* You should have received a copy of the GNU General Public License
* along with this program. If not, see <http://www.gnu.org/licenses/>.
*/
-/** @file lib/uaudio-oss.c
+/** @file lib/uaudio-rtp.c
* @brief Support for RTP network play backend */
#include "common.h"
-#include <pthread.h>
+#include <errno.h>
+#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 "uaudio.h"
#include "mem.h"
#include "log.h"
#include "syscalls.h"
+#include "rtp.h"
+#include "addr.h"
+#include "ifreq.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 RTP payload type */
+static int rtp_payload;
+
+/** @brief RTP output socket */
+static int rtp_fd;
+
+/** @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;
+
+/** @brief Network error count
+ *
+ * If too many errors occur in too short a time, we give up.
+ */
+static int rtp_errors;
+
+/** @brief Set while paused */
+static volatile int rtp_paused;
static const char *const rtp_options[] = {
+ "rtp-destination",
+ "rtp-destination-port",
+ "rtp-source",
+ "rtp-source-port",
+ "multicast-ttl",
+ "multicast-loop",
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, 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 = 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;
+ while(u < limit) {
+ *u = htons(*u);
+ ++u;
+ }
+#endif
+ vec[0].iov_base = (void *)&header;
+ vec[0].iov_len = sizeof header;
+ vec[1].iov_base = buffer;
+ vec[1].iov_len = nsamples * uaudio_sample_size;
+ const uint32_t timestamp = uaudio_schedule_sync();
+ header.timestamp = htonl(rtp_base + (uint32_t)timestamp);
+ /* If we're paused don't actually end a packet, we just pretend */
+ if(flags & UAUDIO_PAUSED) {
+ uaudio_schedule_sent(nsamples);
+ return nsamples;
+ }
+ 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;
+ } 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 rtp_open(void) {
+ struct addrinfo *res, *sres;
+ static const int one = 1;
+ int sndbuf, target_sndbuf = 131072;
+ socklen_t len;
+ struct netaddress dst[1], src[1];
+
+ /* Get configuration */
+ rtp_get_netconfig("rtp-destination-af",
+ "rtp-destination",
+ "rtp-destination-port",
+ dst);
+ rtp_get_netconfig("rtp-source-af",
+ "rtp-source",
+ "rtp-source-port",
+ src);
+ /* ...microseconds */
+
+ /* Resolve addresses */
+ 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 */
+ 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)) {
+ /* 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) {
+ 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");
+ if(setsockopt(rtp_fd, IPPROTO_IP, IP_MULTICAST_LOOP,
+ &loop, sizeof loop) < 0)
+ 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");
+ if(setsockopt(rtp_fd, IPPROTO_IP, IPV6_MULTICAST_LOOP,
+ &loop, sizeof loop) < 0)
+ 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",
+ format_sockaddr(res->ai_addr), 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;
+ }
+ 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)",
+ format_sockaddr(res->ai_addr), ifs->ifa_name);
+ } else
+ info("unicasting on %s", format_sockaddr(res->ai_addr));
+ }
+ /* 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);
+ /* 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",
+ format_sockaddr(sres->ai_addr));
+ if(connect(rtp_fd, res->ai_addr, res->ai_addrlen) < 0)
+ fatal(errno, "error connecting broadcast socket to %s",
+ format_sockaddr(res->ai_addr));
+}
+
static void rtp_start(uaudio_callback *callback,
void *userdata) {
- (void)callback;
- (void)userdata;
- /* TODO */
+ /* We only support L16 (but we do stereo and mono and will convert sign) */
+ if(uaudio_channels == 2
+ && uaudio_bits == 16
+ && uaudio_rate == 44100)
+ rtp_payload = 10;
+ else if(uaudio_channels == 1
+ && uaudio_bits == 16
+ && 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);
+ /* 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);
+ rtp_open();
+ uaudio_schedule_init();
+ uaudio_thread_start(callback,
+ userdata,
+ rtp_play,
+ 256 / uaudio_sample_size,
+ (NETWORK_BYTES - sizeof(struct rtp_header))
+ / uaudio_sample_size,
+ 0);
}
static void rtp_stop(void) {
- /* TODO */
+ uaudio_thread_stop();
+ close(rtp_fd);
+ rtp_fd = -1;
}
-static void rtp_activate(void) {
- /* TODO */
-}
+static void rtp_configure(void) {
+ char buffer[64];
-static void rtp_deactivate(void) {
- /* TODO */
+ 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");
}
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,
};
/*
~mdw / disorder / blobdiff