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. */
235 nsamples += p->nsamples;
236 pthread_cond_broadcast(&cond);
237 p = 0; /* we've consumed this packet */
239 pthread_mutex_unlock(&lock);
243 #if HAVE_COREAUDIO_AUDIOHARDWARE_H
244 /** @brief Callback from Core Audio */
245 static OSStatus adioproc(AudioDeviceID inDevice,
246 const AudioTimeStamp *inNow,
247 const AudioBufferList *inInputData,
248 const AudioTimeStamp *inInputTime,
249 AudioBufferList *outOutputData,
250 const AudioTimeStamp *inOutputTime,
251 void *inClientData) {
252 UInt32 nbuffers = outOutputData->mNumberBuffers;
253 AudioBuffer *ab = outOutputData->mBuffers;
254 float *samplesOut; /* where to write samples to */
255 size_t samplesOutLeft; /* space left */
256 size_t samplesInLeft;
257 size_t samplesToCopy;
259 pthread_mutex_lock(&lock);
260 samplesOut = ab->data;
261 samplesOutLeft = ab->mDataByteSize / sizeof (float);
262 while(packets && nbuffers > 0) {
263 if(packets->used == packets->nsamples) {
264 /* TODO if we dropped a packet then we should introduce a gap here */
265 struct packet *const p = packets;
268 pthread_cond_broadcast(&cond);
271 if(samplesOutLeft == 0) {
274 samplesOut = ab->data;
275 samplesOutLeft = ab->mDataByteSize / sizeof (float);
278 /* Now: (1) there is some data left to read
279 * (2) there is some space to put it */
280 samplesInLeft = packets->nsamples - packets->used;
281 samplesToCopy = (samplesInLeft < samplesOutLeft
282 ? samplesInLeft : samplesOutLeft);
283 memcpy(samplesOut, packet->samples + packets->used, samplesToCopy);
284 packets->used += samplesToCopy;
285 samplesOut += samplesToCopy;
286 samesOutLeft -= samplesToCopy;
288 pthread_mutex_unlock(&lock);
293 /** @brief Play an RTP stream
295 * This is the guts of the program. It is responsible for:
296 * - starting the listening thread
297 * - opening the audio device
298 * - reading ahead to build up a buffer
299 * - arranging for audio to be played
300 * - detecting when the buffer has got too small and re-buffering
302 static void play_rtp(void) {
305 /* We receive and convert audio data in a background thread */
306 pthread_create(<id, 0, listen_thread, 0);
310 snd_pcm_hw_params_t *hwparams;
311 snd_pcm_sw_params_t *swparams;
312 /* Only support one format for now */
313 const int sample_format = SND_PCM_FORMAT_S16_BE;
314 unsigned rate = 44100;
315 const int channels = 2;
316 const int samplesize = channels * sizeof(uint16_t);
317 snd_pcm_uframes_t pcm_bufsize = MAXSAMPLES * samplesize * 3;
318 /* If we can write more than this many samples we'll get a wakeup */
319 const int avail_min = 256;
320 snd_pcm_sframes_t frames_written;
321 size_t samples_written;
324 int infilling = 0, escape = 0;
326 uint32_t packet_start, packet_end;
329 if((err = snd_pcm_open(&pcm,
330 device ? device : "default",
331 SND_PCM_STREAM_PLAYBACK,
333 fatal(0, "error from snd_pcm_open: %d", err);
334 /* Set up 'hardware' parameters */
335 snd_pcm_hw_params_alloca(&hwparams);
336 if((err = snd_pcm_hw_params_any(pcm, hwparams)) < 0)
337 fatal(0, "error from snd_pcm_hw_params_any: %d", err);
338 if((err = snd_pcm_hw_params_set_access(pcm, hwparams,
339 SND_PCM_ACCESS_RW_INTERLEAVED)) < 0)
340 fatal(0, "error from snd_pcm_hw_params_set_access: %d", err);
341 if((err = snd_pcm_hw_params_set_format(pcm, hwparams,
343 fatal(0, "error from snd_pcm_hw_params_set_format (%d): %d",
345 if((err = snd_pcm_hw_params_set_rate_near(pcm, hwparams, &rate, 0)) < 0)
346 fatal(0, "error from snd_pcm_hw_params_set_rate (%d): %d",
348 if((err = snd_pcm_hw_params_set_channels(pcm, hwparams,
350 fatal(0, "error from snd_pcm_hw_params_set_channels (%d): %d",
352 if((err = snd_pcm_hw_params_set_buffer_size_near(pcm, hwparams,
354 fatal(0, "error from snd_pcm_hw_params_set_buffer_size (%d): %d",
355 MAXSAMPLES * samplesize * 3, err);
356 if((err = snd_pcm_hw_params(pcm, hwparams)) < 0)
357 fatal(0, "error calling snd_pcm_hw_params: %d", err);
358 /* Set up 'software' parameters */
359 snd_pcm_sw_params_alloca(&swparams);
360 if((err = snd_pcm_sw_params_current(pcm, swparams)) < 0)
361 fatal(0, "error calling snd_pcm_sw_params_current: %d", err);
362 if((err = snd_pcm_sw_params_set_avail_min(pcm, swparams, avail_min)) < 0)
363 fatal(0, "error calling snd_pcm_sw_params_set_avail_min %d: %d",
365 if((err = snd_pcm_sw_params(pcm, swparams)) < 0)
366 fatal(0, "error calling snd_pcm_sw_params: %d", err);
371 pthread_mutex_lock(&lock);
373 /* Wait for the buffer to fill up a bit */
375 info("%lu samples in buffer (%lus)", nsamples,
376 nsamples / (44100 * 2));
377 info("Buffering...");
378 while(nsamples < readahead)
379 pthread_cond_wait(&cond, &lock);
381 if((err = snd_pcm_prepare(pcm)))
382 fatal(0, "error calling snd_pcm_prepare: %d", err);
385 /* Start at the first available packet */
386 next_timestamp = packets->timestamp;
391 info("%lu samples in buffer (%lus)", nsamples,
392 nsamples / (44100 * 2));
394 /* Wait until the buffer empties out */
395 while(nsamples >= minbuffer && !escape) {
397 if(now > logged + 10) {
399 info("%lu samples in buffer (%lus)", nsamples,
400 nsamples / (44100 * 2));
403 && ge(next_timestamp, packets->timestamp + packets->nsamples)) {
404 struct packet *p = packets;
406 info("dropping buffered past packet %"PRIx32" < %"PRIx32,
407 packets->timestamp, next_timestamp);
411 assert(lt(p->timestamp, packets->timestamp));
412 nsamples -= p->nsamples;
414 pthread_cond_broadcast(&cond);
417 /* Wait for ALSA to ask us for more data */
418 pthread_mutex_unlock(&lock);
419 snd_pcm_wait(pcm, -1);
420 pthread_mutex_lock(&lock);
421 /* ALSA is ready for more data */
422 packet_start = packets->timestamp;
423 packet_end = packets->timestamp + packets->nsamples;
424 if(ge(next_timestamp, packet_start)
425 && lt(next_timestamp, packet_end)) {
426 /* The target timestamp is somewhere in this packet */
427 const uint32_t offset = next_timestamp - packets->timestamp;
428 const uint32_t samples_available = (packets->timestamp + packets->nsamples) - next_timestamp;
429 const size_t frames_available = samples_available / 2;
431 frames_written = snd_pcm_writei(pcm,
432 packets->samples_raw + offset,
434 if(frames_written < 0) {
435 switch(frames_written) {
437 info("snd_pcm_wait() returned but we got -EAGAIN!");
440 error(0, "error calling snd_pcm_writei: %ld",
441 (long)frames_written);
445 fatal(0, "error calling snd_pcm_writei: %ld",
446 (long)frames_written);
449 samples_written = frames_written * 2;
450 next_timestamp += samples_written;
451 if(ge(next_timestamp, packet_end)) {
452 /* We're done with this packet */
453 struct packet *p = packets;
457 assert(lt(p->timestamp, packets->timestamp));
458 nsamples -= p->nsamples;
460 pthread_cond_broadcast(&cond);
465 /* We don't have anything to play! We'd better play some 0s. */
466 static const uint16_t zeros[INFILL_SAMPLES];
467 size_t samples_available = INFILL_SAMPLES, frames_available;
469 /* If the maximum infill would take us past the start of the next
470 * packet then we truncate the infill to the right amount. */
471 if(lt(packets->timestamp,
472 next_timestamp + samples_available))
473 samples_available = packets->timestamp - next_timestamp;
474 if((int)samples_available < 0) {
475 info("packets->timestamp: %"PRIx32" next_timestamp: %"PRIx32" next+max: %"PRIx32" available: %"PRIx32,
476 packets->timestamp, next_timestamp,
477 next_timestamp + INFILL_SAMPLES, samples_available);
479 frames_available = samples_available / 2;
481 info("Infilling %d samples, next=%"PRIx32" packet=[%"PRIx32",%"PRIx32"]",
482 samples_available, next_timestamp,
483 packets->timestamp, packets->timestamp + packets->nsamples);
486 frames_written = snd_pcm_writei(pcm,
489 if(frames_written < 0) {
490 switch(frames_written) {
492 info("snd_pcm_wait() returned but we got -EAGAIN!");
495 error(0, "error calling snd_pcm_writei: %ld",
496 (long)frames_written);
500 fatal(0, "error calling snd_pcm_writei: %ld",
501 (long)frames_written);
504 samples_written = frames_written * 2;
505 next_timestamp += samples_written;
510 /* We stop playing for a bit until the buffer re-fills */
511 pthread_mutex_unlock(&lock);
512 if((err = snd_pcm_nonblock(pcm, 0)))
513 fatal(0, "error calling snd_pcm_nonblock: %d", err);
515 if((err = snd_pcm_drop(pcm)))
516 fatal(0, "error calling snd_pcm_drop: %d", err);
519 if((err = snd_pcm_drain(pcm)))
520 fatal(0, "error calling snd_pcm_drain: %d", err);
521 if((err = snd_pcm_nonblock(pcm, 1)))
522 fatal(0, "error calling snd_pcm_nonblock: %d", err);
524 pthread_mutex_lock(&lock);
528 #elif HAVE_COREAUDIO_AUDIOHARDWARE_H
533 AudioStreamBasicDescription asbd;
535 /* If this looks suspiciously like libao's macosx driver there's an
536 * excellent reason for that... */
538 /* TODO report errors as strings not numbers */
539 propertySize = sizeof adid;
540 status = AudioHardwareGetProperty(kAudioHardwarePropertyDefaultOutputDevice,
541 &propertySize, &adid);
543 fatal(0, "AudioHardwareGetProperty: %d", (int)status);
544 if(adid == kAudioDeviceUnknown)
545 fatal(0, "no output device");
546 propertySize = sizeof asbd;
547 status = AudioDeviceGetProperty(adid, 0, false,
548 kAudioDevicePropertyStreamFormat,
549 &propertySize, &asbd);
551 fatal(0, "AudioHardwareGetProperty: %d", (int)status);
552 D(("mSampleRate %f", asbd.mSampleRate));
553 D(("mFormatID %08"PRIx32, asbd.mFormatID));
554 D(("mFormatFlags %08"PRIx32, asbd.mFormatFlags));
555 D(("mBytesPerPacket %08"PRIx32, asbd.mBytesPerPacket));
556 D(("mFramesPerPacket %08"PRIx32, asbd.mFramesPerPacket));
557 D(("mBytesPerFrame %08"PRIx32, asbd.mBytesPerFrame));
558 D(("mChannelsPerFrame %08"PRIx32, asbd.mChannelsPerFrame));
559 D(("mBitsPerChannel %08"PRIx32, asbd.mBitsPerChannel));
560 D(("mReserved %08"PRIx32, asbd.mReserved));
561 if(asbd.mFormatID != kAudioFormatLinearPCM)
562 fatal(0, "audio device does not support kAudioFormatLinearPCM");
563 status = AudioDeviceAddIOProc(adid, adioproc, 0);
565 fatal(0, "AudioDeviceAddIOProc: %d", (int)status);
566 pthread_mutex_lock(&lock);
568 /* Wait for the buffer to fill up a bit */
569 while(nsamples < readahead)
570 pthread_cond_wait(&cond, &lock);
571 /* Start playing now */
572 status = AudioDeviceStart(adid, adioproc);
574 fatal(0, "AudioDeviceStart: %d", (int)status);
575 /* Wait until the buffer empties out */
576 while(nsamples >= minbuffer)
577 pthread_cond_wait(&cond, &lock);
578 /* Stop playing for a bit until the buffer re-fills */
579 status = AudioDeviceStop(adid, adioproc);
581 fatal(0, "AudioDeviceStop: %d", (int)status);
586 # error No known audio API
590 /* display usage message and terminate */
591 static void help(void) {
593 " disorder-playrtp [OPTIONS] ADDRESS [PORT]\n"
595 " --help, -h Display usage message\n"
596 " --version, -V Display version number\n"
597 " --debug, -d Turn on debugging\n"
598 " --device, -D DEVICE Output device\n"
599 " --min, -m FRAMES Buffer low water mark\n"
600 " --buffer, -b FRAMES Buffer high water mark\n");
605 /* display version number and terminate */
606 static void version(void) {
607 xprintf("disorder-playrtp version %s\n", disorder_version_string);
612 int main(int argc, char **argv) {
614 struct addrinfo *res;
615 struct stringlist sl;
618 static const struct addrinfo prefs = {
630 if(!setlocale(LC_CTYPE, "")) fatal(errno, "error calling setlocale");
631 while((n = getopt_long(argc, argv, "hVdD:m:b:", options, 0)) >= 0) {
635 case 'd': debugging = 1; break;
636 case 'D': device = optarg; break;
637 case 'm': minbuffer = 2 * atol(optarg); break;
638 case 'b': readahead = 2 * atol(optarg); break;
639 default: fatal(0, "invalid option");
644 if(argc < 1 || argc > 2)
645 fatal(0, "usage: disorder-playrtp [OPTIONS] ADDRESS [PORT]");
648 /* Listen for inbound audio data */
649 if(!(res = get_address(&sl, &prefs, &sockname)))
651 if((rtpfd = socket(res->ai_family,
653 res->ai_protocol)) < 0)
654 fatal(errno, "error creating socket");
655 if(bind(rtpfd, res->ai_addr, res->ai_addrlen) < 0)
656 fatal(errno, "error binding socket to %s", sockname);