#include "addr.h"
#include "ifreq.h"
#include "timeval.h"
+#include "configuration.h"
/** @brief Bytes to send per network packet
*
/** @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.
"rtp-source-port",
"multicast-ttl",
"multicast-loop",
- "rtp-delay-threshold",
+ "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;
+ header.mpt = (uaudio_schedule_reactivated ? 0x80 : 0x00) | rtp_payload;
#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);
- }
- }
- header.timestamp = htonl((uint32_t)rtp_timestamp);
+ uaudio_schedule_synchronize();
+ header.timestamp = htonl((uint32_t)uaudio_schedule_timestamp);
int written_bytes;
do {
written_bytes = writev(rtp_fd, vec, 2);
} 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;
+ const size_t written_samples = written_bytes / uaudio_sample_size;
+ uaudio_schedule_update(written_samples);
return written_samples;
}
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");
+ 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");
- src.s[1] = uaudio_get("rtp-source-port");
+ 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])
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_delay_threshold = atoi(uaudio_get("rtp-delay-threshold", "1000"));
+ /* ...microseconds */
/* Resolve addresses */
res = get_address(&dst, &pref, &sockname);
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(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,
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_sequence, sizeof rtp_sequence);
rtp_open();
+ uaudio_schedule_init();
uaudio_thread_start(callback,
userdata,
rtp_play,
}
static void rtp_activate(void) {
- rtp_reactivated = 1;
+ uaudio_schedule_reactivated = 1;
uaudio_thread_activate();
}
uaudio_thread_deactivate();
}
+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);
+ }
+ 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