/*
* This file is part of DisOrder
- * Copyright (C) 2005, 2006, 2007 Richard Kettlewell
+ * Copyright (C) 2005-2009 Richard Kettlewell
+ * Portions (C) 2007 Mark Wooding
*
- * This program is free software; you can redistribute it and/or modify
+ * This program is free software: you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
+ * the Free Software Foundation, either version 3 of the License, or
* (at your option) any later version.
- *
- * This program is distributed in the hope that it will be useful, but
- * WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
- * General Public License for more details.
- *
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
* You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307
- * USA
+ * along with this program. If not, see <http://www.gnu.org/licenses/>.
*/
/** @file server/speaker.c
- * @brief Speaker processs
+ * @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 via file descriptor
- * passing from the main server and plays them in the right order.
+ * 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 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.
*
- * @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.)
+ * 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.
*
- * When communicating with a subprocess, <a
- * href="http://sox.sourceforge.net/">sox</a> is invoked to convert the inbound
- * data to a single consistent format. The same applies for network (RTP)
- * play, though in that case currently only 44.1KHz 16-bit stereo is supported.
+ * @b Encodings. The encodings supported depend entirely on the uaudio backend
+ * chosen. See @ref uaudio.h, etc.
*
- * The inbound data starts with a structure defining the data format. Note
- * that this is NOT portable between different platforms or even necessarily
- * between versions; the speaker is assumed to be built from the same source
- * and run on the same host as the main server.
+ * 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
* 2-byte samples.
*/
-#include <config.h>
-#include "types.h"
+#include "common.h"
#include <getopt.h>
-#include <stdio.h>
-#include <stdlib.h>
#include <locale.h>
#include <syslog.h>
#include <unistd.h>
#include <errno.h>
#include <ao/ao.h>
-#include <string.h>
-#include <assert.h>
#include <sys/select.h>
#include <sys/wait.h>
#include <time.h>
#include <fcntl.h>
#include <poll.h>
-#include <sys/socket.h>
-#include <netdb.h>
-#include <gcrypt.h>
-#include <sys/uio.h>
+#include <sys/un.h>
+#include <sys/stat.h>
+#include <pthread.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 "addr.h"
-#include "timeval.h"
-#include "rtp.h"
-
-#if API_ALSA
-#include <alsa/asoundlib.h>
-#endif
-
-#ifdef WORDS_BIGENDIAN
-# define MACHINE_AO_FMT AO_FMT_BIG
-#else
-# define MACHINE_AO_FMT AO_FMT_LITTLE
-#endif
-
-/** @brief How many seconds of input to buffer
- *
- * While any given connection has this much audio buffered, no more reads will
- * be issued for that connection. The decoder will have to wait.
- */
-#define BUFFER_SECONDS 5
-
-#define FRAMES 4096 /* Frame batch size */
-
-/** @brief Bytes to send per network packet
- *
- * Don't make this too big or arithmetic will start to overflow.
- */
-#define NETWORK_BYTES (1024+sizeof(struct rtp_header))
-
-/** @brief Maximum RTP playahead (ms) */
-#define RTP_AHEAD_MS 1000
+#include "printf.h"
+#include "version.h"
+#include "uaudio.h"
/** @brief Maximum number of FDs to poll for */
-#define NFDS 256
+#define NFDS 1024
/** @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.
*/
-static struct track {
- struct track *next; /* next track */
+struct track {
+ /** @brief Next track */
+ struct track *next;
+
+ /** @brief Input file descriptor */
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 */
-static time_t last_report; /* when we last reported */
-static int paused; /* pause status */
-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 Track ID */
+ char id[24];
-/** @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 Start position of data in buffer */
+ size_t start;
-static int forceplay; /* frames to force play */
-static int cmdfd = -1; /* child process input */
-static int bfd = -1; /* broadcast FD */
+ /** @brief Number of bytes of data in buffer */
+ size_t used;
-/** @brief RTP timestamp
- *
- * This counts the number of samples played (NB not the number of frames
- * played).
- *
- * The timestamp in the packet header is only 32 bits wide. With 44100Hz
- * stereo, that only gives about half a day before wrapping, which is not
- * particularly convenient for certain debugging purposes. Therefore the
- * timestamp is maintained as a 64-bit integer, giving around six million years
- * before wrapping, and truncated to 32 bits when transmitting.
- */
-static uint64_t rtp_time;
+ /** @brief Set @c fd is at EOF */
+ int eof;
-/** @brief RTP base timestamp
- *
- * This is the real time correspoding to an @ref rtp_time of 0. It is used
- * to recalculate the timestamp after idle periods.
- */
-static struct timeval rtp_time_0;
+ /** @brief Total number of samples played */
+ unsigned long long played;
-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 Slot in @ref fds */
+ int slot;
-/** @brief Structure of a backend */
-struct speaker_backend {
- /** @brief Which backend this is
+ /** @brief Set when playable
*
- * @c -1 terminates the list.
+ * 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 backend;
-
- /** @brief Flags
- *
- * Possible values
- * - @ref FIXED_FORMAT
- */
- unsigned flags;
-/** @brief Lock to configured sample format */
-#define FIXED_FORMAT 0x0001
+ int playable;
- /** @brief Initialization
- *
- * Called once at startup. This is responsible for one-time setup
- * operations, for instance opening a network socket to transmit to.
+ /** @brief Input buffer
*
- * When writing to a native sound API this might @b not imply opening the
- * native sound device - that might be done by @c activate below.
+ * 1Mbyte is enough for nearly 6s of 44100Hz 16-bit stereo
*/
- void (*init)(void);
+ char buffer[1048576];
+};
- /** @brief Activation
- * @return 0 on success, non-0 on error
- *
- * Called to activate the output device.
- *
- * After this function succeeds, @ref ready should be non-0. As well as
- * opening the audio device, this function is responsible for reconfiguring
- * if it necessary to cope with different samples formats (for backends that
- * don't demand a single fixed sample format for the lifetime of the server).
- */
- int (*activate)(void);
+/** @brief Lock protecting data structures
+ *
+ * This lock protects values shared between the main thread and the callback.
+ * It is needed e.g. if changing @ref playing or if modifying buffer pointers.
+ * It is not needed to add a new track, to read values only modified in the
+ * same thread, etc.
+ */
+static pthread_mutex_t lock = PTHREAD_MUTEX_INITIALIZER;
- /** @brief Play sound
- * @param frames Number of frames to play
- * @return Number of frames actually played
- */
- size_t (*play)(size_t frames);
-
- /** @brief Deactivation
- *
- * Called to deactivate the sound device. This is the inverse of
- * @c activate above.
- */
- void (*deactivate)(void);
+/** @brief Linked list of all prepared tracks */
+static struct track *tracks;
- /** @brief Called before poll()
- *
- * Called before the call to poll(). Should call addfd() to update the FD
- * array and stash the slot number somewhere safe.
- */
- void (*beforepoll)(void);
-};
+/** @brief Playing track, or NULL
+ *
+ * This means the DESIRED playing track. It does not reflect any other state
+ * (e.g. activation of uaudio backend).
+ */
+static struct track *playing;
+
+/** @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 speaker_backend *backend;
+static const struct uaudio *backend;
static const struct option options[] = {
{ "help", no_argument, 0, 'h' },
{ "config", required_argument, 0, 'c' },
{ "debug", no_argument, 0, 'd' },
{ "no-debug", no_argument, 0, 'D' },
+ { "syslog", no_argument, 0, 's' },
+ { "no-syslog", no_argument, 0, 'S' },
{ 0, 0, 0, 0 }
};
" --version, -V Display version number\n"
" --config PATH, -c PATH Set configuration file\n"
" --debug, -d Turn on debugging\n"
+ " --[no-]syslog Force logging\n"
"\n"
"Speaker process for DisOrder. Not intended to be run\n"
"directly.\n");
exit(0);
}
-/* Display version number and terminate. */
-static void version(void) {
- xprintf("disorder-speaker version %s\n", disorder_version_string);
- xfclose(stdout);
- exit(0);
-}
-
-/** @brief Return the number of bytes per frame in @p format */
-static size_t bytes_per_frame(const ao_sample_format *format) {
- return format->channels * format->bits / 8;
-}
-
-/** @brief Find track @p id, maybe creating it if not found */
+/** @brief Find track @p id, maybe creating it if not found
+ * @param id Track ID to find
+ * @param create If nonzero, create track structure of @p id not found
+ * @return Pointer to track structure or NULL
+ */
static struct track *findtrack(const char *id, int create) {
struct track *t;
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;
}
-/** @brief Remove track @p id (but do not destroy it) */
+/** @brief Remove track @p id (but do not destroy it)
+ * @param id Track ID to remove
+ * @return Track structure or NULL if not found
+ */
static struct track *removetrack(const char *id) {
struct track *t, **tt;
return t;
}
-/** @brief Destroy a track */
+/** @brief Destroy a track
+ * @param t Track structure
+ */
static void destroy(struct track *t) {
D(("destroy %s", t->id));
- if(t->fd != -1) xclose(t->fd);
- if(t->buffer != (void *)&t->format) free(t->buffer);
- free(t);
-}
-
-/** @brief Notice a new connection */
-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);
-}
-
-/** @brief 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);
-}
-
-/** @brief Compute arguments to sox */
-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;
- }
-}
-
-/** @brief Enable format translation
- *
- * If necessary, replaces a tracks inbound file descriptor with one connected
- * to a sox invocation, which performs the required translation.
- */
-static void enable_translation(struct track *t) {
- if((backend->flags & FIXED_FORMAT)
- && !formats_equal(&t->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, &t->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) {
- signal(SIGPIPE, SIG_DFL);
- xdup2(t->fd, 0);
- xdup2(soxpipe[1], 1);
- fcntl(0, F_SETFL, fcntl(0, F_GETFL) & ~O_NONBLOCK);
- close(soxpipe[0]);
- close(soxpipe[1]);
- close(t->fd);
- execvp("sox", (char **)av);
- _exit(1);
- }
- D(("forking sox for format conversion (kid = %d)", soxkid));
- close(t->fd);
- close(soxpipe[1]);
- t->fd = soxpipe[0];
- t->format = config->sample_format;
- }
+ free(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.
+ * 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) {
+static int speaker_fill(struct track *t) {
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;
+ pthread_mutex_lock(&lock);
+ 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;
+ pthread_mutex_unlock(&lock);
do {
n = read(t->fd, t->buffer + where, left);
} while(n < 0 && errno == EINTR);
+ pthread_mutex_lock(&lock);
if(n < 0) {
- if(errno != EAGAIN) fatal(errno, "error reading sample stream");
- return 0;
- }
- if(n == 0) {
+ if(errno != EAGAIN)
+ fatal(errno, "error reading sample stream");
+ rc = 0;
+ } else if(n == 0) {
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");
- /* If the input format is unsuitable, arrange to translate it */
- enable_translation(t);
- /* 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;
+ pthread_mutex_unlock(&lock);
+ return rc;
}
-/** @brief Close the sound device */
-static void idle(void) {
- D(("idle"));
- if(backend->deactivate)
- backend->deactivate();
- idled = 1;
- ready = 0;
-}
-
-/** @brief 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
-/** @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
+/** @brief Return nonzero if we want to play some audio
*
- * Makes sure the sound device is open and has the right sample format. Return
- * 0 on success and -1 on error.
+ * 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.
*/
-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;
- }
- return backend->activate();
-}
-
-/* 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) {
- 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));
+static int playable(void) {
+ return playing
+ && !paused
+ && playing->playable;
}
-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() */
- 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 */
- avail_bytes = playing->size - playing->start;
- else
- /* The ring buffer is not wrapped, can play the lot */
- avail_bytes = playing->used;
- avail_frames = avail_bytes / bpf;
- /* Only play up to the requested amount */
- if(avail_frames > frames)
- avail_frames = frames;
- if(!avail_frames)
- return;
- /* Play it, Sam */
- written_frames = backend->play(avail_frames);
- written_bytes = written_frames * bpf;
- /* written_bytes and written_frames had better both be set and correct by
- * this point */
- playing->start += written_bytes;
- playing->used -= written_bytes;
- playing->played += written_frames;
- /* If the pointer is at the end of the buffer (or the buffer is completely
- * empty) wrap it back to the start. */
- if(!playing->used || playing->start == playing->size)
- playing->start = 0;
- frames -= written_frames;
-}
-
-/* Notify the server what we're up to. */
+/** @brief Notify the server what we're up to */
static void report(void) {
struct speaker_message sm;
- if(playing && 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);
+ pthread_mutex_lock(&lock);
+ sm.data = playing->played / (uaudio_rate * uaudio_channels);
+ pthread_mutex_unlock(&lock);
+ speaker_send(1, &sm);
}
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;
return -1;
}
-#if API_ALSA
-/** @brief ALSA backend initialization */
-static void alsa_init(void) {
- info("selected ALSA backend");
-}
-
-/** @brief ALSA backend activation */
-static int 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) {
- 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;
-}
-
-/** @brief Play via ALSA */
-static size_t alsa_play(size_t frames) {
- snd_pcm_sframes_t pcm_written_frames;
- int err;
-
- pcm_written_frames = snd_pcm_writei(pcm,
- playing->buffer + playing->start,
- frames);
- D(("actually play %zu frames, wrote %d",
- 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 0;
- case -EAGAIN:
- return 0;
- default:
- fatal(0, "error calling snd_pcm_writei: %d",
- (int)pcm_written_frames);
- }
- } else
- return pcm_written_frames;
-}
-
-static int alsa_slots, alsa_nslots = -1;
-
-/** @brief Fill in poll fd array for ALSA */
-static void alsa_beforepoll(void) {
- /* 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, err;
-
- 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;
-}
-
-/** @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"));
- }
-}
-#endif
-
-/** @brief Command backend initialization */
-static void command_init(void) {
- info("selected command backend");
- fork_cmd();
-}
-
-/** @brief Play to a subprocess */
-static size_t command_play(size_t frames) {
- size_t bytes = frames * bpf;
- int written_bytes;
-
- written_bytes = write(cmdfd, playing->buffer + playing->start, bytes);
- D(("actually play %zu bytes, wrote %d",
- bytes, written_bytes));
- if(written_bytes < 0) {
- switch(errno) {
- case EPIPE:
- error(0, "hmm, command died; trying another");
- fork_cmd();
- return 0;
- case EAGAIN:
- return 0;
- default:
- fatal(errno, "error writing to subprocess");
- }
- } else
- return written_bytes / bpf;
-}
-
-static int cmdfd_slot;
-
-/** @brief Update poll array for writing to subprocess */
-static void command_beforepoll(void) {
- /* 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);
-}
-
-/** @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 */
-static void network_init(void) {
- 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;
- int sndbuf, target_sndbuf = 131072;
- socklen_t len;
- char *sockname, *ssockname;
-
- res = get_address(&config->broadcast, &pref, &sockname);
- if(!res) exit(-1);
- if(config->broadcast_from.n) {
- sres = get_address(&config->broadcast_from, &prefbind, &ssockname);
- if(!sres) exit(-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 setting SO_BROADCAST on broadcast socket");
- len = sizeof sndbuf;
- if(getsockopt(bfd, SOL_SOCKET, SO_SNDBUF,
- &sndbuf, &len) < 0)
- fatal(errno, "error getting SO_SNDBUF");
- if(target_sndbuf > sndbuf) {
- if(setsockopt(bfd, SOL_SOCKET, SO_SNDBUF,
- &target_sndbuf, sizeof target_sndbuf) < 0)
- error(errno, "error setting SO_SNDBUF to %d", target_sndbuf);
- else
- info("changed socket send buffer size from %d to %d",
- sndbuf, target_sndbuf);
- } else
- info("default socket send buffer is %d",
- sndbuf);
- /* 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;
- }
-}
-
-/** @brief Play over the network */
-static size_t network_play(size_t frames) {
- struct rtp_header header;
- struct iovec vec[2];
- size_t bytes = frames * bpf, written_frames;
- int written_bytes;
- /* We transmit using RTP (RFC3550) and attempt to conform to the internet
- * AVT profile (RFC3551). */
-
- if(idled) {
- /* There may have been a gap. Fix up the RTP time accordingly. */
- struct timeval now;
- uint64_t delta;
- uint64_t target_rtp_time;
-
- /* Find the current time */
- xgettimeofday(&now, 0);
- /* Find the number of microseconds elapsed since rtp_time=0 */
- delta = tvsub_us(now, rtp_time_0);
- assert(delta <= UINT64_MAX / 88200);
- target_rtp_time = (delta * playing->format.rate
- * playing->format.channels) / 1000000;
- /* Overflows at ~6 years uptime with 44100Hz stereo */
-
- /* rtp_time is the number of samples we've played. NB that we play
- * RTP_AHEAD_MS ahead of ourselves, so it may legitimately be ahead of
- * the value we deduce from time comparison.
- *
- * Suppose we have 1s track started at t=0, and another track begins to
- * play at t=2s. Suppose RTP_AHEAD_MS=1000 and 44100Hz stereo. In that
- * case we'll send 1s of audio as fast as we can, giving rtp_time=88200.
- * rtp_time stops at this point.
- *
- * At t=2s we'll have calculated target_rtp_time=176400. In this case we
- * set rtp_time=176400 and the player can correctly conclude that it
- * should leave 1s between the tracks.
- *
- * Suppose instead that the second track arrives at t=0.5s, and that
- * we've managed to transmit the whole of the first track already. We'll
- * have target_rtp_time=44100.
- *
- * The desired behaviour is to play the second track back to back with
- * first. In this case therefore we do not modify rtp_time.
- *
- * Is it ever right to reduce rtp_time? No; for that would imply
- * transmitting packets with overlapping timestamp ranges, which does not
- * make sense.
- */
- if(target_rtp_time > rtp_time) {
- /* More time has elapsed than we've transmitted samples. That implies
- * we've been 'sending' silence. */
- info("advancing rtp_time by %"PRIu64" samples",
- target_rtp_time - rtp_time);
- rtp_time = target_rtp_time;
- } else if(target_rtp_time < rtp_time) {
- const int64_t samples_ahead = ((uint64_t)RTP_AHEAD_MS
- * config->sample_format.rate
- * config->sample_format.channels
- / 1000);
-
- if(target_rtp_time + samples_ahead < rtp_time) {
- info("reversing rtp_time by %"PRIu64" samples",
- rtp_time - target_rtp_time);
- }
+/** @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)
+ write(sigpipe[1], "", 1);
+ provided_samples = bytes / uaudio_sample_size;
+ playing->played += provided_samples;
}
}
- header.vpxcc = 2 << 6; /* V=2, P=0, X=0, CC=0 */
- header.seq = htons(rtp_seq++);
- header.timestamp = htonl((uint32_t)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(bytes > NETWORK_BYTES - sizeof header) {
- bytes = NETWORK_BYTES - sizeof header;
- /* Always send a whole number of frames */
- bytes -= bytes % bpf;
+ /* 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;
}
- /* "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 = bytes;
- 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 0;
- } else
- audio_errors /= 2;
- written_bytes -= sizeof (struct rtp_header);
- written_frames = written_bytes / bpf;
- /* Advance RTP's notion of the time */
- rtp_time += written_frames * playing->format.channels;
- return written_frames;
+ pthread_mutex_unlock(&lock);
+ return provided_samples;
}
-static int bfd_slot;
-
-/** @brief Set up poll array for network play */
-static void network_beforepoll(void) {
- struct timeval now;
- uint64_t target_us;
- uint64_t target_rtp_time;
- const int64_t samples_ahead = ((uint64_t)RTP_AHEAD_MS
- * config->sample_format.rate
- * config->sample_format.channels
- / 1000);
-
- /* If we're starting then initialize the base time */
- if(!rtp_time)
- xgettimeofday(&rtp_time_0, 0);
- /* We send audio data whenever we get RTP_AHEAD seconds or more
- * behind */
- xgettimeofday(&now, 0);
- target_us = tvsub_us(now, rtp_time_0);
- assert(target_us <= UINT64_MAX / 88200);
- target_rtp_time = (target_us * config->sample_format.rate
- * config->sample_format.channels)
- / 1000000;
- if((int64_t)(rtp_time - target_rtp_time) < samples_ahead)
- bfd_slot = addfd(bfd, POLLOUT);
-}
-
-/** @brief Table of speaker backends */
-static const struct speaker_backend backends[] = {
-#if API_ALSA
- {
- BACKEND_ALSA,
- 0,
- alsa_init,
- alsa_activate,
- alsa_play,
- alsa_deactivate,
- alsa_beforepoll
- },
-#endif
- {
- BACKEND_COMMAND,
- FIXED_FORMAT,
- command_init,
- generic_activate,
- command_play,
- 0, /* deactivate */
- command_beforepoll
- },
- {
- BACKEND_NETWORK,
- FIXED_FORMAT,
- network_init,
- generic_activate,
- network_play,
- 0, /* deactivate */
- network_beforepoll
- },
- { -1, 0, 0, 0, 0, 0, 0 }
-};
-
-int main(int argc, char **argv) {
- int n, fd, stdin_slot, poke, timeout;
+/** @brief Main event loop */
+static void mainloop(void) {
struct track *t;
struct speaker_message sm;
-#if API_ALSA
- int err;
-#endif
+ int n, fd, stdin_slot, timeout, listen_slot, sigpipe_slot;
- 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()) fatal(0, "cannot read configuration");
- /* ignore SIGPIPE */
- signal(SIGPIPE, SIG_IGN);
- /* reap kids */
- signal(SIGCHLD, reap);
- /* set nice value */
- xnice(config->nice_speaker);
- /* change user */
- become_mortal();
- /* make sure we're not root, whatever the config says */
- if(getuid() == 0 || geteuid() == 0) fatal(0, "do not run as root");
- /* identify the backend used to play */
- for(n = 0; backends[n].backend != -1; ++n)
- if(backends[n].backend == config->speaker_backend)
- break;
- if(backends[n].backend == -1)
- fatal(0, "unsupported backend %d", config->speaker_backend);
- backend = &backends[n];
- /* backend-specific initialization */
- backend->init();
+ /* Keep going while our parent process is alive */
while(getppid() != 1) {
+ int force_report = 0;
+
fdno = 0;
+ /* By default we will wait up to a second before thinking about current
+ * state. */
+ timeout = 1000;
/* 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;
- 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 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;
}
+ sigpipe_slot = addfd(sigpipe[1], POLLIN);
/* 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 */
- 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) {
+ 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*/);
+ if (write(fd, "", 1) < 0) /* write an ack */
+ error(errno, "writing ack to inbound connection");
+ if(t->fd != -1) {
+ error(0, "%s: already got a connection", id);
+ xclose(fd);
+ } else {
+ nonblock(fd);
+ t->fd = fd; /* yay */
+ }
+ }
} else
- poke = 1;
- break;
- }
- if(poke) {
- /* Some attempt to play must have failed */
- if(playing && !paused)
- play(forceplay);
- else
- forceplay = 0; /* just in case */
+ 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");
+ 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);
- report();
+ force_report = 1;
break;
case SM_PAUSE:
D(("SM_PAUSE"));
paused = 1;
- report();
+ force_report = 1;
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:
- D(("SM_CANCEL %s", sm.id));
+ D(("SM_CANCEL %s", sm.id));
t = removetrack(sm.id);
if(t) {
+ pthread_mutex_lock(&lock);
if(t == playing) {
+ /* scratching the playing track */
sm.type = SM_FINISHED;
- strcpy(sm.id, playing->id);
- speaker_send(1, &sm, 0);
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. */
+ info("SM_CANCEL for nonplaying track %s", sm.id);
+ sm.type = SM_STILLBORN;
}
+ strcpy(sm.id, t->id);
destroy(t);
- } else
+ pthread_mutex_unlock(&lock);
+ } 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;
error(0, "SM_CANCEL for unknown track %s", sm.id);
- report();
+ }
+ speaker_send(1, &sm);
+ force_report = 1;
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)))
- 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];
+
+ read(sigpipe[0], buffer, sizeof buffer);
+ }
+ if(playing && playing->used == 0 && playing->eof) {
+ /* The playing track is done. Tell the server, and destroy it. */
+ memset(&sm, 0, sizeof sm);
+ sm.type = SM_FINISHED;
+ strcpy(sm.id, playing->id);
+ speaker_send(1, &sm);
+ removetrack(playing->id);
+ pthread_mutex_lock(&lock);
+ destroy(playing);
+ playing = 0;
+ pthread_mutex_unlock(&lock);
+ /* The server will presumalby send as an SM_PLAY by return */
+ }
+ /* Impose any state change required by the above */
+ if(playable()) {
+ if(!activated) {
+ activated = 1;
+ backend->activate();
+ }
+ } else {
+ if(activated) {
+ activated = 0;
+ backend->deactivate();
+ }
+ }
+ /* If we've not reported our state for a second do so now. */
+ if(force_report || time(0) > last_report)
report();
}
+}
+
+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, "")) 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: 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)) 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)
+ 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)
+ fatal(errno, "getrlimit RLIMIT_NOFILE");
+ if(rl->rlim_cur > NFDS) {
+ rl->rlim_cur = NFDS;
+ if(setrlimit(RLIMIT_NOFILE, rl) < 0)
+ fatal(errno, "setrlimit to reduce RLIMIT_NOFILE to %lu",
+ (unsigned long)rl->rlim_cur);
+ info("set RLIM_NOFILE to %lu", (unsigned long)rl->rlim_cur);
+ } else
+ info("RLIM_NOFILE is %lu", (unsigned long)rl->rlim_cur);
+ /* 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);
+ /* TODO other parameters! */
+ backend = uaudio_find(config->api);
+ /* backend-specific initialization */
+ 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)
+ 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)
+ 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);
+ memset(&sm, 0, sizeof sm);
+ sm.type = SM_READY;
+ speaker_send(1, &sm);
+ mainloop();
info("stopped (parent terminated)");
exit(0);
}
~mdw / disorder / blobdiff