/*
* This file is part of DisOrder
- * Copyright (C) 2005, 2006, 2007 Richard Kettlewell
+ * Copyright (C) 2005-2008 Richard Kettlewell
+ * Portions (C) 2007 Mark Wooding
*
* 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
* USA
*/
/** @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.
*
- * For the <a href="http://www.alsa-project.org/">ALSA</a> API, 8- and 16- bit
- * stereo and mono are supported, with any sample rate (within the limits that
- * ALSA can deal with.)
+ * @b 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.)
*
- * 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.
+ * 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).
*
- * 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.
+7 * @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.
*
- * 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 <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 "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"
+#include "speaker.h"
+#include "printf.h"
+#include "version.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;
-/** @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
+/** @brief Number of bytes pre frame */
+size_t bpf;
-#define FRAMES 4096 /* Frame batch size */
+/** @brief Array of file descriptors for poll() */
+struct pollfd fds[NFDS];
-/** @brief Bytes to send per network packet
- *
- * Don't make this too big or arithmetic will start to overflow.
- */
-#define NETWORK_BYTES 1024
-
-/** @brief Maximum RTP playahead (seconds) */
-#define RTP_AHEAD 2
-
-/** @brief Maximum number of FDs to poll for */
-#define NFDS 256
+/** @brief Next free slot in @ref fds */
+int fdno;
-/** @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 */
- 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 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 cmdfd = -1; /* child process input */
-static int bfd = -1; /* broadcast FD */
-/** @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 The current device state */
+enum device_states device_state;
-/** @brief RTP base timestamp
+/** @brief Set when idled
*
- * This is the real time correspoding to an @ref rtp_time of 0. It is used
- * to recalculate the timestamp after idle periods.
+ * This is set when the sound device is deliberately closed by idle().
*/
-static struct timeval rtp_time_0;
+int idled;
-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 Selected backend */
+static const struct speaker_backend *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) {
+static size_t bytes_per_frame(const struct stream_header *format) {
return format->channels * format->bits / 8;
}
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;
}
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) {
- switch(config->speaker_backend) {
- case BACKEND_COMMAND:
- case BACKEND_NETWORK:
- /* These backends need a specific sample format */
- break;
- case BACKEND_ALSA:
- /* ALSA can cope */
- return;
- }
- if(!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;
- ready = 0;
- }
-}
-
/** @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;
- 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);
if(n == 0) {
D(("fill %s: eof detected", t->id));
t->eof = 1;
+ t->playable = 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));
- }
+ if(t->used == sizeof t->buffer)
+ t->playable = 1;
}
return 0;
}
-/** @brief 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_backend == BACKEND_ALSA && pcm) {
- int err;
-
- if((err = snd_pcm_nonblock(pcm, 0)) < 0)
- fatal(0, "error calling snd_pcm_nonblock: %d", err);
- D(("draining pcm"));
- snd_pcm_drain(pcm);
- D(("closing pcm"));
- snd_pcm_close(pcm);
- pcm = 0;
- forceplay = 0;
- D(("released audio device"));
- }
-#endif
+ if(backend->deactivate)
+ backend->deactivate();
+ else
+ device_state = device_closed;
idled = 1;
- ready = 0;
}
/** @brief Abandon the current track */
-static void abandon(void) {
+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
-/** @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) {
- /* If we don't know the format yet we cannot start. */
- if(!playing->got_format) {
- D((" - not got format for %s", playing->id));
- return -1;
- }
- switch(config->speaker_backend) {
- case BACKEND_COMMAND:
- case BACKEND_NETWORK:
- if(!ready) {
- pcm_format = config->sample_format;
- bufsize = 3 * FRAMES;
- bpf = bytes_per_frame(&config->sample_format);
- D(("acquired audio device"));
- ready = 1;
- }
- return 0;
- case BACKEND_ALSA:
-#if API_ALSA
- /* If we need to change format then close the current device. */
- if(pcm && !formats_equal(&playing->format, &pcm_format))
- idle();
- if(!pcm) {
- snd_pcm_hw_params_t *hwparams;
- snd_pcm_sw_params_t *swparams;
- snd_pcm_uframes_t pcm_bufsize;
- int err;
- int sample_format = 0;
- unsigned rate;
-
- D(("snd_pcm_open"));
- if((err = snd_pcm_open(&pcm,
- config->device,
- SND_PCM_STREAM_PLAYBACK,
- SND_PCM_NONBLOCK))) {
- error(0, "error from snd_pcm_open: %d", err);
- goto error;
- }
- snd_pcm_hw_params_alloca(&hwparams);
- D(("set up hw params"));
- if((err = snd_pcm_hw_params_any(pcm, hwparams)) < 0)
- fatal(0, "error from snd_pcm_hw_params_any: %d", err);
- if((err = snd_pcm_hw_params_set_access(pcm, hwparams,
- SND_PCM_ACCESS_RW_INTERLEAVED)) < 0)
- fatal(0, "error from snd_pcm_hw_params_set_access: %d", err);
- switch(playing->format.bits) {
- case 8:
- sample_format = SND_PCM_FORMAT_S8;
- break;
- case 16:
- switch(playing->format.byte_format) {
- case AO_FMT_NATIVE: sample_format = SND_PCM_FORMAT_S16; break;
- case AO_FMT_LITTLE: sample_format = SND_PCM_FORMAT_S16_LE; break;
- case AO_FMT_BIG: sample_format = SND_PCM_FORMAT_S16_BE; break;
- error(0, "unrecognized byte format %d", playing->format.byte_format);
- goto fatal;
- }
- break;
- default:
- error(0, "unsupported sample size %d", playing->format.bits);
- goto fatal;
- }
- if((err = snd_pcm_hw_params_set_format(pcm, hwparams,
- sample_format)) < 0) {
- error(0, "error from snd_pcm_hw_params_set_format (%d): %d",
- sample_format, err);
- goto fatal;
- }
- rate = playing->format.rate;
- if((err = snd_pcm_hw_params_set_rate_near(pcm, hwparams, &rate, 0)) < 0) {
- error(0, "error from snd_pcm_hw_params_set_rate (%d): %d",
- playing->format.rate, err);
- goto fatal;
- }
- if(rate != (unsigned)playing->format.rate)
- info("want rate %d, got %u", playing->format.rate, rate);
- if((err = snd_pcm_hw_params_set_channels(pcm, hwparams,
- playing->format.channels)) < 0) {
- error(0, "error from snd_pcm_hw_params_set_channels (%d): %d",
- playing->format.channels, err);
- goto fatal;
- }
- bufsize = 3 * FRAMES;
- pcm_bufsize = bufsize;
- if((err = snd_pcm_hw_params_set_buffer_size_near(pcm, hwparams,
- &pcm_bufsize)) < 0)
- fatal(0, "error from snd_pcm_hw_params_set_buffer_size (%d): %d",
- 3 * FRAMES, err);
- if(pcm_bufsize != 3 * FRAMES && pcm_bufsize != last_pcm_bufsize)
- info("asked for PCM buffer of %d frames, got %d",
- 3 * FRAMES, (int)pcm_bufsize);
- last_pcm_bufsize = pcm_bufsize;
- if((err = snd_pcm_hw_params(pcm, hwparams)) < 0)
- fatal(0, "error calling snd_pcm_hw_params: %d", err);
- D(("set up sw params"));
- snd_pcm_sw_params_alloca(&swparams);
- if((err = snd_pcm_sw_params_current(pcm, swparams)) < 0)
- fatal(0, "error calling snd_pcm_sw_params_current: %d", err);
- if((err = snd_pcm_sw_params_set_avail_min(pcm, swparams, FRAMES)) < 0)
- fatal(0, "error calling snd_pcm_sw_params_set_avail_min %d: %d",
- FRAMES, err);
- if((err = snd_pcm_sw_params(pcm, swparams)) < 0)
- fatal(0, "error calling snd_pcm_sw_params: %d", err);
- pcm_format = playing->format;
- bpf = bytes_per_frame(&pcm_format);
- D(("acquired audio device"));
- log_params(hwparams, swparams);
- ready = 1;
- }
- return 0;
- fatal:
- abandon();
- error:
- /* We assume the error is temporary and that we'll retry in a bit. */
- if(pcm) {
- snd_pcm_close(pcm);
- pcm = 0;
- }
- return -1;
-#endif
- default:
- assert(!"reached");
- }
+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_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 Return nonzero if we want to play some audio
+ *
+ * We want to play audio if there is a current track; and it is not paused; and
+ * it is playable according to the rules for @ref track::playable.
+ */
+static int playable(void) {
+ return playing
+ && !paused
+ && playing->playable;
}
-static void play(size_t frames) {
- size_t avail_bytes, write_bytes, written_frames;
+/** @brief Play up to @p frames frames of audio
+ *
+ * It is always safe to call this function.
+ * - If @ref playing is 0 then it will just return
+ * - If @ref paused is non-0 then it will just return
+ * - If @ref device_state != @ref device_open then it will call activate() and
+ * return if it it fails.
+ * - If there is not enough audio to play then it play what is available.
+ *
+ * If there are not enough frames to play then whatever is available is played
+ * instead. It is up to mainloop() to ensure that speaker_play() is not called
+ * when unreasonably only an small amounts of data is available to play.
+ */
+static void speaker_play(size_t frames) {
+ size_t avail_frames, avail_bytes, written_frames;
ssize_t written_bytes;
- struct rtp_header header;
- struct iovec vec[2];
- 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 paused */
+ if(!playable())
return;
+ /* Make sure the output device is open */
+ if(device_state != device_open) {
+ activate();
+ if(device_state != device_open)
+ return;
}
D(("play: play %zu/%zu%s %dHz %db %dc", frames, playing->used / bpf,
playing->eof ? " EOF" : "",
- 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;
-
- switch(config->speaker_backend) {
-#if API_ALSA
- case BACKEND_ALSA: {
- snd_pcm_sframes_t pcm_written_frames;
- size_t avail_frames;
- int err;
-
- avail_frames = avail_bytes / bpf;
- if(avail_frames > frames)
- avail_frames = frames;
- if(!avail_frames)
- return;
- pcm_written_frames = snd_pcm_writei(pcm,
- playing->buffer + playing->start,
- avail_frames);
- D(("actually play %zu frames, wrote %d",
- avail_frames, (int)pcm_written_frames));
- if(pcm_written_frames < 0) {
- switch(pcm_written_frames) {
- case -EPIPE: /* underrun */
- error(0, "snd_pcm_writei reports underrun");
- if((err = snd_pcm_prepare(pcm)) < 0)
- fatal(0, "error calling snd_pcm_prepare: %d", err);
- return;
- case -EAGAIN:
- return;
- default:
- fatal(0, "error calling snd_pcm_writei: %d",
- (int)pcm_written_frames);
- }
- }
- written_frames = pcm_written_frames;
- written_bytes = written_frames * bpf;
- break;
- }
-#endif
- case BACKEND_COMMAND:
- if(avail_bytes > frames * bpf)
- avail_bytes = frames * bpf;
- written_bytes = write(cmdfd, playing->buffer + playing->start,
- avail_bytes);
- D(("actually play %zu bytes, wrote %d",
- avail_bytes, (int)written_bytes));
- if(written_bytes < 0) {
- switch(errno) {
- case EPIPE:
- error(0, "hmm, command died; trying another");
- fork_cmd();
- return;
- case EAGAIN:
- return;
- }
- }
- written_frames = written_bytes / bpf; /* good enough */
- break;
- case BACKEND_NETWORK:
- /* We transmit using RTP (RFC3550) and attempt to conform to the internet
- * AVT profile (RFC3551). */
-
- if(idled) {
- /* There's 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 */
- if(target_rtp_time > rtp_time)
- info("advancing rtp_time by %"PRIu64" samples",
- target_rtp_time - rtp_time);
- else if(target_rtp_time < rtp_time)
- info("reversing rtp_time by %"PRIu64" samples",
- rtp_time - target_rtp_time);
- rtp_time = target_rtp_time;
- }
- 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(avail_bytes > NETWORK_BYTES - sizeof header) {
- avail_bytes = NETWORK_BYTES - sizeof header;
- /* Always send a whole number of frames */
- avail_bytes -= avail_bytes % bpf;
- }
- /* "The RTP clock rate used for generating the RTP timestamp is independent
- * of the number of channels and the encoding; it equals the number of
- * sampling periods per second. For N-channel encodings, each sampling
- * period (say, 1/8000 of a second) generates N samples. (This terminology
- * is standard, but somewhat confusing, as the total number of samples
- * generated per second is then the sampling rate times the channel
- * count.)"
- */
- write_bytes = avail_bytes;
- if(write_bytes) {
- vec[0].iov_base = (void *)&header;
- vec[0].iov_len = sizeof header;
- vec[1].iov_base = playing->buffer + playing->start;
- vec[1].iov_len = avail_bytes;
- 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;
- }
- } else
- audio_errors /= 2;
- written_bytes = avail_bytes;
- written_frames = written_bytes / bpf;
- /* Advance RTP's notion of the time */
- rtp_time += written_frames * playing->format.channels;
- break;
- default:
- assert(!"reached");
- }
+ 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->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;
+ /* If the buffer emptied out mark the track as unplayably */
+ if(!playing->used && !playing->eof) {
+ error(0, "track buffer emptied");
+ playing->playable = 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);
}
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, cmdfd_slot, bfd_slot, poke, timeout;
+/** @brief Table of speaker backends */
+static const struct speaker_backend *backends[] = {
+#if HAVE_ALSA_ASOUNDLIB_H
+ &alsa_backend,
+#endif
+ &command_backend,
+ &network_backend,
+#if HAVE_COREAUDIO_AUDIOHARDWARE_H
+ &coreaudio_backend,
+#endif
+#if HAVE_SYS_SOUNDCARD_H
+ &oss_backend,
+#endif
+ 0
+};
+
+/** @brief Main event loop */
+static void mainloop(void) {
struct track *t;
struct speaker_message sm;
- struct addrinfo *res, *sres;
- static const struct addrinfo pref = {
- 0,
- PF_INET,
- SOCK_DGRAM,
- IPPROTO_UDP,
- 0,
- 0,
- 0,
- 0
- };
- static const struct addrinfo prefbind = {
- AI_PASSIVE,
- PF_INET,
- SOCK_DGRAM,
- IPPROTO_UDP,
- 0,
- 0,
- 0,
- 0
- };
- static const int one = 1;
- char *sockname, *ssockname;
-#if API_ALSA
- int alsa_nslots = -1, err;
-#endif
+ int n, fd, stdin_slot, timeout, listen_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");
- switch(config->speaker_backend) {
- case BACKEND_ALSA:
- info("selected ALSA backend");
- case BACKEND_COMMAND:
- info("selected command backend");
- fork_cmd();
- break;
- case BACKEND_NETWORK:
- res = get_address(&config->broadcast, &pref, &sockname);
- if(!res) return -1;
- if(config->broadcast_from.n) {
- sres = get_address(&config->broadcast_from, &prefbind, &ssockname);
- if(!sres) return -1;
- } else
- sres = 0;
- if((bfd = socket(res->ai_family,
- res->ai_socktype,
- res->ai_protocol)) < 0)
- fatal(errno, "error creating broadcast socket");
- if(setsockopt(bfd, SOL_SOCKET, SO_BROADCAST, &one, sizeof one) < 0)
- fatal(errno, "error settting SO_BROADCAST on broadcast socket");
- /* We might well want to set additional broadcast- or multicast-related
- * options here */
- if(sres && bind(bfd, sres->ai_addr, sres->ai_addrlen) < 0)
- fatal(errno, "error binding broadcast socket to %s", ssockname);
- if(connect(bfd, res->ai_addr, res->ai_addrlen) < 0)
- fatal(errno, "error connecting broadcast socket to %s", sockname);
- /* Select an SSRC */
- gcry_randomize(&rtp_id, sizeof rtp_id, GCRY_STRONG_RANDOM);
- info("selected network backend, sending to %s", sockname);
- if(config->sample_format.byte_format != AO_FMT_BIG) {
- info("forcing big-endian sample format");
- config->sample_format.byte_format = AO_FMT_BIG;
- }
- break;
- default:
- fatal(0, "unknown backend %d", config->speaker_backend);
- }
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;
- cmdfd_slot = -1;
- bfd_slot = -1;
- /* By default we will wait up to a second before thinking about current
- * state. */
- timeout = 1000;
- if(ready && !forceplay) {
- switch(config->speaker_backend) {
- case BACKEND_COMMAND:
- /* We send sample data to the subprocess as fast as it can accept it.
- * This isn't ideal as pause latency can be very high as a result. */
- if(cmdfd >= 0)
- cmdfd_slot = addfd(cmdfd, POLLOUT);
- break;
- case BACKEND_NETWORK: {
- struct timeval now;
- uint64_t target_us;
- uint64_t target_rtp_time;
- const uint64_t ahead = RTP_AHEAD * config->sample_format.rate
- * config->sample_format.channels;
-
- static unsigned logit;
-
- /* 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;
- /* TODO remove logging guff */
- if(!(logit++ & 1023))
- info("rtp_time %llu target %llu difference %lld [%lld]",
- rtp_time, target_rtp_time,
- rtp_time - target_rtp_time,
- ahead);
- if(rtp_time < target_rtp_time
- || rtp_time - target_rtp_time < ahead)
- bfd_slot = addfd(bfd, POLLOUT);
- break;
- }
-#if API_ALSA
- case BACKEND_ALSA: {
- /* We send sample data to ALSA as fast as it can accept it, relying on
- * the fact that it has a relatively small buffer to minimize pause
- * latency. */
- int retry = 3;
-
- alsa_slots = fdno;
- do {
- retry = 0;
- alsa_nslots = snd_pcm_poll_descriptors(pcm, &fds[fdno], NFDS - fdno);
- if((alsa_nslots <= 0
- || !(fds[alsa_slots].events & POLLOUT))
- && snd_pcm_state(pcm) == SND_PCM_STATE_XRUN) {
- error(0, "underrun detected after call to snd_pcm_poll_descriptors()");
- if((err = snd_pcm_prepare(pcm)))
- fatal(0, "error calling snd_pcm_prepare: %d", err);
- } else
- break;
- } while(retry-- > 0);
- if(alsa_nslots >= 0)
- fdno += alsa_nslots;
- break;
- }
-#endif
- default:
- assert(!"unknown backend");
- }
+ if(playable()) {
+ /* We want to play some audio. If the device is closed then we attempt
+ * to open it. */
+ if(device_state == device_closed)
+ activate();
+ /* If the device is (now) open then we will wait up until it is ready for
+ * more. If something went wrong then we should have device_error
+ * instead, but the post-poll code will cope even if it's
+ * device_closed. */
+ if(device_state == device_open)
+ backend->beforepoll(&timeout);
}
/* 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;
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);
- } else
- poke = 1;
- break;
+ if(playable()) {
+ /* We want to play some audio */
+ if(device_state == device_open) {
+ if(backend->ready())
+ speaker_play(3 * FRAMES);
+ } else {
+ /* We must be in _closed or _error, and it should be the latter, but we
+ * cope with either.
+ *
+ * We most likely timed out, so now is a good time to retry.
+ * speaker_play() knows to re-activate the device if necessary.
+ */
+ speaker_play(3 * FRAMES);
+ }
}
- if(poke) {
- /* 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) {
+ 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*/);
+ write(fd, "", 1); /* write an ack */
+ if(t->fd != -1) {
+ error(0, "%s: already got a connection", id);
+ 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.
+ * speaker_play() is clever enough to perform any activation that is
+ * required. */
+ speaker_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);
+ speaker_play(3 * FRAMES);
}
report();
break;
case SM_CANCEL:
- D(("SM_CANCEL %s", sm.id));
+ D(("SM_CANCEL %s", sm.id));
t = removetrack(sm.id);
if(t) {
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
+ } 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);
+ }
+ speaker_send(1, &sm);
report();
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);
+ if(t->fd != -1
+ && t->slot != -1
+ && (fds[t->slot].revents & (POLLIN | POLLHUP)))
+ speaker_fill(t);
/* Maybe we finished playing a track somewhere in the above */
maybe_finished();
/* If we don't need the sound device for now then close it for the benefit
* of anyone else who wants it. */
- if((!playing || paused) && 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, logsyslog = !isatty(2);
+ struct sockaddr_un addr;
+ static const int one = 1;
+ struct speaker_message sm;
+ const char *d;
+ char *dir;
+
+ 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;
+ }
+ 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->api)
+ break;
+ if(!backends[n])
+ fatal(0, "unsupported api %d", config->api);
+ backend = backends[n];
+ /* backend-specific initialization */
+ backend->init();
+ /* 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