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
27 #include <sys/socket.h>
28 #include <sys/types.h>
29 #include <sys/socket.h>
36 #include "configuration.h"
42 #if HAVE_COREAUDIO_AUDIOHARDWARE_H
43 # include <CoreAudio/AudioHardware.h>
46 #include <alsa/asoundlib.h>
49 #define readahead linux_headers_are_borked
51 /** @brief RTP socket */
54 /** @brief Log output */
57 /** @brief Output device */
58 static const char *device;
60 /** @brief Maximum samples per packet we'll support
62 * NB that two channels = two samples in this program.
64 #define MAXSAMPLES 2048
66 /** @brief Minimum low watermark
68 * We'll stop playing if there's only this many samples in the buffer. */
69 static unsigned minbuffer = 2 * 44100 / 10; /* 0.2 seconds */
71 /** @brief Maximum sample size
73 * The maximum supported size (in bytes) of one sample. */
74 #define MAXSAMPLESIZE 2
76 /** @brief Buffer high watermark
78 * We'll only start playing when this many samples are available. */
79 static unsigned readahead = 2 * 2 * 44100;
81 /** @brief Maximum buffer size
83 * We'll stop reading from the network if we have this many samples. */
84 static unsigned maxbuffer;
86 /** @brief Number of samples to infill by in one go */
87 #define INFILL_SAMPLES (44100 * 2) /* 1s */
89 /** @brief Received packet
91 * Packets are recorded in an ordered linked list. */
93 /** @brief Pointer to next packet
94 * The next packet might not be immediately next: if packets are dropped
95 * or mis-ordered there may be gaps at any given moment. */
97 /** @brief Number of samples in this packet */
99 /** @brief Timestamp from RTP packet
101 * NB that "timestamps" are really sample counters.*/
103 #if HAVE_COREAUDIO_AUDIOHARDWARE_H
104 /** @brief Converted sample data */
105 float samples_float[MAXSAMPLES];
107 /** @brief Raw sample data */
108 unsigned char samples_raw[MAXSAMPLES * MAXSAMPLESIZE];
112 /** @brief Total number of samples available */
113 static unsigned long nsamples;
115 /** @brief Linked list of packets
117 * In ascending order of timestamp. Really this should be a heap for more
118 * efficient access. */
119 static struct packet *packets;
121 /** @brief Timestamp of next packet to play.
123 * This is set to the timestamp of the last packet, plus the number of
124 * samples it contained. Only valid if @ref active is nonzero.
126 static uint32_t next_timestamp;
128 /** @brief True if actively playing
130 * This is true when playing and false when just buffering. */
133 /** @brief Lock protecting @ref packets */
134 static pthread_mutex_t lock = PTHREAD_MUTEX_INITIALIZER;
136 /** @brief Condition variable signalled whenever @ref packets is changed */
137 static pthread_cond_t cond = PTHREAD_COND_INITIALIZER;
139 static const struct option options[] = {
140 { "help", no_argument, 0, 'h' },
141 { "version", no_argument, 0, 'V' },
142 { "debug", no_argument, 0, 'd' },
143 { "device", required_argument, 0, 'D' },
144 { "min", required_argument, 0, 'm' },
145 { "max", required_argument, 0, 'x' },
146 { "buffer", required_argument, 0, 'b' },
150 /** @brief Return true iff a < b in sequence-space arithmetic */
151 static inline int lt(uint32_t a, uint32_t b) {
152 return (uint32_t)(a - b) & 0x80000000;
155 /** @brief Return true iff a >= b in sequence-space arithmetic */
156 static inline int ge(uint32_t a, uint32_t b) {
160 /** @brief Return true iff a > b in sequence-space arithmetic */
161 static inline int gt(uint32_t a, uint32_t b) {
165 /** @brief Return true iff a <= b in sequence-space arithmetic */
166 static inline int le(uint32_t a, uint32_t b) {
170 /** @brief Drop the packet at the head of the queue */
171 static void drop_first_packet(void) {
172 struct packet *const p = packets;
174 nsamples -= p->nsamples;
176 pthread_cond_broadcast(&cond);
179 /** @brief Background thread collecting samples
181 * This function collects samples, perhaps converts them to the target format,
182 * and adds them to the packet list. */
183 static void *listen_thread(void attribute((unused)) *arg) {
184 struct packet *p = 0, **pp;
187 struct rtp_header header;
188 uint8_t bytes[sizeof(uint16_t) * MAXSAMPLES + sizeof (struct rtp_header)];
190 const uint16_t *const samples = (uint16_t *)(packet.bytes
191 + sizeof (struct rtp_header));
195 p = xmalloc(sizeof *p);
196 n = read(rtpfd, packet.bytes, sizeof packet.bytes);
202 fatal(errno, "error reading from socket");
205 /* Ignore too-short packets */
206 if((size_t)n <= sizeof (struct rtp_header)) {
207 info("ignored a short packet");
210 p->timestamp = ntohl(packet.header.timestamp);
211 /* Ignore packets in the past */
212 if(active && lt(p->timestamp, next_timestamp)) {
213 info("dropping old packet, timestamp=%"PRIx32" < %"PRIx32,
214 p->timestamp, next_timestamp);
217 /* Convert to target format */
218 switch(packet.header.mpt & 0x7F) {
220 p->nsamples = (n - sizeof (struct rtp_header)) / sizeof(uint16_t);
221 #if HAVE_COREAUDIO_AUDIOHARDWARE_H
222 /* Convert to what Core Audio expects */
226 for(i = 0; i < p->nsamples; ++i)
227 p->samples_float[i] = (int16_t)ntohs(samples[i]) * (0.5f / 32767);
230 /* ALSA can do any necessary conversion itself (though it might be better
231 * to do any necessary conversion in the background) */
232 memcpy(p->samples_raw, samples, n - sizeof (struct rtp_header));
235 /* TODO support other RFC3551 media types (when the speaker does) */
237 fatal(0, "unsupported RTP payload type %d",
238 packet.header.mpt & 0x7F);
241 fprintf(logfp, "sequence %u timestamp %"PRIx32" length %"PRIx32" end %"PRIx32"\n",
242 ntohs(packet.header.seq),
243 p->timestamp, p->nsamples, p->timestamp + p->nsamples);
244 pthread_mutex_lock(&lock);
245 /* Stop reading if we've reached the maximum.
247 * This is rather unsatisfactory: it means that if packets get heavily
248 * out of order then we guarantee dropouts. But for now... */
249 if(nsamples >= maxbuffer) {
251 while(nsamples >= maxbuffer)
252 pthread_cond_wait(&cond, &lock);
255 *pp && lt((*pp)->timestamp, p->timestamp);
258 /* So now either !*pp or *pp >= p */
259 if(*pp && p->timestamp == (*pp)->timestamp) {
260 /* *pp == p; a duplicate. Ideally we avoid the translation step here,
261 * but we'll worry about that another time. */
262 info("dropped a duplicated");
265 info("receiving packets out of order");
268 nsamples += p->nsamples;
269 pthread_cond_broadcast(&cond);
270 p = 0; /* we've consumed this packet */
272 pthread_mutex_unlock(&lock);
276 #if HAVE_COREAUDIO_AUDIOHARDWARE_H
277 /** @brief Callback from Core Audio */
278 static OSStatus adioproc
279 (AudioDeviceID attribute((unused)) inDevice,
280 const AudioTimeStamp attribute((unused)) *inNow,
281 const AudioBufferList attribute((unused)) *inInputData,
282 const AudioTimeStamp attribute((unused)) *inInputTime,
283 AudioBufferList *outOutputData,
284 const AudioTimeStamp attribute((unused)) *inOutputTime,
285 void attribute((unused)) *inClientData) {
286 UInt32 nbuffers = outOutputData->mNumberBuffers;
287 AudioBuffer *ab = outOutputData->mBuffers;
289 pthread_mutex_lock(&lock);
290 while(nbuffers > 0) {
291 float *samplesOut = ab->mData;
292 size_t samplesOutLeft = ab->mDataByteSize / sizeof (float);
294 while(samplesOutLeft > 0) {
296 /* There's a packet */
297 const uint32_t packet_start = packets->timestamp;
298 const uint32_t packet_end = packets->timestamp + packets->nsamples;
300 if(le(packet_end, next_timestamp)) {
301 /* This packet is in the past */
302 info("dropping buffered past packet %"PRIx32" < %"PRIx32,
303 packet_start, next_timestamp);
307 if(ge(next_timestamp, packet_start)
308 && lt(next_timestamp, packet_end)) {
309 /* This packet is suitable */
310 const uint32_t offset = next_timestamp - packet_start;
311 uint32_t samples_available = packet_end - next_timestamp;
312 if(samples_available > samplesOutLeft)
313 samples_available = samplesOutLeft;
315 packets->samples_float + offset,
316 samples_available * sizeof(float));
317 samplesOut += samples_available;
318 next_timestamp += samples_available;
319 samplesOutLeft -= samples_available;
320 if(ge(next_timestamp, packet_end))
325 /* We didn't find a suitable packet (though there might still be
326 * unsuitable ones). We infill with 0s. */
328 /* There is a next packet, only infill up to that point */
329 uint32_t samples_available = packets->timestamp - next_timestamp;
331 if(samples_available > samplesOutLeft)
332 samples_available = samplesOutLeft;
333 info("infill by %"PRIu32, samples_available);
334 /* Convniently the buffer is 0 to start with */
335 next_timestamp += samples_available;
336 samplesOut += samples_available;
337 samplesOutLeft -= samples_available;
339 /* There's no next packet at all */
340 info("infilled by %zu", samplesOutLeft);
341 next_timestamp += samplesOutLeft;
342 samplesOut += samplesOutLeft;
349 pthread_mutex_unlock(&lock);
354 /** @brief Play an RTP stream
356 * This is the guts of the program. It is responsible for:
357 * - starting the listening thread
358 * - opening the audio device
359 * - reading ahead to build up a buffer
360 * - arranging for audio to be played
361 * - detecting when the buffer has got too small and re-buffering
363 static void play_rtp(void) {
366 /* We receive and convert audio data in a background thread */
367 pthread_create(<id, 0, listen_thread, 0);
371 snd_pcm_hw_params_t *hwparams;
372 snd_pcm_sw_params_t *swparams;
373 /* Only support one format for now */
374 const int sample_format = SND_PCM_FORMAT_S16_BE;
375 unsigned rate = 44100;
376 const int channels = 2;
377 const int samplesize = channels * sizeof(uint16_t);
378 snd_pcm_uframes_t pcm_bufsize = MAXSAMPLES * samplesize * 3;
379 /* If we can write more than this many samples we'll get a wakeup */
380 const int avail_min = 256;
381 snd_pcm_sframes_t frames_written;
382 size_t samples_written;
385 int infilling = 0, escape = 0;
387 uint32_t packet_start, packet_end;
390 if((err = snd_pcm_open(&pcm,
391 device ? device : "default",
392 SND_PCM_STREAM_PLAYBACK,
394 fatal(0, "error from snd_pcm_open: %d", err);
395 /* Set up 'hardware' parameters */
396 snd_pcm_hw_params_alloca(&hwparams);
397 if((err = snd_pcm_hw_params_any(pcm, hwparams)) < 0)
398 fatal(0, "error from snd_pcm_hw_params_any: %d", err);
399 if((err = snd_pcm_hw_params_set_access(pcm, hwparams,
400 SND_PCM_ACCESS_RW_INTERLEAVED)) < 0)
401 fatal(0, "error from snd_pcm_hw_params_set_access: %d", err);
402 if((err = snd_pcm_hw_params_set_format(pcm, hwparams,
404 fatal(0, "error from snd_pcm_hw_params_set_format (%d): %d",
406 if((err = snd_pcm_hw_params_set_rate_near(pcm, hwparams, &rate, 0)) < 0)
407 fatal(0, "error from snd_pcm_hw_params_set_rate (%d): %d",
409 if((err = snd_pcm_hw_params_set_channels(pcm, hwparams,
411 fatal(0, "error from snd_pcm_hw_params_set_channels (%d): %d",
413 if((err = snd_pcm_hw_params_set_buffer_size_near(pcm, hwparams,
415 fatal(0, "error from snd_pcm_hw_params_set_buffer_size (%d): %d",
416 MAXSAMPLES * samplesize * 3, err);
417 if((err = snd_pcm_hw_params(pcm, hwparams)) < 0)
418 fatal(0, "error calling snd_pcm_hw_params: %d", err);
419 /* Set up 'software' parameters */
420 snd_pcm_sw_params_alloca(&swparams);
421 if((err = snd_pcm_sw_params_current(pcm, swparams)) < 0)
422 fatal(0, "error calling snd_pcm_sw_params_current: %d", err);
423 if((err = snd_pcm_sw_params_set_avail_min(pcm, swparams, avail_min)) < 0)
424 fatal(0, "error calling snd_pcm_sw_params_set_avail_min %d: %d",
426 if((err = snd_pcm_sw_params(pcm, swparams)) < 0)
427 fatal(0, "error calling snd_pcm_sw_params: %d", err);
432 pthread_mutex_lock(&lock);
434 /* Wait for the buffer to fill up a bit */
436 info("%lu samples in buffer (%lus)", nsamples,
437 nsamples / (44100 * 2));
438 info("Buffering...");
439 while(nsamples < readahead)
440 pthread_cond_wait(&cond, &lock);
442 if((err = snd_pcm_prepare(pcm)))
443 fatal(0, "error calling snd_pcm_prepare: %d", err);
446 /* Start at the first available packet */
447 next_timestamp = packets->timestamp;
452 info("%lu samples in buffer (%lus)", nsamples,
453 nsamples / (44100 * 2));
455 /* Wait until the buffer empties out */
456 while(nsamples >= minbuffer && !escape) {
458 if(now > logged + 10) {
460 info("%lu samples in buffer (%lus)", nsamples,
461 nsamples / (44100 * 2));
464 && ge(next_timestamp, packets->timestamp + packets->nsamples)) {
465 info("dropping buffered past packet %"PRIx32" < %"PRIx32,
466 packets->timestamp, next_timestamp);
470 /* Wait for ALSA to ask us for more data */
471 pthread_mutex_unlock(&lock);
472 write(2, ".", 1); /* TODO remove me sometime */
473 switch(err = snd_pcm_wait(pcm, -1)) {
475 info("snd_pcm_wait timed out");
480 fatal(0, "snd_pcm_wait returned %d", err);
482 pthread_mutex_lock(&lock);
483 /* ALSA is ready for more data */
484 packet_start = packets->timestamp;
485 packet_end = packets->timestamp + packets->nsamples;
486 if(ge(next_timestamp, packet_start)
487 && lt(next_timestamp, packet_end)) {
488 /* The target timestamp is somewhere in this packet */
489 const uint32_t offset = next_timestamp - packets->timestamp;
490 const uint32_t samples_available = (packets->timestamp + packets->nsamples) - next_timestamp;
491 const size_t frames_available = samples_available / 2;
493 frames_written = snd_pcm_writei(pcm,
494 packets->samples_raw + offset,
496 if(frames_written < 0) {
497 switch(frames_written) {
499 info("snd_pcm_wait() returned but we got -EAGAIN!");
502 error(0, "error calling snd_pcm_writei: %ld",
503 (long)frames_written);
507 fatal(0, "error calling snd_pcm_writei: %ld",
508 (long)frames_written);
511 samples_written = frames_written * 2;
512 next_timestamp += samples_written;
513 if(ge(next_timestamp, packet_end))
518 /* We don't have anything to play! We'd better play some 0s. */
519 static const uint16_t zeros[INFILL_SAMPLES];
520 size_t samples_available = INFILL_SAMPLES, frames_available;
522 /* If the maximum infill would take us past the start of the next
523 * packet then we truncate the infill to the right amount. */
524 if(lt(packets->timestamp,
525 next_timestamp + samples_available))
526 samples_available = packets->timestamp - next_timestamp;
527 if((int)samples_available < 0) {
528 info("packets->timestamp: %"PRIx32" next_timestamp: %"PRIx32" next+max: %"PRIx32" available: %"PRIx32,
529 packets->timestamp, next_timestamp,
530 next_timestamp + INFILL_SAMPLES, samples_available);
532 frames_available = samples_available / 2;
534 info("Infilling %d samples, next=%"PRIx32" packet=[%"PRIx32",%"PRIx32"]",
535 samples_available, next_timestamp,
536 packets->timestamp, packets->timestamp + packets->nsamples);
539 frames_written = snd_pcm_writei(pcm,
542 if(frames_written < 0) {
543 switch(frames_written) {
545 info("snd_pcm_wait() returned but we got -EAGAIN!");
548 error(0, "error calling snd_pcm_writei: %ld",
549 (long)frames_written);
553 fatal(0, "error calling snd_pcm_writei: %ld",
554 (long)frames_written);
557 samples_written = frames_written * 2;
558 next_timestamp += samples_written;
563 /* We stop playing for a bit until the buffer re-fills */
564 pthread_mutex_unlock(&lock);
565 if((err = snd_pcm_nonblock(pcm, 0)))
566 fatal(0, "error calling snd_pcm_nonblock: %d", err);
568 if((err = snd_pcm_drop(pcm)))
569 fatal(0, "error calling snd_pcm_drop: %d", err);
572 if((err = snd_pcm_drain(pcm)))
573 fatal(0, "error calling snd_pcm_drain: %d", err);
574 if((err = snd_pcm_nonblock(pcm, 1)))
575 fatal(0, "error calling snd_pcm_nonblock: %d", err);
577 pthread_mutex_lock(&lock);
581 #elif HAVE_COREAUDIO_AUDIOHARDWARE_H
586 AudioStreamBasicDescription asbd;
588 /* If this looks suspiciously like libao's macosx driver there's an
589 * excellent reason for that... */
591 /* TODO report errors as strings not numbers */
592 propertySize = sizeof adid;
593 status = AudioHardwareGetProperty(kAudioHardwarePropertyDefaultOutputDevice,
594 &propertySize, &adid);
596 fatal(0, "AudioHardwareGetProperty: %d", (int)status);
597 if(adid == kAudioDeviceUnknown)
598 fatal(0, "no output device");
599 propertySize = sizeof asbd;
600 status = AudioDeviceGetProperty(adid, 0, false,
601 kAudioDevicePropertyStreamFormat,
602 &propertySize, &asbd);
604 fatal(0, "AudioHardwareGetProperty: %d", (int)status);
605 D(("mSampleRate %f", asbd.mSampleRate));
606 D(("mFormatID %08lx", asbd.mFormatID));
607 D(("mFormatFlags %08lx", asbd.mFormatFlags));
608 D(("mBytesPerPacket %08lx", asbd.mBytesPerPacket));
609 D(("mFramesPerPacket %08lx", asbd.mFramesPerPacket));
610 D(("mBytesPerFrame %08lx", asbd.mBytesPerFrame));
611 D(("mChannelsPerFrame %08lx", asbd.mChannelsPerFrame));
612 D(("mBitsPerChannel %08lx", asbd.mBitsPerChannel));
613 D(("mReserved %08lx", asbd.mReserved));
614 if(asbd.mFormatID != kAudioFormatLinearPCM)
615 fatal(0, "audio device does not support kAudioFormatLinearPCM");
616 status = AudioDeviceAddIOProc(adid, adioproc, 0);
618 fatal(0, "AudioDeviceAddIOProc: %d", (int)status);
619 pthread_mutex_lock(&lock);
621 /* Wait for the buffer to fill up a bit */
622 info("Buffering...");
623 while(nsamples < readahead)
624 pthread_cond_wait(&cond, &lock);
625 /* Start playing now */
627 next_timestamp = packets->timestamp;
629 status = AudioDeviceStart(adid, adioproc);
631 fatal(0, "AudioDeviceStart: %d", (int)status);
632 /* Wait until the buffer empties out */
633 while(nsamples >= minbuffer)
634 pthread_cond_wait(&cond, &lock);
635 /* Stop playing for a bit until the buffer re-fills */
636 status = AudioDeviceStop(adid, adioproc);
638 fatal(0, "AudioDeviceStop: %d", (int)status);
644 # error No known audio API
648 /* display usage message and terminate */
649 static void help(void) {
651 " disorder-playrtp [OPTIONS] ADDRESS [PORT]\n"
653 " --device, -D DEVICE Output device\n"
654 " --min, -m FRAMES Buffer low water mark\n"
655 " --buffer, -b FRAMES Buffer high water mark\n"
656 " --max, -x FRAMES Buffer maximum size\n"
657 " --help, -h Display usage message\n"
658 " --version, -V Display version number\n"
664 /* display version number and terminate */
665 static void version(void) {
666 xprintf("disorder-playrtp version %s\n", disorder_version_string);
671 int main(int argc, char **argv) {
673 struct addrinfo *res;
674 struct stringlist sl;
677 static const struct addrinfo prefs = {
689 if(!setlocale(LC_CTYPE, "")) fatal(errno, "error calling setlocale");
690 while((n = getopt_long(argc, argv, "hVdD:m:b:x:L:", options, 0)) >= 0) {
694 case 'd': debugging = 1; break;
695 case 'D': device = optarg; break;
696 case 'm': minbuffer = 2 * atol(optarg); break;
697 case 'b': readahead = 2 * atol(optarg); break;
698 case 'x': maxbuffer = 2 * atol(optarg); break;
699 case 'L': logfp = fopen(optarg, "w"); break;
700 default: fatal(0, "invalid option");
704 maxbuffer = 4 * readahead;
707 if(argc < 1 || argc > 2)
708 fatal(0, "usage: disorder-playrtp [OPTIONS] ADDRESS [PORT]");
711 /* Listen for inbound audio data */
712 if(!(res = get_address(&sl, &prefs, &sockname)))
714 if((rtpfd = socket(res->ai_family,
716 res->ai_protocol)) < 0)
717 fatal(errno, "error creating socket");
718 if(bind(rtpfd, res->ai_addr, res->ai_addrlen) < 0)
719 fatal(errno, "error binding socket to %s", sockname);