2 * This file is part of DisOrder
3 * Copyright (C) 2005, 2006, 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 server/speaker.c
21 * @brief Speaker processs
23 * This program is responsible for transmitting a single coherent audio stream
24 * to its destination (over the network, to some sound API, to some
25 * subprocess). It receives connections from decoders via file descriptor
26 * passing from the main server and plays them in the right order.
28 * For the <a href="http://www.alsa-project.org/">ALSA</a> API, 8- and 16- bit
29 * stereo and mono are supported, with any sample rate (within the limits that
30 * ALSA can deal with.)
32 * When communicating with a subprocess, <a
33 * href="http://sox.sourceforge.net/">sox</a> is invoked to convert the inbound
34 * data to a single consistent format. The same applies for network (RTP)
35 * play, though in that case currently only 44.1KHz 16-bit stereo is supported.
37 * The inbound data starts with a structure defining the data format. Note
38 * that this is NOT portable between different platforms or even necessarily
39 * between versions; the speaker is assumed to be built from the same source
40 * and run on the same host as the main server.
42 * This program deliberately does not use the garbage collector even though it
43 * might be convenient to do so. This is for two reasons. Firstly some sound
44 * APIs use thread threads and we do not want to have to deal with potential
45 * interactions between threading and garbage collection. Secondly this
46 * process needs to be able to respond quickly and this is not compatible with
47 * the collector hanging the program even relatively briefly.
63 #include <sys/select.h>
68 #include <sys/socket.h>
73 #include "configuration.h"
85 #include <alsa/asoundlib.h>
88 #ifdef WORDS_BIGENDIAN
89 # define MACHINE_AO_FMT AO_FMT_BIG
91 # define MACHINE_AO_FMT AO_FMT_LITTLE
94 /** @brief How many seconds of input to buffer
96 * While any given connection has this much audio buffered, no more reads will
97 * be issued for that connection. The decoder will have to wait.
99 #define BUFFER_SECONDS 5
101 #define FRAMES 4096 /* Frame batch size */
103 /** @brief Bytes to send per network packet
105 * Don't make this too big or arithmetic will start to overflow.
107 #define NETWORK_BYTES (1024+sizeof(struct rtp_header))
109 /** @brief Maximum RTP playahead (ms) */
110 #define RTP_AHEAD_MS 1000
112 /** @brief Maximum number of FDs to poll for */
115 /** @brief Track structure
117 * Known tracks are kept in a linked list. Usually there will be at most two
118 * of these but rearranging the queue can cause there to be more.
120 static struct track
{
121 struct track
*next
; /* next track */
122 int fd
; /* input FD */
123 char id
[24]; /* ID */
124 size_t start
, used
; /* start + bytes used */
125 int eof
; /* input is at EOF */
126 int got_format
; /* got format yet? */
127 ao_sample_format format
; /* sample format */
128 unsigned long long played
; /* number of frames played */
129 char *buffer
; /* sample buffer */
130 size_t size
; /* sample buffer size */
131 int slot
; /* poll array slot */
132 } *tracks
, *playing
; /* all tracks + playing track */
134 static time_t last_report
; /* when we last reported */
135 static int paused
; /* pause status */
136 static size_t bpf
; /* bytes per frame */
137 static struct pollfd fds
[NFDS
]; /* if we need more than that */
138 static int fdno
; /* fd number */
139 static size_t bufsize
; /* buffer size */
141 /** @brief The current PCM handle */
142 static snd_pcm_t
*pcm
;
143 static snd_pcm_uframes_t last_pcm_bufsize
; /* last seen buffer size */
144 static ao_sample_format pcm_format
; /* current format if aodev != 0 */
147 /** @brief Ready to send audio
149 * This is set when the destination is ready to receive audio. Generally
150 * this implies that the sound device is open. In the ALSA backend it
151 * does @b not necessarily imply that is has the right sample format.
155 static int forceplay
; /* frames to force play */
156 static int cmdfd
= -1; /* child process input */
157 static int bfd
= -1; /* broadcast FD */
159 /** @brief RTP timestamp
161 * This counts the number of samples played (NB not the number of frames
164 * The timestamp in the packet header is only 32 bits wide. With 44100Hz
165 * stereo, that only gives about half a day before wrapping, which is not
166 * particularly convenient for certain debugging purposes. Therefore the
167 * timestamp is maintained as a 64-bit integer, giving around six million years
168 * before wrapping, and truncated to 32 bits when transmitting.
170 static uint64_t rtp_time
;
172 /** @brief RTP base timestamp
174 * This is the real time correspoding to an @ref rtp_time of 0. It is used
175 * to recalculate the timestamp after idle periods.
177 static struct timeval rtp_time_0
;
179 static uint16_t rtp_seq
; /* frame sequence number */
180 static uint32_t rtp_id
; /* RTP SSRC */
181 static int idled
; /* set when idled */
182 static int audio_errors
; /* audio error counter */
184 /** @brief Structure of a backend */
185 struct speaker_backend
{
186 /** @brief Which backend this is
188 * @c -1 terminates the list.
195 * - @ref FIXED_FORMAT
198 /** @brief Lock to configured sample format */
199 #define FIXED_FORMAT 0x0001
201 /** @brief Initialization
203 * Called once at startup. This is responsible for one-time setup
204 * operations, for instance opening a network socket to transmit to.
206 * When writing to a native sound API this might @b not imply opening the
207 * native sound device - that might be done by @c activate below.
211 /** @brief Activation
212 * @return 0 on success, non-0 on error
214 * Called to activate the output device.
216 * After this function succeeds, @ref ready should be non-0. As well as
217 * opening the audio device, this function is responsible for reconfiguring
218 * if it necessary to cope with different samples formats (for backends that
219 * don't demand a single fixed sample format for the lifetime of the server).
221 int (*activate
)(void);
223 /** @brief Play sound
224 * @param frames Number of frames to play
225 * @return Number of frames actually played
227 size_t (*play
)(size_t frames
);
229 /** @brief Deactivation
231 * Called to deactivate the sound device. This is the inverse of
234 void (*deactivate
)(void);
237 /** @brief Selected backend */
238 static const struct speaker_backend
*backend
;
240 static const struct option options
[] = {
241 { "help", no_argument
, 0, 'h' },
242 { "version", no_argument
, 0, 'V' },
243 { "config", required_argument
, 0, 'c' },
244 { "debug", no_argument
, 0, 'd' },
245 { "no-debug", no_argument
, 0, 'D' },
249 /* Display usage message and terminate. */
250 static void help(void) {
252 " disorder-speaker [OPTIONS]\n"
254 " --help, -h Display usage message\n"
255 " --version, -V Display version number\n"
256 " --config PATH, -c PATH Set configuration file\n"
257 " --debug, -d Turn on debugging\n"
259 "Speaker process for DisOrder. Not intended to be run\n"
265 /* Display version number and terminate. */
266 static void version(void) {
267 xprintf("disorder-speaker version %s\n", disorder_version_string
);
272 /** @brief Return the number of bytes per frame in @p format */
273 static size_t bytes_per_frame(const ao_sample_format
*format
) {
274 return format
->channels
* format
->bits
/ 8;
277 /** @brief Find track @p id, maybe creating it if not found */
278 static struct track
*findtrack(const char *id
, int create
) {
281 D(("findtrack %s %d", id
, create
));
282 for(t
= tracks
; t
&& strcmp(id
, t
->id
); t
= t
->next
)
285 t
= xmalloc(sizeof *t
);
290 /* The initial input buffer will be the sample format. */
291 t
->buffer
= (void *)&t
->format
;
292 t
->size
= sizeof t
->format
;
297 /** @brief Remove track @p id (but do not destroy it) */
298 static struct track
*removetrack(const char *id
) {
299 struct track
*t
, **tt
;
301 D(("removetrack %s", id
));
302 for(tt
= &tracks
; (t
= *tt
) && strcmp(id
, t
->id
); tt
= &t
->next
)
309 /** @brief Destroy a track */
310 static void destroy(struct track
*t
) {
311 D(("destroy %s", t
->id
));
312 if(t
->fd
!= -1) xclose(t
->fd
);
313 if(t
->buffer
!= (void *)&t
->format
) free(t
->buffer
);
317 /** @brief Notice a new connection */
318 static void acquire(struct track
*t
, int fd
) {
319 D(("acquire %s %d", t
->id
, fd
));
326 /** @brief Return true if A and B denote identical libao formats, else false */
327 static int formats_equal(const ao_sample_format
*a
,
328 const ao_sample_format
*b
) {
329 return (a
->bits
== b
->bits
330 && a
->rate
== b
->rate
331 && a
->channels
== b
->channels
332 && a
->byte_format
== b
->byte_format
);
335 /** @brief Compute arguments to sox */
336 static void soxargs(const char ***pp
, char **qq
, ao_sample_format
*ao
) {
341 *(*pp
)++ = *qq
; n
= sprintf(*qq
, "-r%d", ao
->rate
); *qq
+= n
+ 1;
342 *(*pp
)++ = *qq
; n
= sprintf(*qq
, "-c%d", ao
->channels
); *qq
+= n
+ 1;
343 /* sox 12.17.9 insists on -b etc; CVS sox insists on -<n> etc; both are
345 switch(config
->sox_generation
) {
348 && ao
->byte_format
!= AO_FMT_NATIVE
349 && ao
->byte_format
!= MACHINE_AO_FMT
) {
353 case 8: *(*pp
)++ = "-b"; break;
354 case 16: *(*pp
)++ = "-w"; break;
355 case 32: *(*pp
)++ = "-l"; break;
356 case 64: *(*pp
)++ = "-d"; break;
357 default: fatal(0, "cannot handle sample size %d", (int)ao
->bits
);
361 switch(ao
->byte_format
) {
362 case AO_FMT_NATIVE
: break;
363 case AO_FMT_BIG
: *(*pp
)++ = "-B"; break;
364 case AO_FMT_LITTLE
: *(*pp
)++ = "-L"; break;
366 *(*pp
)++ = *qq
; n
= sprintf(*qq
, "-%d", ao
->bits
/8); *qq
+= n
+ 1;
371 /** @brief Enable format translation
373 * If necessary, replaces a tracks inbound file descriptor with one connected
374 * to a sox invocation, which performs the required translation.
376 static void enable_translation(struct track
*t
) {
377 if((backend
->flags
& FIXED_FORMAT
)
378 && !formats_equal(&t
->format
, &config
->sample_format
)) {
379 char argbuf
[1024], *q
= argbuf
;
380 const char *av
[18], **pp
= av
;
385 soxargs(&pp
, &q
, &t
->format
);
387 soxargs(&pp
, &q
, &config
->sample_format
);
391 for(pp
= av
; *pp
; pp
++)
392 D(("sox arg[%d] = %s", pp
- av
, *pp
));
398 signal(SIGPIPE
, SIG_DFL
);
400 xdup2(soxpipe
[1], 1);
401 fcntl(0, F_SETFL
, fcntl(0, F_GETFL
) & ~O_NONBLOCK
);
405 execvp("sox", (char **)av
);
408 D(("forking sox for format conversion (kid = %d)", soxkid
));
412 t
->format
= config
->sample_format
;
416 /** @brief Read data into a sample buffer
417 * @param t Pointer to track
418 * @return 0 on success, -1 on EOF
420 * This is effectively the read callback on @c t->fd.
422 static int fill(struct track
*t
) {
426 D(("fill %s: eof=%d used=%zu size=%zu got_format=%d",
427 t
->id
, t
->eof
, t
->used
, t
->size
, t
->got_format
));
428 if(t
->eof
) return -1;
429 if(t
->used
< t
->size
) {
430 /* there is room left in the buffer */
431 where
= (t
->start
+ t
->used
) % t
->size
;
433 /* We are reading audio data, get as much as we can */
434 if(where
>= t
->start
) left
= t
->size
- where
;
435 else left
= t
->start
- where
;
437 /* We are still waiting for the format, only get that */
438 left
= sizeof (ao_sample_format
) - t
->used
;
440 n
= read(t
->fd
, t
->buffer
+ where
, left
);
441 } while(n
< 0 && errno
== EINTR
);
443 if(errno
!= EAGAIN
) fatal(errno
, "error reading sample stream");
447 D(("fill %s: eof detected", t
->id
));
452 if(!t
->got_format
&& t
->used
>= sizeof (ao_sample_format
)) {
453 assert(t
->used
== sizeof (ao_sample_format
));
454 /* Check that our assumptions are met. */
455 if(t
->format
.bits
& 7)
456 fatal(0, "bits per sample not a multiple of 8");
457 /* If the input format is unsuitable, arrange to translate it */
458 enable_translation(t
);
459 /* Make a new buffer for audio data. */
460 t
->size
= bytes_per_frame(&t
->format
) * t
->format
.rate
* BUFFER_SECONDS
;
461 t
->buffer
= xmalloc(t
->size
);
464 D(("got format for %s", t
->id
));
470 /** @brief Close the sound device */
471 static void idle(void) {
473 if(backend
->deactivate
)
474 backend
->deactivate();
479 /** @brief Abandon the current track */
480 static void abandon(void) {
481 struct speaker_message sm
;
484 memset(&sm
, 0, sizeof sm
);
485 sm
.type
= SM_FINISHED
;
486 strcpy(sm
.id
, playing
->id
);
487 speaker_send(1, &sm
, 0);
488 removetrack(playing
->id
);
495 /** @brief Log ALSA parameters */
496 static void log_params(snd_pcm_hw_params_t
*hwparams
,
497 snd_pcm_sw_params_t
*swparams
) {
501 return; /* too verbose */
506 snd_pcm_sw_params_get_silence_size(swparams
, &f
);
507 info("sw silence_size=%lu", (unsigned long)f
);
508 snd_pcm_sw_params_get_silence_threshold(swparams
, &f
);
509 info("sw silence_threshold=%lu", (unsigned long)f
);
510 snd_pcm_sw_params_get_sleep_min(swparams
, &u
);
511 info("sw sleep_min=%lu", (unsigned long)u
);
512 snd_pcm_sw_params_get_start_threshold(swparams
, &f
);
513 info("sw start_threshold=%lu", (unsigned long)f
);
514 snd_pcm_sw_params_get_stop_threshold(swparams
, &f
);
515 info("sw stop_threshold=%lu", (unsigned long)f
);
516 snd_pcm_sw_params_get_xfer_align(swparams
, &f
);
517 info("sw xfer_align=%lu", (unsigned long)f
);
522 /** @brief Enable sound output
524 * Makes sure the sound device is open and has the right sample format. Return
525 * 0 on success and -1 on error.
527 static int activate(void) {
528 /* If we don't know the format yet we cannot start. */
529 if(!playing
->got_format
) {
530 D((" - not got format for %s", playing
->id
));
533 return backend
->activate();
536 /* Check to see whether the current track has finished playing */
537 static void maybe_finished(void) {
540 && (!playing
->got_format
541 || playing
->used
< bytes_per_frame(&playing
->format
)))
545 static void fork_cmd(void) {
548 if(cmdfd
!= -1) close(cmdfd
);
552 signal(SIGPIPE
, SIG_DFL
);
556 execl("/bin/sh", "sh", "-c", config
->speaker_command
, (char *)0);
557 fatal(errno
, "error execing /bin/sh");
561 D(("forked cmd %d, fd = %d", cmdpid
, cmdfd
));
564 static void play(size_t frames
) {
565 size_t avail_frames
, avail_bytes
, written_frames
;
566 ssize_t written_bytes
;
568 /* Make sure the output device is activated */
573 forceplay
= 0; /* Must have called abandon() */
576 D(("play: play %zu/%zu%s %dHz %db %dc", frames
, playing
->used
/ bpf
,
577 playing
->eof ?
" EOF" : "",
578 playing
->format
.rate
,
579 playing
->format
.bits
,
580 playing
->format
.channels
));
581 /* If we haven't got enough bytes yet wait until we have. Exception: when
583 if(playing
->used
< frames
* bpf
&& !playing
->eof
) {
587 /* We have got enough data so don't force play again */
589 /* Figure out how many frames there are available to write */
590 if(playing
->start
+ playing
->used
> playing
->size
)
591 /* The ring buffer is currently wrapped, only play up to the wrap point */
592 avail_bytes
= playing
->size
- playing
->start
;
594 /* The ring buffer is not wrapped, can play the lot */
595 avail_bytes
= playing
->used
;
596 avail_frames
= avail_bytes
/ bpf
;
597 /* Only play up to the requested amount */
598 if(avail_frames
> frames
)
599 avail_frames
= frames
;
603 written_frames
= backend
->play(avail_frames
);
604 written_bytes
= written_frames
* bpf
;
605 /* written_bytes and written_frames had better both be set and correct by
607 playing
->start
+= written_bytes
;
608 playing
->used
-= written_bytes
;
609 playing
->played
+= written_frames
;
610 /* If the pointer is at the end of the buffer (or the buffer is completely
611 * empty) wrap it back to the start. */
612 if(!playing
->used
|| playing
->start
== playing
->size
)
614 frames
-= written_frames
;
617 /* Notify the server what we're up to. */
618 static void report(void) {
619 struct speaker_message sm
;
621 if(playing
&& playing
->buffer
!= (void *)&playing
->format
) {
622 memset(&sm
, 0, sizeof sm
);
623 sm
.type
= paused ? SM_PAUSED
: SM_PLAYING
;
624 strcpy(sm
.id
, playing
->id
);
625 sm
.data
= playing
->played
/ playing
->format
.rate
;
626 speaker_send(1, &sm
, 0);
631 static void reap(int __attribute__((unused
)) sig
) {
636 cmdpid
= waitpid(-1, &st
, WNOHANG
);
638 signal(SIGCHLD
, reap
);
641 static int addfd(int fd
, int events
) {
644 fds
[fdno
].events
= events
;
651 /** @brief ALSA backend initialization */
652 static void alsa_init(void) {
653 info("selected ALSA backend");
656 /** @brief ALSA backend activation */
657 static int alsa_activate(void) {
658 /* If we need to change format then close the current device. */
659 if(pcm
&& !formats_equal(&playing
->format
, &pcm_format
))
662 snd_pcm_hw_params_t
*hwparams
;
663 snd_pcm_sw_params_t
*swparams
;
664 snd_pcm_uframes_t pcm_bufsize
;
666 int sample_format
= 0;
670 if((err
= snd_pcm_open(&pcm
,
672 SND_PCM_STREAM_PLAYBACK
,
673 SND_PCM_NONBLOCK
))) {
674 error(0, "error from snd_pcm_open: %d", err
);
677 snd_pcm_hw_params_alloca(&hwparams
);
678 D(("set up hw params"));
679 if((err
= snd_pcm_hw_params_any(pcm
, hwparams
)) < 0)
680 fatal(0, "error from snd_pcm_hw_params_any: %d", err
);
681 if((err
= snd_pcm_hw_params_set_access(pcm
, hwparams
,
682 SND_PCM_ACCESS_RW_INTERLEAVED
)) < 0)
683 fatal(0, "error from snd_pcm_hw_params_set_access: %d", err
);
684 switch(playing
->format
.bits
) {
686 sample_format
= SND_PCM_FORMAT_S8
;
689 switch(playing
->format
.byte_format
) {
690 case AO_FMT_NATIVE
: sample_format
= SND_PCM_FORMAT_S16
; break;
691 case AO_FMT_LITTLE
: sample_format
= SND_PCM_FORMAT_S16_LE
; break;
692 case AO_FMT_BIG
: sample_format
= SND_PCM_FORMAT_S16_BE
; break;
693 error(0, "unrecognized byte format %d", playing
->format
.byte_format
);
698 error(0, "unsupported sample size %d", playing
->format
.bits
);
701 if((err
= snd_pcm_hw_params_set_format(pcm
, hwparams
,
702 sample_format
)) < 0) {
703 error(0, "error from snd_pcm_hw_params_set_format (%d): %d",
707 rate
= playing
->format
.rate
;
708 if((err
= snd_pcm_hw_params_set_rate_near(pcm
, hwparams
, &rate
, 0)) < 0) {
709 error(0, "error from snd_pcm_hw_params_set_rate (%d): %d",
710 playing
->format
.rate
, err
);
713 if(rate
!= (unsigned)playing
->format
.rate
)
714 info("want rate %d, got %u", playing
->format
.rate
, rate
);
715 if((err
= snd_pcm_hw_params_set_channels(pcm
, hwparams
,
716 playing
->format
.channels
)) < 0) {
717 error(0, "error from snd_pcm_hw_params_set_channels (%d): %d",
718 playing
->format
.channels
, err
);
721 bufsize
= 3 * FRAMES
;
722 pcm_bufsize
= bufsize
;
723 if((err
= snd_pcm_hw_params_set_buffer_size_near(pcm
, hwparams
,
725 fatal(0, "error from snd_pcm_hw_params_set_buffer_size (%d): %d",
727 if(pcm_bufsize
!= 3 * FRAMES
&& pcm_bufsize
!= last_pcm_bufsize
)
728 info("asked for PCM buffer of %d frames, got %d",
729 3 * FRAMES
, (int)pcm_bufsize
);
730 last_pcm_bufsize
= pcm_bufsize
;
731 if((err
= snd_pcm_hw_params(pcm
, hwparams
)) < 0)
732 fatal(0, "error calling snd_pcm_hw_params: %d", err
);
733 D(("set up sw params"));
734 snd_pcm_sw_params_alloca(&swparams
);
735 if((err
= snd_pcm_sw_params_current(pcm
, swparams
)) < 0)
736 fatal(0, "error calling snd_pcm_sw_params_current: %d", err
);
737 if((err
= snd_pcm_sw_params_set_avail_min(pcm
, swparams
, FRAMES
)) < 0)
738 fatal(0, "error calling snd_pcm_sw_params_set_avail_min %d: %d",
740 if((err
= snd_pcm_sw_params(pcm
, swparams
)) < 0)
741 fatal(0, "error calling snd_pcm_sw_params: %d", err
);
742 pcm_format
= playing
->format
;
743 bpf
= bytes_per_frame(&pcm_format
);
744 D(("acquired audio device"));
745 log_params(hwparams
, swparams
);
752 /* We assume the error is temporary and that we'll retry in a bit. */
760 /** @brief Play via ALSA */
761 static size_t alsa_play(size_t frames
) {
762 snd_pcm_sframes_t pcm_written_frames
;
765 pcm_written_frames
= snd_pcm_writei(pcm
,
766 playing
->buffer
+ playing
->start
,
768 D(("actually play %zu frames, wrote %d",
769 frames
, (int)pcm_written_frames
));
770 if(pcm_written_frames
< 0) {
771 switch(pcm_written_frames
) {
772 case -EPIPE
: /* underrun */
773 error(0, "snd_pcm_writei reports underrun");
774 if((err
= snd_pcm_prepare(pcm
)) < 0)
775 fatal(0, "error calling snd_pcm_prepare: %d", err
);
780 fatal(0, "error calling snd_pcm_writei: %d",
781 (int)pcm_written_frames
);
784 return pcm_written_frames
;
787 /** @brief ALSA deactivation */
788 static void alsa_deactivate(void) {
792 if((err
= snd_pcm_nonblock(pcm
, 0)) < 0)
793 fatal(0, "error calling snd_pcm_nonblock: %d", err
);
800 D(("released audio device"));
805 /** @brief Command backend initialization */
806 static void command_init(void) {
807 info("selected command backend");
811 /** @brief Play to a subprocess */
812 static size_t command_play(size_t frames
) {
813 size_t bytes
= frames
* bpf
;
816 written_bytes
= write(cmdfd
, playing
->buffer
+ playing
->start
, bytes
);
817 D(("actually play %zu bytes, wrote %d",
818 bytes
, written_bytes
));
819 if(written_bytes
< 0) {
822 error(0, "hmm, command died; trying another");
828 fatal(errno
, "error writing to subprocess");
831 return written_bytes
/ bpf
;
834 /** @brief Command/network backend activation */
835 static int generic_activate(void) {
837 bufsize
= 3 * FRAMES
;
838 bpf
= bytes_per_frame(&config
->sample_format
);
839 D(("acquired audio device"));
845 /** @brief Network backend initialization */
846 static void network_init(void) {
847 struct addrinfo
*res
, *sres
;
848 static const struct addrinfo pref
= {
858 static const struct addrinfo prefbind
= {
868 static const int one
= 1;
869 int sndbuf
, target_sndbuf
= 131072;
871 char *sockname
, *ssockname
;
873 res
= get_address(&config
->broadcast
, &pref
, &sockname
);
875 if(config
->broadcast_from
.n
) {
876 sres
= get_address(&config
->broadcast_from
, &prefbind
, &ssockname
);
880 if((bfd
= socket(res
->ai_family
,
882 res
->ai_protocol
)) < 0)
883 fatal(errno
, "error creating broadcast socket");
884 if(setsockopt(bfd
, SOL_SOCKET
, SO_BROADCAST
, &one
, sizeof one
) < 0)
885 fatal(errno
, "error setting SO_BROADCAST on broadcast socket");
887 if(getsockopt(bfd
, SOL_SOCKET
, SO_SNDBUF
,
889 fatal(errno
, "error getting SO_SNDBUF");
890 if(target_sndbuf
> sndbuf
) {
891 if(setsockopt(bfd
, SOL_SOCKET
, SO_SNDBUF
,
892 &target_sndbuf
, sizeof target_sndbuf
) < 0)
893 error(errno
, "error setting SO_SNDBUF to %d", target_sndbuf
);
895 info("changed socket send buffer size from %d to %d",
896 sndbuf
, target_sndbuf
);
898 info("default socket send buffer is %d",
900 /* We might well want to set additional broadcast- or multicast-related
902 if(sres
&& bind(bfd
, sres
->ai_addr
, sres
->ai_addrlen
) < 0)
903 fatal(errno
, "error binding broadcast socket to %s", ssockname
);
904 if(connect(bfd
, res
->ai_addr
, res
->ai_addrlen
) < 0)
905 fatal(errno
, "error connecting broadcast socket to %s", sockname
);
907 gcry_randomize(&rtp_id
, sizeof rtp_id
, GCRY_STRONG_RANDOM
);
908 info("selected network backend, sending to %s", sockname
);
909 if(config
->sample_format
.byte_format
!= AO_FMT_BIG
) {
910 info("forcing big-endian sample format");
911 config
->sample_format
.byte_format
= AO_FMT_BIG
;
915 /** @brief Play over the network */
916 static size_t network_play(size_t frames
) {
917 struct rtp_header header
;
919 size_t bytes
= frames
* bpf
, written_frames
;
921 /* We transmit using RTP (RFC3550) and attempt to conform to the internet
922 * AVT profile (RFC3551). */
925 /* There may have been a gap. Fix up the RTP time accordingly. */
928 uint64_t target_rtp_time
;
930 /* Find the current time */
931 xgettimeofday(&now
, 0);
932 /* Find the number of microseconds elapsed since rtp_time=0 */
933 delta
= tvsub_us(now
, rtp_time_0
);
934 assert(delta
<= UINT64_MAX
/ 88200);
935 target_rtp_time
= (delta
* playing
->format
.rate
936 * playing
->format
.channels
) / 1000000;
937 /* Overflows at ~6 years uptime with 44100Hz stereo */
939 /* rtp_time is the number of samples we've played. NB that we play
940 * RTP_AHEAD_MS ahead of ourselves, so it may legitimately be ahead of
941 * the value we deduce from time comparison.
943 * Suppose we have 1s track started at t=0, and another track begins to
944 * play at t=2s. Suppose RTP_AHEAD_MS=1000 and 44100Hz stereo. In that
945 * case we'll send 1s of audio as fast as we can, giving rtp_time=88200.
946 * rtp_time stops at this point.
948 * At t=2s we'll have calculated target_rtp_time=176400. In this case we
949 * set rtp_time=176400 and the player can correctly conclude that it
950 * should leave 1s between the tracks.
952 * Suppose instead that the second track arrives at t=0.5s, and that
953 * we've managed to transmit the whole of the first track already. We'll
954 * have target_rtp_time=44100.
956 * The desired behaviour is to play the second track back to back with
957 * first. In this case therefore we do not modify rtp_time.
959 * Is it ever right to reduce rtp_time? No; for that would imply
960 * transmitting packets with overlapping timestamp ranges, which does not
963 if(target_rtp_time
> rtp_time
) {
964 /* More time has elapsed than we've transmitted samples. That implies
965 * we've been 'sending' silence. */
966 info("advancing rtp_time by %"PRIu64
" samples",
967 target_rtp_time
- rtp_time
);
968 rtp_time
= target_rtp_time
;
969 } else if(target_rtp_time
< rtp_time
) {
970 const int64_t samples_ahead
= ((uint64_t)RTP_AHEAD_MS
971 * config
->sample_format
.rate
972 * config
->sample_format
.channels
975 if(target_rtp_time
+ samples_ahead
< rtp_time
) {
976 info("reversing rtp_time by %"PRIu64
" samples",
977 rtp_time
- target_rtp_time
);
981 header
.vpxcc
= 2 << 6; /* V=2, P=0, X=0, CC=0 */
982 header
.seq
= htons(rtp_seq
++);
983 header
.timestamp
= htonl((uint32_t)rtp_time
);
984 header
.ssrc
= rtp_id
;
985 header
.mpt
= (idled ?
0x80 : 0x00) | 10;
986 /* 10 = L16 = 16-bit x 2 x 44100KHz. We ought to deduce this value from
987 * the sample rate (in a library somewhere so that configuration.c can rule
988 * out invalid rates).
991 if(bytes
> NETWORK_BYTES
- sizeof header
) {
992 bytes
= NETWORK_BYTES
- sizeof header
;
993 /* Always send a whole number of frames */
994 bytes
-= bytes
% bpf
;
996 /* "The RTP clock rate used for generating the RTP timestamp is independent
997 * of the number of channels and the encoding; it equals the number of
998 * sampling periods per second. For N-channel encodings, each sampling
999 * period (say, 1/8000 of a second) generates N samples. (This terminology
1000 * is standard, but somewhat confusing, as the total number of samples
1001 * generated per second is then the sampling rate times the channel
1004 vec
[0].iov_base
= (void *)&header
;
1005 vec
[0].iov_len
= sizeof header
;
1006 vec
[1].iov_base
= playing
->buffer
+ playing
->start
;
1007 vec
[1].iov_len
= bytes
;
1009 written_bytes
= writev(bfd
, vec
, 2);
1010 } while(written_bytes
< 0 && errno
== EINTR
);
1011 if(written_bytes
< 0) {
1012 error(errno
, "error transmitting audio data");
1014 if(audio_errors
== 10)
1015 fatal(0, "too many audio errors");
1019 written_bytes
-= sizeof (struct rtp_header
);
1020 written_frames
= written_bytes
/ bpf
;
1021 /* Advance RTP's notion of the time */
1022 rtp_time
+= written_frames
* playing
->format
.channels
;
1023 return written_frames
;
1026 /** @brief Table of speaker backends */
1027 static const struct speaker_backend backends
[] = {
1054 { -1, 0, 0, 0, 0, 0 }
1057 int main(int argc
, char **argv
) {
1058 int n
, fd
, stdin_slot
, alsa_slots
, cmdfd_slot
, bfd_slot
, poke
, timeout
;
1060 struct speaker_message sm
;
1062 int alsa_nslots
= -1, err
;
1066 if(!setlocale(LC_CTYPE
, "")) fatal(errno
, "error calling setlocale");
1067 while((n
= getopt_long(argc
, argv
, "hVc:dD", options
, 0)) >= 0) {
1070 case 'V': version();
1071 case 'c': configfile
= optarg
; break;
1072 case 'd': debugging
= 1; break;
1073 case 'D': debugging
= 0; break;
1074 default: fatal(0, "invalid option");
1077 if(getenv("DISORDER_DEBUG_SPEAKER")) debugging
= 1;
1078 /* If stderr is a TTY then log there, otherwise to syslog. */
1080 openlog(progname
, LOG_PID
, LOG_DAEMON
);
1081 log_default
= &log_syslog
;
1083 if(config_read()) fatal(0, "cannot read configuration");
1084 /* ignore SIGPIPE */
1085 signal(SIGPIPE
, SIG_IGN
);
1087 signal(SIGCHLD
, reap
);
1088 /* set nice value */
1089 xnice(config
->nice_speaker
);
1092 /* make sure we're not root, whatever the config says */
1093 if(getuid() == 0 || geteuid() == 0) fatal(0, "do not run as root");
1094 /* identify the backend used to play */
1095 for(n
= 0; backends
[n
].backend
!= -1; ++n
)
1096 if(backends
[n
].backend
== config
->speaker_backend
)
1098 if(backends
[n
].backend
== -1)
1099 fatal(0, "unsupported backend %d", config
->speaker_backend
);
1100 backend
= &backends
[n
];
1101 /* backend-specific initialization */
1103 while(getppid() != 1) {
1105 /* Always ready for commands from the main server. */
1106 stdin_slot
= addfd(0, POLLIN
);
1107 /* Try to read sample data for the currently playing track if there is
1109 if(playing
&& !playing
->eof
&& playing
->used
< playing
->size
) {
1110 playing
->slot
= addfd(playing
->fd
, POLLIN
);
1113 /* If forceplay is set then wait until it succeeds before waiting on the
1118 /* By default we will wait up to a second before thinking about current
1121 if(ready
&& !forceplay
) {
1122 switch(config
->speaker_backend
) {
1123 case BACKEND_COMMAND
:
1124 /* We send sample data to the subprocess as fast as it can accept it.
1125 * This isn't ideal as pause latency can be very high as a result. */
1127 cmdfd_slot
= addfd(cmdfd
, POLLOUT
);
1129 case BACKEND_NETWORK
: {
1132 uint64_t target_rtp_time
;
1133 const int64_t samples_ahead
= ((uint64_t)RTP_AHEAD_MS
1134 * config
->sample_format
.rate
1135 * config
->sample_format
.channels
1138 static unsigned logit
;
1141 /* If we're starting then initialize the base time */
1143 xgettimeofday(&rtp_time_0
, 0);
1144 /* We send audio data whenever we get RTP_AHEAD seconds or more
1146 xgettimeofday(&now
, 0);
1147 target_us
= tvsub_us(now
, rtp_time_0
);
1148 assert(target_us
<= UINT64_MAX
/ 88200);
1149 target_rtp_time
= (target_us
* config
->sample_format
.rate
1150 * config
->sample_format
.channels
)
1154 /* TODO remove logging guff */
1155 if(!(logit
++ & 1023))
1156 info("rtp_time %llu target %llu difference %lld [%lld]",
1157 rtp_time
, target_rtp_time
,
1158 rtp_time
- target_rtp_time
,
1161 if((int64_t)(rtp_time
- target_rtp_time
) < samples_ahead
)
1162 bfd_slot
= addfd(bfd
, POLLOUT
);
1166 case BACKEND_ALSA
: {
1167 /* We send sample data to ALSA as fast as it can accept it, relying on
1168 * the fact that it has a relatively small buffer to minimize pause
1175 alsa_nslots
= snd_pcm_poll_descriptors(pcm
, &fds
[fdno
], NFDS
- fdno
);
1176 if((alsa_nslots
<= 0
1177 || !(fds
[alsa_slots
].events
& POLLOUT
))
1178 && snd_pcm_state(pcm
) == SND_PCM_STATE_XRUN
) {
1179 error(0, "underrun detected after call to snd_pcm_poll_descriptors()");
1180 if((err
= snd_pcm_prepare(pcm
)))
1181 fatal(0, "error calling snd_pcm_prepare: %d", err
);
1184 } while(retry
-- > 0);
1185 if(alsa_nslots
>= 0)
1186 fdno
+= alsa_nslots
;
1191 assert(!"unknown backend");
1194 /* If any other tracks don't have a full buffer, try to read sample data
1196 for(t
= tracks
; t
; t
= t
->next
)
1198 if(!t
->eof
&& t
->used
< t
->size
) {
1199 t
->slot
= addfd(t
->fd
, POLLIN
| POLLHUP
);
1203 /* Wait for something interesting to happen */
1204 n
= poll(fds
, fdno
, timeout
);
1206 if(errno
== EINTR
) continue;
1207 fatal(errno
, "error calling poll");
1209 /* Play some sound before doing anything else */
1211 switch(config
->speaker_backend
) {
1214 if(alsa_slots
!= -1) {
1215 unsigned short alsa_revents
;
1217 if((err
= snd_pcm_poll_descriptors_revents(pcm
,
1220 &alsa_revents
)) < 0)
1221 fatal(0, "error calling snd_pcm_poll_descriptors_revents: %d", err
);
1222 if(alsa_revents
& (POLLOUT
| POLLERR
))
1228 case BACKEND_COMMAND
:
1229 if(cmdfd_slot
!= -1) {
1230 if(fds
[cmdfd_slot
].revents
& (POLLOUT
| POLLERR
))
1235 case BACKEND_NETWORK
:
1236 if(bfd_slot
!= -1) {
1237 if(fds
[bfd_slot
].revents
& (POLLOUT
| POLLERR
))
1244 /* Some attempt to play must have failed */
1245 if(playing
&& !paused
)
1248 forceplay
= 0; /* just in case */
1250 /* Perhaps we have a command to process */
1251 if(fds
[stdin_slot
].revents
& POLLIN
) {
1252 n
= speaker_recv(0, &sm
, &fd
);
1256 D(("SM_PREPARE %s %d", sm
.id
, fd
));
1257 if(fd
== -1) fatal(0, "got SM_PREPARE but no file descriptor");
1258 t
= findtrack(sm
.id
, 1);
1262 D(("SM_PLAY %s %d", sm
.id
, fd
));
1263 if(playing
) fatal(0, "got SM_PLAY but already playing something");
1264 t
= findtrack(sm
.id
, 1);
1265 if(fd
!= -1) acquire(t
, fd
);
1285 D(("SM_CANCEL %s", sm
.id
));
1286 t
= removetrack(sm
.id
);
1289 sm
.type
= SM_FINISHED
;
1290 strcpy(sm
.id
, playing
->id
);
1291 speaker_send(1, &sm
, 0);
1296 error(0, "SM_CANCEL for unknown track %s", sm
.id
);
1301 if(config_read()) error(0, "cannot read configuration");
1302 info("reloaded configuration");
1305 error(0, "unknown message type %d", sm
.type
);
1308 /* Read in any buffered data */
1309 for(t
= tracks
; t
; t
= t
->next
)
1310 if(t
->slot
!= -1 && (fds
[t
->slot
].revents
& (POLLIN
| POLLHUP
)))
1312 /* We might be able to play now */
1313 if(ready
&& forceplay
&& playing
&& !paused
)
1315 /* Maybe we finished playing a track somewhere in the above */
1317 /* If we don't need the sound device for now then close it for the benefit
1318 * of anyone else who wants it. */
1319 if((!playing
|| paused
) && ready
)
1321 /* If we've not reported out state for a second do so now. */
1322 if(time(0) > last_report
)
1325 info("stopped (parent terminated)");
1334 indent-tabs-mode:nil