/*
* This file is part of DisOrder.
- * Copyright (C) 2007 Richard Kettlewell
+ * Copyright (C) 2007-2009 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
* systems. There is no support for Microsoft Windows yet, and that will in
* fact probably an entirely separate program.
*
- * The program runs (at least) three threads. listen_thread() is responsible
- * for reading RTP packets off the wire and adding them to the linked list @ref
- * received_packets, assuming they are basically sound. queue_thread() takes
- * packets off this linked list and adds them to @ref packets (an operation
- * which might be much slower due to contention for @ref lock).
+ * The program runs (at least) three threads:
+ *
+ * listen_thread() is responsible for reading RTP packets off the wire and
+ * adding them to the linked list @ref received_packets, assuming they are
+ * basically sound.
+ *
+ * queue_thread() takes packets off this linked list and adds them to @ref
+ * packets (an operation which might be much slower due to contention for @ref
+ * lock).
*
- * The main thread is responsible for actually playing audio. In ALSA this
- * means it waits until ALSA says it's ready for more audio which it then
- * plays. See @ref clients/playrtp-alsa.c.
+ * control_thread() accepts commands from Disobedience (or anything else).
*
- * In Core Audio the main thread is only responsible for starting and stopping
- * play: the system does the actual playback in its own private thread, and
- * calls adioproc() to fetch the audio data. See @ref
- * clients/playrtp-coreaudio.c.
+ * The main thread activates and deactivates audio playing via the @ref
+ * lib/uaudio.h API (which probably implies at least one further thread).
*
* Sometimes it happens that there is no audio available to play. This may
* because the server went away, or a packet was dropped, or the server
* - 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 <math.h>
#include "log.h"
#include "mem.h"
#include "client.h"
#include "playrtp.h"
#include "inputline.h"
-
-#define readahead linux_headers_are_borked
+#include "version.h"
+#include "uaudio.h"
/** @brief Obsolete synonym */
#ifndef IPV6_JOIN_GROUP
static FILE *logfp;
/** @brief Output device */
-const char *device;
-
-/** @brief Minimum low watermark
- *
- * We'll stop playing if there's only this many samples in the buffer. */
-unsigned minbuffer = 2 * 44100 / 10; /* 0.2 seconds */
-/** @brief Buffer high watermark
- *
- * We'll only start playing when this many samples are available. */
-static unsigned readahead = 2 * 2 * 44100;
+/** @brief Buffer low watermark in samples */
+unsigned minbuffer = 4 * (2 * 44100) / 10; /* 0.4 seconds */
-/** @brief Maximum buffer size
+/** @brief Maximum buffer size in samples
*
- * We'll stop reading from the network if we have this many samples. */
+ * We'll stop reading from the network if we have this many samples.
+ */
static unsigned maxbuffer;
/** @brief Received packets
/** @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' },
{ "device", required_argument, 0, 'D' },
{ "min", required_argument, 0, 'm' },
{ "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' },
+ { "command", required_argument, 0, 'e' },
+ { "pause-mode", required_argument, 0, 'P' },
{ "socket", required_argument, 0, 's' },
{ "config", required_argument, 0, 'C' },
+ { "monitor", no_argument, 0, 'M' },
{ 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)
pthread_cond_broadcast(&cond);
pthread_mutex_unlock(&lock);
}
+#if HAVE_STUPID_GCC44
+ return NULL;
+#endif
}
/** @brief Background thread collecting samples
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) */
fatal(0, "unsupported RTP payload type %d",
header.mpt & 0x7F);
}
+ /* See if packet is silent */
+ const uint16_t *s = p->samples_raw;
+ n = p->nsamples;
+ for(; n > 0; --n)
+ if(*s++)
+ break;
+ if(!n)
+ p->flags |= SILENT;
if(logfp)
fprintf(logfp, "sequence %u timestamp %"PRIx32" length %"PRIx32" end %"PRIx32"\n",
seq, timestamp, p->nsamples, timestamp + p->nsamples);
* 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 */
* Must be called with @ref lock held.
*/
void playrtp_fill_buffer(void) {
- while(nsamples)
+ /* Discard current buffer contents */
+ while(nsamples) {
+ //fprintf(stderr, "%8u/%u (%u) DROPPING\n", nsamples, maxbuffer, minbuffer);
drop_first_packet();
+ }
info("Buffering...");
- while(nsamples < readahead)
+ /* Wait until there's at least minbuffer samples available */
+ while(nsamples < minbuffer) {
+ //fprintf(stderr, "%8u/%u (%u) FILLING\n", nsamples, maxbuffer, minbuffer);
pthread_cond_wait(&cond, &lock);
+ }
+ /* Start from whatever is earliest */
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"
#if HAVE_COREAUDIO_AUDIOHARDWARE_H
" --core-audio, -c Use Core Audio to play audio\n"
#endif
+ " --command, -e COMMAND Pipe audio to command.\n"
+ " --pause-mode, -P silence For -e: pauses send silence (default)\n"
+ " --pause-mode, -P suspend For -e: pauses suspend writes\n"
" --help, -h Display usage message\n"
" --version, -V Display version number\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;
+ int silent = 0;
+
+ 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;
+ }
+ silent = !!(p->flags & SILENT);
+ } 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);
+ silent = 1;
+ }
+ /* 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;
+ /* If we're getting behind then try to drop just silent packets
+ *
+ * In theory this shouldn't be necessary. The server is supposed to send
+ * packets at the right rate and compares the number of samples sent with the
+ * time in order to ensure this.
+ *
+ * However, various things could throw this off:
+ *
+ * - the server's clock could advance at the wrong rate. This would cause it
+ * to mis-estimate the right number of samples to have sent and
+ * inappropriately throttle or speed up.
+ *
+ * - playback could happen at the wrong rate. If the playback host's sound
+ * card has a slightly incorrect clock then eventually it will get out
+ * of step.
+ *
+ * So if we play back slightly slower than the server sends for either of
+ * these reasons then eventually our buffer, and the socket's buffer, will
+ * fill, and the kernel will start dropping packets. The result is audible
+ * and not very nice.
+ *
+ * Therefore if we're getting behind, we pre-emptively drop silent packets,
+ * since a change in the duration of a silence is less noticeable than a
+ * dropped packet from the middle of continuous music.
+ *
+ * (If things go wrong the other way then eventually we run out of packets to
+ * play and are forced to play silence. This doesn't seem to happen in
+ * practice but if it does then in the same way we can artificially extend
+ * silent packets to compensate.)
+ *
+ * Dropped packets are always logged; use 'disorder-playrtp --monitor' to
+ * track how close to target buffer occupancy we are on a once-a-minute
+ * basis.
+ */
+ if(nsamples > minbuffer && silent) {
+ info("dropping %zu samples (%"PRIu32" > %"PRIu32")",
+ samples, nsamples, minbuffer);
+ samples = 0;
+ }
+ /* Junk obsolete packets */
+ playrtp_next_packet();
+ pthread_mutex_unlock(&lock);
+ return samples;
}
int main(int argc, char **argv) {
struct addrinfo *res;
struct stringlist sl;
char *sockname;
- int rcvbuf, target_rcvbuf = 131072;
+ int rcvbuf, target_rcvbuf = 0;
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;
+ pthread_t ltid;
+ int monitor = 0;
+ static const int one = 1;
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
};
+ /* Timing information is often important to debugging playrtp, so we include
+ * timestamps in the logs */
+ logdate = 1;
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:x:L:R:aocC:re:P:M", 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 'D': uaudio_set("device", optarg); break;
case 'm': minbuffer = 2 * atol(optarg); break;
- case 'b': readahead = 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;
+ case 'e': backend = &uaudio_command; uaudio_set("command", optarg); break;
+ case 'P': uaudio_set("pause-mode", optarg); break;
+ case 'M': monitor = 1; break;
default: fatal(0, "invalid option");
}
}
- if(config_read(0)) fatal(0, "cannot read configuration");
+ if(config_read(0, NULL)) fatal(0, "cannot read configuration");
if(!maxbuffer)
- maxbuffer = 4 * readahead;
+ maxbuffer = 2 * minbuffer;
argc -= optind;
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");
- 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) {
+ /* Allow multiple listeners */
+ xsetsockopt(rtpfd, SOL_SOCKET, SO_REUSEADDR, &one, sizeof one);
+ is_multicast = multicast(res->ai_addr);
+ /* The multicast and unicast/broadcast cases are different enough that they
+ * are totally split. Trying to find commonality between them causes more
+ * trouble that it's worth. */
+ if(is_multicast) {
+ /* Stash the multicast group address */
+ 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;
+ default:
+ fatal(0, "unsupported family %d", (int)res->ai_addr->sa_family);
+ }
+ /* Bind to to the multicast group address */
+ if(bind(rtpfd, res->ai_addr, res->ai_addrlen) < 0)
+ fatal(errno, "error binding socket to %s", format_sockaddr(res->ai_addr));
+ /* Add multicast group membership */
+ 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);
}
+ /* Report what we did */
+ info("listening on %s multicast group %s",
+ format_sockaddr(res->ai_addr), format_sockaddr(&mgroup.sa));
+ } else {
+ /* Bind to 0/port */
+ switch(res->ai_addr->sa_family) {
+ case AF_INET: {
+ struct sockaddr_in *in = (struct sockaddr_in *)res->ai_addr;
+
+ memset(&in->sin_addr, 0, sizeof (struct in_addr));
+ break;
+ }
+ case AF_INET6: {
+ struct sockaddr_in6 *in6 = (struct sockaddr_in6 *)res->ai_addr;
+
+ memset(&in6->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", format_sockaddr(res->ai_addr));
+ /* Report what we did */
+ info("listening on %s", format_sockaddr(res->ai_addr));
}
len = sizeof rcvbuf;
if(getsockopt(rtpfd, SOL_SOCKET, SO_RCVBUF, &rcvbuf, &len) < 0)
rcvbuf, target_rcvbuf);
} else
info("default socket receive buffer %d", rcvbuf);
+ //info("minbuffer %u maxbuffer %u", minbuffer, maxbuffer);
if(logfp)
info("WARNING: -L option can impact performance");
if(control_socket) {
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);
+ }
+ /* 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);
+ time_t lastlog = 0;
+ 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;
+ pthread_mutex_unlock(&lock);
+ backend->activate();
+ pthread_mutex_lock(&lock);
+ /* Wait until the buffer empties out
+ *
+ * If there's a packet that we can play right now then we definitely
+ * continue.
+ *
+ * Also if there's at least minbuffer samples we carry on regardless and
+ * insert silence. The assumption is there's been a pause but more data
+ * is now available.
+ */
+ while(nsamples >= minbuffer
+ || (nsamples > 0
+ && contains(pheap_first(&packets), next_timestamp))) {
+ if(monitor) {
+ time_t now = xtime(0);
+
+ if(now >= lastlog + 60) {
+ int offset = nsamples - minbuffer;
+ double offtime = (double)offset / (uaudio_rate * uaudio_channels);
+ info("%+d samples off (%d.%02ds, %d bytes)",
+ offset,
+ (int)fabs(offtime) * (offtime < 0 ? -1 : 1),
+ (int)(fabs(offtime) * 100) % 100,
+ offset * uaudio_bits / CHAR_BIT);
+ lastlog = now;
+ }
+ }
+ //fprintf(stderr, "%8u/%u (%u) PLAYING\n", nsamples, maxbuffer, minbuffer);
+ pthread_cond_wait(&cond, &lock);
+ }
+#if 0
+ if(nsamples) {
+ struct packet *p = pheap_first(&packets);
+ fprintf(stderr, "nsamples=%u (%u) next_timestamp=%"PRIx32", first packet is [%"PRIx32",%"PRIx32")\n",
+ nsamples, minbuffer, next_timestamp,p->timestamp,p->timestamp+p->nsamples);
+ }
+#endif
+ /* Stop playing for a bit until the buffer re-fills */
+ pthread_mutex_unlock(&lock);
+ backend->deactivate();
+ pthread_mutex_lock(&lock);
+ active = 0;
+ /* Go back round */
+ }
return 0;
}
~mdw / disorder / blobdiff