* 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 <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 "addr.h"
#include "ifreq.h"
#include "timeval.h"
+#include "configuration.h"
/** @brief Bytes to send per network packet
*
/** @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 RTP timestamp
- *
- * This is the timestamp that will be used on the next outbound packet.
- *
- * The timestamp in the packet header is only 32 bits wide. With 44100Hz
- * stereo, that only gives about half a day before wrapping, which is not
- * particularly convenient for certain debugging purposes. Therefore the
- * timestamp is maintained as a 64-bit integer, giving around six million years
- * before wrapping, and truncated to 32 bits when transmitting.
- */
-static uint64_t rtp_timestamp;
-
-/** @brief Actual time corresponding to @ref rtp_timestamp
- *
- * This is the time, on this machine, at which the sample at @ref rtp_timestamp
- * ought to be sent, interpreted as the time the last packet was sent plus the
- * time length of the packet. */
-static struct timeval rtp_timeval;
-
-/** @brief Set when we (re-)activate, to provoke timestamp resync */
-static int rtp_reactivated;
-
/** @brief Network error count
*
* If too many errors occur in too short a time, we give up.
*/
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 Set while paused */
+static volatile int rtp_paused;
static const char *const rtp_options[] = {
"rtp-destination",
"rtp-source-port",
"multicast-ttl",
"multicast-loop",
- "rtp-delay-threshold",
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))
+ 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];
- struct timeval now;
-
+
+#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_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;
-retry:
- xgettimeofday(&now, NULL);
- if(rtp_reactivated) {
- /* We've been deactivated for some unknown interval. We need to advance
- * rtp_timestamp to account for the dead air. */
- /* On the first run through we'll set the start time. */
- if(!rtp_timeval.tv_sec)
- rtp_timeval = now;
- /* See how much time we missed.
- *
- * This will be 0 on the first run through, in which case we'll not modify
- * anything.
- *
- * It'll be negative in the (rare) situation where the deactivation
- * interval is shorter than the last packet we sent. In this case we wait
- * for that much time and then return having sent no samples, which will
- * cause uaudio_play_thread_fn() to retry.
- *
- * In the normal case it will be positive.
- */
- const int64_t delay = tvsub_us(now, rtp_timeval); /* microseconds */
- if(delay < 0) {
- usleep(-delay);
- goto retry;
- }
- /* Advance the RTP timestamp to the present. With 44.1KHz stereo this will
- * overflow the intermediate value with a delay of a bit over 6 years.
- * This seems acceptable. */
- uint64_t update = (delay * uaudio_rate * uaudio_channels) / 1000000;
- /* Don't throw off channel synchronization */
- update -= update % uaudio_channels;
- /* We log nontrivial changes */
- if(update)
- info("advancing rtp_time by %"PRIu64" samples", update);
- rtp_timestamp += update;
- rtp_timeval = now;
- rtp_reactivated = 0;
- } else {
- /* Chances are we've been called right on the heels of the previous packet.
- * If we just sent packets as fast as we got audio data we'd get way ahead
- * of the player and some buffer somewhere would fill (or at least become
- * unreasonably large).
- *
- * First find out how far ahead of the target time we are.
- */
- const int64_t ahead = tvsub_us(now, rtp_timeval); /* microseconds */
- /* Only delay at all if we are nontrivially ahead. */
- if(ahead > rtp_delay_threshold) {
- /* Don't delay by the full amount */
- usleep(ahead - rtp_delay_threshold / 2);
- /* Refetch time (so we don't get out of step with reality) */
- xgettimeofday(&now, NULL);
- }
+ 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;
}
- header.timestamp = htonl((uint32_t)rtp_timestamp);
int written_bytes;
do {
written_bytes = writev(rtp_fd, vec, 2);
return 0;
} else
rtp_errors /= 2; /* gradual decay */
- written_bytes -= sizeof (struct rtp_header);
- size_t written_samples = written_bytes / uaudio_sample_size;
- /* rtp_timestamp and rtp_timestamp are supposed to refer to the first sample
- * of the next packet */
- rtp_timestamp += written_samples;
- const unsigned usec = (rtp_timeval.tv_usec
- + 1000000 * written_samples / (uaudio_rate
- * uaudio_channels));
- /* ...will only overflow 32 bits if one packet is more than about half an
- * hour long, which is not plausible. */
- rtp_timeval.tv_sec += usec / 1000000;
- rtp_timeval.tv_usec = usec % 1000000;
- return written_samples;
+ /* 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 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;
- const char *delay;
+ 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");
- dst.s[1] = uaudio_get("rtp-destination-port");
- src.n = 2;
- src.s = xcalloc(2, sizeof *dst.s);
- src.s[0] = uaudio_get("rtp-source");
- src.s[1] = uaudio_get("rtp-source-port");
- 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;
- if((delay = uaudio_get("rtp-delay-threshold")))
- rtp_delay_threshold = atoi(delay);
- else
- rtp_delay_threshold = 1000; /* microseconds */
+ 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 = 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(errno, "error creating broadcast socket");
if(multicast(res->ai_addr)) {
/* Enable multicast options */
- const char *ttls = uaudio_get("multicast-ttl");
- const int ttl = ttls ? atoi(ttls) : 1;
- const char *loops = uaudio_get("multicast-loop");
- const int loop = loops ? !strcmp(loops, "yes") : 1;
+ 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,
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);
- /* 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_sequence, sizeof rtp_sequence);
- gcry_create_nonce(&rtp_timestamp, sizeof rtp_timestamp);
- /* rtp_play() will spot this and choose an initial value */
- rtp_timeval.tv_sec = 0;
+ fatal(errno, "error connecting broadcast socket to %s",
+ format_sockaddr(res->ai_addr));
}
static void rtp_start(uaudio_callback *callback,
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);
+ / uaudio_sample_size,
+ 0);
}
static void rtp_stop(void) {
rtp_fd = -1;
}
-static void rtp_activate(void) {
- rtp_reactivated = 1;
- uaudio_thread_activate();
-}
+static void rtp_configure(void) {
+ char buffer[64];
-static void rtp_deactivate(void) {
- uaudio_thread_deactivate();
+ 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