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further debugging info() calls; marked TODO to remove when bug fixed
[disorder]
/
server
/
speaker-network.c
diff --git
a/server/speaker-network.c
b/server/speaker-network.c
index
c81b7db
..
c8edbfe
100644
(file)
--- a/
server/speaker-network.c
+++ b/
server/speaker-network.c
@@
-32,6
+32,7
@@
#include <assert.h>
#include <net/if.h>
#include <ifaddrs.h>
#include <assert.h>
#include <net/if.h>
#include <ifaddrs.h>
+#include <errno.h>
#include "configuration.h"
#include "syscalls.h"
#include "configuration.h"
#include "syscalls.h"
@@
-106,11
+107,6
@@
static void network_init(void) {
socklen_t len;
char *sockname, *ssockname;
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 = get_address(&config->broadcast, &pref, &sockname);
if(!res) exit(-1);
if(config->broadcast_from.n) {
@@
-155,7
+151,11
@@
static void network_init(void) {
if(getifaddrs(&ifs) < 0)
fatal(errno, "error calling getifaddrs");
while(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)
if((ifs->ifa_flags & IFF_BROADCAST)
+ && ifs->ifa_broadaddr
&& sockaddr_equal(ifs->ifa_broadaddr, res->ai_addr))
break;
ifs = ifs->ifa_next;
&& sockaddr_equal(ifs->ifa_broadaddr, res->ai_addr))
break;
ifs = ifs->ifa_next;
@@
-189,14
+189,13
@@
static void network_init(void) {
fatal(errno, "error connecting broadcast socket to %s", sockname);
/* Select an SSRC */
gcry_randomize(&rtp_id, sizeof rtp_id, GCRY_STRONG_RANDOM);
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];
}
/** @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). */
int written_bytes;
/* We transmit using RTP (RFC3550) and attempt to conform to the internet
* AVT profile (RFC3551). */
@@
-212,8
+211,8
@@
static size_t network_play(size_t frames) {
/* Find the number of microseconds elapsed since rtp_time=0 */
delta = tvsub_us(now, rtp_time_0);
assert(delta <= UINT64_MAX / 88200);
/* 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
/* Overflows at ~6 years uptime with 44100Hz stereo */
/* rtp_time is the number of samples we've played. NB that we play
@@
-272,7
+271,7
@@
static size_t network_play(size_t frames) {
if(bytes > NETWORK_BYTES - sizeof header) {
bytes = NETWORK_BYTES - sizeof header;
/* Always send a whole number of frames */
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
}
/* "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
@@
-298,23
+297,25
@@
static size_t network_play(size_t frames) {
} else
audio_errors /= 2;
written_bytes -= sizeof (struct rtp_header);
} 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 */
/* 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 */
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;
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
const int64_t samples_ahead = ((uint64_t)RTP_AHEAD_MS
- * config->sample_format.rate
- * config->sample_format.channels
+ * samples_per_second
/ 1000);
/ 1000);
+ int64_t lead, ahead_ms;
/* If we're starting then initialize the base time */
if(!rtp_time)
/* If we're starting then initialize the base time */
if(!rtp_time)
@@
-327,8
+328,17
@@
static void network_beforepoll(void) {
target_rtp_time = (target_us * config->sample_format.rate
* config->sample_format.channels)
/ 1000000;
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);
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 */
}
/** @brief Process poll() results for network play */
@@
-341,7
+351,7
@@
static int network_ready(void) {
const struct speaker_backend network_backend = {
BACKEND_NETWORK,
const struct speaker_backend network_backend = {
BACKEND_NETWORK,
-
FIXED_FORMAT
,
+
0
,
network_init,
0, /* activate */
network_play,
network_init,
0, /* activate */
network_play,