Correct ordering of configuration key table.

[disorder] / server / speaker.c

diff --git a/server/speaker.c b/server/speaker.c
index f7e2f590a41ab1be6a99e6c93f60aa233ee00da2..ccdf1e11aecff9115670a2dc46049ecd7bf7d294 100644 (file)
--- a/server/speaker.c
+++ b/server/speaker.c
@@ -1,124 +1,217 @@
 /*
  * This file is part of DisOrder
 /*
  * 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
  * 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.
+ * 
+ * You should have received a copy of the GNU General Public License
+ * along with this program.  If not, see <http://www.gnu.org/licenses/>.
+ */
+/** @file server/speaker.c
+ * @brief Speaker process

  *
  *

- * 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 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.

  *
  *

- * 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
+ * @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.

  */
 
  */
 

-/* 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 libao
- * drivers are implemented using 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. */
-
-#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 <ao/ao.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/select.h>
 #include <sys/wait.h>
 #include <time.h>
 #include <fcntl.h>
 #include <poll.h>

-#include <sys/socket.h>
-#include <netdb.h>
+#include <sys/un.h>
+#include <sys/stat.h>
+#include <pthread.h>
+#include <sys/resource.h>

 #include <gcrypt.h>
 #include <gcrypt.h>

-#include <sys/uio.h>

 
 #include "configuration.h"
 #include "syscalls.h"
 #include "log.h"
 #include "defs.h"
 #include "mem.h"
 
 #include "configuration.h"
 #include "syscalls.h"
 #include "log.h"
 #include "defs.h"
 #include "mem.h"

-#include "speaker.h"
+#include "speaker-protocol.h"

 #include "user.h"
 #include "user.h"

-#include "addr.h"
-#include "timeval.h"
-#include "rtp.h"
+#include "printf.h"
+#include "version.h"
+#include "uaudio.h"

 
 

-#if API_ALSA
-#include <alsa/asoundlib.h>
-#endif
+/** @brief Maximum number of FDs to poll for */
+#define NFDS 1024

 
 

-#ifdef WORDS_BIGENDIAN
-# define MACHINE_AO_FMT AO_FMT_BIG
-#else
-# define MACHINE_AO_FMT AO_FMT_LITTLE
-#endif
+/** @brief Number of bytes before end of track to send SM_FINISHED
+ *
+ * Generally set to 1 second.
+ */
+static size_t early_finish;

 
 

-#define BUFFER_SECONDS 5                /* How many seconds of input to
-                                         * buffer. */
+/** @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;

 
 

-#define FRAMES 4096                     /* Frame batch size */
+  /** @brief Input file descriptor */
+  int fd;                               /* input FD */

 
 

-#define NETWORK_BYTES 1024              /* Bytes to send per network packet */
-/* (don't make this too big or arithmetic will start to overflow) */
+  /** @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];
+};

 
 

-#define RTP_AHEAD 2                     /* Max RTP playahead (seconds) */
+/** @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;

 
 

-#define NFDS 256                        /* Max FDs to poll for */
+/** @brief Linked list of all prepared tracks */
+static struct track *tracks;

 
 

-/* Known tracks are kept in a linked list.  We don't normally to have
- * more than two - maybe three at the outside. */
-static struct track {
-  struct track *next;                   /* next track */
-  int fd;                               /* input FD */
-  char id[24];                          /* ID */
-  size_t start, used;                   /* start + bytes used */
-  int eof;                              /* input is at EOF */
-  int got_format;                       /* got format yet? */
-  ao_sample_format format;              /* sample format */
-  unsigned long long played;            /* number of frames played */
-  char *buffer;                         /* sample buffer */
-  size_t size;                          /* sample buffer size */
-  int slot;                             /* poll array slot */
-} *tracks, *playing;                    /* all tracks + playing track */
+/** @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;

 
 

-static time_t last_report;              /* when we last reported */
-static int paused;                      /* pause status */
-static ao_sample_format pcm_format;     /* current format if aodev != 0 */
-static size_t bpf;                      /* bytes per frame */
-static struct pollfd fds[NFDS];         /* if we need more than that */
-static int fdno;                        /* fd number */
-static size_t bufsize;                  /* buffer size */
-#if API_ALSA
-static snd_pcm_t *pcm;                  /* current pcm handle */
-static snd_pcm_uframes_t last_pcm_bufsize; /* last seen buffer size */
-#endif
-static int ready;                       /* ready to send audio */
-static int forceplay;                   /* frames to force play */
-static int cmdfd = -1;                  /* child process input */
-static int bfd = -1;                    /* broadcast FD */
-static uint32_t rtp_time;               /* RTP timestamp */
-static struct timeval rtp_time_real;    /* corresponding real time */
-static uint16_t rtp_seq;                /* frame sequence number */
-static uint32_t rtp_id;                 /* RTP SSRC */
-static int idled;                       /* set when idled */
-static int audio_errors;                /* audio error counter */
+/** @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' },
 
 static const struct option options[] = {
   { "help", no_argument, 0, 'h' },


@@ -126,6 +219,8 @@ static const struct option options[] = {
   { "config", required_argument, 0, 'c' },
   { "debug", no_argument, 0, 'd' },
   { "no-debug", no_argument, 0, 'D' },
   { "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 }
 };
 
   { 0, 0, 0, 0 }
 };
 


@@ -138,6 +233,7 @@ static void help(void) {
          "  --version, -V           Display version number\n"
          "  --config PATH, -c PATH  Set configuration file\n"
          "  --debug, -d             Turn on debugging\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");
           "\n"
           "Speaker process for DisOrder.  Not intended to be run\n"
           "directly.\n");


@@ -145,19 +241,11 @@ 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);
-}
-
-/* Return the number of bytes per frame in FORMAT. */
-static size_t bytes_per_frame(const ao_sample_format *format) {
-  return format->channels * format->bits / 8;
-}
-
-/* Find track 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;
 
 static struct track *findtrack(const char *id, int create) {
   struct track *t;
 


@@ -170,14 +258,14 @@ static struct track *findtrack(const char *id, int create) {
     strcpy(t->id, id);
     t->fd = -1;
     tracks = t;
     strcpy(t->id, id);
     t->fd = -1;
     tracks = t;

-    /* The initial input buffer will be the sample format. */
-    t->buffer = (void *)&t->format;
-    t->size = sizeof t->format;

   }
   return t;
 }
 
   }
   return t;
 }
 

-/* Remove track 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;
 


@@ -189,572 +277,108 @@ static struct track *removetrack(const char *id) {
   return t;
 }
 
   return t;
 }
 

-/* 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->buffer != (void *)&t->format) free(t->buffer);
-  free(t);
-}
-
-/* Notice a new FD. */
-static void acquire(struct track *t, int fd) {
-  D(("acquire %s %d", t->id, fd));

   if(t->fd != -1)
     xclose(t->fd);
   if(t->fd != -1)
     xclose(t->fd);

-  t->fd = fd;
-  nonblock(fd);
+  free(t);

 }
 
 }
 

-/* Read data into a sample buffer.  Return 0 on success, -1 on EOF. */
-static int fill(struct track *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;
   size_t where, left;

-  int n;
+  int n, rc;

 
 

-  D(("fill %s: eof=%d used=%zu size=%zu  got_format=%d",
-     t->id, t->eof, t->used, t->size, t->got_format));
-  if(t->eof) return -1;
-  if(t->used < t->size) {
+  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 */
     /* there is room left in the buffer */

-    where = (t->start + t->used) % t->size;
-    if(t->got_format) {
-      /* We are reading audio data, get as much as we can */
-      if(where >= t->start) left = t->size - where;
-      else left = t->start - where;
-    } else
-      /* We are still waiting for the format, only get that */
-      left = sizeof (ao_sample_format) - t->used;
+    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);
     do {
       n = read(t->fd, t->buffer + where, left);
     } while(n < 0 && errno == EINTR);

+    pthread_mutex_lock(&lock);

     if(n < 0) {
     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;
       D(("fill %s: eof detected", t->id));
       t->eof = 1;

-      return -1;
-    }
-    t->used += n;
-    if(!t->got_format && t->used >= sizeof (ao_sample_format)) {
-      assert(t->used == sizeof (ao_sample_format));
-      /* Check that our assumptions are met. */
-      if(t->format.bits & 7)
-        fatal(0, "bits per sample not a multiple of 8");
-      /* Make a new buffer for audio data. */
-      t->size = bytes_per_frame(&t->format) * t->format.rate * BUFFER_SECONDS;
-      t->buffer = xmalloc(t->size);
-      t->used = 0;
-      t->got_format = 1;
-      D(("got format for %s", t->id));
+      /* 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 0;
+  return rc;

 }
 
 }
 

-/* Return true if A and B denote identical libao formats, else false. */
-static int formats_equal(const ao_sample_format *a,
-                         const ao_sample_format *b) {
-  return (a->bits == b->bits
-          && a->rate == b->rate
-          && a->channels == b->channels
-          && a->byte_format == b->byte_format);
-}
-
-/* Close the sound device. */
-static void idle(void) {
-  D(("idle"));
-#if API_ALSA
-  if(config->speaker_backend == BACKEND_ALSA && pcm) {
-    int  err;
-
-    if((err = snd_pcm_nonblock(pcm, 0)) < 0)
-      fatal(0, "error calling snd_pcm_nonblock: %d", err);
-    D(("draining pcm"));
-    snd_pcm_drain(pcm);
-    D(("closing pcm"));
-    snd_pcm_close(pcm);
-    pcm = 0;
-    forceplay = 0;
-    D(("released audio device"));
-  }
-#endif
-  idled = 1;
-  ready = 0;
-}
-
-/* Abandon the current track */
-static 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, 0);
-  removetrack(playing->id);
-  destroy(playing);
-  playing = 0;
-  forceplay = 0;
-}
-
-#if API_ALSA
-static void log_params(snd_pcm_hw_params_t *hwparams,
-                       snd_pcm_sw_params_t *swparams) {
-  snd_pcm_uframes_t f;
-  unsigned u;
-
-  return;                               /* too verbose */
-  if(hwparams) {
-    /* TODO */
-  }
-  if(swparams) {
-    snd_pcm_sw_params_get_silence_size(swparams, &f);
-    info("sw silence_size=%lu", (unsigned long)f);
-    snd_pcm_sw_params_get_silence_threshold(swparams, &f);
-    info("sw silence_threshold=%lu", (unsigned long)f);
-    snd_pcm_sw_params_get_sleep_min(swparams, &u);
-    info("sw sleep_min=%lu", (unsigned long)u);
-    snd_pcm_sw_params_get_start_threshold(swparams, &f);
-    info("sw start_threshold=%lu", (unsigned long)f);
-    snd_pcm_sw_params_get_stop_threshold(swparams, &f);
-    info("sw stop_threshold=%lu", (unsigned long)f);
-    snd_pcm_sw_params_get_xfer_align(swparams, &f);
-    info("sw xfer_align=%lu", (unsigned long)f);
-  }
-}
-#endif
-
-static void soxargs(const char ***pp, char **qq, ao_sample_format *ao) {
-  int n;
-
-  *(*pp)++ = "-t.raw";
-  *(*pp)++ = "-s";
-  *(*pp)++ = *qq; n = sprintf(*qq, "-r%d", ao->rate); *qq += n + 1;
-  *(*pp)++ = *qq; n = sprintf(*qq, "-c%d", ao->channels); *qq += n + 1;
-  /* sox 12.17.9 insists on -b etc; CVS sox insists on -<n> etc; both are
-   * deployed! */
-  switch(config->sox_generation) {
-  case 0:
-    if(ao->bits != 8
-       && ao->byte_format != AO_FMT_NATIVE
-       && ao->byte_format != MACHINE_AO_FMT) {
-      *(*pp)++ = "-x";
-    }
-    switch(ao->bits) {
-    case 8: *(*pp)++ = "-b"; break;
-    case 16: *(*pp)++ = "-w"; break;
-    case 32: *(*pp)++ = "-l"; break;
-    case 64: *(*pp)++ = "-d"; break;
-    default: fatal(0, "cannot handle sample size %d", (int)ao->bits);
-    }
-    break;
-  case 1:
-    switch(ao->byte_format) {
-    case AO_FMT_NATIVE: break;
-    case AO_FMT_BIG: *(*pp)++ = "-B"; break;
-    case AO_FMT_LITTLE: *(*pp)++ = "-L"; break;
-    }
-    *(*pp)++ = *qq; n = sprintf(*qq, "-%d", ao->bits/8); *qq += n + 1;
-    break;
-  }
-}
-
-/* Make sure the sound device is open and has the right sample format.  Return
- * 0 on success and -1 on error. */
-static int activate(void) {
-  /* If we don't know the format yet we cannot start. */
-  if(!playing->got_format) {
-    D((" - not got format for %s", playing->id));
-    return -1;
-  }
-  switch(config->speaker_backend) {
-  case BACKEND_COMMAND:
-  case BACKEND_NETWORK:
-    /* If we pass audio on to some other agent then we enforce the configured
-     * sample format on the *inbound* audio data. */
-    if(!formats_equal(&playing->format, &config->sample_format)) {
-      char argbuf[1024], *q = argbuf;
-      const char *av[18], **pp = av;
-      int soxpipe[2];
-      pid_t soxkid;
-      *pp++ = "sox";
-      soxargs(&pp, &q, &playing->format);
-      *pp++ = "-";
-      soxargs(&pp, &q, &config->sample_format);
-      *pp++ = "-";
-      *pp++ = 0;
-      if(debugging) {
-        for(pp = av; *pp; pp++)
-          D(("sox arg[%d] = %s", pp - av, *pp));
-        D(("end args"));
-      }
-      xpipe(soxpipe);
-      soxkid = xfork();
-      if(soxkid == 0) {
-        xdup2(playing->fd, 0);
-        xdup2(soxpipe[1], 1);
-        fcntl(0, F_SETFL, fcntl(0, F_GETFL) & ~O_NONBLOCK);
-        close(soxpipe[0]);
-        close(soxpipe[1]);
-        close(playing->fd);
-        execvp("sox", (char **)av);
-        _exit(1);
-      }
-      D(("forking sox for format conversion (kid = %d)", soxkid));
-      close(playing->fd);
-      close(soxpipe[1]);
-      playing->fd = soxpipe[0];
-      playing->format = config->sample_format;
-      ready = 0;
-    }
-    if(!ready) {
-      pcm_format = config->sample_format;
-      bufsize = 3 * FRAMES;
-      bpf = bytes_per_frame(&config->sample_format);
-      D(("acquired audio device"));
-      ready = 1;
-    }
-    return 0;
-  case BACKEND_ALSA:
-#if API_ALSA
-    /* If we need to change format then close the current device. */
-    if(pcm && !formats_equal(&playing->format, &pcm_format))
-      idle();
-    if(!pcm) {
-      snd_pcm_hw_params_t *hwparams;
-      snd_pcm_sw_params_t *swparams;
-      snd_pcm_uframes_t pcm_bufsize;
-      int err;
-      int sample_format = 0;
-      unsigned rate;
-
-      D(("snd_pcm_open"));
-      if((err = snd_pcm_open(&pcm,
-                             config->device,
-                             SND_PCM_STREAM_PLAYBACK,
-                             SND_PCM_NONBLOCK))) {
-        error(0, "error from snd_pcm_open: %d", err);
-        goto error;
-      }
-      snd_pcm_hw_params_alloca(&hwparams);
-      D(("set up hw params"));
-      if((err = snd_pcm_hw_params_any(pcm, hwparams)) < 0)
-        fatal(0, "error from snd_pcm_hw_params_any: %d", err);
-      if((err = snd_pcm_hw_params_set_access(pcm, hwparams,
-                                             SND_PCM_ACCESS_RW_INTERLEAVED)) < 0)
-        fatal(0, "error from snd_pcm_hw_params_set_access: %d", err);
-      switch(playing->format.bits) {
-      case 8:
-        sample_format = SND_PCM_FORMAT_S8;
-        break;
-      case 16:
-        switch(playing->format.byte_format) {
-        case AO_FMT_NATIVE: sample_format = SND_PCM_FORMAT_S16; break;
-        case AO_FMT_LITTLE: sample_format = SND_PCM_FORMAT_S16_LE; break;
-        case AO_FMT_BIG: sample_format = SND_PCM_FORMAT_S16_BE; break;
-          error(0, "unrecognized byte format %d", playing->format.byte_format);
-          goto fatal;
-        }
-        break;
-      default:
-        error(0, "unsupported sample size %d", playing->format.bits);
-        goto fatal;
-      }
-      if((err = snd_pcm_hw_params_set_format(pcm, hwparams,
-                                             sample_format)) < 0) {
-        error(0, "error from snd_pcm_hw_params_set_format (%d): %d",
-              sample_format, err);
-        goto fatal;
-      }
-      rate = playing->format.rate;
-      if((err = snd_pcm_hw_params_set_rate_near(pcm, hwparams, &rate, 0)) < 0) {
-        error(0, "error from snd_pcm_hw_params_set_rate (%d): %d",
-              playing->format.rate, err);
-        goto fatal;
-      }
-      if(rate != (unsigned)playing->format.rate)
-        info("want rate %d, got %u", playing->format.rate, rate);
-      if((err = snd_pcm_hw_params_set_channels(pcm, hwparams,
-                                               playing->format.channels)) < 0) {
-        error(0, "error from snd_pcm_hw_params_set_channels (%d): %d",
-              playing->format.channels, err);
-        goto fatal;
-      }
-      bufsize = 3 * FRAMES;
-      pcm_bufsize = bufsize;
-      if((err = snd_pcm_hw_params_set_buffer_size_near(pcm, hwparams,
-                                                       &pcm_bufsize)) < 0)
-        fatal(0, "error from snd_pcm_hw_params_set_buffer_size (%d): %d",
-              3 * FRAMES, err);
-      if(pcm_bufsize != 3 * FRAMES && pcm_bufsize != last_pcm_bufsize)
-        info("asked for PCM buffer of %d frames, got %d",
-             3 * FRAMES, (int)pcm_bufsize);
-      last_pcm_bufsize = pcm_bufsize;
-      if((err = snd_pcm_hw_params(pcm, hwparams)) < 0)
-        fatal(0, "error calling snd_pcm_hw_params: %d", err);
-      D(("set up sw params"));
-      snd_pcm_sw_params_alloca(&swparams);
-      if((err = snd_pcm_sw_params_current(pcm, swparams)) < 0)
-        fatal(0, "error calling snd_pcm_sw_params_current: %d", err);
-      if((err = snd_pcm_sw_params_set_avail_min(pcm, swparams, FRAMES)) < 0)
-        fatal(0, "error calling snd_pcm_sw_params_set_avail_min %d: %d",
-              FRAMES, err);
-      if((err = snd_pcm_sw_params(pcm, swparams)) < 0)
-        fatal(0, "error calling snd_pcm_sw_params: %d", err);
-      pcm_format = playing->format;
-      bpf = bytes_per_frame(&pcm_format);
-      D(("acquired audio device"));
-      log_params(hwparams, swparams);
-      ready = 1;
-    }
-    return 0;
-  fatal:
-    abandon();
-  error:
-    /* We assume the error is temporary and that we'll retry in a bit. */
-    if(pcm) {
-      snd_pcm_close(pcm);
-      pcm = 0;
-    }
-    return -1;
-#endif
-  default:
-    assert(!"reached");
-  }
-}
-
-/* Check to see whether the current track has finished playing */
-static void maybe_finished(void) {
-  if(playing
-     && playing->eof
-     && (!playing->got_format
-         || playing->used < bytes_per_frame(&playing->format)))
-    abandon();
-}
-
-static void fork_cmd(void) {
-  pid_t cmdpid;
-  int pfd[2];
-  if(cmdfd != -1) close(cmdfd);
-  xpipe(pfd);
-  cmdpid = xfork();
-  if(!cmdpid) {
-    xdup2(pfd[0], 0);
-    close(pfd[0]);
-    close(pfd[1]);
-    execl("/bin/sh", "sh", "-c", config->speaker_command, (char *)0);
-    fatal(errno, "error execing /bin/sh");
-  }
-  close(pfd[0]);
-  cmdfd = pfd[1];
-  D(("forked cmd %d, fd = %d", cmdpid, cmdfd));
-}
-
-static void play(size_t frames) {
-  size_t avail_bytes, written_frames;
-  ssize_t written_bytes;
-  struct rtp_header header;
-  struct iovec vec[2];
-
-  if(activate()) {
-    if(playing)
-      forceplay = frames;
-    else
-      forceplay = 0;                    /* Must have called abandon() */
-    return;
-  }
-  D(("play: play %zu/%zu%s %dHz %db %dc",  frames, playing->used / bpf,
-     playing->eof ? " EOF" : "",
-     playing->format.rate,
-     playing->format.bits,
-     playing->format.channels));
-  /* If we haven't got enough bytes yet wait until we have.  Exception: when
-   * we are at eof. */
-  if(playing->used < frames * bpf && !playing->eof) {
-    forceplay = frames;
-    return;
-  }
-  /* We have got enough data so don't force play again */
-  forceplay = 0;
-  /* Figure out how many frames there are available to write */
-  if(playing->start + playing->used > playing->size)
-    avail_bytes = playing->size - playing->start;
-  else
-    avail_bytes = playing->used;
-
-  switch(config->speaker_backend) {
-#if API_ALSA
-  case BACKEND_ALSA: {
-    snd_pcm_sframes_t pcm_written_frames;
-    size_t avail_frames;
-    int err;
-
-    avail_frames = avail_bytes / bpf;
-    if(avail_frames > frames)
-      avail_frames = frames;
-    if(!avail_frames)
-      return;
-    pcm_written_frames = snd_pcm_writei(pcm,
-                                        playing->buffer + playing->start,
-                                        avail_frames);
-    D(("actually play %zu frames, wrote %d",
-       avail_frames, (int)pcm_written_frames));
-    if(pcm_written_frames < 0) {
-      switch(pcm_written_frames) {
-        case -EPIPE:                        /* underrun */
-          error(0, "snd_pcm_writei reports underrun");
-          if((err = snd_pcm_prepare(pcm)) < 0)
-            fatal(0, "error calling snd_pcm_prepare: %d", err);
-          return;
-        case -EAGAIN:
-          return;
-        default:
-          fatal(0, "error calling snd_pcm_writei: %d",
-                (int)pcm_written_frames);
-      }
-    }
-    written_frames = pcm_written_frames;
-    written_bytes = written_frames * bpf;
-    break;
-  }
-#endif
-  case BACKEND_COMMAND:
-    if(avail_bytes > frames * bpf)
-      avail_bytes = frames * bpf;
-    written_bytes = write(cmdfd, playing->buffer + playing->start,
-                          avail_bytes);
-    D(("actually play %zu bytes, wrote %d",
-       avail_bytes, (int)written_bytes));
-    if(written_bytes < 0) {
-      switch(errno) {
-        case EPIPE:
-          error(0, "hmm, command died; trying another");
-          fork_cmd();
-          return;
-        case EAGAIN:
-          return;
-      }
-    }
-    written_frames = written_bytes / bpf; /* good enough */
-    break;
-  case BACKEND_NETWORK:
-    /* We transmit using RTP (RFC3550) and attempt to conform to the internet
-     * AVT profile (RFC3551). */
-    if(rtp_time_real.tv_sec == 0)
-      xgettimeofday(&rtp_time_real, 0);
-    if(idled) {
-      struct timeval now;
-      xgettimeofday(&now, 0);
-      /* There's been a gap.  Fix up the RTP time accordingly. */
-      const long offset =  (((now.tv_sec + now.tv_usec /1000000.0)
-                             - (rtp_time_real.tv_sec + rtp_time_real.tv_usec / 1000000.0)) 
-                            * playing->format.rate * playing->format.channels);
-      info("offset RTP timestamp by %ld", offset);
-      rtp_time += offset;
-    }
-    header.vpxcc = 2 << 6;              /* V=2, P=0, X=0, CC=0 */
-    header.seq = htons(rtp_seq++);
-    header.timestamp = htonl(rtp_time);
-    header.ssrc = rtp_id;
-    header.mpt = (idled ? 0x80 : 0x00) | 10;
-    /* 10 = L16 = 16-bit x 2 x 44100KHz.  We ought to deduce this value from
-     * the sample rate (in a library somewhere so that configuration.c can rule
-     * out invalid rates).
-     */
-    idled = 0;
-    if(avail_bytes > NETWORK_BYTES - sizeof header) {
-      avail_bytes = NETWORK_BYTES - sizeof header;
-      avail_bytes -= avail_bytes % bpf;
-    }
-    /* "The RTP clock rate used for generating the RTP timestamp is independent
-     * of the number of channels and the encoding; it equals the number of
-     * sampling periods per second.  For N-channel encodings, each sampling
-     * period (say, 1/8000 of a second) generates N samples. (This terminology
-     * is standard, but somewhat confusing, as the total number of samples
-     * generated per second is then the sampling rate times the channel
-     * count.)"
-     */
-    vec[0].iov_base = (void *)&header;
-    vec[0].iov_len = sizeof header;
-    vec[1].iov_base = playing->buffer + playing->start;
-    vec[1].iov_len = avail_bytes;
-#if 0
-    {
-      char buffer[3 * sizeof header + 1];
-      size_t n;
-      const uint8_t *ptr = (void *)&header;
-      
-      for(n = 0; n < sizeof header; ++n)
-        sprintf(&buffer[3 * n], "%02x ", *ptr++);
-      info(buffer);
-    }
-#endif
-    do {
-      written_bytes = writev(bfd,
-                             vec,
-                             2);
-    } while(written_bytes < 0 && errno == EINTR);
-    if(written_bytes < 0) {
-      error(errno, "error transmitting audio data");
-      ++audio_errors;
-      if(audio_errors == 10)
-        fatal(0, "too many audio errors");
-      return;
-    }
-    audio_errors /= 2;
-    written_bytes = avail_bytes;
-    written_frames = written_bytes / bpf;
-    /* Advance RTP's notion of the time */
-    rtp_time += written_frames * playing->format.channels;
-    /* Advance the corresponding real time */
-    assert(NETWORK_BYTES <= 2000);      /* else risk overflowing 32 bits */
-    rtp_time_real.tv_usec += written_frames * 1000000 / playing->format.rate;
-    if(rtp_time_real.tv_usec >= 1000000) {
-      ++rtp_time_real.tv_sec;
-      rtp_time_real.tv_usec -= 1000000;
-    }
-    break;
-  default:
-    assert(!"reached");
-  }
-  /* 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 == playing->size)
-    playing->start = 0;
-  frames -= written_frames;
+/** @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;

 }
 
 }
 

-/* 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;
 
 static void report(void) {
   struct speaker_message sm;
 

-  if(playing && playing->buffer != (void *)&playing->format) {
+  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 / playing->format.rate;
-    speaker_send(1, &sm, 0);
+    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);

 }
 
 }
 

+/** @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;
 static int addfd(int fd, int events) {
   if(fdno < NFDS) {
     fds[fdno].fd = fd;


@@ -764,308 +388,408 @@ static int addfd(int fd, int events) {
     return -1;
 }
 
     return -1;
 }
 

-int main(int argc, char **argv) {
-  int n, fd, stdin_slot, alsa_slots, cmdfd_slot, bfd_slot, poke, timeout;
-  struct timeval now, delta;
-  struct track *t;
-  struct speaker_message sm;
-  struct addrinfo *res, *sres;
-  static const struct addrinfo pref = {
-    0,
-    PF_INET,
-    SOCK_DGRAM,
-    IPPROTO_UDP,
-    0,
-    0,
-    0,
-    0
-  };
-  static const struct addrinfo prefbind = {
-    AI_PASSIVE,
-    PF_INET,
-    SOCK_DGRAM,
-    IPPROTO_UDP,
-    0,
-    0,
-    0,
-    0
-  };
-  static const int one = 1;
-  char *sockname, *ssockname;
-#if API_ALSA
-  int alsa_nslots = -1, err;
-#endif
-
-  set_progname(argv);
-  if(!setlocale(LC_CTYPE, "")) fatal(errno, "error calling setlocale");
-  while((n = getopt_long(argc, argv, "hVc:dD", options, 0)) >= 0) {
-    switch(n) {
-    case 'h': help();
-    case 'V': version();
-    case 'c': configfile = optarg; break;
-    case 'd': debugging = 1; break;
-    case 'D': debugging = 0; break;
-    default: fatal(0, "invalid option");
+/** @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(getenv("DISORDER_DEBUG_SPEAKER")) debugging = 1;
-  /* If stderr is a TTY then log there, otherwise to syslog. */
-  if(!isatty(2)) {
-    openlog(progname, LOG_PID, LOG_DAEMON);
-    log_default = &log_syslog;
-  }
-  if(config_read()) 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");
-  switch(config->speaker_backend) {
-  case BACKEND_ALSA:
-    info("selected ALSA backend");
-  case BACKEND_COMMAND:
-    info("selected command backend");
-    fork_cmd();
-    break;
-  case BACKEND_NETWORK:
-    res = get_address(&config->broadcast, &pref, &sockname);
-    if(!res) return -1;
-    if(config->broadcast_from.n) {
-      sres = get_address(&config->broadcast_from, &prefbind, &ssockname);
-      if(!sres) return -1;
-    } else
-      sres = 0;
-    if((bfd = socket(res->ai_family,
-                     res->ai_socktype,
-                     res->ai_protocol)) < 0)
-      fatal(errno, "error creating broadcast socket");
-    if(setsockopt(bfd, SOL_SOCKET, SO_BROADCAST, &one, sizeof one) < 0)
-      fatal(errno, "error settting SO_BROADCAST on broadcast socket");
-    /* We might well want to set additional broadcast- or multicast-related
-     * options here */
-    if(sres && bind(bfd, sres->ai_addr, sres->ai_addrlen) < 0)
-      fatal(errno, "error binding broadcast socket to %s", ssockname);
-    if(connect(bfd, res->ai_addr, res->ai_addrlen) < 0)
-      fatal(errno, "error connecting broadcast socket to %s", sockname);
-    /* Select an SSRC */
-    gcry_randomize(&rtp_id, sizeof rtp_id, GCRY_STRONG_RANDOM);
-    info("selected network backend, sending to %s", sockname);
-    if(config->sample_format.byte_format != AO_FMT_BIG) {
-      info("forcing big-endian sample format");
-      config->sample_format.byte_format = AO_FMT_BIG;
-    }
-    break;
-  default:
-    fatal(0, "unknown backend %d", config->speaker_backend);
+  /* 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) {
   while(getppid() != 1) {

+    int force_report = 0;
+

     fdno = 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);
     /* 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. */
     /* Try to read sample data for the currently playing track if there is
      * buffer space. */

-    if(playing && !playing->eof && playing->used < playing->size) {
+    if(playing
+       && playing->fd >= 0
+       && !playing->eof
+       && playing->used < (sizeof playing->buffer))

       playing->slot = addfd(playing->fd, POLLIN);
       playing->slot = addfd(playing->fd, POLLIN);

-    } else if(playing)
+    else if(playing)

       playing->slot = -1;
       playing->slot = -1;

-    /* If forceplay is set then wait until it succeeds before waiting on the
-     * sound device. */
-    alsa_slots = -1;
-    cmdfd_slot = -1;
-    bfd_slot = -1;
-    /* By default we will wait up to a second before thinking about current
-     * state. */
-    timeout = 1000;
-    if(ready && !forceplay) {
-      switch(config->speaker_backend) {
-      case BACKEND_COMMAND:
-        /* We send sample data to the subprocess as fast as it can accept it.
-         * This isn't ideal as pause latency can be very high as a result. */
-        if(cmdfd >= 0)
-          cmdfd_slot = addfd(cmdfd, POLLOUT);
-        break;
-      case BACKEND_NETWORK:
-        /* We want to keep the notional playing point somewhere in the near
-         * future.  If it's too near then clients that attempt even the
-         * slightest amount of read-ahead will never catch up, and those that
-         * don't will skip whenever there's a trivial network delay.  If it's
-         * too far ahead then pause latency will be too high.
-         */
-        xgettimeofday(&now, 0);
-        delta = tvsub(rtp_time_real, now);
-        if(delta.tv_sec < RTP_AHEAD) {
-          D(("delta = %ld.%06ld", (long)delta.tv_sec, (long)delta.tv_usec));
-          bfd_slot = addfd(bfd, POLLOUT);
-          if(delta.tv_sec < 0)
-            rtp_time_real = now;        /* catch up */
-        }
-        break;
-#if API_ALSA
-      case BACKEND_ALSA: {
-        /* We send sample data to ALSA as fast as it can accept it, relying on
-         * the fact that it has a relatively small buffer to minimize pause
-         * latency. */
-        int retry = 3;
-        
-        alsa_slots = fdno;
-        do {
-          retry = 0;
-          alsa_nslots = snd_pcm_poll_descriptors(pcm, &fds[fdno], NFDS - fdno);
-          if((alsa_nslots <= 0
-              || !(fds[alsa_slots].events & POLLOUT))
-             && snd_pcm_state(pcm) == SND_PCM_STATE_XRUN) {
-            error(0, "underrun detected after call to snd_pcm_poll_descriptors()");
-            if((err = snd_pcm_prepare(pcm)))
-              fatal(0, "error calling snd_pcm_prepare: %d", err);
-          } else
-            break;
-        } while(retry-- > 0);
-        if(alsa_nslots >= 0)
-          fdno += alsa_nslots;
-        break;
-      }
-#endif
-      default:
-        assert(!"unknown backend");
-      }
-    }

     /* If any other tracks don't have a full buffer, try to read sample data
     /* If any other tracks don't have a full buffer, try to read sample data

-     * from them. */
+     * 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) {
     for(t = tracks; t; t = t->next)
       if(t != playing) {

-        if(!t->eof && t->used < t->size) {
+        if(t->fd >= 0
+           && !t->eof
+           && t->used < sizeof t->buffer) {

           t->slot = addfd(t->fd,  POLLIN | POLLHUP);
         } else
           t->slot = -1;
       }
           t->slot = addfd(t->fd,  POLLIN | POLLHUP);
         } else
           t->slot = -1;
       }

+    sigpipe_slot = addfd(sigpipe[0], POLLIN);

     /* Wait for something interesting to happen */
     /* 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");
+      disorder_fatal(errno, "error calling poll");

     }
     }

-    /* Play some sound before doing anything else */
-    poke = 0;
-    switch(config->speaker_backend) {
-#if API_ALSA
-    case BACKEND_ALSA:
-      if(alsa_slots != -1) {
-        unsigned short alsa_revents;
-        
-        if((err = snd_pcm_poll_descriptors_revents(pcm,
-                                                   &fds[alsa_slots],
-                                                   alsa_nslots,
-                                                   &alsa_revents)) < 0)
-          fatal(0, "error calling snd_pcm_poll_descriptors_revents: %d", err);
-        if(alsa_revents & (POLLOUT | POLLERR))
-          play(3 * FRAMES);
-      } else
-        poke = 1;
-      break;
-#endif
-    case BACKEND_COMMAND:
-      if(cmdfd_slot != -1) {
-        if(fds[cmdfd_slot].revents & (POLLOUT | POLLERR))
-          play(3 * FRAMES);
-      } else
-        poke = 1;
-      break;
-    case BACKEND_NETWORK:
-      if(bfd_slot != -1) {
-        if(fds[bfd_slot].revents & (POLLOUT | POLLERR))
-          play(3 * FRAMES);
+    /* 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 */
+          }
+          /* Notify the server that the connection arrived */
+          sm.type = SM_ARRIVED;
+          strcpy(sm.id, id);
+          speaker_send(1, &sm);
+        }

       } else
       } else

-        poke = 1;
-      break;
-    }
-    if(poke) {
-      /* Some attempt to play must have failed */
-      if(playing && !paused)
-        play(forceplay);
-      else
-        forceplay = 0;                  /* just in case */
+        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) {

-      n = speaker_recv(0, &sm, &fd);
+      /* 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)
       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) {
        switch(sm.type) {

-       case SM_PREPARE:
-          D(("SM_PREPARE %s %d", sm.id, fd));
-         if(fd == -1) fatal(0, "got SM_PREPARE but no file descriptor");
-         t = findtrack(sm.id, 1);
-          acquire(t, fd);
-         break;

        case SM_PLAY:
        case SM_PLAY:

-          D(("SM_PLAY %s %d", sm.id, fd));
-          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);
          t = findtrack(sm.id, 1);

-          if(fd != -1) acquire(t, fd);
-          playing = t;
-          play(bufsize);
-          report();
+          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;
          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;
-            if(playing)
-              play(bufsize);
-          }
-          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, 0);
-             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()) 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 */
     for(t = tracks; t; t = t->next)
         }
     }
     /* Read in any buffered data */
     for(t = tracks; t; t = t->next)

-      if(t->slot != -1 && (fds[t->slot].revents & (POLLIN | POLLHUP)))
-         fill(t);
-    /* We might be able to play now */
-    if(ready && forceplay && playing && !paused)
-      play(forceplay);
-    /* 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) && ready)
-      idle();
-    /* If we've not reported out state for a second do so now. */
-    if(time(0) > last_report)
+      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();
   }
       report();
   }

-  info("stopped (parent terminated)");
+}
+
+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);
 }
 
   exit(0);
 }