| 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 | |
| 21 | #include <config.h> |
| 22 | #include "types.h" |
| 23 | |
| 24 | #include <getopt.h> |
| 25 | #include <stdio.h> |
| 26 | #include <stdlib.h> |
| 27 | #include <sys/socket.h> |
| 28 | #include <sys/types.h> |
| 29 | #include <sys/socket.h> |
| 30 | #include <netdb.h> |
| 31 | #include <pthread.h> |
| 32 | #include <locale.h> |
| 33 | #include <sys/uio.h> |
| 34 | |
| 35 | #include "log.h" |
| 36 | #include "mem.h" |
| 37 | #include "configuration.h" |
| 38 | #include "addr.h" |
| 39 | #include "syscalls.h" |
| 40 | #include "rtp.h" |
| 41 | #include "defs.h" |
| 42 | |
| 43 | #if HAVE_COREAUDIO_AUDIOHARDWARE_H |
| 44 | # include <CoreAudio/AudioHardware.h> |
| 45 | #endif |
| 46 | #if API_ALSA |
| 47 | #include <alsa/asoundlib.h> |
| 48 | #endif |
| 49 | |
| 50 | #define readahead linux_headers_are_borked |
| 51 | |
| 52 | /** @brief RTP socket */ |
| 53 | static int rtpfd; |
| 54 | |
| 55 | /** @brief Log output */ |
| 56 | static FILE *logfp; |
| 57 | |
| 58 | /** @brief Output device */ |
| 59 | static const char *device; |
| 60 | |
| 61 | /** @brief Maximum samples per packet we'll support |
| 62 | * |
| 63 | * NB that two channels = two samples in this program. |
| 64 | */ |
| 65 | #define MAXSAMPLES 2048 |
| 66 | |
| 67 | /** @brief Minimum low watermark |
| 68 | * |
| 69 | * We'll stop playing if there's only this many samples in the buffer. */ |
| 70 | static unsigned minbuffer = 2 * 44100 / 10; /* 0.2 seconds */ |
| 71 | |
| 72 | /** @brief Maximum sample size |
| 73 | * |
| 74 | * The maximum supported size (in bytes) of one sample. */ |
| 75 | #define MAXSAMPLESIZE 2 |
| 76 | |
| 77 | /** @brief Buffer high watermark |
| 78 | * |
| 79 | * We'll only start playing when this many samples are available. */ |
| 80 | static unsigned readahead = 2 * 2 * 44100; |
| 81 | |
| 82 | /** @brief Maximum buffer size |
| 83 | * |
| 84 | * We'll stop reading from the network if we have this many samples. */ |
| 85 | static unsigned maxbuffer; |
| 86 | |
| 87 | /** @brief Number of samples to infill by in one go */ |
| 88 | #define INFILL_SAMPLES (44100 * 2) /* 1s */ |
| 89 | |
| 90 | /** @brief Received packet */ |
| 91 | struct packet { |
| 92 | /** @brief Number of samples in this packet */ |
| 93 | uint32_t nsamples; |
| 94 | /** @brief Timestamp from RTP packet |
| 95 | * |
| 96 | * NB that "timestamps" are really sample counters.*/ |
| 97 | uint32_t timestamp; |
| 98 | /** @brief Raw sample data */ |
| 99 | unsigned char samples_raw[MAXSAMPLES * MAXSAMPLESIZE]; |
| 100 | }; |
| 101 | |
| 102 | /** @brief Total number of samples available */ |
| 103 | static unsigned long nsamples; |
| 104 | |
| 105 | /** @brief Mapping of sequence numbers to packets |
| 106 | * |
| 107 | * This isn't very efficient - 256KB on 32-bit machines, 512KB if you do a |
| 108 | * 64-bit build for some reason. It can be optimized later if need be. */ |
| 109 | static struct packet *packets[65536]; |
| 110 | |
| 111 | /** @brief Total number of packets */ |
| 112 | static unsigned npackets; |
| 113 | |
| 114 | /** @brief Timestamp of next packet to play. |
| 115 | * |
| 116 | * This is set to the timestamp of the last packet, plus the number of |
| 117 | * samples it contained. Only valid if @ref active is nonzero. |
| 118 | */ |
| 119 | static uint32_t next_timestamp; |
| 120 | |
| 121 | /** @brief True if actively playing |
| 122 | * |
| 123 | * This is true when playing and false when just buffering. */ |
| 124 | static int active; |
| 125 | |
| 126 | /** @brief Sequence number of next packet we expxect to play */ |
| 127 | static uint16_t sequence; |
| 128 | |
| 129 | /** @brief Structure of free packet list */ |
| 130 | union free_packet { |
| 131 | struct packet p; |
| 132 | union free_packet *next; |
| 133 | }; |
| 134 | |
| 135 | /** @brief Linked list of free packets */ |
| 136 | static union free_packet *free_packets; |
| 137 | |
| 138 | /** @brief Array of new free packets */ |
| 139 | static union free_packet *next_free_packet; |
| 140 | |
| 141 | /** @brief Count of new free packets */ |
| 142 | static size_t count_free_packets; |
| 143 | |
| 144 | /** @brief Lock protecting @ref packets */ |
| 145 | static pthread_mutex_t lock = PTHREAD_MUTEX_INITIALIZER; |
| 146 | |
| 147 | /** @brief Condition variable signalled whenever @ref packets is changed */ |
| 148 | static pthread_cond_t cond = PTHREAD_COND_INITIALIZER; |
| 149 | |
| 150 | static const struct option options[] = { |
| 151 | { "help", no_argument, 0, 'h' }, |
| 152 | { "version", no_argument, 0, 'V' }, |
| 153 | { "debug", no_argument, 0, 'd' }, |
| 154 | { "device", required_argument, 0, 'D' }, |
| 155 | { "min", required_argument, 0, 'm' }, |
| 156 | { "max", required_argument, 0, 'x' }, |
| 157 | { "buffer", required_argument, 0, 'b' }, |
| 158 | { 0, 0, 0, 0 } |
| 159 | }; |
| 160 | |
| 161 | /** @brief Return a new packet |
| 162 | * |
| 163 | * Assumes that @ref lock is held. */ |
| 164 | static struct packet *new_packet(void) { |
| 165 | struct packet *p; |
| 166 | |
| 167 | if(free_packets) { |
| 168 | p = &free_packets->p; |
| 169 | free_packets = free_packets->next; |
| 170 | } else { |
| 171 | if(!count_free_packets) { |
| 172 | next_free_packet = xcalloc(1024, sizeof (union free_packet)); |
| 173 | count_free_packets = 1024; |
| 174 | } |
| 175 | p = &(next_free_packet++)->p; |
| 176 | --count_free_packets; |
| 177 | } |
| 178 | return p; |
| 179 | } |
| 180 | |
| 181 | /** @brief Free a packet |
| 182 | * |
| 183 | * Assumes that @ref lock is held. */ |
| 184 | static void free_packet(struct packet *p) { |
| 185 | union free_packet *u = (union free_packet *)p; |
| 186 | u->next = free_packets; |
| 187 | free_packets = u; |
| 188 | } |
| 189 | |
| 190 | /** @brief Return true iff a < b in sequence-space arithmetic */ |
| 191 | static inline int lt(uint32_t a, uint32_t b) { |
| 192 | return (uint32_t)(a - b) & 0x80000000; |
| 193 | } |
| 194 | |
| 195 | /** @brief Return true iff a >= b in sequence-space arithmetic */ |
| 196 | static inline int ge(uint32_t a, uint32_t b) { |
| 197 | return !lt(a, b); |
| 198 | } |
| 199 | |
| 200 | /** @brief Return true iff a > b in sequence-space arithmetic */ |
| 201 | static inline int gt(uint32_t a, uint32_t b) { |
| 202 | return lt(b, a); |
| 203 | } |
| 204 | |
| 205 | /** @brief Return true iff a <= b in sequence-space arithmetic */ |
| 206 | static inline int le(uint32_t a, uint32_t b) { |
| 207 | return !lt(b, a); |
| 208 | } |
| 209 | |
| 210 | /** @brief Drop the packet at the head of the queue */ |
| 211 | static void drop_packet(unsigned sequence) { |
| 212 | if(packets[sequence]) { |
| 213 | nsamples -= packets[sequence]->nsamples; |
| 214 | free_packet(packets[sequence]); |
| 215 | packets[sequence] = 0; |
| 216 | pthread_cond_broadcast(&cond); |
| 217 | --npackets; |
| 218 | } |
| 219 | } |
| 220 | |
| 221 | /** @brief Background thread collecting samples |
| 222 | * |
| 223 | * This function collects samples, perhaps converts them to the target format, |
| 224 | * and adds them to the packet list. */ |
| 225 | static void *listen_thread(void attribute((unused)) *arg) { |
| 226 | struct packet *p = 0; |
| 227 | int n; |
| 228 | struct rtp_header header; |
| 229 | uint16_t seq; |
| 230 | uint32_t timestamp; |
| 231 | struct iovec iov[2]; |
| 232 | |
| 233 | for(;;) { |
| 234 | if(!p) { |
| 235 | pthread_mutex_lock(&lock); |
| 236 | p = new_packet(); |
| 237 | pthread_mutex_unlock(&lock); |
| 238 | } |
| 239 | iov[0].iov_base = &header; |
| 240 | iov[0].iov_len = sizeof header; |
| 241 | iov[1].iov_base = p->samples_raw; |
| 242 | iov[1].iov_len = sizeof p->samples_raw; |
| 243 | n = readv(rtpfd, iov, 2); |
| 244 | if(n < 0) { |
| 245 | switch(errno) { |
| 246 | case EINTR: |
| 247 | continue; |
| 248 | default: |
| 249 | fatal(errno, "error reading from socket"); |
| 250 | } |
| 251 | } |
| 252 | /* Ignore too-short packets */ |
| 253 | if((size_t)n <= sizeof (struct rtp_header)) { |
| 254 | info("ignored a short packet"); |
| 255 | continue; |
| 256 | } |
| 257 | timestamp = htonl(header.timestamp); |
| 258 | seq = htons(header.seq); |
| 259 | /* Ignore packets in the past */ |
| 260 | if(active && lt(timestamp, next_timestamp)) { |
| 261 | info("dropping old packet, timestamp=%"PRIx32" < %"PRIx32, |
| 262 | timestamp, next_timestamp); |
| 263 | continue; |
| 264 | } |
| 265 | pthread_mutex_lock(&lock); |
| 266 | p = new_packet(); |
| 267 | p->timestamp = timestamp; |
| 268 | /* Convert to target format */ |
| 269 | switch(header.mpt & 0x7F) { |
| 270 | case 10: |
| 271 | p->nsamples = (n - sizeof header) / sizeof(uint16_t); |
| 272 | /* ALSA can do any necessary conversion itself (though it might be better |
| 273 | * to do any necessary conversion in the background) */ |
| 274 | /* TODO we could readv into the buffer */ |
| 275 | break; |
| 276 | /* TODO support other RFC3551 media types (when the speaker does) */ |
| 277 | default: |
| 278 | fatal(0, "unsupported RTP payload type %d", |
| 279 | header.mpt & 0x7F); |
| 280 | } |
| 281 | if(logfp) |
| 282 | fprintf(logfp, "sequence %u timestamp %"PRIx32" length %"PRIx32" end %"PRIx32"\n", |
| 283 | seq, timestamp, p->nsamples, timestamp + p->nsamples); |
| 284 | /* Stop reading if we've reached the maximum. |
| 285 | * |
| 286 | * This is rather unsatisfactory: it means that if packets get heavily |
| 287 | * out of order then we guarantee dropouts. But for now... */ |
| 288 | if(nsamples >= maxbuffer) { |
| 289 | info("buffer full"); |
| 290 | while(nsamples >= maxbuffer) |
| 291 | pthread_cond_wait(&cond, &lock); |
| 292 | } |
| 293 | /* If there's a packet there already we overwrite it; perhaps it is left |
| 294 | * over from an earlier stage. */ |
| 295 | drop_packet(seq); |
| 296 | /* Record this packet */ |
| 297 | packets[seq] = p; |
| 298 | /* If we currently have no idea where to start playing, this is it */ |
| 299 | if(!npackets) |
| 300 | sequence = seq; |
| 301 | ++npackets; |
| 302 | nsamples += p->nsamples; |
| 303 | pthread_cond_broadcast(&cond); |
| 304 | pthread_mutex_unlock(&lock); |
| 305 | } |
| 306 | } |
| 307 | |
| 308 | /** @brief Return true if @p p contains @p timestamp */ |
| 309 | static inline int contains(const struct packet *p, uint32_t timestamp) { |
| 310 | const uint32_t packet_start = p->timestamp; |
| 311 | const uint32_t packet_end = p->timestamp + p->nsamples; |
| 312 | |
| 313 | return (ge(timestamp, packet_start) |
| 314 | && lt(timestamp, packet_end)); |
| 315 | } |
| 316 | |
| 317 | #if HAVE_COREAUDIO_AUDIOHARDWARE_H |
| 318 | /** @brief Callback from Core Audio */ |
| 319 | static OSStatus adioproc |
| 320 | (AudioDeviceID attribute((unused)) inDevice, |
| 321 | const AudioTimeStamp attribute((unused)) *inNow, |
| 322 | const AudioBufferList attribute((unused)) *inInputData, |
| 323 | const AudioTimeStamp attribute((unused)) *inInputTime, |
| 324 | AudioBufferList *outOutputData, |
| 325 | const AudioTimeStamp attribute((unused)) *inOutputTime, |
| 326 | void attribute((unused)) *inClientData) { |
| 327 | UInt32 nbuffers = outOutputData->mNumberBuffers; |
| 328 | AudioBuffer *ab = outOutputData->mBuffers; |
| 329 | const struct packet *p; |
| 330 | |
| 331 | pthread_mutex_lock(&lock); |
| 332 | while(nbuffers > 0) { |
| 333 | float *samplesOut = ab->mData; |
| 334 | size_t samplesOutLeft = ab->mDataByteSize / sizeof (float); |
| 335 | |
| 336 | while(samplesOutLeft > 0) { |
| 337 | /* Look for a suitable packet, dropping any unsuitable ones along the |
| 338 | * way. Unsuitable packets are ones that are in the past. */ |
| 339 | while(npackets |
| 340 | && (!packets[sequence] |
| 341 | || le(packets[sequence]->timestamp |
| 342 | + packets[sequence]->nsamples, |
| 343 | next_timestamp))) |
| 344 | drop_packet(sequence++); |
| 345 | p = packets[sequence]; |
| 346 | if(p) { |
| 347 | if(contains(p, next_timestamp)) { |
| 348 | /* This packet is suitable */ |
| 349 | const uint32_t packet_end = p->timestamp + p->nsamples; |
| 350 | const uint32_t offset = next_timestamp - p->timestamp; |
| 351 | const uint16_t *ptr = |
| 352 | (void *)(p->samples_raw + offset * sizeof (uint16_t)); |
| 353 | uint32_t samples_available = packet_end - next_timestamp; |
| 354 | if(samples_available > samplesOutLeft) |
| 355 | samples_available = samplesOutLeft; |
| 356 | next_timestamp += samples_available; |
| 357 | samplesOutLeft -= samples_available; |
| 358 | while(samples_available-- > 0) |
| 359 | *samplesOut++ = (int16_t)ntohs(*ptr++) * (0.5 / 32767); |
| 360 | /* We don't bother junking the packet or advancing sequence - that'll |
| 361 | * be dealt with next time round */ |
| 362 | continue; |
| 363 | } |
| 364 | } |
| 365 | /* We didn't find a suitable packet (though there might still be |
| 366 | * unsuitable ones). We infill with 0s. */ |
| 367 | if(p) { |
| 368 | /* There is a next packet, only infill up to that point */ |
| 369 | uint32_t samples_available = p->timestamp - next_timestamp; |
| 370 | |
| 371 | if(samples_available > samplesOutLeft) |
| 372 | samples_available = samplesOutLeft; |
| 373 | info("infill by %"PRIu32, samples_available); |
| 374 | /* Convniently the buffer is 0 to start with */ |
| 375 | next_timestamp += samples_available; |
| 376 | samplesOut += samples_available; |
| 377 | samplesOutLeft -= samples_available; |
| 378 | } else { |
| 379 | /* There's no next packet at all */ |
| 380 | info("infilled by %zu", samplesOutLeft); |
| 381 | next_timestamp += samplesOutLeft; |
| 382 | samplesOut += samplesOutLeft; |
| 383 | samplesOutLeft = 0; |
| 384 | } |
| 385 | } |
| 386 | ++ab; |
| 387 | --nbuffers; |
| 388 | } |
| 389 | pthread_mutex_unlock(&lock); |
| 390 | return 0; |
| 391 | } |
| 392 | #endif |
| 393 | |
| 394 | /** @brief Play an RTP stream |
| 395 | * |
| 396 | * This is the guts of the program. It is responsible for: |
| 397 | * - starting the listening thread |
| 398 | * - opening the audio device |
| 399 | * - reading ahead to build up a buffer |
| 400 | * - arranging for audio to be played |
| 401 | * - detecting when the buffer has got too small and re-buffering |
| 402 | */ |
| 403 | static void play_rtp(void) { |
| 404 | pthread_t ltid; |
| 405 | |
| 406 | /* We receive and convert audio data in a background thread */ |
| 407 | pthread_create(<id, 0, listen_thread, 0); |
| 408 | #if API_ALSA |
| 409 | { |
| 410 | snd_pcm_t *pcm; |
| 411 | snd_pcm_hw_params_t *hwparams; |
| 412 | snd_pcm_sw_params_t *swparams; |
| 413 | /* Only support one format for now */ |
| 414 | const int sample_format = SND_PCM_FORMAT_S16_BE; |
| 415 | unsigned rate = 44100; |
| 416 | const int channels = 2; |
| 417 | const int samplesize = channels * sizeof(uint16_t); |
| 418 | snd_pcm_uframes_t pcm_bufsize = MAXSAMPLES * samplesize * 3; |
| 419 | /* If we can write more than this many samples we'll get a wakeup */ |
| 420 | const int avail_min = 256; |
| 421 | snd_pcm_sframes_t frames_written; |
| 422 | size_t samples_written; |
| 423 | int prepared = 1; |
| 424 | int err; |
| 425 | int infilling = 0, escape = 0; |
| 426 | time_t logged, now; |
| 427 | uint32_t packet_start, packet_end; |
| 428 | |
| 429 | /* Open ALSA */ |
| 430 | if((err = snd_pcm_open(&pcm, |
| 431 | device ? device : "default", |
| 432 | SND_PCM_STREAM_PLAYBACK, |
| 433 | SND_PCM_NONBLOCK))) |
| 434 | fatal(0, "error from snd_pcm_open: %d", err); |
| 435 | /* Set up 'hardware' parameters */ |
| 436 | snd_pcm_hw_params_alloca(&hwparams); |
| 437 | if((err = snd_pcm_hw_params_any(pcm, hwparams)) < 0) |
| 438 | fatal(0, "error from snd_pcm_hw_params_any: %d", err); |
| 439 | if((err = snd_pcm_hw_params_set_access(pcm, hwparams, |
| 440 | SND_PCM_ACCESS_RW_INTERLEAVED)) < 0) |
| 441 | fatal(0, "error from snd_pcm_hw_params_set_access: %d", err); |
| 442 | if((err = snd_pcm_hw_params_set_format(pcm, hwparams, |
| 443 | sample_format)) < 0) |
| 444 | fatal(0, "error from snd_pcm_hw_params_set_format (%d): %d", |
| 445 | sample_format, err); |
| 446 | if((err = snd_pcm_hw_params_set_rate_near(pcm, hwparams, &rate, 0)) < 0) |
| 447 | fatal(0, "error from snd_pcm_hw_params_set_rate (%d): %d", |
| 448 | rate, err); |
| 449 | if((err = snd_pcm_hw_params_set_channels(pcm, hwparams, |
| 450 | channels)) < 0) |
| 451 | fatal(0, "error from snd_pcm_hw_params_set_channels (%d): %d", |
| 452 | channels, err); |
| 453 | if((err = snd_pcm_hw_params_set_buffer_size_near(pcm, hwparams, |
| 454 | &pcm_bufsize)) < 0) |
| 455 | fatal(0, "error from snd_pcm_hw_params_set_buffer_size (%d): %d", |
| 456 | MAXSAMPLES * samplesize * 3, err); |
| 457 | if((err = snd_pcm_hw_params(pcm, hwparams)) < 0) |
| 458 | fatal(0, "error calling snd_pcm_hw_params: %d", err); |
| 459 | /* Set up 'software' parameters */ |
| 460 | snd_pcm_sw_params_alloca(&swparams); |
| 461 | if((err = snd_pcm_sw_params_current(pcm, swparams)) < 0) |
| 462 | fatal(0, "error calling snd_pcm_sw_params_current: %d", err); |
| 463 | if((err = snd_pcm_sw_params_set_avail_min(pcm, swparams, avail_min)) < 0) |
| 464 | fatal(0, "error calling snd_pcm_sw_params_set_avail_min %d: %d", |
| 465 | avail_min, err); |
| 466 | if((err = snd_pcm_sw_params(pcm, swparams)) < 0) |
| 467 | fatal(0, "error calling snd_pcm_sw_params: %d", err); |
| 468 | |
| 469 | /* Ready to go */ |
| 470 | |
| 471 | time(&logged); |
| 472 | pthread_mutex_lock(&lock); |
| 473 | for(;;) { |
| 474 | /* Wait for the buffer to fill up a bit */ |
| 475 | logged = now; |
| 476 | info("%lu samples in buffer (%lus)", nsamples, |
| 477 | nsamples / (44100 * 2)); |
| 478 | info("Buffering..."); |
| 479 | while(nsamples < readahead) |
| 480 | pthread_cond_wait(&cond, &lock); |
| 481 | if(!prepared) { |
| 482 | if((err = snd_pcm_prepare(pcm))) |
| 483 | fatal(0, "error calling snd_pcm_prepare: %d", err); |
| 484 | prepared = 1; |
| 485 | } |
| 486 | assert(sequence != -1); |
| 487 | /* Start at the first available packet */ |
| 488 | next_timestamp = packets[sequence]->timestamp; |
| 489 | active = 1; |
| 490 | infilling = 0; |
| 491 | escape = 0; |
| 492 | logged = now; |
| 493 | info("%lu samples in buffer (%lus)", nsamples, |
| 494 | nsamples / (44100 * 2)); |
| 495 | info("Playing..."); |
| 496 | /* Wait until the buffer empties out */ |
| 497 | while(nsamples >= minbuffer && !escape) { |
| 498 | time(&now); |
| 499 | if(now > logged + 10) { |
| 500 | logged = now; |
| 501 | info("%lu samples in buffer (%lus)", nsamples, |
| 502 | nsamples / (44100 * 2)); |
| 503 | } |
| 504 | if(packets |
| 505 | && ge(next_timestamp, packets->timestamp + packets->nsamples)) { |
| 506 | info("dropping buffered past packet %"PRIx32" < %"PRIx32, |
| 507 | packets->timestamp, next_timestamp); |
| 508 | drop_first_packet(); |
| 509 | continue; |
| 510 | } |
| 511 | /* Wait for ALSA to ask us for more data */ |
| 512 | pthread_mutex_unlock(&lock); |
| 513 | write(2, ".", 1); /* TODO remove me sometime */ |
| 514 | switch(err = snd_pcm_wait(pcm, -1)) { |
| 515 | case 0: |
| 516 | info("snd_pcm_wait timed out"); |
| 517 | break; |
| 518 | case 1: |
| 519 | break; |
| 520 | default: |
| 521 | fatal(0, "snd_pcm_wait returned %d", err); |
| 522 | } |
| 523 | pthread_mutex_lock(&lock); |
| 524 | /* ALSA is ready for more data */ |
| 525 | packet_start = packets->timestamp; |
| 526 | packet_end = packets->timestamp + packets->nsamples; |
| 527 | if(ge(next_timestamp, packet_start) |
| 528 | && lt(next_timestamp, packet_end)) { |
| 529 | /* The target timestamp is somewhere in this packet */ |
| 530 | const uint32_t offset = next_timestamp - packets->timestamp; |
| 531 | const uint32_t samples_available = (packets->timestamp + packets->nsamples) - next_timestamp; |
| 532 | const size_t frames_available = samples_available / 2; |
| 533 | |
| 534 | frames_written = snd_pcm_writei(pcm, |
| 535 | packets->samples_raw + offset, |
| 536 | frames_available); |
| 537 | if(frames_written < 0) { |
| 538 | switch(frames_written) { |
| 539 | case -EAGAIN: |
| 540 | info("snd_pcm_wait() returned but we got -EAGAIN!"); |
| 541 | break; |
| 542 | case -EPIPE: |
| 543 | error(0, "error calling snd_pcm_writei: %ld", |
| 544 | (long)frames_written); |
| 545 | escape = 1; |
| 546 | break; |
| 547 | default: |
| 548 | fatal(0, "error calling snd_pcm_writei: %ld", |
| 549 | (long)frames_written); |
| 550 | } |
| 551 | } else { |
| 552 | samples_written = frames_written * 2; |
| 553 | next_timestamp += samples_written; |
| 554 | if(ge(next_timestamp, packet_end)) |
| 555 | drop_first_packet(); |
| 556 | infilling = 0; |
| 557 | } |
| 558 | } else { |
| 559 | /* We don't have anything to play! We'd better play some 0s. */ |
| 560 | static const uint16_t zeros[INFILL_SAMPLES]; |
| 561 | size_t samples_available = INFILL_SAMPLES, frames_available; |
| 562 | |
| 563 | /* If the maximum infill would take us past the start of the next |
| 564 | * packet then we truncate the infill to the right amount. */ |
| 565 | if(lt(packets->timestamp, |
| 566 | next_timestamp + samples_available)) |
| 567 | samples_available = packets->timestamp - next_timestamp; |
| 568 | if((int)samples_available < 0) { |
| 569 | info("packets->timestamp: %"PRIx32" next_timestamp: %"PRIx32" next+max: %"PRIx32" available: %"PRIx32, |
| 570 | packets->timestamp, next_timestamp, |
| 571 | next_timestamp + INFILL_SAMPLES, samples_available); |
| 572 | } |
| 573 | frames_available = samples_available / 2; |
| 574 | if(!infilling) { |
| 575 | info("Infilling %d samples, next=%"PRIx32" packet=[%"PRIx32",%"PRIx32"]", |
| 576 | samples_available, next_timestamp, |
| 577 | packets->timestamp, packets->timestamp + packets->nsamples); |
| 578 | //infilling++; |
| 579 | } |
| 580 | frames_written = snd_pcm_writei(pcm, |
| 581 | zeros, |
| 582 | frames_available); |
| 583 | if(frames_written < 0) { |
| 584 | switch(frames_written) { |
| 585 | case -EAGAIN: |
| 586 | info("snd_pcm_wait() returned but we got -EAGAIN!"); |
| 587 | break; |
| 588 | case -EPIPE: |
| 589 | error(0, "error calling snd_pcm_writei: %ld", |
| 590 | (long)frames_written); |
| 591 | escape = 1; |
| 592 | break; |
| 593 | default: |
| 594 | fatal(0, "error calling snd_pcm_writei: %ld", |
| 595 | (long)frames_written); |
| 596 | } |
| 597 | } else { |
| 598 | samples_written = frames_written * 2; |
| 599 | next_timestamp += samples_written; |
| 600 | } |
| 601 | } |
| 602 | } |
| 603 | active = 0; |
| 604 | /* We stop playing for a bit until the buffer re-fills */ |
| 605 | pthread_mutex_unlock(&lock); |
| 606 | if((err = snd_pcm_nonblock(pcm, 0))) |
| 607 | fatal(0, "error calling snd_pcm_nonblock: %d", err); |
| 608 | if(escape) { |
| 609 | if((err = snd_pcm_drop(pcm))) |
| 610 | fatal(0, "error calling snd_pcm_drop: %d", err); |
| 611 | escape = 0; |
| 612 | } else |
| 613 | if((err = snd_pcm_drain(pcm))) |
| 614 | fatal(0, "error calling snd_pcm_drain: %d", err); |
| 615 | if((err = snd_pcm_nonblock(pcm, 1))) |
| 616 | fatal(0, "error calling snd_pcm_nonblock: %d", err); |
| 617 | prepared = 0; |
| 618 | pthread_mutex_lock(&lock); |
| 619 | } |
| 620 | |
| 621 | } |
| 622 | #elif HAVE_COREAUDIO_AUDIOHARDWARE_H |
| 623 | { |
| 624 | OSStatus status; |
| 625 | UInt32 propertySize; |
| 626 | AudioDeviceID adid; |
| 627 | AudioStreamBasicDescription asbd; |
| 628 | |
| 629 | /* If this looks suspiciously like libao's macosx driver there's an |
| 630 | * excellent reason for that... */ |
| 631 | |
| 632 | /* TODO report errors as strings not numbers */ |
| 633 | propertySize = sizeof adid; |
| 634 | status = AudioHardwareGetProperty(kAudioHardwarePropertyDefaultOutputDevice, |
| 635 | &propertySize, &adid); |
| 636 | if(status) |
| 637 | fatal(0, "AudioHardwareGetProperty: %d", (int)status); |
| 638 | if(adid == kAudioDeviceUnknown) |
| 639 | fatal(0, "no output device"); |
| 640 | propertySize = sizeof asbd; |
| 641 | status = AudioDeviceGetProperty(adid, 0, false, |
| 642 | kAudioDevicePropertyStreamFormat, |
| 643 | &propertySize, &asbd); |
| 644 | if(status) |
| 645 | fatal(0, "AudioHardwareGetProperty: %d", (int)status); |
| 646 | D(("mSampleRate %f", asbd.mSampleRate)); |
| 647 | D(("mFormatID %08lx", asbd.mFormatID)); |
| 648 | D(("mFormatFlags %08lx", asbd.mFormatFlags)); |
| 649 | D(("mBytesPerPacket %08lx", asbd.mBytesPerPacket)); |
| 650 | D(("mFramesPerPacket %08lx", asbd.mFramesPerPacket)); |
| 651 | D(("mBytesPerFrame %08lx", asbd.mBytesPerFrame)); |
| 652 | D(("mChannelsPerFrame %08lx", asbd.mChannelsPerFrame)); |
| 653 | D(("mBitsPerChannel %08lx", asbd.mBitsPerChannel)); |
| 654 | D(("mReserved %08lx", asbd.mReserved)); |
| 655 | if(asbd.mFormatID != kAudioFormatLinearPCM) |
| 656 | fatal(0, "audio device does not support kAudioFormatLinearPCM"); |
| 657 | status = AudioDeviceAddIOProc(adid, adioproc, 0); |
| 658 | if(status) |
| 659 | fatal(0, "AudioDeviceAddIOProc: %d", (int)status); |
| 660 | pthread_mutex_lock(&lock); |
| 661 | for(;;) { |
| 662 | /* Wait for the buffer to fill up a bit */ |
| 663 | info("Buffering..."); |
| 664 | while(nsamples < readahead) |
| 665 | pthread_cond_wait(&cond, &lock); |
| 666 | /* Start playing now */ |
| 667 | info("Playing..."); |
| 668 | next_timestamp = packets[sequence]->timestamp; |
| 669 | active = 1; |
| 670 | status = AudioDeviceStart(adid, adioproc); |
| 671 | if(status) |
| 672 | fatal(0, "AudioDeviceStart: %d", (int)status); |
| 673 | /* Wait until the buffer empties out */ |
| 674 | while(nsamples >= minbuffer) |
| 675 | pthread_cond_wait(&cond, &lock); |
| 676 | /* Stop playing for a bit until the buffer re-fills */ |
| 677 | status = AudioDeviceStop(adid, adioproc); |
| 678 | if(status) |
| 679 | fatal(0, "AudioDeviceStop: %d", (int)status); |
| 680 | active = 0; |
| 681 | /* Go back round */ |
| 682 | } |
| 683 | } |
| 684 | #else |
| 685 | # error No known audio API |
| 686 | #endif |
| 687 | } |
| 688 | |
| 689 | /* display usage message and terminate */ |
| 690 | static void help(void) { |
| 691 | xprintf("Usage:\n" |
| 692 | " disorder-playrtp [OPTIONS] ADDRESS [PORT]\n" |
| 693 | "Options:\n" |
| 694 | " --device, -D DEVICE Output device\n" |
| 695 | " --min, -m FRAMES Buffer low water mark\n" |
| 696 | " --buffer, -b FRAMES Buffer high water mark\n" |
| 697 | " --max, -x FRAMES Buffer maximum size\n" |
| 698 | " --help, -h Display usage message\n" |
| 699 | " --version, -V Display version number\n" |
| 700 | ); |
| 701 | xfclose(stdout); |
| 702 | exit(0); |
| 703 | } |
| 704 | |
| 705 | /* display version number and terminate */ |
| 706 | static void version(void) { |
| 707 | xprintf("disorder-playrtp version %s\n", disorder_version_string); |
| 708 | xfclose(stdout); |
| 709 | exit(0); |
| 710 | } |
| 711 | |
| 712 | int main(int argc, char **argv) { |
| 713 | int n; |
| 714 | struct addrinfo *res; |
| 715 | struct stringlist sl; |
| 716 | char *sockname; |
| 717 | |
| 718 | static const struct addrinfo prefs = { |
| 719 | AI_PASSIVE, |
| 720 | PF_INET, |
| 721 | SOCK_DGRAM, |
| 722 | IPPROTO_UDP, |
| 723 | 0, |
| 724 | 0, |
| 725 | 0, |
| 726 | 0 |
| 727 | }; |
| 728 | |
| 729 | mem_init(); |
| 730 | if(!setlocale(LC_CTYPE, "")) fatal(errno, "error calling setlocale"); |
| 731 | while((n = getopt_long(argc, argv, "hVdD:m:b:x:L:", options, 0)) >= 0) { |
| 732 | switch(n) { |
| 733 | case 'h': help(); |
| 734 | case 'V': version(); |
| 735 | case 'd': debugging = 1; break; |
| 736 | case 'D': device = optarg; break; |
| 737 | case 'm': minbuffer = 2 * atol(optarg); break; |
| 738 | case 'b': readahead = 2 * atol(optarg); break; |
| 739 | case 'x': maxbuffer = 2 * atol(optarg); break; |
| 740 | case 'L': logfp = fopen(optarg, "w"); break; |
| 741 | default: fatal(0, "invalid option"); |
| 742 | } |
| 743 | } |
| 744 | if(!maxbuffer) |
| 745 | maxbuffer = 4 * readahead; |
| 746 | argc -= optind; |
| 747 | argv += optind; |
| 748 | if(argc < 1 || argc > 2) |
| 749 | fatal(0, "usage: disorder-playrtp [OPTIONS] ADDRESS [PORT]"); |
| 750 | sl.n = argc; |
| 751 | sl.s = argv; |
| 752 | /* Listen for inbound audio data */ |
| 753 | if(!(res = get_address(&sl, &prefs, &sockname))) |
| 754 | exit(1); |
| 755 | if((rtpfd = socket(res->ai_family, |
| 756 | res->ai_socktype, |
| 757 | res->ai_protocol)) < 0) |
| 758 | fatal(errno, "error creating socket"); |
| 759 | if(bind(rtpfd, res->ai_addr, res->ai_addrlen) < 0) |
| 760 | fatal(errno, "error binding socket to %s", sockname); |
| 761 | play_rtp(); |
| 762 | return 0; |
| 763 | } |
| 764 | |
| 765 | /* |
| 766 | Local Variables: |
| 767 | c-basic-offset:2 |
| 768 | comment-column:40 |
| 769 | fill-column:79 |
| 770 | indent-tabs-mode:nil |
| 771 | End: |
| 772 | */ |