/*
* 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
#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 "timeval.h"
#include "configuration.h"
-/** @brief Bytes to send per network packet
- *
- * This is the maximum number of bytes we pass to write(2); to determine actual
- * packet sizes, add a UDP header and an IP header (and a link layer header if
- * it's the link layer size you care about).
- *
- * Don't make this too big or arithmetic will start to overflow.
- */
-#define NETWORK_BYTES (1500-8/*UDP*/-40/*IP*/-8/*conservatism*/)
+/** @brief Bytes to send per network packet */
+static int rtp_max_payload;
/** @brief RTP payload type */
static int rtp_payload;
-/** @brief RTP output socket */
-static int rtp_fd;
+/** @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;
*/
static int rtp_errors;
-/** @brief Delay threshold in microseconds
- *
- * rtp_play() never attempts to introduce a delay shorter than this.
- */
-static int64_t rtp_delay_threshold;
+/** @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-source-port",
"multicast-ttl",
"multicast-loop",
- "delay-threshold",
+ "rtp-mode",
+ "rtp-max-payload",
+ "rtp-mtu-discovery",
NULL
};
-static size_t rtp_play(void *buffer, size_t nsamples) {
+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 = (uaudio_schedule_reactivated ? 0x80 : 0x00) | rtp_payload;
+ 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;
vec[0].iov_len = sizeof header;
vec[1].iov_base = buffer;
vec[1].iov_len = nsamples * uaudio_sample_size;
- uaudio_schedule_synchronize();
- header.timestamp = htonl((uint32_t)uaudio_schedule_timestamp);
- int written_bytes;
- do {
- written_bytes = writev(rtp_fd, vec, 2);
- } while(written_bytes < 0 && errno == EINTR);
- if(written_bytes < 0) {
- error(errno, "error transmitting audio data");
- ++rtp_errors;
- if(rtp_errors == 10)
- fatal(0, "too many audio tranmission errors");
- return 0;
+ 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
- rtp_errors /= 2; /* gradual decay */
- written_bytes -= sizeof (struct rtp_header);
- const size_t written_samples = written_bytes / uaudio_sample_size;
- uaudio_schedule_update(written_samples);
- return written_samples;
+ disorder_info("default socket send buffer on %s socket is %d",
+ what, sndbuf);
}
static void rtp_open(void) {
- struct addrinfo *res, *sres;
- static const struct addrinfo pref = {
- .ai_flags = 0,
- .ai_family = PF_INET,
- .ai_socktype = SOCK_DGRAM,
- .ai_protocol = IPPROTO_UDP,
- };
- static const struct addrinfo prefbind = {
- .ai_flags = AI_PASSIVE,
- .ai_family = PF_INET,
- .ai_socktype = SOCK_DGRAM,
- .ai_protocol = IPPROTO_UDP,
- };
+ struct resolved *dres, *sres;
+ size_t ndres, nsres;
static const int one = 1;
- int sndbuf, target_sndbuf = 131072;
- socklen_t len;
- char *sockname, *ssockname;
- struct stringlist dst, src;
+ struct netaddress dst[1], src[1];
+ const char *mode;
+#ifdef IP_MTU_DISCOVER
+ const char *mtu_disc;
+ int opt;
+#endif
- /* Get configuration */
- dst.n = 2;
- dst.s = xcalloc(2, sizeof *dst.s);
- dst.s[0] = uaudio_get("rtp-destination", NULL);
- dst.s[1] = uaudio_get("rtp-destination-port", NULL);
- src.n = 2;
- src.s = xcalloc(2, sizeof *dst.s);
- src.s[0] = uaudio_get("rtp-source", NULL);
- src.s[1] = uaudio_get("rtp-source-port", NULL);
- if(!dst.s[0])
- fatal(0, "'rtp-destination' not set");
- if(!dst.s[1])
- fatal(0, "'rtp-destination-port' not set");
- if(src.s[0]) {
- if(!src.s[1])
- fatal(0, "'rtp-source-port' not set");
- src.n = 2;
- } else
- src.n = 0;
- rtp_delay_threshold = atoi(uaudio_get("rtp-delay-threshold", "1000"));
- /* ...microseconds */
-
- /* Resolve addresses */
- res = get_address(&dst, &pref, &sockname);
- if(!res) exit(-1);
- if(src.n) {
- sres = get_address(&src, &prefbind, &ssockname);
- if(!sres) exit(-1);
- } else
+ /* 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;
- /* Create the socket */
- if((rtp_fd = socket(res->ai_family,
- res->ai_socktype,
- res->ai_protocol)) < 0)
- fatal(errno, "error creating broadcast socket");
- if(multicast(res->ai_addr)) {
+ 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(res->ai_family) {
+ switch(dres->sa->sa_family) {
case PF_INET: {
if(setsockopt(rtp_fd, IPPROTO_IP, IP_MULTICAST_TTL,
&ttl, sizeof ttl) < 0)
- fatal(errno, "error setting IP_MULTICAST_TTL on multicast socket");
+ disorder_fatal(errno, "error setting IP_MULTICAST_TTL on multicast socket");
if(setsockopt(rtp_fd, IPPROTO_IP, IP_MULTICAST_LOOP,
&loop, sizeof loop) < 0)
- fatal(errno, "error setting IP_MULTICAST_LOOP on multicast socket");
+ 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)
- fatal(errno, "error setting IPV6_MULTICAST_HOPS on multicast socket");
+ disorder_fatal(errno, "error setting IPV6_MULTICAST_HOPS on multicast socket");
if(setsockopt(rtp_fd, IPPROTO_IP, IPV6_MULTICAST_LOOP,
&loop, sizeof loop) < 0)
- fatal(errno, "error setting IPV6_MULTICAST_LOOP on multicast socket");
+ disorder_fatal(errno, "error setting IPV6_MULTICAST_LOOP on multicast socket");
break;
}
default:
- fatal(0, "unsupported address family %d", res->ai_family);
- }
- info("multicasting on %s TTL=%d loop=%s",
- sockname, ttl, loop ? "yes" : "no");
- } else {
- struct ifaddrs *ifs;
-
- if(getifaddrs(&ifs) < 0)
- 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, res->ai_addr))
- break;
- ifs = ifs->ifa_next;
+ disorder_fatal(0, "unsupported address family %d", dres->sa->sa_family);
}
- if(ifs) {
- if(setsockopt(rtp_fd, SOL_SOCKET, SO_BROADCAST, &one, sizeof one) < 0)
- fatal(errno, "error setting SO_BROADCAST on broadcast socket");
- info("broadcasting on %s (%s)", sockname, ifs->ifa_name);
- } else
- info("unicasting on %s", sockname);
+ disorder_info("multicasting on %s TTL=%d loop=%s",
+ format_sockaddr(dres->sa), ttl, loop ? "yes" : "no");
+ break;
}
- /* Enlarge the socket buffer */
- len = sizeof sndbuf;
- if(getsockopt(rtp_fd, SOL_SOCKET, SO_SNDBUF,
- &sndbuf, &len) < 0)
- fatal(errno, "error getting SO_SNDBUF");
- if(target_sndbuf > sndbuf) {
- if(setsockopt(rtp_fd, SOL_SOCKET, SO_SNDBUF,
- &target_sndbuf, sizeof target_sndbuf) < 0)
- error(errno, "error setting SO_SNDBUF to %d", target_sndbuf);
- else
- info("changed socket send buffer size from %d to %d",
- sndbuf, target_sndbuf);
- } else
- info("default socket send buffer is %d",
- sndbuf);
+ 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(sres && bind(rtp_fd, sres->ai_addr, sres->ai_addrlen) < 0)
- fatal(errno, "error binding broadcast socket to %s", ssockname);
- if(connect(rtp_fd, res->ai_addr, res->ai_addrlen) < 0)
- fatal(errno, "error connecting broadcast socket to %s", sockname);
+ 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,
&& uaudio_rate == 44100)
rtp_payload = 11;
else
- fatal(0, "asked for %d/%d/%d 16/44100/1 and 16/44100/2",
- uaudio_bits, uaudio_rate, uaudio_channels);
+ 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,
- (NETWORK_BYTES - sizeof(struct rtp_header))
- / 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) {
uaudio_thread_stop();
- close(rtp_fd);
- rtp_fd = -1;
-}
-
-static void rtp_activate(void) {
- uaudio_schedule_reactivated = 1;
- uaudio_thread_activate();
-}
-
-static void rtp_deactivate(void) {
- uaudio_thread_deactivate();
+ 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_configure(void) {
char buffer[64];
- uaudio_set("rtp-destination", config->broadcast.s[0]);
- uaudio_set("rtp-destination-port", config->broadcast.s[1]);
- if(config->broadcast_from.n) {
- uaudio_set("rtp-source", config->broadcast_from.s[0]);
- uaudio_set("rtp-source-port", config->broadcast_from.s[0]);
- } else {
- uaudio_set("rtp-source", NULL);
- uaudio_set("rtp-source-port", NULL);
- }
+ 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_delay_threshold);
- uaudio_set("delay-threshold", buffer);
+ 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");
+}
+
+/** @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 = {
.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,
};
/*
~mdw / disorder / blobdiff