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 buffer size
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 size
75 * We'll only start playing when this many samples are available. */
76 static unsigned readahead = 2 * 2 * 44100;
78 /** @brief Number of samples to infill by in one go */
79 #define INFILL_SAMPLES (44100 * 2) /* 1s */
81 #define MAXBUFFER (3 * 88200) /* maximum buffer contents */
83 /** @brief Received packet
85 * Packets are recorded in an ordered linked list. */
87 /** @brief Pointer to next packet
88 * The next packet might not be immediately next: if packets are dropped
89 * or mis-ordered there may be gaps at any given moment. */
91 /** @brief Number of samples in this packet */
93 /** @brief Timestamp from RTP packet
95 * NB that "timestamps" are really sample counters.*/
97 #if HAVE_COREAUDIO_AUDIOHARDWARE_H
98 /** @brief Converted sample data */
99 float samples_float[MAXSAMPLES];
101 /** @brief Raw sample data */
102 unsigned char samples_raw[MAXSAMPLES * MAXSAMPLESIZE];
106 /** @brief Total number of samples available */
107 static unsigned long nsamples;
109 /** @brief Linked list of packets
111 * In ascending order of timestamp. */
112 static struct packet *packets;
114 /** @brief Timestamp of next packet to play.
116 * This is set to the timestamp of the last packet, plus the number of
117 * samples it contained. Only valid if @ref active is nonzero.
119 static uint32_t next_timestamp;
121 /** @brief True if actively playing
123 * This is true when playing and false when just buffering. */
126 /** @brief Lock protecting @ref packets */
127 static pthread_mutex_t lock = PTHREAD_MUTEX_INITIALIZER;
129 /** @brief Condition variable signalled whenever @ref packets is changed */
130 static pthread_cond_t cond = PTHREAD_COND_INITIALIZER;
132 static const struct option options[] = {
133 { "help", no_argument, 0, 'h' },
134 { "version", no_argument, 0, 'V' },
135 { "debug", no_argument, 0, 'd' },
136 { "device", required_argument, 0, 'D' },
137 { "min", required_argument, 0, 'm' },
138 { "buffer", required_argument, 0, 'b' },
142 /** @brief Return true iff a < b in sequence-space arithmetic */
143 static inline int lt(uint32_t a, uint32_t b) {
144 return (uint32_t)(a - b) & 0x80000000;
147 /** @brief Return true iff a >= b in sequence-space arithmetic */
148 static inline int ge(uint32_t a, uint32_t b) {
152 /** @brief Return true iff a > b in sequence-space arithmetic */
153 static inline int gt(uint32_t a, uint32_t b) {
157 /** @brief Return true iff a <= b in sequence-space arithmetic */
158 static inline int le(uint32_t a, uint32_t b) {
162 /** @brief Background thread collecting samples
164 * This function collects samples, perhaps converts them to the target format,
165 * and adds them to the packet list. */
166 static void *listen_thread(void attribute((unused)) *arg) {
167 struct packet *p = 0, **pp;
170 struct rtp_header header;
171 uint8_t bytes[sizeof(uint16_t) * MAXSAMPLES + sizeof (struct rtp_header)];
173 const uint16_t *const samples = (uint16_t *)(packet.bytes
174 + sizeof (struct rtp_header));
178 p = xmalloc(sizeof *p);
179 n = read(rtpfd, packet.bytes, sizeof packet.bytes);
185 fatal(errno, "error reading from socket");
188 /* Ignore too-short packets */
189 if((size_t)n <= sizeof (struct rtp_header))
191 p->timestamp = ntohl(packet.header.timestamp);
192 /* Ignore packets in the past */
193 if(active && lt(p->timestamp, next_timestamp)) {
194 info("dropping old packet, timestamp=%"PRIx32" < %"PRIx32,
195 p->timestamp, next_timestamp);
198 /* Convert to target format */
199 switch(packet.header.mpt & 0x7F) {
201 p->nsamples = (n - sizeof (struct rtp_header)) / sizeof(uint16_t);
202 #if HAVE_COREAUDIO_AUDIOHARDWARE_H
203 /* Convert to what Core Audio expects */
204 for(n = 0; n < p->nsamples; ++n)
205 p->samples_float[n] = (int16_t)ntohs(samples[n]) * (0.5f / 32767);
207 /* ALSA can do any necessary conversion itself (though it might be better
208 * to do any necessary conversion in the background) */
209 memcpy(p->samples_raw, samples, n - sizeof (struct rtp_header));
212 /* TODO support other RFC3551 media types (when the speaker does) */
214 fatal(0, "unsupported RTP payload type %d",
215 packet.header.mpt & 0x7F);
217 pthread_mutex_lock(&lock);
218 /* Stop reading if we've reached the maximum.
220 * This is rather unsatisfactory: it means that if packets get heavily
221 * out of order then we guarantee dropouts. But for now... */
222 while(nsamples >= MAXBUFFER)
223 pthread_cond_wait(&cond, &lock);
225 *pp && lt((*pp)->timestamp, p->timestamp);
228 /* So now either !*pp or *pp >= p */
229 if(*pp && p->timestamp == (*pp)->timestamp) {
230 /* *pp == p; a duplicate. Ideally we avoid the translation step here,
231 * but we'll worry about that another time. */
232 info("dropped a duplicated");
235 info("receiving packets out of order");
238 nsamples += p->nsamples;
239 pthread_cond_broadcast(&cond);
240 p = 0; /* we've consumed this packet */
242 pthread_mutex_unlock(&lock);
246 #if HAVE_COREAUDIO_AUDIOHARDWARE_H
247 /** @brief Callback from Core Audio */
248 static OSStatus adioproc(AudioDeviceID inDevice,
249 const AudioTimeStamp *inNow,
250 const AudioBufferList *inInputData,
251 const AudioTimeStamp *inInputTime,
252 AudioBufferList *outOutputData,
253 const AudioTimeStamp *inOutputTime,
254 void *inClientData) {
255 UInt32 nbuffers = outOutputData->mNumberBuffers;
256 AudioBuffer *ab = outOutputData->mBuffers;
257 float *samplesOut; /* where to write samples to */
258 size_t samplesOutLeft; /* space left */
259 size_t samplesInLeft;
260 size_t samplesToCopy;
262 pthread_mutex_lock(&lock);
263 samplesOut = ab->data;
264 samplesOutLeft = ab->mDataByteSize / sizeof (float);
265 while(packets && nbuffers > 0) {
266 if(packets->used == packets->nsamples) {
267 /* TODO if we dropped a packet then we should introduce a gap here */
268 struct packet *const p = packets;
271 pthread_cond_broadcast(&cond);
274 if(samplesOutLeft == 0) {
277 samplesOut = ab->data;
278 samplesOutLeft = ab->mDataByteSize / sizeof (float);
281 /* Now: (1) there is some data left to read
282 * (2) there is some space to put it */
283 samplesInLeft = packets->nsamples - packets->used;
284 samplesToCopy = (samplesInLeft < samplesOutLeft
285 ? samplesInLeft : samplesOutLeft);
286 memcpy(samplesOut, packet->samples + packets->used, samplesToCopy);
287 packets->used += samplesToCopy;
288 samplesOut += samplesToCopy;
289 samesOutLeft -= samplesToCopy;
291 pthread_mutex_unlock(&lock);
296 /** @brief Play an RTP stream
298 * This is the guts of the program. It is responsible for:
299 * - starting the listening thread
300 * - opening the audio device
301 * - reading ahead to build up a buffer
302 * - arranging for audio to be played
303 * - detecting when the buffer has got too small and re-buffering
305 static void play_rtp(void) {
308 /* We receive and convert audio data in a background thread */
309 pthread_create(<id, 0, listen_thread, 0);
313 snd_pcm_hw_params_t *hwparams;
314 snd_pcm_sw_params_t *swparams;
315 /* Only support one format for now */
316 const int sample_format = SND_PCM_FORMAT_S16_BE;
317 unsigned rate = 44100;
318 const int channels = 2;
319 const int samplesize = channels * sizeof(uint16_t);
320 snd_pcm_uframes_t pcm_bufsize = MAXSAMPLES * samplesize * 3;
321 /* If we can write more than this many samples we'll get a wakeup */
322 const int avail_min = 256;
323 snd_pcm_sframes_t frames_written;
324 size_t samples_written;
327 int infilling = 0, escape = 0;
329 uint32_t packet_start, packet_end;
332 if((err = snd_pcm_open(&pcm,
333 device ? device : "default",
334 SND_PCM_STREAM_PLAYBACK,
336 fatal(0, "error from snd_pcm_open: %d", err);
337 /* Set up 'hardware' parameters */
338 snd_pcm_hw_params_alloca(&hwparams);
339 if((err = snd_pcm_hw_params_any(pcm, hwparams)) < 0)
340 fatal(0, "error from snd_pcm_hw_params_any: %d", err);
341 if((err = snd_pcm_hw_params_set_access(pcm, hwparams,
342 SND_PCM_ACCESS_RW_INTERLEAVED)) < 0)
343 fatal(0, "error from snd_pcm_hw_params_set_access: %d", err);
344 if((err = snd_pcm_hw_params_set_format(pcm, hwparams,
346 fatal(0, "error from snd_pcm_hw_params_set_format (%d): %d",
348 if((err = snd_pcm_hw_params_set_rate_near(pcm, hwparams, &rate, 0)) < 0)
349 fatal(0, "error from snd_pcm_hw_params_set_rate (%d): %d",
351 if((err = snd_pcm_hw_params_set_channels(pcm, hwparams,
353 fatal(0, "error from snd_pcm_hw_params_set_channels (%d): %d",
355 if((err = snd_pcm_hw_params_set_buffer_size_near(pcm, hwparams,
357 fatal(0, "error from snd_pcm_hw_params_set_buffer_size (%d): %d",
358 MAXSAMPLES * samplesize * 3, err);
359 if((err = snd_pcm_hw_params(pcm, hwparams)) < 0)
360 fatal(0, "error calling snd_pcm_hw_params: %d", err);
361 /* Set up 'software' parameters */
362 snd_pcm_sw_params_alloca(&swparams);
363 if((err = snd_pcm_sw_params_current(pcm, swparams)) < 0)
364 fatal(0, "error calling snd_pcm_sw_params_current: %d", err);
365 if((err = snd_pcm_sw_params_set_avail_min(pcm, swparams, avail_min)) < 0)
366 fatal(0, "error calling snd_pcm_sw_params_set_avail_min %d: %d",
368 if((err = snd_pcm_sw_params(pcm, swparams)) < 0)
369 fatal(0, "error calling snd_pcm_sw_params: %d", err);
374 pthread_mutex_lock(&lock);
376 /* Wait for the buffer to fill up a bit */
378 info("%lu samples in buffer (%lus)", nsamples,
379 nsamples / (44100 * 2));
380 info("Buffering...");
381 while(nsamples < readahead)
382 pthread_cond_wait(&cond, &lock);
384 if((err = snd_pcm_prepare(pcm)))
385 fatal(0, "error calling snd_pcm_prepare: %d", err);
388 /* Start at the first available packet */
389 next_timestamp = packets->timestamp;
394 info("%lu samples in buffer (%lus)", nsamples,
395 nsamples / (44100 * 2));
397 /* Wait until the buffer empties out */
398 while(nsamples >= minbuffer && !escape) {
400 if(now > logged + 10) {
402 info("%lu samples in buffer (%lus)", nsamples,
403 nsamples / (44100 * 2));
406 && ge(next_timestamp, packets->timestamp + packets->nsamples)) {
407 struct packet *p = packets;
409 info("dropping buffered past packet %"PRIx32" < %"PRIx32,
410 packets->timestamp, next_timestamp);
414 assert(lt(p->timestamp, packets->timestamp));
415 nsamples -= p->nsamples;
417 pthread_cond_broadcast(&cond);
420 /* Wait for ALSA to ask us for more data */
421 pthread_mutex_unlock(&lock);
422 write(2, ".", 1); /* TODO remove me sometime */
423 switch(err = snd_pcm_wait(pcm, -1)) {
425 info("snd_pcm_wait timed out");
430 fatal(0, "snd_pcm_wait returned %d", err);
432 pthread_mutex_lock(&lock);
433 /* ALSA is ready for more data */
434 packet_start = packets->timestamp;
435 packet_end = packets->timestamp + packets->nsamples;
436 if(ge(next_timestamp, packet_start)
437 && lt(next_timestamp, packet_end)) {
438 /* The target timestamp is somewhere in this packet */
439 const uint32_t offset = next_timestamp - packets->timestamp;
440 const uint32_t samples_available = (packets->timestamp + packets->nsamples) - next_timestamp;
441 const size_t frames_available = samples_available / 2;
443 frames_written = snd_pcm_writei(pcm,
444 packets->samples_raw + offset,
446 if(frames_written < 0) {
447 switch(frames_written) {
449 info("snd_pcm_wait() returned but we got -EAGAIN!");
452 error(0, "error calling snd_pcm_writei: %ld",
453 (long)frames_written);
457 fatal(0, "error calling snd_pcm_writei: %ld",
458 (long)frames_written);
461 samples_written = frames_written * 2;
462 next_timestamp += samples_written;
463 if(ge(next_timestamp, packet_end)) {
464 /* We're done with this packet */
465 struct packet *p = packets;
469 assert(lt(p->timestamp, packets->timestamp));
470 nsamples -= p->nsamples;
472 pthread_cond_broadcast(&cond);
477 /* We don't have anything to play! We'd better play some 0s. */
478 static const uint16_t zeros[INFILL_SAMPLES];
479 size_t samples_available = INFILL_SAMPLES, frames_available;
481 /* If the maximum infill would take us past the start of the next
482 * packet then we truncate the infill to the right amount. */
483 if(lt(packets->timestamp,
484 next_timestamp + samples_available))
485 samples_available = packets->timestamp - next_timestamp;
486 if((int)samples_available < 0) {
487 info("packets->timestamp: %"PRIx32" next_timestamp: %"PRIx32" next+max: %"PRIx32" available: %"PRIx32,
488 packets->timestamp, next_timestamp,
489 next_timestamp + INFILL_SAMPLES, samples_available);
491 frames_available = samples_available / 2;
493 info("Infilling %d samples, next=%"PRIx32" packet=[%"PRIx32",%"PRIx32"]",
494 samples_available, next_timestamp,
495 packets->timestamp, packets->timestamp + packets->nsamples);
498 frames_written = snd_pcm_writei(pcm,
501 if(frames_written < 0) {
502 switch(frames_written) {
504 info("snd_pcm_wait() returned but we got -EAGAIN!");
507 error(0, "error calling snd_pcm_writei: %ld",
508 (long)frames_written);
512 fatal(0, "error calling snd_pcm_writei: %ld",
513 (long)frames_written);
516 samples_written = frames_written * 2;
517 next_timestamp += samples_written;
522 /* We stop playing for a bit until the buffer re-fills */
523 pthread_mutex_unlock(&lock);
524 if((err = snd_pcm_nonblock(pcm, 0)))
525 fatal(0, "error calling snd_pcm_nonblock: %d", err);
527 if((err = snd_pcm_drop(pcm)))
528 fatal(0, "error calling snd_pcm_drop: %d", err);
531 if((err = snd_pcm_drain(pcm)))
532 fatal(0, "error calling snd_pcm_drain: %d", err);
533 if((err = snd_pcm_nonblock(pcm, 1)))
534 fatal(0, "error calling snd_pcm_nonblock: %d", err);
536 pthread_mutex_lock(&lock);
540 #elif HAVE_COREAUDIO_AUDIOHARDWARE_H
545 AudioStreamBasicDescription asbd;
547 /* If this looks suspiciously like libao's macosx driver there's an
548 * excellent reason for that... */
550 /* TODO report errors as strings not numbers */
551 propertySize = sizeof adid;
552 status = AudioHardwareGetProperty(kAudioHardwarePropertyDefaultOutputDevice,
553 &propertySize, &adid);
555 fatal(0, "AudioHardwareGetProperty: %d", (int)status);
556 if(adid == kAudioDeviceUnknown)
557 fatal(0, "no output device");
558 propertySize = sizeof asbd;
559 status = AudioDeviceGetProperty(adid, 0, false,
560 kAudioDevicePropertyStreamFormat,
561 &propertySize, &asbd);
563 fatal(0, "AudioHardwareGetProperty: %d", (int)status);
564 D(("mSampleRate %f", asbd.mSampleRate));
565 D(("mFormatID %08"PRIx32, asbd.mFormatID));
566 D(("mFormatFlags %08"PRIx32, asbd.mFormatFlags));
567 D(("mBytesPerPacket %08"PRIx32, asbd.mBytesPerPacket));
568 D(("mFramesPerPacket %08"PRIx32, asbd.mFramesPerPacket));
569 D(("mBytesPerFrame %08"PRIx32, asbd.mBytesPerFrame));
570 D(("mChannelsPerFrame %08"PRIx32, asbd.mChannelsPerFrame));
571 D(("mBitsPerChannel %08"PRIx32, asbd.mBitsPerChannel));
572 D(("mReserved %08"PRIx32, asbd.mReserved));
573 if(asbd.mFormatID != kAudioFormatLinearPCM)
574 fatal(0, "audio device does not support kAudioFormatLinearPCM");
575 status = AudioDeviceAddIOProc(adid, adioproc, 0);
577 fatal(0, "AudioDeviceAddIOProc: %d", (int)status);
578 pthread_mutex_lock(&lock);
580 /* Wait for the buffer to fill up a bit */
581 while(nsamples < readahead)
582 pthread_cond_wait(&cond, &lock);
583 /* Start playing now */
584 status = AudioDeviceStart(adid, adioproc);
586 fatal(0, "AudioDeviceStart: %d", (int)status);
587 /* Wait until the buffer empties out */
588 while(nsamples >= minbuffer)
589 pthread_cond_wait(&cond, &lock);
590 /* Stop playing for a bit until the buffer re-fills */
591 status = AudioDeviceStop(adid, adioproc);
593 fatal(0, "AudioDeviceStop: %d", (int)status);
598 # error No known audio API
602 /* display usage message and terminate */
603 static void help(void) {
605 " disorder-playrtp [OPTIONS] ADDRESS [PORT]\n"
607 " --help, -h Display usage message\n"
608 " --version, -V Display version number\n"
609 " --debug, -d Turn on debugging\n"
610 " --device, -D DEVICE Output device\n"
611 " --min, -m FRAMES Buffer low water mark\n"
612 " --buffer, -b FRAMES Buffer high water mark\n");
617 /* display version number and terminate */
618 static void version(void) {
619 xprintf("disorder-playrtp version %s\n", disorder_version_string);
624 int main(int argc, char **argv) {
626 struct addrinfo *res;
627 struct stringlist sl;
630 static const struct addrinfo prefs = {
642 if(!setlocale(LC_CTYPE, "")) fatal(errno, "error calling setlocale");
643 while((n = getopt_long(argc, argv, "hVdD:m:b:", options, 0)) >= 0) {
647 case 'd': debugging = 1; break;
648 case 'D': device = optarg; break;
649 case 'm': minbuffer = 2 * atol(optarg); break;
650 case 'b': readahead = 2 * atol(optarg); break;
651 default: fatal(0, "invalid option");
656 if(argc < 1 || argc > 2)
657 fatal(0, "usage: disorder-playrtp [OPTIONS] ADDRESS [PORT]");
660 /* Listen for inbound audio data */
661 if(!(res = get_address(&sl, &prefs, &sockname)))
663 if((rtpfd = socket(res->ai_family,
665 res->ai_protocol)) < 0)
666 fatal(errno, "error creating socket");
667 if(bind(rtpfd, res->ai_addr, res->ai_addrlen) < 0)
668 fatal(errno, "error binding socket to %s", sockname);