X-Git-Url: http://www.chiark.greenend.org.uk/ucgi/~mdw/git/disorder/blobdiff_plain/6d2d327ca57fefaddceba10eb323451f8150e95d..3fbdc96d45fbf2abcc93ed2e8ad206bc540be92b:/server/speaker.c
diff --git a/server/speaker.c b/server/speaker.c
index aa09c02..4a8966d 100644
--- a/server/speaker.c
+++ b/server/speaker.c
@@ -1,6 +1,7 @@
/*
* This file is part of DisOrder
* Copyright (C) 2005, 2006, 2007 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
@@ -22,8 +23,9 @@
*
* 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 Encodings. For the ALSA API,
* 8- and 16- bit stereo and mono are supported, with any sample rate (within
@@ -33,7 +35,7 @@
* 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
+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
@@ -66,6 +68,8 @@
#include
#include
#include
+#include
+#include
#include "configuration.h"
#include "syscalls.h"
@@ -75,6 +79,8 @@
#include "speaker-protocol.h"
#include "user.h"
#include "speaker.h"
+#include "printf.h"
+#include "version.h"
/** @brief Linked list of all prepared tracks */
struct track *tracks;
@@ -91,6 +97,9 @@ struct pollfd fds[NFDS];
/** @brief Next free slot in @ref fds */
int fdno;
+/** @brief Listen socket */
+static int listenfd;
+
static time_t last_report; /* when we last reported */
static int paused; /* pause status */
@@ -112,6 +121,8 @@ static const struct option options[] = {
{ "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 }
};
@@ -124,6 +135,7 @@ static void help(void) {
" --version, -V Display version number\n"
" --config PATH, -c PATH Set configuration file\n"
" --debug, -d Turn on debugging\n"
+ " --[no-]syslog Force logging\n"
"\n"
"Speaker process for DisOrder. Not intended to be run\n"
"directly.\n");
@@ -131,13 +143,6 @@ static void help(void) {
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 struct stream_header *format) {
return format->channels * format->bits / 8;
@@ -179,15 +184,6 @@ static void destroy(struct track *t) {
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 Read data into a sample buffer
* @param t Pointer to track
* @return 0 on success, -1 on EOF
@@ -196,7 +192,7 @@ static void acquire(struct track *t, int fd) {
* 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;
@@ -219,9 +215,12 @@ static int fill(struct track *t) {
if(n == 0) {
D(("fill %s: eof detected", t->id));
t->eof = 1;
+ t->playable = 1;
return -1;
}
t->used += n;
+ if(t->used == sizeof t->buffer)
+ t->playable = 1;
}
return 0;
}
@@ -249,7 +248,7 @@ void abandon(void) {
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;
@@ -281,6 +280,17 @@ static void maybe_finished(void) {
abandon();
}
+/** @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;
+}
+
/** @brief Play up to @p frames frames of audio
*
* It is always safe to call this function.
@@ -291,15 +301,15 @@ static void maybe_finished(void) {
* - If there is not enough audio to play then it play what is available.
*
* If there are not enough frames to play then whatever is available is played
- * instead. It is up to mainloop() to ensure that play() is not called when
- * unreasonably only an small amounts of data is available to play.
+ * 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 play(size_t frames) {
+static void speaker_play(size_t frames) {
size_t avail_frames, avail_bytes, written_frames;
ssize_t written_bytes;
- /* Make sure there's a track to play and it is not pasued */
- if(!playing || paused)
+ /* 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) {
@@ -337,6 +347,11 @@ static void play(size_t frames) {
* empty) wrap it back to the start. */
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;
}
@@ -350,7 +365,7 @@ static void report(void) {
sm.type = paused ? SM_PAUSED : SM_PLAYING;
strcpy(sm.id, playing->id);
sm.data = playing->played / config->sample_format.rate;
- speaker_send(1, &sm, 0);
+ speaker_send(1, &sm);
}
time(&last_report);
}
@@ -376,31 +391,25 @@ int addfd(int fd, int events) {
/** @brief Table of speaker backends */
static const struct speaker_backend *backends[] = {
-#if API_ALSA
+#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 Return nonzero if we want to play some audio
- *
- * We want to play audio if there is a current track; and it is not paused; and
- * there are at least @ref FRAMES frames of audio to play, or we are in sight
- * of the end of the current track.
- */
-static int playable(void) {
- return playing
- && !paused
- && (playing->used >= FRAMES || playing->eof);
-}
-
/** @brief Main event loop */
static void mainloop(void) {
struct track *t;
struct speaker_message sm;
- int n, fd, stdin_slot, timeout;
+ int n, fd, stdin_slot, timeout, listen_slot;
while(getppid() != 1) {
fdno = 0;
@@ -409,9 +418,14 @@ static void mainloop(void) {
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 < (sizeof playing->buffer))
+ if(playing
+ && playing->fd >= 0
+ && !playing->eof
+ && playing->used < (sizeof playing->buffer))
playing->slot = addfd(playing->fd, POLLIN);
else if(playing)
playing->slot = -1;
@@ -425,14 +439,16 @@ static void mainloop(void) {
* instead, but the post-poll code will cope even if it's
* device_closed. */
if(device_state == device_open)
- backend->beforepoll();
+ backend->beforepoll(&timeout);
}
/* If any other tracks don't have a full buffer, try to read sample data
* 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 < sizeof t->buffer) {
+ if(t->fd >= 0
+ && !t->eof
+ && t->used < sizeof t->buffer) {
t->slot = addfd(t->fd, POLLIN | POLLHUP);
} else
t->slot = -1;
@@ -448,41 +464,71 @@ static void mainloop(void) {
/* We want to play some audio */
if(device_state == device_open) {
if(backend->ready())
- play(3 * FRAMES);
+ 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. play()
- * knows to re-activate the device if necessary.
+ * We most likely timed out, so now is a good time to retry.
+ * speaker_play() knows to re-activate the device if necessary.
*/
- play(3 * FRAMES);
+ speaker_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*/);
+ 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) {
/* There might (in theory) be several commands queued up, but in general
* this won't be the case, so we don't bother looping around to pick them
* all up. */
- n = speaker_recv(0, &sm, &fd);
+ 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;
/* We attempt to play straight away rather than going round the loop.
- * play() is clever enough to perform any activation that is
+ * speaker_play() is clever enough to perform any activation that is
* required. */
- play(3 * FRAMES);
+ speaker_play(3 * FRAMES);
report();
break;
case SM_PAUSE:
@@ -496,28 +542,41 @@ static void mainloop(void) {
paused = 0;
/* As for SM_PLAY we attempt to play straight away. */
if(playing)
- play(3 * FRAMES);
+ 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:
@@ -526,8 +585,10 @@ static void mainloop(void) {
}
/* Read in any buffered data */
for(t = tracks; t; t = t->next)
- if(t->slot != -1 && (fds[t->slot].revents & (POLLIN | POLLHUP)))
- fill(t);
+ 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
@@ -541,27 +602,33 @@ static void mainloop(void) {
}
int main(int argc, char **argv) {
- int n;
+ 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:dD", options, 0)) >= 0) {
+ while((n = getopt_long(argc, argv, "hVc:dDSs", options, 0)) >= 0) {
switch(n) {
case 'h': help();
- case 'V': version();
+ 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(getenv("DISORDER_DEBUG_SPEAKER")) debugging = 1;
- /* If stderr is a TTY then log there, otherwise to syslog. */
- if(!isatty(2)) {
+ if((d = getenv("DISORDER_DEBUG_SPEAKER"))) debugging = atoi(d);
+ if(logsyslog) {
openlog(progname, LOG_PID, LOG_DAEMON);
log_default = &log_syslog;
}
- if(config_read()) fatal(0, "cannot read configuration");
+ if(config_read(1)) fatal(0, "cannot read configuration");
bpf = bytes_per_frame(&config->sample_format);
/* ignore SIGPIPE */
signal(SIGPIPE, SIG_IGN);
@@ -575,13 +642,35 @@ int main(int argc, char **argv) {
if(getuid() == 0 || geteuid() == 0) fatal(0, "do not run as root");
/* identify the backend used to play */
for(n = 0; backends[n]; ++n)
- if(backends[n]->backend == config->speaker_backend)
+ if(backends[n]->backend == config->api)
break;
if(!backends[n])
- fatal(0, "unsupported backend %d", config->speaker_backend);
+ 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);