/*
* This file is part of DisOrder
- * Copyright (C) 2005, 2006, 2007 Richard Kettlewell
+ * Copyright (C) 2005-2009 Richard Kettlewell
+ * Portions (C) 2007 Mark Wooding
*
- * This program is free software; you can redistribute it and/or modify
+ * 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 2 of the License, or
+ * 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.
- *
+ *
+ * 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, write to the Free Software
- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307
- * USA
+ * along with this program. If not, see <http://www.gnu.org/licenses/>.
*/
/** @file server/speaker.c
* @brief Speaker process
* process that is about to become disorder-normalize) and plays them in the
* right order.
*
- * @b Encodings. For the <a href="http://www.alsa-project.org/">ALSA</a> API,
- * 8- and 16- bit stereo and mono are supported, with any sample rate (within
- * the limits that ALSA can deal with.)
+ * @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
* 2-byte samples.
*/
-#include <config.h>
-#include "types.h"
+#include "common.h"
#include <getopt.h>
-#include <stdio.h>
-#include <stdlib.h>
#include <locale.h>
#include <syslog.h>
#include <unistd.h>
#include <errno.h>
#include <ao/ao.h>
-#include <string.h>
-#include <assert.h>
#include <sys/select.h>
#include <sys/wait.h>
#include <time.h>
#include <fcntl.h>
#include <poll.h>
#include <sys/un.h>
+#include <sys/stat.h>
+#include <pthread.h>
+#include <sys/resource.h>
+#include <gcrypt.h>
#include "configuration.h"
#include "syscalls.h"
#include "mem.h"
#include "speaker-protocol.h"
#include "user.h"
-#include "speaker.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 */
-struct track *tracks;
+static struct track *tracks;
-/** @brief Playing track, or NULL */
-struct track *playing;
+/** @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 Number of bytes pre frame */
-size_t bpf;
+/** @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() */
-struct pollfd fds[NFDS];
+static struct pollfd fds[NFDS];
/** @brief Next free slot in @ref fds */
-int fdno;
+static int fdno;
/** @brief Listen socket */
static int listenfd;
-static time_t last_report; /* when we last reported */
-static int paused; /* pause status */
+/** @brief Timestamp of last potential report to server */
+static time_t last_report;
-/** @brief The current device state */
-enum device_states device_state;
+/** @brief Set when paused */
+static int paused;
-/** @brief Set when idled
- *
- * This is set when the sound device is deliberately closed by idle().
- */
-int idled;
+/** @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 speaker_backend *backend;
+static const struct uaudio *backend;
static const struct option options[] = {
{ "help", no_argument, 0, 'h' },
{ "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 }
};
" --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");
exit(0);
}
-/* Display version number and terminate. */
-static void version(void) {
- xprintf("disorder-speaker version %s\n", disorder_version_string);
- xfclose(stdout);
- exit(0);
-}
-
-/** @brief Return the number of bytes per frame in @p format */
-static size_t bytes_per_frame(const struct stream_header *format) {
- return format->channels * format->bits / 8;
-}
-
-/** @brief Find track @p id, maybe creating it if not found */
+/** @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;
return t;
}
-/** @brief Remove track @p id (but do not destroy it) */
+/** @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;
return t;
}
-/** @brief Destroy a track */
+/** @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);
+ if(t->fd != -1)
+ xclose(t->fd);
free(t);
}
* main loop whenever the track's file descriptor is readable, assuming the
* buffer has not reached the maximum allowed occupancy.
*/
-static int fill(struct track *t) {
+static int speaker_fill(struct track *t) {
size_t where, left;
- int n;
+ int n, rc;
D(("fill %s: eof=%d used=%zu",
t->id, t->eof, t->used));
- if(t->eof) return -1;
+ 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;
+ 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) fatal(errno, "error reading sample stream");
- return 0;
- }
- 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;
- return -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;
}
- t->used += n;
}
- return 0;
-}
-
-/** @brief Close the sound device
- *
- * This is called to deactivate the output device when pausing, and also by the
- * ALSA backend when changing encoding (in which case the sound device will be
- * immediately reactivated).
- */
-static void idle(void) {
- D(("idle"));
- if(backend->deactivate)
- backend->deactivate();
- else
- device_state = device_closed;
- idled = 1;
-}
-
-/** @brief Abandon the current track */
-void abandon(void) {
- struct speaker_message sm;
-
- D(("abandon"));
- memset(&sm, 0, sizeof sm);
- sm.type = SM_FINISHED;
- strcpy(sm.id, playing->id);
- speaker_send(1, &sm);
- removetrack(playing->id);
- destroy(playing);
- playing = 0;
-}
-
-/** @brief Enable sound output
- *
- * Makes sure the sound device is open and has the right sample format. Return
- * 0 on success and -1 on error.
- */
-static void activate(void) {
- if(backend->activate)
- backend->activate();
- else
- device_state = device_open;
-}
-
-/** @brief Check whether the current track has finished
- *
- * The current track is determined to have finished either if the input stream
- * eded before the format could be determined (i.e. it is malformed) or the
- * input is at end of file and there is less than a frame left unplayed. (So
- * it copes with decoders that crash mid-frame.)
- */
-static void maybe_finished(void) {
- if(playing
- && playing->eof
- && playing->used < bytes_per_frame(&config->sample_format))
- abandon();
+ return rc;
}
-/** @brief Play up to @p frames frames of audio
+/** @brief Return nonzero if we want to play some audio
*
- * It is always safe to call this function.
- * - If @ref playing is 0 then it will just return
- * - If @ref paused is non-0 then it will just return
- * - If @ref device_state != @ref device_open then it will call activate() and
- * return if it it fails.
- * - If there is not enough audio to play then it play what is available.
+ * 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.
*
- * If there are not enough frames to play then whatever is available is played
- * instead. It is up to mainloop() to ensure that play() is not called when
- * unreasonably only an small amounts of data is available to play.
+ * 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 void play(size_t frames) {
- size_t avail_frames, avail_bytes, written_frames;
- ssize_t written_bytes;
-
- /* Make sure there's a track to play and it is not pasued */
- if(!playing || paused)
- return;
- /* Make sure the output device is open */
- if(device_state != device_open) {
- activate();
- if(device_state != device_open)
- return;
- }
- D(("play: play %zu/%zu%s %dHz %db %dc", frames, playing->used / bpf,
- playing->eof ? " EOF" : "",
- config->sample_format.rate,
- config->sample_format.bits,
- config->sample_format.channels));
- /* Figure out how many frames there are available to write */
- if(playing->start + playing->used > sizeof playing->buffer)
- /* The ring buffer is currently wrapped, only play up to the wrap point */
- avail_bytes = (sizeof playing->buffer) - playing->start;
- else
- /* The ring buffer is not wrapped, can play the lot */
- avail_bytes = playing->used;
- avail_frames = avail_bytes / bpf;
- /* Only play up to the requested amount */
- if(avail_frames > frames)
- avail_frames = frames;
- if(!avail_frames)
- return;
- /* Play it, Sam */
- written_frames = backend->play(avail_frames);
- written_bytes = written_frames * bpf;
- /* written_bytes and written_frames had better both be set and correct by
- * this point */
- playing->start += written_bytes;
- playing->used -= written_bytes;
- playing->played += written_frames;
- /* If the pointer is at the end of the buffer (or the buffer is completely
- * empty) wrap it back to the start. */
- if(!playing->used || playing->start == (sizeof playing->buffer))
- playing->start = 0;
- frames -= written_frames;
- return;
+static int playable(void) {
+ return playing
+ && (!paused || playing->finished)
+ && playing->playable;
}
-/* Notify the server what we're up to. */
+/** @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 / config->sample_format.rate;
+ sm.data = playing->played / (uaudio_rate * uaudio_channels);
speaker_send(1, &sm);
+ xtime(&last_report);
}
- time(&last_report);
-}
-
-static void reap(int __attribute__((unused)) sig) {
- pid_t cmdpid;
- int st;
-
- do
- cmdpid = waitpid(-1, &st, WNOHANG);
- while(cmdpid > 0);
- signal(SIGCHLD, reap);
}
-int addfd(int fd, int events) {
+/** @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 -1;
}
-/** @brief Table of speaker backends */
-static const struct speaker_backend *backends[] = {
-#if HAVE_ALSA_ASOUNDLIB_H
- &alsa_backend,
-#endif
- &command_backend,
- &network_backend,
-#if HAVE_COREAUDIO_AUDIOHARDWARE_H
- &coreaudio_backend,
-#endif
-#if HAVE_SYS_SOUNDCARD_H
- &oss_backend,
-#endif
- 0
-};
-
-/** @brief Return nonzero if we want to play some audio
+/** @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
*
- * We want to play audio if there is a current track; and it is not paused; and
- * there are at least @ref FRAMES frames of audio to play, or we are in sight
- * of the end of the current track.
+ * See uaudio_callback().
*/
-static int playable(void) {
- return playing
- && !paused
- && (playing->used >= FRAMES || playing->eof);
+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;
+ 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 a second before thinking about current
- * state. */
- timeout = 1000;
+ /* 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 */
playing->slot = addfd(playing->fd, POLLIN);
else if(playing)
playing->slot = -1;
- if(playable()) {
- /* We want to play some audio. If the device is closed then we attempt
- * to open it. */
- if(device_state == device_closed)
- activate();
- /* If the device is (now) open then we will wait up until it is ready for
- * more. If something went wrong then we should have device_error
- * instead, but the post-poll code will cope even if it's
- * device_closed. */
- if(device_state == device_open)
- backend->beforepoll();
- }
/* 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. */
} 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;
- fatal(errno, "error calling poll");
- }
- /* Play some sound before doing anything else */
- if(playable()) {
- /* We want to play some audio */
- if(device_state == device_open) {
- if(backend->ready())
- play(3 * FRAMES);
- } else {
- /* We must be in _closed or _error, and it should be the latter, but we
- * cope with either.
- *
- * We most likely timed out, so now is a good time to retry. play()
- * knows to re-activate the device if necessary.
- */
- play(3 * FRAMES);
- }
+ disorder_fatal(errno, "error calling poll");
}
/* Perhaps a connection has arrived */
if(fds[listen_slot].revents & POLLIN) {
if((fd = accept(listenfd, (struct sockaddr *)&addr, &addrlen)) >= 0) {
blocking(fd);
if(read(fd, &l, sizeof l) < 4) {
- error(errno, "reading length from inbound connection");
+ disorder_error(errno, "reading length from inbound connection");
xclose(fd);
} else if(l >= sizeof id) {
- error(0, "id length too long");
+ disorder_error(0, "id length too long");
xclose(fd);
} else if(read(fd, id, l) < (ssize_t)l) {
- error(errno, "reading id from inbound connection");
+ 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*/);
- write(fd, "", 1); /* write an ack */
+ if (write(fd, "", 1) < 0) /* write an ack */
+ disorder_error(errno, "writing ack to inbound connection");
if(t->fd != -1) {
- error(0, "got a connection for a track that already has one");
+ disorder_error(0, "%s: already got a connection", id);
xclose(fd);
} else {
nonblock(fd);
t->fd = fd; /* yay */
}
+ /* Notify the server that the connection arrived */
+ sm.type = SM_ARRIVED;
+ strcpy(sm.id, id);
+ speaker_send(1, &sm);
}
} else
- error(errno, "accept");
+ disorder_error(errno, "accept");
}
/* Perhaps we have a command to process */
if(fds[stdin_slot].revents & POLLIN) {
* this won't be the case, so we don't bother looping around to pick them
* all up. */
n = speaker_recv(0, &sm);
- /* TODO */
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:
- if(playing) fatal(0, "got SM_PLAY but already playing something");
+ /* 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)
- error(0, "cannot play track because no connection arrived");
- playing = t;
- /* We attempt to play straight away rather than going round the loop.
- * play() is clever enough to perform any activation that is
- * required. */
- play(3 * FRAMES);
- report();
+ 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;
- report();
+ force_report = 1;
break;
case SM_RESUME:
D(("SM_RESUME"));
- if(paused) {
- paused = 0;
- /* As for SM_PLAY we attempt to play straight away. */
- if(playing)
- play(3 * FRAMES);
- }
- report();
+ paused = 0;
+ force_report = 1;
break;
case SM_CANCEL:
- D(("SM_CANCEL %s", sm.id));
+ D(("SM_CANCEL %s", sm.id));
t = removetrack(sm.id);
if(t) {
- if(t == playing) {
+ 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;
- strcpy(sm.id, playing->id);
- speaker_send(1, &sm);
- playing = 0;
+ 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
- error(0, "SM_CANCEL for unknown track %s", sm.id);
- report();
+ } 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)) error(0, "cannot read configuration");
- info("reloaded configuration");
+ if(config_read(1, NULL))
+ disorder_error(0, "cannot read configuration");
+ disorder_info("reloaded configuration");
break;
default:
- error(0, "unknown message type %d", sm.type);
+ disorder_error(0, "unknown message type %d", sm.type);
}
}
/* Read in any buffered data */
if(t->fd != -1
&& t->slot != -1
&& (fds[t->slot].revents & (POLLIN | POLLHUP)))
- fill(t);
- /* Maybe we finished playing a track somewhere in the above */
- maybe_finished();
- /* If we don't need the sound device for now then close it for the benefit
- * of anyone else who wants it. */
- if((!playing || paused) && device_state == device_open)
- idle();
- /* If we've not reported out state for a second do so now. */
- if(time(0) > last_report)
+ 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;
+ 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, "")) fatal(errno, "error calling setlocale");
- while((n = getopt_long(argc, argv, "hVc:dD", options, 0)) >= 0) {
+ 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();
+ case 'V': version("disorder-speaker");
case 'c': configfile = optarg; break;
case 'd': debugging = 1; break;
case 'D': debugging = 0; break;
- default: fatal(0, "invalid option");
+ case 'S': logsyslog = 0; break;
+ case 's': logsyslog = 1; break;
+ default: disorder_fatal(0, "invalid option");
}
}
- if(getenv("DISORDER_DEBUG_SPEAKER")) debugging = 1;
- /* If stderr is a TTY then log there, otherwise to syslog. */
- if(!isatty(2)) {
+ if((d = getenv("DISORDER_DEBUG_SPEAKER"))) debugging = atoi(d);
+ if(logsyslog) {
openlog(progname, LOG_PID, LOG_DAEMON);
log_default = &log_syslog;
}
- if(config_read(1)) fatal(0, "cannot read configuration");
- bpf = bytes_per_frame(&config->sample_format);
+ config_uaudio_apis = uaudio_apis;
+ if(config_read(1, NULL)) disorder_fatal(0, "cannot read configuration");
/* ignore SIGPIPE */
signal(SIGPIPE, SIG_IGN);
- /* reap kids */
- signal(SIGCHLD, reap);
/* 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) fatal(0, "do not run as root");
- /* identify the backend used to play */
- for(n = 0; backends[n]; ++n)
- if(backends[n]->backend == config->speaker_backend)
- break;
- if(!backends[n])
- fatal(0, "unsupported backend %d", config->speaker_backend);
- backend = backends[n];
+ 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 */
- backend->init();
+ 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",
+ snprintf(addr.sun_path, sizeof addr.sun_path, "%s/speaker/socket",
config->home);
if(unlink(addr.sun_path) < 0 && errno != ENOENT)
- error(errno, "removing %s", addr.sun_path);
+ 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)
- fatal(errno, "error binding socket to %s", addr.sun_path);
+ disorder_fatal(errno, "error binding socket to %s", addr.sun_path);
xlisten(listenfd, 128);
nonblock(listenfd);
- info("listening on %s", addr.sun_path);
+ disorder_info("listening on %s", addr.sun_path);
memset(&sm, 0, sizeof sm);
sm.type = SM_READY;
speaker_send(1, &sm);
mainloop();
- info("stopped (parent terminated)");
+ disorder_info("stopped (parent terminated)");
exit(0);
}
~mdw / disorder / blobdiff