#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 "addr.h"
#include "ifreq.h"
#include "timeval.h"
+#include "configuration.h"
/** @brief Bytes to send per network packet
*
NULL
};
+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))
+ 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) {
struct rtp_header header;
struct iovec vec[2];
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,
- };
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];
/* 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_get_netconfig("rtp-destination-af",
+ "rtp-destination",
+ "rtp-destination-port",
+ dst);
+ rtp_get_netconfig("rtp-source-af",
+ "rtp-source",
+ "rtp-source-port",
+ src);
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);
+ res = netaddress_resolve(dst, 0, IPPROTO_UDP);
+ if(!res)
+ exit(-1);
+ if(src->af != -1) {
+ sres = netaddress_resolve(src, 1, IPPROTO_UDP);
+ if(!sres)
+ exit(-1);
} else
sres = 0;
/* Create the socket */
fatal(0, "unsupported address family %d", res->ai_family);
}
info("multicasting on %s TTL=%d loop=%s",
- sockname, ttl, loop ? "yes" : "no");
+ format_sockaddr(res->ai_addr), ttl, loop ? "yes" : "no");
} else {
struct ifaddrs *ifs;
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);
+ info("broadcasting on %s (%s)",
+ format_sockaddr(res->ai_addr), ifs->ifa_name);
} else
- info("unicasting on %s", sockname);
+ info("unicasting on %s", format_sockaddr(res->ai_addr));
}
/* Enlarge the socket buffer */
len = sizeof sndbuf;
/* 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);
+ fatal(errno, "error binding broadcast socket to %s",
+ format_sockaddr(sres->ai_addr));
if(connect(rtp_fd, res->ai_addr, res->ai_addrlen) < 0)
- fatal(errno, "error connecting broadcast socket to %s", sockname);
+ fatal(errno, "error connecting broadcast socket to %s",
+ format_sockaddr(res->ai_addr));
}
static void rtp_start(uaudio_callback *callback,
rtp_play,
256 / uaudio_sample_size,
(NETWORK_BYTES - sizeof(struct rtp_header))
- / uaudio_sample_size);
+ / uaudio_sample_size,
+ 0);
}
static void rtp_stop(void) {
uaudio_thread_deactivate();
}
+static void rtp_configure(void) {
+ char buffer[64];
+
+ 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);
+}
+
const struct uaudio uaudio_rtp = {
.name = "rtp",
.options = rtp_options,
.start = rtp_start,
.stop = rtp_stop,
.activate = rtp_activate,
- .deactivate = rtp_deactivate
+ .deactivate = rtp_deactivate,
+ .configure = rtp_configure,
};
/*
~mdw / disorder / blobdiff