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e83d0967 RK |
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 | */ | |
28bacdc0 RK |
20 | /** @file clients/playrtp.c |
21 | * @brief RTP player | |
22 | * | |
b0fdc63d | 23 | * This player supports Linux (<a href="http://www.alsa-project.org/">ALSA</a>) |
24 | * and Apple Mac (<a | |
25 | * href="http://developer.apple.com/audio/coreaudio.html">Core Audio</a>) | |
26 | * systems. There is no support for Microsoft Windows yet, and that will in | |
27 | * fact probably an entirely separate program. | |
28 | * | |
189e9830 RK |
29 | * The program runs (at least) three threads. listen_thread() is responsible |
30 | * for reading RTP packets off the wire and adding them to the linked list @ref | |
31 | * received_packets, assuming they are basically sound. queue_thread() takes | |
32 | * packets off this linked list and adds them to @ref packets (an operation | |
33 | * which might be much slower due to contention for @ref lock). | |
b0fdc63d | 34 | * |
35 | * The main thread is responsible for actually playing audio. In ALSA this | |
36 | * means it waits until ALSA says it's ready for more audio which it then | |
37 | * plays. | |
38 | * | |
8e3fe3d8 | 39 | * In Core Audio the main thread is only responsible for starting and stopping |
b0fdc63d | 40 | * play: the system does the actual playback in its own private thread, and |
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. | |
189e9830 RK |
46 | * |
47 | * Assumptions: | |
48 | * - it is safe to read uint32_t values without a lock protecting them | |
28bacdc0 | 49 | */ |
e83d0967 RK |
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> | |
0b75463f | 62 | #include <locale.h> |
2c7c9eae | 63 | #include <sys/uio.h> |
28bacdc0 | 64 | #include <string.h> |
8e3fe3d8 | 65 | #include <assert.h> |
e83d0967 RK |
66 | |
67 | #include "log.h" | |
68 | #include "mem.h" | |
69 | #include "configuration.h" | |
70 | #include "addr.h" | |
71 | #include "syscalls.h" | |
72 | #include "rtp.h" | |
0b75463f | 73 | #include "defs.h" |
28bacdc0 RK |
74 | #include "vector.h" |
75 | #include "heap.h" | |
189e9830 | 76 | #include "timeval.h" |
8e3fe3d8 | 77 | #include "playrtp.h" |
e83d0967 RK |
78 | |
79 | #if HAVE_COREAUDIO_AUDIOHARDWARE_H | |
80 | # include <CoreAudio/AudioHardware.h> | |
81 | #endif | |
0b75463f | 82 | #if API_ALSA |
83 | #include <alsa/asoundlib.h> | |
84 | #endif | |
e83d0967 | 85 | |
1153fd23 | 86 | #define readahead linux_headers_are_borked |
87 | ||
0b75463f | 88 | /** @brief RTP socket */ |
e83d0967 RK |
89 | static int rtpfd; |
90 | ||
345ebe66 RK |
91 | /** @brief Log output */ |
92 | static FILE *logfp; | |
93 | ||
0b75463f | 94 | /** @brief Output device */ |
8e3fe3d8 | 95 | const char *device; |
0b75463f | 96 | |
9086a105 | 97 | /** @brief Minimum low watermark |
0b75463f | 98 | * |
99 | * We'll stop playing if there's only this many samples in the buffer. */ | |
1153fd23 | 100 | static unsigned minbuffer = 2 * 44100 / 10; /* 0.2 seconds */ |
0b75463f | 101 | |
9086a105 | 102 | /** @brief Buffer high watermark |
1153fd23 | 103 | * |
104 | * We'll only start playing when this many samples are available. */ | |
8d0c14d7 | 105 | static unsigned readahead = 2 * 2 * 44100; |
0b75463f | 106 | |
9086a105 RK |
107 | /** @brief Maximum buffer size |
108 | * | |
109 | * We'll stop reading from the network if we have this many samples. */ | |
110 | static unsigned maxbuffer; | |
111 | ||
189e9830 RK |
112 | /** @brief Received packets |
113 | * Protected by @ref receive_lock | |
114 | * | |
115 | * Received packets are added to this list, and queue_thread() picks them off | |
116 | * it and adds them to @ref packets. Whenever a packet is added to it, @ref | |
117 | * receive_cond is signalled. | |
118 | */ | |
8e3fe3d8 | 119 | struct packet *received_packets; |
189e9830 RK |
120 | |
121 | /** @brief Tail of @ref received_packets | |
122 | * Protected by @ref receive_lock | |
123 | */ | |
8e3fe3d8 | 124 | struct packet **received_tail = &received_packets; |
189e9830 RK |
125 | |
126 | /** @brief Lock protecting @ref received_packets | |
127 | * | |
128 | * Only listen_thread() and queue_thread() ever hold this lock. It is vital | |
129 | * that queue_thread() not hold it any longer than it strictly has to. */ | |
8e3fe3d8 | 130 | pthread_mutex_t receive_lock = PTHREAD_MUTEX_INITIALIZER; |
189e9830 RK |
131 | |
132 | /** @brief Condition variable signalled when @ref received_packets is updated | |
133 | * | |
134 | * Used by listen_thread() to notify queue_thread() that it has added another | |
135 | * packet to @ref received_packets. */ | |
8e3fe3d8 | 136 | pthread_cond_t receive_cond = PTHREAD_COND_INITIALIZER; |
189e9830 RK |
137 | |
138 | /** @brief Length of @ref received_packets */ | |
8e3fe3d8 | 139 | uint32_t nreceived; |
28bacdc0 RK |
140 | |
141 | /** @brief Binary heap of received packets */ | |
8e3fe3d8 | 142 | struct pheap packets; |
28bacdc0 | 143 | |
189e9830 RK |
144 | /** @brief Total number of samples available |
145 | * | |
146 | * We make this volatile because we inspect it without a protecting lock, | |
147 | * so the usual pthread_* guarantees aren't available. | |
148 | */ | |
8e3fe3d8 | 149 | volatile uint32_t nsamples; |
0b75463f | 150 | |
151 | /** @brief Timestamp of next packet to play. | |
152 | * | |
153 | * This is set to the timestamp of the last packet, plus the number of | |
09ee2f0d | 154 | * samples it contained. Only valid if @ref active is nonzero. |
0b75463f | 155 | */ |
8e3fe3d8 | 156 | uint32_t next_timestamp; |
e83d0967 | 157 | |
09ee2f0d | 158 | /** @brief True if actively playing |
159 | * | |
160 | * This is true when playing and false when just buffering. */ | |
8e3fe3d8 | 161 | int active; |
09ee2f0d | 162 | |
189e9830 | 163 | /** @brief Lock protecting @ref packets */ |
8e3fe3d8 | 164 | pthread_mutex_t lock = PTHREAD_MUTEX_INITIALIZER; |
189e9830 RK |
165 | |
166 | /** @brief Condition variable signalled whenever @ref packets is changed */ | |
8e3fe3d8 | 167 | pthread_cond_t cond = PTHREAD_COND_INITIALIZER; |
2c7c9eae | 168 | |
8e3fe3d8 | 169 | HEAP_DEFINE(pheap, struct packet *, lt_packet); |
e83d0967 RK |
170 | |
171 | static const struct option options[] = { | |
172 | { "help", no_argument, 0, 'h' }, | |
173 | { "version", no_argument, 0, 'V' }, | |
174 | { "debug", no_argument, 0, 'd' }, | |
0b75463f | 175 | { "device", required_argument, 0, 'D' }, |
1153fd23 | 176 | { "min", required_argument, 0, 'm' }, |
9086a105 | 177 | { "max", required_argument, 0, 'x' }, |
1153fd23 | 178 | { "buffer", required_argument, 0, 'b' }, |
1f10f780 | 179 | { "rcvbuf", required_argument, 0, 'R' }, |
23205f9c | 180 | { "multicast", required_argument, 0, 'M' }, |
e83d0967 RK |
181 | { 0, 0, 0, 0 } |
182 | }; | |
183 | ||
28bacdc0 RK |
184 | /** @brief Drop the first packet |
185 | * | |
186 | * Assumes that @ref lock is held. | |
187 | */ | |
188 | static void drop_first_packet(void) { | |
189 | if(pheap_count(&packets)) { | |
190 | struct packet *const p = pheap_remove(&packets); | |
191 | nsamples -= p->nsamples; | |
192 | free_packet(p); | |
2c7c9eae | 193 | pthread_cond_broadcast(&cond); |
2c7c9eae | 194 | } |
9086a105 RK |
195 | } |
196 | ||
189e9830 RK |
197 | /** @brief Background thread adding packets to heap |
198 | * | |
199 | * This just transfers packets from @ref received_packets to @ref packets. It | |
200 | * is important that it holds @ref receive_lock for as little time as possible, | |
201 | * in order to minimize the interval between calls to read() in | |
202 | * listen_thread(). | |
203 | */ | |
204 | static void *queue_thread(void attribute((unused)) *arg) { | |
205 | struct packet *p; | |
206 | ||
207 | for(;;) { | |
208 | /* Get the next packet */ | |
209 | pthread_mutex_lock(&receive_lock); | |
210 | while(!received_packets) | |
211 | pthread_cond_wait(&receive_cond, &receive_lock); | |
212 | p = received_packets; | |
213 | received_packets = p->next; | |
214 | if(!received_packets) | |
215 | received_tail = &received_packets; | |
216 | --nreceived; | |
217 | pthread_mutex_unlock(&receive_lock); | |
218 | /* Add it to the heap */ | |
219 | pthread_mutex_lock(&lock); | |
220 | pheap_insert(&packets, p); | |
221 | nsamples += p->nsamples; | |
222 | pthread_cond_broadcast(&cond); | |
223 | pthread_mutex_unlock(&lock); | |
224 | } | |
225 | } | |
226 | ||
09ee2f0d | 227 | /** @brief Background thread collecting samples |
0b75463f | 228 | * |
229 | * This function collects samples, perhaps converts them to the target format, | |
b0fdc63d | 230 | * and adds them to the packet list. |
231 | * | |
232 | * It is crucial that the gap between successive calls to read() is as small as | |
233 | * possible: otherwise packets will be dropped. | |
234 | * | |
235 | * We use a binary heap to ensure that the unavoidable effort is at worst | |
236 | * logarithmic in the total number of packets - in fact if packets are mostly | |
237 | * received in order then we will largely do constant work per packet since the | |
238 | * newest packet will always be last. | |
239 | * | |
240 | * Of more concern is that we must acquire the lock on the heap to add a packet | |
241 | * to it. If this proves a problem in practice then the answer would be | |
242 | * (probably doubly) linked list with new packets added the end and a second | |
243 | * thread which reads packets off the list and adds them to the heap. | |
244 | * | |
245 | * We keep memory allocation (mostly) very fast by keeping pre-allocated | |
246 | * packets around; see @ref new_packet(). | |
247 | */ | |
0b75463f | 248 | static void *listen_thread(void attribute((unused)) *arg) { |
2c7c9eae | 249 | struct packet *p = 0; |
0b75463f | 250 | int n; |
2c7c9eae RK |
251 | struct rtp_header header; |
252 | uint16_t seq; | |
253 | uint32_t timestamp; | |
254 | struct iovec iov[2]; | |
e83d0967 RK |
255 | |
256 | for(;;) { | |
189e9830 | 257 | if(!p) |
2c7c9eae | 258 | p = new_packet(); |
2c7c9eae RK |
259 | iov[0].iov_base = &header; |
260 | iov[0].iov_len = sizeof header; | |
261 | iov[1].iov_base = p->samples_raw; | |
b64efe7e | 262 | iov[1].iov_len = sizeof p->samples_raw / sizeof *p->samples_raw; |
2c7c9eae | 263 | n = readv(rtpfd, iov, 2); |
e83d0967 RK |
264 | if(n < 0) { |
265 | switch(errno) { | |
266 | case EINTR: | |
267 | continue; | |
268 | default: | |
269 | fatal(errno, "error reading from socket"); | |
270 | } | |
271 | } | |
0b75463f | 272 | /* Ignore too-short packets */ |
345ebe66 RK |
273 | if((size_t)n <= sizeof (struct rtp_header)) { |
274 | info("ignored a short packet"); | |
0b75463f | 275 | continue; |
345ebe66 | 276 | } |
2c7c9eae RK |
277 | timestamp = htonl(header.timestamp); |
278 | seq = htons(header.seq); | |
09ee2f0d | 279 | /* Ignore packets in the past */ |
2c7c9eae | 280 | if(active && lt(timestamp, next_timestamp)) { |
c0e41690 | 281 | info("dropping old packet, timestamp=%"PRIx32" < %"PRIx32, |
2c7c9eae | 282 | timestamp, next_timestamp); |
09ee2f0d | 283 | continue; |
c0e41690 | 284 | } |
189e9830 | 285 | p->next = 0; |
58b5a68f | 286 | p->flags = 0; |
2c7c9eae | 287 | p->timestamp = timestamp; |
e83d0967 | 288 | /* Convert to target format */ |
58b5a68f RK |
289 | if(header.mpt & 0x80) |
290 | p->flags |= IDLE; | |
2c7c9eae | 291 | switch(header.mpt & 0x7F) { |
e83d0967 | 292 | case 10: |
2c7c9eae | 293 | p->nsamples = (n - sizeof header) / sizeof(uint16_t); |
e83d0967 RK |
294 | break; |
295 | /* TODO support other RFC3551 media types (when the speaker does) */ | |
296 | default: | |
0b75463f | 297 | fatal(0, "unsupported RTP payload type %d", |
2c7c9eae | 298 | header.mpt & 0x7F); |
e83d0967 | 299 | } |
345ebe66 RK |
300 | if(logfp) |
301 | fprintf(logfp, "sequence %u timestamp %"PRIx32" length %"PRIx32" end %"PRIx32"\n", | |
2c7c9eae | 302 | seq, timestamp, p->nsamples, timestamp + p->nsamples); |
0b75463f | 303 | /* Stop reading if we've reached the maximum. |
304 | * | |
305 | * This is rather unsatisfactory: it means that if packets get heavily | |
306 | * out of order then we guarantee dropouts. But for now... */ | |
345ebe66 | 307 | if(nsamples >= maxbuffer) { |
189e9830 | 308 | pthread_mutex_lock(&lock); |
345ebe66 RK |
309 | while(nsamples >= maxbuffer) |
310 | pthread_cond_wait(&cond, &lock); | |
189e9830 | 311 | pthread_mutex_unlock(&lock); |
345ebe66 | 312 | } |
189e9830 RK |
313 | /* Add the packet to the receive queue */ |
314 | pthread_mutex_lock(&receive_lock); | |
315 | *received_tail = p; | |
316 | received_tail = &p->next; | |
317 | ++nreceived; | |
318 | pthread_cond_signal(&receive_cond); | |
319 | pthread_mutex_unlock(&receive_lock); | |
58b5a68f RK |
320 | /* We'll need a new packet */ |
321 | p = 0; | |
e83d0967 RK |
322 | } |
323 | } | |
324 | ||
b0fdc63d | 325 | /** @brief Return true if @p p contains @p timestamp |
326 | * | |
327 | * Containment implies that a sample @p timestamp exists within the packet. | |
328 | */ | |
2c7c9eae RK |
329 | static inline int contains(const struct packet *p, uint32_t timestamp) { |
330 | const uint32_t packet_start = p->timestamp; | |
331 | const uint32_t packet_end = p->timestamp + p->nsamples; | |
332 | ||
333 | return (ge(timestamp, packet_start) | |
334 | && lt(timestamp, packet_end)); | |
335 | } | |
336 | ||
5626f6d2 RK |
337 | /** @brief Wait until the buffer is adequately full |
338 | * | |
339 | * Must be called with @ref lock held. | |
340 | */ | |
341 | static void fill_buffer(void) { | |
bfd27c14 RK |
342 | while(nsamples) |
343 | drop_first_packet(); | |
5626f6d2 RK |
344 | info("Buffering..."); |
345 | while(nsamples < readahead) | |
346 | pthread_cond_wait(&cond, &lock); | |
347 | next_timestamp = pheap_first(&packets)->timestamp; | |
348 | active = 1; | |
349 | } | |
350 | ||
351 | /** @brief Find next packet | |
352 | * @return Packet to play or NULL if none found | |
353 | * | |
354 | * The return packet is merely guaranteed not to be in the past: it might be | |
355 | * the first packet in the future rather than one that is actually suitable to | |
356 | * play. | |
357 | * | |
358 | * Must be called with @ref lock held. | |
359 | */ | |
360 | static struct packet *next_packet(void) { | |
361 | while(pheap_count(&packets)) { | |
362 | struct packet *const p = pheap_first(&packets); | |
363 | if(le(p->timestamp + p->nsamples, next_timestamp)) { | |
364 | /* This packet is in the past. Drop it and try another one. */ | |
365 | drop_first_packet(); | |
366 | } else | |
367 | /* This packet is NOT in the past. (It might be in the future | |
368 | * however.) */ | |
369 | return p; | |
370 | } | |
371 | return 0; | |
372 | } | |
373 | ||
e83d0967 | 374 | #if HAVE_COREAUDIO_AUDIOHARDWARE_H |
09ee2f0d | 375 | /** @brief Callback from Core Audio */ |
9086a105 RK |
376 | static OSStatus adioproc |
377 | (AudioDeviceID attribute((unused)) inDevice, | |
378 | const AudioTimeStamp attribute((unused)) *inNow, | |
379 | const AudioBufferList attribute((unused)) *inInputData, | |
380 | const AudioTimeStamp attribute((unused)) *inInputTime, | |
381 | AudioBufferList *outOutputData, | |
382 | const AudioTimeStamp attribute((unused)) *inOutputTime, | |
383 | void attribute((unused)) *inClientData) { | |
e83d0967 RK |
384 | UInt32 nbuffers = outOutputData->mNumberBuffers; |
385 | AudioBuffer *ab = outOutputData->mBuffers; | |
28bacdc0 | 386 | uint32_t samples_available; |
e83d0967 | 387 | |
0b75463f | 388 | pthread_mutex_lock(&lock); |
9086a105 RK |
389 | while(nbuffers > 0) { |
390 | float *samplesOut = ab->mData; | |
391 | size_t samplesOutLeft = ab->mDataByteSize / sizeof (float); | |
2c7c9eae | 392 | |
9086a105 | 393 | while(samplesOutLeft > 0) { |
5626f6d2 | 394 | const struct packet *p = next_packet(); |
28bacdc0 RK |
395 | if(p && contains(p, next_timestamp)) { |
396 | /* This packet is ready to play */ | |
397 | const uint32_t packet_end = p->timestamp + p->nsamples; | |
398 | const uint32_t offset = next_timestamp - p->timestamp; | |
b64efe7e | 399 | const uint16_t *ptr = (void *)(p->samples_raw + offset); |
28bacdc0 RK |
400 | |
401 | samples_available = packet_end - next_timestamp; | |
402 | if(samples_available > samplesOutLeft) | |
403 | samples_available = samplesOutLeft; | |
404 | next_timestamp += samples_available; | |
405 | samplesOutLeft -= samples_available; | |
406 | while(samples_available-- > 0) | |
407 | *samplesOut++ = (int16_t)ntohs(*ptr++) * (0.5 / 32767); | |
408 | /* We don't bother junking the packet - that'll be dealt with next time | |
409 | * round */ | |
410 | } else { | |
411 | /* No packet is ready to play (and there might be no packet at all) */ | |
412 | samples_available = p ? p->timestamp - next_timestamp | |
413 | : samplesOutLeft; | |
9086a105 RK |
414 | if(samples_available > samplesOutLeft) |
415 | samples_available = samplesOutLeft; | |
58b5a68f | 416 | //info("infill by %"PRIu32, samples_available); |
28bacdc0 | 417 | /* Conveniently the buffer is 0 to start with */ |
9086a105 RK |
418 | next_timestamp += samples_available; |
419 | samplesOut += samples_available; | |
420 | samplesOutLeft -= samples_available; | |
9086a105 | 421 | } |
e83d0967 | 422 | } |
9086a105 RK |
423 | ++ab; |
424 | --nbuffers; | |
e83d0967 RK |
425 | } |
426 | pthread_mutex_unlock(&lock); | |
427 | return 0; | |
428 | } | |
429 | #endif | |
430 | ||
b64efe7e | 431 | |
432 | #if API_ALSA | |
433 | /** @brief PCM handle */ | |
434 | static snd_pcm_t *pcm; | |
435 | ||
436 | /** @brief True when @ref pcm is up and running */ | |
437 | static int alsa_prepared = 1; | |
438 | ||
439 | /** @brief Initialize @ref pcm */ | |
440 | static void setup_alsa(void) { | |
441 | snd_pcm_hw_params_t *hwparams; | |
442 | snd_pcm_sw_params_t *swparams; | |
443 | /* Only support one format for now */ | |
444 | const int sample_format = SND_PCM_FORMAT_S16_BE; | |
445 | unsigned rate = 44100; | |
446 | const int channels = 2; | |
447 | const int samplesize = channels * sizeof(uint16_t); | |
448 | snd_pcm_uframes_t pcm_bufsize = MAXSAMPLES * samplesize * 3; | |
449 | /* If we can write more than this many samples we'll get a wakeup */ | |
450 | const int avail_min = 256; | |
451 | int err; | |
452 | ||
453 | /* Open ALSA */ | |
454 | if((err = snd_pcm_open(&pcm, | |
455 | device ? device : "default", | |
456 | SND_PCM_STREAM_PLAYBACK, | |
457 | SND_PCM_NONBLOCK))) | |
458 | fatal(0, "error from snd_pcm_open: %d", err); | |
459 | /* Set up 'hardware' parameters */ | |
460 | snd_pcm_hw_params_alloca(&hwparams); | |
461 | if((err = snd_pcm_hw_params_any(pcm, hwparams)) < 0) | |
462 | fatal(0, "error from snd_pcm_hw_params_any: %d", err); | |
463 | if((err = snd_pcm_hw_params_set_access(pcm, hwparams, | |
464 | SND_PCM_ACCESS_RW_INTERLEAVED)) < 0) | |
465 | fatal(0, "error from snd_pcm_hw_params_set_access: %d", err); | |
466 | if((err = snd_pcm_hw_params_set_format(pcm, hwparams, | |
467 | sample_format)) < 0) | |
468 | ||
469 | fatal(0, "error from snd_pcm_hw_params_set_format (%d): %d", | |
470 | sample_format, err); | |
471 | if((err = snd_pcm_hw_params_set_rate_near(pcm, hwparams, &rate, 0)) < 0) | |
472 | fatal(0, "error from snd_pcm_hw_params_set_rate (%d): %d", | |
473 | rate, err); | |
474 | if((err = snd_pcm_hw_params_set_channels(pcm, hwparams, | |
475 | channels)) < 0) | |
476 | fatal(0, "error from snd_pcm_hw_params_set_channels (%d): %d", | |
477 | channels, err); | |
478 | if((err = snd_pcm_hw_params_set_buffer_size_near(pcm, hwparams, | |
479 | &pcm_bufsize)) < 0) | |
480 | fatal(0, "error from snd_pcm_hw_params_set_buffer_size (%d): %d", | |
481 | MAXSAMPLES * samplesize * 3, err); | |
482 | if((err = snd_pcm_hw_params(pcm, hwparams)) < 0) | |
483 | fatal(0, "error calling snd_pcm_hw_params: %d", err); | |
484 | /* Set up 'software' parameters */ | |
485 | snd_pcm_sw_params_alloca(&swparams); | |
486 | if((err = snd_pcm_sw_params_current(pcm, swparams)) < 0) | |
487 | fatal(0, "error calling snd_pcm_sw_params_current: %d", err); | |
488 | if((err = snd_pcm_sw_params_set_avail_min(pcm, swparams, avail_min)) < 0) | |
489 | fatal(0, "error calling snd_pcm_sw_params_set_avail_min %d: %d", | |
490 | avail_min, err); | |
491 | if((err = snd_pcm_sw_params(pcm, swparams)) < 0) | |
492 | fatal(0, "error calling snd_pcm_sw_params: %d", err); | |
493 | } | |
494 | ||
495 | /** @brief Wait until ALSA wants some audio */ | |
496 | static void wait_alsa(void) { | |
497 | struct pollfd fds[64]; | |
498 | int nfds, err; | |
499 | unsigned short events; | |
500 | ||
501 | for(;;) { | |
502 | do { | |
503 | if((nfds = snd_pcm_poll_descriptors(pcm, | |
504 | fds, sizeof fds / sizeof *fds)) < 0) | |
505 | fatal(0, "error calling snd_pcm_poll_descriptors: %d", nfds); | |
506 | } while(poll(fds, nfds, -1) < 0 && errno == EINTR); | |
507 | if((err = snd_pcm_poll_descriptors_revents(pcm, fds, nfds, &events))) | |
508 | fatal(0, "error calling snd_pcm_poll_descriptors_revents: %d", err); | |
509 | if(events & POLLOUT) | |
510 | return; | |
511 | } | |
512 | } | |
513 | ||
b0fdc63d | 514 | /** @brief Play some sound via ALSA |
b64efe7e | 515 | * @param s Pointer to sample data |
516 | * @param n Number of samples | |
517 | * @return 0 on success, -1 on non-fatal error | |
518 | */ | |
519 | static int alsa_writei(const void *s, size_t n) { | |
520 | /* Do the write */ | |
521 | const snd_pcm_sframes_t frames_written = snd_pcm_writei(pcm, s, n / 2); | |
522 | if(frames_written < 0) { | |
523 | /* Something went wrong */ | |
524 | switch(frames_written) { | |
525 | case -EAGAIN: | |
526 | return 0; | |
527 | case -EPIPE: | |
528 | error(0, "error calling snd_pcm_writei: %ld", | |
529 | (long)frames_written); | |
530 | return -1; | |
531 | default: | |
532 | fatal(0, "error calling snd_pcm_writei: %ld", | |
533 | (long)frames_written); | |
534 | } | |
535 | } else { | |
536 | /* Success */ | |
537 | next_timestamp += frames_written * 2; | |
538 | return 0; | |
539 | } | |
540 | } | |
541 | ||
542 | /** @brief Play the relevant part of a packet | |
543 | * @param p Packet to play | |
544 | * @return 0 on success, -1 on non-fatal error | |
545 | */ | |
546 | static int alsa_play(const struct packet *p) { | |
b64efe7e | 547 | return alsa_writei(p->samples_raw + next_timestamp - p->timestamp, |
548 | (p->timestamp + p->nsamples) - next_timestamp); | |
549 | } | |
550 | ||
551 | /** @brief Play some silence | |
552 | * @param p Next packet or NULL | |
553 | * @return 0 on success, -1 on non-fatal error | |
554 | */ | |
555 | static int alsa_infill(const struct packet *p) { | |
556 | static const uint16_t zeros[INFILL_SAMPLES]; | |
557 | size_t samples_available = INFILL_SAMPLES; | |
558 | ||
559 | if(p && samples_available > p->timestamp - next_timestamp) | |
560 | samples_available = p->timestamp - next_timestamp; | |
b64efe7e | 561 | return alsa_writei(zeros, samples_available); |
562 | } | |
563 | ||
564 | /** @brief Reset ALSA state after we lost synchronization */ | |
565 | static void alsa_reset(int hard_reset) { | |
566 | int err; | |
567 | ||
568 | if((err = snd_pcm_nonblock(pcm, 0))) | |
569 | fatal(0, "error calling snd_pcm_nonblock: %d", err); | |
570 | if(hard_reset) { | |
571 | if((err = snd_pcm_drop(pcm))) | |
572 | fatal(0, "error calling snd_pcm_drop: %d", err); | |
573 | } else | |
574 | if((err = snd_pcm_drain(pcm))) | |
575 | fatal(0, "error calling snd_pcm_drain: %d", err); | |
576 | if((err = snd_pcm_nonblock(pcm, 1))) | |
577 | fatal(0, "error calling snd_pcm_nonblock: %d", err); | |
578 | alsa_prepared = 0; | |
579 | } | |
580 | #endif | |
581 | ||
09ee2f0d | 582 | /** @brief Play an RTP stream |
583 | * | |
584 | * This is the guts of the program. It is responsible for: | |
585 | * - starting the listening thread | |
586 | * - opening the audio device | |
587 | * - reading ahead to build up a buffer | |
588 | * - arranging for audio to be played | |
589 | * - detecting when the buffer has got too small and re-buffering | |
590 | */ | |
0b75463f | 591 | static void play_rtp(void) { |
592 | pthread_t ltid; | |
e83d0967 RK |
593 | |
594 | /* We receive and convert audio data in a background thread */ | |
0b75463f | 595 | pthread_create(<id, 0, listen_thread, 0); |
189e9830 RK |
596 | /* We have a second thread to add received packets to the queue */ |
597 | pthread_create(<id, 0, queue_thread, 0); | |
e83d0967 | 598 | #if API_ALSA |
0b75463f | 599 | { |
b64efe7e | 600 | struct packet *p; |
601 | int escape, err; | |
602 | ||
603 | /* Open the sound device */ | |
604 | setup_alsa(); | |
0b75463f | 605 | pthread_mutex_lock(&lock); |
606 | for(;;) { | |
607 | /* Wait for the buffer to fill up a bit */ | |
b64efe7e | 608 | fill_buffer(); |
609 | if(!alsa_prepared) { | |
0b75463f | 610 | if((err = snd_pcm_prepare(pcm))) |
611 | fatal(0, "error calling snd_pcm_prepare: %d", err); | |
b64efe7e | 612 | alsa_prepared = 1; |
0b75463f | 613 | } |
c0e41690 | 614 | escape = 0; |
ed13cbc8 | 615 | info("Playing..."); |
b64efe7e | 616 | /* Keep playing until the buffer empties out, or ALSA tells us to get |
617 | * lost */ | |
ca8d597c RK |
618 | while((nsamples >= minbuffer |
619 | || (nsamples > 0 | |
620 | && contains(pheap_first(&packets), next_timestamp))) | |
621 | && !escape) { | |
0b75463f | 622 | /* Wait for ALSA to ask us for more data */ |
623 | pthread_mutex_unlock(&lock); | |
b64efe7e | 624 | wait_alsa(); |
0b75463f | 625 | pthread_mutex_lock(&lock); |
b64efe7e | 626 | /* ALSA is ready for more data, find something to play */ |
627 | p = next_packet(); | |
628 | /* Play it or play some silence */ | |
629 | if(contains(p, next_timestamp)) | |
630 | escape = alsa_play(p); | |
631 | else | |
632 | escape = alsa_infill(p); | |
0b75463f | 633 | } |
09ee2f0d | 634 | active = 0; |
0b75463f | 635 | /* We stop playing for a bit until the buffer re-fills */ |
636 | pthread_mutex_unlock(&lock); | |
b64efe7e | 637 | alsa_reset(escape); |
0b75463f | 638 | pthread_mutex_lock(&lock); |
639 | } | |
640 | ||
641 | } | |
e83d0967 RK |
642 | #elif HAVE_COREAUDIO_AUDIOHARDWARE_H |
643 | { | |
644 | OSStatus status; | |
645 | UInt32 propertySize; | |
646 | AudioDeviceID adid; | |
647 | AudioStreamBasicDescription asbd; | |
648 | ||
649 | /* If this looks suspiciously like libao's macosx driver there's an | |
650 | * excellent reason for that... */ | |
651 | ||
652 | /* TODO report errors as strings not numbers */ | |
653 | propertySize = sizeof adid; | |
654 | status = AudioHardwareGetProperty(kAudioHardwarePropertyDefaultOutputDevice, | |
655 | &propertySize, &adid); | |
656 | if(status) | |
657 | fatal(0, "AudioHardwareGetProperty: %d", (int)status); | |
658 | if(adid == kAudioDeviceUnknown) | |
659 | fatal(0, "no output device"); | |
660 | propertySize = sizeof asbd; | |
661 | status = AudioDeviceGetProperty(adid, 0, false, | |
662 | kAudioDevicePropertyStreamFormat, | |
663 | &propertySize, &asbd); | |
664 | if(status) | |
665 | fatal(0, "AudioHardwareGetProperty: %d", (int)status); | |
666 | D(("mSampleRate %f", asbd.mSampleRate)); | |
9086a105 RK |
667 | D(("mFormatID %08lx", asbd.mFormatID)); |
668 | D(("mFormatFlags %08lx", asbd.mFormatFlags)); | |
669 | D(("mBytesPerPacket %08lx", asbd.mBytesPerPacket)); | |
670 | D(("mFramesPerPacket %08lx", asbd.mFramesPerPacket)); | |
671 | D(("mBytesPerFrame %08lx", asbd.mBytesPerFrame)); | |
672 | D(("mChannelsPerFrame %08lx", asbd.mChannelsPerFrame)); | |
673 | D(("mBitsPerChannel %08lx", asbd.mBitsPerChannel)); | |
674 | D(("mReserved %08lx", asbd.mReserved)); | |
e83d0967 RK |
675 | if(asbd.mFormatID != kAudioFormatLinearPCM) |
676 | fatal(0, "audio device does not support kAudioFormatLinearPCM"); | |
677 | status = AudioDeviceAddIOProc(adid, adioproc, 0); | |
678 | if(status) | |
679 | fatal(0, "AudioDeviceAddIOProc: %d", (int)status); | |
680 | pthread_mutex_lock(&lock); | |
681 | for(;;) { | |
682 | /* Wait for the buffer to fill up a bit */ | |
b64efe7e | 683 | fill_buffer(); |
e83d0967 | 684 | /* Start playing now */ |
8dcb5ff0 | 685 | info("Playing..."); |
28bacdc0 | 686 | next_timestamp = pheap_first(&packets)->timestamp; |
8dcb5ff0 | 687 | active = 1; |
e83d0967 RK |
688 | status = AudioDeviceStart(adid, adioproc); |
689 | if(status) | |
690 | fatal(0, "AudioDeviceStart: %d", (int)status); | |
691 | /* Wait until the buffer empties out */ | |
ca8d597c RK |
692 | while(nsamples >= minbuffer |
693 | || (nsamples > 0 | |
694 | && contains(pheap_first(&packets), next_timestamp))) | |
e83d0967 RK |
695 | pthread_cond_wait(&cond, &lock); |
696 | /* Stop playing for a bit until the buffer re-fills */ | |
697 | status = AudioDeviceStop(adid, adioproc); | |
698 | if(status) | |
699 | fatal(0, "AudioDeviceStop: %d", (int)status); | |
8dcb5ff0 | 700 | active = 0; |
e83d0967 RK |
701 | /* Go back round */ |
702 | } | |
703 | } | |
704 | #else | |
705 | # error No known audio API | |
706 | #endif | |
707 | } | |
708 | ||
709 | /* display usage message and terminate */ | |
710 | static void help(void) { | |
711 | xprintf("Usage:\n" | |
712 | " disorder-playrtp [OPTIONS] ADDRESS [PORT]\n" | |
713 | "Options:\n" | |
1153fd23 | 714 | " --device, -D DEVICE Output device\n" |
715 | " --min, -m FRAMES Buffer low water mark\n" | |
9086a105 RK |
716 | " --buffer, -b FRAMES Buffer high water mark\n" |
717 | " --max, -x FRAMES Buffer maximum size\n" | |
1f10f780 | 718 | " --rcvbuf, -R BYTES Socket receive buffer size\n" |
23205f9c | 719 | " --multicast, -M GROUP Join multicast group\n" |
9086a105 RK |
720 | " --help, -h Display usage message\n" |
721 | " --version, -V Display version number\n" | |
722 | ); | |
e83d0967 RK |
723 | xfclose(stdout); |
724 | exit(0); | |
725 | } | |
726 | ||
727 | /* display version number and terminate */ | |
728 | static void version(void) { | |
729 | xprintf("disorder-playrtp version %s\n", disorder_version_string); | |
730 | xfclose(stdout); | |
731 | exit(0); | |
732 | } | |
733 | ||
734 | int main(int argc, char **argv) { | |
735 | int n; | |
736 | struct addrinfo *res; | |
737 | struct stringlist sl; | |
0b75463f | 738 | char *sockname; |
1f10f780 RK |
739 | int rcvbuf, target_rcvbuf = 131072; |
740 | socklen_t len; | |
23205f9c RK |
741 | char *multicast_group = 0; |
742 | struct ip_mreq mreq; | |
743 | struct ipv6_mreq mreq6; | |
e83d0967 | 744 | |
0b75463f | 745 | static const struct addrinfo prefs = { |
e83d0967 RK |
746 | AI_PASSIVE, |
747 | PF_INET, | |
748 | SOCK_DGRAM, | |
749 | IPPROTO_UDP, | |
750 | 0, | |
751 | 0, | |
752 | 0, | |
753 | 0 | |
754 | }; | |
755 | ||
756 | mem_init(); | |
757 | if(!setlocale(LC_CTYPE, "")) fatal(errno, "error calling setlocale"); | |
23205f9c | 758 | while((n = getopt_long(argc, argv, "hVdD:m:b:x:L:R:M:", options, 0)) >= 0) { |
e83d0967 RK |
759 | switch(n) { |
760 | case 'h': help(); | |
761 | case 'V': version(); | |
762 | case 'd': debugging = 1; break; | |
0b75463f | 763 | case 'D': device = optarg; break; |
1153fd23 | 764 | case 'm': minbuffer = 2 * atol(optarg); break; |
765 | case 'b': readahead = 2 * atol(optarg); break; | |
9086a105 | 766 | case 'x': maxbuffer = 2 * atol(optarg); break; |
345ebe66 | 767 | case 'L': logfp = fopen(optarg, "w"); break; |
1f10f780 | 768 | case 'R': target_rcvbuf = atoi(optarg); break; |
23205f9c | 769 | case 'M': multicast_group = optarg; break; |
e83d0967 RK |
770 | default: fatal(0, "invalid option"); |
771 | } | |
772 | } | |
9086a105 RK |
773 | if(!maxbuffer) |
774 | maxbuffer = 4 * readahead; | |
e83d0967 RK |
775 | argc -= optind; |
776 | argv += optind; | |
777 | if(argc < 1 || argc > 2) | |
778 | fatal(0, "usage: disorder-playrtp [OPTIONS] ADDRESS [PORT]"); | |
779 | sl.n = argc; | |
780 | sl.s = argv; | |
781 | /* Listen for inbound audio data */ | |
0b75463f | 782 | if(!(res = get_address(&sl, &prefs, &sockname))) |
e83d0967 RK |
783 | exit(1); |
784 | if((rtpfd = socket(res->ai_family, | |
785 | res->ai_socktype, | |
786 | res->ai_protocol)) < 0) | |
787 | fatal(errno, "error creating socket"); | |
788 | if(bind(rtpfd, res->ai_addr, res->ai_addrlen) < 0) | |
789 | fatal(errno, "error binding socket to %s", sockname); | |
23205f9c RK |
790 | if(multicast_group) { |
791 | if((n = getaddrinfo(multicast_group, 0, &prefs, &res))) | |
792 | fatal(0, "getaddrinfo %s: %s", multicast_group, gai_strerror(n)); | |
793 | switch(res->ai_family) { | |
794 | case PF_INET: | |
795 | mreq.imr_multiaddr = ((struct sockaddr_in *)res->ai_addr)->sin_addr; | |
796 | mreq.imr_interface.s_addr = 0; /* use primary interface */ | |
797 | if(setsockopt(rtpfd, IPPROTO_IP, IP_ADD_MEMBERSHIP, | |
798 | &mreq, sizeof mreq) < 0) | |
799 | fatal(errno, "error calling setsockopt IP_ADD_MEMBERSHIP"); | |
800 | break; | |
801 | case PF_INET6: | |
802 | mreq6.ipv6mr_multiaddr = ((struct sockaddr_in6 *)res->ai_addr)->sin6_addr; | |
803 | memset(&mreq6.ipv6mr_interface, 0, sizeof mreq6.ipv6mr_interface); | |
804 | if(setsockopt(rtpfd, IPPROTO_IPV6, IPV6_JOIN_GROUP, | |
805 | &mreq6, sizeof mreq6) < 0) | |
806 | fatal(errno, "error calling setsockopt IPV6_JOIN_GROUP"); | |
807 | break; | |
808 | default: | |
809 | fatal(0, "unsupported address family %d", res->ai_family); | |
810 | } | |
811 | } | |
1f10f780 RK |
812 | len = sizeof rcvbuf; |
813 | if(getsockopt(rtpfd, SOL_SOCKET, SO_RCVBUF, &rcvbuf, &len) < 0) | |
814 | fatal(errno, "error calling getsockopt SO_RCVBUF"); | |
f0bae611 | 815 | if(target_rcvbuf > rcvbuf) { |
1f10f780 RK |
816 | if(setsockopt(rtpfd, SOL_SOCKET, SO_RCVBUF, |
817 | &target_rcvbuf, sizeof target_rcvbuf) < 0) | |
818 | error(errno, "error calling setsockopt SO_RCVBUF %d", | |
819 | target_rcvbuf); | |
820 | /* We try to carry on anyway */ | |
821 | else | |
822 | info("changed socket receive buffer from %d to %d", | |
823 | rcvbuf, target_rcvbuf); | |
824 | } else | |
825 | info("default socket receive buffer %d", rcvbuf); | |
826 | if(logfp) | |
827 | info("WARNING: -L option can impact performance"); | |
e83d0967 RK |
828 | play_rtp(); |
829 | return 0; | |
830 | } | |
831 | ||
832 | /* | |
833 | Local Variables: | |
834 | c-basic-offset:2 | |
835 | comment-column:40 | |
836 | fill-column:79 | |
837 | indent-tabs-mode:nil | |
838 | End: | |
839 | */ |