X-Git-Url: https://git.distorted.org.uk/~mdw/disorder/blobdiff_plain/6ba5f1eaee022b36d76f245c7cbcf78b72e9cf35..3a23a6a5d9a9973ebb9d62644d2d68a79483d31d:/server/speaker.c diff --git a/server/speaker.c b/server/speaker.c index 994c4b4..aebd930 100644 --- a/server/speaker.c +++ b/server/speaker.c @@ -18,7 +18,7 @@ * 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 @@ -82,86 +82,56 @@ #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" #if API_ALSA #include #endif -#ifdef WORDS_BIGENDIAN -# define MACHINE_AO_FMT AO_FMT_BIG -#else -# define MACHINE_AO_FMT AO_FMT_LITTLE -#endif - -/** @brief How many seconds of input to buffer - * - * While any given connection has this much audio buffered, no more reads will - * be issued for that connection. The decoder will have to wait. - */ -#define BUFFER_SECONDS 5 - -#define FRAMES 4096 /* Frame batch size */ - -/** @brief Bytes to send per network packet - * - * Don't make this too big or arithmetic will start to overflow. - */ -#define NETWORK_BYTES (1024+sizeof(struct rtp_header)) +/** @brief Linked list of all prepared tracks */ +struct track *tracks; -/** @brief Maximum RTP playahead (ms) */ -#define RTP_AHEAD_MS 1000 - -/** @brief Maximum number of FDs to poll for */ -#define NFDS 256 - -/** @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 Playing track, or NULL */ +struct track *playing; static time_t last_report; /* when we last reported */ static int paused; /* pause status */ static size_t bpf; /* bytes per frame */ static struct pollfd fds[NFDS]; /* if we need more than that */ static int fdno; /* fd number */ -static size_t bufsize; /* buffer size */ #if API_ALSA /** @brief The current PCM handle */ static snd_pcm_t *pcm; static snd_pcm_uframes_t last_pcm_bufsize; /* last seen buffer size */ -static ao_sample_format pcm_format; /* current format if aodev != 0 */ #endif -/** @brief Ready to send audio +/** @brief The current device state */ +enum device_states device_state; + +/** @brief The current device sample format * - * This is set when the destination is ready to receive audio. Generally - * this implies that the sound device is open. In the ALSA backend it - * does @b not necessarily imply that is has the right sample format. + * Only meaningful if @ref device_state = @ref device_open or perhaps @ref + * device_error. For @ref FIXED_FORMAT backends, this should always match @c + * config->sample_format. */ -static int ready; +ao_sample_format device_format; -static int forceplay; /* frames to force play */ -static int cmdfd = -1; /* child process input */ -static int bfd = -1; /* broadcast FD */ +/** @brief Pipe to subprocess + * + * This is the file descriptor to write to for @ref BACKEND_COMMAND. + */ +static int cmdfd = -1; + +/** @brief Network socket + * + * This is the file descriptor to write to for @ref BACKEND_NETWORK. + */ +static int bfd = -1; /** @brief RTP timestamp * @@ -183,70 +153,20 @@ static uint64_t rtp_time; */ static struct timeval rtp_time_0; -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 RTP packet sequence number */ +static uint16_t rtp_seq; -/** @brief Structure of a backend */ -struct speaker_backend { - /** @brief Which backend this is - * - * @c -1 terminates the list. - */ - int backend; +/** @brief RTP SSRC */ +static uint32_t rtp_id; - /** @brief Flags - * - * Possible values - * - @ref FIXED_FORMAT - */ - unsigned flags; -/** @brief Lock to configured sample format */ -#define FIXED_FORMAT 0x0001 - - /** @brief Initialization - * - * Called once at startup. This is responsible for one-time setup - * operations, for instance opening a network socket to transmit to. - * - * When writing to a native sound API this might @b not imply opening the - * native sound device - that might be done by @c activate below. - */ - void (*init)(void); - - /** @brief Activation - * @return 0 on success, non-0 on error - * - * Called to activate the output device. - * - * After this function succeeds, @ref ready should be non-0. As well as - * opening the audio device, this function is responsible for reconfiguring - * if it necessary to cope with different samples formats (for backends that - * don't demand a single fixed sample format for the lifetime of the server). - */ - int (*activate)(void); - - /** @brief Play sound - * @param frames Number of frames to play - * @return Number of frames actually played - */ - size_t (*play)(size_t frames); - - /** @brief Deactivation - * - * Called to deactivate the sound device. This is the inverse of - * @c activate above. - */ - void (*deactivate)(void); +/** @brief Set when idled + * + * This is set when the sound device is deliberately closed by idle(). + */ +static int idled; /* set when idled */ - /** @brief Called before poll() - * - * Called before the call to poll(). Should call addfd() to update the FD - * array and stash the slot number somewhere safe. - */ - void (*beforepoll)(void); -}; +/** @brief Error counter */ +static int audio_errors; /** @brief Selected backend */ static const struct speaker_backend *backend; @@ -431,7 +351,9 @@ static void enable_translation(struct track *t) { * @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) { size_t where, left; @@ -481,13 +403,19 @@ static int fill(struct track *t) { 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(backend->deactivate) + if(backend->deactivate) backend->deactivate(); + else + device_state = device_closed; idled = 1; - ready = 0; } /** @brief Abandon the current track */ @@ -502,52 +430,39 @@ static void abandon(void) { 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) { +static void activate(void) { /* If we don't know the format yet we cannot start. */ if(!playing->got_format) { D((" - not got format for %s", playing->id)); - return -1; + return; + } + if(backend->flags & FIXED_FORMAT) + device_format = config->sample_format; + if(backend->activate) { + backend->activate(); + } else { + assert(backend->flags & FIXED_FORMAT); + /* ...otherwise device_format not set */ + device_state = device_open; } - return backend->activate(); + if(device_state == device_open) + bpf = bytes_per_frame(&device_format); } -/* 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 @@ -556,50 +471,38 @@ static void maybe_finished(void) { 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 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 play() is not called when + * unreasonably only an small amounts of data is available to play. + */ static void play(size_t frames) { size_t avail_frames, avail_bytes, written_frames; ssize_t written_bytes; - /* Make sure the output device is activated */ - if(activate()) { - if(playing) - forceplay = frames; - else - forceplay = 0; /* Must have called abandon() */ + /* Make sure there's a track to play and it is not pasued */ + if(!playing || paused) return; + /* Make sure the output device is open and has the right sample format */ + if(device_state != device_open + || !formats_equal(&device_format, &playing->format)) { + 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; /* Figure out how many frames there are available to write */ if(playing->start + playing->used > playing->size) /* The ring buffer is currently wrapped, only play up to the wrap point */ @@ -626,6 +529,7 @@ static void play(size_t frames) { if(!playing->used || playing->start == playing->size) playing->start = 0; frames -= written_frames; + return; } /* Notify the server what we're up to. */ @@ -667,11 +571,55 @@ static void alsa_init(void) { info("selected ALSA backend"); } +/** @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); + } +} + +/** @brief ALSA deactivation */ +static void alsa_deactivate(void) { + if(pcm) { + int err; + + if((err = snd_pcm_nonblock(pcm, 0)) < 0) + fatal(0, "error calling snd_pcm_nonblock: %d", err); + D(("draining pcm")); + snd_pcm_drain(pcm); + D(("closing pcm")); + snd_pcm_close(pcm); + pcm = 0; + device_state = device_closed; + D(("released audio device")); + } +} + /** @brief ALSA backend activation */ -static int alsa_activate(void) { +static void alsa_activate(void) { /* If we need to change format then close the current device. */ - if(pcm && !formats_equal(&playing->format, &pcm_format)) - idle(); + if(pcm && !formats_equal(&playing->format, &device_format)) + alsa_deactivate(); + /* Now if the sound device is open it must have the right format */ if(!pcm) { snd_pcm_hw_params_t *hwparams; snd_pcm_sw_params_t *swparams; @@ -732,8 +680,7 @@ static int alsa_activate(void) { playing->format.channels, err); goto fatal; } - bufsize = 3 * FRAMES; - pcm_bufsize = bufsize; + pcm_bufsize = 3 * FRAMES; 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", @@ -753,13 +700,12 @@ static int alsa_activate(void) { 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); + device_format = playing->format; D(("acquired audio device")); log_params(hwparams, swparams); - ready = 1; + device_state = device_open; } - return 0; + return; fatal: abandon(); error: @@ -767,8 +713,9 @@ error: if(pcm) { snd_pcm_close(pcm); pcm = 0; + device_state = device_error; } - return -1; + return; } /** @brief Play via ALSA */ @@ -824,24 +771,44 @@ static void alsa_beforepoll(void) { fdno += alsa_nslots; } -/** @brief ALSA deactivation */ -static void alsa_deactivate(void) { - if(pcm) { - int err; - - if((err = snd_pcm_nonblock(pcm, 0)) < 0) - fatal(0, "error calling snd_pcm_nonblock: %d", err); - D(("draining pcm")); - snd_pcm_drain(pcm); - D(("closing pcm")); - snd_pcm_close(pcm); - pcm = 0; - forceplay = 0; - D(("released audio device")); - } +/** @brief Process poll() results for ALSA */ +static int alsa_ready(void) { + int err; + + 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)) + return 1; + else + return 0; } #endif +/** @brief Start the subprocess for @ref BACKEND_COMMAND */ +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 Command backend initialization */ static void command_init(void) { info("selected command backend"); @@ -881,15 +848,12 @@ static void command_beforepoll(void) { cmdfd_slot = addfd(cmdfd, POLLOUT); } -/** @brief Command/network backend activation */ -static int generic_activate(void) { - if(!ready) { - bufsize = 3 * FRAMES; - bpf = bytes_per_frame(&config->sample_format); - D(("acquired audio device")); - ready = 1; - } - return 0; +/** @brief Process poll() results for subprocess play */ +static int command_ready(void) { + if(fds[cmdfd_slot].revents & (POLLOUT | POLLERR)) + return 1; + else + return 0; } /** @brief Network backend initialization */ @@ -1010,6 +974,7 @@ static size_t network_play(size_t frames) { * transmitting packets with overlapping timestamp ranges, which does not * make sense. */ + target_rtp_time &= ~(uint64_t)1; /* stereo! */ if(target_rtp_time > rtp_time) { /* More time has elapsed than we've transmitted samples. That implies * we've been 'sending' silence. */ @@ -1100,6 +1065,14 @@ static void network_beforepoll(void) { bfd_slot = addfd(bfd, POLLOUT); } +/** @brief Process poll() results for network play */ +static int network_ready(void) { + if(fds[bfd_slot].revents & (POLLOUT | POLLERR)) + return 1; + else + return 0; +} + /** @brief Table of speaker backends */ static const struct speaker_backend backends[] = { #if API_ALSA @@ -1110,100 +1083,79 @@ static const struct speaker_backend backends[] = { alsa_activate, alsa_play, alsa_deactivate, - alsa_beforepoll + alsa_beforepoll, + alsa_ready }, #endif { BACKEND_COMMAND, FIXED_FORMAT, command_init, - generic_activate, + 0, /* activate */ command_play, 0, /* deactivate */ - command_beforepoll + command_beforepoll, + command_ready }, { BACKEND_NETWORK, FIXED_FORMAT, network_init, - generic_activate, + 0, /* activate */ network_play, 0, /* deactivate */ - network_beforepoll + network_beforepoll, + network_ready }, - { -1, 0, 0, 0, 0, 0, 0 } + { -1, 0, 0, 0, 0, 0, 0, 0 } /* end of list */ }; -int main(int argc, char **argv) { - int n, fd, stdin_slot, poke, timeout; +/** @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; -#if API_ALSA - int err; -#endif + int n, fd, stdin_slot, timeout; - set_progname(argv); - if(!setlocale(LC_CTYPE, "")) fatal(errno, "error calling setlocale"); - while((n = getopt_long(argc, argv, "hVc:dD", options, 0)) >= 0) { - switch(n) { - case 'h': help(); - case 'V': version(); - case 'c': configfile = optarg; break; - case 'd': debugging = 1; break; - case 'D': debugging = 0; break; - default: fatal(0, "invalid option"); - } - } - if(getenv("DISORDER_DEBUG_SPEAKER")) debugging = 1; - /* If stderr is a TTY then log there, otherwise to syslog. */ - if(!isatty(2)) { - openlog(progname, LOG_PID, LOG_DAEMON); - log_default = &log_syslog; - } - if(config_read()) fatal(0, "cannot read configuration"); - /* ignore SIGPIPE */ - signal(SIGPIPE, SIG_IGN); - /* reap kids */ - signal(SIGCHLD, reap); - /* set nice value */ - xnice(config->nice_speaker); - /* change user */ - become_mortal(); - /* make sure we're not root, whatever the config says */ - if(getuid() == 0 || geteuid() == 0) fatal(0, "do not run as root"); - /* identify the backend used to play */ - for(n = 0; backends[n].backend != -1; ++n) - if(backends[n].backend == config->speaker_backend) - break; - if(backends[n].backend == -1) - fatal(0, "unsupported backend %d", config->speaker_backend); - backend = &backends[n]; - /* backend-specific initialization */ - backend->init(); 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); /* 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->eof && playing->used < playing->size) playing->slot = addfd(playing->fd, POLLIN); - } else if(playing) + else if(playing) playing->slot = -1; - /* If forceplay is set then wait until it succeeds before waiting on the - * sound device. */ - alsa_slots = -1; - cmdfd_slot = -1; - bfd_slot = -1; - /* By default we will wait up to a second before thinking about current - * state. */ - timeout = 1000; - /* We'll break the poll as soon as the underlying sound device is ready for - * more data */ - if(ready && !forceplay) - backend->beforepoll(); + if(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(); + } /* 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) { @@ -1218,48 +1170,26 @@ int main(int argc, char **argv) { 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)) + if(playable()) { + /* We want to play some audio */ + if(device_state == device_open) { + if(backend->ready()) 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(poke) { - /* Some attempt to play must have failed */ - if(playing && !paused) - play(forceplay); - else - forceplay = 0; /* just in case */ + } 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. + */ + play(3 * FRAMES); + } } /* 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); if(n > 0) switch(sm.type) { @@ -1275,7 +1205,10 @@ int main(int argc, char **argv) { t = findtrack(sm.id, 1); if(fd != -1) acquire(t, fd); playing = t; - play(bufsize); + /* We attempt to play straight away rather than going round the loop. + * play() is clever enough to perform any activation that is + * required. */ + play(3 * FRAMES); report(); break; case SM_PAUSE: @@ -1287,8 +1220,9 @@ int main(int argc, char **argv) { D(("SM_RESUME")); if(paused) { paused = 0; + /* As for SM_PLAY we attempt to play straight away. */ if(playing) - play(bufsize); + play(3 * FRAMES); } report(); break; @@ -1320,19 +1254,60 @@ int main(int argc, char **argv) { for(t = tracks; t; t = t->next) if(t->slot != -1 && (fds[t->slot].revents & (POLLIN | POLLHUP))) fill(t); - /* We might be able to play now */ - if(ready && forceplay && playing && !paused) - play(forceplay); /* Maybe we finished playing a track somewhere in the above */ maybe_finished(); /* If we don't need the sound device for now then close it for the benefit * of anyone else who wants it. */ - if((!playing || paused) && ready) + 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; + + set_progname(argv); + if(!setlocale(LC_CTYPE, "")) fatal(errno, "error calling setlocale"); + while((n = getopt_long(argc, argv, "hVc:dD", options, 0)) >= 0) { + switch(n) { + case 'h': help(); + case 'V': version(); + case 'c': configfile = optarg; break; + case 'd': debugging = 1; break; + case 'D': debugging = 0; break; + default: fatal(0, "invalid option"); + } + } + if(getenv("DISORDER_DEBUG_SPEAKER")) debugging = 1; + /* If stderr is a TTY then log there, otherwise to syslog. */ + if(!isatty(2)) { + openlog(progname, LOG_PID, LOG_DAEMON); + log_default = &log_syslog; + } + if(config_read()) fatal(0, "cannot read configuration"); + /* ignore SIGPIPE */ + signal(SIGPIPE, SIG_IGN); + /* reap kids */ + signal(SIGCHLD, reap); + /* set nice value */ + xnice(config->nice_speaker); + /* change user */ + become_mortal(); + /* make sure we're not root, whatever the config says */ + if(getuid() == 0 || geteuid() == 0) fatal(0, "do not run as root"); + /* identify the backend used to play */ + for(n = 0; backends[n].backend != -1; ++n) + if(backends[n].backend == config->speaker_backend) + break; + if(backends[n].backend == -1) + fatal(0, "unsupported backend %d", config->speaker_backend); + backend = &backends[n]; + /* backend-specific initialization */ + backend->init(); + mainloop(); info("stopped (parent terminated)"); exit(0); }