X-Git-Url: http://www.chiark.greenend.org.uk/ucgi/~mdw/git/disorder/blobdiff_plain/3a23a6a5d9a9973ebb9d62644d2d68a79483d31d..6d2d327ca57fefaddceba10eb323451f8150e95d:/server/speaker.c diff --git a/server/speaker.c b/server/speaker.c index aebd930..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 @@ -72,10 +66,6 @@ #include #include #include -#include -#include -#include -#include #include "configuration.h" #include "syscalls.h" @@ -84,89 +74,34 @@ #include "mem.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 - /** @brief Linked list of all prepared tracks */ struct track *tracks; /** @brief Playing track, or NULL */ struct track *playing; +/** @brief Number of bytes pre frame */ +size_t bpf; + +/** @brief Array of file descriptors for poll() */ +struct pollfd fds[NFDS]; + +/** @brief Next free slot in @ref fds */ +int fdno; + 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 */ -#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 */ -#endif /** @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 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 - * - * This counts the number of samples played (NB not the number of frames - * played). - * - * 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. - */ -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; - -/** @brief RTP packet sequence number */ -static uint16_t rtp_seq; - -/** @brief RTP SSRC */ -static uint32_t rtp_id; - /** @brief Set when idled * * This is set when the sound device is deliberately closed by idle(). */ -static int idled; /* set when idled */ - -/** @brief Error counter */ -static int audio_errors; +int idled; /** @brief Selected backend */ static const struct speaker_backend *backend; @@ -204,7 +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; } @@ -221,9 +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; } @@ -244,7 +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); } @@ -257,96 +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 */ -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 @@ -359,19 +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); @@ -385,20 +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; } @@ -419,7 +242,7 @@ static void idle(void) { } /** @brief Abandon the current track */ -static void abandon(void) { +void abandon(void) { struct speaker_message sm; D(("abandon")); @@ -438,22 +261,10 @@ static 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) - bpf = bytes_per_frame(&device_format); } /** @brief Check whether the current track has finished @@ -466,8 +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(); } @@ -491,22 +301,21 @@ 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 / 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; @@ -526,7 +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; @@ -536,11 +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); @@ -556,7 +365,7 @@ 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; @@ -565,549 +374,14 @@ 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 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 void alsa_activate(void) { - /* If we need to change format then close the current device. */ - 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; - 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; - } - 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", - 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); - device_format = playing->format; - D(("acquired audio device")); - log_params(hwparams, swparams); - device_state = device_open; - } - return; -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; - device_state = device_error; - } - return; -} - -/** @brief Play via ALSA */ -static size_t alsa_play(size_t frames) { - snd_pcm_sframes_t pcm_written_frames; - int err; - - pcm_written_frames = snd_pcm_writei(pcm, - playing->buffer + playing->start, - frames); - D(("actually play %zu frames, wrote %d", - 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 0; - case -EAGAIN: - return 0; - default: - fatal(0, "error calling snd_pcm_writei: %d", - (int)pcm_written_frames); - } - } else - return pcm_written_frames; -} - -static int alsa_slots, alsa_nslots = -1; - -/** @brief Fill in poll fd array for ALSA */ -static void alsa_beforepoll(void) { - /* 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, err; - - 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; -} - -/** @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"); - fork_cmd(); -} - -/** @brief Play to a subprocess */ -static size_t command_play(size_t frames) { - size_t bytes = frames * bpf; - int written_bytes; - - written_bytes = write(cmdfd, playing->buffer + playing->start, bytes); - D(("actually play %zu bytes, wrote %d", - bytes, written_bytes)); - if(written_bytes < 0) { - switch(errno) { - case EPIPE: - error(0, "hmm, command died; trying another"); - fork_cmd(); - return 0; - case EAGAIN: - return 0; - default: - fatal(errno, "error writing to subprocess"); - } - } else - return written_bytes / bpf; -} - -static int cmdfd_slot; - -/** @brief Update poll array for writing to subprocess */ -static void command_beforepoll(void) { - /* 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); -} - -/** @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 */ -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 Play over the network */ -static size_t network_play(size_t frames) { - struct rtp_header header; - struct iovec vec[2]; - size_t bytes = frames * bpf, written_frames; - int written_bytes; - /* 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. - */ - 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. */ - 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(bytes > NETWORK_BYTES - sizeof header) { - bytes = NETWORK_BYTES - sizeof header; - /* Always send a whole number of frames */ - bytes -= 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.)" - */ - vec[0].iov_base = (void *)&header; - vec[0].iov_len = sizeof header; - vec[1].iov_base = playing->buffer + playing->start; - vec[1].iov_len = 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 0; - } else - audio_errors /= 2; - written_bytes -= sizeof (struct rtp_header); - written_frames = written_bytes / bpf; - /* Advance RTP's notion of the time */ - rtp_time += written_frames * playing->format.channels; - return written_frames; -} - -static int bfd_slot; - -/** @brief Set up poll array for network play */ -static void network_beforepoll(void) { - 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 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((int64_t)(rtp_time - target_rtp_time) < samples_ahead) - 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[] = { +static const struct speaker_backend *backends[] = { #if API_ALSA - { - BACKEND_ALSA, - 0, - alsa_init, - alsa_activate, - alsa_play, - alsa_deactivate, - alsa_beforepoll, - alsa_ready - }, + &alsa_backend, #endif - { - BACKEND_COMMAND, - FIXED_FORMAT, - command_init, - 0, /* activate */ - command_play, - 0, /* deactivate */ - command_beforepoll, - command_ready - }, - { - BACKEND_NETWORK, - FIXED_FORMAT, - network_init, - 0, /* activate */ - network_play, - 0, /* deactivate */ - network_beforepoll, - network_ready - }, - { -1, 0, 0, 0, 0, 0, 0, 0 } /* end of list */ + &command_backend, + &network_backend, + 0 }; /** @brief Return nonzero if we want to play some audio @@ -1137,7 +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; @@ -1158,7 +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; @@ -1288,6 +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 */ @@ -1299,12 +574,12 @@ int main(int argc, char **argv) { /* 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) + for(n = 0; backends[n]; ++n) + if(backends[n]->backend == config->speaker_backend) break; - if(backends[n].backend == -1) + if(!backends[n]) fatal(0, "unsupported backend %d", config->speaker_backend); - backend = &backends[n]; + backend = backends[n]; /* backend-specific initialization */ backend->init(); mainloop();