X-Git-Url: http://www.chiark.greenend.org.uk/ucgi/~mdw/git/disorder/blobdiff_plain/763d5e6ad88ef3ba1cd1d7742d060e4f1e54c6b8..ae5b28b95df354760fc05130db1d816691cc58ad:/server/speaker.c diff --git a/server/speaker.c b/server/speaker.c index 6296169..3aef0b0 100644 --- a/server/speaker.c +++ b/server/speaker.c @@ -1,6 +1,6 @@ /* * This file is part of DisOrder - * Copyright (C) 2005, 2006 Richard Kettlewell + * Copyright (C) 2005, 2006, 2007 Richard Kettlewell * * 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 @@ -17,14 +17,35 @@ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 * USA */ - -/* 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 libao - * drivers are implemented using 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. */ +/** @file server/speaker.c + * @brief Speaker processs + * + * 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. + * + * 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. + * + * 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. + * + * 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. + */ #include #include "types.h" @@ -40,8 +61,14 @@ #include #include #include +#include #include -#include +#include +#include +#include +#include +#include +#include #include "configuration.h" #include "syscalls.h" @@ -50,16 +77,46 @@ #include "mem.h" #include "speaker.h" #include "user.h" +#include "addr.h" +#include "timeval.h" +#include "rtp.h" -#define BUFFER_SECONDS 5 /* How many seconds of input to - * buffer. */ +#if API_ALSA +#include +#endif + +#ifdef WORDS_BIGENDIAN +# define MACHINE_AO_FMT AO_FMT_BIG +#else +# define MACHINE_AO_FMT AO_FMT_LITTLE +#endif + +/** @brief How many seconds of input to buffer + * + * While any given connection has this much audio buffered, no more reads will + * be issued for that connection. The decoder will have to wait. + */ +#define BUFFER_SECONDS 5 #define FRAMES 4096 /* Frame batch size */ -#define NFDS 256 /* Max FDs to poll for */ +/** @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 Maximum number of FDs to poll for */ +#define NFDS 256 -/* Known tracks are kept in a linked list. We don't normally to have - * more than two - maybe three at the outside. */ +/** @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 */ @@ -76,13 +133,44 @@ static struct track { static time_t last_report; /* when we last reported */ static int paused; /* pause status */ -static snd_pcm_t *pcm; /* current pcm handle */ static ao_sample_format pcm_format; /* current format if aodev != 0 */ static size_t bpf; /* bytes per frame */ static struct pollfd fds[NFDS]; /* if we need more than that */ static int fdno; /* fd number */ -static snd_pcm_uframes_t pcm_bufsize; /* buffer size */ +static size_t bufsize; /* buffer size */ +#if API_ALSA +static snd_pcm_t *pcm; /* current pcm handle */ +static snd_pcm_uframes_t last_pcm_bufsize; /* last seen buffer size */ +#endif +static int ready; /* ready to send audio */ 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). + * + * 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; + +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 */ static const struct option options[] = { { "help", no_argument, 0, 'h' }, @@ -116,12 +204,12 @@ static void version(void) { exit(0); } -/* Return the number of bytes per frame in FORMAT. */ +/** @brief Return the number of bytes per frame in @p format */ static size_t bytes_per_frame(const ao_sample_format *format) { return format->channels * format->bits / 8; } -/* Find track ID, maybe creating it if not found. */ +/** @brief Find track @p id, maybe creating it if not found */ static struct track *findtrack(const char *id, int create) { struct track *t; @@ -141,7 +229,7 @@ static struct track *findtrack(const char *id, int create) { return t; } -/* Remove track ID (but do not destroy it). */ +/** @brief Remove track @p id (but do not destroy it) */ static struct track *removetrack(const char *id) { struct track *t, **tt; @@ -153,7 +241,7 @@ static struct track *removetrack(const char *id) { return t; } -/* Destroy a track. */ +/** @brief Destroy a track */ static void destroy(struct track *t) { D(("destroy %s", t->id)); if(t->fd != -1) xclose(t->fd); @@ -161,7 +249,7 @@ static void destroy(struct track *t) { free(t); } -/* Notice a new FD. */ +/** @brief Notice a new connection */ static void acquire(struct track *t, int fd) { D(("acquire %s %d", t->id, fd)); if(t->fd != -1) @@ -170,7 +258,111 @@ static void acquire(struct track *t, int fd) { nonblock(fd); } -/* Read data into a sample buffer. Return 0 on success, -1 on EOF. */ +/** @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) { + switch(config->speaker_backend) { + case BACKEND_COMMAND: + case BACKEND_NETWORK: + /* These backends need a specific sample format */ + break; + case BACKEND_ALSA: + /* ALSA can cope */ + return; + } + if(!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; + ready = 0; + } +} + +/** @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. + */ static int fill(struct track *t) { size_t where, left; int n; @@ -206,6 +398,8 @@ static int fill(struct track *t) { /* 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); @@ -217,21 +411,13 @@ static int fill(struct track *t) { return 0; } -/* 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); -} - -/* Close the sound device. */ +/** @brief Close the sound device */ static void idle(void) { - int err; - D(("idle")); - if(pcm) { +#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")); @@ -242,9 +428,12 @@ static void idle(void) { forceplay = 0; D(("released audio device")); } +#endif + idled = 1; + ready = 0; } -/* Abandon the current track */ +/** @brief Abandon the current track */ static void abandon(void) { struct speaker_message sm; @@ -259,109 +448,162 @@ static void abandon(void) { forceplay = 0; } -/* Make sure the sound device is open and has the right sample format. Return - * 0 on success and -1 on error. */ -static int activate(void) { - int err; - snd_pcm_hw_params_t *hwparams; - snd_pcm_sw_params_t *swparams; - int sample_format = 0; - unsigned rate; +#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; } - /* If we need to change format then close the current device. */ - if(pcm && !formats_equal(&playing->format, &pcm_format)) - idle(); - if(!pcm) { - 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; + switch(config->speaker_backend) { + case BACKEND_COMMAND: + case BACKEND_NETWORK: + if(!ready) { + pcm_format = config->sample_format; + bufsize = 3 * FRAMES; + bpf = bytes_per_frame(&config->sample_format); + D(("acquired audio device")); + ready = 1; } - 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); + return 0; + case BACKEND_ALSA: +#if API_ALSA + /* 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; } - 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((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; } - 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; + 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; } - 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) - info("asked for PCM buffer of %d frames, got %d", - 3 * FRAMES, (int)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")); - } - 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 + default: + assert(!"reached"); } - return -1; } /* Check to see whether the current track has finished playing */ @@ -373,10 +615,30 @@ static void maybe_finished(void) { abandon(); } +static void fork_cmd(void) { + pid_t cmdpid; + int pfd[2]; + if(cmdfd != -1) close(cmdfd); + xpipe(pfd); + cmdpid = xfork(); + if(!cmdpid) { + signal(SIGPIPE, SIG_DFL); + xdup2(pfd[0], 0); + close(pfd[0]); + close(pfd[1]); + execl("/bin/sh", "sh", "-c", config->speaker_command, (char *)0); + fatal(errno, "error execing /bin/sh"); + } + close(pfd[0]); + cmdfd = pfd[1]; + D(("forked cmd %d, fd = %d", cmdpid, cmdfd)); +} + static void play(size_t frames) { - snd_pcm_sframes_t written_frames; - size_t avail_bytes, avail_frames, written_bytes; - int err; + size_t avail_bytes, write_bytes, written_frames; + ssize_t written_bytes; + struct rtp_header header; + struct iovec vec[2]; if(activate()) { if(playing) @@ -403,30 +665,141 @@ static void play(size_t frames) { avail_bytes = playing->size - playing->start; else avail_bytes = playing->used; - avail_frames = avail_bytes / bpf; - if(avail_frames > frames) - avail_frames = frames; - if(!avail_frames) - return; - written_frames = snd_pcm_writei(pcm, - playing->buffer + playing->start, - avail_frames); - D(("actually play %zu frames, wrote %d", - avail_frames, (int)written_frames)); - if(written_frames < 0) { - switch(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: + + 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; - default: - fatal(0, "error calling snd_pcm_writei: %d", (int)written_frames); + 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's 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 */ + if(target_rtp_time > rtp_time) + info("advancing rtp_time by %"PRIu64" samples", + target_rtp_time - rtp_time); + else if(target_rtp_time < rtp_time) + info("reversing rtp_time by %"PRIu64" samples", + rtp_time - target_rtp_time); + 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"); } - written_bytes = written_frames * bpf; + /* written_bytes and written_frames had better both be set and correct by + * this point */ playing->start += written_bytes; playing->used -= written_bytes; playing->played += written_frames; @@ -451,6 +824,16 @@ static void report(void) { time(&last_report); } +static void reap(int __attribute__((unused)) sig) { + pid_t cmdpid; + int st; + + do + cmdpid = waitpid(-1, &st, WNOHANG); + while(cmdpid > 0); + signal(SIGCHLD, reap); +} + static int addfd(int fd, int events) { if(fdno < NFDS) { fds[fdno].fd = fd; @@ -461,13 +844,39 @@ static int addfd(int fd, int events) { } int main(int argc, char **argv) { - int n, fd, stdin_slot, alsa_slots, alsa_nslots = -1, err; - unsigned short alsa_revents; + int n, fd, stdin_slot, alsa_slots, cmdfd_slot, bfd_slot, poke, timeout; struct track *t; struct speaker_message sm; + 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; +#if API_ALSA + int alsa_nslots = -1, err; +#endif set_progname(argv); - mem_init(0); if(!setlocale(LC_CTYPE, "")) fatal(errno, "error calling setlocale"); while((n = getopt_long(argc, argv, "hVc:dD", options, 0)) >= 0) { switch(n) { @@ -488,13 +897,62 @@ int main(int argc, char **argv) { 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"); - info("started"); + switch(config->speaker_backend) { + case BACKEND_ALSA: + info("selected ALSA backend"); + case BACKEND_COMMAND: + info("selected command backend"); + fork_cmd(); + break; + case BACKEND_NETWORK: + res = get_address(&config->broadcast, &pref, &sockname); + if(!res) return -1; + if(config->broadcast_from.n) { + sres = get_address(&config->broadcast_from, &prefbind, &ssockname); + if(!sres) return -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(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); + /* 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; + } + break; + default: + fatal(0, "unknown backend %d", config->speaker_backend); + } while(getppid() != 1) { fdno = 0; /* Always ready for commands from the main server. */ @@ -507,37 +965,135 @@ int main(int argc, char **argv) { playing->slot = -1; /* If forceplay is set then wait until it succeeds before waiting on the * sound device. */ - if(pcm && !forceplay) { - alsa_slots = fdno; - alsa_nslots = snd_pcm_poll_descriptors(pcm, &fds[fdno], NFDS - fdno); - fdno += alsa_nslots; - } else - alsa_slots = -1; + 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 any other tracks don't have a full buffer, try to read sample data * from them. */ for(t = tracks; t; t = t->next) if(t != playing) { if(!t->eof && t->used < t->size) { - t->slot = addfd(t->fd, POLLIN); + t->slot = addfd(t->fd, POLLIN | POLLHUP); } else t->slot = -1; } - /* Wait up to a second before thinking about current state */ - n = poll(fds, fdno, 1000); + /* Wait for something interesting to happen */ + n = poll(fds, fdno, timeout); if(n < 0) { if(errno == EINTR) continue; fatal(errno, "error calling poll"); } /* Play some sound before doing anything else */ - if(alsa_slots != -1) { - 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) - play(3 * FRAMES); - } 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(poke) { /* Some attempt to play must have failed */ if(playing && !paused) play(forceplay); @@ -561,7 +1117,7 @@ int main(int argc, char **argv) { t = findtrack(sm.id, 1); if(fd != -1) acquire(t, fd); playing = t; - play(pcm_bufsize); + play(bufsize); report(); break; case SM_PAUSE: @@ -574,7 +1130,7 @@ int main(int argc, char **argv) { if(paused) { paused = 0; if(playing) - play(pcm_bufsize); + play(bufsize); } report(); break; @@ -604,16 +1160,16 @@ 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)) + if(t->slot != -1 && (fds[t->slot].revents & (POLLIN | POLLHUP))) fill(t); /* We might be able to play now */ - if(pcm && forceplay && playing && !paused) + 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) && pcm) + if((!playing || paused) && ready) idle(); /* If we've not reported out state for a second do so now. */ if(time(0) > last_report)