/* * This file is part of DisOrder * Copyright (C) 2005-2009 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 * 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. * * You should have received a copy of the GNU General Public License * along with this program. If not, see . */ /** @file server/speaker.c * @brief Speaker process * * This program is responsible for transmitting a single coherent audio stream * to its destination (over the network, to some sound API, to some * subprocess). It receives connections from decoders (or rather from the * process that is about to become disorder-normalize) and plays them in the * right order. * * @b 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. * * 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. * * To implement gapless playback, the server is notified that a track has * finished slightly early. @ref SM_PLAY is therefore allowed to arrive while * the previous track is still playing provided an early @ref SM_FINISHED has * been sent for it. * * @b Encodings. The encodings supported depend entirely on the uaudio backend * chosen. See @ref uaudio.h, etc. * * Inbound data is expected to match @c config->sample_format. In normal use * this is arranged by the @c disorder-normalize program (see @ref * server/normalize.c). * * @b Garbage @b Collection. This program deliberately does not use the * garbage collector even though it might be convenient to do so. This is for * two reasons. Firstly some sound APIs use thread threads and we do not want * to have to deal with potential interactions between threading and garbage * collection. Secondly this process needs to be able to respond quickly and * this is not compatible with the collector hanging the program even * relatively briefly. * * @b Units. This program thinks at various times in three different units. * Bytes are obvious. A sample is a single sample on a single channel. A * frame is several samples on different channels at the same point in time. * So (for instance) a 16-bit stereo frame is 4 bytes and consists of a pair of * 2-byte samples. */ #include "common.h" #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include "configuration.h" #include "syscalls.h" #include "log.h" #include "defs.h" #include "mem.h" #include "speaker-protocol.h" #include "user.h" #include "printf.h" #include "version.h" #include "uaudio.h" /** @brief Maximum number of FDs to poll for */ #define NFDS 1024 /** @brief Number of bytes before end of track to send SM_FINISHED * * Generally set to 1 second. */ static size_t early_finish; /** @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. */ struct track { /** @brief Next track */ struct track *next; /** @brief Input file descriptor */ int fd; /* input FD */ /** @brief Track ID */ char id[24]; /** @brief Start position of data in buffer */ size_t start; /** @brief Number of bytes of data in buffer */ size_t used; /** @brief Set @c fd is at EOF */ int eof; /** @brief Total number of samples played */ unsigned long long played; /** @brief Slot in @ref fds */ int slot; /** @brief Set when playable * * 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 playable; /** @brief Set when finished * * This is set when we've notified the server that the track is finished. * Once this has happened (typically very late in the track's lifetime) the * track cannot be paused or cancelled. */ int finished; /** @brief Input buffer * * 1Mbyte is enough for nearly 6s of 44100Hz 16-bit stereo */ char buffer[1048576]; }; /** @brief Lock protecting data structures * * This lock protects values shared between the main thread and the callback. * * It is held 'all' the time by the main thread, the exceptions being when * called activate/deactivate callbacks and when calling (potentially) slow * system calls (in particular poll(), where in fact the main thread will spend * most of its time blocked). * * The callback holds it when it's running. */ static pthread_mutex_t lock = PTHREAD_MUTEX_INITIALIZER; /** @brief Linked list of all prepared tracks */ static struct track *tracks; /** @brief Playing track, or NULL * * This means the track the speaker process intends to play. It does not * reflect any other state (e.g. activation of uaudio backend). */ static struct track *playing; /** @brief Pending playing track, or NULL * * This means the track the server wants the speaker to play. */ static struct track *pending_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 uaudio *backend; static const struct option options[] = { { "help", no_argument, 0, 'h' }, { "version", no_argument, 0, 'V' }, { "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 } }; /* Display usage message and terminate. */ static void help(void) { xprintf("Usage:\n" " disorder-speaker [OPTIONS]\n" "Options:\n" " --help, -h Display usage message\n" " --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"); xfclose(stdout); exit(0); } /** @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; D(("findtrack %s %d", id, create)); for(t = tracks; t && strcmp(id, t->id); t = t->next) ; if(!t && create) { t = xmalloc(sizeof *t); t->next = tracks; strcpy(t->id, id); t->fd = -1; tracks = t; } return t; } /** @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; D(("removetrack %s", id)); for(tt = &tracks; (t = *tt) && strcmp(id, t->id); tt = &t->next) ; if(t) *tt = t->next; return t; } /** @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); 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. 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 speaker_fill(struct track *t) { size_t where, left; int n, rc; D(("fill %s: eof=%d used=%zu", t->id, t->eof, t->used)); if(t->eof) return -1; if(t->used < sizeof t->buffer) { /* there is room left in the buffer */ 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) disorder_fatal(errno, "error reading sample stream"); rc = 0; } else if(n == 0) { D(("fill %s: eof detected", t->id)); t->eof = 1; /* 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 rc; } /** @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. * * We don't allow tracks to be paused if we've already told the server we've * finished them; that would cause such tracks to survive much longer than the * few samples they're supposed to, with report() remaining silent for the * duration. */ static int playable(void) { return playing && (!paused || playing->finished) && playing->playable; } /** @brief Notify the server what we're up to */ static void report(void) { struct speaker_message sm; if(playing) { /* Had better not send a report for a track that the server thinks has * finished, that would be confusing. */ if(playing->finished) return; memset(&sm, 0, sizeof sm); sm.type = paused ? SM_PAUSED : SM_PLAYING; strcpy(sm.id, playing->id); sm.data = playing->played / (uaudio_rate * uaudio_channels); speaker_send(1, &sm); xtime(&last_report); } } /** @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; fds[fdno].events = events; return fdno++; } else return -1; } /** @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) { int ignored = write(sigpipe[1], "", 1); (void) ignored; } provided_samples = bytes / uaudio_sample_size; playing->played += provided_samples; } } /* 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; if(playing) disorder_info("%zu samples silence, playing->used=%zu", provided_samples, playing->used); else disorder_info("%zu samples silence, playing=NULL", provided_samples); } pthread_mutex_unlock(&lock); return provided_samples; } /** @brief Main event loop */ static void mainloop(void) { struct track *t; struct speaker_message sm; int n, fd, stdin_slot, timeout, listen_slot, sigpipe_slot; /* Keep going while our parent process is alive */ pthread_mutex_lock(&lock); while(getppid() != 1) { int force_report = 0; fdno = 0; /* By default we will wait up to half a second before thinking about * current state. */ timeout = 500; /* 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->fd >= 0 && !playing->eof && playing->used < (sizeof playing->buffer)) playing->slot = addfd(playing->fd, POLLIN); else if(playing) playing->slot = -1; /* 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->fd >= 0 && !t->eof && t->used < sizeof t->buffer) { t->slot = addfd(t->fd, POLLIN | POLLHUP); } else t->slot = -1; } sigpipe_slot = addfd(sigpipe[0], POLLIN); /* Wait for something interesting to happen */ pthread_mutex_unlock(&lock); n = poll(fds, fdno, timeout); pthread_mutex_lock(&lock); if(n < 0) { if(errno == EINTR) continue; disorder_fatal(errno, "error calling poll"); } /* 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) { disorder_error(errno, "reading length from inbound connection"); xclose(fd); } else if(l >= sizeof id) { disorder_error(0, "id length too long"); xclose(fd); } else if(read(fd, id, l) < (ssize_t)l) { disorder_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 */ disorder_error(errno, "writing ack to inbound connection"); if(t->fd != -1) { disorder_error(0, "%s: already got a connection", id); xclose(fd); } else { nonblock(fd); t->fd = fd; /* yay */ } /* Notify the server that the connection arrived */ sm.type = SM_ARRIVED; strcpy(sm.id, id); speaker_send(1, &sm); } } else disorder_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); if(n > 0) /* As a rule we don't send success replies to most commands - we just * force the regular status update to be sent immediately rather than * on schedule. */ switch(sm.type) { case SM_PLAY: /* SM_PLAY is only allowed if the server reasonably believes that * nothing is playing */ if(playing) { /* If finished isn't set then the server can't believe that this * track has finished */ if(!playing->finished) disorder_fatal(0, "got SM_PLAY but already playing something"); /* If pending_playing is set then the server must believe that that * is playing */ if(pending_playing) disorder_fatal(0, "got SM_PLAY but have a pending playing track"); } t = findtrack(sm.id, 1); D(("SM_PLAY %s fd %d", t->id, t->fd)); if(t->fd == -1) disorder_error(0, "cannot play track because no connection arrived"); /* TODO as things stand we often report this error message but then * appear to proceed successfully. Understanding why requires a look * at play.c: we call prepare() which makes the connection in a child * process, and then sends the SM_PLAY in the parent process. The * latter may well be faster. As it happens this is harmless; we'll * just sit around sending silence until the decoder connects and * starts sending some sample data. But is is annoying and ought to * be fixed. */ pending_playing = t; /* If nothing is currently playing then we'll switch to the pending * track below so there's no point distinguishing the situations * here. */ break; case SM_PAUSE: D(("SM_PAUSE")); paused = 1; force_report = 1; break; case SM_RESUME: D(("SM_RESUME")); paused = 0; force_report = 1; break; case SM_CANCEL: D(("SM_CANCEL %s", sm.id)); t = removetrack(sm.id); if(t) { if(t == playing || t == pending_playing) { /* Scratching the track that the server believes is playing, * which might either be the actual playing track or a pending * playing track */ sm.type = SM_FINISHED; if(t == playing) playing = 0; else pending_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. */ disorder_info("SM_CANCEL for nonplaying track %s", sm.id); sm.type = SM_STILLBORN; } strcpy(sm.id, t->id); destroy(t); } 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; disorder_error(0, "SM_CANCEL for unknown track %s", sm.id); } speaker_send(1, &sm); force_report = 1; break; case SM_RELOAD: D(("SM_RELOAD")); if(config_read(1, NULL)) disorder_error(0, "cannot read configuration"); disorder_info("reloaded configuration"); break; default: disorder_error(0, "unknown message type %d", sm.type); } } /* Read in any buffered data */ for(t = tracks; t; t = t->next) 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]; int ignored; (void)ignored; ignored = read(sigpipe[0], buffer, sizeof buffer); } /* Send SM_FINISHED when we're near the end of the track. * * This is how we implement gapless play; we hope that the SM_PLAY from the * server arrives before the remaining bytes of the track play out. */ if(playing && playing->eof && !playing->finished && playing->used <= early_finish) { memset(&sm, 0, sizeof sm); sm.type = SM_FINISHED; strcpy(sm.id, playing->id); speaker_send(1, &sm); playing->finished = 1; } /* When the track is actually finished, deconfigure it */ if(playing && playing->eof && !playing->used) { removetrack(playing->id); destroy(playing); playing = 0; } /* Act on the pending SM_PLAY */ if(!playing && pending_playing) { playing = pending_playing; pending_playing = 0; force_report = 1; } /* Impose any state change required by the above */ if(playable()) { if(!activated) { activated = 1; pthread_mutex_unlock(&lock); backend->activate(); pthread_mutex_lock(&lock); } } else { if(activated) { activated = 0; pthread_mutex_unlock(&lock); backend->deactivate(); pthread_mutex_lock(&lock); } } /* If we've not reported our state for a second do so now. */ if(force_report || xtime(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, "")) disorder_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: disorder_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, NULL)) disorder_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) disorder_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) disorder_fatal(errno, "getrlimit RLIMIT_NOFILE"); if(rl->rlim_cur > NFDS) { rl->rlim_cur = NFDS; if(setrlimit(RLIMIT_NOFILE, rl) < 0) disorder_fatal(errno, "setrlimit to reduce RLIMIT_NOFILE to %lu", (unsigned long)rl->rlim_cur); disorder_info("set RLIM_NOFILE to %lu", (unsigned long)rl->rlim_cur); } else disorder_info("RLIM_NOFILE is %lu", (unsigned long)rl->rlim_cur); /* gcrypt initialization */ if(!gcry_check_version(NULL)) disorder_fatal(0, "gcry_check_version failed"); gcry_control(GCRYCTL_INIT_SECMEM, 0); gcry_control (GCRYCTL_INITIALIZATION_FINISHED, 0); /* 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); early_finish = uaudio_sample_size * uaudio_channels * uaudio_rate; /* TODO other parameters! */ backend = uaudio_find(config->api); /* backend-specific initialization */ if(backend->configure) backend->configure(); 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) disorder_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) disorder_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) disorder_fatal(errno, "error binding socket to %s", addr.sun_path); xlisten(listenfd, 128); nonblock(listenfd); disorder_info("listening on %s", addr.sun_path); memset(&sm, 0, sizeof sm); sm.type = SM_READY; speaker_send(1, &sm); mainloop(); disorder_info("stopped (parent terminated)"); exit(0); } /* Local Variables: c-basic-offset:2 comment-column:40 fill-column:79 indent-tabs-mode:nil End: */