/*
* This file is part of DisOrder
* Copyright (C) 2005-2009 Richard Kettlewell
* Portions (C) 2007 Mark Wooding
*
* This program is free software: you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation, either version 3 of the License, or
* (at your option) any later version.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program. If not, see .
*/
/** @file server/speaker.c
* @brief Speaker process
*
* This program is responsible for transmitting a single coherent audio stream
* to its destination (over the network, to some sound API, to some
* subprocess). It receives connections from decoders (or rather from the
* process that is about to become disorder-normalize) and plays them in the
* right order.
*
* @b Model. mainloop() implements a select loop awaiting commands from the
* main server, new connections to the speaker socket, and audio data on those
* connections. Each connection starts with a queue ID (with a 32-bit
* native-endian length word), allowing it to be referred to in commands from
* the server.
*
* Data read on connections is buffered, up to a limit (currently 1Mbyte per
* track). No attempt is made here to limit the number of tracks, it is
* assumed that the main server won't start outrageously many decoders.
*
* Audio is supplied from this buffer to the uaudio play callback. Playback is
* enabled when a track is to be played and disabled when the its last bytes
* have been return by the callback; pause and resume is implemneted the
* obvious way. If the callback finds itself required to play when there is no
* playing track it returns dead air.
*
* To implement gapless playback, the server is notified that a track has
* finished slightly early. @ref SM_PLAY is therefore allowed to arrive while
* the previous track is still playing provided an early @ref SM_FINISHED has
* been sent for it.
*
* @b Encodings. The encodings supported depend entirely on the uaudio backend
* chosen. See @ref uaudio.h, etc.
*
* Inbound data is expected to match @c config->sample_format. In normal use
* this is arranged by the @c disorder-normalize program (see @ref
* server/normalize.c).
*
* @b Garbage @b Collection. This program deliberately does not use the
* garbage collector even though it might be convenient to do so. This is for
* two reasons. Firstly some sound APIs use thread threads and we do not want
* to have to deal with potential interactions between threading and garbage
* collection. Secondly this process needs to be able to respond quickly and
* this is not compatible with the collector hanging the program even
* relatively briefly.
*
* @b Units. This program thinks at various times in three different units.
* Bytes are obvious. A sample is a single sample on a single channel. A
* frame is several samples on different channels at the same point in time.
* So (for instance) a 16-bit stereo frame is 4 bytes and consists of a pair of
* 2-byte samples.
*/
#include "common.h"
#include
#include
#include
#include
#include
#include
#include
#include
#include
#include
#include
#include
#include
#include
#include
#include
#include "configuration.h"
#include "syscalls.h"
#include "log.h"
#include "defs.h"
#include "mem.h"
#include "speaker-protocol.h"
#include "user.h"
#include "printf.h"
#include "version.h"
#include "uaudio.h"
/** @brief Maximum number of FDs to poll for */
#define NFDS 1024
/** @brief Number of bytes before end of track to send SM_FINISHED
*
* Generally set to 1 second.
*/
static size_t early_finish;
/** @brief Track structure
*
* Known tracks are kept in a linked list. Usually there will be at most two
* of these but rearranging the queue can cause there to be more.
*/
struct track {
/** @brief Next track */
struct track *next;
/** @brief Input file descriptor */
int fd; /* input FD */
/** @brief Track ID */
char id[24];
/** @brief Start position of data in buffer */
size_t start;
/** @brief Number of bytes of data in buffer */
size_t used;
/** @brief Set @c fd is at EOF */
int eof;
/** @brief Total number of samples played */
unsigned long long played;
/** @brief Slot in @ref fds */
int slot;
/** @brief Set when playable
*
* A track becomes playable whenever it fills its buffer or reaches EOF; it
* stops being playable when it entirely empties its buffer. Tracks start
* out life not playable.
*/
int playable;
/** @brief Set when finished
*
* This is set when we've notified the server that the track is finished.
* Once this has happened (typically very late in the track's lifetime) the
* track cannot be paused or cancelled.
*/
int finished;
/** @brief Input buffer
*
* 1Mbyte is enough for nearly 6s of 44100Hz 16-bit stereo
*/
char buffer[1048576];
};
/** @brief Lock protecting data structures
*
* This lock protects values shared between the main thread and the callback.
*
* It is held 'all' the time by the main thread, the exceptions being when
* called activate/deactivate callbacks and when calling (potentially) slow
* system calls (in particular poll(), where in fact the main thread will spend
* most of its time blocked).
*
* The callback holds it when it's running.
*/
static pthread_mutex_t lock = PTHREAD_MUTEX_INITIALIZER;
/** @brief Linked list of all prepared tracks */
static struct track *tracks;
/** @brief Playing track, or NULL
*
* This means the track the speaker process intends to play. It does not
* reflect any other state (e.g. activation of uaudio backend).
*/
static struct track *playing;
/** @brief Pending playing track, or NULL
*
* This means the track the server wants the speaker to play.
*/
static struct track *pending_playing;
/** @brief Array of file descriptors for poll() */
static struct pollfd fds[NFDS];
/** @brief Next free slot in @ref fds */
static int fdno;
/** @brief Listen socket */
static int listenfd;
/** @brief Timestamp of last potential report to server */
static time_t last_report;
/** @brief Set when paused */
static int paused;
/** @brief Set when back end activated */
static int activated;
/** @brief Signal pipe back into the poll() loop */
static int sigpipe[2];
/** @brief Selected backend */
static const struct uaudio *backend;
static const struct option options[] = {
{ "help", no_argument, 0, 'h' },
{ "version", no_argument, 0, 'V' },
{ "config", required_argument, 0, 'c' },
{ "debug", no_argument, 0, 'd' },
{ "no-debug", no_argument, 0, 'D' },
{ "syslog", no_argument, 0, 's' },
{ "no-syslog", no_argument, 0, 'S' },
{ 0, 0, 0, 0 }
};
/* Display usage message and terminate. */
static void help(void) {
xprintf("Usage:\n"
" disorder-speaker [OPTIONS]\n"
"Options:\n"
" --help, -h Display usage message\n"
" --version, -V Display version number\n"
" --config PATH, -c PATH Set configuration file\n"
" --debug, -d Turn on debugging\n"
" --[no-]syslog Force logging\n"
"\n"
"Speaker process for DisOrder. Not intended to be run\n"
"directly.\n");
xfclose(stdout);
exit(0);
}
/** @brief Find track @p id, maybe creating it if not found
* @param id Track ID to find
* @param create If nonzero, create track structure of @p id not found
* @return Pointer to track structure or NULL
*/
static struct track *findtrack(const char *id, int create) {
struct track *t;
D(("findtrack %s %d", id, create));
for(t = tracks; t && strcmp(id, t->id); t = t->next)
;
if(!t && create) {
t = xmalloc(sizeof *t);
t->next = tracks;
strcpy(t->id, id);
t->fd = -1;
tracks = t;
}
return t;
}
/** @brief Remove track @p id (but do not destroy it)
* @param id Track ID to remove
* @return Track structure or NULL if not found
*/
static struct track *removetrack(const char *id) {
struct track *t, **tt;
D(("removetrack %s", id));
for(tt = &tracks; (t = *tt) && strcmp(id, t->id); tt = &t->next)
;
if(t)
*tt = t->next;
return t;
}
/** @brief Destroy a track
* @param t Track structure
*/
static void destroy(struct track *t) {
D(("destroy %s", t->id));
if(t->fd != -1)
xclose(t->fd);
free(t);
}
/** @brief Read data into a sample buffer
* @param t Pointer to track
* @return 0 on success, -1 on EOF
*
* This is effectively the read callback on @c t->fd. It is called from the
* main loop whenever the track's file descriptor is readable, assuming the
* buffer has not reached the maximum allowed occupancy.
*/
static int speaker_fill(struct track *t) {
size_t where, left;
int n, rc;
D(("fill %s: eof=%d used=%zu",
t->id, t->eof, t->used));
if(t->eof)
return -1;
if(t->used < sizeof t->buffer) {
/* there is room left in the buffer */
where = (t->start + t->used) % sizeof t->buffer;
/* Get as much data as we can */
if(where >= t->start)
left = (sizeof t->buffer) - where;
else
left = t->start - where;
pthread_mutex_unlock(&lock);
do {
n = read(t->fd, t->buffer + where, left);
} while(n < 0 && errno == EINTR);
pthread_mutex_lock(&lock);
if(n < 0) {
if(errno != EAGAIN)
disorder_fatal(errno, "error reading sample stream");
rc = 0;
} else if(n == 0) {
D(("fill %s: eof detected", t->id));
t->eof = 1;
/* A track always becomes playable at EOF; we're not going to see any
* more data. */
t->playable = 1;
rc = -1;
} else {
t->used += n;
/* A track becomes playable when it (first) fills its buffer. For
* 44.1KHz 16-bit stereo this is ~6s of audio data. The latency will
* depend how long that takes to decode (hopefuly not very!) */
if(t->used == sizeof t->buffer)
t->playable = 1;
rc = 0;
}
}
return rc;
}
/** @brief Return nonzero if we want to play some audio
*
* We want to play audio if there is a current track; and it is not paused; and
* it is playable according to the rules for @ref track::playable.
*
* We don't allow tracks to be paused if we've already told the server we've
* finished them; that would cause such tracks to survive much longer than the
* few samples they're supposed to, with report() remaining silent for the
* duration.
*/
static int playable(void) {
return playing
&& (!paused || playing->finished)
&& playing->playable;
}
/** @brief Notify the server what we're up to */
static void report(void) {
struct speaker_message sm;
if(playing) {
/* Had better not send a report for a track that the server thinks has
* finished, that would be confusing. */
if(playing->finished)
return;
memset(&sm, 0, sizeof sm);
sm.type = paused ? SM_PAUSED : SM_PLAYING;
strcpy(sm.id, playing->id);
sm.data = playing->played / (uaudio_rate * uaudio_channels);
speaker_send(1, &sm);
xtime(&last_report);
}
}
/** @brief Add a file descriptor to the set to poll() for
* @param fd File descriptor
* @param events Events to wait for e.g. @c POLLIN
* @return Slot number
*/
static int addfd(int fd, int events) {
if(fdno < NFDS) {
fds[fdno].fd = fd;
fds[fdno].events = events;
return fdno++;
} else
return -1;
}
/** @brief Callback to return some sampled data
* @param buffer Where to put sample data
* @param max_samples How many samples to return
* @param userdata User data
* @return Number of samples written
*
* See uaudio_callback().
*/
static size_t speaker_callback(void *buffer,
size_t max_samples,
void attribute((unused)) *userdata) {
const size_t max_bytes = max_samples * uaudio_sample_size;
size_t provided_samples = 0;
pthread_mutex_lock(&lock);
/* TODO perhaps we should immediately go silent if we've been asked to pause
* or cancel the playing track (maybe block in the cancel case and see what
* else turns up?) */
if(playing) {
if(playing->used > 0) {
size_t bytes;
/* Compute size of largest contiguous chunk. We get called as often as
* necessary so there's no need for cleverness here. */
if(playing->start + playing->used > sizeof playing->buffer)
bytes = sizeof playing->buffer - playing->start;
else
bytes = playing->used;
/* Limit to what we were asked for */
if(bytes > max_bytes)
bytes = max_bytes;
/* Provide it */
memcpy(buffer, playing->buffer + playing->start, bytes);
playing->start += bytes;
playing->used -= bytes;
/* Wrap around to start of buffer */
if(playing->start == sizeof playing->buffer)
playing->start = 0;
/* See if we've reached the end of the track */
if(playing->used == 0 && playing->eof) {
int ignored = write(sigpipe[1], "", 1);
(void) ignored;
}
provided_samples = bytes / uaudio_sample_size;
playing->played += provided_samples;
}
}
/* If we couldn't provide anything at all, play dead air */
/* TODO maybe it would be better to block, in some cases? */
if(!provided_samples) {
memset(buffer, 0, max_bytes);
provided_samples = max_samples;
if(playing)
disorder_info("%zu samples silence, playing->used=%zu",
provided_samples, playing->used);
else
disorder_info("%zu samples silence, playing=NULL", provided_samples);
}
pthread_mutex_unlock(&lock);
return provided_samples;
}
/** @brief Main event loop */
static void mainloop(void) {
struct track *t;
struct speaker_message sm;
int n, fd, stdin_slot, timeout, listen_slot, sigpipe_slot;
/* Keep going while our parent process is alive */
pthread_mutex_lock(&lock);
while(getppid() != 1) {
int force_report = 0;
fdno = 0;
/* By default we will wait up to half a second before thinking about
* current state. */
timeout = 500;
/* Always ready for commands from the main server. */
stdin_slot = addfd(0, POLLIN);
/* Also always ready for inbound connections */
listen_slot = addfd(listenfd, POLLIN);
/* Try to read sample data for the currently playing track if there is
* buffer space. */
if(playing
&& playing->fd >= 0
&& !playing->eof
&& playing->used < (sizeof playing->buffer))
playing->slot = addfd(playing->fd, POLLIN);
else if(playing)
playing->slot = -1;
/* If any other tracks don't have a full buffer, try to read sample data
* from them. We do this last of all, so that if we run out of slots,
* nothing important can't be monitored. */
for(t = tracks; t; t = t->next)
if(t != playing) {
if(t->fd >= 0
&& !t->eof
&& t->used < sizeof t->buffer) {
t->slot = addfd(t->fd, POLLIN | POLLHUP);
} else
t->slot = -1;
}
sigpipe_slot = addfd(sigpipe[0], POLLIN);
/* Wait for something interesting to happen */
pthread_mutex_unlock(&lock);
n = poll(fds, fdno, timeout);
pthread_mutex_lock(&lock);
if(n < 0) {
if(errno == EINTR) continue;
disorder_fatal(errno, "error calling poll");
}
/* Perhaps a connection has arrived */
if(fds[listen_slot].revents & POLLIN) {
struct sockaddr_un addr;
socklen_t addrlen = sizeof addr;
uint32_t l;
char id[24];
if((fd = accept(listenfd, (struct sockaddr *)&addr, &addrlen)) >= 0) {
blocking(fd);
if(read(fd, &l, sizeof l) < 4) {
disorder_error(errno, "reading length from inbound connection");
xclose(fd);
} else if(l >= sizeof id) {
disorder_error(0, "id length too long");
xclose(fd);
} else if(read(fd, id, l) < (ssize_t)l) {
disorder_error(errno, "reading id from inbound connection");
xclose(fd);
} else {
id[l] = 0;
D(("id %s fd %d", id, fd));
t = findtrack(id, 1/*create*/);
if (write(fd, "", 1) < 0) /* write an ack */
disorder_error(errno, "writing ack to inbound connection");
if(t->fd != -1) {
disorder_error(0, "%s: already got a connection", id);
xclose(fd);
} else {
nonblock(fd);
t->fd = fd; /* yay */
}
}
} else
disorder_error(errno, "accept");
}
/* Perhaps we have a command to process */
if(fds[stdin_slot].revents & POLLIN) {
/* There might (in theory) be several commands queued up, but in general
* this won't be the case, so we don't bother looping around to pick them
* all up. */
n = speaker_recv(0, &sm);
if(n > 0)
/* As a rule we don't send success replies to most commands - we just
* force the regular status update to be sent immediately rather than
* on schedule. */
switch(sm.type) {
case SM_PLAY:
/* SM_PLAY is only allowed if the server reasonably believes that
* nothing is playing */
if(playing) {
/* If finished isn't set then the server can't believe that this
* track has finished */
if(!playing->finished)
disorder_fatal(0, "got SM_PLAY but already playing something");
/* If pending_playing is set then the server must believe that that
* is playing */
if(pending_playing)
disorder_fatal(0, "got SM_PLAY but have a pending playing track");
}
t = findtrack(sm.id, 1);
D(("SM_PLAY %s fd %d", t->id, t->fd));
if(t->fd == -1)
disorder_error(0,
"cannot play track because no connection arrived");
/* TODO as things stand we often report this error message but then
* appear to proceed successfully. Understanding why requires a look
* at play.c: we call prepare() which makes the connection in a child
* process, and then sends the SM_PLAY in the parent process. The
* latter may well be faster. As it happens this is harmless; we'll
* just sit around sending silence until the decoder connects and
* starts sending some sample data. But is is annoying and ought to
* be fixed. */
pending_playing = t;
/* If nothing is currently playing then we'll switch to the pending
* track below so there's no point distinguishing the situations
* here. */
break;
case SM_PAUSE:
D(("SM_PAUSE"));
paused = 1;
force_report = 1;
break;
case SM_RESUME:
D(("SM_RESUME"));
paused = 0;
force_report = 1;
break;
case SM_CANCEL:
D(("SM_CANCEL %s", sm.id));
t = removetrack(sm.id);
if(t) {
if(t == playing || t == pending_playing) {
/* Scratching the track that the server believes is playing,
* which might either be the actual playing track or a pending
* playing track */
sm.type = SM_FINISHED;
if(t == playing)
playing = 0;
else
pending_playing = 0;
} else {
/* Could be scratching the playing track before it's quite got
* going, or could be just removing a track from the queue. We
* log more because there's been a bug here recently than because
* it's particularly interesting; the log message will be removed
* if no further problems show up. */
disorder_info("SM_CANCEL for nonplaying track %s", sm.id);
sm.type = SM_STILLBORN;
}
strcpy(sm.id, t->id);
destroy(t);
} else {
/* Probably scratching the playing track well before it's got
* going, but could indicate a bug, so we log this as an error. */
sm.type = SM_UNKNOWN;
disorder_error(0, "SM_CANCEL for unknown track %s", sm.id);
}
speaker_send(1, &sm);
force_report = 1;
break;
case SM_RELOAD:
D(("SM_RELOAD"));
if(config_read(1, NULL))
disorder_error(0, "cannot read configuration");
disorder_info("reloaded configuration");
break;
default:
disorder_error(0, "unknown message type %d", sm.type);
}
}
/* Read in any buffered data */
for(t = tracks; t; t = t->next)
if(t->fd != -1
&& t->slot != -1
&& (fds[t->slot].revents & (POLLIN | POLLHUP)))
speaker_fill(t);
/* Drain the signal pipe. We don't care about its contents, merely that it
* interrupted poll(). */
if(fds[sigpipe_slot].revents & POLLIN) {
char buffer[64];
int ignored; (void)ignored;
ignored = read(sigpipe[0], buffer, sizeof buffer);
}
/* Send SM_FINISHED when we're near the end of the track.
*
* This is how we implement gapless play; we hope that the SM_PLAY from the
* server arrives before the remaining bytes of the track play out.
*/
if(playing
&& playing->eof
&& !playing->finished
&& playing->used <= early_finish) {
memset(&sm, 0, sizeof sm);
sm.type = SM_FINISHED;
strcpy(sm.id, playing->id);
speaker_send(1, &sm);
playing->finished = 1;
}
/* When the track is actually finished, deconfigure it */
if(playing && playing->eof && !playing->used) {
removetrack(playing->id);
destroy(playing);
playing = 0;
}
/* Act on the pending SM_PLAY */
if(!playing && pending_playing) {
playing = pending_playing;
pending_playing = 0;
force_report = 1;
}
/* Impose any state change required by the above */
if(playable()) {
if(!activated) {
activated = 1;
pthread_mutex_unlock(&lock);
backend->activate();
pthread_mutex_lock(&lock);
}
} else {
if(activated) {
activated = 0;
pthread_mutex_unlock(&lock);
backend->deactivate();
pthread_mutex_lock(&lock);
}
}
/* If we've not reported our state for a second do so now. */
if(force_report || xtime(0) > last_report)
report();
}
}
int main(int argc, char **argv) {
int n, logsyslog = !isatty(2);
struct sockaddr_un addr;
static const int one = 1;
struct speaker_message sm;
const char *d;
char *dir;
struct rlimit rl[1];
set_progname(argv);
if(!setlocale(LC_CTYPE, "")) disorder_fatal(errno, "error calling setlocale");
while((n = getopt_long(argc, argv, "hVc:dDSs", options, 0)) >= 0) {
switch(n) {
case 'h': help();
case 'V': version("disorder-speaker");
case 'c': configfile = optarg; break;
case 'd': debugging = 1; break;
case 'D': debugging = 0; break;
case 'S': logsyslog = 0; break;
case 's': logsyslog = 1; break;
default: disorder_fatal(0, "invalid option");
}
}
if((d = getenv("DISORDER_DEBUG_SPEAKER"))) debugging = atoi(d);
if(logsyslog) {
openlog(progname, LOG_PID, LOG_DAEMON);
log_default = &log_syslog;
}
config_uaudio_apis = uaudio_apis;
if(config_read(1, NULL)) disorder_fatal(0, "cannot read configuration");
/* ignore SIGPIPE */
signal(SIGPIPE, SIG_IGN);
/* set nice value */
xnice(config->nice_speaker);
/* change user */
become_mortal();
/* make sure we're not root, whatever the config says */
if(getuid() == 0 || geteuid() == 0)
disorder_fatal(0, "do not run as root");
/* Make sure we can't have more than NFDS files open (it would bust our
* poll() array) */
if(getrlimit(RLIMIT_NOFILE, rl) < 0)
disorder_fatal(errno, "getrlimit RLIMIT_NOFILE");
if(rl->rlim_cur > NFDS) {
rl->rlim_cur = NFDS;
if(setrlimit(RLIMIT_NOFILE, rl) < 0)
disorder_fatal(errno, "setrlimit to reduce RLIMIT_NOFILE to %lu",
(unsigned long)rl->rlim_cur);
disorder_info("set RLIM_NOFILE to %lu", (unsigned long)rl->rlim_cur);
} else
disorder_info("RLIM_NOFILE is %lu", (unsigned long)rl->rlim_cur);
/* gcrypt initialization */
if(!gcry_check_version(NULL))
disorder_fatal(0, "gcry_check_version failed");
gcry_control(GCRYCTL_INIT_SECMEM, 0);
gcry_control (GCRYCTL_INITIALIZATION_FINISHED, 0);
/* create a pipe between the backend callback and the poll() loop */
xpipe(sigpipe);
nonblock(sigpipe[0]);
/* set up audio backend */
uaudio_set_format(config->sample_format.rate,
config->sample_format.channels,
config->sample_format.bits,
config->sample_format.bits != 8);
early_finish = uaudio_sample_size * uaudio_channels * uaudio_rate;
/* TODO other parameters! */
backend = uaudio_find(config->api);
/* backend-specific initialization */
if(backend->configure)
backend->configure();
backend->start(speaker_callback, NULL);
/* create the socket directory */
byte_xasprintf(&dir, "%s/speaker", config->home);
unlink(dir); /* might be a leftover socket */
if(mkdir(dir, 0700) < 0 && errno != EEXIST)
disorder_fatal(errno, "error creating %s", dir);
/* set up the listen socket */
listenfd = xsocket(PF_UNIX, SOCK_STREAM, 0);
memset(&addr, 0, sizeof addr);
addr.sun_family = AF_UNIX;
snprintf(addr.sun_path, sizeof addr.sun_path, "%s/speaker/socket",
config->home);
if(unlink(addr.sun_path) < 0 && errno != ENOENT)
disorder_error(errno, "removing %s", addr.sun_path);
xsetsockopt(listenfd, SOL_SOCKET, SO_REUSEADDR, &one, sizeof one);
if(bind(listenfd, (const struct sockaddr *)&addr, sizeof addr) < 0)
disorder_fatal(errno, "error binding socket to %s", addr.sun_path);
xlisten(listenfd, 128);
nonblock(listenfd);
disorder_info("listening on %s", addr.sun_path);
memset(&sm, 0, sizeof sm);
sm.type = SM_READY;
speaker_send(1, &sm);
mainloop();
disorder_info("stopped (parent terminated)");
exit(0);
}
/*
Local Variables:
c-basic-offset:2
comment-column:40
fill-column:79
indent-tabs-mode:nil
End:
*/