X-Git-Url: https://git.distorted.org.uk/~mdw/disorder/blobdiff_plain/ea410ba1d3b06ba6f60305c7d31369da134906dd..5a7c42a822d0d3c873e6d3bc3f06e31fbdde43f4:/server/speaker.c diff --git a/server/speaker.c b/server/speaker.c index 2114db4..4bb1d2c 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 @@ -87,96 +87,39 @@ #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 - -/** @brief Frame batch size - * - * This controls how many frames are written in one go. - * - * For ALSA we request a buffer of three times this size and set the low - * watermark to this amount. The goal is then to keep between 1 and 3 times - * this many frames in play. - * - * For all backends we attempt to play up to three times this many frames per - * shot. In practice we will often only send much less than this. - */ -#define FRAMES 4096 - -/** @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 Maximum RTP playahead (ms) */ -#define RTP_AHEAD_MS 1000 +/** @brief Linked list of all prepared tracks */ +struct track *tracks; -/** @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 - * - * 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; -/** @brief Frames to force-play +/** @brief The current device sample format * - * If this is nonzero, and playing is enabled, then the main loop will attempt - * to play this many frames without checking whether the output device is - * ready. + * 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 forceplay; +ao_sample_format device_format; /** @brief Pipe to subprocess * @@ -219,84 +162,12 @@ static uint32_t rtp_id; /** @brief Set when idled * * This is set when the sound device is deliberately closed by idle(). - * @ref ready is set to 0 at the same time. */ static int idled; /* set when idled */ /** @brief Error counter */ static int audio_errors; -/** @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); - - /** @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 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 Called after poll() - * @return 0 if we could play, non-0 if not - * - * Called after the call to poll(). Should arrange to play some audio if the - * output device is ready. - * - * The return value should be 0 if the device was ready to play, or nonzero - * if it was not. - */ - int (*afterpoll)(void); -}; - /** @brief Selected backend */ static const struct speaker_backend *backend; @@ -540,10 +411,11 @@ static int fill(struct track *t) { */ 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 */ @@ -558,49 +430,30 @@ 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); } /** @brief Check whether the current track has finished @@ -618,52 +471,38 @@ static void maybe_finished(void) { abandon(); } -/** @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 Play up to @p frames frames of audio */ +/** @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 */ @@ -690,6 +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. */ @@ -731,11 +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; @@ -796,8 +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", @@ -817,13 +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: @@ -831,8 +713,9 @@ error: if(pcm) { snd_pcm_close(pcm); pcm = 0; + device_state = device_error; } - return -1; + return; } /** @brief Play via ALSA */ @@ -889,41 +772,42 @@ static void alsa_beforepoll(void) { } /** @brief Process poll() results for ALSA */ -static int alsa_afterpoll(void) { +static int alsa_ready(void) { int err; - 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); - return 0; - } else + 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 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 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)); } -#endif /** @brief Command backend initialization */ static void command_init(void) { @@ -965,24 +849,11 @@ static void command_beforepoll(void) { } /** @brief Process poll() results for subprocess play */ -static int command_afterpoll(void) { - if(cmdfd_slot != -1) { - if(fds[cmdfd_slot].revents & (POLLOUT | POLLERR)) - play(3 * FRAMES); +static int command_ready(void) { + if(fds[cmdfd_slot].revents & (POLLOUT | POLLERR)) + return 1; + else return 0; - } else - return -1; -} - -/** @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 Network backend initialization */ @@ -1194,13 +1065,11 @@ static void network_beforepoll(void) { } /** @brief Process poll() results for network play */ -static int network_afterpoll(void) { - if(bfd_slot != -1) { - if(fds[bfd_slot].revents & (POLLOUT | POLLERR)) - play(3 * FRAMES); - return 0; - } else +static int network_ready(void) { + if(fds[bfd_slot].revents & (POLLOUT | POLLERR)) return 1; + else + return 0; } /** @brief Table of speaker backends */ @@ -1214,105 +1083,78 @@ static const struct speaker_backend backends[] = { alsa_play, alsa_deactivate, alsa_beforepoll, - alsa_afterpoll + alsa_ready }, #endif { BACKEND_COMMAND, FIXED_FORMAT, command_init, - generic_activate, + 0, /* activate */ command_play, 0, /* deactivate */ command_beforepoll, - command_afterpoll + command_ready }, { BACKEND_NETWORK, FIXED_FORMAT, network_init, - generic_activate, + 0, /* activate */ network_play, 0, /* deactivate */ network_beforepoll, - network_afterpoll + network_ready }, { -1, 0, 0, 0, 0, 0, 0, 0 } /* end of list */ }; -/** @brief Main event loop - * - * This has grown in a rather bizarre and ad-hoc way is very sensitive to - * changes... - * - * Firstly the loop is terminated when the parent process exits. Therefore the - * speaker process has the same lifetime as the main server. This and the - * reading of data from decoders is comprehensible enough. - * - * The playing of audio is more complicated however. - * - * On the first run through when a track is ready to be played, @ref ready and - * @ref forceplay will both be zero. Therefore @c beforepoll is not called. - * - * @c afterpoll on the other hand @b is called and will return nonzero. The - * result is that we call @c play(0). This will call activate(), setting - * @ref ready nonzero, but otherwise has no immediate effect. - * - * We then deal with stdin and the decoders. - * - * We then reach the second place we might play some audio. @ref forceplay is - * 0 so nothing happens here again. - * - * On the next iteration through however @ref ready is nonzero, and @ref - * forceplay is 0, so we call @c beforepoll. After the @c poll() we call @c - * afterpoll and actually get some audio played. - * - * This is surely @b far more complicated than it needs to be! - * - * If at any call to play(), activate() fails, or if there aren't enough bytes - * in the buffer to satisfy the request, then @ref forceplay is set non-0. On - * the next pass through the event loop @c beforepoll is not called. This - * means that (if none of the other FDs trigger) the @c poll() call will block - * for up to a second. @c afterpoll will return nonzero, since @c beforepoll - * wasn't called, and consequently play() is called with @ref forceplay as its - * argument. +/** @brief Return nonzero if we want to play some audio * - * The effect is to attempt to restart playing audio - including the activate() - * step, which may have failed at the previous attempt - at least once a second - * after an error has disabled it. The delay prevents busy-waiting on whatever - * condition has rendered the audio device uncooperative. + * 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; - int n, fd, stdin_slot, poke, timeout; + int n, fd, stdin_slot, timeout; 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. */ -#if API_ALSA - alsa_slots = -1; -#endif - 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) { @@ -1327,16 +1169,26 @@ static void mainloop(void) { fatal(errno, "error calling poll"); } /* Play some sound before doing anything else */ - poke = backend->afterpoll(); - if(poke) { - /* Some attempt to play must have failed */ - if(playing && !paused) - play(forceplay); - else - forceplay = 0; /* just in case */ + if(playable()) { + /* We want to play some audio */ + if(device_state == device_open) { + if(backend->ready()) + 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. 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) { @@ -1352,7 +1204,10 @@ static void mainloop(void) { 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: @@ -1364,8 +1219,9 @@ static void mainloop(void) { 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; @@ -1397,14 +1253,11 @@ static void mainloop(void) { 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)