X-Git-Url: https://git.distorted.org.uk/~mdw/disorder/blobdiff_plain/ec57f6c97b41d54ade912f7e3b9f727b40e38e16..d76bbdea349bf3856172ce9599e657d566e91162:/lib/uaudio-rtp.c

diff --git a/lib/uaudio-rtp.c b/lib/uaudio-rtp.c
index ea1d1ca..31e8f57 100644
--- a/lib/uaudio-rtp.c
+++ b/lib/uaudio-rtp.c
@@ -23,6 +23,8 @@
 #include <sys/socket.h>
 #include <ifaddrs.h>
 #include <net/if.h>
+#include <arpa/inet.h>
+#include <netinet/in.h>
 #include <gcrypt.h>
 #include <unistd.h>
 #include <time.h>
@@ -36,6 +38,7 @@
 #include "addr.h"
 #include "ifreq.h"
 #include "timeval.h"
+#include "configuration.h"
 
 /** @brief Bytes to send per network packet
  *
@@ -143,17 +146,16 @@ static void rtp_open(void) {
   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])
@@ -164,10 +166,8 @@ static void rtp_open(void) {
     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);
@@ -184,10 +184,8 @@ static void rtp_open(void) {
     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,
@@ -283,7 +281,8 @@ static void rtp_start(uaudio_callback *callback,
                       rtp_play,
                       256 / uaudio_sample_size,
                       (NETWORK_BYTES - sizeof(struct rtp_header))
-                      / uaudio_sample_size);
+                      / uaudio_sample_size,
+                      0);
 }
 
 static void rtp_stop(void) {
@@ -301,13 +300,33 @@ static void rtp_deactivate(void) {
   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,
 };
 
 /*