/*
* 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
* 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 process
+ *
+ * This program is responsible for transmitting a single coherent audio stream
+ * to its destination (over the network, to some sound API, to some
+ * subprocess). It receives connections from decoders via file descriptor
+ * passing from the main server and plays them in the right order.
+ *
+ * @b Encodings. For the <a href="http://www.alsa-project.org/">ALSA</a> 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, <a
+ * href="http://sox.sourceforge.net/">sox</a> 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.
+ *
+ * @b Garbage @b Collection. This program deliberately does not use the
+ * garbage collector even though it might be convenient to do so. This is for
+ * two reasons. Firstly some sound APIs use thread threads and we do not want
+ * to have to deal with potential interactions between threading and garbage
+ * collection. Secondly this process needs to be able to respond quickly and
+ * this is not compatible with the collector hanging the program even
+ * relatively briefly.
+ *
+ * @b Units. This program thinks at various times in three different units.
+ * Bytes are obvious. A sample is a single sample on a single channel. A
+ * frame is several samples on different channels at the same point in time.
+ * So (for instance) a 16-bit stereo frame is 4 bytes and consists of a pair of
+ * 2-byte samples.
+ */
#include <config.h>
#include "types.h"
#include <string.h>
#include <assert.h>
#include <sys/select.h>
+#include <sys/wait.h>
#include <time.h>
-#include <alsa/asoundlib.h>
+#include <fcntl.h>
+#include <poll.h>
+#include <sys/socket.h>
+#include <netdb.h>
+#include <gcrypt.h>
+#include <sys/uio.h>
#include "configuration.h"
#include "syscalls.h"
#include "log.h"
#include "defs.h"
#include "mem.h"
-#include "speaker.h"
+#include "speaker-protocol.h"
#include "user.h"
+#include "addr.h"
+#include "timeval.h"
+#include "rtp.h"
+#include "speaker.h"
+
+#if API_ALSA
+#include <alsa/asoundlib.h>
+#endif
-#define BUFFER_SECONDS 5 /* How many seconds of input to
- * buffer. */
-
-#define FRAMES 4096 /* Frame batch size */
-
-#define NFDS 256 /* Max FDs to poll for */
-
-/* Known tracks are kept in a linked list. We don't normally to have
- * more than two - maybe three at the outside. */
-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 Linked list of all prepared tracks */
+struct track *tracks;
+
+/** @brief Playing track, or NULL */
+struct track *playing;
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 int forceplay; /* frames to force play */
+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 Frames to force-play
+ *
+ * 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.
+ */
+static int forceplay;
+
+/** @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().
+ * @ref ready is set to 0 at the same time.
+ */
+static int idled; /* set when idled */
+
+/** @brief Error counter */
+static int audio_errors;
+
+/** @brief Selected backend */
+static const struct speaker_backend *backend;
static const struct option options[] = {
{ "help", no_argument, 0, 'h' },
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;
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;
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);
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)
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 -<n> etc; both are
+ * deployed! */
+ switch(config->sox_generation) {
+ case 0:
+ if(ao->bits != 8
+ && ao->byte_format != AO_FMT_NATIVE
+ && ao->byte_format != MACHINE_AO_FMT) {
+ *(*pp)++ = "-x";
+ }
+ switch(ao->bits) {
+ case 8: *(*pp)++ = "-b"; break;
+ case 16: *(*pp)++ = "-w"; break;
+ case 32: *(*pp)++ = "-l"; break;
+ case 64: *(*pp)++ = "-d"; break;
+ default: fatal(0, "cannot handle sample size %d", (int)ao->bits);
+ }
+ break;
+ case 1:
+ switch(ao->byte_format) {
+ case AO_FMT_NATIVE: break;
+ case AO_FMT_BIG: *(*pp)++ = "-B"; break;
+ case AO_FMT_LITTLE: *(*pp)++ = "-L"; break;
+ }
+ *(*pp)++ = *qq; n = sprintf(*qq, "-%d", ao->bits/8); *qq += n + 1;
+ break;
+ }
+}
+
+/** @brief Enable format translation
+ *
+ * If necessary, replaces a tracks inbound file descriptor with one connected
+ * to a sox invocation, which performs the required translation.
+ */
+static void enable_translation(struct track *t) {
+ if((backend->flags & FIXED_FORMAT)
+ && !formats_equal(&t->format, &config->sample_format)) {
+ char argbuf[1024], *q = argbuf;
+ const char *av[18], **pp = av;
+ int soxpipe[2];
+ pid_t soxkid;
+
+ *pp++ = "sox";
+ soxargs(&pp, &q, &t->format);
+ *pp++ = "-";
+ soxargs(&pp, &q, &config->sample_format);
+ *pp++ = "-";
+ *pp++ = 0;
+ if(debugging) {
+ for(pp = av; *pp; pp++)
+ D(("sox arg[%d] = %s", pp - av, *pp));
+ D(("end args"));
+ }
+ xpipe(soxpipe);
+ soxkid = xfork();
+ if(soxkid == 0) {
+ signal(SIGPIPE, SIG_DFL);
+ xdup2(t->fd, 0);
+ xdup2(soxpipe[1], 1);
+ fcntl(0, F_SETFL, fcntl(0, F_GETFL) & ~O_NONBLOCK);
+ close(soxpipe[0]);
+ close(soxpipe[1]);
+ close(t->fd);
+ execvp("sox", (char **)av);
+ _exit(1);
+ }
+ D(("forking sox for format conversion (kid = %d)", soxkid));
+ close(t->fd);
+ close(soxpipe[1]);
+ t->fd = soxpipe[0];
+ t->format = config->sample_format;
+ }
+}
+
+/** @brief Read data into a sample buffer
+ * @param t Pointer to track
+ * @return 0 on success, -1 on EOF
+ *
+ * This is effectively the read callback on @c t->fd. It is called from the
+ * main loop whenever the track's file descriptor is readable, assuming the
+ * buffer has not reached the maximum allowed occupancy.
+ */
static int fill(struct track *t) {
size_t where, left;
int n;
/* 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);
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
+ *
+ * This is called to deactivate the output device when pausing, and also by the
+ * ALSA backend when changing encoding (in which case the sound device will be
+ * immediately reactivated).
+ */
static void idle(void) {
- int err;
-
D(("idle"));
- if(pcm) {
- 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"));
- }
+ if(backend->deactivate)
+ backend->deactivate();
+ idled = 1;
+ ready = 0;
}
-/* Abandon the current track */
+/** @brief Abandon the current track */
static void abandon(void) {
struct speaker_message sm;
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;
}
+ return backend->activate();
+}
+
+/** @brief Check whether the current track has finished
+ *
+ * The current track is determined to have finished either if the input stream
+ * eded before the format could be determined (i.e. it is malformed) or the
+ * input is at end of file and there is less than a frame left unplayed. (So
+ * it copes with decoders that crash mid-frame.)
+ */
+static void maybe_finished(void) {
+ if(playing
+ && playing->eof
+ && (!playing->got_format
+ || playing->used < bytes_per_frame(&playing->format)))
+ 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 */
+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() */
+ 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 */
+ avail_bytes = playing->size - playing->start;
+ else
+ /* The ring buffer is not wrapped, can play the lot */
+ avail_bytes = playing->used;
+ avail_frames = avail_bytes / bpf;
+ /* Only play up to the requested amount */
+ if(avail_frames > frames)
+ avail_frames = frames;
+ if(!avail_frames)
+ return;
+ /* Play it, Sam */
+ written_frames = backend->play(avail_frames);
+ written_bytes = written_frames * bpf;
+ /* written_bytes and written_frames had better both be set and correct by
+ * this point */
+ playing->start += written_bytes;
+ playing->used -= written_bytes;
+ 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)
+ playing->start = 0;
+ frames -= written_frames;
+}
+
+/* Notify the server what we're up to. */
+static void report(void) {
+ struct speaker_message sm;
+
+ if(playing && playing->buffer != (void *)&playing->format) {
+ memset(&sm, 0, sizeof sm);
+ sm.type = paused ? SM_PAUSED : SM_PLAYING;
+ strcpy(sm.id, playing->id);
+ sm.data = playing->played / playing->format.rate;
+ speaker_send(1, &sm, 0);
+ }
+ 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;
+ fds[fdno].events = events;
+ return fdno++;
+ } else
+ return -1;
+}
+
+#if API_ALSA
+/** @brief ALSA backend initialization */
+static void alsa_init(void) {
+ info("selected ALSA backend");
+}
+
+/** @brief ALSA backend activation */
+static int alsa_activate(void) {
/* If we need to change format then close the current device. */
if(pcm && !formats_equal(&playing->format, &pcm_format))
- idle();
+ 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,
playing->format.channels, err);
goto fatal;
}
- pcm_bufsize = 3 * FRAMES;
+ 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)
+ 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"));
pcm_format = playing->format;
bpf = bytes_per_frame(&pcm_format);
D(("acquired audio device"));
+ log_params(hwparams, swparams);
+ ready = 1;
}
return 0;
fatal:
return -1;
}
-/* Check to see whether the current track has finished playing */
-static void maybe_finished(void) {
- if(playing
- && playing->eof
- && (!playing->got_format
- || playing->used < bytes_per_frame(&playing->format)))
- abandon();
-}
-
-static void play(size_t frames) {
- snd_pcm_sframes_t written_frames;
- size_t avail_bytes, avail_frames, written_bytes;
+/** @brief Play via ALSA */
+static size_t alsa_play(size_t frames) {
+ snd_pcm_sframes_t pcm_written_frames;
int err;
-
- if(activate()) {
- if(playing)
- forceplay = frames;
- else
- forceplay = 0; /* Must have called abandon() */
- 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)
- 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);
+
+ pcm_written_frames = snd_pcm_writei(pcm,
+ playing->buffer + playing->start,
+ frames);
D(("actually play %zu frames, wrote %d",
- avail_frames, (int)written_frames));
- if(written_frames < 0) {
- switch(written_frames) {
+ 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;
+ return 0;
case -EAGAIN:
- return;
+ return 0;
default:
- fatal(0, "error calling snd_pcm_writei: %d", (int)written_frames);
+ fatal(0, "error calling snd_pcm_writei: %d",
+ (int)pcm_written_frames);
}
- }
- written_bytes = written_frames * bpf;
- playing->start += written_bytes;
- playing->used -= written_bytes;
- 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)
- playing->start = 0;
- frames -= written_frames;
+ } else
+ return pcm_written_frames;
}
-/* Notify the server what we're up to. */
-static void report(void) {
- struct speaker_message sm;
+static int alsa_slots, alsa_nslots = -1;
- if(playing && playing->buffer != (void *)&playing->format) {
- memset(&sm, 0, sizeof sm);
- sm.type = paused ? SM_PAUSED : SM_PLAYING;
- strcpy(sm.id, playing->id);
- sm.data = playing->played / playing->format.rate;
- speaker_send(1, &sm, 0);
+/** @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_afterpoll(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
+ return 1;
+}
+
+/** @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"));
}
- time(&last_report);
}
+#endif
-static int addfd(int fd, int events) {
- if(fdno < NFDS) {
- fds[fdno].fd = fd;
- fds[fdno].events = events;
- return fdno++;
+/** @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_afterpoll(void) {
+ if(cmdfd_slot != -1) {
+ if(fds[cmdfd_slot].revents & (POLLOUT | POLLERR))
+ play(3 * FRAMES);
+ return 0;
} else
return -1;
}
-int main(int argc, char **argv) {
- int n, fd, stdin_slot, alsa_slots, alsa_nslots = -1, err;
- unsigned short alsa_revents;
- struct track *t;
- struct speaker_message sm;
+/** @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;
+}
- 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) {
- case 'h': help();
- case 'V': version();
- case 'c': configfile = optarg; break;
- case 'd': debugging = 1; break;
- case 'D': debugging = 0; break;
- default: fatal(0, "invalid option");
+/** @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.
+ */
+ 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);
+ }
}
}
- if(getenv("DISORDER_DEBUG_SPEAKER")) debugging = 1;
- /* If stderr is a TTY then log there, otherwise to syslog. */
- if(!isatty(2)) {
- openlog(progname, LOG_PID, LOG_DAEMON);
- log_default = &log_syslog;
+ 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;
}
- if(config_read()) fatal(0, "cannot read configuration");
- /* ignore SIGPIPE */
- signal(SIGPIPE, SIG_IGN);
- /* 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");
+ /* "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_afterpoll(void) {
+ if(bfd_slot != -1) {
+ if(fds[bfd_slot].revents & (POLLOUT | POLLERR))
+ play(3 * FRAMES);
+ return 0;
+ } else
+ return 1;
+}
+
+/** @brief Table of speaker backends */
+static const struct speaker_backend backends[] = {
+#if API_ALSA
+ {
+ BACKEND_ALSA,
+ 0,
+ alsa_init,
+ alsa_activate,
+ alsa_play,
+ alsa_deactivate,
+ alsa_beforepoll,
+ alsa_afterpoll
+ },
+#endif
+ {
+ BACKEND_COMMAND,
+ FIXED_FORMAT,
+ command_init,
+ generic_activate,
+ command_play,
+ 0, /* deactivate */
+ command_beforepoll,
+ command_afterpoll
+ },
+ {
+ BACKEND_NETWORK,
+ FIXED_FORMAT,
+ network_init,
+ generic_activate,
+ network_play,
+ 0, /* deactivate */
+ network_beforepoll,
+ network_afterpoll
+ },
+ { -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.
+ *
+ * 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.
+ */
+static void mainloop(void) {
+ struct track *t;
+ struct speaker_message sm;
+ int n, fd, stdin_slot, poke, timeout;
+
while(getppid() != 1) {
fdno = 0;
/* Always ready for commands from the main server. */
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;
+#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 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 = backend->afterpoll();
+ if(poke) {
/* Some attempt to play must have failed */
if(playing && !paused)
play(forceplay);
t = findtrack(sm.id, 1);
if(fd != -1) acquire(t, fd);
playing = t;
- play(pcm_bufsize);
+ play(bufsize);
report();
break;
case SM_PAUSE:
if(paused) {
paused = 0;
if(playing)
- play(pcm_bufsize);
+ play(bufsize);
}
report();
break;
}
/* 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)
report();
}
+}
+
+int main(int argc, char **argv) {
+ int n;
+
+ set_progname(argv);
+ if(!setlocale(LC_CTYPE, "")) fatal(errno, "error calling setlocale");
+ while((n = getopt_long(argc, argv, "hVc:dD", options, 0)) >= 0) {
+ switch(n) {
+ case 'h': help();
+ case 'V': version();
+ case 'c': configfile = optarg; break;
+ case 'd': debugging = 1; break;
+ case 'D': debugging = 0; break;
+ default: fatal(0, "invalid option");
+ }
+ }
+ if(getenv("DISORDER_DEBUG_SPEAKER")) debugging = 1;
+ /* If stderr is a TTY then log there, otherwise to syslog. */
+ if(!isatty(2)) {
+ openlog(progname, LOG_PID, LOG_DAEMON);
+ log_default = &log_syslog;
+ }
+ if(config_read()) fatal(0, "cannot read configuration");
+ /* ignore SIGPIPE */
+ signal(SIGPIPE, SIG_IGN);
+ /* reap kids */
+ signal(SIGCHLD, reap);
+ /* set nice value */
+ xnice(config->nice_speaker);
+ /* change user */
+ become_mortal();
+ /* make sure we're not root, whatever the config says */
+ if(getuid() == 0 || geteuid() == 0) fatal(0, "do not run as root");
+ /* identify the backend used to play */
+ for(n = 0; backends[n].backend != -1; ++n)
+ if(backends[n].backend == config->speaker_backend)
+ break;
+ if(backends[n].backend == -1)
+ fatal(0, "unsupported backend %d", config->speaker_backend);
+ backend = &backends[n];
+ /* backend-specific initialization */
+ backend->init();
+ mainloop();
info("stopped (parent terminated)");
exit(0);
}
indent-tabs-mode:nil
End:
*/
-/* arch-tag:HQ4ayCGCjeBF97RuRnvcyg */
~mdw / disorder / blobdiff