X-Git-Url: http://www.chiark.greenend.org.uk/ucgi/~mdw/git/disorder/blobdiff_plain/a99c4e9a49364934a3f74f003507477b9840acb2..c897bb654f1536c082d077d030c6b0fa5b471fc2:/clients/playrtp.c

diff --git a/clients/playrtp.c b/clients/playrtp.c
index f92d872..f5103a3 100644
--- a/clients/playrtp.c
+++ b/clients/playrtp.c
@@ -1,21 +1,19 @@
 /*
  * This file is part of DisOrder.
- * Copyright (C) 2007 Richard Kettlewell
+ * Copyright (C) 2007, 2008 Richard Kettlewell
  *
- * This program is free software; you can redistribute it and/or modify
+ * 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
- * the Free Software Foundation; either version 2 of the License, or
+ * the Free Software Foundation, either version 3 of the License, or
  * (at your option) any later version.
  *
- * This program is distributed in the hope that it will be useful, but
- * WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * General Public License for more details.
- *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ * 
  * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307
- * USA
+ * along with this program.  If not, see <http://www.gnu.org/licenses/>.
  */
 /** @file clients/playrtp.c
  * @brief RTP player
@@ -49,12 +47,9 @@
  * - it is safe to read uint32_t values without a lock protecting them
  */
 
-#include <config.h>
-#include "types.h"
+#include "common.h"
 
 #include <getopt.h>
-#include <stdio.h>
-#include <stdlib.h>
 #include <sys/socket.h>
 #include <sys/types.h>
 #include <sys/socket.h>
@@ -62,12 +57,13 @@
 #include <pthread.h>
 #include <locale.h>
 #include <sys/uio.h>
-#include <string.h>
-#include <assert.h>
 #include <errno.h>
 #include <netinet/in.h>
 #include <sys/time.h>
 #include <sys/un.h>
+#include <unistd.h>
+#include <sys/mman.h>
+#include <fcntl.h>
 
 #include "log.h"
 #include "mem.h"
@@ -82,6 +78,8 @@
 #include "client.h"
 #include "playrtp.h"
 #include "inputline.h"
+#include "version.h"
+#include "uaudio.h"
 
 #define readahead linux_headers_are_borked
 
@@ -97,7 +95,6 @@ static int rtpfd;
 static FILE *logfp;
 
 /** @brief Output device */
-const char *device;
 
 /** @brief Minimum low watermark
  *
@@ -171,24 +168,35 @@ pthread_mutex_t lock = PTHREAD_MUTEX_INITIALIZER;
 /** @brief Condition variable signalled whenever @ref packets is changed */
 pthread_cond_t cond = PTHREAD_COND_INITIALIZER;
 
-#if HAVE_ALSA_ASOUNDLIB_H
-# define DEFAULT_BACKEND playrtp_alsa
-#elif HAVE_SYS_SOUNDCARD_H || EMPEG_HOST
-# define DEFAULT_BACKEND playrtp_oss
-#elif HAVE_COREAUDIO_AUDIOHARDWARE_H
-# define DEFAULT_BACKEND playrtp_coreaudio
-#else
-# error No known backend
-#endif
-
 /** @brief Backend to play with */
-static void (*backend)(void) = &DEFAULT_BACKEND;
+static const struct uaudio *backend;
 
 HEAP_DEFINE(pheap, struct packet *, lt_packet);
 
 /** @brief Control socket or NULL */
 const char *control_socket;
 
+/** @brief Buffer for debugging dump
+ *
+ * The debug dump is enabled by the @c --dump option.  It records the last 20s
+ * of audio to the specified file (which will be about 3.5Mbytes).  The file is
+ * written as as ring buffer, so the start point will progress through it.
+ *
+ * Use clients/dump2wav to convert this to a WAV file, which can then be loaded
+ * into (e.g.) Audacity for further inspection.
+ *
+ * All three backends (ALSA, OSS, Core Audio) now support this option.
+ *
+ * The idea is to allow the user a few seconds to react to an audible artefact.
+ */
+int16_t *dump_buffer;
+
+/** @brief Current index within debugging dump */
+size_t dump_index;
+
+/** @brief Size of debugging dump in samples */
+size_t dump_size = 44100/*Hz*/ * 2/*channels*/ * 20/*seconds*/;
+
 static const struct option options[] = {
   { "help", no_argument, 0, 'h' },
   { "version", no_argument, 0, 'V' },
@@ -198,7 +206,6 @@ static const struct option options[] = {
   { "max", required_argument, 0, 'x' },
   { "buffer", required_argument, 0, 'b' },
   { "rcvbuf", required_argument, 0, 'R' },
-  { "multicast", required_argument, 0, 'M' },
 #if HAVE_SYS_SOUNDCARD_H || EMPEG_HOST
   { "oss", no_argument, 0, 'o' },
 #endif
@@ -208,6 +215,7 @@ static const struct option options[] = {
 #if HAVE_COREAUDIO_AUDIOHARDWARE_H
   { "core-audio", no_argument, 0, 'c' },
 #endif
+  { "dump", required_argument, 0, 'r' },
   { "socket", required_argument, 0, 's' },
   { "config", required_argument, 0, 'C' },
   { 0, 0, 0, 0 }
@@ -305,8 +313,9 @@ static void *queue_thread(void attribute((unused)) *arg) {
   for(;;) {
     /* Get the next packet */
     pthread_mutex_lock(&receive_lock);
-    while(!received_packets)
+    while(!received_packets) {
       pthread_cond_wait(&receive_cond, &receive_lock);
+    }
     p = received_packets;
     received_packets = p->next;
     if(!received_packets)
@@ -380,6 +389,9 @@ static void *listen_thread(void attribute((unused)) *arg) {
            timestamp, next_timestamp);
       continue;
     }
+    /* Ignore packets with the extension bit set. */
+    if(header.vpxcc & 0x10)
+      continue;
     p->next = 0;
     p->flags = 0;
     p->timestamp = timestamp;
@@ -387,7 +399,7 @@ static void *listen_thread(void attribute((unused)) *arg) {
     if(header.mpt & 0x80)
       p->flags |= IDLE;
     switch(header.mpt & 0x7F) {
-    case 10:
+    case 10:                            /* L16 */
       p->nsamples = (n - sizeof header) / sizeof(uint16_t);
       break;
       /* TODO support other RFC3551 media types (when the speaker does) */
@@ -404,8 +416,9 @@ static void *listen_thread(void attribute((unused)) *arg) {
      * out of order then we guarantee dropouts.  But for now... */
     if(nsamples >= maxbuffer) {
       pthread_mutex_lock(&lock);
-      while(nsamples >= maxbuffer)
+      while(nsamples >= maxbuffer) {
         pthread_cond_wait(&cond, &lock);
+      }
       pthread_mutex_unlock(&lock);
     }
     /* Add the packet to the receive queue */
@@ -428,8 +441,9 @@ void playrtp_fill_buffer(void) {
   while(nsamples)
     drop_first_packet();
   info("Buffering...");
-  while(nsamples < readahead)
+  while(nsamples < readahead) {
     pthread_cond_wait(&cond, &lock);
+  }
   next_timestamp = pheap_first(&packets)->timestamp;
   active = 1;
 }
@@ -457,40 +471,16 @@ struct packet *playrtp_next_packet(void) {
   return 0;
 }
 
-/** @brief Play an RTP stream
- *
- * This is the guts of the program.  It is responsible for:
- * - starting the listening thread
- * - opening the audio device
- * - reading ahead to build up a buffer
- * - arranging for audio to be played
- * - detecting when the buffer has got too small and re-buffering
- */
-static void play_rtp(void) {
-  pthread_t ltid;
-  int err;
-
-  /* We receive and convert audio data in a background thread */
-  if((err = pthread_create(&ltid, 0, listen_thread, 0)))
-    fatal(err, "pthread_create listen_thread");
-  /* We have a second thread to add received packets to the queue */
-  if((err = pthread_create(&ltid, 0, queue_thread, 0)))
-    fatal(err, "pthread_create queue_thread");
-  /* The rest of the work is backend-specific */
-  backend();
-}
-
 /* display usage message and terminate */
 static void help(void) {
   xprintf("Usage:\n"
-	  "  disorder-playrtp [OPTIONS] ADDRESS [PORT]\n"
+	  "  disorder-playrtp [OPTIONS] [[ADDRESS] PORT]\n"
 	  "Options:\n"
           "  --device, -D DEVICE     Output device\n"
           "  --min, -m FRAMES        Buffer low water mark\n"
           "  --buffer, -b FRAMES     Buffer high water mark\n"
           "  --max, -x FRAMES        Buffer maximum size\n"
           "  --rcvbuf, -R BYTES      Socket receive buffer size\n"
-          "  --multicast, -M GROUP   Join multicast group\n"
           "  --config, -C PATH       Set configuration file\n"
 #if HAVE_ALSA_ASOUNDLIB_H
           "  --alsa, -a              Use ALSA to play audio\n"
@@ -508,11 +498,64 @@ static void help(void) {
   exit(0);
 }
 
-/* display version number and terminate */
-static void version(void) {
-  xprintf("disorder-playrtp version %s\n", disorder_version_string);
-  xfclose(stdout);
-  exit(0);
+static size_t playrtp_callback(void *buffer,
+                               size_t max_samples,
+                               void attribute((unused)) *userdata) {
+  size_t samples;
+
+  pthread_mutex_lock(&lock);
+  /* Get the next packet, junking any that are now in the past */
+  const struct packet *p = playrtp_next_packet();
+  if(p && contains(p, next_timestamp)) {
+    /* This packet is ready to play; the desired next timestamp points
+     * somewhere into it. */
+
+    /* Timestamp of end of packet */
+    const uint32_t packet_end = p->timestamp + p->nsamples;
+
+    /* Offset of desired next timestamp into current packet */
+    const uint32_t offset = next_timestamp - p->timestamp;
+
+    /* Pointer to audio data */
+    const uint16_t *ptr = (void *)(p->samples_raw + offset);
+
+    /* Compute number of samples left in packet, limited to output buffer
+     * size */
+    samples = packet_end - next_timestamp;
+    if(samples > max_samples)
+      samples = max_samples;
+
+    /* Copy into buffer, converting to native endianness */
+    size_t i = samples;
+    int16_t *bufptr = buffer;
+    while(i > 0) {
+      *bufptr++ = (int16_t)ntohs(*ptr++);
+      --i;
+    }
+    /* We don't junk the packet here; a subsequent call to
+     * playrtp_next_packet() will dispose of it (if it's actually done with). */
+  } else {
+    /* There is no suitable packet.  We introduce 0s up to the next packet, or
+     * to fill the buffer if there's no next packet or that's too many.  The
+     * comparison with max_samples deals with the otherwise troubling overflow
+     * case. */
+    samples = p ? p->timestamp - next_timestamp : max_samples;
+    if(samples > max_samples)
+      samples = max_samples;
+    //info("infill by %zu", samples);
+    memset(buffer, 0, samples * uaudio_sample_size);
+  }
+  /* Debug dump */
+  if(dump_buffer) {
+    for(size_t i = 0; i < samples; ++i) {
+      dump_buffer[dump_index++] = ((int16_t *)buffer)[i];
+      dump_index %= dump_size;
+    }
+  }
+  /* Advance timestamp */
+  next_timestamp += samples;
+  pthread_mutex_unlock(&lock);
+  return samples;
 }
 
 int main(int argc, char **argv) {
@@ -522,29 +565,35 @@ int main(int argc, char **argv) {
   char *sockname;
   int rcvbuf, target_rcvbuf = 131072;
   socklen_t len;
-  char *multicast_group = 0;
   struct ip_mreq mreq;
   struct ipv6_mreq mreq6;
   disorder_client *c;
   char *address, *port;
+  int is_multicast;
+  union any_sockaddr {
+    struct sockaddr sa;
+    struct sockaddr_in in;
+    struct sockaddr_in6 in6;
+  };
+  union any_sockaddr mgroup;
+  const char *dumpfile = 0;
+  const char *device = 0;
+  pthread_t ltid;
 
   static const struct addrinfo prefs = {
-    AI_PASSIVE,
-    PF_INET,
-    SOCK_DGRAM,
-    IPPROTO_UDP,
-    0,
-    0,
-    0,
-    0
+    .ai_flags = AI_PASSIVE,
+    .ai_family = PF_INET,
+    .ai_socktype = SOCK_DGRAM,
+    .ai_protocol = IPPROTO_UDP
   };
 
   mem_init();
   if(!setlocale(LC_CTYPE, "")) fatal(errno, "error calling setlocale");
-  while((n = getopt_long(argc, argv, "hVdD:m:b:x:L:R:M:aocC:", options, 0)) >= 0) {
+  backend = uaudio_apis[0];
+  while((n = getopt_long(argc, argv, "hVdD:m:b:x:L:R:M:aocC:r", options, 0)) >= 0) {
     switch(n) {
     case 'h': help();
-    case 'V': version();
+    case 'V': version("disorder-playrtp");
     case 'd': debugging = 1; break;
     case 'D': device = optarg; break;
     case 'm': minbuffer = 2 * atol(optarg); break;
@@ -552,18 +601,18 @@ int main(int argc, char **argv) {
     case 'x': maxbuffer = 2 * atol(optarg); break;
     case 'L': logfp = fopen(optarg, "w"); break;
     case 'R': target_rcvbuf = atoi(optarg); break;
-    case 'M': multicast_group = optarg; break;
 #if HAVE_ALSA_ASOUNDLIB_H
-    case 'a': backend = playrtp_alsa; break;
+    case 'a': backend = &uaudio_alsa; break;
 #endif
 #if HAVE_SYS_SOUNDCARD_H || EMPEG_HOST
-    case 'o': backend = playrtp_oss; break;
+    case 'o': backend = &uaudio_oss; break;
 #endif
 #if HAVE_COREAUDIO_AUDIOHARDWARE_H      
-    case 'c': backend = playrtp_coreaudio; break;
+    case 'c': backend = &uaudio_coreaudio; break;
 #endif
     case 'C': configfile = optarg; break;
     case 's': control_socket = optarg; break;
+    case 'r': dumpfile = optarg; break;
     default: fatal(0, "invalid option");
     }
   }
@@ -574,44 +623,70 @@ int main(int argc, char **argv) {
   argv += optind;
   switch(argc) {
   case 0:
-  case 1:
+    /* Get configuration from server */
     if(!(c = disorder_new(1))) exit(EXIT_FAILURE);
     if(disorder_connect(c)) exit(EXIT_FAILURE);
     if(disorder_rtp_address(c, &address, &port)) exit(EXIT_FAILURE);
-    sl.n = 1;
-    sl.s = &port;
-    /* set multicast_group if address is a multicast address */
+    sl.n = 2;
+    sl.s = xcalloc(2, sizeof *sl.s);
+    sl.s[0] = address;
+    sl.s[1] = port;
     break;
+  case 1:
   case 2:
+    /* Use command-line ADDRESS+PORT or just PORT */
     sl.n = argc;
     sl.s = argv;
     break;
   default:
-    fatal(0, "usage: disorder-playrtp [OPTIONS] [ADDRESS [PORT]]");
+    fatal(0, "usage: disorder-playrtp [OPTIONS] [[ADDRESS] PORT]");
   }
-  /* Listen for inbound audio data */
+  /* Look up address and port */
   if(!(res = get_address(&sl, &prefs, &sockname)))
     exit(1);
-  info("listening on %s", sockname);
+  /* Create the socket */
   if((rtpfd = socket(res->ai_family,
                      res->ai_socktype,
                      res->ai_protocol)) < 0)
     fatal(errno, "error creating socket");
+  /* Stash the multicast group address */
+  if((is_multicast = multicast(res->ai_addr))) {
+    memcpy(&mgroup, res->ai_addr, res->ai_addrlen);
+    switch(res->ai_addr->sa_family) {
+    case AF_INET:
+      mgroup.in.sin_port = 0;
+      break;
+    case AF_INET6:
+      mgroup.in6.sin6_port = 0;
+      break;
+    }
+  }
+  /* Bind to 0/port */
+  switch(res->ai_addr->sa_family) {
+  case AF_INET:
+    memset(&((struct sockaddr_in *)res->ai_addr)->sin_addr, 0,
+           sizeof (struct in_addr));
+    break;
+  case AF_INET6:
+    memset(&((struct sockaddr_in6 *)res->ai_addr)->sin6_addr, 0,
+           sizeof (struct in6_addr));
+    break;
+  default:
+    fatal(0, "unsupported family %d", (int)res->ai_addr->sa_family);
+  }
   if(bind(rtpfd, res->ai_addr, res->ai_addrlen) < 0)
     fatal(errno, "error binding socket to %s", sockname);
-  if(multicast_group) {
-    if((n = getaddrinfo(multicast_group, 0, &prefs, &res)))
-      fatal(0, "getaddrinfo %s: %s", multicast_group, gai_strerror(n));
-    switch(res->ai_family) {
+  if(is_multicast) {
+    switch(mgroup.sa.sa_family) {
     case PF_INET:
-      mreq.imr_multiaddr = ((struct sockaddr_in *)res->ai_addr)->sin_addr;
+      mreq.imr_multiaddr = mgroup.in.sin_addr;
       mreq.imr_interface.s_addr = 0;      /* use primary interface */
       if(setsockopt(rtpfd, IPPROTO_IP, IP_ADD_MEMBERSHIP,
                     &mreq, sizeof mreq) < 0)
         fatal(errno, "error calling setsockopt IP_ADD_MEMBERSHIP");
       break;
     case PF_INET6:
-      mreq6.ipv6mr_multiaddr = ((struct sockaddr_in6 *)res->ai_addr)->sin6_addr;
+      mreq6.ipv6mr_multiaddr = mgroup.in6.sin6_addr;
       memset(&mreq6.ipv6mr_interface, 0, sizeof mreq6.ipv6mr_interface);
       if(setsockopt(rtpfd, IPPROTO_IPV6, IPV6_JOIN_GROUP,
                     &mreq6, sizeof mreq6) < 0)
@@ -620,7 +695,10 @@ int main(int argc, char **argv) {
     default:
       fatal(0, "unsupported address family %d", res->ai_family);
     }
-  }
+    info("listening on %s multicast group %s",
+         format_sockaddr(res->ai_addr), format_sockaddr(&mgroup.sa));
+  } else
+    info("listening on %s", format_sockaddr(res->ai_addr));
   len = sizeof rcvbuf;
   if(getsockopt(rtpfd, SOL_SOCKET, SO_RCVBUF, &rcvbuf, &len) < 0)
     fatal(errno, "error calling getsockopt SO_RCVBUF");
@@ -643,7 +721,61 @@ int main(int argc, char **argv) {
     if((err = pthread_create(&tid, 0, control_thread, 0)))
       fatal(err, "pthread_create control_thread");
   }
-  play_rtp();
+  if(dumpfile) {
+    int fd;
+    unsigned char buffer[65536];
+    size_t written;
+
+    if((fd = open(dumpfile, O_RDWR|O_TRUNC|O_CREAT, 0666)) < 0)
+      fatal(errno, "opening %s", dumpfile);
+    /* Fill with 0s to a suitable size */
+    memset(buffer, 0, sizeof buffer);
+    for(written = 0; written < dump_size * sizeof(int16_t);
+        written += sizeof buffer) {
+      if(write(fd, buffer, sizeof buffer) < 0)
+        fatal(errno, "clearing %s", dumpfile);
+    }
+    /* Map the buffer into memory for convenience */
+    dump_buffer = mmap(0, dump_size * sizeof(int16_t), PROT_READ|PROT_WRITE,
+                       MAP_SHARED, fd, 0);
+    if(dump_buffer == (void *)-1)
+      fatal(errno, "mapping %s", dumpfile);
+    info("dumping to %s", dumpfile);
+  }
+  /* Choose output device */
+  if(device)
+    uaudio_set("device", device);
+  /* Set up output.  Currently we only support L16 so there's no harm setting
+   * the format before we know what it is! */
+  uaudio_set_format(44100/*Hz*/, 2/*channels*/,
+                    16/*bits/channel*/, 1/*signed*/);
+  backend->start(playrtp_callback, NULL);
+  /* We receive and convert audio data in a background thread */
+  if((err = pthread_create(&ltid, 0, listen_thread, 0)))
+    fatal(err, "pthread_create listen_thread");
+  /* We have a second thread to add received packets to the queue */
+  if((err = pthread_create(&ltid, 0, queue_thread, 0)))
+    fatal(err, "pthread_create queue_thread");
+  pthread_mutex_lock(&lock);
+  for(;;) {
+    /* Wait for the buffer to fill up a bit */
+    playrtp_fill_buffer();
+    /* Start playing now */
+    info("Playing...");
+    next_timestamp = pheap_first(&packets)->timestamp;
+    active = 1;
+    backend->activate();
+    /* Wait until the buffer empties out */
+    while(nsamples >= minbuffer
+	  || (nsamples > 0
+	      && contains(pheap_first(&packets), next_timestamp))) {
+      pthread_cond_wait(&cond, &lock);
+    }
+    /* Stop playing for a bit until the buffer re-fills */
+    backend->deactivate();
+    active = 0;
+    /* Go back round */
+  }
   return 0;
 }