/*
* 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
* - 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>
#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"
#include "client.h"
#include "playrtp.h"
#include "inputline.h"
+#include "version.h"
+#include "uaudio.h"
#define readahead linux_headers_are_borked
static FILE *logfp;
/** @brief Output device */
-const char *device;
/** @brief Minimum low watermark
*
/** @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' },
{ "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
#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 }
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)
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;
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) */
* 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 */
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;
}
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(<id, 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(<id, 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"
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) {
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;
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");
}
}
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)
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");
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(<id, 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(<id, 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;
}
~mdw / disorder / blobdiff