X-Git-Url: http://www.chiark.greenend.org.uk/ucgi/~mdw/git/disorder/blobdiff_plain/f8f8039fc44104b43b94467d17a8d71fc40f219b..6d2d327ca57fefaddceba10eb323451f8150e95d:/server/speaker.c diff --git a/server/speaker.c b/server/speaker.c index 1dc90e4..aa09c02 100644 --- a/server/speaker.c +++ b/server/speaker.c @@ -29,15 +29,9 @@ * 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. - * - * 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. + * Inbound data is expected to match @c config->sample_format. In normal use + * this is arranged by the @c disorder-normalize program (see @ref + * server/normalize.c). * * @b Garbage @b Collection. This program deliberately does not use the * garbage collector even though it might be convenient to do so. This is for @@ -89,7 +83,7 @@ struct track *tracks; struct track *playing; /** @brief Number of bytes pre frame */ -size_t device_bpf; +size_t bpf; /** @brief Array of file descriptors for poll() */ struct pollfd fds[NFDS]; @@ -103,14 +97,6 @@ static int paused; /* pause status */ /** @brief The current device state */ enum device_states device_state; -/** @brief The current device 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. - */ -ao_sample_format device_format; - /** @brief Set when idled * * This is set when the sound device is deliberately closed by idle(). @@ -153,7 +139,7 @@ static void version(void) { } /** @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; } @@ -170,9 +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; } @@ -193,7 +176,6 @@ 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); } @@ -206,96 +188,6 @@ static void acquire(struct track *t, int fd) { nonblock(fd); } -/** @brief Return true if A and B denote identical libao formats, else false */ -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 @@ -308,19 +200,15 @@ static int 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); @@ -334,20 +222,6 @@ static int fill(struct track *t) { 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)); - } } return 0; } @@ -387,22 +261,10 @@ void abandon(void) { * 0 on success and -1 on error. */ 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; - } - if(backend->flags & FIXED_FORMAT) - device_format = config->sample_format; - if(backend->activate) { + if(backend->activate) backend->activate(); - } else { - assert(backend->flags & FIXED_FORMAT); - /* ...otherwise device_format not set */ + else device_state = device_open; - } - if(device_state == device_open) - device_bpf = bytes_per_frame(&device_format); } /** @brief Check whether the current track has finished @@ -415,8 +277,7 @@ static void activate(void) { 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(); } @@ -440,26 +301,25 @@ static void play(size_t frames) { /* 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)) { + /* 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 / device_bpf, + 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)); + 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) + if(playing->start + playing->used > sizeof playing->buffer) /* The ring buffer is currently wrapped, only play up to the wrap point */ - avail_bytes = playing->size - playing->start; + 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 / device_bpf; + avail_frames = avail_bytes / bpf; /* Only play up to the requested amount */ if(avail_frames > frames) avail_frames = frames; @@ -467,7 +327,7 @@ static void play(size_t frames) { return; /* Play it, Sam */ written_frames = backend->play(avail_frames); - written_bytes = written_frames * device_bpf; + written_bytes = written_frames * bpf; /* written_bytes and written_frames had better both be set and correct by * this point */ playing->start += written_bytes; @@ -475,7 +335,7 @@ 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; frames -= written_frames; return; @@ -485,11 +345,11 @@ static void play(size_t frames) { 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; + sm.data = playing->played / config->sample_format.rate; speaker_send(1, &sm, 0); } time(&last_report); @@ -551,7 +411,7 @@ static void mainloop(void) { 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 < (sizeof playing->buffer)) playing->slot = addfd(playing->fd, POLLIN); else if(playing) playing->slot = -1; @@ -572,7 +432,7 @@ static void mainloop(void) { * 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->eof && t->used < sizeof t->buffer) { t->slot = addfd(t->fd, POLLIN | POLLHUP); } else t->slot = -1; @@ -702,6 +562,7 @@ int main(int argc, char **argv) { log_default = &log_syslog; } if(config_read()) fatal(0, "cannot read configuration"); + bpf = bytes_per_frame(&config->sample_format); /* ignore SIGPIPE */ signal(SIGPIPE, SIG_IGN); /* reap kids */