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
21 /* This program deliberately does not use the garbage collector even though it
22 * might be convenient to do so. This is for two reasons. Firstly some libao
23 * drivers are implemented using threads and we do not want to have to deal
24 * with potential interactions between threading and garbage collection.
25 * Secondly this process needs to be able to respond quickly and this is not
26 * compatible with the collector hanging the program even relatively
42 #include <sys/select.h>
47 #include <sys/socket.h>
52 #include "configuration.h"
64 #include <alsa/asoundlib.h>
67 #ifdef WORDS_BIGENDIAN
68 # define MACHINE_AO_FMT AO_FMT_BIG
70 # define MACHINE_AO_FMT AO_FMT_LITTLE
73 #define BUFFER_SECONDS 5 /* How many seconds of input to
76 #define FRAMES 4096 /* Frame batch size */
78 #define NETWORK_BYTES 1024 /* Bytes to send per network packet */
79 /* (don't make this too big or arithmetic will start to overflow) */
81 #define RTP_AHEAD 2 /* Max RTP playahead (seconds) */
83 #define NFDS 256 /* Max FDs to poll for */
85 /* Known tracks are kept in a linked list. We don't normally to have
86 * more than two - maybe three at the outside. */
88 struct track
*next
; /* next track */
89 int fd
; /* input FD */
91 size_t start
, used
; /* start + bytes used */
92 int eof
; /* input is at EOF */
93 int got_format
; /* got format yet? */
94 ao_sample_format format
; /* sample format */
95 unsigned long long played
; /* number of frames played */
96 char *buffer
; /* sample buffer */
97 size_t size
; /* sample buffer size */
98 int slot
; /* poll array slot */
99 } *tracks
, *playing
; /* all tracks + playing track */
101 static time_t last_report
; /* when we last reported */
102 static int paused
; /* pause status */
103 static ao_sample_format pcm_format
; /* current format if aodev != 0 */
104 static size_t bpf
; /* bytes per frame */
105 static struct pollfd fds
[NFDS
]; /* if we need more than that */
106 static int fdno
; /* fd number */
107 static size_t bufsize
; /* buffer size */
109 static snd_pcm_t
*pcm
; /* current pcm handle */
110 static snd_pcm_uframes_t last_pcm_bufsize
; /* last seen buffer size */
112 static int ready
; /* ready to send audio */
113 static int forceplay
; /* frames to force play */
114 static int cmdfd
= -1; /* child process input */
115 static int bfd
= -1; /* broadcast FD */
116 static uint32_t rtp_time
; /* RTP timestamp */
117 static struct timeval rtp_time_real
; /* corresponding real time */
118 static uint16_t rtp_seq
; /* frame sequence number */
119 static uint32_t rtp_id
; /* RTP SSRC */
120 static int idled
; /* set when idled */
121 static int audio_errors
; /* audio error counter */
123 static const struct option options
[] = {
124 { "help", no_argument
, 0, 'h' },
125 { "version", no_argument
, 0, 'V' },
126 { "config", required_argument
, 0, 'c' },
127 { "debug", no_argument
, 0, 'd' },
128 { "no-debug", no_argument
, 0, 'D' },
132 /* Display usage message and terminate. */
133 static void help(void) {
135 " disorder-speaker [OPTIONS]\n"
137 " --help, -h Display usage message\n"
138 " --version, -V Display version number\n"
139 " --config PATH, -c PATH Set configuration file\n"
140 " --debug, -d Turn on debugging\n"
142 "Speaker process for DisOrder. Not intended to be run\n"
148 /* Display version number and terminate. */
149 static void version(void) {
150 xprintf("disorder-speaker version %s\n", disorder_version_string
);
155 /* Return the number of bytes per frame in FORMAT. */
156 static size_t bytes_per_frame(const ao_sample_format
*format
) {
157 return format
->channels
* format
->bits
/ 8;
160 /* Find track ID, maybe creating it if not found. */
161 static struct track
*findtrack(const char *id
, int create
) {
164 D(("findtrack %s %d", id
, create
));
165 for(t
= tracks
; t
&& strcmp(id
, t
->id
); t
= t
->next
)
168 t
= xmalloc(sizeof *t
);
173 /* The initial input buffer will be the sample format. */
174 t
->buffer
= (void *)&t
->format
;
175 t
->size
= sizeof t
->format
;
180 /* Remove track ID (but do not destroy it). */
181 static struct track
*removetrack(const char *id
) {
182 struct track
*t
, **tt
;
184 D(("removetrack %s", id
));
185 for(tt
= &tracks
; (t
= *tt
) && strcmp(id
, t
->id
); tt
= &t
->next
)
192 /* Destroy a track. */
193 static void destroy(struct track
*t
) {
194 D(("destroy %s", t
->id
));
195 if(t
->fd
!= -1) xclose(t
->fd
);
196 if(t
->buffer
!= (void *)&t
->format
) free(t
->buffer
);
200 /* Notice a new FD. */
201 static void acquire(struct track
*t
, int fd
) {
202 D(("acquire %s %d", t
->id
, fd
));
209 /* Read data into a sample buffer. Return 0 on success, -1 on EOF. */
210 static int fill(struct track
*t
) {
214 D(("fill %s: eof=%d used=%zu size=%zu got_format=%d",
215 t
->id
, t
->eof
, t
->used
, t
->size
, t
->got_format
));
216 if(t
->eof
) return -1;
217 if(t
->used
< t
->size
) {
218 /* there is room left in the buffer */
219 where
= (t
->start
+ t
->used
) % t
->size
;
221 /* We are reading audio data, get as much as we can */
222 if(where
>= t
->start
) left
= t
->size
- where
;
223 else left
= t
->start
- where
;
225 /* We are still waiting for the format, only get that */
226 left
= sizeof (ao_sample_format
) - t
->used
;
228 n
= read(t
->fd
, t
->buffer
+ where
, left
);
229 } while(n
< 0 && errno
== EINTR
);
231 if(errno
!= EAGAIN
) fatal(errno
, "error reading sample stream");
235 D(("fill %s: eof detected", t
->id
));
240 if(!t
->got_format
&& t
->used
>= sizeof (ao_sample_format
)) {
241 assert(t
->used
== sizeof (ao_sample_format
));
242 /* Check that our assumptions are met. */
243 if(t
->format
.bits
& 7)
244 fatal(0, "bits per sample not a multiple of 8");
245 /* Make a new buffer for audio data. */
246 t
->size
= bytes_per_frame(&t
->format
) * t
->format
.rate
* BUFFER_SECONDS
;
247 t
->buffer
= xmalloc(t
->size
);
250 D(("got format for %s", t
->id
));
256 /* Return true if A and B denote identical libao formats, else false. */
257 static int formats_equal(const ao_sample_format
*a
,
258 const ao_sample_format
*b
) {
259 return (a
->bits
== b
->bits
260 && a
->rate
== b
->rate
261 && a
->channels
== b
->channels
262 && a
->byte_format
== b
->byte_format
);
265 /* Close the sound device. */
266 static void idle(void) {
269 if(config
->speaker_backend
== BACKEND_ALSA
&& pcm
) {
272 if((err
= snd_pcm_nonblock(pcm
, 0)) < 0)
273 fatal(0, "error calling snd_pcm_nonblock: %d", err
);
280 D(("released audio device"));
287 /* Abandon the current track */
288 static void abandon(void) {
289 struct speaker_message sm
;
292 memset(&sm
, 0, sizeof sm
);
293 sm
.type
= SM_FINISHED
;
294 strcpy(sm
.id
, playing
->id
);
295 speaker_send(1, &sm
, 0);
296 removetrack(playing
->id
);
303 static void log_params(snd_pcm_hw_params_t
*hwparams
,
304 snd_pcm_sw_params_t
*swparams
) {
308 return; /* too verbose */
313 snd_pcm_sw_params_get_silence_size(swparams
, &f
);
314 info("sw silence_size=%lu", (unsigned long)f
);
315 snd_pcm_sw_params_get_silence_threshold(swparams
, &f
);
316 info("sw silence_threshold=%lu", (unsigned long)f
);
317 snd_pcm_sw_params_get_sleep_min(swparams
, &u
);
318 info("sw sleep_min=%lu", (unsigned long)u
);
319 snd_pcm_sw_params_get_start_threshold(swparams
, &f
);
320 info("sw start_threshold=%lu", (unsigned long)f
);
321 snd_pcm_sw_params_get_stop_threshold(swparams
, &f
);
322 info("sw stop_threshold=%lu", (unsigned long)f
);
323 snd_pcm_sw_params_get_xfer_align(swparams
, &f
);
324 info("sw xfer_align=%lu", (unsigned long)f
);
329 static void soxargs(const char ***pp
, char **qq
, ao_sample_format
*ao
) {
334 *(*pp
)++ = *qq
; n
= sprintf(*qq
, "-r%d", ao
->rate
); *qq
+= n
+ 1;
335 *(*pp
)++ = *qq
; n
= sprintf(*qq
, "-c%d", ao
->channels
); *qq
+= n
+ 1;
336 /* sox 12.17.9 insists on -b etc; CVS sox insists on -<n> etc; both are
338 switch(config
->sox_generation
) {
341 && ao
->byte_format
!= AO_FMT_NATIVE
342 && ao
->byte_format
!= MACHINE_AO_FMT
) {
346 case 8: *(*pp
)++ = "-b"; break;
347 case 16: *(*pp
)++ = "-w"; break;
348 case 32: *(*pp
)++ = "-l"; break;
349 case 64: *(*pp
)++ = "-d"; break;
350 default: fatal(0, "cannot handle sample size %d", (int)ao
->bits
);
354 switch(ao
->byte_format
) {
355 case AO_FMT_NATIVE
: break;
356 case AO_FMT_BIG
: *(*pp
)++ = "-B"; break;
357 case AO_FMT_LITTLE
: *(*pp
)++ = "-L"; break;
359 *(*pp
)++ = *qq
; n
= sprintf(*qq
, "-%d", ao
->bits
/8); *qq
+= n
+ 1;
364 /* Make sure the sound device is open and has the right sample format. Return
365 * 0 on success and -1 on error. */
366 static int activate(void) {
367 /* If we don't know the format yet we cannot start. */
368 if(!playing
->got_format
) {
369 D((" - not got format for %s", playing
->id
));
372 switch(config
->speaker_backend
) {
373 case BACKEND_COMMAND
:
374 case BACKEND_NETWORK
:
375 /* If we pass audio on to some other agent then we enforce the configured
376 * sample format on the *inbound* audio data. */
377 if(!formats_equal(&playing
->format
, &config
->sample_format
)) {
378 char argbuf
[1024], *q
= argbuf
;
379 const char *av
[18], **pp
= av
;
383 soxargs(&pp
, &q
, &playing
->format
);
385 soxargs(&pp
, &q
, &config
->sample_format
);
389 for(pp
= av
; *pp
; pp
++)
390 D(("sox arg[%d] = %s", pp
- av
, *pp
));
396 xdup2(playing
->fd
, 0);
397 xdup2(soxpipe
[1], 1);
398 fcntl(0, F_SETFL
, fcntl(0, F_GETFL
) & ~O_NONBLOCK
);
402 execvp("sox", (char **)av
);
405 D(("forking sox for format conversion (kid = %d)", soxkid
));
408 playing
->fd
= soxpipe
[0];
409 playing
->format
= config
->sample_format
;
413 pcm_format
= config
->sample_format
;
414 bufsize
= 3 * FRAMES
;
415 bpf
= bytes_per_frame(&config
->sample_format
);
416 D(("acquired audio device"));
422 /* If we need to change format then close the current device. */
423 if(pcm
&& !formats_equal(&playing
->format
, &pcm_format
))
426 snd_pcm_hw_params_t
*hwparams
;
427 snd_pcm_sw_params_t
*swparams
;
428 snd_pcm_uframes_t pcm_bufsize
;
430 int sample_format
= 0;
434 if((err
= snd_pcm_open(&pcm
,
436 SND_PCM_STREAM_PLAYBACK
,
437 SND_PCM_NONBLOCK
))) {
438 error(0, "error from snd_pcm_open: %d", err
);
441 snd_pcm_hw_params_alloca(&hwparams
);
442 D(("set up hw params"));
443 if((err
= snd_pcm_hw_params_any(pcm
, hwparams
)) < 0)
444 fatal(0, "error from snd_pcm_hw_params_any: %d", err
);
445 if((err
= snd_pcm_hw_params_set_access(pcm
, hwparams
,
446 SND_PCM_ACCESS_RW_INTERLEAVED
)) < 0)
447 fatal(0, "error from snd_pcm_hw_params_set_access: %d", err
);
448 switch(playing
->format
.bits
) {
450 sample_format
= SND_PCM_FORMAT_S8
;
453 switch(playing
->format
.byte_format
) {
454 case AO_FMT_NATIVE
: sample_format
= SND_PCM_FORMAT_S16
; break;
455 case AO_FMT_LITTLE
: sample_format
= SND_PCM_FORMAT_S16_LE
; break;
456 case AO_FMT_BIG
: sample_format
= SND_PCM_FORMAT_S16_BE
; break;
457 error(0, "unrecognized byte format %d", playing
->format
.byte_format
);
462 error(0, "unsupported sample size %d", playing
->format
.bits
);
465 if((err
= snd_pcm_hw_params_set_format(pcm
, hwparams
,
466 sample_format
)) < 0) {
467 error(0, "error from snd_pcm_hw_params_set_format (%d): %d",
471 rate
= playing
->format
.rate
;
472 if((err
= snd_pcm_hw_params_set_rate_near(pcm
, hwparams
, &rate
, 0)) < 0) {
473 error(0, "error from snd_pcm_hw_params_set_rate (%d): %d",
474 playing
->format
.rate
, err
);
477 if(rate
!= (unsigned)playing
->format
.rate
)
478 info("want rate %d, got %u", playing
->format
.rate
, rate
);
479 if((err
= snd_pcm_hw_params_set_channels(pcm
, hwparams
,
480 playing
->format
.channels
)) < 0) {
481 error(0, "error from snd_pcm_hw_params_set_channels (%d): %d",
482 playing
->format
.channels
, err
);
485 bufsize
= 3 * FRAMES
;
486 pcm_bufsize
= bufsize
;
487 if((err
= snd_pcm_hw_params_set_buffer_size_near(pcm
, hwparams
,
489 fatal(0, "error from snd_pcm_hw_params_set_buffer_size (%d): %d",
491 if(pcm_bufsize
!= 3 * FRAMES
&& pcm_bufsize
!= last_pcm_bufsize
)
492 info("asked for PCM buffer of %d frames, got %d",
493 3 * FRAMES
, (int)pcm_bufsize
);
494 last_pcm_bufsize
= pcm_bufsize
;
495 if((err
= snd_pcm_hw_params(pcm
, hwparams
)) < 0)
496 fatal(0, "error calling snd_pcm_hw_params: %d", err
);
497 D(("set up sw params"));
498 snd_pcm_sw_params_alloca(&swparams
);
499 if((err
= snd_pcm_sw_params_current(pcm
, swparams
)) < 0)
500 fatal(0, "error calling snd_pcm_sw_params_current: %d", err
);
501 if((err
= snd_pcm_sw_params_set_avail_min(pcm
, swparams
, FRAMES
)) < 0)
502 fatal(0, "error calling snd_pcm_sw_params_set_avail_min %d: %d",
504 if((err
= snd_pcm_sw_params(pcm
, swparams
)) < 0)
505 fatal(0, "error calling snd_pcm_sw_params: %d", err
);
506 pcm_format
= playing
->format
;
507 bpf
= bytes_per_frame(&pcm_format
);
508 D(("acquired audio device"));
509 log_params(hwparams
, swparams
);
516 /* We assume the error is temporary and that we'll retry in a bit. */
528 /* Check to see whether the current track has finished playing */
529 static void maybe_finished(void) {
532 && (!playing
->got_format
533 || playing
->used
< bytes_per_frame(&playing
->format
)))
537 static void fork_cmd(void) {
540 if(cmdfd
!= -1) close(cmdfd
);
547 execl("/bin/sh", "sh", "-c", config
->speaker_command
, (char *)0);
548 fatal(errno
, "error execing /bin/sh");
552 D(("forked cmd %d, fd = %d", cmdpid
, cmdfd
));
555 static void play(size_t frames
) {
556 size_t avail_bytes
, written_frames
;
557 ssize_t written_bytes
;
558 struct rtp_header header
;
565 forceplay
= 0; /* Must have called abandon() */
568 D(("play: play %zu/%zu%s %dHz %db %dc", frames
, playing
->used
/ bpf
,
569 playing
->eof ?
" EOF" : "",
570 playing
->format
.rate
,
571 playing
->format
.bits
,
572 playing
->format
.channels
));
573 /* If we haven't got enough bytes yet wait until we have. Exception: when
575 if(playing
->used
< frames
* bpf
&& !playing
->eof
) {
579 /* We have got enough data so don't force play again */
581 /* Figure out how many frames there are available to write */
582 if(playing
->start
+ playing
->used
> playing
->size
)
583 avail_bytes
= playing
->size
- playing
->start
;
585 avail_bytes
= playing
->used
;
587 switch(config
->speaker_backend
) {
590 snd_pcm_sframes_t pcm_written_frames
;
594 avail_frames
= avail_bytes
/ bpf
;
595 if(avail_frames
> frames
)
596 avail_frames
= frames
;
599 pcm_written_frames
= snd_pcm_writei(pcm
,
600 playing
->buffer
+ playing
->start
,
602 D(("actually play %zu frames, wrote %d",
603 avail_frames
, (int)pcm_written_frames
));
604 if(pcm_written_frames
< 0) {
605 switch(pcm_written_frames
) {
606 case -EPIPE
: /* underrun */
607 error(0, "snd_pcm_writei reports underrun");
608 if((err
= snd_pcm_prepare(pcm
)) < 0)
609 fatal(0, "error calling snd_pcm_prepare: %d", err
);
614 fatal(0, "error calling snd_pcm_writei: %d",
615 (int)pcm_written_frames
);
618 written_frames
= pcm_written_frames
;
619 written_bytes
= written_frames
* bpf
;
623 case BACKEND_COMMAND
:
624 if(avail_bytes
> frames
* bpf
)
625 avail_bytes
= frames
* bpf
;
626 written_bytes
= write(cmdfd
, playing
->buffer
+ playing
->start
,
628 D(("actually play %zu bytes, wrote %d",
629 avail_bytes
, (int)written_bytes
));
630 if(written_bytes
< 0) {
633 error(0, "hmm, command died; trying another");
640 written_frames
= written_bytes
/ bpf
; /* good enough */
642 case BACKEND_NETWORK
:
643 /* We transmit using RTP (RFC3550) and attempt to conform to the internet
644 * AVT profile (RFC3551). */
645 if(rtp_time_real
.tv_sec
== 0)
646 xgettimeofday(&rtp_time_real
, 0);
649 xgettimeofday(&now
, 0);
650 /* There's been a gap. Fix up the RTP time accordingly. */
651 const long offset
= (((now
.tv_sec
+ now
.tv_usec
/1000000.0)
652 - (rtp_time_real
.tv_sec
+ rtp_time_real
.tv_usec
/ 1000000.0))
653 * playing
->format
.rate
* playing
->format
.channels
);
655 info("offset RTP timestamp by %ld", offset
);
658 xgettimeofday(&rtp_time_real
, 0);
660 header
.vpxcc
= 2 << 6; /* V=2, P=0, X=0, CC=0 */
661 header
.seq
= htons(rtp_seq
++);
662 header
.timestamp
= htonl(rtp_time
);
663 header
.ssrc
= rtp_id
;
664 header
.mpt
= (idled ?
0x80 : 0x00) | 10;
665 /* 10 = L16 = 16-bit x 2 x 44100KHz. We ought to deduce this value from
666 * the sample rate (in a library somewhere so that configuration.c can rule
667 * out invalid rates).
670 if(avail_bytes
> NETWORK_BYTES
- sizeof header
) {
671 avail_bytes
= NETWORK_BYTES
- sizeof header
;
672 avail_bytes
-= avail_bytes
% bpf
;
674 /* "The RTP clock rate used for generating the RTP timestamp is independent
675 * of the number of channels and the encoding; it equals the number of
676 * sampling periods per second. For N-channel encodings, each sampling
677 * period (say, 1/8000 of a second) generates N samples. (This terminology
678 * is standard, but somewhat confusing, as the total number of samples
679 * generated per second is then the sampling rate times the channel
682 vec
[0].iov_base
= (void *)&header
;
683 vec
[0].iov_len
= sizeof header
;
684 vec
[1].iov_base
= playing
->buffer
+ playing
->start
;
685 vec
[1].iov_len
= avail_bytes
;
688 char buffer
[3 * sizeof header
+ 1];
690 const uint8_t *ptr
= (void *)&header
;
692 for(n
= 0; n
< sizeof header
; ++n
)
693 sprintf(&buffer
[3 * n
], "%02x ", *ptr
++);
698 written_bytes
= writev(bfd
,
701 } while(written_bytes
< 0 && errno
== EINTR
);
702 if(written_bytes
< 0) {
703 error(errno
, "error transmitting audio data");
705 if(audio_errors
== 10)
706 fatal(0, "too many audio errors");
710 written_bytes
= avail_bytes
;
711 written_frames
= written_bytes
/ bpf
;
712 /* Advance RTP's notion of the time */
713 rtp_time
+= written_frames
* playing
->format
.channels
;
714 /* Advance the corresponding real time */
715 assert(NETWORK_BYTES
<= 2000); /* else risk overflowing 32 bits */
716 rtp_time_real
.tv_usec
+= written_frames
* 1000000 / playing
->format
.rate
;
717 if(rtp_time_real
.tv_usec
>= 1000000) {
718 ++rtp_time_real
.tv_sec
;
719 rtp_time_real
.tv_usec
-= 1000000;
721 assert(rtp_time_real
.tv_usec
< 1000000);
726 /* written_bytes and written_frames had better both be set and correct by
728 playing
->start
+= written_bytes
;
729 playing
->used
-= written_bytes
;
730 playing
->played
+= written_frames
;
731 /* If the pointer is at the end of the buffer (or the buffer is completely
732 * empty) wrap it back to the start. */
733 if(!playing
->used
|| playing
->start
== playing
->size
)
735 frames
-= written_frames
;
738 /* Notify the server what we're up to. */
739 static void report(void) {
740 struct speaker_message sm
;
742 if(playing
&& playing
->buffer
!= (void *)&playing
->format
) {
743 memset(&sm
, 0, sizeof sm
);
744 sm
.type
= paused ? SM_PAUSED
: SM_PLAYING
;
745 strcpy(sm
.id
, playing
->id
);
746 sm
.data
= playing
->played
/ playing
->format
.rate
;
747 speaker_send(1, &sm
, 0);
752 static void reap(int __attribute__((unused
)) sig
) {
757 cmdpid
= waitpid(-1, &st
, WNOHANG
);
759 signal(SIGCHLD
, reap
);
762 static int addfd(int fd
, int events
) {
765 fds
[fdno
].events
= events
;
771 int main(int argc
, char **argv
) {
772 int n
, fd
, stdin_slot
, alsa_slots
, cmdfd_slot
, bfd_slot
, poke
, timeout
;
773 struct timeval now
, delta
;
775 struct speaker_message sm
;
776 struct addrinfo
*res
, *sres
;
777 static const struct addrinfo pref
= {
787 static const struct addrinfo prefbind
= {
797 static const int one
= 1;
798 char *sockname
, *ssockname
;
800 int alsa_nslots
= -1, err
;
804 if(!setlocale(LC_CTYPE
, "")) fatal(errno
, "error calling setlocale");
805 while((n
= getopt_long(argc
, argv
, "hVc:dD", options
, 0)) >= 0) {
809 case 'c': configfile
= optarg
; break;
810 case 'd': debugging
= 1; break;
811 case 'D': debugging
= 0; break;
812 default: fatal(0, "invalid option");
815 if(getenv("DISORDER_DEBUG_SPEAKER")) debugging
= 1;
816 /* If stderr is a TTY then log there, otherwise to syslog. */
818 openlog(progname
, LOG_PID
, LOG_DAEMON
);
819 log_default
= &log_syslog
;
821 if(config_read()) fatal(0, "cannot read configuration");
823 signal(SIGPIPE
, SIG_IGN
);
825 signal(SIGCHLD
, reap
);
827 xnice(config
->nice_speaker
);
830 /* make sure we're not root, whatever the config says */
831 if(getuid() == 0 || geteuid() == 0) fatal(0, "do not run as root");
832 switch(config
->speaker_backend
) {
834 info("selected ALSA backend");
835 case BACKEND_COMMAND
:
836 info("selected command backend");
839 case BACKEND_NETWORK
:
840 res
= get_address(&config
->broadcast
, &pref
, &sockname
);
842 if(config
->broadcast_from
.n
) {
843 sres
= get_address(&config
->broadcast_from
, &prefbind
, &ssockname
);
847 if((bfd
= socket(res
->ai_family
,
849 res
->ai_protocol
)) < 0)
850 fatal(errno
, "error creating broadcast socket");
851 if(setsockopt(bfd
, SOL_SOCKET
, SO_BROADCAST
, &one
, sizeof one
) < 0)
852 fatal(errno
, "error settting SO_BROADCAST on broadcast socket");
853 /* We might well want to set additional broadcast- or multicast-related
855 if(sres
&& bind(bfd
, sres
->ai_addr
, sres
->ai_addrlen
) < 0)
856 fatal(errno
, "error binding broadcast socket to %s", ssockname
);
857 if(connect(bfd
, res
->ai_addr
, res
->ai_addrlen
) < 0)
858 fatal(errno
, "error connecting broadcast socket to %s", sockname
);
860 gcry_randomize(&rtp_id
, sizeof rtp_id
, GCRY_STRONG_RANDOM
);
861 info("selected network backend, sending to %s", sockname
);
862 if(config
->sample_format
.byte_format
!= AO_FMT_BIG
) {
863 info("forcing big-endian sample format");
864 config
->sample_format
.byte_format
= AO_FMT_BIG
;
868 fatal(0, "unknown backend %d", config
->speaker_backend
);
870 while(getppid() != 1) {
872 /* Always ready for commands from the main server. */
873 stdin_slot
= addfd(0, POLLIN
);
874 /* Try to read sample data for the currently playing track if there is
876 if(playing
&& !playing
->eof
&& playing
->used
< playing
->size
) {
877 playing
->slot
= addfd(playing
->fd
, POLLIN
);
880 /* If forceplay is set then wait until it succeeds before waiting on the
885 /* By default we will wait up to a second before thinking about current
888 if(ready
&& !forceplay
) {
889 switch(config
->speaker_backend
) {
890 case BACKEND_COMMAND
:
891 /* We send sample data to the subprocess as fast as it can accept it.
892 * This isn't ideal as pause latency can be very high as a result. */
894 cmdfd_slot
= addfd(cmdfd
, POLLOUT
);
896 case BACKEND_NETWORK
:
897 /* We want to keep the notional playing point somewhere in the near
898 * future. If it's too near then clients that attempt even the
899 * slightest amount of read-ahead will never catch up, and those that
900 * don't will skip whenever there's a trivial network delay. If it's
901 * too far ahead then pause latency will be too high.
903 xgettimeofday(&now
, 0);
904 delta
= tvsub(rtp_time_real
, now
);
905 if(delta
.tv_sec
< RTP_AHEAD
) {
906 D(("delta = %ld.%06ld", (long)delta
.tv_sec
, (long)delta
.tv_usec
));
907 bfd_slot
= addfd(bfd
, POLLOUT
);
909 rtp_time_real
= now
; /* catch up */
914 /* We send sample data to ALSA as fast as it can accept it, relying on
915 * the fact that it has a relatively small buffer to minimize pause
922 alsa_nslots
= snd_pcm_poll_descriptors(pcm
, &fds
[fdno
], NFDS
- fdno
);
924 || !(fds
[alsa_slots
].events
& POLLOUT
))
925 && snd_pcm_state(pcm
) == SND_PCM_STATE_XRUN
) {
926 error(0, "underrun detected after call to snd_pcm_poll_descriptors()");
927 if((err
= snd_pcm_prepare(pcm
)))
928 fatal(0, "error calling snd_pcm_prepare: %d", err
);
931 } while(retry
-- > 0);
938 assert(!"unknown backend");
941 /* If any other tracks don't have a full buffer, try to read sample data
943 for(t
= tracks
; t
; t
= t
->next
)
945 if(!t
->eof
&& t
->used
< t
->size
) {
946 t
->slot
= addfd(t
->fd
, POLLIN
| POLLHUP
);
950 /* Wait for something interesting to happen */
951 n
= poll(fds
, fdno
, timeout
);
953 if(errno
== EINTR
) continue;
954 fatal(errno
, "error calling poll");
956 /* Play some sound before doing anything else */
958 switch(config
->speaker_backend
) {
961 if(alsa_slots
!= -1) {
962 unsigned short alsa_revents
;
964 if((err
= snd_pcm_poll_descriptors_revents(pcm
,
968 fatal(0, "error calling snd_pcm_poll_descriptors_revents: %d", err
);
969 if(alsa_revents
& (POLLOUT
| POLLERR
))
975 case BACKEND_COMMAND
:
976 if(cmdfd_slot
!= -1) {
977 if(fds
[cmdfd_slot
].revents
& (POLLOUT
| POLLERR
))
982 case BACKEND_NETWORK
:
984 if(fds
[bfd_slot
].revents
& (POLLOUT
| POLLERR
))
991 /* Some attempt to play must have failed */
992 if(playing
&& !paused
)
995 forceplay
= 0; /* just in case */
997 /* Perhaps we have a command to process */
998 if(fds
[stdin_slot
].revents
& POLLIN
) {
999 n
= speaker_recv(0, &sm
, &fd
);
1003 D(("SM_PREPARE %s %d", sm
.id
, fd
));
1004 if(fd
== -1) fatal(0, "got SM_PREPARE but no file descriptor");
1005 t
= findtrack(sm
.id
, 1);
1009 D(("SM_PLAY %s %d", sm
.id
, fd
));
1010 if(playing
) fatal(0, "got SM_PLAY but already playing something");
1011 t
= findtrack(sm
.id
, 1);
1012 if(fd
!= -1) acquire(t
, fd
);
1032 D(("SM_CANCEL %s", sm
.id
));
1033 t
= removetrack(sm
.id
);
1036 sm
.type
= SM_FINISHED
;
1037 strcpy(sm
.id
, playing
->id
);
1038 speaker_send(1, &sm
, 0);
1043 error(0, "SM_CANCEL for unknown track %s", sm
.id
);
1048 if(config_read()) error(0, "cannot read configuration");
1049 info("reloaded configuration");
1052 error(0, "unknown message type %d", sm
.type
);
1055 /* Read in any buffered data */
1056 for(t
= tracks
; t
; t
= t
->next
)
1057 if(t
->slot
!= -1 && (fds
[t
->slot
].revents
& (POLLIN
| POLLHUP
)))
1059 /* We might be able to play now */
1060 if(ready
&& forceplay
&& playing
&& !paused
)
1062 /* Maybe we finished playing a track somewhere in the above */
1064 /* If we don't need the sound device for now then close it for the benefit
1065 * of anyone else who wants it. */
1066 if((!playing
|| paused
) && ready
)
1068 /* If we've not reported out state for a second do so now. */
1069 if(time(0) > last_report
)
1072 info("stopped (parent terminated)");
1081 indent-tabs-mode:nil