X-Git-Url: https://www.chiark.greenend.org.uk/ucgi/~mdw/git/disorder/blobdiff_plain/4f8132b3d83dc46d4ab2321ef5b838da406bd100..refs/heads/master:/lib/uaudio-rtp.c

diff --git a/lib/uaudio-rtp.c b/lib/uaudio-rtp.c
index 1bbfa9c..dc1fd6c 100644
--- a/lib/uaudio-rtp.c
+++ b/lib/uaudio-rtp.c
@@ -1,6 +1,6 @@
 /*
  * This file is part of DisOrder.
- * Copyright (C) 2009 Richard Kettlewell
+ * Copyright (C) 2009, 2013 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
@@ -15,38 +15,527 @@
  * 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 <pthread.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 */
+static int rtp_max_payload;
+
+/** @brief RTP payload type */
+static int rtp_payload;
+
+/** @brief RTP broadcast/multicast output socket */
+static int rtp_fd = -1;
+
+/** @brief RTP unicast output socket (IPv4) */
+static int rtp_fd4 = -1;
+
+/** @brief RTP unicast output socket (IPv6) */
+static int rtp_fd6 = -1;
+
+/** @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 Network error count
+ *
+ * If too many errors occur in too short a time, we give up.
+ */
+static int rtp_errors;
+
+/** @brief RTP mode */
+static int rtp_mode;
+
+#define RTP_BROADCAST 1
+#define RTP_MULTICAST 2
+#define RTP_UNICAST 3
+#define RTP_REQUEST 4
+#define RTP_AUTO 5
+
+/** @brief A unicast client */
+struct rtp_recipient {
+  struct rtp_recipient *next;
+  struct sockaddr_storage sa;
+};
+
+/** @brief List of unicast clients */
+static struct rtp_recipient *rtp_recipient_list;
+
+/** @brief Mutex protecting data structures */
+static pthread_mutex_t rtp_lock = PTHREAD_MUTEX_INITIALIZER;
 
 static const char *const rtp_options[] = {
+  "rtp-destination",
+  "rtp-destination-port",
+  "rtp-source",
+  "rtp-source-port",
+  "multicast-ttl",
+  "multicast-loop",
+  "rtp-mode",
+  "rtp-max-payload",
+  "rtp-mtu-discovery",
   NULL
 };
 
+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))
+      disorder_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];
+
+#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_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;
+  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;
+  const uint32_t timestamp = uaudio_schedule_sync();
+  header.timestamp = htonl(rtp_base + (uint32_t)timestamp);
+
+  /* We send ~120 packets a second with current arrangements.  So if we log
+   * once every 8192 packets we log about once a minute. */
+
+  if(!(ntohs(header.seq) & 8191)
+     && config->rtp_verbose)
+    disorder_info("RTP: seq %04"PRIx16" %08"PRIx32"+%08"PRIx32"=%08"PRIx32" ns %zu%s",
+                  ntohs(header.seq),
+                  rtp_base,
+                  timestamp,
+                  header.timestamp,
+                  nsamples,
+                  flags & UAUDIO_PAUSED ? " [paused]" : "");
+
+  /* If we're paused don't actually end a packet, we just pretend */
+  if(flags & UAUDIO_PAUSED) {
+    uaudio_schedule_sent(nsamples);
+    return nsamples;
+  }
+  /* Send stuff to explicitly registerd unicast addresses unconditionally */
+  struct rtp_recipient *r;
+  struct msghdr m;
+  memset(&m, 0, sizeof m);
+  m.msg_iov = vec;
+  m.msg_iovlen = 2;
+  pthread_mutex_lock(&rtp_lock);
+  for(r = rtp_recipient_list; r; r = r->next) {
+    m.msg_name = &r->sa;
+    m.msg_namelen = r->sa.ss_family == AF_INET ? 
+      sizeof(struct sockaddr_in) : sizeof (struct sockaddr_in6);
+    sendmsg(r->sa.ss_family == AF_INET ? rtp_fd4 : rtp_fd6,
+            &m, MSG_DONTWAIT|MSG_NOSIGNAL);
+    // TODO similar error handling to other case?
+  }
+  pthread_mutex_unlock(&rtp_lock);
+  if(rtp_mode != RTP_REQUEST) {
+    int written_bytes;
+    do {
+      written_bytes = writev(rtp_fd, vec, 2);
+    } while(written_bytes < 0 && errno == EINTR);
+    if(written_bytes < 0) {
+      disorder_error(errno, "error transmitting audio data");
+      ++rtp_errors;
+      if(rtp_errors == 10)
+        disorder_fatal(0, "too many audio transmission errors");
+      return 0;
+    } else
+      rtp_errors /= 2;                    /* gradual decay */
+  }
+  /* 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 hack_send_buffer_size(int fd, const char *what) {
+  int sndbuf, target_sndbuf = 131072;
+  socklen_t len = sizeof sndbuf;
+
+  if(getsockopt(fd, SOL_SOCKET, SO_SNDBUF,
+                &sndbuf, &len) < 0)
+    disorder_fatal(errno, "error getting SO_SNDBUF on %s socket", what);
+  if(target_sndbuf > sndbuf) {
+    if(setsockopt(fd, SOL_SOCKET, SO_SNDBUF,
+                  &target_sndbuf, sizeof target_sndbuf) < 0)
+      disorder_error(errno, "error setting SO_SNDBUF on %s socket to %d",
+                     what, target_sndbuf);
+    else
+      disorder_info("changed socket send buffer size on %socket from %d to %d",
+                    what, sndbuf, target_sndbuf);
+  } else
+    disorder_info("default socket send buffer on %s socket is %d",
+                  what, sndbuf);
+}
+
+static void rtp_open(void) {
+  struct resolved *dres, *sres;
+  size_t ndres, nsres;
+  static const int one = 1;
+  struct netaddress dst[1], src[1];
+  const char *mode;
+#ifdef IP_MTU_DISCOVER
+  const char *mtu_disc;
+  int opt;
+#endif
+  
+  /* Get the mode */
+  mode = uaudio_get("rtp-mode", "auto");
+  if(!strcmp(mode, "broadcast")) rtp_mode = RTP_BROADCAST;
+  else if(!strcmp(mode, "multicast")) rtp_mode = RTP_MULTICAST;
+  else if(!strcmp(mode, "unicast")) rtp_mode = RTP_UNICAST;
+  else if(!strcmp(mode, "request")) rtp_mode = RTP_REQUEST;
+  else rtp_mode = RTP_AUTO;
+  /* Get the source and destination addresses (which might be missing) */
+  rtp_get_netconfig("rtp-destination-af",
+                    "rtp-destination",
+                    "rtp-destination-port",
+                    dst);
+  rtp_get_netconfig("rtp-source-af",
+                    "rtp-source",
+                    "rtp-source-port",
+                    src);
+  if(dst->af != -1) {
+    if(netaddress_resolve(dst, 0, SOCK_DGRAM, &dres, &ndres))
+      exit(-1);
+  } else {
+    dres = 0;
+    ndres = 0;
+  }
+  if(src->af != -1) {
+    if(netaddress_resolve(src, 0, SOCK_DGRAM, &sres, &nsres))
+      exit(-1);
+  } else {
+    sres = 0;
+    nsres = 0;
+  }
+  /* _AUTO inspects the destination address and acts accordingly */
+  if(rtp_mode == RTP_AUTO) {
+    if(!dres)
+      rtp_mode = RTP_REQUEST;
+    else if(multicast(dres->sa))
+      rtp_mode = RTP_MULTICAST;
+    else {
+      struct ifaddrs *ifs;
+
+      if(getifaddrs(&ifs) < 0)
+        disorder_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, dres->sa))
+          break;
+        ifs = ifs->ifa_next;
+      }
+      if(ifs) 
+        rtp_mode = RTP_BROADCAST;
+      else
+        rtp_mode = RTP_UNICAST;
+    }
+  }
+  rtp_max_payload = atoi(uaudio_get("rtp-max-payload", "-1"));
+  if(rtp_max_payload < 0)
+    rtp_max_payload = 1500 - 8/*UDP*/ - 40/*IP*/ - 8/*conservatism*/;
+  /* Create the sockets */
+  if(rtp_mode != RTP_REQUEST) {
+    if((rtp_fd = socket(dres->sa->sa_family, SOCK_DGRAM, IPPROTO_UDP)) < 0)
+      disorder_fatal(errno, "error creating RTP transmission socket");
+  }
+  if((rtp_fd4 = socket(AF_INET, SOCK_DGRAM, IPPROTO_UDP)) < 0)
+    disorder_fatal(errno, "error creating v4 RTP transmission socket");
+  if((rtp_fd6 = socket(AF_INET6, SOCK_DGRAM, IPPROTO_UDP)) < 0)
+    disorder_fatal(errno, "error creating v6 RTP transmission socket");
+  /* Configure the socket according to the desired mode */
+  switch(rtp_mode) {
+  case RTP_MULTICAST: {
+    /* Enable multicast options */
+    const int ttl = atoi(uaudio_get("multicast-ttl", "1"));
+    const int loop = !strcmp(uaudio_get("multicast-loop", "yes"), "yes");
+    switch(dres->sa->sa_family) {
+    case PF_INET: {
+      if(setsockopt(rtp_fd, IPPROTO_IP, IP_MULTICAST_TTL,
+                    &ttl, sizeof ttl) < 0)
+        disorder_fatal(errno, "error setting IP_MULTICAST_TTL on multicast socket");
+      if(setsockopt(rtp_fd, IPPROTO_IP, IP_MULTICAST_LOOP,
+                    &loop, sizeof loop) < 0)
+        disorder_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)
+        disorder_fatal(errno, "error setting IPV6_MULTICAST_HOPS on multicast socket");
+      if(setsockopt(rtp_fd, IPPROTO_IP, IPV6_MULTICAST_LOOP,
+                    &loop, sizeof loop) < 0)
+        disorder_fatal(errno, "error setting IPV6_MULTICAST_LOOP on multicast socket");
+      break;
+    }
+    default:
+      disorder_fatal(0, "unsupported address family %d", dres->sa->sa_family);
+    }
+    disorder_info("multicasting on %s TTL=%d loop=%s", 
+                  format_sockaddr(dres->sa), ttl, loop ? "yes" : "no");
+    break;
+  }
+  case RTP_UNICAST: {
+    disorder_info("unicasting on %s", format_sockaddr(dres->sa));
+    break;
+  }
+  case RTP_BROADCAST: {
+    if(setsockopt(rtp_fd, SOL_SOCKET, SO_BROADCAST, &one, sizeof one) < 0)
+      disorder_fatal(errno, "error setting SO_BROADCAST on broadcast socket");
+    disorder_info("broadcasting on %s", 
+                  format_sockaddr(dres->sa));
+    break;
+  }
+  case RTP_REQUEST: {
+    disorder_info("will transmit on request");
+    break;
+  }
+  }
+  /* Enlarge the socket buffers */
+  if (rtp_fd != -1) hack_send_buffer_size(rtp_fd, "master socket");
+  hack_send_buffer_size(rtp_fd4, "IPv4 on-demand socket");
+  hack_send_buffer_size(rtp_fd6, "IPv6 on-demand socket");
+  /* We might well want to set additional broadcast- or multicast-related
+   * options here */
+  if(rtp_mode != RTP_REQUEST) {
+    if(sres && bind(rtp_fd, sres->sa, sres->len) < 0)
+      disorder_fatal(errno, "error binding broadcast socket to %s", 
+                     format_sockaddr(sres->sa));
+    if(connect(rtp_fd, dres->sa, dres->len) < 0)
+      disorder_fatal(errno, "error connecting broadcast socket to %s", 
+                     format_sockaddr(dres->sa));
+  }
+#ifdef IP_MTU_DISCOVER
+  mtu_disc = uaudio_get("rtp-mtu-discovery", "default");
+  do {
+    if(!strcmp(mtu_disc, "yes")) opt = IP_PMTUDISC_DO;
+    else if(!strcmp(mtu_disc, "no")) opt = IP_PMTUDISC_DONT;
+    else break;
+    if(setsockopt(rtp_fd4, IPPROTO_IP, IP_MTU_DISCOVER, &opt, sizeof opt))
+      disorder_fatal(errno, "error setting MTU discovery");
+    if(sres->sa->sa_family == AF_INET &&
+        setsockopt(rtp_fd, IPPROTO_IP, IP_MTU_DISCOVER, &opt, sizeof opt))
+      disorder_fatal(errno, "error setting MTU discovery");
+  } while (0);
+#endif
+  if(config->rtp_verbose)
+    disorder_info("RTP: prepared socket");
+}
+
 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
+    disorder_fatal(0, "asked for %d/%d/%d 16/44100/1 and 16/44100/2",
+                   uaudio_bits, uaudio_rate, uaudio_channels); 
+  if(config->rtp_verbose)
+    disorder_info("RTP: %d channels %d bits %d Hz payload type %d",
+                  uaudio_channels, uaudio_bits, uaudio_rate, rtp_payload);
+  /* 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);
+  if(config->rtp_verbose)
+    disorder_info("RTP: id %08"PRIx32" base %08"PRIx32" initial seq %08"PRIx16,
+                  rtp_id, rtp_base, rtp_sequence);
+  rtp_open();
+  uaudio_schedule_init();
+  if(config->rtp_verbose)
+    disorder_info("RTP: initialized schedule");
+  uaudio_thread_start(callback,
+                      userdata,
+                      rtp_play,
+                      256 / uaudio_sample_size,
+                      (rtp_max_payload - sizeof(struct rtp_header))
+                      / uaudio_sample_size,
+                      0);
+  if(config->rtp_verbose)
+    disorder_info("RTP: created thread");
 }
 
 static void rtp_stop(void) {
-  /* TODO */
+  uaudio_thread_stop();
+  if(rtp_fd >= 0) { close(rtp_fd); rtp_fd = -1; }
+  if(rtp_fd4 >= 0) { close(rtp_fd4); rtp_fd4 = -1; }
+  if(rtp_fd6 >= 0) { close(rtp_fd6); rtp_fd6 = -1; }
 }
 
-static void rtp_activate(void) {
-  /* TODO */
+static void rtp_configure(void) {
+  char buffer[64];
+
+  uaudio_set("rtp-mode", config->rtp_mode);
+  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");
+  snprintf(buffer, sizeof buffer, "%ld", config->rtp_max_payload);
+  uaudio_set("rtp-max-payload", buffer);
+  uaudio_set("rtp-mtu-discovery", config->rtp_mtu_discovery);
+  if(config->rtp_verbose)
+    disorder_info("RTP: configured");
 }
 
-static void rtp_deactivate(void) {
-  /* TODO */
+/** @brief Add an RTP recipient address
+ * @param sa Pointer to recipient address
+ * @return 0 on success, -1 on error
+ */
+int rtp_add_recipient(const struct sockaddr_storage *sa) {
+  struct rtp_recipient *r;
+  int rc;
+  pthread_mutex_lock(&rtp_lock);
+  for(r = rtp_recipient_list;
+      r && sockaddrcmp((struct sockaddr *)sa,
+                       (struct sockaddr *)&r->sa);
+      r = r->next)
+    ;
+  if(r)
+    rc = -1;
+  else {
+    r = xmalloc(sizeof *r);
+    memcpy(&r->sa, sa, sizeof *sa);
+    r->next = rtp_recipient_list;
+    rtp_recipient_list = r;
+    rc = 0;
+  }
+  pthread_mutex_unlock(&rtp_lock);
+  return rc;
+}
+
+/** @brief Remove an RTP recipient address
+ * @param sa Pointer to recipient address
+ * @return 0 on success, -1 on error
+ */
+int rtp_remove_recipient(const struct sockaddr_storage *sa) {
+  struct rtp_recipient *r, **rr;
+  int rc;
+  pthread_mutex_lock(&rtp_lock);
+  for(rr = &rtp_recipient_list;
+      (r = *rr) && sockaddrcmp((struct sockaddr *)sa,
+                               (struct sockaddr *)&r->sa);
+      rr = &r->next)
+    ;
+  if(r) {
+    *rr = r->next;
+    xfree(r);
+    rc = 0;
+  } else {
+    disorder_error(0, "bogus rtp_remove_recipient");
+    rc = -1;
+  }
+  pthread_mutex_unlock(&rtp_lock);
+  return rc;
 }
 
 const struct uaudio uaudio_rtp = {
@@ -54,8 +543,10 @@ 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,
+  .flags = UAUDIO_API_SERVER,
 };
 
 /*