* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307
* USA
*/
-
-/* 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. */
+/** @file server/speaker.c
+ * @brief Speaker process
+ *
+ * This program is responsible for transmitting a single coherent audio stream
+ * to its destination (over the network, to some sound API, to some
+ * subprocess). It receives connections from decoders (or rather from the
+ * process that is about to become disorder-normalize) and plays them in the
+ * right order.
+ *
+ * @b 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.)
+ *
+ * Inbound data is expected to match @c config->sample_format. In normal use
+ * this is arranged by the @c disorder-normalize program (see @ref
+ * server/normalize.c).
+ *
+ * @b Garbage @b Collection. This program deliberately does not use the
+ * garbage collector even though it might be convenient to do so. This is for
+ * two reasons. Firstly some sound APIs use thread threads and we do not want
+ * to have to deal with potential interactions between threading and garbage
+ * collection. Secondly this process needs to be able to respond quickly and
+ * this is not compatible with the collector hanging the program even
+ * relatively briefly.
+ *
+ * @b Units. This program thinks at various times in three different units.
+ * Bytes are obvious. A sample is a single sample on a single channel. A
+ * frame is several samples on different channels at the same point in time.
+ * So (for instance) a 16-bit stereo frame is 4 bytes and consists of a pair of
+ * 2-byte samples.
+ */
#include <config.h>
#include "types.h"
#include <time.h>
#include <fcntl.h>
#include <poll.h>
+#include <sys/un.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 "speaker.h"
-#if API_ALSA
-#include <alsa/asoundlib.h>
-#endif
+/** @brief Linked list of all prepared tracks */
+struct track *tracks;
-#ifdef WORDS_BIGENDIAN
-# define MACHINE_AO_FMT AO_FMT_BIG
-#else
-# define MACHINE_AO_FMT AO_FMT_LITTLE
-#endif
+/** @brief Playing track, or NULL */
+struct track *playing;
-#define BUFFER_SECONDS 5 /* How many seconds of input to
- * buffer. */
-
-#define FRAMES 4096 /* Frame batch size */
-
-#define NFDS 256 /* Max FDs to poll for */
-
-/* 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 Number of bytes pre frame */
+size_t bpf;
+
+/** @brief Array of file descriptors for poll() */
+struct pollfd fds[NFDS];
+
+/** @brief Next free slot in @ref fds */
+int fdno;
+
+/** @brief Listen socket */
+static int listenfd;
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 kidfd = -1; /* child process input */
+
+/** @brief The current device state */
+enum device_states device_state;
+
+/** @brief Set when idled
+ *
+ * This is set when the sound device is deliberately closed by idle().
+ */
+int idled;
+
+/** @brief Selected backend */
+static const struct speaker_backend *backend;
static const struct option options[] = {
{ "help", no_argument, 0, 'h' },
exit(0);
}
-/* Return the number of bytes per frame in FORMAT. */
-static size_t bytes_per_frame(const ao_sample_format *format) {
+/** @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;
}
-/* Find track ID, maybe creating it if not found. */
+/** @brief Find track @p id, maybe creating it if not found */
static struct track *findtrack(const char *id, int create) {
struct track *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;
}
-/* Remove track ID (but do not destroy it). */
+/** @brief Remove track @p id (but do not destroy it) */
static struct track *removetrack(const char *id) {
struct track *t, **tt;
return t;
}
-/* Destroy a track. */
+/** @brief Destroy a track */
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);
- t->fd = fd;
- nonblock(fd);
-}
-
-/* Read data into a sample buffer. Return 0 on success, -1 on EOF. */
+/** @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 fill(struct track *t) {
size_t where, left;
int n;
- D(("fill %s: eof=%d used=%zu size=%zu got_format=%d",
- t->id, t->eof, t->used, t->size, t->got_format));
+ D(("fill %s: eof=%d used=%zu",
+ t->id, t->eof, t->used));
if(t->eof) return -1;
- if(t->used < t->size) {
+ if(t->used < sizeof t->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;
do {
n = read(t->fd, t->buffer + where, left);
} while(n < 0 && errno == EINTR);
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));
- }
}
return 0;
}
-/* 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. */
+/** @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 API_ALSA
- if(!config->speaker_command && 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
- ready = 0;
+ if(backend->deactivate)
+ backend->deactivate();
+ else
+ device_state = device_closed;
+ idled = 1;
}
-/* Abandon the current track */
-static void abandon(void) {
+/** @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, 0);
+ speaker_send(1, &sm);
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;
- }
- if(kidfd >= 0) {
- 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;
- }
- if(config->speaker_command)
- return -1;
-#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;
- }
-#endif
- return -1;
+/** @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;
}
-/* Check to see whether the current track has finished playing */
+/** @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->got_format
- || playing->used < bytes_per_frame(&playing->format)))
+ && playing->used < bytes_per_frame(&config->sample_format))
abandon();
}
-static void fork_kid(void) {
- pid_t kid;
- int pfd[2];
- if(kidfd != -1) close(kidfd);
- xpipe(pfd);
- kid = xfork();
- if(!kid) {
- 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]);
- kidfd = pfd[1];
- D(("forked kid %d, fd = %d", kid, kidfd));
-}
-
+/** @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_bytes, written_frames;
+ size_t avail_frames, avail_bytes, written_frames;
ssize_t written_bytes;
- 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 */
+ 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" : "",
- 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;
+ 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 > playing->size)
- avail_bytes = playing->size - playing->start;
+ 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;
-
- if(!config->speaker_command) {
-#if API_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;
-#else
- assert(!"reached");
-#endif
- } else {
- if(avail_bytes > frames * bpf)
- avail_bytes = frames * bpf;
- written_bytes = write(kidfd, 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, kid died; trying another");
- fork_kid();
- return;
- case EAGAIN:
- return;
- }
- }
- written_frames = written_bytes / bpf; /* good enough */
- }
+ 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 == playing->size)
+ if(!playing->used || playing->start == (sizeof playing->buffer))
playing->start = 0;
frames -= written_frames;
+ return;
}
/* Notify the server what we're up to. */
static void report(void) {
struct speaker_message sm;
- if(playing && playing->buffer != (void *)&playing->format) {
+ if(playing) {
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 / config->sample_format.rate;
+ speaker_send(1, &sm);
}
time(&last_report);
}
static void reap(int __attribute__((unused)) sig) {
- pid_t kid;
+ pid_t cmdpid;
int st;
do
- kid = waitpid(-1, &st, WNOHANG);
- while(kid > 0);
+ cmdpid = waitpid(-1, &st, WNOHANG);
+ while(cmdpid > 0);
signal(SIGCHLD, reap);
}
-static int addfd(int fd, int events) {
+int addfd(int fd, int events) {
if(fdno < NFDS) {
fds[fdno].fd = fd;
fds[fdno].events = events;
return -1;
}
-int main(int argc, char **argv) {
- int n, fd, stdin_slot, alsa_slots, kid_slot;
- struct track *t;
- struct speaker_message sm;
+/** @brief Table of speaker backends */
+static const struct speaker_backend *backends[] = {
#if API_ALSA
- int alsa_nslots = -1, err;
+ &alsa_backend,
#endif
+ &command_backend,
+ &network_backend,
+ 0
+};
+
+/** @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
+ * 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, timeout, listen_slot;
- set_progname(argv);
- mem_init(0);
- 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");
- }
- }
- 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");
- info("started");
- if(config->speaker_command)
- fork_kid();
- else {
-#if API_ALSA
- /* ok */
-#else
- fatal(0, "invoked speaker but no speaker_command and no known sound API");
- #endif
- }
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);
+ /* Also always ready for inbound connections */
+ listen_slot = addfd(listenfd, POLLIN);
/* Try to read sample data for the currently playing track if there is
* buffer space. */
- if(playing && !playing->eof && playing->used < playing->size) {
+ if(playing
+ && playing->fd >= 0
+ && !playing->eof
+ && playing->used < (sizeof playing->buffer))
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. */
- alsa_slots = -1;
- kid_slot = -1;
- if(ready && !forceplay) {
- if(config->speaker_command) {
- if(kidfd >= 0)
- kid_slot = addfd(kidfd, POLLOUT);
- } else {
-#if API_ALSA
- 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;
-#endif
- }
+ 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) {
+ if(t->fd >= 0
+ && !t->eof
+ && t->used < sizeof t->buffer) {
t->slot = addfd(t->fd, POLLIN | POLLHUP);
} else
t->slot = -1;
}
- /* Wait up to a second before thinking about current state */
- n = poll(fds, fdno, 1000);
+ /* Wait for something interesting to happen */
+ n = poll(fds, fdno, timeout);
if(n < 0) {
if(errno == EINTR) continue;
fatal(errno, "error calling poll");
}
/* Play some sound before doing anything else */
- if(alsa_slots != -1) {
-#if API_ALSA
- 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);
-#endif
- } else if(kid_slot != -1) {
- if(fds[kid_slot].revents & (POLLOUT | POLLERR))
+ 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);
- } else {
- /* Some attempt to play must have failed */
- if(playing && !paused)
- play(forceplay);
- else
- forceplay = 0; /* just in case */
+ }
+ }
+ /* 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) {
+ if(read(fd, &l, sizeof l) < 4) {
+ error(errno, "reading length from inbound connection");
+ xclose(fd);
+ } else if(l >= sizeof id) {
+ error(0, "id length too long");
+ xclose(fd);
+ } else if(read(fd, id, l) < (ssize_t)l) {
+ error(errno, "reading id from inbound connection");
+ xclose(fd);
+ } else {
+ id[l] = 0;
+ D(("id %s fd %d", id, fd));
+ t = findtrack(id, 1/*create*/);
+ write(fd, "", 1); /* write an ack */
+ if(t->fd != -1) {
+ error(0, "got a connection for a track that already has one");
+ xclose(fd);
+ } else {
+ nonblock(fd);
+ t->fd = fd; /* yay */
+ }
+ }
+ } else
+ error(errno, "accept");
}
/* 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);
+ /* TODO */
if(n > 0)
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:
- D(("SM_PLAY %s %d", sm.id, fd));
if(playing) fatal(0, "got SM_PLAY but already playing something");
t = findtrack(sm.id, 1);
- if(fd != -1) acquire(t, fd);
+ 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;
- 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;
if(t == playing) {
sm.type = SM_FINISHED;
strcpy(sm.id, playing->id);
- speaker_send(1, &sm, 0);
+ speaker_send(1, &sm);
playing = 0;
}
destroy(t);
break;
case SM_RELOAD:
D(("SM_RELOAD"));
- if(config_read()) error(0, "cannot read configuration");
+ if(config_read(1)) error(0, "cannot read configuration");
info("reloaded configuration");
break;
default:
}
/* Read in any buffered data */
for(t = tracks; t; t = t->next)
- if(t->slot != -1 && (fds[t->slot].revents & (POLLIN | POLLHUP)))
+ if(t->fd != -1
+ && 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)
report();
}
+}
+
+int main(int argc, char **argv) {
+ int n;
+ struct sockaddr_un addr;
+ static const int one = 1;
+
+ 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");
+ }
+ }
+ 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(1)) fatal(0, "cannot read configuration");
+ bpf = bytes_per_frame(&config->sample_format);
+ /* ignore SIGPIPE */
+ signal(SIGPIPE, SIG_IGN);
+ /* reap kids */
+ signal(SIGCHLD, reap);
+ /* set nice value */
+ xnice(config->nice_speaker);
+ /* change user */
+ become_mortal();
+ /* make sure we're not root, whatever the config says */
+ if(getuid() == 0 || geteuid() == 0) fatal(0, "do not run as root");
+ /* identify the backend used to play */
+ for(n = 0; backends[n]; ++n)
+ if(backends[n]->backend == config->speaker_backend)
+ break;
+ if(!backends[n])
+ fatal(0, "unsupported backend %d", config->speaker_backend);
+ backend = backends[n];
+ /* backend-specific initialization */
+ backend->init();
+ /* 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",
+ config->home);
+ if(unlink(addr.sun_path) < 0 && errno != ENOENT)
+ error(errno, "removing %s", addr.sun_path);
+ xsetsockopt(listenfd, SOL_SOCKET, SO_REUSEADDR, &one, sizeof one);
+ if(bind(listenfd, (const struct sockaddr *)&addr, sizeof addr) < 0)
+ fatal(errno, "error binding socket to %s", addr.sun_path);
+ xlisten(listenfd, 128);
+ nonblock(listenfd);
+ info("listening on %s", addr.sun_path);
+ mainloop();
info("stopped (parent terminated)");
exit(0);
}
~mdw / disorder / blobdiff