* 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 <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.
+ *
+ * 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 <config.h>
#include "types.h"
# define MACHINE_AO_FMT AO_FMT_LITTLE
#endif
-#define BUFFER_SECONDS 5 /* How many seconds of input to
- * buffer. */
+/** @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 NETWORK_BYTES 1024 /* Bytes to send per network packet */
-/* (don't make this too big or arithmetic will start to overflow) */
+/** @brief Bytes to send per network packet
+ *
+ * Don't make this too big or arithmetic will start to overflow.
+ */
+#define NETWORK_BYTES 1024
-#define RTP_AHEAD 2 /* Max RTP playahead (seconds) */
+/** @brief Maximum RTP playahead (seconds) */
+#define RTP_AHEAD 2
-#define NFDS 256 /* Max FDs to poll for */
+/** @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 */
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) {
+ 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;
/* 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 */
static void idle(void) {
D(("idle"));
#if API_ALSA
ready = 0;
}
-/* Abandon the current track */
+/** @brief Abandon the current track */
static void abandon(void) {
struct speaker_message sm;
}
#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;
}
#endif
-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;
- }
-}
-
-/* Make sure the sound device is open and has the right sample format. Return
- * 0 on success and -1 on error. */
+/** @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) {
switch(config->speaker_backend) {
case BACKEND_COMMAND:
case BACKEND_NETWORK:
- /* If we pass audio on to some other agent then we enforce the configured
- * sample format on the *inbound* audio data. */
- if(!formats_equal(&playing->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, &playing->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) {
- xdup2(playing->fd, 0);
- xdup2(soxpipe[1], 1);
- fcntl(0, F_SETFL, fcntl(0, F_GETFL) & ~O_NONBLOCK);
- close(soxpipe[0]);
- close(soxpipe[1]);
- close(playing->fd);
- execvp("sox", (char **)av);
- _exit(1);
- }
- D(("forking sox for format conversion (kid = %d)", soxkid));
- close(playing->fd);
- close(soxpipe[1]);
- playing->fd = soxpipe[0];
- playing->format = config->sample_format;
- ready = 0;
- }
if(!ready) {
pcm_format = config->sample_format;
bufsize = 3 * FRAMES;
xpipe(pfd);
cmdpid = xfork();
if(!cmdpid) {
+ signal(SIGPIPE, SIG_DFL);
xdup2(pfd[0], 0);
close(pfd[0]);
close(pfd[1]);
}
static void play(size_t frames) {
- size_t avail_bytes, written_frames;
+ size_t avail_bytes, write_bytes, written_frames;
ssize_t written_bytes;
- struct rtp header;
+ struct rtp_header header;
struct iovec vec[2];
if(activate()) {
switch(config->speaker_backend) {
#if API_ALSA
- case BACKEND_ALSA:
+ case BACKEND_ALSA: {
snd_pcm_sframes_t pcm_written_frames;
size_t avail_frames;
int err;
written_frames = pcm_written_frames;
written_bytes = written_frames * bpf;
break;
+ }
#endif
case BACKEND_COMMAND:
if(avail_bytes > frames * bpf)
struct timeval now;
xgettimeofday(&now, 0);
/* There's been a gap. Fix up the RTP time accordingly. */
- rtp_time += (((now.tv_sec + now.tv_usec /1000000.0)
- - (rtp_time_real.tv_sec + rtp_time_real.tv_usec / 1000000.0))
- * playing->format.rate * playing->format.channels);
+ const long offset = (((now.tv_sec + now.tv_usec /1000000.0)
+ - (rtp_time_real.tv_sec + rtp_time_real.tv_usec / 1000000.0))
+ * playing->format.rate * playing->format.channels);
+ if(offset >= 0) {
+ info("offset RTP timestamp by %ld", offset);
+ rtp_time += offset;
+ }
+ rtp_time_real = now;
}
header.vpxcc = 2 << 6; /* V=2, P=0, X=0, CC=0 */
header.seq = htons(rtp_seq++);
* 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 = avail_bytes;
+ write_bytes = avail_bytes;
#if 0
- {
- char buffer[3 * sizeof header + 1];
- size_t n;
- const uint8_t *ptr = (void *)&header;
-
- for(n = 0; n < sizeof header; ++n)
- sprintf(&buffer[3 * n], "%02x ", *ptr++);
- info(buffer);
- }
+ while(write_bytes > 0 && (uint32_t)(playing->buffer + playing->start + write_bytes - 4) == 0)
+ write_bytes -= 4;
#endif
- 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");
+ 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;
+#if 0
+ {
+ char buffer[3 * sizeof header + 1];
+ size_t n;
+ const uint8_t *ptr = (void *)&header;
+
+ for(n = 0; n < sizeof header; ++n)
+ sprintf(&buffer[3 * n], "%02x ", *ptr++);
+ info(buffer);
+ }
+#endif
+ 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;
++rtp_time_real.tv_sec;
rtp_time_real.tv_usec -= 1000000;
}
+ assert(rtp_time_real.tv_usec < 1000000);
break;
default:
assert(!"reached");
}
break;
#if API_ALSA
- case BACKEND_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. */
if(alsa_nslots >= 0)
fdno += alsa_nslots;
break;
+ }
#endif
default:
assert(!"unknown backend");
~mdw / disorder / blobdiff