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 Output device */
55 static const char *device;
57 /** @brief Maximum samples per packet we'll support
59 * NB that two channels = two samples in this program.
61 #define MAXSAMPLES 2048
63 /** @brief Minimum low watermark
65 * We'll stop playing if there's only this many samples in the buffer. */
66 static unsigned minbuffer = 2 * 44100 / 10; /* 0.2 seconds */
68 /** @brief Maximum sample size
70 * The maximum supported size (in bytes) of one sample. */
71 #define MAXSAMPLESIZE 2
73 /** @brief Buffer high watermark
75 * We'll only start playing when this many samples are available. */
76 static unsigned readahead = 2 * 2 * 44100;
78 /** @brief Maximum buffer size
80 * We'll stop reading from the network if we have this many samples. */
81 static unsigned maxbuffer;
83 /** @brief Number of samples to infill by in one go */
84 #define INFILL_SAMPLES (44100 * 2) /* 1s */
86 /** @brief Received packet
88 * Packets are recorded in an ordered linked list. */
90 /** @brief Pointer to next packet
91 * The next packet might not be immediately next: if packets are dropped
92 * or mis-ordered there may be gaps at any given moment. */
94 /** @brief Number of samples in this packet */
96 /** @brief Timestamp from RTP packet
98 * NB that "timestamps" are really sample counters.*/
100 #if HAVE_COREAUDIO_AUDIOHARDWARE_H
101 /** @brief Converted sample data */
102 float samples_float[MAXSAMPLES];
104 /** @brief Raw sample data */
105 unsigned char samples_raw[MAXSAMPLES * MAXSAMPLESIZE];
109 /** @brief Total number of samples available */
110 static unsigned long nsamples;
112 /** @brief Linked list of packets
114 * In ascending order of timestamp. Really this should be a heap for more
115 * efficient access. */
116 static struct packet *packets;
118 /** @brief Timestamp of next packet to play.
120 * This is set to the timestamp of the last packet, plus the number of
121 * samples it contained. Only valid if @ref active is nonzero.
123 static uint32_t next_timestamp;
125 /** @brief True if actively playing
127 * This is true when playing and false when just buffering. */
130 /** @brief Lock protecting @ref packets */
131 static pthread_mutex_t lock = PTHREAD_MUTEX_INITIALIZER;
133 /** @brief Condition variable signalled whenever @ref packets is changed */
134 static pthread_cond_t cond = PTHREAD_COND_INITIALIZER;
136 static const struct option options[] = {
137 { "help", no_argument, 0, 'h' },
138 { "version", no_argument, 0, 'V' },
139 { "debug", no_argument, 0, 'd' },
140 { "device", required_argument, 0, 'D' },
141 { "min", required_argument, 0, 'm' },
142 { "max", required_argument, 0, 'x' },
143 { "buffer", required_argument, 0, 'b' },
147 /** @brief Return true iff a < b in sequence-space arithmetic */
148 static inline int lt(uint32_t a, uint32_t b) {
149 return (uint32_t)(a - b) & 0x80000000;
152 /** @brief Return true iff a >= b in sequence-space arithmetic */
153 static inline int ge(uint32_t a, uint32_t b) {
157 /** @brief Return true iff a > b in sequence-space arithmetic */
158 static inline int gt(uint32_t a, uint32_t b) {
162 /** @brief Return true iff a <= b in sequence-space arithmetic */
163 static inline int le(uint32_t a, uint32_t b) {
167 /** @brief Drop the packet at the head of the queue */
168 static void drop_first_packet(void) {
169 struct packet *const p = packets;
171 nsamples -= p->nsamples;
173 pthread_cond_broadcast(&cond);
176 /** @brief Background thread collecting samples
178 * This function collects samples, perhaps converts them to the target format,
179 * and adds them to the packet list. */
180 static void *listen_thread(void attribute((unused)) *arg) {
181 struct packet *p = 0, **pp;
184 struct rtp_header header;
185 uint8_t bytes[sizeof(uint16_t) * MAXSAMPLES + sizeof (struct rtp_header)];
187 const uint16_t *const samples = (uint16_t *)(packet.bytes
188 + sizeof (struct rtp_header));
192 p = xmalloc(sizeof *p);
193 n = read(rtpfd, packet.bytes, sizeof packet.bytes);
199 fatal(errno, "error reading from socket");
202 /* Ignore too-short packets */
203 if((size_t)n <= sizeof (struct rtp_header))
205 p->timestamp = ntohl(packet.header.timestamp);
206 /* Ignore packets in the past */
207 if(active && lt(p->timestamp, next_timestamp)) {
208 info("dropping old packet, timestamp=%"PRIx32" < %"PRIx32,
209 p->timestamp, next_timestamp);
212 /* Convert to target format */
213 switch(packet.header.mpt & 0x7F) {
215 p->nsamples = (n - sizeof (struct rtp_header)) / sizeof(uint16_t);
216 #if HAVE_COREAUDIO_AUDIOHARDWARE_H
217 /* Convert to what Core Audio expects */
221 for(i = 0; i < p->nsamples; ++n)
222 p->samples_float[i] = (int16_t)ntohs(samples[i]) * (0.5f / 32767);
225 /* ALSA can do any necessary conversion itself (though it might be better
226 * to do any necessary conversion in the background) */
227 memcpy(p->samples_raw, samples, n - sizeof (struct rtp_header));
230 /* TODO support other RFC3551 media types (when the speaker does) */
232 fatal(0, "unsupported RTP payload type %d",
233 packet.header.mpt & 0x7F);
235 pthread_mutex_lock(&lock);
236 /* Stop reading if we've reached the maximum.
238 * This is rather unsatisfactory: it means that if packets get heavily
239 * out of order then we guarantee dropouts. But for now... */
240 while(nsamples >= maxbuffer)
241 pthread_cond_wait(&cond, &lock);
243 *pp && lt((*pp)->timestamp, p->timestamp);
246 /* So now either !*pp or *pp >= p */
247 if(*pp && p->timestamp == (*pp)->timestamp) {
248 /* *pp == p; a duplicate. Ideally we avoid the translation step here,
249 * but we'll worry about that another time. */
250 info("dropped a duplicated");
253 info("receiving packets out of order");
256 nsamples += p->nsamples;
257 pthread_cond_broadcast(&cond);
258 p = 0; /* we've consumed this packet */
260 pthread_mutex_unlock(&lock);
264 #if HAVE_COREAUDIO_AUDIOHARDWARE_H
265 /** @brief Callback from Core Audio */
266 static OSStatus adioproc
267 (AudioDeviceID attribute((unused)) inDevice,
268 const AudioTimeStamp attribute((unused)) *inNow,
269 const AudioBufferList attribute((unused)) *inInputData,
270 const AudioTimeStamp attribute((unused)) *inInputTime,
271 AudioBufferList *outOutputData,
272 const AudioTimeStamp attribute((unused)) *inOutputTime,
273 void attribute((unused)) *inClientData) {
274 UInt32 nbuffers = outOutputData->mNumberBuffers;
275 AudioBuffer *ab = outOutputData->mBuffers;
277 pthread_mutex_lock(&lock);
278 while(nbuffers > 0) {
279 float *samplesOut = ab->mData;
280 size_t samplesOutLeft = ab->mDataByteSize / sizeof (float);
282 while(samplesOutLeft > 0) {
284 /* There's a packet */
285 const uint32_t packet_start = packets->timestamp;
286 const uint32_t packet_end = packets->timestamp + packets->nsamples;
288 if(le(packet_end, next_timestamp)) {
289 /* This packet is in the past */
290 info("dropping buffered past packet %"PRIx32" < %"PRIx32,
291 packets->timestamp, next_timestamp);
294 if(ge(next_timestamp, packet_start)
295 && lt(next_timestamp, packet_end)) {
296 /* This packet is suitable */
297 const uint32_t offset = next_timestamp - packets->timestamp;
298 uint32_t samples_available = packet_end - next_timestamp;
299 if(samples_available > samplesOutLeft)
300 samples_available = samplesOutLeft;
302 packets->samples_float + offset,
303 samples_available * sizeof(float));
304 samplesOut += samples_available;
305 next_timestamp += samples_available;
306 if(ge(next_timestamp, packet_end))
311 /* We didn't find a suitable packet (though there might still be
312 * unsuitable ones). We infill with 0s. */
314 /* There is a next packet, only infill up to that point */
315 uint32_t samples_available = packets->timestamp - next_timestamp;
317 if(samples_available > samplesOutLeft)
318 samples_available = samplesOutLeft;
319 /* Convniently the buffer is 0 to start with */
320 next_timestamp += samples_available;
321 samplesOut += samples_available;
322 samplesOutLeft -= samples_available;
323 /* TODO log infill */
325 /* There's no next packet at all */
326 next_timestamp += samplesOutLeft;
327 samplesOut += samplesOutLeft;
329 /* TODO log infill */
335 pthread_mutex_unlock(&lock);
340 /** @brief Play an RTP stream
342 * This is the guts of the program. It is responsible for:
343 * - starting the listening thread
344 * - opening the audio device
345 * - reading ahead to build up a buffer
346 * - arranging for audio to be played
347 * - detecting when the buffer has got too small and re-buffering
349 static void play_rtp(void) {
352 /* We receive and convert audio data in a background thread */
353 pthread_create(<id, 0, listen_thread, 0);
357 snd_pcm_hw_params_t *hwparams;
358 snd_pcm_sw_params_t *swparams;
359 /* Only support one format for now */
360 const int sample_format = SND_PCM_FORMAT_S16_BE;
361 unsigned rate = 44100;
362 const int channels = 2;
363 const int samplesize = channels * sizeof(uint16_t);
364 snd_pcm_uframes_t pcm_bufsize = MAXSAMPLES * samplesize * 3;
365 /* If we can write more than this many samples we'll get a wakeup */
366 const int avail_min = 256;
367 snd_pcm_sframes_t frames_written;
368 size_t samples_written;
371 int infilling = 0, escape = 0;
373 uint32_t packet_start, packet_end;
376 if((err = snd_pcm_open(&pcm,
377 device ? device : "default",
378 SND_PCM_STREAM_PLAYBACK,
380 fatal(0, "error from snd_pcm_open: %d", err);
381 /* Set up 'hardware' parameters */
382 snd_pcm_hw_params_alloca(&hwparams);
383 if((err = snd_pcm_hw_params_any(pcm, hwparams)) < 0)
384 fatal(0, "error from snd_pcm_hw_params_any: %d", err);
385 if((err = snd_pcm_hw_params_set_access(pcm, hwparams,
386 SND_PCM_ACCESS_RW_INTERLEAVED)) < 0)
387 fatal(0, "error from snd_pcm_hw_params_set_access: %d", err);
388 if((err = snd_pcm_hw_params_set_format(pcm, hwparams,
390 fatal(0, "error from snd_pcm_hw_params_set_format (%d): %d",
392 if((err = snd_pcm_hw_params_set_rate_near(pcm, hwparams, &rate, 0)) < 0)
393 fatal(0, "error from snd_pcm_hw_params_set_rate (%d): %d",
395 if((err = snd_pcm_hw_params_set_channels(pcm, hwparams,
397 fatal(0, "error from snd_pcm_hw_params_set_channels (%d): %d",
399 if((err = snd_pcm_hw_params_set_buffer_size_near(pcm, hwparams,
401 fatal(0, "error from snd_pcm_hw_params_set_buffer_size (%d): %d",
402 MAXSAMPLES * samplesize * 3, err);
403 if((err = snd_pcm_hw_params(pcm, hwparams)) < 0)
404 fatal(0, "error calling snd_pcm_hw_params: %d", err);
405 /* Set up 'software' parameters */
406 snd_pcm_sw_params_alloca(&swparams);
407 if((err = snd_pcm_sw_params_current(pcm, swparams)) < 0)
408 fatal(0, "error calling snd_pcm_sw_params_current: %d", err);
409 if((err = snd_pcm_sw_params_set_avail_min(pcm, swparams, avail_min)) < 0)
410 fatal(0, "error calling snd_pcm_sw_params_set_avail_min %d: %d",
412 if((err = snd_pcm_sw_params(pcm, swparams)) < 0)
413 fatal(0, "error calling snd_pcm_sw_params: %d", err);
418 pthread_mutex_lock(&lock);
420 /* Wait for the buffer to fill up a bit */
422 info("%lu samples in buffer (%lus)", nsamples,
423 nsamples / (44100 * 2));
424 info("Buffering...");
425 while(nsamples < readahead)
426 pthread_cond_wait(&cond, &lock);
428 if((err = snd_pcm_prepare(pcm)))
429 fatal(0, "error calling snd_pcm_prepare: %d", err);
432 /* Start at the first available packet */
433 next_timestamp = packets->timestamp;
438 info("%lu samples in buffer (%lus)", nsamples,
439 nsamples / (44100 * 2));
441 /* Wait until the buffer empties out */
442 while(nsamples >= minbuffer && !escape) {
444 if(now > logged + 10) {
446 info("%lu samples in buffer (%lus)", nsamples,
447 nsamples / (44100 * 2));
450 && ge(next_timestamp, packets->timestamp + packets->nsamples)) {
451 info("dropping buffered past packet %"PRIx32" < %"PRIx32,
452 packets->timestamp, next_timestamp);
456 /* Wait for ALSA to ask us for more data */
457 pthread_mutex_unlock(&lock);
458 write(2, ".", 1); /* TODO remove me sometime */
459 switch(err = snd_pcm_wait(pcm, -1)) {
461 info("snd_pcm_wait timed out");
466 fatal(0, "snd_pcm_wait returned %d", err);
468 pthread_mutex_lock(&lock);
469 /* ALSA is ready for more data */
470 packet_start = packets->timestamp;
471 packet_end = packets->timestamp + packets->nsamples;
472 if(ge(next_timestamp, packet_start)
473 && lt(next_timestamp, packet_end)) {
474 /* The target timestamp is somewhere in this packet */
475 const uint32_t offset = next_timestamp - packets->timestamp;
476 const uint32_t samples_available = (packets->timestamp + packets->nsamples) - next_timestamp;
477 const size_t frames_available = samples_available / 2;
479 frames_written = snd_pcm_writei(pcm,
480 packets->samples_raw + offset,
482 if(frames_written < 0) {
483 switch(frames_written) {
485 info("snd_pcm_wait() returned but we got -EAGAIN!");
488 error(0, "error calling snd_pcm_writei: %ld",
489 (long)frames_written);
493 fatal(0, "error calling snd_pcm_writei: %ld",
494 (long)frames_written);
497 samples_written = frames_written * 2;
498 next_timestamp += samples_written;
499 if(ge(next_timestamp, packet_end))
504 /* We don't have anything to play! We'd better play some 0s. */
505 static const uint16_t zeros[INFILL_SAMPLES];
506 size_t samples_available = INFILL_SAMPLES, frames_available;
508 /* If the maximum infill would take us past the start of the next
509 * packet then we truncate the infill to the right amount. */
510 if(lt(packets->timestamp,
511 next_timestamp + samples_available))
512 samples_available = packets->timestamp - next_timestamp;
513 if((int)samples_available < 0) {
514 info("packets->timestamp: %"PRIx32" next_timestamp: %"PRIx32" next+max: %"PRIx32" available: %"PRIx32,
515 packets->timestamp, next_timestamp,
516 next_timestamp + INFILL_SAMPLES, samples_available);
518 frames_available = samples_available / 2;
520 info("Infilling %d samples, next=%"PRIx32" packet=[%"PRIx32",%"PRIx32"]",
521 samples_available, next_timestamp,
522 packets->timestamp, packets->timestamp + packets->nsamples);
525 frames_written = snd_pcm_writei(pcm,
528 if(frames_written < 0) {
529 switch(frames_written) {
531 info("snd_pcm_wait() returned but we got -EAGAIN!");
534 error(0, "error calling snd_pcm_writei: %ld",
535 (long)frames_written);
539 fatal(0, "error calling snd_pcm_writei: %ld",
540 (long)frames_written);
543 samples_written = frames_written * 2;
544 next_timestamp += samples_written;
549 /* We stop playing for a bit until the buffer re-fills */
550 pthread_mutex_unlock(&lock);
551 if((err = snd_pcm_nonblock(pcm, 0)))
552 fatal(0, "error calling snd_pcm_nonblock: %d", err);
554 if((err = snd_pcm_drop(pcm)))
555 fatal(0, "error calling snd_pcm_drop: %d", err);
558 if((err = snd_pcm_drain(pcm)))
559 fatal(0, "error calling snd_pcm_drain: %d", err);
560 if((err = snd_pcm_nonblock(pcm, 1)))
561 fatal(0, "error calling snd_pcm_nonblock: %d", err);
563 pthread_mutex_lock(&lock);
567 #elif HAVE_COREAUDIO_AUDIOHARDWARE_H
572 AudioStreamBasicDescription asbd;
574 /* If this looks suspiciously like libao's macosx driver there's an
575 * excellent reason for that... */
577 /* TODO report errors as strings not numbers */
578 propertySize = sizeof adid;
579 status = AudioHardwareGetProperty(kAudioHardwarePropertyDefaultOutputDevice,
580 &propertySize, &adid);
582 fatal(0, "AudioHardwareGetProperty: %d", (int)status);
583 if(adid == kAudioDeviceUnknown)
584 fatal(0, "no output device");
585 propertySize = sizeof asbd;
586 status = AudioDeviceGetProperty(adid, 0, false,
587 kAudioDevicePropertyStreamFormat,
588 &propertySize, &asbd);
590 fatal(0, "AudioHardwareGetProperty: %d", (int)status);
591 D(("mSampleRate %f", asbd.mSampleRate));
592 D(("mFormatID %08lx", asbd.mFormatID));
593 D(("mFormatFlags %08lx", asbd.mFormatFlags));
594 D(("mBytesPerPacket %08lx", asbd.mBytesPerPacket));
595 D(("mFramesPerPacket %08lx", asbd.mFramesPerPacket));
596 D(("mBytesPerFrame %08lx", asbd.mBytesPerFrame));
597 D(("mChannelsPerFrame %08lx", asbd.mChannelsPerFrame));
598 D(("mBitsPerChannel %08lx", asbd.mBitsPerChannel));
599 D(("mReserved %08lx", asbd.mReserved));
600 if(asbd.mFormatID != kAudioFormatLinearPCM)
601 fatal(0, "audio device does not support kAudioFormatLinearPCM");
602 status = AudioDeviceAddIOProc(adid, adioproc, 0);
604 fatal(0, "AudioDeviceAddIOProc: %d", (int)status);
605 pthread_mutex_lock(&lock);
607 /* Wait for the buffer to fill up a bit */
608 while(nsamples < readahead)
609 pthread_cond_wait(&cond, &lock);
610 /* Start playing now */
611 status = AudioDeviceStart(adid, adioproc);
613 fatal(0, "AudioDeviceStart: %d", (int)status);
614 /* Wait until the buffer empties out */
615 while(nsamples >= minbuffer)
616 pthread_cond_wait(&cond, &lock);
617 /* Stop playing for a bit until the buffer re-fills */
618 status = AudioDeviceStop(adid, adioproc);
620 fatal(0, "AudioDeviceStop: %d", (int)status);
625 # error No known audio API
629 /* display usage message and terminate */
630 static void help(void) {
632 " disorder-playrtp [OPTIONS] ADDRESS [PORT]\n"
634 " --device, -D DEVICE Output device\n"
635 " --min, -m FRAMES Buffer low water mark\n"
636 " --buffer, -b FRAMES Buffer high water mark\n"
637 " --max, -x FRAMES Buffer maximum size\n"
638 " --help, -h Display usage message\n"
639 " --version, -V Display version number\n"
645 /* display version number and terminate */
646 static void version(void) {
647 xprintf("disorder-playrtp version %s\n", disorder_version_string);
652 int main(int argc, char **argv) {
654 struct addrinfo *res;
655 struct stringlist sl;
658 static const struct addrinfo prefs = {
670 if(!setlocale(LC_CTYPE, "")) fatal(errno, "error calling setlocale");
671 while((n = getopt_long(argc, argv, "hVdD:m:b:x:", options, 0)) >= 0) {
675 case 'd': debugging = 1; break;
676 case 'D': device = optarg; break;
677 case 'm': minbuffer = 2 * atol(optarg); break;
678 case 'b': readahead = 2 * atol(optarg); break;
679 case 'x': maxbuffer = 2 * atol(optarg); break;
680 default: fatal(0, "invalid option");
684 maxbuffer = 4 * readahead;
687 if(argc < 1 || argc > 2)
688 fatal(0, "usage: disorder-playrtp [OPTIONS] ADDRESS [PORT]");
691 /* Listen for inbound audio data */
692 if(!(res = get_address(&sl, &prefs, &sockname)))
694 if((rtpfd = socket(res->ai_family,
696 res->ai_protocol)) < 0)
697 fatal(errno, "error creating socket");
698 if(bind(rtpfd, res->ai_addr, res->ai_addrlen) < 0)
699 fatal(errno, "error binding socket to %s", sockname);