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
/** @brief The current PCM handle */
static snd_pcm_t *pcm;
static snd_pcm_uframes_t last_pcm_bufsize; /* last seen buffer size */
-static ao_sample_format pcm_format; /* current format if aodev != 0 */
#endif
-/** @brief Ready to send audio
- *
- * This is set when the destination is ready to receive audio. Generally
- * this implies that the sound device is open. In the ALSA backend it
- * does @b not necessarily imply that is has the right sample format.
- */
-static int ready;
+/** @brief The current device state */
+enum device_states device_state;
-/** @brief Frames to force-play
+/** @brief The current device sample format
*
- * If this is nonzero, and playing is enabled, then the main loop will attempt
- * to play this many frames without checking whether the output device is
- * ready.
+ * Only meaningful if @ref device_state = @ref device_open or perhaps @ref
+ * device_error. For @ref FIXED_FORMAT backends, this should always match @c
+ * config->sample_format.
*/
-static int forceplay;
+ao_sample_format device_format;
/** @brief Pipe to subprocess
*
/** @brief Set when idled
*
* This is set when the sound device is deliberately closed by idle().
- * @ref ready is set to 0 at the same time.
*/
static int idled; /* set when idled */
*/
static void idle(void) {
D(("idle"));
- if(backend->deactivate)
+ if(backend->deactivate)
backend->deactivate();
+ else
+ device_state = device_closed;
idled = 1;
- ready = 0;
}
/** @brief Abandon the current track */
removetrack(playing->id);
destroy(playing);
playing = 0;
- forceplay = 0;
-}
-
-#if API_ALSA
-/** @brief Log ALSA parameters */
-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
/** @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 int activate(void) {
+static void 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;
+ return;
}
- return backend->activate();
+ if(backend->flags & FIXED_FORMAT)
+ device_format = config->sample_format;
+ if(backend->activate) {
+ backend->activate();
+ } else {
+ assert(backend->flags & FIXED_FORMAT);
+ /* ...otherwise device_format not set */
+ device_state = device_open;
+ }
+ if(device_state == device_open)
+ bpf = bytes_per_frame(&device_format);
}
/** @brief Check whether the current track has finished
abandon();
}
-/** @brief Start the subprocess for @ref BACKEND_COMMAND */
-static void fork_cmd(void) {
- pid_t cmdpid;
- int pfd[2];
- if(cmdfd != -1) close(cmdfd);
- xpipe(pfd);
- cmdpid = xfork();
- if(!cmdpid) {
- signal(SIGPIPE, SIG_DFL);
- 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));
-}
-
-/** @brief Play up to @p frames frames of audio */
+/** @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 play() is not called when
+ * unreasonably only an small amounts of data is available to play.
+ */
static void play(size_t frames) {
size_t avail_frames, avail_bytes, written_frames;
ssize_t written_bytes;
- /* Make sure the output device is activated */
- if(activate()) {
- if(playing)
- forceplay = frames;
- else
- forceplay = 0; /* Must have called abandon() */
+ /* Make sure there's a track to play and it is not pasued */
+ if(!playing || paused)
return;
+ /* Make sure the output device is open and has the right sample format */
+ if(device_state != device_open
+ || !formats_equal(&device_format, &playing->format)) {
+ activate();
+ if(device_state != device_open)
+ 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)
/* The ring buffer is currently wrapped, only play up to the wrap point */
if(!playing->used || playing->start == playing->size)
playing->start = 0;
frames -= written_frames;
+ return;
}
/* Notify the server what we're up to. */
info("selected ALSA backend");
}
+/** @brief Log ALSA parameters */
+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);
+ }
+}
+
+/** @brief ALSA deactivation */
+static void alsa_deactivate(void) {
+ if(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;
+ device_state = device_closed;
+ D(("released audio device"));
+ }
+}
+
/** @brief ALSA backend activation */
-static int alsa_activate(void) {
+static void alsa_activate(void) {
/* If we need to change format then close the current device. */
- if(pcm && !formats_equal(&playing->format, &pcm_format))
- idle();
+ if(pcm && !formats_equal(&playing->format, &device_format))
+ alsa_deactivate();
+ /* Now if the sound device is open it must have the right format */
if(!pcm) {
snd_pcm_hw_params_t *hwparams;
snd_pcm_sw_params_t *swparams;
playing->format.channels, err);
goto fatal;
}
- bufsize = 3 * FRAMES;
- pcm_bufsize = bufsize;
+ pcm_bufsize = 3 * FRAMES;
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",
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);
+ device_format = playing->format;
D(("acquired audio device"));
log_params(hwparams, swparams);
- ready = 1;
+ device_state = device_open;
}
- return 0;
+ return;
fatal:
abandon();
error:
if(pcm) {
snd_pcm_close(pcm);
pcm = 0;
+ device_state = device_error;
}
- return -1;
+ return;
}
/** @brief Play via ALSA */
}
/** @brief Process poll() results for ALSA */
-static int alsa_afterpoll(void) {
+static int alsa_ready(void) {
int err;
- 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);
- return 0;
- } else
+ 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))
return 1;
+ else
+ return 0;
}
+#endif
-/** @brief ALSA deactivation */
-static void alsa_deactivate(void) {
- if(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"));
+/** @brief Start the subprocess for @ref BACKEND_COMMAND */
+static void fork_cmd(void) {
+ pid_t cmdpid;
+ int pfd[2];
+ if(cmdfd != -1) close(cmdfd);
+ xpipe(pfd);
+ cmdpid = xfork();
+ if(!cmdpid) {
+ signal(SIGPIPE, SIG_DFL);
+ 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));
}
-#endif
/** @brief Command backend initialization */
static void command_init(void) {
}
/** @brief Process poll() results for subprocess play */
-static int command_afterpoll(void) {
- if(cmdfd_slot != -1) {
- if(fds[cmdfd_slot].revents & (POLLOUT | POLLERR))
- play(3 * FRAMES);
+static int command_ready(void) {
+ if(fds[cmdfd_slot].revents & (POLLOUT | POLLERR))
+ return 1;
+ else
return 0;
- } else
- return -1;
-}
-
-/** @brief Command/network backend activation */
-static int generic_activate(void) {
- if(!ready) {
- bufsize = 3 * FRAMES;
- bpf = bytes_per_frame(&config->sample_format);
- D(("acquired audio device"));
- ready = 1;
- }
- return 0;
}
/** @brief Network backend initialization */
}
/** @brief Process poll() results for network play */
-static int network_afterpoll(void) {
- if(bfd_slot != -1) {
- if(fds[bfd_slot].revents & (POLLOUT | POLLERR))
- play(3 * FRAMES);
- return 0;
- } else
+static int network_ready(void) {
+ if(fds[bfd_slot].revents & (POLLOUT | POLLERR))
return 1;
+ else
+ return 0;
}
/** @brief Table of speaker backends */
alsa_play,
alsa_deactivate,
alsa_beforepoll,
- alsa_afterpoll
+ alsa_ready
},
#endif
{
BACKEND_COMMAND,
FIXED_FORMAT,
command_init,
- generic_activate,
+ 0, /* activate */
command_play,
0, /* deactivate */
command_beforepoll,
- command_afterpoll
+ command_ready
},
{
BACKEND_NETWORK,
FIXED_FORMAT,
network_init,
- generic_activate,
+ 0, /* activate */
network_play,
0, /* deactivate */
network_beforepoll,
- network_afterpoll
+ network_ready
},
{ -1, 0, 0, 0, 0, 0, 0, 0 } /* end of list */
};
-/** @brief Main event loop
- *
- * This has grown in a rather bizarre and ad-hoc way is very sensitive to
- * changes...
+/** @brief Return nonzero if we want to play some audio
*
- * Firstly the loop is terminated when the parent process exits. Therefore the
- * speaker process has the same lifetime as the main server. This and the
- * reading of data from decoders is comprehensible enough.
- *
- * The playing of audio is more complicated however.
- *
- * On the first run through when a track is ready to be played, @ref ready and
- * @ref forceplay will both be zero. Therefore @c beforepoll is not called.
- *
- * @c afterpoll on the other hand @b is called and will return nonzero. The
- * result is that we call @c play(0). This will call activate(), setting
- * @ref ready nonzero, but otherwise has no immediate effect.
- *
- * We then deal with stdin and the decoders.
- *
- * We then reach the second place we might play some audio. @ref forceplay is
- * 0 so nothing happens here again.
- *
- * On the next iteration through however @ref ready is nonzero, and @ref
- * forceplay is 0, so we call @c beforepoll. After the @c poll() we call @c
- * afterpoll and actually get some audio played.
- *
- * This is surely @b far more complicated than it needs to be!
- *
- * If at any call to play(), activate() fails, or if there aren't enough bytes
- * in the buffer to satisfy the request, then @ref forceplay is set non-0. On
- * the next pass through the event loop @c beforepoll is not called. This
- * means that (if none of the other FDs trigger) the @c poll() call will block
- * for up to a second. @c afterpoll will return nonzero, since @c beforepoll
- * wasn't called, and consequently play() is called with @ref forceplay as its
- * argument.
- *
- * The effect is to attempt to restart playing audio - including the activate()
- * step, which may have failed at the previous attempt - at least once a second
- * after an error has disabled it. The delay prevents busy-waiting on whatever
- * condition has rendered the audio device uncooperative.
+ * We want to play audio if there is a current track; and it is not paused; and
+ * there are at least @ref FRAMES frames of audio to play, or we are in sight
+ * of the end of the current track.
*/
+static int playable(void) {
+ return playing
+ && !paused
+ && (playing->used >= FRAMES || playing->eof);
+}
+
+/** @brief Main event loop */
static void mainloop(void) {
struct track *t;
struct speaker_message sm;
- int n, fd, stdin_slot, poke, timeout;
+ int n, fd, stdin_slot, timeout;
while(getppid() != 1) {
fdno = 0;
+ /* By default we will wait up to a second before thinking about current
+ * state. */
+ timeout = 1000;
/* Always ready for commands from the main server. */
stdin_slot = addfd(0, POLLIN);
/* 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->eof && playing->used < playing->size)
playing->slot = addfd(playing->fd, POLLIN);
- } else if(playing)
+ else if(playing)
playing->slot = -1;
- /* If forceplay is set then wait until it succeeds before waiting on the
- * sound device. */
-#if API_ALSA
- alsa_slots = -1;
-#endif
- cmdfd_slot = -1;
- bfd_slot = -1;
- /* By default we will wait up to a second before thinking about current
- * state. */
- timeout = 1000;
- /* We'll break the poll as soon as the underlying sound device is ready for
- * more data */
- if(ready && !forceplay)
- backend->beforepoll();
+ if(playable()) {
+ /* We want to play some audio. If the device is closed then we attempt
+ * to open it. */
+ if(device_state == device_closed)
+ activate();
+ /* If the device is (now) open then we will wait up until it is ready for
+ * more. If something went wrong then we should have device_error
+ * instead, but the post-poll code will cope even if it's
+ * device_closed. */
+ if(device_state == device_open)
+ backend->beforepoll();
+ }
/* If any other tracks don't have a full buffer, try to read sample data
- * from them. */
+ * from them. We do this last of all, so that if we run out of slots,
+ * nothing important can't be monitored. */
for(t = tracks; t; t = t->next)
if(t != playing) {
if(!t->eof && t->used < t->size) {
fatal(errno, "error calling poll");
}
/* Play some sound before doing anything else */
- poke = backend->afterpoll();
- if(poke) {
- /* Some attempt to play must have failed */
- if(playing && !paused)
- play(forceplay);
- else
- forceplay = 0; /* just in case */
+ if(playable()) {
+ /* We want to play some audio */
+ if(device_state == device_open) {
+ if(backend->ready())
+ play(3 * FRAMES);
+ } else {
+ /* We must be in _closed or _error, and it should be the latter, but we
+ * cope with either.
+ *
+ * We most likely timed out, so now is a good time to retry. play()
+ * knows to re-activate the device if necessary.
+ */
+ play(3 * FRAMES);
+ }
}
/* Perhaps we have a command to process */
if(fds[stdin_slot].revents & POLLIN) {
+ /* There might (in theory) be several commands queued up, but in general
+ * this won't be the case, so we don't bother looping around to pick them
+ * all up. */
n = speaker_recv(0, &sm, &fd);
if(n > 0)
switch(sm.type) {
t = findtrack(sm.id, 1);
if(fd != -1) acquire(t, fd);
playing = t;
- play(bufsize);
+ /* We attempt to play straight away rather than going round the loop.
+ * play() is clever enough to perform any activation that is
+ * required. */
+ play(3 * FRAMES);
report();
break;
case SM_PAUSE:
D(("SM_RESUME"));
if(paused) {
paused = 0;
+ /* As for SM_PLAY we attempt to play straight away. */
if(playing)
- play(bufsize);
+ play(3 * FRAMES);
}
report();
break;
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)
+ 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)
~mdw / disorder / blobdiff