Consistency check for finished tracks.

[disorder] / server / speaker.c

diff --git a/server/speaker.c b/server/speaker.c
index 4082064841abe608e80008d64457eb775dd46104..a8524b4e607864ef66e69de313e1502ab2540b12 100644 (file)
--- a/server/speaker.c
+++ b/server/speaker.c
@@ -1,22 +1,20 @@
 /*
  * This file is part of DisOrder
 /*
  * This file is part of DisOrder

- * Copyright (C) 2005, 2006, 2007 Richard Kettlewell
+ * Copyright (C) 2005-2010 Richard Kettlewell

  * Portions (C) 2007 Mark Wooding
  *
  * 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
  * 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.
  * (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
  * 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
  */
 /** @file server/speaker.c
  * @brief Speaker process


@@ -27,9 +25,29 @@
  * process that is about to become disorder-normalize) and plays them in the
  * right order.
  *
  * 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 returned by the callback; pause and resume is implemented 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
  *
  * Inbound data is expected to match @c config->sample_format.  In normal use
  * this is arranged by the @c disorder-normalize program (see @ref


@@ -50,25 +68,23 @@
  * 2-byte samples.
  */
 
  * 2-byte samples.
  */
 

-#include <config.h>
-#include "types.h"
+#include "common.h"

 
 #include <getopt.h>
 
 #include <getopt.h>

-#include <stdio.h>
-#include <stdlib.h>

 #include <locale.h>
 #include <syslog.h>
 #include <unistd.h>
 #include <errno.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/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 "configuration.h"
 #include "syscalls.h"


@@ -77,40 +93,135 @@
 #include "mem.h"
 #include "speaker-protocol.h"
 #include "user.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;
+
+  /** @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 Linked list of all prepared tracks */
-struct track *tracks;
+/** @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
+ *
+ * This includes @ref playing and @ref pending_playing.
+ */
+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).
+ *
+ * This track remains on @ref track.
+ */
+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.
+ *
+ * This track remains on @p track.
+ */
+static struct track *pending_playing;

 
 /** @brief Array of file descriptors for poll() */
 
 /** @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;
+/** @brief Next free slot in @ref fds
+ *
+ * This is used when filling in the @ref fds array each iteration through the
+ * event loop.
+ */
+static int fdno;

 
 /** @brief Listen socket */
 static int listenfd;
 
 
 /** @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 */
 
 /** @brief Selected backend */

-static const struct speaker_backend *backend;
+static const struct uaudio *backend;

 
 static const struct option options[] = {
   { "help", no_argument, 0, 'h' },
 
 static const struct option options[] = {
   { "help", no_argument, 0, 'h' },


@@ -140,19 +251,11 @@ static void help(void) {
   exit(0);
 }
 
   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 if not found
+ * @return Pointer to track structure or NULL
+ */

 static struct track *findtrack(const char *id, int create) {
   struct track *t;
 
 static struct track *findtrack(const char *id, int create) {
   struct track *t;
 


@@ -169,7 +272,10 @@ static struct track *findtrack(const char *id, int create) {
   return 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;
 
 static struct track *removetrack(const char *id) {
   struct track *t, **tt;
 


@@ -181,10 +287,13 @@ static struct track *removetrack(const char *id) {
   return t;
 }
 
   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));
 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);
 }
 
   free(t);
 }
 


@@ -195,196 +304,101 @@ static void destroy(struct track *t) {
  * 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.
  * 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.

+ *
+ * Errors count as EOF.

  */
 static int speaker_fill(struct track *t) {
   size_t where, left;
  */
 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));
 
   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(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);
     do {
       n = read(t->fd, t->buffer + where, left);
     } while(n < 0 && errno == EINTR);

-    if(n < 0) {
-      if(errno != EAGAIN) fatal(errno, "error reading sample stream");
-      return 0;
-    }
-    if(n == 0) {
-      D(("fill %s: eof detected", t->id));
+    pthread_mutex_lock(&lock);
+    if(n < 0 && errno == EAGAIN) {
+      /* EAGAIN means more later */
+      rc = 0;
+    } else if(n <= 0) {
+      /* n=0 means EOF.  n<0 means some error occurred.  We log the error but
+       * otherwise treat it as identical to EOF. */
+      if(n < 0)
+        disorder_error(errno, "error reading sample stream for %s", t->id);
+      else
+        D(("fill %s: eof detected", t->id));

       t->eof = 1;
       t->eof = 1;

+      /* A track always becomes playable at EOF; we're not going to see any
+       * more data. */

       t->playable = 1;
       t->playable = 1;

-      return -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;
-    if(t->used == sizeof t->buffer)
-      t->playable = 1;
-  }
-  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();
+  } else
+    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.
 }
 
 /** @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.  The effect is that if you hit pause towards the end of a track,
+ * what should happen is that it finished but the next one is paused right at
+ * its start.

  */
 static int playable(void) {
   return playing
  */
 static int playable(void) {
   return playing

-         && !paused
+         && (!paused || playing->finished)

          && playing->playable;
 }
 
          && playing->playable;
 }
 

-/** @brief Play up to @p frames frames of 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.
- *
- * If there are not enough frames to play then whatever is available is played
- * instead.  It is up to mainloop() to ensure that speaker_play() is not called
- * when unreasonably only an small amounts of data is available to play.
- */
-static void speaker_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 paused */
-  if(!playable())
-    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;
-  /* If the buffer emptied out mark the track as unplayably */
-  if(!playing->used) {
-    error(0, "track buffer emptied");
-    playing->playable = 0;
-  }
-  frames -= written_frames;
-  return;
-}
-
-/* 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) {
 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);
     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);
     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;
   if(fdno < NFDS) {
     fds[fdno].fd = fd;
     fds[fdno].events = events;


@@ -393,33 +407,83 @@ int addfd(int fd, int events) {
     return -1;
 }
 
     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 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 so make sure the event
+       * loop wakes up. */
+      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;
 
 /** @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;

 
 

+  pthread_mutex_lock(&lock);
+  /* Keep going while our parent process is alive */

   while(getppid() != 1) {
   while(getppid() != 1) {

+    int force_report = 0;
+

     fdno = 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 */
     /* Always ready for commands from the main server. */
     stdin_slot = addfd(0, POLLIN);
     /* Also always ready for inbound connections */


@@ -433,18 +497,8 @@ static void mainloop(void) {
       playing->slot = addfd(playing->fd, POLLIN);
     else if(playing)
       playing->slot = -1;
       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(&timeout);
-    }
+    /* Allow the poll() to be interrupted at the end of a track */
+    sigpipe_slot = addfd(sigpipe[0], POLLIN);

     /* 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. */
     /* 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. */


@@ -458,26 +512,12 @@ static void mainloop(void) {
           t->slot = -1;
       }
     /* Wait for something interesting to happen */
           t->slot = -1;
       }
     /* Wait for something interesting to happen */

+    pthread_mutex_unlock(&lock);

     n = poll(fds, fdno, timeout);
     n = poll(fds, fdno, timeout);

+    pthread_mutex_lock(&lock);

     if(n < 0) {
       if(errno == EINTR) continue;
     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())
-          speaker_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.
-         * speaker_play() knows to re-activate the device if necessary.
-         */
-        speaker_play(3 * FRAMES);
-      }
+      disorder_fatal(errno, "error calling poll");

     }
     /* Perhaps a connection has arrived */
     if(fds[listen_slot].revents & POLLIN) {
     }
     /* Perhaps a connection has arrived */
     if(fds[listen_slot].revents & POLLIN) {


@@ -487,31 +527,41 @@ static void mainloop(void) {
       char id[24];
 
       if((fd = accept(listenfd, (struct sockaddr *)&addr, &addrlen)) >= 0) {
       char id[24];
 
       if((fd = accept(listenfd, (struct sockaddr *)&addr, &addrlen)) >= 0) {

+        /* We do blocking reads for the header.  In theory this means that the
+         * connecting process could wedge the speaker indefinitely.  In
+         * practice that would mean that the main server was broken anyway.
+         * Still, this is ugly, and a rewrite would be nice. */

         blocking(fd);
         if(read(fd, &l, sizeof l) < 4) {
         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) {
           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) {
           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*/);
           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 for %s",
+                           id);

           if(t->fd != -1) {
           if(t->fd != -1) {

-            error(0, "%s: already got a connection", id);
+            disorder_error(0, "%s: already got a connection", id);

             xclose(fd);
           } else {
             nonblock(fd);
             t->fd = fd;               /* yay */
           }
             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
         }
       } else

-        error(errno, "accept");
+        disorder_error(errno, "accept");

     }
     /* Perhaps we have a command to process */
     if(fds[stdin_slot].revents & POLLIN) {
     }
     /* Perhaps we have a command to process */
     if(fds[stdin_slot].revents & POLLIN) {


@@ -519,59 +569,93 @@ static void mainloop(void) {
        * this won't be the case, so we don't bother looping around to pick them
        * all up. */ 
       n = speaker_recv(0, &sm);
        * 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)
       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:
        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)
          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.
-           * speaker_play() is clever enough to perform any activation that is
-           * required. */
-          speaker_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;
          break;
        case SM_PAUSE:
           D(("SM_PAUSE"));
          paused = 1;

-          report();
+          force_report = 1;

           break;
        case SM_RESUME:
           D(("SM_RESUME"));
           break;
        case SM_RESUME:
           D(("SM_RESUME"));

-          if(paused) {
-            paused = 0;
-            /* As for SM_PLAY we attempt to play straight away. */
-            if(playing)
-              speaker_play(3 * FRAMES);
-          }
-          report();
+          paused = 0;
+          force_report = 1;

          break;
        case SM_CANCEL:
          break;
        case SM_CANCEL:

-          D(("SM_CANCEL %s",  sm.id));
+          D(("SM_CANCEL %s", sm.id));

          t = removetrack(sm.id);
          if(t) {
          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;
               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);
            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"));
          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:
          break;
        default:

-         error(0, "unknown message type %d", sm.type);
+         disorder_error(0, "unknown message type %d", sm.type);

         }
     }
     /* Read in any buffered data */
         }
     }
     /* Read in any buffered data */


@@ -580,14 +664,63 @@ static void mainloop(void) {
          && t->slot != -1
          && (fds[t->slot].revents & (POLLIN | POLLHUP)))
          speaker_fill(t);
          && t->slot != -1
          && (fds[t->slot].revents & (POLLIN | POLLHUP)))
          speaker_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)
+    /* 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) {
+      if(!playing->finished) {
+        /* should never happen but we'd like to know if it does */
+        disorder_fatal(0, "track finish state inconsistent");
+      }
+      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();
   }
 }
       report();
   }
 }


@@ -598,19 +731,21 @@ int main(int argc, char **argv) {
   static const int one = 1;
   struct speaker_message sm;
   const char *d;
   static const int one = 1;
   struct speaker_message sm;
   const char *d;

+  char *dir;
+  struct rlimit rl[1];

 
   set_progname(argv);
 
   set_progname(argv);

-  if(!setlocale(LC_CTYPE, "")) fatal(errno, "error calling setlocale");
+  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();
   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;
     case 'S': logsyslog = 0; break;
     case 's': logsyslog = 1; break;
     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: fatal(0, "invalid option");
+    default: disorder_fatal(0, "invalid option");

     }
   }
   if((d = getenv("DISORDER_DEBUG_SPEAKER"))) debugging = atoi(d);
     }
   }
   if((d = getenv("DISORDER_DEBUG_SPEAKER"))) debugging = atoi(d);


@@ -618,46 +753,73 @@ int main(int argc, char **argv) {
     openlog(progname, LOG_PID, LOG_DAEMON);
     log_default = &log_syslog;
   }
     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);
   /* 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 */
   /* 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-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;
   /* 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)
            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)
   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);
   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();
   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);
 }
 
   exit(0);
 }