#include <string.h>
#include <assert.h>
#include <sys/select.h>
+#include <sys/wait.h>
#include <time.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 "mem.h"
#include "speaker.h"
#include "user.h"
+#include "addr.h"
+#include "timeval.h"
+#include "rtp.h"
-#if BUILD_SPEAKER
+#if API_ALSA
#include <alsa/asoundlib.h>
+#endif
+
+#ifdef WORDS_BIGENDIAN
+# define MACHINE_AO_FMT AO_FMT_BIG
+#else
+# define MACHINE_AO_FMT AO_FMT_LITTLE
+#endif
#define BUFFER_SECONDS 5 /* How many seconds of input to
* buffer. */
#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) */
+
+#define RTP_AHEAD 2 /* Max RTP playahead (seconds) */
+
#define NFDS 256 /* Max FDs to poll for */
/* Known tracks are kept in a linked list. We don't normally to have
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 */
+static uint32_t rtp_time; /* RTP timestamp */
+static struct timeval rtp_time_real; /* corresponding real time */
+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' },
/* 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"));
forceplay = 0;
D(("released audio device"));
}
+#endif
+ idled = 1;
+ ready = 0;
}
/* Abandon the current track */
forceplay = 0;
}
+#if API_ALSA
+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
+
+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. */
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 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;
- }
- 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;
+ 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"));
}
- break;
- default:
- error(0, "unsupported sample size %d", playing->format.bits);
- goto fatal;
+ 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((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;
+ if(!ready) {
+ pcm_format = config->sample_format;
+ bufsize = 3 * FRAMES;
+ bpf = bytes_per_frame(&config->sample_format);
+ D(("acquired audio device"));
+ ready = 1;
}
- 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;
+ 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;
+ }
+ 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 */
abandon();
}
+static void fork_cmd(void) {
+ pid_t cmdpid;
+ int pfd[2];
+ if(cmdfd != -1) close(cmdfd);
+ xpipe(pfd);
+ cmdpid = xfork();
+ if(!cmdpid) {
+ 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, written_frames;
+ ssize_t written_bytes;
+ struct rtp header;
+ struct iovec vec[2];
if(activate()) {
if(playing)
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);
+
+ 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;
- case -EAGAIN:
+ 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(rtp_time_real.tv_sec == 0)
+ xgettimeofday(&rtp_time_real, 0);
+ if(idled) {
+ 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);
+ }
+ header.vpxcc = 2 << 6; /* V=2, P=0, X=0, CC=0 */
+ header.seq = htons(rtp_seq++);
+ header.timestamp = htonl(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;
+ 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.)"
+ */
+ 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;
- default:
- fatal(0, "error calling snd_pcm_writei: %d", (int)written_frames);
}
+ 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;
+ /* Advance the corresponding real time */
+ assert(NETWORK_BYTES <= 2000); /* else risk overflowing 32 bits */
+ rtp_time_real.tv_usec += written_frames * 1000000 / playing->format.rate;
+ if(rtp_time_real.tv_usec >= 1000000) {
+ ++rtp_time_real.tv_sec;
+ rtp_time_real.tv_usec -= 1000000;
+ }
+ 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;
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;
}
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 timeval now, delta;
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;
+ 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) {
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 settting SO_BROADCAST on broadcast socket");
+ /* 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. */
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:
+ /* We want to keep the notional playing point somewhere in the near
+ * future. If it's too near then clients that attempt even the
+ * slightest amount of read-ahead will never catch up, and those that
+ * don't will skip whenever there's a trivial network delay. If it's
+ * too far ahead then pause latency will be too high.
+ */
+ xgettimeofday(&now, 0);
+ delta = tvsub(rtp_time_real, now);
+ if(delta.tv_sec < RTP_AHEAD) {
+ D(("delta = %ld.%06ld", (long)delta.tv_sec, (long)delta.tv_usec));
+ bfd_slot = addfd(bfd, POLLOUT);
+ if(delta.tv_sec < 0)
+ rtp_time_real = now; /* catch up */
+ }
+ 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);
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)
info("stopped (parent terminated)");
exit(0);
}
-#else
-int main(int attribute((unused)) argc, char **argv) {
- set_progname(argv);
- mem_init(0);
- fatal(0, "disorder-speaker not supported on this platform");
-}
-#endif
/*
Local Variables:
~mdw / disorder / blobdiff