#include <sys/socket.h>
#include <sys/uio.h>
#include <assert.h>
+#include <net/if.h>
+#include <ifaddrs.h>
+#include <errno.h>
#include "configuration.h"
#include "syscalls.h"
#include "addr.h"
#include "timeval.h"
#include "rtp.h"
+#include "ifreq.h"
#include "speaker-protocol.h"
#include "speaker.h"
socklen_t len;
char *sockname, *ssockname;
- /* Override sample format */
- config->sample_format.rate = 44100;
- config->sample_format.channels = 2;
- config->sample_format.bits = 16;
- config->sample_format.byte_format = AO_FMT_BIG;
res = get_address(&config->broadcast, &pref, &sockname);
if(!res) exit(-1);
if(config->broadcast_from.n) {
res->ai_socktype,
res->ai_protocol)) < 0)
fatal(errno, "error creating broadcast socket");
- if((res->ai_family == PF_INET
- && IN_MULTICAST(
- ntohl(((struct sockaddr_in *)res->ai_addr)->sin_addr.s_addr)
- ))
- || (res->ai_family == PF_INET6
- && IN6_IS_ADDR_MULTICAST(
- &((struct sockaddr_in6 *)res->ai_addr)->sin6_addr
- ))) {
+ if(multicast(res->ai_addr)) {
/* Multicasting */
switch(res->ai_family) {
case PF_INET: {
default:
fatal(0, "unsupported address family %d", res->ai_family);
}
+ info("multicasting on %s", sockname);
} else {
- /* Presumably just broadcasting */
- if(setsockopt(bfd, SOL_SOCKET, SO_BROADCAST, &one, sizeof one) < 0)
- fatal(errno, "error setting SO_BROADCAST on broadcast socket");
+ 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(bfd, 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);
}
len = sizeof sndbuf;
if(getsockopt(bfd, SOL_SOCKET, SO_SNDBUF,
fatal(errno, "error connecting broadcast socket to %s", sockname);
/* Select an SSRC */
gcry_randomize(&rtp_id, sizeof rtp_id, GCRY_STRONG_RANDOM);
- info("selected network backend, sending to %s", sockname);
}
/** @brief Play over the network */
static size_t network_play(size_t frames) {
struct rtp_header header;
struct iovec vec[2];
- size_t bytes = frames * device_bpf, written_frames;
+ size_t bytes = frames * bpf, written_frames;
int written_bytes;
/* We transmit using RTP (RFC3550) and attempt to conform to the internet
* AVT profile (RFC3551). */
/* Find the number of microseconds elapsed since rtp_time=0 */
delta = tvsub_us(now, rtp_time_0);
assert(delta <= UINT64_MAX / 88200);
- target_rtp_time = (delta * playing->format.rate
- * playing->format.channels) / 1000000;
+ target_rtp_time = (delta * config->sample_format.rate
+ * config->sample_format.channels) / 1000000;
/* Overflows at ~6 years uptime with 44100Hz stereo */
/* rtp_time is the number of samples we've played. NB that we play
if(bytes > NETWORK_BYTES - sizeof header) {
bytes = NETWORK_BYTES - sizeof header;
/* Always send a whole number of frames */
- bytes -= bytes % device_bpf;
+ bytes -= bytes % bpf;
}
/* "The RTP clock rate used for generating the RTP timestamp is independent
* of the number of channels and the encoding; it equals the number of
} else
audio_errors /= 2;
written_bytes -= sizeof (struct rtp_header);
- written_frames = written_bytes / device_bpf;
+ written_frames = written_bytes / bpf;
/* Advance RTP's notion of the time */
- rtp_time += written_frames * playing->format.channels;
+ rtp_time += written_frames * config->sample_format.channels;
return written_frames;
}
static int bfd_slot;
/** @brief Set up poll array for network play */
-static void network_beforepoll(void) {
+static void network_beforepoll(int *timeoutp) {
struct timeval now;
uint64_t target_us;
uint64_t target_rtp_time;
+ const int64_t samples_per_second = config->sample_format.rate
+ * config->sample_format.channels;
const int64_t samples_ahead = ((uint64_t)RTP_AHEAD_MS
- * config->sample_format.rate
- * config->sample_format.channels
+ * samples_per_second
/ 1000);
+ int64_t lead, ahead_ms;
/* If we're starting then initialize the base time */
if(!rtp_time)
target_rtp_time = (target_us * config->sample_format.rate
* config->sample_format.channels)
/ 1000000;
- if((int64_t)(rtp_time - target_rtp_time) < samples_ahead)
+ lead = rtp_time - target_rtp_time;
+ if(lead < samples_ahead)
+ /* We've not reached the desired lead, write as fast as we can */
bfd_slot = addfd(bfd, POLLOUT);
+ else {
+ /* We've reached the desired lead, we can afford to wait a bit even if the
+ * IP stack thinks it can accept more. */
+ ahead_ms = 1000 * (lead - samples_ahead) / samples_per_second;
+ if(ahead_ms < *timeoutp)
+ *timeoutp = ahead_ms;
+ }
}
/** @brief Process poll() results for network play */
const struct speaker_backend network_backend = {
BACKEND_NETWORK,
- FIXED_FORMAT,
+ 0,
network_init,
0, /* activate */
network_play,
~mdw / disorder / blobdiff