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speaker protocol redesign to cope with libao re-opening
[disorder]
/
server
/
speaker-network.c
diff --git
a/server/speaker-network.c
b/server/speaker-network.c
index 5b1ccce0a9d02c486326d280c9563f963ecba8a0..59630ffc1bab747adf485a834a77f2d64861d634 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) {
@@
-200,7
+196,7
@@
static void network_init(void) {
static size_t network_play(size_t frames) {
struct rtp_header header;
struct iovec vec[2];
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). */
@@
-216,8
+212,8
@@
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
@@
-276,7
+272,7
@@
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
@@
-302,9
+298,9
@@
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;
}
return written_frames;
}
@@
-345,7
+341,7
@@
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,