X-Git-Url: http://www.chiark.greenend.org.uk/ucgi/~mdw/git/disorder/blobdiff_plain/7a2c706849ecf6cee19d9e502f8491ffac3322e0..287ad384814351a195696f91e83388c12b265f90:/lib/uaudio-rtp.c

diff --git a/lib/uaudio-rtp.c b/lib/uaudio-rtp.c
index 1bbfa9c..31e8f57 100644
--- a/lib/uaudio-rtp.c
+++ b/lib/uaudio-rtp.c
@@ -15,38 +15,308 @@
  * 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 <pthread.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"
 #include "log.h"
 #include "syscalls.h"
+#include "rtp.h"
+#include "addr.h"
+#include "ifreq.h"
+#include "timeval.h"
+#include "configuration.h"
+
+/** @brief Bytes to send per network packet
+ *
+ * This is the maximum number of bytes we pass to write(2); to determine actual
+ * packet sizes, add a UDP header and an IP header (and a link layer header if
+ * it's the link layer size you care about).
+ *
+ * Don't make this too big or arithmetic will start to overflow.
+ */
+#define NETWORK_BYTES (1500-8/*UDP*/-40/*IP*/-8/*conservatism*/)
+
+/** @brief RTP payload type */
+static int rtp_payload;
+
+/** @brief RTP output socket */
+static int rtp_fd;
+
+/** @brief RTP SSRC */
+static uint32_t rtp_id;
+
+/** @brief RTP sequence number */
+static uint16_t rtp_sequence;
+
+/** @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;
 
 static const char *const rtp_options[] = {
+  "rtp-destination",
+  "rtp-destination-port",
+  "rtp-source",
+  "rtp-source-port",
+  "multicast-ttl",
+  "multicast-loop",
+  "delay-threshold",
   NULL
 };
 
+static size_t rtp_play(void *buffer, size_t nsamples) {
+  struct rtp_header header;
+  struct iovec vec[2];
+  
+  /* 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 = (uaudio_schedule_reactivated ? 0x80 : 0x00) | rtp_payload;
+#if !WORDS_BIGENDIAN
+  /* Convert samples to network byte order */
+  uint16_t *u = buffer, *const limit = u + nsamples;
+  while(u < limit) {
+    *u = htons(*u);
+    ++u;
+  }
+#endif
+  vec[0].iov_base = (void *)&header;
+  vec[0].iov_len = sizeof header;
+  vec[1].iov_base = buffer;
+  vec[1].iov_len = nsamples * uaudio_sample_size;
+  uaudio_schedule_synchronize();
+  header.timestamp = htonl((uint32_t)uaudio_schedule_timestamp);
+  int written_bytes;
+  do {
+    written_bytes = writev(rtp_fd, vec, 2);
+  } while(written_bytes < 0 && errno == EINTR);
+  if(written_bytes < 0) {
+    error(errno, "error transmitting audio data");
+    ++rtp_errors;
+    if(rtp_errors == 10)
+      fatal(0, "too many audio tranmission errors");
+    return 0;
+  } else
+    rtp_errors /= 2;                    /* gradual decay */
+  written_bytes -= sizeof (struct rtp_header);
+  const size_t written_samples = written_bytes / uaudio_sample_size;
+  uaudio_schedule_update(written_samples);
+  return written_samples;
+}
+
+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;
+  
+  /* Get configuration */
+  dst.n = 2;
+  dst.s = xcalloc(2, sizeof *dst.s);
+  dst.s[0] = uaudio_get("rtp-destination", NULL);
+  dst.s[1] = uaudio_get("rtp-destination-port", NULL);
+  src.n = 2;
+  src.s = xcalloc(2, sizeof *dst.s);
+  src.s[0] = uaudio_get("rtp-source", NULL);
+  src.s[1] = uaudio_get("rtp-source-port", NULL);
+  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;
+  rtp_delay_threshold = atoi(uaudio_get("rtp-delay-threshold", "1000"));
+  /* ...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);
+  } else
+    sres = 0;
+  /* Create the socket */
+  if((rtp_fd = socket(res->ai_family,
+                      res->ai_socktype,
+                      res->ai_protocol)) < 0)
+    fatal(errno, "error creating broadcast socket");
+  if(multicast(res->ai_addr)) {
+    /* Enable multicast options */
+    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,
+                    &ttl, sizeof ttl) < 0)
+        fatal(errno, "error setting IP_MULTICAST_TTL on multicast socket");
+      if(setsockopt(rtp_fd, IPPROTO_IP, IP_MULTICAST_LOOP,
+                    &loop, sizeof loop) < 0)
+        fatal(errno, "error setting IP_MULTICAST_LOOP on multicast socket");
+      break;
+    }
+    case PF_INET6: {
+      if(setsockopt(rtp_fd, IPPROTO_IPV6, IPV6_MULTICAST_HOPS,
+                    &ttl, sizeof ttl) < 0)
+        fatal(errno, "error setting IPV6_MULTICAST_HOPS on multicast socket");
+      if(setsockopt(rtp_fd, IPPROTO_IP, IPV6_MULTICAST_LOOP,
+                    &loop, sizeof loop) < 0)
+        fatal(errno, "error setting IPV6_MULTICAST_LOOP on multicast socket");
+      break;
+    }
+    default:
+      fatal(0, "unsupported address family %d", res->ai_family);
+    }
+    info("multicasting on %s TTL=%d loop=%s", 
+         sockname, ttl, loop ? "yes" : "no");
+  } else {
+    struct ifaddrs *ifs;
+
+    if(getifaddrs(&ifs) < 0)
+      fatal(errno, "error calling getifaddrs");
+    while(ifs) {
+      /* (At least on Darwin) IFF_BROADCAST might be set but ifa_broadaddr
+       * still a null pointer.  It turns out that there's a subsequent entry
+       * for he same interface which _does_ have ifa_broadaddr though... */
+      if((ifs->ifa_flags & IFF_BROADCAST)
+         && ifs->ifa_broadaddr
+         && sockaddr_equal(ifs->ifa_broadaddr, res->ai_addr))
+        break;
+      ifs = ifs->ifa_next;
+    }
+    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);
+    } else
+      info("unicasting on %s", sockname);
+  }
+  /* Enlarge the socket buffer */
+  len = sizeof sndbuf;
+  if(getsockopt(rtp_fd, SOL_SOCKET, SO_SNDBUF,
+                &sndbuf, &len) < 0)
+    fatal(errno, "error getting SO_SNDBUF");
+  if(target_sndbuf > sndbuf) {
+    if(setsockopt(rtp_fd, 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(rtp_fd, sres->ai_addr, sres->ai_addrlen) < 0)
+    fatal(errno, "error binding broadcast socket to %s", ssockname);
+  if(connect(rtp_fd, res->ai_addr, res->ai_addrlen) < 0)
+    fatal(errno, "error connecting broadcast socket to %s", sockname);
+}
+
 static void rtp_start(uaudio_callback *callback,
                       void *userdata) {
-  (void)callback;
-  (void)userdata;
-  /* TODO */
+  /* We only support L16 (but we do stereo and mono and will convert sign) */
+  if(uaudio_channels == 2
+     && uaudio_bits == 16
+     && uaudio_rate == 44100)
+    rtp_payload = 10;
+  else if(uaudio_channels == 1
+     && uaudio_bits == 16
+     && uaudio_rate == 44100)
+    rtp_payload = 11;
+  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_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,
+                      0);
 }
 
 static void rtp_stop(void) {
-  /* TODO */
+  uaudio_thread_stop();
+  close(rtp_fd);
+  rtp_fd = -1;
 }
 
 static void rtp_activate(void) {
-  /* TODO */
+  uaudio_schedule_reactivated = 1;
+  uaudio_thread_activate();
 }
 
 static void rtp_deactivate(void) {
-  /* TODO */
+  uaudio_thread_deactivate();
+}
+
+static void rtp_configure(void) {
+  char buffer[64];
+
+  uaudio_set("rtp-destination", config->broadcast.s[0]);
+  uaudio_set("rtp-destination-port", config->broadcast.s[1]);
+  if(config->broadcast_from.n) {
+    uaudio_set("rtp-source", config->broadcast_from.s[0]);
+    uaudio_set("rtp-source-port", config->broadcast_from.s[0]);
+  } else {
+    uaudio_set("rtp-source", NULL);
+    uaudio_set("rtp-source-port", NULL);
+  }
+  snprintf(buffer, sizeof buffer, "%ld", config->multicast_ttl);
+  uaudio_set("multicast-ttl", buffer);
+  uaudio_set("multicast-loop", config->multicast_loop ? "yes" : "no");
+  snprintf(buffer, sizeof buffer, "%ld", config->rtp_delay_threshold);
+  uaudio_set("delay-threshold", buffer);
 }
 
 const struct uaudio uaudio_rtp = {
@@ -55,7 +325,8 @@ const struct uaudio uaudio_rtp = {
   .start = rtp_start,
   .stop = rtp_stop,
   .activate = rtp_activate,
-  .deactivate = rtp_deactivate
+  .deactivate = rtp_deactivate,
+  .configure = rtp_configure,
 };
 
 /*