/*
* This file is part of DisOrder.
* Copyright (C) 2007 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
* the Free Software Foundation; either version 2 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.
*
* 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
*/
/** @file clients/playrtp.c
* @brief RTP player
*
* This player supports Linux (ALSA)
* and Apple Mac (Core Audio)
* 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 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.
*
* 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.
*
* 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
* deliberately did not send any sound because it encountered a silence.
*
* Assumptions:
* - it is safe to read uint32_t values without a lock protecting them
*/
#include
#include "types.h"
#include
#include
#include
#include
#include
#include
#include
#include
#include
#include
#include
#include
#include
#include
#include
#include
#include
#include
#include
#include "log.h"
#include "mem.h"
#include "configuration.h"
#include "addr.h"
#include "syscalls.h"
#include "rtp.h"
#include "defs.h"
#include "vector.h"
#include "heap.h"
#include "timeval.h"
#include "client.h"
#include "playrtp.h"
#include "inputline.h"
#define readahead linux_headers_are_borked
/** @brief Obsolete synonym */
#ifndef IPV6_JOIN_GROUP
# define IPV6_JOIN_GROUP IPV6_ADD_MEMBERSHIP
#endif
/** @brief RTP socket */
static int rtpfd;
/** @brief Log output */
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 Maximum buffer size
*
* We'll stop reading from the network if we have this many samples. */
static unsigned maxbuffer;
/** @brief Received packets
* Protected by @ref receive_lock
*
* Received packets are added to this list, and queue_thread() picks them off
* it and adds them to @ref packets. Whenever a packet is added to it, @ref
* receive_cond is signalled.
*/
struct packet *received_packets;
/** @brief Tail of @ref received_packets
* Protected by @ref receive_lock
*/
struct packet **received_tail = &received_packets;
/** @brief Lock protecting @ref received_packets
*
* Only listen_thread() and queue_thread() ever hold this lock. It is vital
* that queue_thread() not hold it any longer than it strictly has to. */
pthread_mutex_t receive_lock = PTHREAD_MUTEX_INITIALIZER;
/** @brief Condition variable signalled when @ref received_packets is updated
*
* Used by listen_thread() to notify queue_thread() that it has added another
* packet to @ref received_packets. */
pthread_cond_t receive_cond = PTHREAD_COND_INITIALIZER;
/** @brief Length of @ref received_packets */
uint32_t nreceived;
/** @brief Binary heap of received packets */
struct pheap packets;
/** @brief Total number of samples available
*
* We make this volatile because we inspect it without a protecting lock,
* so the usual pthread_* guarantees aren't available.
*/
volatile uint32_t nsamples;
/** @brief Timestamp of next packet to play.
*
* This is set to the timestamp of the last packet, plus the number of
* samples it contained. Only valid if @ref active is nonzero.
*/
uint32_t next_timestamp;
/** @brief True if actively playing
*
* This is true when playing and false when just buffering. */
int active;
/** @brief Lock protecting @ref packets */
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;
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' },
{ "debug", no_argument, 0, 'd' },
{ "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' },
#if HAVE_SYS_SOUNDCARD_H || EMPEG_HOST
{ "oss", no_argument, 0, 'o' },
#endif
#if HAVE_ALSA_ASOUNDLIB_H
{ "alsa", no_argument, 0, 'a' },
#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 }
};
/** @brief Control thread
*
* This thread is responsible for accepting control commands from Disobedience
* (or other controllers) over an AF_UNIX stream socket with a path specified
* by the @c --socket option. The protocol uses simple string commands and
* replies:
*
* - @c stop will shut the player down
* - @c query will send back the reply @c running
* - anything else is ignored
*
* Commands and response strings terminated by shutting down the connection or
* by a newline. No attempt is made to multiplex multiple clients so it is
* important that the command be sent as soon as the connection is made - it is
* assumed that both parties to the protocol are entirely cooperating with one
* another.
*/
static void *control_thread(void attribute((unused)) *arg) {
struct sockaddr_un sa;
int sfd, cfd;
char *line;
socklen_t salen;
FILE *fp;
assert(control_socket);
unlink(control_socket);
memset(&sa, 0, sizeof sa);
sa.sun_family = AF_UNIX;
strcpy(sa.sun_path, control_socket);
sfd = xsocket(PF_UNIX, SOCK_STREAM, 0);
if(bind(sfd, (const struct sockaddr *)&sa, sizeof sa) < 0)
fatal(errno, "error binding to %s", control_socket);
if(listen(sfd, 128) < 0)
fatal(errno, "error calling listen on %s", control_socket);
info("listening on %s", control_socket);
for(;;) {
salen = sizeof sa;
cfd = accept(sfd, (struct sockaddr *)&sa, &salen);
if(cfd < 0) {
switch(errno) {
case EINTR:
case EAGAIN:
break;
default:
fatal(errno, "error calling accept on %s", control_socket);
}
}
if(!(fp = fdopen(cfd, "r+"))) {
error(errno, "error calling fdopen for %s connection", control_socket);
close(cfd);
continue;
}
if(!inputline(control_socket, fp, &line, '\n')) {
if(!strcmp(line, "stop")) {
info("stopped via %s", control_socket);
exit(0); /* terminate immediately */
}
if(!strcmp(line, "query"))
fprintf(fp, "running");
xfree(line);
}
if(fclose(fp) < 0)
error(errno, "error closing %s connection", control_socket);
}
}
/** @brief Drop the first packet
*
* Assumes that @ref lock is held.
*/
static void drop_first_packet(void) {
if(pheap_count(&packets)) {
struct packet *const p = pheap_remove(&packets);
nsamples -= p->nsamples;
playrtp_free_packet(p);
pthread_cond_broadcast(&cond);
}
}
/** @brief Background thread adding packets to heap
*
* This just transfers packets from @ref received_packets to @ref packets. It
* is important that it holds @ref receive_lock for as little time as possible,
* in order to minimize the interval between calls to read() in
* listen_thread().
*/
static void *queue_thread(void attribute((unused)) *arg) {
struct packet *p;
for(;;) {
/* Get the next packet */
pthread_mutex_lock(&receive_lock);
while(!received_packets)
pthread_cond_wait(&receive_cond, &receive_lock);
p = received_packets;
received_packets = p->next;
if(!received_packets)
received_tail = &received_packets;
--nreceived;
pthread_mutex_unlock(&receive_lock);
/* Add it to the heap */
pthread_mutex_lock(&lock);
pheap_insert(&packets, p);
nsamples += p->nsamples;
pthread_cond_broadcast(&cond);
pthread_mutex_unlock(&lock);
}
}
/** @brief Background thread collecting samples
*
* This function collects samples, perhaps converts them to the target format,
* and adds them to the packet list.
*
* It is crucial that the gap between successive calls to read() is as small as
* possible: otherwise packets will be dropped.
*
* We use a binary heap to ensure that the unavoidable effort is at worst
* logarithmic in the total number of packets - in fact if packets are mostly
* received in order then we will largely do constant work per packet since the
* newest packet will always be last.
*
* Of more concern is that we must acquire the lock on the heap to add a packet
* to it. If this proves a problem in practice then the answer would be
* (probably doubly) linked list with new packets added the end and a second
* thread which reads packets off the list and adds them to the heap.
*
* We keep memory allocation (mostly) very fast by keeping pre-allocated
* packets around; see @ref playrtp_new_packet().
*/
static void *listen_thread(void attribute((unused)) *arg) {
struct packet *p = 0;
int n;
struct rtp_header header;
uint16_t seq;
uint32_t timestamp;
struct iovec iov[2];
for(;;) {
if(!p)
p = playrtp_new_packet();
iov[0].iov_base = &header;
iov[0].iov_len = sizeof header;
iov[1].iov_base = p->samples_raw;
iov[1].iov_len = sizeof p->samples_raw / sizeof *p->samples_raw;
n = readv(rtpfd, iov, 2);
if(n < 0) {
switch(errno) {
case EINTR:
continue;
default:
fatal(errno, "error reading from socket");
}
}
/* Ignore too-short packets */
if((size_t)n <= sizeof (struct rtp_header)) {
info("ignored a short packet");
continue;
}
timestamp = htonl(header.timestamp);
seq = htons(header.seq);
/* Ignore packets in the past */
if(active && lt(timestamp, next_timestamp)) {
info("dropping old packet, timestamp=%"PRIx32" < %"PRIx32,
timestamp, next_timestamp);
continue;
}
p->next = 0;
p->flags = 0;
p->timestamp = timestamp;
/* Convert to target format */
if(header.mpt & 0x80)
p->flags |= IDLE;
switch(header.mpt & 0x7F) {
case 10:
p->nsamples = (n - sizeof header) / sizeof(uint16_t);
break;
/* TODO support other RFC3551 media types (when the speaker does) */
default:
fatal(0, "unsupported RTP payload type %d",
header.mpt & 0x7F);
}
if(logfp)
fprintf(logfp, "sequence %u timestamp %"PRIx32" length %"PRIx32" end %"PRIx32"\n",
seq, timestamp, p->nsamples, timestamp + p->nsamples);
/* Stop reading if we've reached the maximum.
*
* This is rather unsatisfactory: it means that if packets get heavily
* out of order then we guarantee dropouts. But for now... */
if(nsamples >= maxbuffer) {
pthread_mutex_lock(&lock);
while(nsamples >= maxbuffer)
pthread_cond_wait(&cond, &lock);
pthread_mutex_unlock(&lock);
}
/* Add the packet to the receive queue */
pthread_mutex_lock(&receive_lock);
*received_tail = p;
received_tail = &p->next;
++nreceived;
pthread_cond_signal(&receive_cond);
pthread_mutex_unlock(&receive_lock);
/* We'll need a new packet */
p = 0;
}
}
/** @brief Wait until the buffer is adequately full
*
* Must be called with @ref lock held.
*/
void playrtp_fill_buffer(void) {
while(nsamples)
drop_first_packet();
info("Buffering...");
while(nsamples < readahead)
pthread_cond_wait(&cond, &lock);
next_timestamp = pheap_first(&packets)->timestamp;
active = 1;
}
/** @brief Find next packet
* @return Packet to play or NULL if none found
*
* The return packet is merely guaranteed not to be in the past: it might be
* the first packet in the future rather than one that is actually suitable to
* play.
*
* Must be called with @ref lock held.
*/
struct packet *playrtp_next_packet(void) {
while(pheap_count(&packets)) {
struct packet *const p = pheap_first(&packets);
if(le(p->timestamp + p->nsamples, next_timestamp)) {
/* This packet is in the past. Drop it and try another one. */
drop_first_packet();
} else
/* This packet is NOT in the past. (It might be in the future
* however.) */
return p;
}
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"
"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"
" --config, -C PATH Set configuration file\n"
#if HAVE_ALSA_ASOUNDLIB_H
" --alsa, -a Use ALSA to play audio\n"
#endif
#if HAVE_SYS_SOUNDCARD_H || EMPEG_HOST
" --oss, -o Use OSS to play audio\n"
#endif
#if HAVE_COREAUDIO_AUDIOHARDWARE_H
" --core-audio, -c Use Core Audio to play audio\n"
#endif
" --help, -h Display usage message\n"
" --version, -V Display version number\n"
);
xfclose(stdout);
exit(0);
}
/* display version number and terminate */
static void version(void) {
xprintf("disorder-playrtp version %s\n", disorder_version_string);
xfclose(stdout);
exit(0);
}
int main(int argc, char **argv) {
int n, err;
struct addrinfo *res;
struct stringlist sl;
char *sockname;
int rcvbuf, target_rcvbuf = 131072;
socklen_t len;
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;
static const struct addrinfo prefs = {
AI_PASSIVE,
PF_INET,
SOCK_DGRAM,
IPPROTO_UDP,
0,
0,
0,
0
};
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:r", options, 0)) >= 0) {
switch(n) {
case 'h': help();
case 'V': version();
case 'd': debugging = 1; break;
case 'D': 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;
#if HAVE_ALSA_ASOUNDLIB_H
case 'a': backend = playrtp_alsa; break;
#endif
#if HAVE_SYS_SOUNDCARD_H || EMPEG_HOST
case 'o': backend = playrtp_oss; break;
#endif
#if HAVE_COREAUDIO_AUDIOHARDWARE_H
case 'c': backend = playrtp_coreaudio; break;
#endif
case 'C': configfile = optarg; break;
case 's': control_socket = optarg; break;
case 'r': dumpfile = optarg; break;
default: fatal(0, "invalid option");
}
}
if(config_read(0)) fatal(0, "cannot read configuration");
if(!maxbuffer)
maxbuffer = 4 * readahead;
argc -= optind;
argv += optind;
switch(argc) {
case 0:
/* 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 = 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]");
}
/* Look up address and port */
if(!(res = get_address(&sl, &prefs, &sockname)))
exit(1);
/* 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(is_multicast) {
switch(mgroup.sa.sa_family) {
case PF_INET:
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 = 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)
fatal(errno, "error calling setsockopt IPV6_JOIN_GROUP");
break;
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(target_rcvbuf > rcvbuf) {
if(setsockopt(rtpfd, SOL_SOCKET, SO_RCVBUF,
&target_rcvbuf, sizeof target_rcvbuf) < 0)
error(errno, "error calling setsockopt SO_RCVBUF %d",
target_rcvbuf);
/* We try to carry on anyway */
else
info("changed socket receive buffer from %d to %d",
rcvbuf, target_rcvbuf);
} else
info("default socket receive buffer %d", rcvbuf);
if(logfp)
info("WARNING: -L option can impact performance");
if(control_socket) {
pthread_t tid;
if((err = pthread_create(&tid, 0, control_thread, 0)))
fatal(err, "pthread_create control_thread");
}
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);
}
play_rtp();
return 0;
}
/*
Local Variables:
c-basic-offset:2
comment-column:40
fill-column:79
indent-tabs-mode:nil
End:
*/