X-Git-Url: https://git.distorted.org.uk/~mdw/disorder/blobdiff_plain/4f8132b3d83dc46d4ab2321ef5b838da406bd100..401d3485a7923ae9b83b23399bee0ef8d19d0444:/lib/uaudio-rtp.c?ds=inline
diff --git a/lib/uaudio-rtp.c b/lib/uaudio-rtp.c
index 1bbfa9c..9980ba0 100644
--- a/lib/uaudio-rtp.c
+++ b/lib/uaudio-rtp.c
@@ -15,38 +15,377 @@
* You should have received a copy of the GNU General Public License
* along with this program. If not, see .
*/
-/** @file lib/uaudio-oss.c
+/** @file lib/uaudio-rtp.c
* @brief Support for RTP network play backend */
#include "common.h"
-#include
+#include
+#include
+#include
+#include
+#include
+#include
+#include
+#include
#include "uaudio.h"
#include "mem.h"
#include "log.h"
#include "syscalls.h"
+#include "rtp.h"
+#include "addr.h"
+#include "ifreq.h"
+#include "timeval.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 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;
static const char *const rtp_options[] = {
+ "rtp-destination",
+ "rtp-destination-port",
+ "rtp-source",
+ "rtp-source-port",
+ "multicast-ttl",
+ "multicast-loop",
+ "rtp-delay-threshold",
NULL
};
+static size_t rtp_play(void *buffer, size_t nsamples) {
+ struct rtp_header header;
+ struct iovec vec[2];
+ struct timeval now;
+
+ /* 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;
+#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;
+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);
+ }
+ }
+ header.timestamp = htonl((uint32_t)rtp_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;
+ } 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;
+}
+
+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;
+
+ /* 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 */
+
+ /* 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
+ 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 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;
+ 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",
+ 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;
+ }
+ 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);
+ }
+ /* 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", ssockname);
+ 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;
+}
+
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);
+ rtp_open();
+ uaudio_thread_start(callback,
+ userdata,
+ rtp_play,
+ 256 / uaudio_sample_size,
+ (NETWORK_BYTES - sizeof(struct rtp_header))
+ / uaudio_sample_size);
}
static void rtp_stop(void) {
- /* TODO */
+ uaudio_thread_stop();
+ close(rtp_fd);
+ rtp_fd = -1;
}
static void rtp_activate(void) {
- /* TODO */
+ rtp_reactivated = 1;
+ uaudio_thread_activate();
}
static void rtp_deactivate(void) {
- /* TODO */
+ uaudio_thread_deactivate();
}
const struct uaudio uaudio_rtp = {