lib/configuration.c, lib/uaudio-rtp.c: Allow configuring payload size.

This is useful in unicast (or multicast-routing?) situations where
packets will traverse networks with unusually small MTU.

diff --git a/doc/disorder_config.5.in b/doc/disorder_config.5.in
index e57e06ac246e5e9a956b3f32e52f8ec1db95885c..0822669a3e10848a8e0351654b63f4eb2ab774aa 100644 (file)
--- a/doc/disorder_config.5.in
+++ b/doc/disorder_config.5.in
@@ -686,6 +686,16 @@ The default value is
 This option is experimental,
 and may change or be removed in a future release.
 .TP
 This option is experimental,
 and may change or be removed in a future release.
 .TP

+.B rtp_max_payload \fBYTES\fR
+Don't send RTP packets with a UDP payload larger than
+.I BYTES
+(including the 12-byte RTP header).  If you know that you will be transmitting
+RTP over networks with an unusually low MTU size, then it is probably useful to
+set this option.
+.IP
+This option is experimental,
+and may change or be removed in a future release.
+.TP

 .B rtp_minbuffer \fIFRAMES\fR
 Set
 .BR disorder-playrtp (1)'s
 .B rtp_minbuffer \fIFRAMES\fR
 Set
 .BR disorder-playrtp (1)'s

diff --git a/lib/configuration.c b/lib/configuration.c
index d1f14e90b001ef28a1fb60d3ef2dae54c38acc74..a87957e4ea707f1ad1252f638174ad70281bfdec 100644 (file)
--- a/lib/configuration.c
+++ b/lib/configuration.c
@@ -1093,6 +1093,7 @@ static const struct conf conf[] = {
   { C(replay_min),       &type_integer,          validate_non_negative },
   { C(rtp_always_request), &type_boolean,       validate_any },
   { C(rtp_delay_threshold), &type_integer,       validate_positive },
   { C(replay_min),       &type_integer,          validate_non_negative },
   { C(rtp_always_request), &type_boolean,       validate_any },
   { C(rtp_delay_threshold), &type_integer,       validate_positive },

+  { C(rtp_max_payload),         &type_integer,          validate_positive },

   { C(rtp_maxbuffer),   &type_integer,          validate_non_negative },
   { C(rtp_minbuffer),   &type_integer,          validate_non_negative },
   { C(rtp_mode),         &type_string,           validate_any },
   { C(rtp_maxbuffer),   &type_integer,          validate_non_negative },
   { C(rtp_minbuffer),   &type_integer,          validate_non_negative },
   { C(rtp_mode),         &type_string,           validate_any },


@@ -1409,6 +1410,7 @@ static struct config *config_default(void) {
   c->listen.af = -1;
   c->connect.af = -1;
   c->rtp_mode = xstrdup("auto");
   c->listen.af = -1;
   c->connect.af = -1;
   c->rtp_mode = xstrdup("auto");

+  c->rtp_max_payload = -1;

   return c;
 }
 
   return c;
 }
 

diff --git a/lib/configuration.h b/lib/configuration.h
index 5c33047d125c4a5552bec3bf97bf305328155b7b..2b0db524361ae4a4996d171b8500162cfd6d0329 100644 (file)
--- a/lib/configuration.h
+++ b/lib/configuration.h
@@ -259,6 +259,16 @@ struct config {
   /** @brief Whether to loop back multicast packets */
   int multicast_loop;
 
   /** @brief Whether to loop back multicast packets */
   int multicast_loop;
 

+  /** @brief Maximum size of RTP payload to send
+   *
+   * This is the maximum number of bytes we pass to write(2); to determine
+   * actual packet sizes, add a UDP header and an IP header (and a link layer
+   * header if it's the link layer size you care about).
+   *
+   * Don't make this too big or arithmetic will start to overflow.
+   */
+  long rtp_max_payload;
+

   /** @brief Login lifetime in seconds */
   long cookie_login_lifetime;
 
   /** @brief Login lifetime in seconds */
   long cookie_login_lifetime;
 

diff --git a/lib/uaudio-rtp.c b/lib/uaudio-rtp.c
index 8559bb228dbaa7c97c15e9bb50c84ca250189111..73677d595daf4037e40bca043b1e7cd6f7bcfa62 100644 (file)
--- a/lib/uaudio-rtp.c
+++ b/lib/uaudio-rtp.c
@@ -41,15 +41,8 @@
 #include "timeval.h"
 #include "configuration.h"
 
 #include "timeval.h"
 #include "configuration.h"
 

-/** @brief Bytes to send per network packet
- *
- * This is the maximum number of bytes we pass to write(2); to determine actual
- * packet sizes, add a UDP header and an IP header (and a link layer header if
- * it's the link layer size you care about).
- *
- * Don't make this too big or arithmetic will start to overflow.
- */
-#define NETWORK_BYTES (1500-8/*UDP*/-40/*IP*/-8/*conservatism*/)
+/** @brief Bytes to send per network packet */
+static int rtp_max_payload;

 
 /** @brief RTP payload type */
 static int rtp_payload;
 
 /** @brief RTP payload type */
 static int rtp_payload;


@@ -107,6 +100,7 @@ static const char *const rtp_options[] = {
   "multicast-ttl",
   "multicast-loop",
   "rtp-mode",
   "multicast-ttl",
   "multicast-loop",
   "rtp-mode",

+  "rtp-max-payload",

   NULL
 };
 
   NULL
 };
 


@@ -326,6 +320,9 @@ static void rtp_open(void) {
         rtp_mode = RTP_UNICAST;
     }
   }
         rtp_mode = RTP_UNICAST;
     }
   }

+  rtp_max_payload = atoi(uaudio_get("rtp-max-payload", "-1"));
+  if(rtp_max_payload < 0)
+    rtp_max_payload = 1500 - 8/*UDP*/ - 40/*IP*/ - 8/*conservatism*/;

   /* Create the sockets */
   if(rtp_mode != RTP_REQUEST) {
     if((rtp_fd = socket(dres->ai_family,
   /* Create the sockets */
   if(rtp_mode != RTP_REQUEST) {
     if((rtp_fd = socket(dres->ai_family,


@@ -437,7 +434,7 @@ static void rtp_start(uaudio_callback *callback,
                       userdata,
                       rtp_play,
                       256 / uaudio_sample_size,
                       userdata,
                       rtp_play,
                       256 / uaudio_sample_size,

-                      (NETWORK_BYTES - sizeof(struct rtp_header))
+                      (rtp_max_payload - sizeof(struct rtp_header))

                       / uaudio_sample_size,
                       0);
   if(config->rtp_verbose)
                       / uaudio_sample_size,
                       0);
   if(config->rtp_verbose)


@@ -464,6 +461,8 @@ static void rtp_configure(void) {
   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->multicast_ttl);
   uaudio_set("multicast-ttl", buffer);
   uaudio_set("multicast-loop", config->multicast_loop ? "yes" : "no");

+  snprintf(buffer, sizeof buffer, "%ld", config->rtp_max_payload);
+  uaudio_set("rtp-max-payload", buffer);

   if(config->rtp_verbose)
     disorder_info("RTP: configured");
 }
   if(config->rtp_verbose)
     disorder_info("RTP: configured");
 }