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
, write_bytes
, written_frames
;
566 ssize_t written_bytes
;
567 struct rtp_header header
;
570 /* Make sure the output device is activated */
575 forceplay
= 0; /* Must have called abandon() */
578 D(("play: play %zu/%zu%s %dHz %db %dc", frames
, playing
->used
/ bpf
,
579 playing
->eof ?
" EOF" : "",
580 playing
->format
.rate
,
581 playing
->format
.bits
,
582 playing
->format
.channels
));
583 /* If we haven't got enough bytes yet wait until we have. Exception: when
585 if(playing
->used
< frames
* bpf
&& !playing
->eof
) {
589 /* We have got enough data so don't force play again */
591 /* Figure out how many frames there are available to write */
592 if(playing
->start
+ playing
->used
> playing
->size
)
593 /* The ring buffer is currently wrapped, only play up to the wrap point */
594 avail_bytes
= playing
->size
- playing
->start
;
596 /* The ring buffer is not wrapped, can play the lot */
597 avail_bytes
= playing
->used
;
598 avail_frames
= avail_bytes
/ bpf
;
599 /* Only play up to the requested amount */
600 if(avail_frames
> frames
)
601 avail_frames
= frames
;
605 switch(config
->speaker_backend
) {
608 snd_pcm_sframes_t pcm_written_frames
;
611 pcm_written_frames
= snd_pcm_writei(pcm
,
612 playing
->buffer
+ playing
->start
,
614 D(("actually play %zu frames, wrote %d",
615 avail_frames
, (int)pcm_written_frames
));
616 if(pcm_written_frames
< 0) {
617 switch(pcm_written_frames
) {
618 case -EPIPE
: /* underrun */
619 error(0, "snd_pcm_writei reports underrun");
620 if((err
= snd_pcm_prepare(pcm
)) < 0)
621 fatal(0, "error calling snd_pcm_prepare: %d", err
);
626 fatal(0, "error calling snd_pcm_writei: %d",
627 (int)pcm_written_frames
);
630 written_frames
= pcm_written_frames
;
631 written_bytes
= written_frames
* bpf
;
635 case BACKEND_COMMAND
:
636 if(avail_bytes
> frames
* bpf
)
637 avail_bytes
= frames
* bpf
;
638 written_bytes
= write(cmdfd
, playing
->buffer
+ playing
->start
,
640 D(("actually play %zu bytes, wrote %d",
641 avail_bytes
, (int)written_bytes
));
642 if(written_bytes
< 0) {
645 error(0, "hmm, command died; trying another");
652 written_frames
= written_bytes
/ bpf
; /* good enough */
654 case BACKEND_NETWORK
:
655 /* We transmit using RTP (RFC3550) and attempt to conform to the internet
656 * AVT profile (RFC3551). */
659 /* There may have been a gap. Fix up the RTP time accordingly. */
662 uint64_t target_rtp_time
;
664 /* Find the current time */
665 xgettimeofday(&now
, 0);
666 /* Find the number of microseconds elapsed since rtp_time=0 */
667 delta
= tvsub_us(now
, rtp_time_0
);
668 assert(delta
<= UINT64_MAX
/ 88200);
669 target_rtp_time
= (delta
* playing
->format
.rate
670 * playing
->format
.channels
) / 1000000;
671 /* Overflows at ~6 years uptime with 44100Hz stereo */
673 /* rtp_time is the number of samples we've played. NB that we play
674 * RTP_AHEAD_MS ahead of ourselves, so it may legitimately be ahead of
675 * the value we deduce from time comparison.
677 * Suppose we have 1s track started at t=0, and another track begins to
678 * play at t=2s. Suppose RTP_AHEAD_MS=1000 and 44100Hz stereo. In that
679 * case we'll send 1s of audio as fast as we can, giving rtp_time=88200.
680 * rtp_time stops at this point.
682 * At t=2s we'll have calculated target_rtp_time=176400. In this case we
683 * set rtp_time=176400 and the player can correctly conclude that it
684 * should leave 1s between the tracks.
686 * Suppose instead that the second track arrives at t=0.5s, and that
687 * we've managed to transmit the whole of the first track already. We'll
688 * have target_rtp_time=44100.
690 * The desired behaviour is to play the second track back to back with
691 * first. In this case therefore we do not modify rtp_time.
693 * Is it ever right to reduce rtp_time? No; for that would imply
694 * transmitting packets with overlapping timestamp ranges, which does not
697 if(target_rtp_time
> rtp_time
) {
698 /* More time has elapsed than we've transmitted samples. That implies
699 * we've been 'sending' silence. */
700 info("advancing rtp_time by %"PRIu64
" samples",
701 target_rtp_time
- rtp_time
);
702 rtp_time
= target_rtp_time
;
703 } else if(target_rtp_time
< rtp_time
) {
704 const int64_t samples_ahead
= ((uint64_t)RTP_AHEAD_MS
705 * config
->sample_format
.rate
706 * config
->sample_format
.channels
709 if(target_rtp_time
+ samples_ahead
< rtp_time
) {
710 info("reversing rtp_time by %"PRIu64
" samples",
711 rtp_time
- target_rtp_time
);
715 header
.vpxcc
= 2 << 6; /* V=2, P=0, X=0, CC=0 */
716 header
.seq
= htons(rtp_seq
++);
717 header
.timestamp
= htonl((uint32_t)rtp_time
);
718 header
.ssrc
= rtp_id
;
719 header
.mpt
= (idled ?
0x80 : 0x00) | 10;
720 /* 10 = L16 = 16-bit x 2 x 44100KHz. We ought to deduce this value from
721 * the sample rate (in a library somewhere so that configuration.c can rule
722 * out invalid rates).
725 if(avail_bytes
> NETWORK_BYTES
- sizeof header
) {
726 avail_bytes
= NETWORK_BYTES
- sizeof header
;
727 /* Always send a whole number of frames */
728 avail_bytes
-= avail_bytes
% bpf
;
730 /* "The RTP clock rate used for generating the RTP timestamp is independent
731 * of the number of channels and the encoding; it equals the number of
732 * sampling periods per second. For N-channel encodings, each sampling
733 * period (say, 1/8000 of a second) generates N samples. (This terminology
734 * is standard, but somewhat confusing, as the total number of samples
735 * generated per second is then the sampling rate times the channel
738 write_bytes
= avail_bytes
;
740 vec
[0].iov_base
= (void *)&header
;
741 vec
[0].iov_len
= sizeof header
;
742 vec
[1].iov_base
= playing
->buffer
+ playing
->start
;
743 vec
[1].iov_len
= avail_bytes
;
745 written_bytes
= writev(bfd
,
748 } while(written_bytes
< 0 && errno
== EINTR
);
749 if(written_bytes
< 0) {
750 error(errno
, "error transmitting audio data");
752 if(audio_errors
== 10)
753 fatal(0, "too many audio errors");
758 written_bytes
= avail_bytes
;
759 written_frames
= written_bytes
/ bpf
;
760 /* Advance RTP's notion of the time */
761 rtp_time
+= written_frames
* playing
->format
.channels
;
766 /* written_bytes and written_frames had better both be set and correct by
768 playing
->start
+= written_bytes
;
769 playing
->used
-= written_bytes
;
770 playing
->played
+= written_frames
;
771 /* If the pointer is at the end of the buffer (or the buffer is completely
772 * empty) wrap it back to the start. */
773 if(!playing
->used
|| playing
->start
== playing
->size
)
775 frames
-= written_frames
;
778 /* Notify the server what we're up to. */
779 static void report(void) {
780 struct speaker_message sm
;
782 if(playing
&& playing
->buffer
!= (void *)&playing
->format
) {
783 memset(&sm
, 0, sizeof sm
);
784 sm
.type
= paused ? SM_PAUSED
: SM_PLAYING
;
785 strcpy(sm
.id
, playing
->id
);
786 sm
.data
= playing
->played
/ playing
->format
.rate
;
787 speaker_send(1, &sm
, 0);
792 static void reap(int __attribute__((unused
)) sig
) {
797 cmdpid
= waitpid(-1, &st
, WNOHANG
);
799 signal(SIGCHLD
, reap
);
802 static int addfd(int fd
, int events
) {
805 fds
[fdno
].events
= events
;
812 /** @brief ALSA backend initialization */
813 static void alsa_init(void) {
814 info("selected ALSA backend");
817 /** @brief ALSA backend activation */
818 static int alsa_activate(void) {
819 /* If we need to change format then close the current device. */
820 if(pcm
&& !formats_equal(&playing
->format
, &pcm_format
))
823 snd_pcm_hw_params_t
*hwparams
;
824 snd_pcm_sw_params_t
*swparams
;
825 snd_pcm_uframes_t pcm_bufsize
;
827 int sample_format
= 0;
831 if((err
= snd_pcm_open(&pcm
,
833 SND_PCM_STREAM_PLAYBACK
,
834 SND_PCM_NONBLOCK
))) {
835 error(0, "error from snd_pcm_open: %d", err
);
838 snd_pcm_hw_params_alloca(&hwparams
);
839 D(("set up hw params"));
840 if((err
= snd_pcm_hw_params_any(pcm
, hwparams
)) < 0)
841 fatal(0, "error from snd_pcm_hw_params_any: %d", err
);
842 if((err
= snd_pcm_hw_params_set_access(pcm
, hwparams
,
843 SND_PCM_ACCESS_RW_INTERLEAVED
)) < 0)
844 fatal(0, "error from snd_pcm_hw_params_set_access: %d", err
);
845 switch(playing
->format
.bits
) {
847 sample_format
= SND_PCM_FORMAT_S8
;
850 switch(playing
->format
.byte_format
) {
851 case AO_FMT_NATIVE
: sample_format
= SND_PCM_FORMAT_S16
; break;
852 case AO_FMT_LITTLE
: sample_format
= SND_PCM_FORMAT_S16_LE
; break;
853 case AO_FMT_BIG
: sample_format
= SND_PCM_FORMAT_S16_BE
; break;
854 error(0, "unrecognized byte format %d", playing
->format
.byte_format
);
859 error(0, "unsupported sample size %d", playing
->format
.bits
);
862 if((err
= snd_pcm_hw_params_set_format(pcm
, hwparams
,
863 sample_format
)) < 0) {
864 error(0, "error from snd_pcm_hw_params_set_format (%d): %d",
868 rate
= playing
->format
.rate
;
869 if((err
= snd_pcm_hw_params_set_rate_near(pcm
, hwparams
, &rate
, 0)) < 0) {
870 error(0, "error from snd_pcm_hw_params_set_rate (%d): %d",
871 playing
->format
.rate
, err
);
874 if(rate
!= (unsigned)playing
->format
.rate
)
875 info("want rate %d, got %u", playing
->format
.rate
, rate
);
876 if((err
= snd_pcm_hw_params_set_channels(pcm
, hwparams
,
877 playing
->format
.channels
)) < 0) {
878 error(0, "error from snd_pcm_hw_params_set_channels (%d): %d",
879 playing
->format
.channels
, err
);
882 bufsize
= 3 * FRAMES
;
883 pcm_bufsize
= bufsize
;
884 if((err
= snd_pcm_hw_params_set_buffer_size_near(pcm
, hwparams
,
886 fatal(0, "error from snd_pcm_hw_params_set_buffer_size (%d): %d",
888 if(pcm_bufsize
!= 3 * FRAMES
&& pcm_bufsize
!= last_pcm_bufsize
)
889 info("asked for PCM buffer of %d frames, got %d",
890 3 * FRAMES
, (int)pcm_bufsize
);
891 last_pcm_bufsize
= pcm_bufsize
;
892 if((err
= snd_pcm_hw_params(pcm
, hwparams
)) < 0)
893 fatal(0, "error calling snd_pcm_hw_params: %d", err
);
894 D(("set up sw params"));
895 snd_pcm_sw_params_alloca(&swparams
);
896 if((err
= snd_pcm_sw_params_current(pcm
, swparams
)) < 0)
897 fatal(0, "error calling snd_pcm_sw_params_current: %d", err
);
898 if((err
= snd_pcm_sw_params_set_avail_min(pcm
, swparams
, FRAMES
)) < 0)
899 fatal(0, "error calling snd_pcm_sw_params_set_avail_min %d: %d",
901 if((err
= snd_pcm_sw_params(pcm
, swparams
)) < 0)
902 fatal(0, "error calling snd_pcm_sw_params: %d", err
);
903 pcm_format
= playing
->format
;
904 bpf
= bytes_per_frame(&pcm_format
);
905 D(("acquired audio device"));
906 log_params(hwparams
, swparams
);
913 /* We assume the error is temporary and that we'll retry in a bit. */
921 /** @brief Play via ALSA */
922 static size_t alsa_play(size_t frames
) {
926 /** @brief ALSA deactivation */
927 static void alsa_deactivate(void) {
931 if((err
= snd_pcm_nonblock(pcm
, 0)) < 0)
932 fatal(0, "error calling snd_pcm_nonblock: %d", err
);
939 D(("released audio device"));
944 /** @brief Command backend initialization */
945 static void command_init(void) {
946 info("selected command backend");
950 /** @brief Play to a subprocess */
951 static size_t command_play(size_t frames
) {
955 /** @brief Command/network backend activation */
956 static int generic_activate(void) {
958 bufsize
= 3 * FRAMES
;
959 bpf
= bytes_per_frame(&config
->sample_format
);
960 D(("acquired audio device"));
966 /** @brief Network backend initialization */
967 static void network_init(void) {
968 struct addrinfo
*res
, *sres
;
969 static const struct addrinfo pref
= {
979 static const struct addrinfo prefbind
= {
989 static const int one
= 1;
990 int sndbuf
, target_sndbuf
= 131072;
992 char *sockname
, *ssockname
;
994 res
= get_address(&config
->broadcast
, &pref
, &sockname
);
996 if(config
->broadcast_from
.n
) {
997 sres
= get_address(&config
->broadcast_from
, &prefbind
, &ssockname
);
1001 if((bfd
= socket(res
->ai_family
,
1003 res
->ai_protocol
)) < 0)
1004 fatal(errno
, "error creating broadcast socket");
1005 if(setsockopt(bfd
, SOL_SOCKET
, SO_BROADCAST
, &one
, sizeof one
) < 0)
1006 fatal(errno
, "error setting SO_BROADCAST on broadcast socket");
1007 len
= sizeof sndbuf
;
1008 if(getsockopt(bfd
, SOL_SOCKET
, SO_SNDBUF
,
1010 fatal(errno
, "error getting SO_SNDBUF");
1011 if(target_sndbuf
> sndbuf
) {
1012 if(setsockopt(bfd
, SOL_SOCKET
, SO_SNDBUF
,
1013 &target_sndbuf
, sizeof target_sndbuf
) < 0)
1014 error(errno
, "error setting SO_SNDBUF to %d", target_sndbuf
);
1016 info("changed socket send buffer size from %d to %d",
1017 sndbuf
, target_sndbuf
);
1019 info("default socket send buffer is %d",
1021 /* We might well want to set additional broadcast- or multicast-related
1023 if(sres
&& bind(bfd
, sres
->ai_addr
, sres
->ai_addrlen
) < 0)
1024 fatal(errno
, "error binding broadcast socket to %s", ssockname
);
1025 if(connect(bfd
, res
->ai_addr
, res
->ai_addrlen
) < 0)
1026 fatal(errno
, "error connecting broadcast socket to %s", sockname
);
1027 /* Select an SSRC */
1028 gcry_randomize(&rtp_id
, sizeof rtp_id
, GCRY_STRONG_RANDOM
);
1029 info("selected network backend, sending to %s", sockname
);
1030 if(config
->sample_format
.byte_format
!= AO_FMT_BIG
) {
1031 info("forcing big-endian sample format");
1032 config
->sample_format
.byte_format
= AO_FMT_BIG
;
1036 /** @brief Play over the network */
1037 static size_t network_play(size_t frames
) {
1041 /** @brief Table of speaker backends */
1042 static const struct speaker_backend backends
[] = {
1069 { -1, 0, 0, 0, 0, 0 }
1072 int main(int argc
, char **argv
) {
1073 int n
, fd
, stdin_slot
, alsa_slots
, cmdfd_slot
, bfd_slot
, poke
, timeout
;
1075 struct speaker_message sm
;
1077 int alsa_nslots
= -1, err
;
1081 if(!setlocale(LC_CTYPE
, "")) fatal(errno
, "error calling setlocale");
1082 while((n
= getopt_long(argc
, argv
, "hVc:dD", options
, 0)) >= 0) {
1085 case 'V': version();
1086 case 'c': configfile
= optarg
; break;
1087 case 'd': debugging
= 1; break;
1088 case 'D': debugging
= 0; break;
1089 default: fatal(0, "invalid option");
1092 if(getenv("DISORDER_DEBUG_SPEAKER")) debugging
= 1;
1093 /* If stderr is a TTY then log there, otherwise to syslog. */
1095 openlog(progname
, LOG_PID
, LOG_DAEMON
);
1096 log_default
= &log_syslog
;
1098 if(config_read()) fatal(0, "cannot read configuration");
1099 /* ignore SIGPIPE */
1100 signal(SIGPIPE
, SIG_IGN
);
1102 signal(SIGCHLD
, reap
);
1103 /* set nice value */
1104 xnice(config
->nice_speaker
);
1107 /* make sure we're not root, whatever the config says */
1108 if(getuid() == 0 || geteuid() == 0) fatal(0, "do not run as root");
1109 /* identify the backend used to play */
1110 for(n
= 0; backends
[n
].backend
!= -1; ++n
)
1111 if(backends
[n
].backend
== config
->speaker_backend
)
1113 if(backends
[n
].backend
== -1)
1114 fatal(0, "unsupported backend %d", config
->speaker_backend
);
1115 backend
= &backends
[n
];
1116 /* backend-specific initialization */
1118 while(getppid() != 1) {
1120 /* Always ready for commands from the main server. */
1121 stdin_slot
= addfd(0, POLLIN
);
1122 /* Try to read sample data for the currently playing track if there is
1124 if(playing
&& !playing
->eof
&& playing
->used
< playing
->size
) {
1125 playing
->slot
= addfd(playing
->fd
, POLLIN
);
1128 /* If forceplay is set then wait until it succeeds before waiting on the
1133 /* By default we will wait up to a second before thinking about current
1136 if(ready
&& !forceplay
) {
1137 switch(config
->speaker_backend
) {
1138 case BACKEND_COMMAND
:
1139 /* We send sample data to the subprocess as fast as it can accept it.
1140 * This isn't ideal as pause latency can be very high as a result. */
1142 cmdfd_slot
= addfd(cmdfd
, POLLOUT
);
1144 case BACKEND_NETWORK
: {
1147 uint64_t target_rtp_time
;
1148 const int64_t samples_ahead
= ((uint64_t)RTP_AHEAD_MS
1149 * config
->sample_format
.rate
1150 * config
->sample_format
.channels
1153 static unsigned logit
;
1156 /* If we're starting then initialize the base time */
1158 xgettimeofday(&rtp_time_0
, 0);
1159 /* We send audio data whenever we get RTP_AHEAD seconds or more
1161 xgettimeofday(&now
, 0);
1162 target_us
= tvsub_us(now
, rtp_time_0
);
1163 assert(target_us
<= UINT64_MAX
/ 88200);
1164 target_rtp_time
= (target_us
* config
->sample_format
.rate
1165 * config
->sample_format
.channels
)
1169 /* TODO remove logging guff */
1170 if(!(logit
++ & 1023))
1171 info("rtp_time %llu target %llu difference %lld [%lld]",
1172 rtp_time
, target_rtp_time
,
1173 rtp_time
- target_rtp_time
,
1176 if((int64_t)(rtp_time
- target_rtp_time
) < samples_ahead
)
1177 bfd_slot
= addfd(bfd
, POLLOUT
);
1181 case BACKEND_ALSA
: {
1182 /* We send sample data to ALSA as fast as it can accept it, relying on
1183 * the fact that it has a relatively small buffer to minimize pause
1190 alsa_nslots
= snd_pcm_poll_descriptors(pcm
, &fds
[fdno
], NFDS
- fdno
);
1191 if((alsa_nslots
<= 0
1192 || !(fds
[alsa_slots
].events
& POLLOUT
))
1193 && snd_pcm_state(pcm
) == SND_PCM_STATE_XRUN
) {
1194 error(0, "underrun detected after call to snd_pcm_poll_descriptors()");
1195 if((err
= snd_pcm_prepare(pcm
)))
1196 fatal(0, "error calling snd_pcm_prepare: %d", err
);
1199 } while(retry
-- > 0);
1200 if(alsa_nslots
>= 0)
1201 fdno
+= alsa_nslots
;
1206 assert(!"unknown backend");
1209 /* If any other tracks don't have a full buffer, try to read sample data
1211 for(t
= tracks
; t
; t
= t
->next
)
1213 if(!t
->eof
&& t
->used
< t
->size
) {
1214 t
->slot
= addfd(t
->fd
, POLLIN
| POLLHUP
);
1218 /* Wait for something interesting to happen */
1219 n
= poll(fds
, fdno
, timeout
);
1221 if(errno
== EINTR
) continue;
1222 fatal(errno
, "error calling poll");
1224 /* Play some sound before doing anything else */
1226 switch(config
->speaker_backend
) {
1229 if(alsa_slots
!= -1) {
1230 unsigned short alsa_revents
;
1232 if((err
= snd_pcm_poll_descriptors_revents(pcm
,
1235 &alsa_revents
)) < 0)
1236 fatal(0, "error calling snd_pcm_poll_descriptors_revents: %d", err
);
1237 if(alsa_revents
& (POLLOUT
| POLLERR
))
1243 case BACKEND_COMMAND
:
1244 if(cmdfd_slot
!= -1) {
1245 if(fds
[cmdfd_slot
].revents
& (POLLOUT
| POLLERR
))
1250 case BACKEND_NETWORK
:
1251 if(bfd_slot
!= -1) {
1252 if(fds
[bfd_slot
].revents
& (POLLOUT
| POLLERR
))
1259 /* Some attempt to play must have failed */
1260 if(playing
&& !paused
)
1263 forceplay
= 0; /* just in case */
1265 /* Perhaps we have a command to process */
1266 if(fds
[stdin_slot
].revents
& POLLIN
) {
1267 n
= speaker_recv(0, &sm
, &fd
);
1271 D(("SM_PREPARE %s %d", sm
.id
, fd
));
1272 if(fd
== -1) fatal(0, "got SM_PREPARE but no file descriptor");
1273 t
= findtrack(sm
.id
, 1);
1277 D(("SM_PLAY %s %d", sm
.id
, fd
));
1278 if(playing
) fatal(0, "got SM_PLAY but already playing something");
1279 t
= findtrack(sm
.id
, 1);
1280 if(fd
!= -1) acquire(t
, fd
);
1300 D(("SM_CANCEL %s", sm
.id
));
1301 t
= removetrack(sm
.id
);
1304 sm
.type
= SM_FINISHED
;
1305 strcpy(sm
.id
, playing
->id
);
1306 speaker_send(1, &sm
, 0);
1311 error(0, "SM_CANCEL for unknown track %s", sm
.id
);
1316 if(config_read()) error(0, "cannot read configuration");
1317 info("reloaded configuration");
1320 error(0, "unknown message type %d", sm
.type
);
1323 /* Read in any buffered data */
1324 for(t
= tracks
; t
; t
= t
->next
)
1325 if(t
->slot
!= -1 && (fds
[t
->slot
].revents
& (POLLIN
| POLLHUP
)))
1327 /* We might be able to play now */
1328 if(ready
&& forceplay
&& playing
&& !paused
)
1330 /* Maybe we finished playing a track somewhere in the above */
1332 /* If we don't need the sound device for now then close it for the benefit
1333 * of anyone else who wants it. */
1334 if((!playing
|| paused
) && ready
)
1336 /* If we've not reported out state for a second do so now. */
1337 if(time(0) > last_report
)
1340 info("stopped (parent terminated)");
1349 indent-tabs-mode:nil