2 * This file is part of DisOrder.
3 * Copyright (C) 2007 Richard Kettlewell
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.
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.
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
20 /** @file clients/playrtp.c
23 * This player supports Linux (<a href="http://www.alsa-project.org/">ALSA</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.
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).
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
39 * InCore Audio the main thread is only responsible for starting and stopping
40 * play: the system does the actual playback in its own private thread, and
41 * calls adioproc() to fetch the audio data.
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.
48 * - it is safe to read uint32_t values without a lock protecting them
57 #include <sys/socket.h>
58 #include <sys/types.h>
59 #include <sys/socket.h>
68 #include "configuration.h"
77 #if HAVE_COREAUDIO_AUDIOHARDWARE_H
78 # include <CoreAudio/AudioHardware.h>
81 #include <alsa/asoundlib.h>
84 #define readahead linux_headers_are_borked
86 /** @brief RTP socket */
89 /** @brief Log output */
92 /** @brief Output device */
93 static const char *device;
95 /** @brief Maximum samples per packet we'll support
97 * NB that two channels = two samples in this program.
99 #define MAXSAMPLES 2048
101 /** @brief Minimum low watermark
103 * We'll stop playing if there's only this many samples in the buffer. */
104 static unsigned minbuffer = 2 * 44100 / 10; /* 0.2 seconds */
106 /** @brief Buffer high watermark
108 * We'll only start playing when this many samples are available. */
109 static unsigned readahead = 2 * 2 * 44100;
111 /** @brief Maximum buffer size
113 * We'll stop reading from the network if we have this many samples. */
114 static unsigned maxbuffer;
116 /** @brief Number of samples to infill by in one go
118 * This is an upper bound - in practice we expect the underlying audio API to
119 * only ask for a much smaller number of samples in any one go.
121 #define INFILL_SAMPLES (44100 * 2) /* 1s */
123 /** @brief Received packet
125 * Received packets are kept in a binary heap (see @ref pheap) ordered by
129 /** @brief Next packet in @ref next_free_packet or @ref received_packets */
132 /** @brief Number of samples in this packet */
135 /** @brief Timestamp from RTP packet
137 * NB that "timestamps" are really sample counters. Use lt() or lt_packet()
138 * to compare timestamps.
145 * - @ref IDLE - the idle bit was set in the RTP packet
148 /** @brief idle bit set in RTP packet*/
151 /** @brief Raw sample data
153 * Only the first @p nsamples samples are defined; the rest is uninitialized
156 uint16_t samples_raw[MAXSAMPLES];
159 /** @brief Return true iff \f$a < b\f$ in sequence-space arithmetic
161 * Specifically it returns true if \f$(a-b) mod 2^{32} < 2^{31}\f$.
163 * See also lt_packet().
165 static inline int lt(uint32_t a, uint32_t b) {
166 return (uint32_t)(a - b) & 0x80000000;
169 /** @brief Return true iff a >= b in sequence-space arithmetic */
170 static inline int ge(uint32_t a, uint32_t b) {
174 /** @brief Return true iff a > b in sequence-space arithmetic */
175 static inline int gt(uint32_t a, uint32_t b) {
179 /** @brief Return true iff a <= b in sequence-space arithmetic */
180 static inline int le(uint32_t a, uint32_t b) {
184 /** @brief Ordering for packets, used by @ref pheap */
185 static inline int lt_packet(const struct packet *a, const struct packet *b) {
186 return lt(a->timestamp, b->timestamp);
189 /** @brief Received packets
190 * Protected by @ref receive_lock
192 * Received packets are added to this list, and queue_thread() picks them off
193 * it and adds them to @ref packets. Whenever a packet is added to it, @ref
194 * receive_cond is signalled.
196 static struct packet *received_packets;
198 /** @brief Tail of @ref received_packets
199 * Protected by @ref receive_lock
201 static struct packet **received_tail = &received_packets;
203 /** @brief Lock protecting @ref received_packets
205 * Only listen_thread() and queue_thread() ever hold this lock. It is vital
206 * that queue_thread() not hold it any longer than it strictly has to. */
207 static pthread_mutex_t receive_lock = PTHREAD_MUTEX_INITIALIZER;
209 /** @brief Condition variable signalled when @ref received_packets is updated
211 * Used by listen_thread() to notify queue_thread() that it has added another
212 * packet to @ref received_packets. */
213 static pthread_cond_t receive_cond = PTHREAD_COND_INITIALIZER;
215 /** @brief Length of @ref received_packets */
216 static uint32_t nreceived;
219 * @brief Binary heap of packets ordered by timestamp */
220 HEAP_TYPE(pheap, struct packet *, lt_packet);
222 /** @brief Binary heap of received packets */
223 static struct pheap packets;
225 /** @brief Total number of samples available
227 * We make this volatile because we inspect it without a protecting lock,
228 * so the usual pthread_* guarantees aren't available.
230 static volatile uint32_t nsamples;
232 /** @brief Timestamp of next packet to play.
234 * This is set to the timestamp of the last packet, plus the number of
235 * samples it contained. Only valid if @ref active is nonzero.
237 static uint32_t next_timestamp;
239 /** @brief True if actively playing
241 * This is true when playing and false when just buffering. */
244 /** @brief Lock protecting @ref packets */
245 static pthread_mutex_t lock = PTHREAD_MUTEX_INITIALIZER;
247 /** @brief Condition variable signalled whenever @ref packets is changed */
248 static pthread_cond_t cond = PTHREAD_COND_INITIALIZER;
250 /** @brief Structure of free packet list */
253 union free_packet *next;
256 /** @brief Linked list of free packets
258 * This is a linked list of formerly used packets. For preference we re-use
259 * packets that have already been used rather than unused ones, to limit the
260 * size of the program's working set. If there are no free packets in the list
261 * we try @ref next_free_packet instead.
263 * Must hold @ref lock when accessing this.
265 static union free_packet *free_packets;
267 /** @brief Array of new free packets
269 * There are @ref count_free_packets ready to use at this address. If there
270 * are none left we allocate more memory.
272 * Must hold @ref lock when accessing this.
274 static union free_packet *next_free_packet;
276 /** @brief Count of new free packets at @ref next_free_packet
278 * Must hold @ref lock when accessing this.
280 static size_t count_free_packets;
282 /** @brief Lock protecting packet allocator */
283 static pthread_mutex_t mem_lock = PTHREAD_MUTEX_INITIALIZER;
285 static const struct option options[] = {
286 { "help", no_argument, 0, 'h' },
287 { "version", no_argument, 0, 'V' },
288 { "debug", no_argument, 0, 'd' },
289 { "device", required_argument, 0, 'D' },
290 { "min", required_argument, 0, 'm' },
291 { "max", required_argument, 0, 'x' },
292 { "buffer", required_argument, 0, 'b' },
293 { "rcvbuf", required_argument, 0, 'R' },
297 /** @brief Return a new packet */
298 static struct packet *new_packet(void) {
301 pthread_mutex_lock(&mem_lock);
303 p = &free_packets->p;
304 free_packets = free_packets->next;
306 if(!count_free_packets) {
307 next_free_packet = xcalloc(1024, sizeof (union free_packet));
308 count_free_packets = 1024;
310 p = &(next_free_packet++)->p;
311 --count_free_packets;
313 pthread_mutex_unlock(&mem_lock);
317 /** @brief Free a packet */
318 static void free_packet(struct packet *p) {
319 union free_packet *u = (union free_packet *)p;
320 pthread_mutex_lock(&mem_lock);
321 u->next = free_packets;
323 pthread_mutex_unlock(&mem_lock);
326 /** @brief Drop the first packet
328 * Assumes that @ref lock is held.
330 static void drop_first_packet(void) {
331 if(pheap_count(&packets)) {
332 struct packet *const p = pheap_remove(&packets);
333 nsamples -= p->nsamples;
335 pthread_cond_broadcast(&cond);
339 /** @brief Background thread adding packets to heap
341 * This just transfers packets from @ref received_packets to @ref packets. It
342 * is important that it holds @ref receive_lock for as little time as possible,
343 * in order to minimize the interval between calls to read() in
346 static void *queue_thread(void attribute((unused)) *arg) {
350 /* Get the next packet */
351 pthread_mutex_lock(&receive_lock);
352 while(!received_packets)
353 pthread_cond_wait(&receive_cond, &receive_lock);
354 p = received_packets;
355 received_packets = p->next;
356 if(!received_packets)
357 received_tail = &received_packets;
359 pthread_mutex_unlock(&receive_lock);
360 /* Add it to the heap */
361 pthread_mutex_lock(&lock);
362 pheap_insert(&packets, p);
363 nsamples += p->nsamples;
364 pthread_cond_broadcast(&cond);
365 pthread_mutex_unlock(&lock);
369 /** @brief Background thread collecting samples
371 * This function collects samples, perhaps converts them to the target format,
372 * and adds them to the packet list.
374 * It is crucial that the gap between successive calls to read() is as small as
375 * possible: otherwise packets will be dropped.
377 * We use a binary heap to ensure that the unavoidable effort is at worst
378 * logarithmic in the total number of packets - in fact if packets are mostly
379 * received in order then we will largely do constant work per packet since the
380 * newest packet will always be last.
382 * Of more concern is that we must acquire the lock on the heap to add a packet
383 * to it. If this proves a problem in practice then the answer would be
384 * (probably doubly) linked list with new packets added the end and a second
385 * thread which reads packets off the list and adds them to the heap.
387 * We keep memory allocation (mostly) very fast by keeping pre-allocated
388 * packets around; see @ref new_packet().
390 static void *listen_thread(void attribute((unused)) *arg) {
391 struct packet *p = 0;
393 struct rtp_header header;
401 iov[0].iov_base = &header;
402 iov[0].iov_len = sizeof header;
403 iov[1].iov_base = p->samples_raw;
404 iov[1].iov_len = sizeof p->samples_raw / sizeof *p->samples_raw;
405 n = readv(rtpfd, iov, 2);
411 fatal(errno, "error reading from socket");
414 /* Ignore too-short packets */
415 if((size_t)n <= sizeof (struct rtp_header)) {
416 info("ignored a short packet");
419 timestamp = htonl(header.timestamp);
420 seq = htons(header.seq);
421 /* Ignore packets in the past */
422 if(active && lt(timestamp, next_timestamp)) {
423 info("dropping old packet, timestamp=%"PRIx32" < %"PRIx32,
424 timestamp, next_timestamp);
429 p->timestamp = timestamp;
430 /* Convert to target format */
431 if(header.mpt & 0x80)
433 switch(header.mpt & 0x7F) {
435 p->nsamples = (n - sizeof header) / sizeof(uint16_t);
437 /* TODO support other RFC3551 media types (when the speaker does) */
439 fatal(0, "unsupported RTP payload type %d",
443 fprintf(logfp, "sequence %u timestamp %"PRIx32" length %"PRIx32" end %"PRIx32"\n",
444 seq, timestamp, p->nsamples, timestamp + p->nsamples);
445 /* Stop reading if we've reached the maximum.
447 * This is rather unsatisfactory: it means that if packets get heavily
448 * out of order then we guarantee dropouts. But for now... */
449 if(nsamples >= maxbuffer) {
450 pthread_mutex_lock(&lock);
451 while(nsamples >= maxbuffer)
452 pthread_cond_wait(&cond, &lock);
453 pthread_mutex_unlock(&lock);
455 /* Add the packet to the receive queue */
456 pthread_mutex_lock(&receive_lock);
458 received_tail = &p->next;
460 pthread_cond_signal(&receive_cond);
461 pthread_mutex_unlock(&receive_lock);
462 /* We'll need a new packet */
467 /** @brief Return true if @p p contains @p timestamp
469 * Containment implies that a sample @p timestamp exists within the packet.
471 static inline int contains(const struct packet *p, uint32_t timestamp) {
472 const uint32_t packet_start = p->timestamp;
473 const uint32_t packet_end = p->timestamp + p->nsamples;
475 return (ge(timestamp, packet_start)
476 && lt(timestamp, packet_end));
479 /** @brief Wait until the buffer is adequately full
481 * Must be called with @ref lock held.
483 static void fill_buffer(void) {
486 info("Buffering...");
487 while(nsamples < readahead)
488 pthread_cond_wait(&cond, &lock);
489 next_timestamp = pheap_first(&packets)->timestamp;
493 /** @brief Find next packet
494 * @return Packet to play or NULL if none found
496 * The return packet is merely guaranteed not to be in the past: it might be
497 * the first packet in the future rather than one that is actually suitable to
500 * Must be called with @ref lock held.
502 static struct packet *next_packet(void) {
503 while(pheap_count(&packets)) {
504 struct packet *const p = pheap_first(&packets);
505 if(le(p->timestamp + p->nsamples, next_timestamp)) {
506 /* This packet is in the past. Drop it and try another one. */
509 /* This packet is NOT in the past. (It might be in the future
516 #if HAVE_COREAUDIO_AUDIOHARDWARE_H
517 /** @brief Callback from Core Audio */
518 static OSStatus adioproc
519 (AudioDeviceID attribute((unused)) inDevice,
520 const AudioTimeStamp attribute((unused)) *inNow,
521 const AudioBufferList attribute((unused)) *inInputData,
522 const AudioTimeStamp attribute((unused)) *inInputTime,
523 AudioBufferList *outOutputData,
524 const AudioTimeStamp attribute((unused)) *inOutputTime,
525 void attribute((unused)) *inClientData) {
526 UInt32 nbuffers = outOutputData->mNumberBuffers;
527 AudioBuffer *ab = outOutputData->mBuffers;
528 uint32_t samples_available;
530 pthread_mutex_lock(&lock);
531 while(nbuffers > 0) {
532 float *samplesOut = ab->mData;
533 size_t samplesOutLeft = ab->mDataByteSize / sizeof (float);
535 while(samplesOutLeft > 0) {
536 const struct packet *p = next_packet();
537 if(p && contains(p, next_timestamp)) {
538 /* This packet is ready to play */
539 const uint32_t packet_end = p->timestamp + p->nsamples;
540 const uint32_t offset = next_timestamp - p->timestamp;
541 const uint16_t *ptr = (void *)(p->samples_raw + offset);
543 samples_available = packet_end - next_timestamp;
544 if(samples_available > samplesOutLeft)
545 samples_available = samplesOutLeft;
546 next_timestamp += samples_available;
547 samplesOutLeft -= samples_available;
548 while(samples_available-- > 0)
549 *samplesOut++ = (int16_t)ntohs(*ptr++) * (0.5 / 32767);
550 /* We don't bother junking the packet - that'll be dealt with next time
553 /* No packet is ready to play (and there might be no packet at all) */
554 samples_available = p ? p->timestamp - next_timestamp
556 if(samples_available > samplesOutLeft)
557 samples_available = samplesOutLeft;
558 //info("infill by %"PRIu32, samples_available);
559 /* Conveniently the buffer is 0 to start with */
560 next_timestamp += samples_available;
561 samplesOut += samples_available;
562 samplesOutLeft -= samples_available;
568 pthread_mutex_unlock(&lock);
575 /** @brief PCM handle */
576 static snd_pcm_t *pcm;
578 /** @brief True when @ref pcm is up and running */
579 static int alsa_prepared = 1;
581 /** @brief Initialize @ref pcm */
582 static void setup_alsa(void) {
583 snd_pcm_hw_params_t *hwparams;
584 snd_pcm_sw_params_t *swparams;
585 /* Only support one format for now */
586 const int sample_format = SND_PCM_FORMAT_S16_BE;
587 unsigned rate = 44100;
588 const int channels = 2;
589 const int samplesize = channels * sizeof(uint16_t);
590 snd_pcm_uframes_t pcm_bufsize = MAXSAMPLES * samplesize * 3;
591 /* If we can write more than this many samples we'll get a wakeup */
592 const int avail_min = 256;
596 if((err = snd_pcm_open(&pcm,
597 device ? device : "default",
598 SND_PCM_STREAM_PLAYBACK,
600 fatal(0, "error from snd_pcm_open: %d", err);
601 /* Set up 'hardware' parameters */
602 snd_pcm_hw_params_alloca(&hwparams);
603 if((err = snd_pcm_hw_params_any(pcm, hwparams)) < 0)
604 fatal(0, "error from snd_pcm_hw_params_any: %d", err);
605 if((err = snd_pcm_hw_params_set_access(pcm, hwparams,
606 SND_PCM_ACCESS_RW_INTERLEAVED)) < 0)
607 fatal(0, "error from snd_pcm_hw_params_set_access: %d", err);
608 if((err = snd_pcm_hw_params_set_format(pcm, hwparams,
611 fatal(0, "error from snd_pcm_hw_params_set_format (%d): %d",
613 if((err = snd_pcm_hw_params_set_rate_near(pcm, hwparams, &rate, 0)) < 0)
614 fatal(0, "error from snd_pcm_hw_params_set_rate (%d): %d",
616 if((err = snd_pcm_hw_params_set_channels(pcm, hwparams,
618 fatal(0, "error from snd_pcm_hw_params_set_channels (%d): %d",
620 if((err = snd_pcm_hw_params_set_buffer_size_near(pcm, hwparams,
622 fatal(0, "error from snd_pcm_hw_params_set_buffer_size (%d): %d",
623 MAXSAMPLES * samplesize * 3, err);
624 if((err = snd_pcm_hw_params(pcm, hwparams)) < 0)
625 fatal(0, "error calling snd_pcm_hw_params: %d", err);
626 /* Set up 'software' parameters */
627 snd_pcm_sw_params_alloca(&swparams);
628 if((err = snd_pcm_sw_params_current(pcm, swparams)) < 0)
629 fatal(0, "error calling snd_pcm_sw_params_current: %d", err);
630 if((err = snd_pcm_sw_params_set_avail_min(pcm, swparams, avail_min)) < 0)
631 fatal(0, "error calling snd_pcm_sw_params_set_avail_min %d: %d",
633 if((err = snd_pcm_sw_params(pcm, swparams)) < 0)
634 fatal(0, "error calling snd_pcm_sw_params: %d", err);
637 /** @brief Wait until ALSA wants some audio */
638 static void wait_alsa(void) {
639 struct pollfd fds[64];
641 unsigned short events;
645 if((nfds = snd_pcm_poll_descriptors(pcm,
646 fds, sizeof fds / sizeof *fds)) < 0)
647 fatal(0, "error calling snd_pcm_poll_descriptors: %d", nfds);
648 } while(poll(fds, nfds, -1) < 0 && errno == EINTR);
649 if((err = snd_pcm_poll_descriptors_revents(pcm, fds, nfds, &events)))
650 fatal(0, "error calling snd_pcm_poll_descriptors_revents: %d", err);
656 /** @brief Play some sound via ALSA
657 * @param s Pointer to sample data
658 * @param n Number of samples
659 * @return 0 on success, -1 on non-fatal error
661 static int alsa_writei(const void *s, size_t n) {
663 const snd_pcm_sframes_t frames_written = snd_pcm_writei(pcm, s, n / 2);
664 if(frames_written < 0) {
665 /* Something went wrong */
666 switch(frames_written) {
670 error(0, "error calling snd_pcm_writei: %ld",
671 (long)frames_written);
674 fatal(0, "error calling snd_pcm_writei: %ld",
675 (long)frames_written);
679 next_timestamp += frames_written * 2;
684 /** @brief Play the relevant part of a packet
685 * @param p Packet to play
686 * @return 0 on success, -1 on non-fatal error
688 static int alsa_play(const struct packet *p) {
689 return alsa_writei(p->samples_raw + next_timestamp - p->timestamp,
690 (p->timestamp + p->nsamples) - next_timestamp);
693 /** @brief Play some silence
694 * @param p Next packet or NULL
695 * @return 0 on success, -1 on non-fatal error
697 static int alsa_infill(const struct packet *p) {
698 static const uint16_t zeros[INFILL_SAMPLES];
699 size_t samples_available = INFILL_SAMPLES;
701 if(p && samples_available > p->timestamp - next_timestamp)
702 samples_available = p->timestamp - next_timestamp;
703 return alsa_writei(zeros, samples_available);
706 /** @brief Reset ALSA state after we lost synchronization */
707 static void alsa_reset(int hard_reset) {
710 if((err = snd_pcm_nonblock(pcm, 0)))
711 fatal(0, "error calling snd_pcm_nonblock: %d", err);
713 if((err = snd_pcm_drop(pcm)))
714 fatal(0, "error calling snd_pcm_drop: %d", err);
716 if((err = snd_pcm_drain(pcm)))
717 fatal(0, "error calling snd_pcm_drain: %d", err);
718 if((err = snd_pcm_nonblock(pcm, 1)))
719 fatal(0, "error calling snd_pcm_nonblock: %d", err);
724 /** @brief Play an RTP stream
726 * This is the guts of the program. It is responsible for:
727 * - starting the listening thread
728 * - opening the audio device
729 * - reading ahead to build up a buffer
730 * - arranging for audio to be played
731 * - detecting when the buffer has got too small and re-buffering
733 static void play_rtp(void) {
736 /* We receive and convert audio data in a background thread */
737 pthread_create(<id, 0, listen_thread, 0);
738 /* We have a second thread to add received packets to the queue */
739 pthread_create(<id, 0, queue_thread, 0);
745 /* Open the sound device */
747 pthread_mutex_lock(&lock);
749 /* Wait for the buffer to fill up a bit */
752 if((err = snd_pcm_prepare(pcm)))
753 fatal(0, "error calling snd_pcm_prepare: %d", err);
758 /* Keep playing until the buffer empties out, or ALSA tells us to get
760 while((nsamples >= minbuffer
762 && contains(pheap_first(&packets), next_timestamp)))
764 /* Wait for ALSA to ask us for more data */
765 pthread_mutex_unlock(&lock);
767 pthread_mutex_lock(&lock);
768 /* ALSA is ready for more data, find something to play */
770 /* Play it or play some silence */
771 if(contains(p, next_timestamp))
772 escape = alsa_play(p);
774 escape = alsa_infill(p);
777 /* We stop playing for a bit until the buffer re-fills */
778 pthread_mutex_unlock(&lock);
780 pthread_mutex_lock(&lock);
784 #elif HAVE_COREAUDIO_AUDIOHARDWARE_H
789 AudioStreamBasicDescription asbd;
791 /* If this looks suspiciously like libao's macosx driver there's an
792 * excellent reason for that... */
794 /* TODO report errors as strings not numbers */
795 propertySize = sizeof adid;
796 status = AudioHardwareGetProperty(kAudioHardwarePropertyDefaultOutputDevice,
797 &propertySize, &adid);
799 fatal(0, "AudioHardwareGetProperty: %d", (int)status);
800 if(adid == kAudioDeviceUnknown)
801 fatal(0, "no output device");
802 propertySize = sizeof asbd;
803 status = AudioDeviceGetProperty(adid, 0, false,
804 kAudioDevicePropertyStreamFormat,
805 &propertySize, &asbd);
807 fatal(0, "AudioHardwareGetProperty: %d", (int)status);
808 D(("mSampleRate %f", asbd.mSampleRate));
809 D(("mFormatID %08lx", asbd.mFormatID));
810 D(("mFormatFlags %08lx", asbd.mFormatFlags));
811 D(("mBytesPerPacket %08lx", asbd.mBytesPerPacket));
812 D(("mFramesPerPacket %08lx", asbd.mFramesPerPacket));
813 D(("mBytesPerFrame %08lx", asbd.mBytesPerFrame));
814 D(("mChannelsPerFrame %08lx", asbd.mChannelsPerFrame));
815 D(("mBitsPerChannel %08lx", asbd.mBitsPerChannel));
816 D(("mReserved %08lx", asbd.mReserved));
817 if(asbd.mFormatID != kAudioFormatLinearPCM)
818 fatal(0, "audio device does not support kAudioFormatLinearPCM");
819 status = AudioDeviceAddIOProc(adid, adioproc, 0);
821 fatal(0, "AudioDeviceAddIOProc: %d", (int)status);
822 pthread_mutex_lock(&lock);
824 /* Wait for the buffer to fill up a bit */
826 /* Start playing now */
828 next_timestamp = pheap_first(&packets)->timestamp;
830 status = AudioDeviceStart(adid, adioproc);
832 fatal(0, "AudioDeviceStart: %d", (int)status);
833 /* Wait until the buffer empties out */
834 while(nsamples >= minbuffer
836 && contains(pheap_first(&packets), next_timestamp)))
837 pthread_cond_wait(&cond, &lock);
838 /* Stop playing for a bit until the buffer re-fills */
839 status = AudioDeviceStop(adid, adioproc);
841 fatal(0, "AudioDeviceStop: %d", (int)status);
847 # error No known audio API
851 /* display usage message and terminate */
852 static void help(void) {
854 " disorder-playrtp [OPTIONS] ADDRESS [PORT]\n"
856 " --device, -D DEVICE Output device\n"
857 " --min, -m FRAMES Buffer low water mark\n"
858 " --buffer, -b FRAMES Buffer high water mark\n"
859 " --max, -x FRAMES Buffer maximum size\n"
860 " --rcvbuf, -R BYTES Socket receive buffer size\n"
861 " --help, -h Display usage message\n"
862 " --version, -V Display version number\n"
868 /* display version number and terminate */
869 static void version(void) {
870 xprintf("disorder-playrtp version %s\n", disorder_version_string);
875 int main(int argc, char **argv) {
877 struct addrinfo *res;
878 struct stringlist sl;
880 int rcvbuf, target_rcvbuf = 131072;
883 static const struct addrinfo prefs = {
895 if(!setlocale(LC_CTYPE, "")) fatal(errno, "error calling setlocale");
896 while((n = getopt_long(argc, argv, "hVdD:m:b:x:L:R:", options, 0)) >= 0) {
900 case 'd': debugging = 1; break;
901 case 'D': device = optarg; break;
902 case 'm': minbuffer = 2 * atol(optarg); break;
903 case 'b': readahead = 2 * atol(optarg); break;
904 case 'x': maxbuffer = 2 * atol(optarg); break;
905 case 'L': logfp = fopen(optarg, "w"); break;
906 case 'R': target_rcvbuf = atoi(optarg); break;
907 default: fatal(0, "invalid option");
911 maxbuffer = 4 * readahead;
914 if(argc < 1 || argc > 2)
915 fatal(0, "usage: disorder-playrtp [OPTIONS] ADDRESS [PORT]");
918 /* Listen for inbound audio data */
919 if(!(res = get_address(&sl, &prefs, &sockname)))
921 if((rtpfd = socket(res->ai_family,
923 res->ai_protocol)) < 0)
924 fatal(errno, "error creating socket");
925 if(bind(rtpfd, res->ai_addr, res->ai_addrlen) < 0)
926 fatal(errno, "error binding socket to %s", sockname);
928 if(getsockopt(rtpfd, SOL_SOCKET, SO_RCVBUF, &rcvbuf, &len) < 0)
929 fatal(errno, "error calling getsockopt SO_RCVBUF");
930 if(target_rcvbuf > rcvbuf) {
931 if(setsockopt(rtpfd, SOL_SOCKET, SO_RCVBUF,
932 &target_rcvbuf, sizeof target_rcvbuf) < 0)
933 error(errno, "error calling setsockopt SO_RCVBUF %d",
935 /* We try to carry on anyway */
937 info("changed socket receive buffer from %d to %d",
938 rcvbuf, target_rcvbuf);
940 info("default socket receive buffer %d", rcvbuf);
942 info("WARNING: -L option can impact performance");