X-Git-Url: http://www.chiark.greenend.org.uk/ucgi/~mdw/git/disorder/blobdiff_plain/0763e1f402fe7bb6c2b7618efb4838810c7ff0e6..05b75f8d50b83e943af3be4071449304d82dbdcd:/server/speaker.c diff --git a/server/speaker.c b/server/speaker.c index bbaedb9..3021652 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 + * Copyright (C) 2005-2008 Richard Kettlewell + * Portions (C) 2007 Mark Wooding * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by @@ -18,208 +19,93 @@ * USA */ /** @file server/speaker.c - * @brief Speaker processs + * @brief Speaker process * * This program is responsible for transmitting a single coherent audio stream * to its destination (over the network, to some sound API, to some - * subprocess). It receives connections from decoders via file descriptor - * passing from the main server and plays them in the right order. + * subprocess). It receives connections from decoders (or rather from the + * process that is about to become disorder-normalize) and plays them in the + * right order. * - * For the ALSA API, 8- and 16- bit - * stereo and mono are supported, with any sample rate (within the limits that - * ALSA can deal with.) + * @b Encodings. For the ALSA API, + * 8- and 16- bit stereo and mono are supported, with any sample rate (within + * the limits that ALSA can deal with.) * - * When communicating with a subprocess, sox is invoked to convert the inbound - * data to a single consistent format. The same applies for network (RTP) - * play, though in that case currently only 44.1KHz 16-bit stereo is supported. + * Inbound data is expected to match @c config->sample_format. In normal use + * this is arranged by the @c disorder-normalize program (see @ref + * server/normalize.c). * - * The inbound data starts with a structure defining the data format. Note - * that this is NOT portable between different platforms or even necessarily - * between versions; the speaker is assumed to be built from the same source - * and run on the same host as the main server. +7 * @b Garbage @b Collection. This program deliberately does not use the + * garbage collector even though it might be convenient to do so. This is for + * two reasons. Firstly some sound APIs use thread threads and we do not want + * to have to deal with potential interactions between threading and garbage + * collection. Secondly this process needs to be able to respond quickly and + * this is not compatible with the collector hanging the program even + * relatively briefly. * - * This program deliberately does not use the garbage collector even though it - * might be convenient to do so. This is for two reasons. Firstly some sound - * APIs use thread threads and we do not want to have to deal with potential - * interactions between threading and garbage collection. Secondly this - * process needs to be able to respond quickly and this is not compatible with - * the collector hanging the program even relatively briefly. + * @b Units. This program thinks at various times in three different units. + * Bytes are obvious. A sample is a single sample on a single channel. A + * frame is several samples on different channels at the same point in time. + * So (for instance) a 16-bit stereo frame is 4 bytes and consists of a pair of + * 2-byte samples. */ -#include -#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 #include "configuration.h" #include "syscalls.h" #include "log.h" #include "defs.h" #include "mem.h" -#include "speaker.h" +#include "speaker-protocol.h" #include "user.h" -#include "addr.h" -#include "timeval.h" -#include "rtp.h" +#include "speaker.h" +#include "printf.h" +#include "version.h" -#if API_ALSA -#include -#endif +/** @brief Linked list of all prepared tracks */ +struct track *tracks; -#ifdef WORDS_BIGENDIAN -# define MACHINE_AO_FMT AO_FMT_BIG -#else -# define MACHINE_AO_FMT AO_FMT_LITTLE -#endif +/** @brief Playing track, or NULL */ +struct track *playing; -/** @brief How many seconds of input to buffer - * - * While any given connection has this much audio buffered, no more reads will - * be issued for that connection. The decoder will have to wait. - */ -#define BUFFER_SECONDS 5 +/** @brief Number of bytes pre frame */ +size_t bpf; -#define FRAMES 4096 /* Frame batch size */ +/** @brief Array of file descriptors for poll() */ +struct pollfd fds[NFDS]; -/** @brief Bytes to send per network packet - * - * Don't make this too big or arithmetic will start to overflow. - */ -#define NETWORK_BYTES (1024+sizeof(struct rtp_header)) +/** @brief Next free slot in @ref fds */ +int fdno; -/** @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 Listen socket */ +static int listenfd; 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 - * - * 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. - */ -static int ready; +/** @brief The current device state */ +enum device_states device_state; -static int forceplay; /* frames to force play */ -static int cmdfd = -1; /* child process input */ -static int bfd = -1; /* broadcast FD */ - -/** @brief RTP timestamp - * - * This counts the number of samples played (NB not the number of frames - * played). +/** @brief Set when idled * - * The timestamp in the packet header is only 32 bits wide. With 44100Hz - * stereo, that only gives about half a day before wrapping, which is not - * particularly convenient for certain debugging purposes. Therefore the - * timestamp is maintained as a 64-bit integer, giving around six million years - * before wrapping, and truncated to 32 bits when transmitting. + * This is set when the sound device is deliberately closed by idle(). */ -static uint64_t rtp_time; - -/** @brief RTP base timestamp - * - * This is the real time correspoding to an @ref rtp_time of 0. It is used - * to recalculate the timestamp after idle periods. - */ -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 Structure of a backend */ -struct speaker_backend { - /** @brief Which backend this is - * - * @c -1 terminates the list. - */ - int backend; - - /** @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); -}; +int idled; /** @brief Selected backend */ static const struct speaker_backend *backend; @@ -230,6 +116,8 @@ 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 } }; @@ -242,6 +130,7 @@ 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"); @@ -249,15 +138,8 @@ 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 ao_sample_format *format) { +static size_t bytes_per_frame(const struct stream_header *format) { return format->channels * format->bits / 8; } @@ -274,9 +156,6 @@ static struct track *findtrack(const char *id, int create) { strcpy(t->id, id); t->fd = -1; tracks = t; - /* The initial input buffer will be the sample format. */ - t->buffer = (void *)&t->format; - t->size = sizeof t->format; } return t; } @@ -297,132 +176,30 @@ static struct track *removetrack(const char *id) { static void destroy(struct track *t) { D(("destroy %s", t->id)); if(t->fd != -1) xclose(t->fd); - if(t->buffer != (void *)&t->format) free(t->buffer); free(t); } -/** @brief Notice a new connection */ -static void acquire(struct track *t, int fd) { - D(("acquire %s %d", t->id, fd)); - if(t->fd != -1) - xclose(t->fd); - t->fd = fd; - nonblock(fd); -} - -/** @brief Return true if A and B denote identical libao formats, else false */ -static int formats_equal(const ao_sample_format *a, - const ao_sample_format *b) { - return (a->bits == b->bits - && a->rate == b->rate - && a->channels == b->channels - && a->byte_format == b->byte_format); -} - -/** @brief Compute arguments to sox */ -static void soxargs(const char ***pp, char **qq, ao_sample_format *ao) { - int n; - - *(*pp)++ = "-t.raw"; - *(*pp)++ = "-s"; - *(*pp)++ = *qq; n = sprintf(*qq, "-r%d", ao->rate); *qq += n + 1; - *(*pp)++ = *qq; n = sprintf(*qq, "-c%d", ao->channels); *qq += n + 1; - /* sox 12.17.9 insists on -b etc; CVS sox insists on - etc; both are - * deployed! */ - switch(config->sox_generation) { - case 0: - if(ao->bits != 8 - && ao->byte_format != AO_FMT_NATIVE - && ao->byte_format != MACHINE_AO_FMT) { - *(*pp)++ = "-x"; - } - switch(ao->bits) { - case 8: *(*pp)++ = "-b"; break; - case 16: *(*pp)++ = "-w"; break; - case 32: *(*pp)++ = "-l"; break; - case 64: *(*pp)++ = "-d"; break; - default: fatal(0, "cannot handle sample size %d", (int)ao->bits); - } - break; - case 1: - switch(ao->byte_format) { - case AO_FMT_NATIVE: break; - case AO_FMT_BIG: *(*pp)++ = "-B"; break; - case AO_FMT_LITTLE: *(*pp)++ = "-L"; break; - } - *(*pp)++ = *qq; n = sprintf(*qq, "-%d", ao->bits/8); *qq += n + 1; - break; - } -} - -/** @brief Enable format translation - * - * If necessary, replaces a tracks inbound file descriptor with one connected - * to a sox invocation, which performs the required translation. - */ -static void enable_translation(struct track *t) { - if((backend->flags & FIXED_FORMAT) - && !formats_equal(&t->format, &config->sample_format)) { - char argbuf[1024], *q = argbuf; - const char *av[18], **pp = av; - int soxpipe[2]; - pid_t soxkid; - - *pp++ = "sox"; - soxargs(&pp, &q, &t->format); - *pp++ = "-"; - soxargs(&pp, &q, &config->sample_format); - *pp++ = "-"; - *pp++ = 0; - if(debugging) { - for(pp = av; *pp; pp++) - D(("sox arg[%d] = %s", pp - av, *pp)); - D(("end args")); - } - xpipe(soxpipe); - soxkid = xfork(); - if(soxkid == 0) { - signal(SIGPIPE, SIG_DFL); - xdup2(t->fd, 0); - xdup2(soxpipe[1], 1); - fcntl(0, F_SETFL, fcntl(0, F_GETFL) & ~O_NONBLOCK); - close(soxpipe[0]); - close(soxpipe[1]); - close(t->fd); - execvp("sox", (char **)av); - _exit(1); - } - D(("forking sox for format conversion (kid = %d)", soxkid)); - close(t->fd); - close(soxpipe[1]); - t->fd = soxpipe[0]; - t->format = config->sample_format; - } -} - /** @brief Read data into a sample buffer * @param t Pointer to track * @return 0 on success, -1 on EOF * - * This is effectively the read callback on @c t->fd. + * This is effectively the read callback on @c t->fd. It is called from the + * main loop whenever the track's file descriptor is readable, assuming the + * buffer has not reached the maximum allowed occupancy. */ -static int fill(struct track *t) { +static int speaker_fill(struct track *t) { size_t where, left; int n; - D(("fill %s: eof=%d used=%zu size=%zu got_format=%d", - t->id, t->eof, t->used, t->size, t->got_format)); + D(("fill %s: eof=%d used=%zu", + t->id, t->eof, t->used)); if(t->eof) return -1; - if(t->used < t->size) { + if(t->used < sizeof t->buffer) { /* there is room left in the buffer */ - where = (t->start + t->used) % t->size; - if(t->got_format) { - /* We are reading audio data, get as much as we can */ - if(where >= t->start) left = t->size - where; - else left = t->start - where; - } else - /* We are still waiting for the format, only get that */ - left = sizeof (ao_sample_format) - t->used; + where = (t->start + t->used) % sizeof t->buffer; + /* Get as much data as we can */ + if(where >= t->start) left = (sizeof t->buffer) - where; + else left = t->start - where; do { n = read(t->fd, t->buffer + where, left); } while(n < 0 && errno == EINTR); @@ -433,333 +210,129 @@ 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->got_format && t->used >= sizeof (ao_sample_format)) { - assert(t->used == sizeof (ao_sample_format)); - /* Check that our assumptions are met. */ - if(t->format.bits & 7) - fatal(0, "bits per sample not a multiple of 8"); - /* If the input format is unsuitable, arrange to translate it */ - enable_translation(t); - /* Make a new buffer for audio data. */ - t->size = bytes_per_frame(&t->format) * t->format.rate * BUFFER_SECONDS; - t->buffer = xmalloc(t->size); - t->used = 0; - t->got_format = 1; - D(("got format for %s", t->id)); - } + if(t->used == sizeof t->buffer) + t->playable = 1; } return 0; } -/** @brief Close the sound device */ +/** @brief Close the sound device + * + * This is called to deactivate the output device when pausing, and also by the + * ALSA backend when changing encoding (in which case the sound device will be + * immediately reactivated). + */ static void idle(void) { D(("idle")); -#if API_ALSA - if(config->speaker_backend == BACKEND_ALSA && pcm) { - int err; - - if((err = snd_pcm_nonblock(pcm, 0)) < 0) - fatal(0, "error calling snd_pcm_nonblock: %d", err); - D(("draining pcm")); - snd_pcm_drain(pcm); - D(("closing pcm")); - snd_pcm_close(pcm); - pcm = 0; - forceplay = 0; - D(("released audio device")); - } -#endif + if(backend->deactivate) + backend->deactivate(); + else + device_state = device_closed; idled = 1; - ready = 0; } /** @brief Abandon the current track */ -static void abandon(void) { +void abandon(void) { struct speaker_message sm; D(("abandon")); memset(&sm, 0, sizeof sm); sm.type = SM_FINISHED; strcpy(sm.id, playing->id); - speaker_send(1, &sm, 0); + speaker_send(1, &sm); removetrack(playing->id); destroy(playing); playing = 0; - forceplay = 0; } -#if API_ALSA -/** @brief Log ALSA parameters */ -static void log_params(snd_pcm_hw_params_t *hwparams, - snd_pcm_sw_params_t *swparams) { - snd_pcm_uframes_t f; - unsigned u; - - return; /* too verbose */ - if(hwparams) { - /* TODO */ - } - if(swparams) { - snd_pcm_sw_params_get_silence_size(swparams, &f); - info("sw silence_size=%lu", (unsigned long)f); - snd_pcm_sw_params_get_silence_threshold(swparams, &f); - info("sw silence_threshold=%lu", (unsigned long)f); - snd_pcm_sw_params_get_sleep_min(swparams, &u); - info("sw sleep_min=%lu", (unsigned long)u); - snd_pcm_sw_params_get_start_threshold(swparams, &f); - info("sw start_threshold=%lu", (unsigned long)f); - snd_pcm_sw_params_get_stop_threshold(swparams, &f); - info("sw stop_threshold=%lu", (unsigned long)f); - snd_pcm_sw_params_get_xfer_align(swparams, &f); - info("sw xfer_align=%lu", (unsigned long)f); - } -} -#endif - /** @brief Enable sound output * * Makes sure the sound device is open and has the right sample format. Return * 0 on success and -1 on error. */ -static int activate(void) { - /* If we don't know the format yet we cannot start. */ - if(!playing->got_format) { - D((" - not got format for %s", playing->id)); - return -1; - } - return backend->activate(); +static void activate(void) { + if(backend->activate) + backend->activate(); + else + device_state = device_open; } -/* Check to see whether the current track has finished playing */ +/** @brief Check whether the current track has finished + * + * The current track is determined to have finished either if the input stream + * eded before the format could be determined (i.e. it is malformed) or the + * input is at end of file and there is less than a frame left unplayed. (So + * it copes with decoders that crash mid-frame.) + */ static void maybe_finished(void) { if(playing && playing->eof - && (!playing->got_format - || playing->used < bytes_per_frame(&playing->format))) + && playing->used < bytes_per_frame(&config->sample_format)) abandon(); } -static void fork_cmd(void) { - pid_t cmdpid; - int pfd[2]; - if(cmdfd != -1) close(cmdfd); - xpipe(pfd); - cmdpid = xfork(); - if(!cmdpid) { - signal(SIGPIPE, SIG_DFL); - xdup2(pfd[0], 0); - close(pfd[0]); - close(pfd[1]); - execl("/bin/sh", "sh", "-c", config->speaker_command, (char *)0); - fatal(errno, "error execing /bin/sh"); - } - close(pfd[0]); - cmdfd = pfd[1]; - D(("forked cmd %d, fd = %d", cmdpid, cmdfd)); +/** @brief Return nonzero if we want to play some audio + * + * We want to play audio if there is a current track; and it is not paused; and + * it is playable according to the rules for @ref track::playable. + */ +static int playable(void) { + return playing + && !paused + && playing->playable; } -static void play(size_t frames) { - size_t avail_bytes, write_bytes, written_frames; +/** @brief Play up to @p frames frames of audio + * + * It is always safe to call this function. + * - If @ref playing is 0 then it will just return + * - If @ref paused is non-0 then it will just return + * - If @ref device_state != @ref device_open then it will call activate() and + * return if it it fails. + * - If there is not enough audio to play then it play what is available. + * + * If there are not enough frames to play then whatever is available is played + * instead. It is up to mainloop() to ensure that speaker_play() is not called + * when unreasonably only an small amounts of data is available to play. + */ +static void speaker_play(size_t frames) { + size_t avail_frames, avail_bytes, written_frames; ssize_t written_bytes; - struct rtp_header header; - struct iovec vec[2]; - if(activate()) { - if(playing) - forceplay = frames; - else - forceplay = 0; /* Must have called abandon() */ + /* Make sure there's a track to play and it is not paused */ + if(!playable()) return; + /* Make sure the output device is open */ + if(device_state != device_open) { + activate(); + if(device_state != device_open) + return; } D(("play: play %zu/%zu%s %dHz %db %dc", frames, playing->used / bpf, playing->eof ? " EOF" : "", - playing->format.rate, - playing->format.bits, - playing->format.channels)); - /* If we haven't got enough bytes yet wait until we have. Exception: when - * we are at eof. */ - if(playing->used < frames * bpf && !playing->eof) { - forceplay = frames; - return; - } - /* We have got enough data so don't force play again */ - forceplay = 0; + config->sample_format.rate, + config->sample_format.bits, + config->sample_format.channels)); /* Figure out how many frames there are available to write */ - if(playing->start + playing->used > playing->size) - avail_bytes = playing->size - playing->start; + if(playing->start + playing->used > sizeof playing->buffer) + /* The ring buffer is currently wrapped, only play up to the wrap point */ + avail_bytes = (sizeof playing->buffer) - playing->start; else + /* The ring buffer is not wrapped, can play the lot */ avail_bytes = playing->used; - - switch(config->speaker_backend) { -#if API_ALSA - case BACKEND_ALSA: { - snd_pcm_sframes_t pcm_written_frames; - size_t avail_frames; - int err; - - avail_frames = avail_bytes / bpf; - if(avail_frames > frames) - avail_frames = frames; - if(!avail_frames) - return; - pcm_written_frames = snd_pcm_writei(pcm, - playing->buffer + playing->start, - avail_frames); - D(("actually play %zu frames, wrote %d", - avail_frames, (int)pcm_written_frames)); - if(pcm_written_frames < 0) { - switch(pcm_written_frames) { - case -EPIPE: /* underrun */ - error(0, "snd_pcm_writei reports underrun"); - if((err = snd_pcm_prepare(pcm)) < 0) - fatal(0, "error calling snd_pcm_prepare: %d", err); - return; - case -EAGAIN: - return; - default: - fatal(0, "error calling snd_pcm_writei: %d", - (int)pcm_written_frames); - } - } - written_frames = pcm_written_frames; - written_bytes = written_frames * bpf; - break; - } -#endif - case BACKEND_COMMAND: - if(avail_bytes > frames * bpf) - avail_bytes = frames * bpf; - written_bytes = write(cmdfd, playing->buffer + playing->start, - avail_bytes); - D(("actually play %zu bytes, wrote %d", - avail_bytes, (int)written_bytes)); - if(written_bytes < 0) { - switch(errno) { - case EPIPE: - error(0, "hmm, command died; trying another"); - fork_cmd(); - return; - case EAGAIN: - return; - } - } - written_frames = written_bytes / bpf; /* good enough */ - break; - case BACKEND_NETWORK: - /* We transmit using RTP (RFC3550) and attempt to conform to the internet - * AVT profile (RFC3551). */ - - if(idled) { - /* There may have been a gap. Fix up the RTP time accordingly. */ - struct timeval now; - uint64_t delta; - uint64_t target_rtp_time; - - /* Find the current time */ - xgettimeofday(&now, 0); - /* Find the number of microseconds elapsed since rtp_time=0 */ - delta = tvsub_us(now, rtp_time_0); - assert(delta <= UINT64_MAX / 88200); - target_rtp_time = (delta * playing->format.rate - * playing->format.channels) / 1000000; - /* Overflows at ~6 years uptime with 44100Hz stereo */ - - /* rtp_time is the number of samples we've played. NB that we play - * RTP_AHEAD_MS ahead of ourselves, so it may legitimately be ahead of - * the value we deduce from time comparison. - * - * Suppose we have 1s track started at t=0, and another track begins to - * play at t=2s. Suppose RTP_AHEAD_MS=1000 and 44100Hz stereo. In that - * case we'll send 1s of audio as fast as we can, giving rtp_time=88200. - * rtp_time stops at this point. - * - * At t=2s we'll have calculated target_rtp_time=176400. In this case we - * set rtp_time=176400 and the player can correctly conclude that it - * should leave 1s between the tracks. - * - * Suppose instead that the second track arrives at t=0.5s, and that - * we've managed to transmit the whole of the first track already. We'll - * have target_rtp_time=44100. - * - * The desired behaviour is to play the second track back to back with - * first. In this case therefore we do not modify rtp_time. - * - * Is it ever right to reduce rtp_time? No; for that would imply - * transmitting packets with overlapping timestamp ranges, which does not - * make sense. - */ - if(target_rtp_time > rtp_time) { - /* More time has elapsed than we've transmitted samples. That implies - * we've been 'sending' silence. */ - info("advancing rtp_time by %"PRIu64" samples", - target_rtp_time - rtp_time); - rtp_time = target_rtp_time; - } else if(target_rtp_time < rtp_time) { - const int64_t samples_ahead = ((uint64_t)RTP_AHEAD_MS - * config->sample_format.rate - * config->sample_format.channels - / 1000); - - if(target_rtp_time + samples_ahead < rtp_time) { - info("reversing rtp_time by %"PRIu64" samples", - rtp_time - target_rtp_time); - } - } - } - header.vpxcc = 2 << 6; /* V=2, P=0, X=0, CC=0 */ - header.seq = htons(rtp_seq++); - header.timestamp = htonl((uint32_t)rtp_time); - header.ssrc = rtp_id; - header.mpt = (idled ? 0x80 : 0x00) | 10; - /* 10 = L16 = 16-bit x 2 x 44100KHz. We ought to deduce this value from - * the sample rate (in a library somewhere so that configuration.c can rule - * out invalid rates). - */ - idled = 0; - if(avail_bytes > NETWORK_BYTES - sizeof header) { - avail_bytes = NETWORK_BYTES - sizeof header; - /* Always send a whole number of frames */ - avail_bytes -= avail_bytes % bpf; - } - /* "The RTP clock rate used for generating the RTP timestamp is independent - * of the number of channels and the encoding; it equals the number of - * sampling periods per second. For N-channel encodings, each sampling - * period (say, 1/8000 of a second) generates N samples. (This terminology - * is standard, but somewhat confusing, as the total number of samples - * generated per second is then the sampling rate times the channel - * count.)" - */ - write_bytes = avail_bytes; - if(write_bytes) { - vec[0].iov_base = (void *)&header; - vec[0].iov_len = sizeof header; - vec[1].iov_base = playing->buffer + playing->start; - vec[1].iov_len = avail_bytes; - do { - written_bytes = writev(bfd, - vec, - 2); - } while(written_bytes < 0 && errno == EINTR); - if(written_bytes < 0) { - error(errno, "error transmitting audio data"); - ++audio_errors; - if(audio_errors == 10) - fatal(0, "too many audio errors"); - return; - } - } else - audio_errors /= 2; - written_bytes = avail_bytes; - written_frames = written_bytes / bpf; - /* Advance RTP's notion of the time */ - rtp_time += written_frames * playing->format.channels; - break; - default: - assert(!"reached"); - } + avail_frames = avail_bytes / bpf; + /* Only play up to the requested amount */ + if(avail_frames > frames) + avail_frames = frames; + if(!avail_frames) + return; + /* Play it, Sam */ + written_frames = backend->play(avail_frames); + written_bytes = written_frames * bpf; /* written_bytes and written_frames had better both be set and correct by * this point */ playing->start += written_bytes; @@ -767,21 +340,27 @@ static void play(size_t frames) { playing->played += written_frames; /* If the pointer is at the end of the buffer (or the buffer is completely * empty) wrap it back to the start. */ - if(!playing->used || playing->start == playing->size) + if(!playing->used || playing->start == (sizeof playing->buffer)) playing->start = 0; + /* If the buffer emptied out mark the track as unplayably */ + if(!playing->used && !playing->eof) { + error(0, "track buffer emptied"); + playing->playable = 0; + } frames -= written_frames; + return; } /* Notify the server what we're up to. */ static void report(void) { struct speaker_message sm; - if(playing && playing->buffer != (void *)&playing->format) { + if(playing) { memset(&sm, 0, sizeof sm); sm.type = paused ? SM_PAUSED : SM_PLAYING; strcpy(sm.id, playing->id); - sm.data = playing->played / playing->format.rate; - speaker_send(1, &sm, 0); + sm.data = playing->played / config->sample_format.rate; + speaker_send(1, &sm); } time(&last_report); } @@ -796,7 +375,7 @@ static void reap(int __attribute__((unused)) sig) { signal(SIGCHLD, reap); } -static int addfd(int fd, int events) { +int addfd(int fd, int events) { if(fdno < NFDS) { fds[fdno].fd = fd; fds[fdno].events = events; @@ -805,382 +384,66 @@ static int addfd(int fd, int events) { return -1; } -#if API_ALSA -/** @brief ALSA backend initialization */ -static void alsa_init(void) { - info("selected ALSA backend"); -} - -/** @brief ALSA backend activation */ -static int 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) { - snd_pcm_hw_params_t *hwparams; - snd_pcm_sw_params_t *swparams; - snd_pcm_uframes_t pcm_bufsize; - int err; - int sample_format = 0; - unsigned rate; - - D(("snd_pcm_open")); - if((err = snd_pcm_open(&pcm, - config->device, - SND_PCM_STREAM_PLAYBACK, - SND_PCM_NONBLOCK))) { - error(0, "error from snd_pcm_open: %d", err); - goto error; - } - snd_pcm_hw_params_alloca(&hwparams); - D(("set up hw params")); - if((err = snd_pcm_hw_params_any(pcm, hwparams)) < 0) - fatal(0, "error from snd_pcm_hw_params_any: %d", err); - if((err = snd_pcm_hw_params_set_access(pcm, hwparams, - SND_PCM_ACCESS_RW_INTERLEAVED)) < 0) - fatal(0, "error from snd_pcm_hw_params_set_access: %d", err); - switch(playing->format.bits) { - case 8: - sample_format = SND_PCM_FORMAT_S8; - break; - case 16: - switch(playing->format.byte_format) { - case AO_FMT_NATIVE: sample_format = SND_PCM_FORMAT_S16; break; - case AO_FMT_LITTLE: sample_format = SND_PCM_FORMAT_S16_LE; break; - case AO_FMT_BIG: sample_format = SND_PCM_FORMAT_S16_BE; break; - error(0, "unrecognized byte format %d", playing->format.byte_format); - goto fatal; - } - break; - default: - error(0, "unsupported sample size %d", playing->format.bits); - goto fatal; - } - if((err = snd_pcm_hw_params_set_format(pcm, hwparams, - sample_format)) < 0) { - error(0, "error from snd_pcm_hw_params_set_format (%d): %d", - sample_format, err); - goto fatal; - } - rate = playing->format.rate; - if((err = snd_pcm_hw_params_set_rate_near(pcm, hwparams, &rate, 0)) < 0) { - error(0, "error from snd_pcm_hw_params_set_rate (%d): %d", - playing->format.rate, err); - goto fatal; - } - if(rate != (unsigned)playing->format.rate) - info("want rate %d, got %u", playing->format.rate, rate); - if((err = snd_pcm_hw_params_set_channels(pcm, hwparams, - playing->format.channels)) < 0) { - error(0, "error from snd_pcm_hw_params_set_channels (%d): %d", - playing->format.channels, err); - goto fatal; - } - bufsize = 3 * FRAMES; - pcm_bufsize = bufsize; - if((err = snd_pcm_hw_params_set_buffer_size_near(pcm, hwparams, - &pcm_bufsize)) < 0) - fatal(0, "error from snd_pcm_hw_params_set_buffer_size (%d): %d", - 3 * FRAMES, err); - if(pcm_bufsize != 3 * FRAMES && pcm_bufsize != last_pcm_bufsize) - info("asked for PCM buffer of %d frames, got %d", - 3 * FRAMES, (int)pcm_bufsize); - last_pcm_bufsize = pcm_bufsize; - if((err = snd_pcm_hw_params(pcm, hwparams)) < 0) - fatal(0, "error calling snd_pcm_hw_params: %d", err); - D(("set up sw params")); - snd_pcm_sw_params_alloca(&swparams); - if((err = snd_pcm_sw_params_current(pcm, swparams)) < 0) - fatal(0, "error calling snd_pcm_sw_params_current: %d", err); - if((err = snd_pcm_sw_params_set_avail_min(pcm, swparams, FRAMES)) < 0) - fatal(0, "error calling snd_pcm_sw_params_set_avail_min %d: %d", - FRAMES, err); - if((err = snd_pcm_sw_params(pcm, swparams)) < 0) - fatal(0, "error calling snd_pcm_sw_params: %d", err); - pcm_format = playing->format; - bpf = bytes_per_frame(&pcm_format); - D(("acquired audio device")); - log_params(hwparams, swparams); - ready = 1; - } - return 0; -fatal: - abandon(); -error: - /* We assume the error is temporary and that we'll retry in a bit. */ - if(pcm) { - snd_pcm_close(pcm); - pcm = 0; - } - return -1; -} -#endif - -/** @brief Command backend initialization */ -static void command_init(void) { - info("selected command backend"); - fork_cmd(); -} - -/** @brief Command backend activation */ -static int command_activate(void) { - if(!ready) { - bufsize = 3 * FRAMES; - bpf = bytes_per_frame(&config->sample_format); - D(("acquired audio device")); - ready = 1; - } - return 0; -} - -/** @brief Network backend initialization */ -static void network_init(void) { - struct addrinfo *res, *sres; - static const struct addrinfo pref = { - 0, - PF_INET, - SOCK_DGRAM, - IPPROTO_UDP, - 0, - 0, - 0, - 0 - }; - static const struct addrinfo prefbind = { - AI_PASSIVE, - PF_INET, - SOCK_DGRAM, - IPPROTO_UDP, - 0, - 0, - 0, - 0 - }; - static const int one = 1; - int sndbuf, target_sndbuf = 131072; - socklen_t len; - char *sockname, *ssockname; - - res = get_address(&config->broadcast, &pref, &sockname); - if(!res) exit(-1); - if(config->broadcast_from.n) { - sres = get_address(&config->broadcast_from, &prefbind, &ssockname); - if(!sres) exit(-1); - } else - sres = 0; - if((bfd = socket(res->ai_family, - res->ai_socktype, - res->ai_protocol)) < 0) - fatal(errno, "error creating broadcast socket"); - if(setsockopt(bfd, SOL_SOCKET, SO_BROADCAST, &one, sizeof one) < 0) - fatal(errno, "error setting SO_BROADCAST on broadcast socket"); - len = sizeof sndbuf; - if(getsockopt(bfd, SOL_SOCKET, SO_SNDBUF, - &sndbuf, &len) < 0) - fatal(errno, "error getting SO_SNDBUF"); - if(target_sndbuf > sndbuf) { - if(setsockopt(bfd, SOL_SOCKET, SO_SNDBUF, - &target_sndbuf, sizeof target_sndbuf) < 0) - error(errno, "error setting SO_SNDBUF to %d", target_sndbuf); - else - info("changed socket send buffer size from %d to %d", - sndbuf, target_sndbuf); - } else - info("default socket send buffer is %d", - sndbuf); - /* We might well want to set additional broadcast- or multicast-related - * options here */ - if(sres && bind(bfd, sres->ai_addr, sres->ai_addrlen) < 0) - fatal(errno, "error binding broadcast socket to %s", ssockname); - if(connect(bfd, res->ai_addr, res->ai_addrlen) < 0) - fatal(errno, "error connecting broadcast socket to %s", sockname); - /* Select an SSRC */ - gcry_randomize(&rtp_id, sizeof rtp_id, GCRY_STRONG_RANDOM); - info("selected network backend, sending to %s", sockname); - if(config->sample_format.byte_format != AO_FMT_BIG) { - info("forcing big-endian sample format"); - config->sample_format.byte_format = AO_FMT_BIG; - } -} - -/** @brief Network backend activation */ -static int network_activate(void) { - if(!ready) { - bufsize = 3 * FRAMES; - bpf = bytes_per_frame(&config->sample_format); - D(("acquired audio device")); - ready = 1; - } - return 0; -} - /** @brief Table of speaker backends */ -static const struct speaker_backend backends[] = { -#if API_ALSA - { - BACKEND_ALSA, - 0, - alsa_init, - alsa_activate - }, +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 - { - BACKEND_COMMAND, - FIXED_FORMAT, - command_init, - command_activate - }, - { - BACKEND_NETWORK, - FIXED_FORMAT, - network_init, - network_activate - }, - { -1, 0, 0, 0 } + 0 }; -int main(int argc, char **argv) { - int n, fd, stdin_slot, alsa_slots, cmdfd_slot, bfd_slot, poke, timeout; +/** @brief Main event loop */ +static void mainloop(void) { struct track *t; struct speaker_message sm; -#if API_ALSA - int alsa_nslots = -1, err; -#endif + int n, fd, stdin_slot, timeout, listen_slot; - set_progname(argv); - if(!setlocale(LC_CTYPE, "")) fatal(errno, "error calling setlocale"); - while((n = getopt_long(argc, argv, "hVc:dD", options, 0)) >= 0) { - switch(n) { - case 'h': help(); - case 'V': version(); - case 'c': configfile = optarg; break; - case 'd': debugging = 1; break; - case 'D': debugging = 0; break; - default: fatal(0, "invalid option"); - } - } - if(getenv("DISORDER_DEBUG_SPEAKER")) debugging = 1; - /* If stderr is a TTY then log there, otherwise to syslog. */ - if(!isatty(2)) { - openlog(progname, LOG_PID, LOG_DAEMON); - log_default = &log_syslog; - } - if(config_read()) fatal(0, "cannot read configuration"); - /* ignore SIGPIPE */ - signal(SIGPIPE, SIG_IGN); - /* reap kids */ - signal(SIGCHLD, reap); - /* set nice value */ - xnice(config->nice_speaker); - /* change user */ - become_mortal(); - /* make sure we're not root, whatever the config says */ - if(getuid() == 0 || geteuid() == 0) fatal(0, "do not run as root"); - /* 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); + /* Also always ready for inbound connections */ + listen_slot = addfd(listenfd, POLLIN); /* Try to read sample data for the currently playing track if there is * buffer space. */ - if(playing && !playing->eof && playing->used < playing->size) { + if(playing + && playing->fd >= 0 + && !playing->eof + && playing->used < (sizeof playing->buffer)) playing->slot = addfd(playing->fd, POLLIN); - } else if(playing) + else if(playing) playing->slot = -1; - /* If forceplay is set then wait until it succeeds before waiting on the - * sound device. */ - alsa_slots = -1; - cmdfd_slot = -1; - bfd_slot = -1; - /* By default we will wait up to a second before thinking about current - * state. */ - timeout = 1000; - if(ready && !forceplay) { - switch(config->speaker_backend) { - case BACKEND_COMMAND: - /* We send sample data to the subprocess as fast as it can accept it. - * This isn't ideal as pause latency can be very high as a result. */ - if(cmdfd >= 0) - cmdfd_slot = addfd(cmdfd, POLLOUT); - break; - case BACKEND_NETWORK: { - struct timeval now; - uint64_t target_us; - uint64_t target_rtp_time; - const int64_t samples_ahead = ((uint64_t)RTP_AHEAD_MS - * config->sample_format.rate - * config->sample_format.channels - / 1000); -#if 0 - static unsigned logit; -#endif - - /* If we're starting then initialize the base time */ - if(!rtp_time) - xgettimeofday(&rtp_time_0, 0); - /* We send audio data whenever we get RTP_AHEAD seconds or more - * behind */ - xgettimeofday(&now, 0); - target_us = tvsub_us(now, rtp_time_0); - assert(target_us <= UINT64_MAX / 88200); - target_rtp_time = (target_us * config->sample_format.rate - * config->sample_format.channels) - - / 1000000; -#if 0 - /* TODO remove logging guff */ - if(!(logit++ & 1023)) - info("rtp_time %llu target %llu difference %lld [%lld]", - rtp_time, target_rtp_time, - rtp_time - target_rtp_time, - samples_ahead); -#endif - if((int64_t)(rtp_time - target_rtp_time) < samples_ahead) - bfd_slot = addfd(bfd, POLLOUT); - break; - } -#if API_ALSA - case BACKEND_ALSA: { - /* We send sample data to ALSA as fast as it can accept it, relying on - * the fact that it has a relatively small buffer to minimize pause - * latency. */ - int retry = 3; - - alsa_slots = fdno; - do { - retry = 0; - alsa_nslots = snd_pcm_poll_descriptors(pcm, &fds[fdno], NFDS - fdno); - if((alsa_nslots <= 0 - || !(fds[alsa_slots].events & POLLOUT)) - && snd_pcm_state(pcm) == SND_PCM_STATE_XRUN) { - error(0, "underrun detected after call to snd_pcm_poll_descriptors()"); - if((err = snd_pcm_prepare(pcm))) - fatal(0, "error calling snd_pcm_prepare: %d", err); - } else - break; - } while(retry-- > 0); - if(alsa_nslots >= 0) - fdno += alsa_nslots; - break; - } -#endif - default: - assert(!"unknown backend"); - } + if(playable()) { + /* We want to play some audio. If the device is closed then we attempt + * to open it. */ + if(device_state == device_closed) + activate(); + /* If the device is (now) open then we will wait up until it is ready for + * more. If something went wrong then we should have device_error + * instead, but the post-poll code will cope even if it's + * device_closed. */ + if(device_state == device_open) + backend->beforepoll(&timeout); } /* If any other tracks don't have a full buffer, try to read sample data - * from them. */ + * from them. We do this last of all, so that if we run out of slots, + * nothing important can't be monitored. */ for(t = tracks; t; t = t->next) if(t != playing) { - if(!t->eof && t->used < t->size) { + if(t->fd >= 0 + && !t->eof + && t->used < sizeof t->buffer) { t->slot = addfd(t->fd, POLLIN | POLLHUP); } else t->slot = -1; @@ -1192,64 +455,75 @@ 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)) - play(3 * FRAMES); - } else - poke = 1; - break; - case BACKEND_NETWORK: - if(bfd_slot != -1) { - if(fds[bfd_slot].revents & (POLLOUT | POLLERR)) - play(3 * FRAMES); - } else - poke = 1; - break; + if(playable()) { + /* We want to play some audio */ + if(device_state == device_open) { + if(backend->ready()) + speaker_play(3 * FRAMES); + } else { + /* We must be in _closed or _error, and it should be the latter, but we + * cope with either. + * + * We most likely timed out, so now is a good time to retry. + * speaker_play() knows to re-activate the device if necessary. + */ + speaker_play(3 * FRAMES); + } } - if(poke) { - /* Some attempt to play must have failed */ - if(playing && !paused) - play(forceplay); - else - forceplay = 0; /* just in case */ + /* Perhaps a connection has arrived */ + if(fds[listen_slot].revents & POLLIN) { + struct sockaddr_un addr; + socklen_t addrlen = sizeof addr; + uint32_t l; + char id[24]; + + if((fd = accept(listenfd, (struct sockaddr *)&addr, &addrlen)) >= 0) { + blocking(fd); + if(read(fd, &l, sizeof l) < 4) { + error(errno, "reading length from inbound connection"); + xclose(fd); + } else if(l >= sizeof id) { + error(0, "id length too long"); + xclose(fd); + } else if(read(fd, id, l) < (ssize_t)l) { + error(errno, "reading id from inbound connection"); + xclose(fd); + } else { + id[l] = 0; + D(("id %s fd %d", id, fd)); + t = findtrack(id, 1/*create*/); + write(fd, "", 1); /* write an ack */ + if(t->fd != -1) { + error(0, "%s: already got a connection", id); + xclose(fd); + } else { + nonblock(fd); + t->fd = fd; /* yay */ + } + } + } else + error(errno, "accept"); } /* Perhaps we have a command to process */ if(fds[stdin_slot].revents & POLLIN) { - n = speaker_recv(0, &sm, &fd); + /* There might (in theory) be several commands queued up, but in general + * this won't be the case, so we don't bother looping around to pick them + * all up. */ + n = speaker_recv(0, &sm); + /* TODO */ if(n > 0) switch(sm.type) { - case SM_PREPARE: - D(("SM_PREPARE %s %d", sm.id, fd)); - if(fd == -1) fatal(0, "got SM_PREPARE but no file descriptor"); - t = findtrack(sm.id, 1); - acquire(t, fd); - break; case SM_PLAY: - D(("SM_PLAY %s %d", sm.id, fd)); if(playing) fatal(0, "got SM_PLAY but already playing something"); t = findtrack(sm.id, 1); - if(fd != -1) acquire(t, fd); + D(("SM_PLAY %s fd %d", t->id, t->fd)); + if(t->fd == -1) + error(0, "cannot play track because no connection arrived"); playing = t; - play(bufsize); + /* We attempt to play straight away rather than going round the loop. + * speaker_play() is clever enough to perform any activation that is + * required. */ + speaker_play(3 * FRAMES); report(); break; case SM_PAUSE: @@ -1261,29 +535,43 @@ 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); + 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: @@ -1292,21 +580,93 @@ int main(int argc, char **argv) { } /* Read in any buffered data */ for(t = tracks; t; t = t->next) - if(t->slot != -1 && (fds[t->slot].revents & (POLLIN | POLLHUP))) - fill(t); - /* We might be able to play now */ - if(ready && forceplay && playing && !paused) - play(forceplay); + if(t->fd != -1 + && t->slot != -1 + && (fds[t->slot].revents & (POLLIN | POLLHUP))) + speaker_fill(t); /* Maybe we finished playing a track somewhere in the above */ maybe_finished(); /* If we don't need the sound device for now then close it for the benefit * of anyone else who wants it. */ - if((!playing || paused) && ready) + if((!playing || paused) && device_state == device_open) idle(); /* If we've not reported out state for a second do so now. */ if(time(0) > last_report) report(); } +} + +int main(int argc, char **argv) { + int n, logsyslog = !isatty(2); + struct sockaddr_un addr; + static const int one = 1; + struct speaker_message sm; + const char *d; + char *dir; + + set_progname(argv); + if(!setlocale(LC_CTYPE, "")) fatal(errno, "error calling setlocale"); + while((n = getopt_long(argc, argv, "hVc:dDSs", options, 0)) >= 0) { + switch(n) { + case 'h': help(); + case 'V': version("disorder-speaker"); + case 'c': configfile = optarg; break; + case 'd': debugging = 1; break; + case 'D': debugging = 0; break; + case 'S': logsyslog = 0; break; + case 's': logsyslog = 1; break; + default: fatal(0, "invalid option"); + } + } + if((d = getenv("DISORDER_DEBUG_SPEAKER"))) debugging = atoi(d); + if(logsyslog) { + openlog(progname, LOG_PID, LOG_DAEMON); + log_default = &log_syslog; + } + if(config_read(1)) fatal(0, "cannot read configuration"); + bpf = bytes_per_frame(&config->sample_format); + /* ignore SIGPIPE */ + signal(SIGPIPE, SIG_IGN); + /* reap kids */ + signal(SIGCHLD, reap); + /* set nice value */ + xnice(config->nice_speaker); + /* change user */ + become_mortal(); + /* make sure we're not root, whatever the config says */ + if(getuid() == 0 || geteuid() == 0) fatal(0, "do not run as root"); + /* identify the backend used to play */ + for(n = 0; backends[n]; ++n) + if(backends[n]->backend == config->api) + break; + if(!backends[n]) + fatal(0, "unsupported api %d", config->api); + backend = backends[n]; + /* backend-specific initialization */ + backend->init(); + /* create the socket directory */ + byte_xasprintf(&dir, "%s/speaker", config->home); + unlink(dir); /* might be a leftover socket */ + if(mkdir(dir, 0700) < 0 && errno != EEXIST) + fatal(errno, "error creating %s", dir); + /* set up the listen socket */ + listenfd = xsocket(PF_UNIX, SOCK_STREAM, 0); + memset(&addr, 0, sizeof addr); + addr.sun_family = AF_UNIX; + snprintf(addr.sun_path, sizeof addr.sun_path, "%s/speaker/socket", + config->home); + if(unlink(addr.sun_path) < 0 && errno != ENOENT) + error(errno, "removing %s", addr.sun_path); + xsetsockopt(listenfd, SOL_SOCKET, SO_REUSEADDR, &one, sizeof one); + if(bind(listenfd, (const struct sockaddr *)&addr, sizeof addr) < 0) + fatal(errno, "error binding socket to %s", addr.sun_path); + xlisten(listenfd, 128); + nonblock(listenfd); + info("listening on %s", addr.sun_path); + memset(&sm, 0, sizeof sm); + sm.type = SM_READY; + speaker_send(1, &sm); + mainloop(); info("stopped (parent terminated)"); exit(0); }