| 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) two threads. listen_thread() is responsible for |
| 30 | * reading RTP packets off the wire and adding them to the binary heap @ref |
| 31 | * packets, assuming they are basically sound. |
| 32 | * |
| 33 | * The main thread is responsible for actually playing audio. In ALSA this |
| 34 | * means it waits until ALSA says it's ready for more audio which it then |
| 35 | * plays. |
| 36 | * |
| 37 | * InCore Audio the main thread is only responsible for starting and stopping |
| 38 | * play: the system does the actual playback in its own private thread, and |
| 39 | * calls adioproc() to fetch the audio data. |
| 40 | * |
| 41 | * Sometimes it happens that there is no audio available to play. This may |
| 42 | * because the server went away, or a packet was dropped, or the server |
| 43 | * deliberately did not send any sound because it encountered a silence. |
| 44 | */ |
| 45 | |
| 46 | #include <config.h> |
| 47 | #include "types.h" |
| 48 | |
| 49 | #include <getopt.h> |
| 50 | #include <stdio.h> |
| 51 | #include <stdlib.h> |
| 52 | #include <sys/socket.h> |
| 53 | #include <sys/types.h> |
| 54 | #include <sys/socket.h> |
| 55 | #include <netdb.h> |
| 56 | #include <pthread.h> |
| 57 | #include <locale.h> |
| 58 | #include <sys/uio.h> |
| 59 | #include <string.h> |
| 60 | |
| 61 | #include "log.h" |
| 62 | #include "mem.h" |
| 63 | #include "configuration.h" |
| 64 | #include "addr.h" |
| 65 | #include "syscalls.h" |
| 66 | #include "rtp.h" |
| 67 | #include "defs.h" |
| 68 | #include "vector.h" |
| 69 | #include "heap.h" |
| 70 | |
| 71 | #if HAVE_COREAUDIO_AUDIOHARDWARE_H |
| 72 | # include <CoreAudio/AudioHardware.h> |
| 73 | #endif |
| 74 | #if API_ALSA |
| 75 | #include <alsa/asoundlib.h> |
| 76 | #endif |
| 77 | |
| 78 | #define readahead linux_headers_are_borked |
| 79 | |
| 80 | /** @brief RTP socket */ |
| 81 | static int rtpfd; |
| 82 | |
| 83 | /** @brief Log output */ |
| 84 | static FILE *logfp; |
| 85 | |
| 86 | /** @brief Output device */ |
| 87 | static const char *device; |
| 88 | |
| 89 | /** @brief Maximum samples per packet we'll support |
| 90 | * |
| 91 | * NB that two channels = two samples in this program. |
| 92 | */ |
| 93 | #define MAXSAMPLES 2048 |
| 94 | |
| 95 | /** @brief Minimum low watermark |
| 96 | * |
| 97 | * We'll stop playing if there's only this many samples in the buffer. */ |
| 98 | static unsigned minbuffer = 2 * 44100 / 10; /* 0.2 seconds */ |
| 99 | |
| 100 | /** @brief Buffer high watermark |
| 101 | * |
| 102 | * We'll only start playing when this many samples are available. */ |
| 103 | static unsigned readahead = 2 * 2 * 44100; |
| 104 | |
| 105 | /** @brief Maximum buffer size |
| 106 | * |
| 107 | * We'll stop reading from the network if we have this many samples. */ |
| 108 | static unsigned maxbuffer; |
| 109 | |
| 110 | /** @brief Number of samples to infill by in one go |
| 111 | * |
| 112 | * This is an upper bound - in practice we expect the underlying audio API to |
| 113 | * only ask for a much smaller number of samples in any one go. |
| 114 | */ |
| 115 | #define INFILL_SAMPLES (44100 * 2) /* 1s */ |
| 116 | |
| 117 | /** @brief Received packet |
| 118 | * |
| 119 | * Received packets are kept in a binary heap (see @ref pheap) ordered by |
| 120 | * timestamp. |
| 121 | */ |
| 122 | struct packet { |
| 123 | /** @brief Number of samples in this packet */ |
| 124 | uint32_t nsamples; |
| 125 | |
| 126 | /** @brief Timestamp from RTP packet |
| 127 | * |
| 128 | * NB that "timestamps" are really sample counters. Use lt() or lt_packet() |
| 129 | * to compare timestamps. |
| 130 | */ |
| 131 | uint32_t timestamp; |
| 132 | |
| 133 | /** @brief Flags |
| 134 | * |
| 135 | * Valid values are: |
| 136 | * - @ref IDLE - the idle bit was set in the RTP packet |
| 137 | */ |
| 138 | unsigned flags; |
| 139 | /** @brief idle bit set in RTP packet*/ |
| 140 | #define IDLE 0x0001 |
| 141 | |
| 142 | /** @brief Raw sample data |
| 143 | * |
| 144 | * Only the first @p nsamples samples are defined; the rest is uninitialized |
| 145 | * data. |
| 146 | */ |
| 147 | uint16_t samples_raw[MAXSAMPLES]; |
| 148 | }; |
| 149 | |
| 150 | /** @brief Return true iff \f$a < b\f$ in sequence-space arithmetic |
| 151 | * |
| 152 | * Specifically it returns true if \f$(a-b) mod 2^{32} < 2^{31}\f$. |
| 153 | * |
| 154 | * See also lt_packet(). |
| 155 | */ |
| 156 | static inline int lt(uint32_t a, uint32_t b) { |
| 157 | return (uint32_t)(a - b) & 0x80000000; |
| 158 | } |
| 159 | |
| 160 | /** @brief Return true iff a >= b in sequence-space arithmetic */ |
| 161 | static inline int ge(uint32_t a, uint32_t b) { |
| 162 | return !lt(a, b); |
| 163 | } |
| 164 | |
| 165 | /** @brief Return true iff a > b in sequence-space arithmetic */ |
| 166 | static inline int gt(uint32_t a, uint32_t b) { |
| 167 | return lt(b, a); |
| 168 | } |
| 169 | |
| 170 | /** @brief Return true iff a <= b in sequence-space arithmetic */ |
| 171 | static inline int le(uint32_t a, uint32_t b) { |
| 172 | return !lt(b, a); |
| 173 | } |
| 174 | |
| 175 | /** @brief Ordering for packets, used by @ref pheap */ |
| 176 | static inline int lt_packet(const struct packet *a, const struct packet *b) { |
| 177 | return lt(a->timestamp, b->timestamp); |
| 178 | } |
| 179 | |
| 180 | /** @struct pheap |
| 181 | * @brief Binary heap of packets ordered by timestamp */ |
| 182 | HEAP_TYPE(pheap, struct packet *, lt_packet); |
| 183 | |
| 184 | /** @brief Binary heap of received packets */ |
| 185 | static struct pheap packets; |
| 186 | |
| 187 | /** @brief Total number of samples available */ |
| 188 | static unsigned long nsamples; |
| 189 | |
| 190 | /** @brief Timestamp of next packet to play. |
| 191 | * |
| 192 | * This is set to the timestamp of the last packet, plus the number of |
| 193 | * samples it contained. Only valid if @ref active is nonzero. |
| 194 | */ |
| 195 | static uint32_t next_timestamp; |
| 196 | |
| 197 | /** @brief True if actively playing |
| 198 | * |
| 199 | * This is true when playing and false when just buffering. */ |
| 200 | static int active; |
| 201 | |
| 202 | /** @brief Structure of free packet list */ |
| 203 | union free_packet { |
| 204 | struct packet p; |
| 205 | union free_packet *next; |
| 206 | }; |
| 207 | |
| 208 | /** @brief Linked list of free packets |
| 209 | * |
| 210 | * This is a linked list of formerly used packets. For preference we re-use |
| 211 | * packets that have already been used rather than unused ones, to limit the |
| 212 | * size of the program's working set. If there are no free packets in the list |
| 213 | * we try @ref next_free_packet instead. |
| 214 | * |
| 215 | * Must hold @ref lock when accessing this. |
| 216 | */ |
| 217 | static union free_packet *free_packets; |
| 218 | |
| 219 | /** @brief Array of new free packets |
| 220 | * |
| 221 | * There are @ref count_free_packets ready to use at this address. If there |
| 222 | * are none left we allocate more memory. |
| 223 | * |
| 224 | * Must hold @ref lock when accessing this. |
| 225 | */ |
| 226 | static union free_packet *next_free_packet; |
| 227 | |
| 228 | /** @brief Count of new free packets at @ref next_free_packet |
| 229 | * |
| 230 | * Must hold @ref lock when accessing this. |
| 231 | */ |
| 232 | static size_t count_free_packets; |
| 233 | |
| 234 | /** @brief Lock protecting @ref packets |
| 235 | * |
| 236 | * This also protects the packet memory allocation infrastructure, @ref |
| 237 | * free_packets and @ref next_free_packet. */ |
| 238 | static pthread_mutex_t lock = PTHREAD_MUTEX_INITIALIZER; |
| 239 | |
| 240 | /** @brief Condition variable signalled whenever @ref packets is changed */ |
| 241 | static pthread_cond_t cond = PTHREAD_COND_INITIALIZER; |
| 242 | |
| 243 | static const struct option options[] = { |
| 244 | { "help", no_argument, 0, 'h' }, |
| 245 | { "version", no_argument, 0, 'V' }, |
| 246 | { "debug", no_argument, 0, 'd' }, |
| 247 | { "device", required_argument, 0, 'D' }, |
| 248 | { "min", required_argument, 0, 'm' }, |
| 249 | { "max", required_argument, 0, 'x' }, |
| 250 | { "buffer", required_argument, 0, 'b' }, |
| 251 | { 0, 0, 0, 0 } |
| 252 | }; |
| 253 | |
| 254 | /** @brief Return a new packet |
| 255 | * |
| 256 | * Assumes that @ref lock is held. */ |
| 257 | static struct packet *new_packet(void) { |
| 258 | struct packet *p; |
| 259 | |
| 260 | if(free_packets) { |
| 261 | p = &free_packets->p; |
| 262 | free_packets = free_packets->next; |
| 263 | } else { |
| 264 | if(!count_free_packets) { |
| 265 | next_free_packet = xcalloc(1024, sizeof (union free_packet)); |
| 266 | count_free_packets = 1024; |
| 267 | } |
| 268 | p = &(next_free_packet++)->p; |
| 269 | --count_free_packets; |
| 270 | } |
| 271 | return p; |
| 272 | } |
| 273 | |
| 274 | /** @brief Free a packet |
| 275 | * |
| 276 | * Assumes that @ref lock is held. */ |
| 277 | static void free_packet(struct packet *p) { |
| 278 | union free_packet *u = (union free_packet *)p; |
| 279 | u->next = free_packets; |
| 280 | free_packets = u; |
| 281 | } |
| 282 | |
| 283 | /** @brief Drop the first packet |
| 284 | * |
| 285 | * Assumes that @ref lock is held. |
| 286 | */ |
| 287 | static void drop_first_packet(void) { |
| 288 | if(pheap_count(&packets)) { |
| 289 | struct packet *const p = pheap_remove(&packets); |
| 290 | nsamples -= p->nsamples; |
| 291 | free_packet(p); |
| 292 | pthread_cond_broadcast(&cond); |
| 293 | } |
| 294 | } |
| 295 | |
| 296 | /** @brief Background thread collecting samples |
| 297 | * |
| 298 | * This function collects samples, perhaps converts them to the target format, |
| 299 | * and adds them to the packet list. |
| 300 | * |
| 301 | * It is crucial that the gap between successive calls to read() is as small as |
| 302 | * possible: otherwise packets will be dropped. |
| 303 | * |
| 304 | * We use a binary heap to ensure that the unavoidable effort is at worst |
| 305 | * logarithmic in the total number of packets - in fact if packets are mostly |
| 306 | * received in order then we will largely do constant work per packet since the |
| 307 | * newest packet will always be last. |
| 308 | * |
| 309 | * Of more concern is that we must acquire the lock on the heap to add a packet |
| 310 | * to it. If this proves a problem in practice then the answer would be |
| 311 | * (probably doubly) linked list with new packets added the end and a second |
| 312 | * thread which reads packets off the list and adds them to the heap. |
| 313 | * |
| 314 | * We keep memory allocation (mostly) very fast by keeping pre-allocated |
| 315 | * packets around; see @ref new_packet(). |
| 316 | */ |
| 317 | static void *listen_thread(void attribute((unused)) *arg) { |
| 318 | struct packet *p = 0; |
| 319 | int n; |
| 320 | struct rtp_header header; |
| 321 | uint16_t seq; |
| 322 | uint32_t timestamp; |
| 323 | struct iovec iov[2]; |
| 324 | |
| 325 | for(;;) { |
| 326 | if(!p) { |
| 327 | pthread_mutex_lock(&lock); |
| 328 | p = new_packet(); |
| 329 | pthread_mutex_unlock(&lock); |
| 330 | } |
| 331 | iov[0].iov_base = &header; |
| 332 | iov[0].iov_len = sizeof header; |
| 333 | iov[1].iov_base = p->samples_raw; |
| 334 | iov[1].iov_len = sizeof p->samples_raw / sizeof *p->samples_raw; |
| 335 | n = readv(rtpfd, iov, 2); |
| 336 | if(n < 0) { |
| 337 | switch(errno) { |
| 338 | case EINTR: |
| 339 | continue; |
| 340 | default: |
| 341 | fatal(errno, "error reading from socket"); |
| 342 | } |
| 343 | } |
| 344 | /* Ignore too-short packets */ |
| 345 | if((size_t)n <= sizeof (struct rtp_header)) { |
| 346 | info("ignored a short packet"); |
| 347 | continue; |
| 348 | } |
| 349 | timestamp = htonl(header.timestamp); |
| 350 | seq = htons(header.seq); |
| 351 | /* Ignore packets in the past */ |
| 352 | if(active && lt(timestamp, next_timestamp)) { |
| 353 | info("dropping old packet, timestamp=%"PRIx32" < %"PRIx32, |
| 354 | timestamp, next_timestamp); |
| 355 | continue; |
| 356 | } |
| 357 | pthread_mutex_lock(&lock); |
| 358 | p->flags = 0; |
| 359 | p->timestamp = timestamp; |
| 360 | /* Convert to target format */ |
| 361 | if(header.mpt & 0x80) |
| 362 | p->flags |= IDLE; |
| 363 | switch(header.mpt & 0x7F) { |
| 364 | case 10: |
| 365 | p->nsamples = (n - sizeof header) / sizeof(uint16_t); |
| 366 | break; |
| 367 | /* TODO support other RFC3551 media types (when the speaker does) */ |
| 368 | default: |
| 369 | fatal(0, "unsupported RTP payload type %d", |
| 370 | header.mpt & 0x7F); |
| 371 | } |
| 372 | if(logfp) |
| 373 | fprintf(logfp, "sequence %u timestamp %"PRIx32" length %"PRIx32" end %"PRIx32"\n", |
| 374 | seq, timestamp, p->nsamples, timestamp + p->nsamples); |
| 375 | /* Stop reading if we've reached the maximum. |
| 376 | * |
| 377 | * This is rather unsatisfactory: it means that if packets get heavily |
| 378 | * out of order then we guarantee dropouts. But for now... */ |
| 379 | if(nsamples >= maxbuffer) { |
| 380 | info("Buffer full"); |
| 381 | while(nsamples >= maxbuffer) |
| 382 | pthread_cond_wait(&cond, &lock); |
| 383 | } |
| 384 | /* Add the packet to the heap */ |
| 385 | pheap_insert(&packets, p); |
| 386 | nsamples += p->nsamples; |
| 387 | /* We'll need a new packet */ |
| 388 | p = 0; |
| 389 | pthread_cond_broadcast(&cond); |
| 390 | pthread_mutex_unlock(&lock); |
| 391 | } |
| 392 | } |
| 393 | |
| 394 | /** @brief Return true if @p p contains @p timestamp |
| 395 | * |
| 396 | * Containment implies that a sample @p timestamp exists within the packet. |
| 397 | */ |
| 398 | static inline int contains(const struct packet *p, uint32_t timestamp) { |
| 399 | const uint32_t packet_start = p->timestamp; |
| 400 | const uint32_t packet_end = p->timestamp + p->nsamples; |
| 401 | |
| 402 | return (ge(timestamp, packet_start) |
| 403 | && lt(timestamp, packet_end)); |
| 404 | } |
| 405 | |
| 406 | #if HAVE_COREAUDIO_AUDIOHARDWARE_H |
| 407 | /** @brief Callback from Core Audio */ |
| 408 | static OSStatus adioproc |
| 409 | (AudioDeviceID attribute((unused)) inDevice, |
| 410 | const AudioTimeStamp attribute((unused)) *inNow, |
| 411 | const AudioBufferList attribute((unused)) *inInputData, |
| 412 | const AudioTimeStamp attribute((unused)) *inInputTime, |
| 413 | AudioBufferList *outOutputData, |
| 414 | const AudioTimeStamp attribute((unused)) *inOutputTime, |
| 415 | void attribute((unused)) *inClientData) { |
| 416 | UInt32 nbuffers = outOutputData->mNumberBuffers; |
| 417 | AudioBuffer *ab = outOutputData->mBuffers; |
| 418 | const struct packet *p; |
| 419 | uint32_t samples_available; |
| 420 | struct timeval in, out; |
| 421 | |
| 422 | gettimeofday(&in, 0); |
| 423 | pthread_mutex_lock(&lock); |
| 424 | while(nbuffers > 0) { |
| 425 | float *samplesOut = ab->mData; |
| 426 | size_t samplesOutLeft = ab->mDataByteSize / sizeof (float); |
| 427 | |
| 428 | while(samplesOutLeft > 0) { |
| 429 | /* Look for a suitable packet, dropping any unsuitable ones along the |
| 430 | * way. Unsuitable packets are ones that are in the past. */ |
| 431 | while(pheap_count(&packets)) { |
| 432 | p = pheap_first(&packets); |
| 433 | if(le(p->timestamp + p->nsamples, next_timestamp)) |
| 434 | /* This packet is in the past. Drop it and try another one. */ |
| 435 | drop_first_packet(); |
| 436 | else |
| 437 | /* This packet is NOT in the past. (It might be in the future |
| 438 | * however.) */ |
| 439 | break; |
| 440 | } |
| 441 | p = pheap_count(&packets) ? pheap_first(&packets) : 0; |
| 442 | if(p && contains(p, next_timestamp)) { |
| 443 | if(p->flags & IDLE) |
| 444 | fprintf(stderr, "\nIDLE\n"); |
| 445 | /* This packet is ready to play */ |
| 446 | const uint32_t packet_end = p->timestamp + p->nsamples; |
| 447 | const uint32_t offset = next_timestamp - p->timestamp; |
| 448 | const uint16_t *ptr = (void *)(p->samples_raw + offset); |
| 449 | |
| 450 | samples_available = packet_end - next_timestamp; |
| 451 | if(samples_available > samplesOutLeft) |
| 452 | samples_available = samplesOutLeft; |
| 453 | next_timestamp += samples_available; |
| 454 | samplesOutLeft -= samples_available; |
| 455 | while(samples_available-- > 0) |
| 456 | *samplesOut++ = (int16_t)ntohs(*ptr++) * (0.5 / 32767); |
| 457 | /* We don't bother junking the packet - that'll be dealt with next time |
| 458 | * round */ |
| 459 | write(2, ".", 1); |
| 460 | } else { |
| 461 | /* No packet is ready to play (and there might be no packet at all) */ |
| 462 | samples_available = p ? p->timestamp - next_timestamp |
| 463 | : samplesOutLeft; |
| 464 | if(samples_available > samplesOutLeft) |
| 465 | samples_available = samplesOutLeft; |
| 466 | //info("infill by %"PRIu32, samples_available); |
| 467 | /* Conveniently the buffer is 0 to start with */ |
| 468 | next_timestamp += samples_available; |
| 469 | samplesOut += samples_available; |
| 470 | samplesOutLeft -= samples_available; |
| 471 | write(2, "?", 1); |
| 472 | } |
| 473 | } |
| 474 | ++ab; |
| 475 | --nbuffers; |
| 476 | } |
| 477 | pthread_mutex_unlock(&lock); |
| 478 | gettimeofday(&out, 0); |
| 479 | { |
| 480 | static double max; |
| 481 | double thistime = (out.tv_sec - in.tv_sec) + (out.tv_usec - in.tv_usec) / 1000000.0; |
| 482 | if(thistime > max) |
| 483 | fprintf(stderr, "adioproc: %8.8fs\n", max = thistime); |
| 484 | } |
| 485 | return 0; |
| 486 | } |
| 487 | #endif |
| 488 | |
| 489 | |
| 490 | #if API_ALSA |
| 491 | /** @brief PCM handle */ |
| 492 | static snd_pcm_t *pcm; |
| 493 | |
| 494 | /** @brief True when @ref pcm is up and running */ |
| 495 | static int alsa_prepared = 1; |
| 496 | |
| 497 | /** @brief Initialize @ref pcm */ |
| 498 | static void setup_alsa(void) { |
| 499 | snd_pcm_hw_params_t *hwparams; |
| 500 | snd_pcm_sw_params_t *swparams; |
| 501 | /* Only support one format for now */ |
| 502 | const int sample_format = SND_PCM_FORMAT_S16_BE; |
| 503 | unsigned rate = 44100; |
| 504 | const int channels = 2; |
| 505 | const int samplesize = channels * sizeof(uint16_t); |
| 506 | snd_pcm_uframes_t pcm_bufsize = MAXSAMPLES * samplesize * 3; |
| 507 | /* If we can write more than this many samples we'll get a wakeup */ |
| 508 | const int avail_min = 256; |
| 509 | int err; |
| 510 | |
| 511 | /* Open ALSA */ |
| 512 | if((err = snd_pcm_open(&pcm, |
| 513 | device ? device : "default", |
| 514 | SND_PCM_STREAM_PLAYBACK, |
| 515 | SND_PCM_NONBLOCK))) |
| 516 | fatal(0, "error from snd_pcm_open: %d", err); |
| 517 | /* Set up 'hardware' parameters */ |
| 518 | snd_pcm_hw_params_alloca(&hwparams); |
| 519 | if((err = snd_pcm_hw_params_any(pcm, hwparams)) < 0) |
| 520 | fatal(0, "error from snd_pcm_hw_params_any: %d", err); |
| 521 | if((err = snd_pcm_hw_params_set_access(pcm, hwparams, |
| 522 | SND_PCM_ACCESS_RW_INTERLEAVED)) < 0) |
| 523 | fatal(0, "error from snd_pcm_hw_params_set_access: %d", err); |
| 524 | if((err = snd_pcm_hw_params_set_format(pcm, hwparams, |
| 525 | sample_format)) < 0) |
| 526 | |
| 527 | fatal(0, "error from snd_pcm_hw_params_set_format (%d): %d", |
| 528 | sample_format, err); |
| 529 | if((err = snd_pcm_hw_params_set_rate_near(pcm, hwparams, &rate, 0)) < 0) |
| 530 | fatal(0, "error from snd_pcm_hw_params_set_rate (%d): %d", |
| 531 | rate, err); |
| 532 | if((err = snd_pcm_hw_params_set_channels(pcm, hwparams, |
| 533 | channels)) < 0) |
| 534 | fatal(0, "error from snd_pcm_hw_params_set_channels (%d): %d", |
| 535 | channels, err); |
| 536 | if((err = snd_pcm_hw_params_set_buffer_size_near(pcm, hwparams, |
| 537 | &pcm_bufsize)) < 0) |
| 538 | fatal(0, "error from snd_pcm_hw_params_set_buffer_size (%d): %d", |
| 539 | MAXSAMPLES * samplesize * 3, err); |
| 540 | if((err = snd_pcm_hw_params(pcm, hwparams)) < 0) |
| 541 | fatal(0, "error calling snd_pcm_hw_params: %d", err); |
| 542 | /* Set up 'software' parameters */ |
| 543 | snd_pcm_sw_params_alloca(&swparams); |
| 544 | if((err = snd_pcm_sw_params_current(pcm, swparams)) < 0) |
| 545 | fatal(0, "error calling snd_pcm_sw_params_current: %d", err); |
| 546 | if((err = snd_pcm_sw_params_set_avail_min(pcm, swparams, avail_min)) < 0) |
| 547 | fatal(0, "error calling snd_pcm_sw_params_set_avail_min %d: %d", |
| 548 | avail_min, err); |
| 549 | if((err = snd_pcm_sw_params(pcm, swparams)) < 0) |
| 550 | fatal(0, "error calling snd_pcm_sw_params: %d", err); |
| 551 | } |
| 552 | |
| 553 | /** @brief Wait until ALSA wants some audio */ |
| 554 | static void wait_alsa(void) { |
| 555 | struct pollfd fds[64]; |
| 556 | int nfds, err; |
| 557 | unsigned short events; |
| 558 | |
| 559 | for(;;) { |
| 560 | do { |
| 561 | if((nfds = snd_pcm_poll_descriptors(pcm, |
| 562 | fds, sizeof fds / sizeof *fds)) < 0) |
| 563 | fatal(0, "error calling snd_pcm_poll_descriptors: %d", nfds); |
| 564 | } while(poll(fds, nfds, -1) < 0 && errno == EINTR); |
| 565 | if((err = snd_pcm_poll_descriptors_revents(pcm, fds, nfds, &events))) |
| 566 | fatal(0, "error calling snd_pcm_poll_descriptors_revents: %d", err); |
| 567 | if(events & POLLOUT) |
| 568 | return; |
| 569 | } |
| 570 | } |
| 571 | |
| 572 | /** @brief Play some sound via ALSA |
| 573 | * @param s Pointer to sample data |
| 574 | * @param n Number of samples |
| 575 | * @return 0 on success, -1 on non-fatal error |
| 576 | */ |
| 577 | static int alsa_writei(const void *s, size_t n) { |
| 578 | /* Do the write */ |
| 579 | const snd_pcm_sframes_t frames_written = snd_pcm_writei(pcm, s, n / 2); |
| 580 | if(frames_written < 0) { |
| 581 | /* Something went wrong */ |
| 582 | switch(frames_written) { |
| 583 | case -EAGAIN: |
| 584 | write(2, "#", 1); |
| 585 | return 0; |
| 586 | case -EPIPE: |
| 587 | error(0, "error calling snd_pcm_writei: %ld", |
| 588 | (long)frames_written); |
| 589 | return -1; |
| 590 | default: |
| 591 | fatal(0, "error calling snd_pcm_writei: %ld", |
| 592 | (long)frames_written); |
| 593 | } |
| 594 | } else { |
| 595 | /* Success */ |
| 596 | next_timestamp += frames_written * 2; |
| 597 | return 0; |
| 598 | } |
| 599 | } |
| 600 | |
| 601 | /** @brief Play the relevant part of a packet |
| 602 | * @param p Packet to play |
| 603 | * @return 0 on success, -1 on non-fatal error |
| 604 | */ |
| 605 | static int alsa_play(const struct packet *p) { |
| 606 | if(p->flags & IDLE) |
| 607 | write(2, "I", 1); |
| 608 | write(2, ".", 1); |
| 609 | return alsa_writei(p->samples_raw + next_timestamp - p->timestamp, |
| 610 | (p->timestamp + p->nsamples) - next_timestamp); |
| 611 | } |
| 612 | |
| 613 | /** @brief Play some silence |
| 614 | * @param p Next packet or NULL |
| 615 | * @return 0 on success, -1 on non-fatal error |
| 616 | */ |
| 617 | static int alsa_infill(const struct packet *p) { |
| 618 | static const uint16_t zeros[INFILL_SAMPLES]; |
| 619 | size_t samples_available = INFILL_SAMPLES; |
| 620 | |
| 621 | if(p && samples_available > p->timestamp - next_timestamp) |
| 622 | samples_available = p->timestamp - next_timestamp; |
| 623 | write(2, "?", 1); |
| 624 | return alsa_writei(zeros, samples_available); |
| 625 | } |
| 626 | |
| 627 | /** @brief Reset ALSA state after we lost synchronization */ |
| 628 | static void alsa_reset(int hard_reset) { |
| 629 | int err; |
| 630 | |
| 631 | if((err = snd_pcm_nonblock(pcm, 0))) |
| 632 | fatal(0, "error calling snd_pcm_nonblock: %d", err); |
| 633 | if(hard_reset) { |
| 634 | if((err = snd_pcm_drop(pcm))) |
| 635 | fatal(0, "error calling snd_pcm_drop: %d", err); |
| 636 | } else |
| 637 | if((err = snd_pcm_drain(pcm))) |
| 638 | fatal(0, "error calling snd_pcm_drain: %d", err); |
| 639 | if((err = snd_pcm_nonblock(pcm, 1))) |
| 640 | fatal(0, "error calling snd_pcm_nonblock: %d", err); |
| 641 | alsa_prepared = 0; |
| 642 | } |
| 643 | #endif |
| 644 | |
| 645 | /** @brief Wait until the buffer is adequately full |
| 646 | * |
| 647 | * Must be called with @ref lock held. |
| 648 | */ |
| 649 | static void fill_buffer(void) { |
| 650 | info("Buffering..."); |
| 651 | while(nsamples < readahead) |
| 652 | pthread_cond_wait(&cond, &lock); |
| 653 | next_timestamp = pheap_first(&packets)->timestamp; |
| 654 | active = 1; |
| 655 | } |
| 656 | |
| 657 | /** @brief Find next packet |
| 658 | * @return Packet to play or NULL if none found |
| 659 | * |
| 660 | * The return packet is merely guaranteed not to be in the past: it might be |
| 661 | * the first packet in the future rather than one that is actually suitable to |
| 662 | * play. |
| 663 | * |
| 664 | * Must be called with @ref lock held. |
| 665 | */ |
| 666 | static struct packet *next_packet(void) { |
| 667 | while(pheap_count(&packets)) { |
| 668 | struct packet *const p = pheap_first(&packets); |
| 669 | if(le(p->timestamp + p->nsamples, next_timestamp)) { |
| 670 | /* This packet is in the past. Drop it and try another one. */ |
| 671 | drop_first_packet(); |
| 672 | } else |
| 673 | /* This packet is NOT in the past. (It might be in the future |
| 674 | * however.) */ |
| 675 | return p; |
| 676 | } |
| 677 | return 0; |
| 678 | } |
| 679 | |
| 680 | /** @brief Play an RTP stream |
| 681 | * |
| 682 | * This is the guts of the program. It is responsible for: |
| 683 | * - starting the listening thread |
| 684 | * - opening the audio device |
| 685 | * - reading ahead to build up a buffer |
| 686 | * - arranging for audio to be played |
| 687 | * - detecting when the buffer has got too small and re-buffering |
| 688 | */ |
| 689 | static void play_rtp(void) { |
| 690 | pthread_t ltid; |
| 691 | |
| 692 | /* We receive and convert audio data in a background thread */ |
| 693 | pthread_create(<id, 0, listen_thread, 0); |
| 694 | #if API_ALSA |
| 695 | { |
| 696 | struct packet *p; |
| 697 | int escape, err; |
| 698 | |
| 699 | /* Open the sound device */ |
| 700 | setup_alsa(); |
| 701 | pthread_mutex_lock(&lock); |
| 702 | for(;;) { |
| 703 | /* Wait for the buffer to fill up a bit */ |
| 704 | fill_buffer(); |
| 705 | if(!alsa_prepared) { |
| 706 | if((err = snd_pcm_prepare(pcm))) |
| 707 | fatal(0, "error calling snd_pcm_prepare: %d", err); |
| 708 | alsa_prepared = 1; |
| 709 | } |
| 710 | escape = 0; |
| 711 | info("Playing..."); |
| 712 | /* Keep playing until the buffer empties out, or ALSA tells us to get |
| 713 | * lost */ |
| 714 | while(nsamples >= minbuffer && !escape) { |
| 715 | /* Wait for ALSA to ask us for more data */ |
| 716 | pthread_mutex_unlock(&lock); |
| 717 | wait_alsa(); |
| 718 | pthread_mutex_lock(&lock); |
| 719 | /* ALSA is ready for more data, find something to play */ |
| 720 | p = next_packet(); |
| 721 | /* Play it or play some silence */ |
| 722 | if(contains(p, next_timestamp)) |
| 723 | escape = alsa_play(p); |
| 724 | else |
| 725 | escape = alsa_infill(p); |
| 726 | } |
| 727 | active = 0; |
| 728 | /* We stop playing for a bit until the buffer re-fills */ |
| 729 | pthread_mutex_unlock(&lock); |
| 730 | alsa_reset(escape); |
| 731 | pthread_mutex_lock(&lock); |
| 732 | } |
| 733 | |
| 734 | } |
| 735 | #elif HAVE_COREAUDIO_AUDIOHARDWARE_H |
| 736 | { |
| 737 | OSStatus status; |
| 738 | UInt32 propertySize; |
| 739 | AudioDeviceID adid; |
| 740 | AudioStreamBasicDescription asbd; |
| 741 | |
| 742 | /* If this looks suspiciously like libao's macosx driver there's an |
| 743 | * excellent reason for that... */ |
| 744 | |
| 745 | /* TODO report errors as strings not numbers */ |
| 746 | propertySize = sizeof adid; |
| 747 | status = AudioHardwareGetProperty(kAudioHardwarePropertyDefaultOutputDevice, |
| 748 | &propertySize, &adid); |
| 749 | if(status) |
| 750 | fatal(0, "AudioHardwareGetProperty: %d", (int)status); |
| 751 | if(adid == kAudioDeviceUnknown) |
| 752 | fatal(0, "no output device"); |
| 753 | propertySize = sizeof asbd; |
| 754 | status = AudioDeviceGetProperty(adid, 0, false, |
| 755 | kAudioDevicePropertyStreamFormat, |
| 756 | &propertySize, &asbd); |
| 757 | if(status) |
| 758 | fatal(0, "AudioHardwareGetProperty: %d", (int)status); |
| 759 | D(("mSampleRate %f", asbd.mSampleRate)); |
| 760 | D(("mFormatID %08lx", asbd.mFormatID)); |
| 761 | D(("mFormatFlags %08lx", asbd.mFormatFlags)); |
| 762 | D(("mBytesPerPacket %08lx", asbd.mBytesPerPacket)); |
| 763 | D(("mFramesPerPacket %08lx", asbd.mFramesPerPacket)); |
| 764 | D(("mBytesPerFrame %08lx", asbd.mBytesPerFrame)); |
| 765 | D(("mChannelsPerFrame %08lx", asbd.mChannelsPerFrame)); |
| 766 | D(("mBitsPerChannel %08lx", asbd.mBitsPerChannel)); |
| 767 | D(("mReserved %08lx", asbd.mReserved)); |
| 768 | if(asbd.mFormatID != kAudioFormatLinearPCM) |
| 769 | fatal(0, "audio device does not support kAudioFormatLinearPCM"); |
| 770 | status = AudioDeviceAddIOProc(adid, adioproc, 0); |
| 771 | if(status) |
| 772 | fatal(0, "AudioDeviceAddIOProc: %d", (int)status); |
| 773 | pthread_mutex_lock(&lock); |
| 774 | for(;;) { |
| 775 | /* Wait for the buffer to fill up a bit */ |
| 776 | fill_buffer(); |
| 777 | /* Start playing now */ |
| 778 | info("Playing..."); |
| 779 | next_timestamp = pheap_first(&packets)->timestamp; |
| 780 | active = 1; |
| 781 | status = AudioDeviceStart(adid, adioproc); |
| 782 | if(status) |
| 783 | fatal(0, "AudioDeviceStart: %d", (int)status); |
| 784 | /* Wait until the buffer empties out */ |
| 785 | while(nsamples >= minbuffer) |
| 786 | pthread_cond_wait(&cond, &lock); |
| 787 | /* Stop playing for a bit until the buffer re-fills */ |
| 788 | status = AudioDeviceStop(adid, adioproc); |
| 789 | if(status) |
| 790 | fatal(0, "AudioDeviceStop: %d", (int)status); |
| 791 | active = 0; |
| 792 | /* Go back round */ |
| 793 | } |
| 794 | } |
| 795 | #else |
| 796 | # error No known audio API |
| 797 | #endif |
| 798 | } |
| 799 | |
| 800 | /* display usage message and terminate */ |
| 801 | static void help(void) { |
| 802 | xprintf("Usage:\n" |
| 803 | " disorder-playrtp [OPTIONS] ADDRESS [PORT]\n" |
| 804 | "Options:\n" |
| 805 | " --device, -D DEVICE Output device\n" |
| 806 | " --min, -m FRAMES Buffer low water mark\n" |
| 807 | " --buffer, -b FRAMES Buffer high water mark\n" |
| 808 | " --max, -x FRAMES Buffer maximum size\n" |
| 809 | " --help, -h Display usage message\n" |
| 810 | " --version, -V Display version number\n" |
| 811 | ); |
| 812 | xfclose(stdout); |
| 813 | exit(0); |
| 814 | } |
| 815 | |
| 816 | /* display version number and terminate */ |
| 817 | static void version(void) { |
| 818 | xprintf("disorder-playrtp version %s\n", disorder_version_string); |
| 819 | xfclose(stdout); |
| 820 | exit(0); |
| 821 | } |
| 822 | |
| 823 | int main(int argc, char **argv) { |
| 824 | int n; |
| 825 | struct addrinfo *res; |
| 826 | struct stringlist sl; |
| 827 | char *sockname; |
| 828 | |
| 829 | static const struct addrinfo prefs = { |
| 830 | AI_PASSIVE, |
| 831 | PF_INET, |
| 832 | SOCK_DGRAM, |
| 833 | IPPROTO_UDP, |
| 834 | 0, |
| 835 | 0, |
| 836 | 0, |
| 837 | 0 |
| 838 | }; |
| 839 | |
| 840 | mem_init(); |
| 841 | if(!setlocale(LC_CTYPE, "")) fatal(errno, "error calling setlocale"); |
| 842 | while((n = getopt_long(argc, argv, "hVdD:m:b:x:L:", options, 0)) >= 0) { |
| 843 | switch(n) { |
| 844 | case 'h': help(); |
| 845 | case 'V': version(); |
| 846 | case 'd': debugging = 1; break; |
| 847 | case 'D': device = optarg; break; |
| 848 | case 'm': minbuffer = 2 * atol(optarg); break; |
| 849 | case 'b': readahead = 2 * atol(optarg); break; |
| 850 | case 'x': maxbuffer = 2 * atol(optarg); break; |
| 851 | case 'L': logfp = fopen(optarg, "w"); break; |
| 852 | default: fatal(0, "invalid option"); |
| 853 | } |
| 854 | } |
| 855 | if(!maxbuffer) |
| 856 | maxbuffer = 4 * readahead; |
| 857 | argc -= optind; |
| 858 | argv += optind; |
| 859 | if(argc < 1 || argc > 2) |
| 860 | fatal(0, "usage: disorder-playrtp [OPTIONS] ADDRESS [PORT]"); |
| 861 | sl.n = argc; |
| 862 | sl.s = argv; |
| 863 | /* Listen for inbound audio data */ |
| 864 | if(!(res = get_address(&sl, &prefs, &sockname))) |
| 865 | exit(1); |
| 866 | if((rtpfd = socket(res->ai_family, |
| 867 | res->ai_socktype, |
| 868 | res->ai_protocol)) < 0) |
| 869 | fatal(errno, "error creating socket"); |
| 870 | if(bind(rtpfd, res->ai_addr, res->ai_addrlen) < 0) |
| 871 | fatal(errno, "error binding socket to %s", sockname); |
| 872 | play_rtp(); |
| 873 | return 0; |
| 874 | } |
| 875 | |
| 876 | /* |
| 877 | Local Variables: |
| 878 | c-basic-offset:2 |
| 879 | comment-column:40 |
| 880 | fill-column:79 |
| 881 | indent-tabs-mode:nil |
| 882 | End: |
| 883 | */ |