 'x': maxbuffer = 2 * atol(optarg); break; |
| 894 | case 'L': logfp = fopen(optarg, "w"); break; |
| 895 | default: fatal(0, "invalid option"); |
| 896 | } |
| 897 | } |
| 898 | if(!maxbuffer) |
| 899 | maxbuffer = 4 * readahead; |
| 900 | argc -= optind; |
| 901 | argv += optind; |
| 902 | if(argc < 1 || argc > 2) |
| 903 | fatal(0, "usage: disorder-playrtp [OPTIONS] ADDRESS [PORT]"); |
| 904 | sl.n = argc; |
| 905 | sl.s = argv; |
| 906 | /* Listen for inbound audio data */ |
| 907 | if(!(res = get_address(&sl, &prefs, &sockname))) |
| 908 | exit(1); |
| 909 | if((rtpfd = socket(res->ai_family, |
| 910 | res->ai_socktype, |
| 911 | res->ai_protocol)) < 0) |
| 912 | fatal(errno, "error creating socket"); |
| 913 | if(bind(rtpfd, res->ai_addr, res->ai_addrlen) < 0) |
| 914 | fatal(errno, "error binding socket to %s", sockname); |
| 915 | play_rtp(); |
| 916 | return 0; |
| 917 | } |
| 918 | |
| 919 | /* |
| 920 | Local Variables: |
| 921 | c-basic-offset:2 |
| 922 | comment-column:40 |
| 923 | fill-column:79 |
| 924 | indent-tabs-mode:nil |
| 925 | End: |
| 926 | */ |