X-Git-Url: http://www.chiark.greenend.org.uk/ucgi/~mdw/git/disorder/blobdiff_plain/71b70599a2cd81c13cc4326499a5d0c45358cd7d..4942ee7d61bf22ba38bf026c7d05028cb7db0d54:/server/speaker.c diff --git a/server/speaker.c b/server/speaker.c index d21ad7c..892e33c 100644 --- a/server/speaker.c +++ b/server/speaker.c @@ -1,22 +1,20 @@ /* * 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 . */ /** @file server/speaker.c * @brief Speaker process @@ -27,9 +25,29 @@ * 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 - * the limits that ALSA can deal with.) + * @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 @@ -50,25 +68,24 @@ * 2-byte samples. */ -#include -#include "types.h" +#include "common.h" #include -#include -#include #include #include #include #include #include -#include -#include #include #include #include #include #include #include +#include +#include +#include +#include #include "configuration.h" #include "syscalls.h" @@ -77,40 +94,124 @@ #include "mem.h" #include "speaker-protocol.h" #include "user.h" -#include "speaker.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 */ -struct track *tracks; +static struct track *tracks; -/** @brief Playing track, or NULL */ -struct track *playing; +/** @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 Number of bytes pre frame */ -size_t bpf; +/** @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() */ -struct pollfd fds[NFDS]; +static struct pollfd fds[NFDS]; /** @brief Next free slot in @ref fds */ -int fdno; +static int fdno; /** @brief Listen socket */ static int listenfd; -static time_t last_report; /* when we last reported */ -static int paused; /* pause status */ +/** @brief Timestamp of last potential report to server */ +static time_t last_report; -/** @brief The current device state */ -enum device_states device_state; +/** @brief Set when paused */ +static int paused; -/** @brief Set when idled - * - * This is set when the sound device is deliberately closed by idle(). - */ -int idled; +/** @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' }, @@ -140,19 +241,11 @@ static void help(void) { exit(0); } -/* Display version number and terminate. */ -static void version(void) { - xprintf("%s", 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; -} - -/** @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; @@ -169,7 +262,10 @@ static struct track *findtrack(const char *id, int create) { 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; @@ -181,10 +277,13 @@ static struct track *removetrack(const char *id) { 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->fd != -1) + xclose(t->fd); free(t); } @@ -198,193 +297,89 @@ static void destroy(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", t->id, t->eof, t->used)); - if(t->eof) return -1; + 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; + 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; + /* A track always becomes playable at EOF; we're not going to see any + * more data. */ t->playable = 1; - return -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; } - t->used += n; - if(t->used == sizeof t->buffer) - t->playable = 1; } - return 0; -} - -/** @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(backend->deactivate) - backend->deactivate(); - else - device_state = device_closed; - idled = 1; -} - -/** @brief Abandon the current track */ -void abandon(void) { - struct speaker_message sm; - - D(("abandon")); - memset(&sm, 0, sizeof sm); - sm.type = SM_FINISHED; - strcpy(sm.id, playing->id); - speaker_send(1, &sm); - removetrack(playing->id); - destroy(playing); - playing = 0; -} - -/** @brief Enable sound output - * - * Makes sure the sound device is open and has the right sample format. Return - * 0 on success and -1 on error. - */ -static void activate(void) { - if(backend->activate) - backend->activate(); - else - device_state = device_open; -} - -/** @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->used < bytes_per_frame(&config->sample_format)) - abandon(); + 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 + && (!paused || playing->finished) && playing->playable; } -/** @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; - - /* 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" : "", - 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 > 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; - 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 == (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. */ +/** @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 / config->sample_format.rate; + sm.data = playing->played / (uaudio_rate * uaudio_channels); speaker_send(1, &sm); + xtime(&last_report); } - 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); } -int addfd(int fd, int events) { +/** @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; @@ -393,33 +388,81 @@ int addfd(int fd, int events) { return -1; } -/** @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 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) + info("%zu samples silence, playing->used=%zu", provided_samples, playing->used); + else + 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; + 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 a second before thinking about current - * state. */ - timeout = 1000; + /* 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 */ @@ -433,18 +476,6 @@ static void mainloop(void) { playing->slot = addfd(playing->fd, POLLIN); else if(playing) playing->slot = -1; - 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. We do this last of all, so that if we run out of slots, * nothing important can't be monitored. */ @@ -457,28 +488,15 @@ static void mainloop(void) { } 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; fatal(errno, "error calling poll"); } - /* Play some sound before doing anything else */ - 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); - } - } /* Perhaps a connection has arrived */ if(fds[listen_slot].revents & POLLIN) { struct sockaddr_un addr; @@ -501,7 +519,8 @@ static void mainloop(void) { id[l] = 0; D(("id %s fd %d", id, fd)); t = findtrack(id, 1/*create*/); - write(fd, "", 1); /* write an ack */ + 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); @@ -519,55 +538,88 @@ static void mainloop(void) { * 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) + /* 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: - if(playing) fatal(0, "got SM_PLAY but already playing something"); + /* 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) + 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) + 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) error(0, "cannot play track because no connection arrived"); - playing = t; - /* 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(); + /* 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; - report(); + force_report = 1; break; case SM_RESUME: D(("SM_RESUME")); - if(paused) { - paused = 0; - /* As for SM_PLAY we attempt to play straight away. */ - if(playing) - speaker_play(3 * FRAMES); - } - 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) { - if(t == playing) { + 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; - strcpy(sm.id, playing->id); - speaker_send(1, &sm); - playing = 0; + 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. */ + 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); - report(); + } + speaker_send(1, &sm); + force_report = 1; break; case SM_RELOAD: D(("SM_RELOAD")); - if(config_read(1)) error(0, "cannot read configuration"); + if(config_read(1, NULL)) + error(0, "cannot read configuration"); info("reloaded configuration"); break; default: @@ -580,14 +632,59 @@ static void mainloop(void) { && 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) && device_state == device_open) - idle(); - /* If we've not reported out state for a second do so now. */ - if(time(0) > last_report) + /* 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(); } } @@ -598,13 +695,15 @@ int main(int argc, char **argv) { 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(); + case 'V': version("disorder-speaker"); case 'c': configfile = optarg; break; case 'd': debugging = 1; break; case 'D': debugging = 0; break; @@ -618,32 +717,59 @@ int main(int argc, char **argv) { 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); + config_uaudio_apis = uaudio_apis; + if(config_read(1, NULL)) 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]; ++n) - if(backends[n]->backend == config->speaker_backend) - break; - if(!backends[n]) - fatal(0, "unsupported backend %d", config->speaker_backend); - backend = backends[n]; + 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); + /* 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 */ - backend->init(); + 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) + 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", + 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);