X-Git-Url: http://www.chiark.greenend.org.uk/ucgi/~mdw/git/disorder/blobdiff_plain/dfa51bb78ea10f22e9fb02397466943befdcc2e6..a202463f02ef9fe101e14bd5ca264de34d50a405:/lib/uaudio-rtp.c

diff --git a/lib/uaudio-rtp.c b/lib/uaudio-rtp.c
index 4ea4329..e1768a8 100644
--- a/lib/uaudio-rtp.c
+++ b/lib/uaudio-rtp.c
@@ -15,16 +15,20 @@
  * You should have received a copy of the GNU General Public License
  * along with this program.  If not, see <http://www.gnu.org/licenses/>.
  */
-/** @file lib/uaudio-oss.c
+/** @file lib/uaudio-rtp.c
  * @brief Support for RTP network play backend */
 #include "common.h"
 
 #include <errno.h>
+#include <sys/socket.h>
 #include <ifaddrs.h>
 #include <net/if.h>
+#include <arpa/inet.h>
+#include <netinet/in.h>
 #include <gcrypt.h>
 #include <unistd.h>
 #include <time.h>
+#include <sys/uio.h>
 
 #include "uaudio.h"
 #include "mem.h"
@@ -34,6 +38,7 @@
 #include "addr.h"
 #include "ifreq.h"
 #include "timeval.h"
+#include "configuration.h"
 
 /** @brief Bytes to send per network packet
  *
@@ -54,42 +59,20 @@ static int rtp_fd;
 /** @brief RTP SSRC */
 static uint32_t rtp_id;
 
+/** @brief Base for timestamp */
+static uint32_t rtp_base;
+
 /** @brief RTP sequence number */
 static uint16_t rtp_sequence;
 
-/** @brief RTP timestamp
- *
- * This is the timestamp that will be used on the next outbound packet.
- *
- * 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_timestamp;
-
-/** @brief Actual time corresponding to @ref rtp_timestamp
- *
- * This is the time, on this machine, at which the sample at @ref rtp_timestamp
- * ought to be sent, interpreted as the time the last packet was sent plus the
- * time length of the packet. */
-static struct timeval rtp_timeval;
-
-/** @brief Set when we (re-)activate, to provoke timestamp resync */
-static int rtp_reactivated;
-
 /** @brief Network error count
  *
  * If too many errors occur in too short a time, we give up.
  */
 static int rtp_errors;
 
-/** @brief Delay threshold in microseconds
- *
- * rtp_play() never attempts to introduce a delay shorter than this.
- */
-static int64_t rtp_delay_threshold;
+/** @brief Set while paused */
+static volatile int rtp_paused;
 
 static const char *const rtp_options[] = {
   "rtp-destination",
@@ -98,21 +81,75 @@ static const char *const rtp_options[] = {
   "rtp-source-port",
   "multicast-ttl",
   "multicast-loop",
-  "rtp-delay-threshold",
   NULL
 };
 
-static size_t rtp_play(void *buffer, size_t nsamples) {
+static void rtp_get_netconfig(const char *af,
+                              const char *addr,
+                              const char *port,
+                              struct netaddress *na) {
+  char *vec[3];
+  
+  vec[0] = uaudio_get(af, NULL);
+  vec[1] = uaudio_get(addr, NULL);
+  vec[2] = uaudio_get(port, NULL);
+  if(!*vec)
+    na->af = -1;
+  else
+    if(netaddress_parse(na, 3, vec))
+      fatal(0, "invalid RTP address");
+}
+
+static void rtp_set_netconfig(const char *af,
+                              const char *addr,
+                              const char *port,
+                              const struct netaddress *na) {
+  uaudio_set(af, NULL);
+  uaudio_set(addr, NULL);
+  uaudio_set(port, NULL);
+  if(na->af != -1) {
+    int nvec;
+    char **vec;
+
+    netaddress_format(na, &nvec, &vec);
+    if(nvec > 0) {
+      uaudio_set(af, vec[0]);
+      xfree(vec[0]);
+    }
+    if(nvec > 1) {
+      uaudio_set(addr, vec[1]);
+      xfree(vec[1]);
+    }
+    if(nvec > 2) {
+      uaudio_set(port, vec[2]);
+      xfree(vec[2]);
+    }
+    xfree(vec);
+  }
+}
+
+static size_t rtp_play(void *buffer, size_t nsamples, unsigned flags) {
   struct rtp_header header;
   struct iovec vec[2];
-  struct timeval now;
-  
+
+#if 0
+  if(flags & (UAUDIO_PAUSE|UAUDIO_RESUME))
+    fprintf(stderr, "rtp_play %zu samples%s%s%s%s\n", nsamples,
+            flags & UAUDIO_PAUSE ? " UAUDIO_PAUSE" : "",
+            flags & UAUDIO_RESUME ? " UAUDIO_RESUME" : "",
+            flags & UAUDIO_PLAYING ? " UAUDIO_PLAYING" : "",
+            flags & UAUDIO_PAUSED ? " UAUDIO_PAUSED" : "");
+#endif
+          
   /* We do as much work as possible before checking what time it is */
   /* Fill out header */
   header.vpxcc = 2 << 6;              /* V=2, P=0, X=0, CC=0 */
   header.seq = htons(rtp_sequence++);
   header.ssrc = rtp_id;
-  header.mpt = (rtp_reactivated ? 0x80 : 0x00) | rtp_payload;
+  header.mpt = rtp_payload;
+  /* If we've come out of a pause, set the marker bit */
+  if(flags & UAUDIO_RESUME)
+    header.mpt |= 0x80;
 #if !WORDS_BIGENDIAN
   /* Convert samples to network byte order */
   uint16_t *u = buffer, *const limit = u + nsamples;
@@ -125,61 +162,13 @@ static size_t rtp_play(void *buffer, size_t nsamples) {
   vec[0].iov_len = sizeof header;
   vec[1].iov_base = buffer;
   vec[1].iov_len = nsamples * uaudio_sample_size;
-retry:
-  xgettimeofday(&now, NULL);
-  if(rtp_reactivated) {
-    /* We've been deactivated for some unknown interval.  We need to advance
-     * rtp_timestamp to account for the dead air. */
-    /* On the first run through we'll set the start time. */
-    if(!rtp_timeval.tv_sec)
-      rtp_timeval = now;
-    /* See how much time we missed.
-     *
-     * This will be 0 on the first run through, in which case we'll not modify
-     * anything.
-     *
-     * It'll be negative in the (rare) situation where the deactivation
-     * interval is shorter than the last packet we sent.  In this case we wait
-     * for that much time and then return having sent no samples, which will
-     * cause uaudio_play_thread_fn() to retry.
-     *
-     * In the normal case it will be positive.
-     */
-    const int64_t delay = tvsub_us(now, rtp_timeval); /* microseconds */
-    if(delay < 0) {
-      usleep(-delay);
-      goto retry;
-    }
-    /* Advance the RTP timestamp to the present.  With 44.1KHz stereo this will
-     * overflow the intermediate value with a delay of a bit over 6 years.
-     * This seems acceptable. */
-    uint64_t update = (delay * uaudio_rate * uaudio_channels) / 1000000;
-    /* Don't throw off channel synchronization */
-    update -= update % uaudio_channels;
-    /* We log nontrivial changes */
-    if(update)
-      info("advancing rtp_time by %"PRIu64" samples", update);
-      rtp_timestamp += update;
-    rtp_timeval = now;
-    rtp_reactivated = 0;
-  } else {
-    /* Chances are we've been called right on the heels of the previous packet.
-     * If we just sent packets as fast as we got audio data we'd get way ahead
-     * of the player and some buffer somewhere would fill (or at least become
-     * unreasonably large).
-     *
-     * First find out how far ahead of the target time we are.
-     */
-    const int64_t ahead = tvsub_us(now, rtp_timeval); /* microseconds */
-    /* Only delay at all if we are nontrivially ahead. */
-    if(ahead > rtp_delay_threshold) {
-      /* Don't delay by the full amount */
-      usleep(ahead - rtp_delay_threshold / 2);
-      /* Refetch time (so we don't get out of step with reality) */
-      xgettimeofday(&now, NULL);
-    }
+  const uint32_t timestamp = uaudio_schedule_sync();
+  header.timestamp = htonl(rtp_base + (uint32_t)timestamp);
+  /* If we're paused don't actually end a packet, we just pretend */
+  if(flags & UAUDIO_PAUSED) {
+    uaudio_schedule_sent(nsamples);
+    return nsamples;
   }
-  header.timestamp = htonl((uint32_t)rtp_timestamp);
   int written_bytes;
   do {
     written_bytes = writev(rtp_fd, vec, 2);
@@ -192,72 +181,38 @@ retry:
     return 0;
   } else
     rtp_errors /= 2;                    /* gradual decay */
-  written_bytes -= sizeof (struct rtp_header);
-  size_t written_samples = written_bytes / uaudio_sample_size;
-  /* rtp_timestamp and rtp_timestamp are supposed to refer to the first sample
-   * of the next packet */
-  rtp_timestamp += written_samples;
-  const unsigned usec = (rtp_timeval.tv_usec
-                         + 1000000 * written_samples / (uaudio_rate
-                                                            * uaudio_channels));
-  /* ...will only overflow 32 bits if one packet is more than about half an
-   * hour long, which is not plausible. */
-  rtp_timeval.tv_sec += usec / 1000000;
-  rtp_timeval.tv_usec = usec % 1000000;
-  return written_samples;
+  /* TODO what can we sensibly do about short writes here?  Really that's just
+   * an error and we ought to be using smaller packets. */
+  uaudio_schedule_sent(nsamples);
+  return nsamples;
 }
 
 static void rtp_open(void) {
   struct addrinfo *res, *sres;
-  static const struct addrinfo pref = {
-    .ai_flags = 0,
-    .ai_family = PF_INET,
-    .ai_socktype = SOCK_DGRAM,
-    .ai_protocol = IPPROTO_UDP,
-  };
-  static const struct addrinfo prefbind = {
-    .ai_flags = AI_PASSIVE,
-    .ai_family = PF_INET,
-    .ai_socktype = SOCK_DGRAM,
-    .ai_protocol = IPPROTO_UDP,
-  };
   static const int one = 1;
   int sndbuf, target_sndbuf = 131072;
   socklen_t len;
-  char *sockname, *ssockname;
-  struct stringlist dst, src;
-  const char *delay;
+  struct netaddress dst[1], src[1];
   
   /* Get configuration */
-  dst.n = 2;
-  dst.s = xcalloc(2, sizeof *dst.s);
-  dst.s[0] = uaudio_get("rtp-destination");
-  dst.s[1] = uaudio_get("rtp-destination-port");
-  src.n = 2;
-  src.s = xcalloc(2, sizeof *dst.s);
-  src.s[0] = uaudio_get("rtp-source");
-  src.s[1] = uaudio_get("rtp-source-port");
-  if(!dst.s[0])
-    fatal(0, "'rtp-destination' not set");
-  if(!dst.s[1])
-    fatal(0, "'rtp-destination-port' not set");
-  if(src.s[0]) {
-    if(!src.s[1])
-      fatal(0, "'rtp-source-port' not set");
-    src.n = 2;
-  } else
-    src.n = 0;
-  if((delay = uaudio_get("rtp-delay-threshold")))
-    rtp_delay_threshold = atoi(delay);
-  else
-    rtp_delay_threshold = 1000;         /* microseconds */
+  rtp_get_netconfig("rtp-destination-af",
+                    "rtp-destination",
+                    "rtp-destination-port",
+                    dst);
+  rtp_get_netconfig("rtp-source-af",
+                    "rtp-source",
+                    "rtp-source-port",
+                    src);
+  /* ...microseconds */
 
   /* Resolve addresses */
-  res = get_address(&dst, &pref, &sockname);
-  if(!res) exit(-1);
-  if(src.n) {
-    sres = get_address(&src, &prefbind, &ssockname);
-    if(!sres) exit(-1);
+  res = netaddress_resolve(dst, 0, IPPROTO_UDP);
+  if(!res)
+    exit(-1);
+  if(src->af != -1) {
+    sres = netaddress_resolve(src, 1, IPPROTO_UDP);
+    if(!sres)
+      exit(-1);
   } else
     sres = 0;
   /* Create the socket */
@@ -267,10 +222,8 @@ static void rtp_open(void) {
     fatal(errno, "error creating broadcast socket");
   if(multicast(res->ai_addr)) {
     /* Enable multicast options */
-    const char *ttls = uaudio_get("multicast-ttl");
-    const int ttl = ttls ? atoi(ttls) : 1;
-    const char *loops = uaudio_get("multicast-loop");
-    const int loop = loops ? !strcmp(loops, "yes") : 1;
+    const int ttl = atoi(uaudio_get("multicast-ttl", "1"));
+    const int loop = !strcmp(uaudio_get("multicast-loop", "yes"), "yes");
     switch(res->ai_family) {
     case PF_INET: {
       if(setsockopt(rtp_fd, IPPROTO_IP, IP_MULTICAST_TTL,
@@ -294,7 +247,7 @@ static void rtp_open(void) {
       fatal(0, "unsupported address family %d", res->ai_family);
     }
     info("multicasting on %s TTL=%d loop=%s", 
-         sockname, ttl, loop ? "yes" : "no");
+         format_sockaddr(res->ai_addr), ttl, loop ? "yes" : "no");
   } else {
     struct ifaddrs *ifs;
 
@@ -313,9 +266,10 @@ static void rtp_open(void) {
     if(ifs) {
       if(setsockopt(rtp_fd, SOL_SOCKET, SO_BROADCAST, &one, sizeof one) < 0)
         fatal(errno, "error setting SO_BROADCAST on broadcast socket");
-      info("broadcasting on %s (%s)", sockname, ifs->ifa_name);
+      info("broadcasting on %s (%s)", 
+           format_sockaddr(res->ai_addr), ifs->ifa_name);
     } else
-      info("unicasting on %s", sockname);
+      info("unicasting on %s", format_sockaddr(res->ai_addr));
   }
   /* Enlarge the socket buffer */
   len = sizeof sndbuf;
@@ -335,17 +289,11 @@ static void rtp_open(void) {
   /* We might well want to set additional broadcast- or multicast-related
    * options here */
   if(sres && bind(rtp_fd, sres->ai_addr, sres->ai_addrlen) < 0)
-    fatal(errno, "error binding broadcast socket to %s", ssockname);
+    fatal(errno, "error binding broadcast socket to %s", 
+          format_sockaddr(sres->ai_addr));
   if(connect(rtp_fd, res->ai_addr, res->ai_addrlen) < 0)
-    fatal(errno, "error connecting broadcast socket to %s", sockname);
-  /* Various fields are required to have random initial values by RFC3550.  The
-   * packet contents are highly public so there's no point asking for very
-   * strong randomness. */
-  gcry_create_nonce(&rtp_id, sizeof rtp_id);
-  gcry_create_nonce(&rtp_sequence, sizeof rtp_sequence);
-  gcry_create_nonce(&rtp_timestamp, sizeof rtp_timestamp);
-  /* rtp_play() will spot this and choose an initial value */
-  rtp_timeval.tv_sec = 0;
+    fatal(errno, "error connecting broadcast socket to %s", 
+          format_sockaddr(res->ai_addr));
 }
 
 static void rtp_start(uaudio_callback *callback,
@@ -362,13 +310,21 @@ static void rtp_start(uaudio_callback *callback,
   else
     fatal(0, "asked for %d/%d/%d 16/44100/1 and 16/44100/2",
           uaudio_bits, uaudio_rate, uaudio_channels); 
+  /* Various fields are required to have random initial values by RFC3550.  The
+   * packet contents are highly public so there's no point asking for very
+   * strong randomness. */
+  gcry_create_nonce(&rtp_id, sizeof rtp_id);
+  gcry_create_nonce(&rtp_base, sizeof rtp_base);
+  gcry_create_nonce(&rtp_sequence, sizeof rtp_sequence);
   rtp_open();
+  uaudio_schedule_init();
   uaudio_thread_start(callback,
                       userdata,
                       rtp_play,
                       256 / uaudio_sample_size,
                       (NETWORK_BYTES - sizeof(struct rtp_header))
-                      / uaudio_sample_size);
+                      / uaudio_sample_size,
+                      0);
 }
 
 static void rtp_stop(void) {
@@ -377,13 +333,18 @@ static void rtp_stop(void) {
   rtp_fd = -1;
 }
 
-static void rtp_activate(void) {
-  rtp_reactivated = 1;
-  uaudio_thread_activate();
-}
+static void rtp_configure(void) {
+  char buffer[64];
 
-static void rtp_deactivate(void) {
-  uaudio_thread_deactivate();
+  rtp_set_netconfig("rtp-destination-af",
+                    "rtp-destination",
+                    "rtp-destination-port", &config->broadcast);
+  rtp_set_netconfig("rtp-source-af",
+                    "rtp-source",
+                    "rtp-source-port", &config->broadcast_from);
+  snprintf(buffer, sizeof buffer, "%ld", config->multicast_ttl);
+  uaudio_set("multicast-ttl", buffer);
+  uaudio_set("multicast-loop", config->multicast_loop ? "yes" : "no");
 }
 
 const struct uaudio uaudio_rtp = {
@@ -391,8 +352,9 @@ const struct uaudio uaudio_rtp = {
   .options = rtp_options,
   .start = rtp_start,
   .stop = rtp_stop,
-  .activate = rtp_activate,
-  .deactivate = rtp_deactivate
+  .activate = uaudio_thread_activate,
+  .deactivate = uaudio_thread_deactivate,
+  .configure = rtp_configure,
 };
 
 /*