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gst/rtsp/: Small cleanups. when multicast is selected as the transport, create UDP sources and connect to the multica...

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Original commit message from CVS:
Based on patch by: Thijs Vermeir <thijs dot vermeir at barco dot com>
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_set_state),
(gst_rtspsrc_media_to_caps), (gst_rtspsrc_stream_setup_rtp),
(gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_open),
(gst_rtspsrc_close), (gst_rtspsrc_play), (gst_rtspsrc_pause):
* gst/rtsp/rtspconnection.c: (inet_aton):
Small cleanups.
when multicast is selected as the transport, create UDP sources and
connect to the multicast group.
Move parsing and setting of caps to a common place.
Fixes #349894.
This commit is contained in:
Thijs Vermeir 2006-09-18 08:59:17 +00:00 committed by Wim Taymans
parent af06a16852
commit 7484c92dfe
3 changed files with 131 additions and 85 deletions
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17
ChangeLog
View file

@ -1,3 +1,18 @@
2006-09-18 Wim Taymans <wim@fluendo.com>
Based on patch by: Thijs Vermeir <thijs dot vermeir at barco dot com>
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_set_state),
(gst_rtspsrc_media_to_caps), (gst_rtspsrc_stream_setup_rtp),
(gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_open),
(gst_rtspsrc_close), (gst_rtspsrc_play), (gst_rtspsrc_pause):
* gst/rtsp/rtspconnection.c: (inet_aton):
Small cleanups.
when multicast is selected as the transport, create UDP sources and
connect to the multicast group.
Move parsing and setting of caps to a common place.
Fixes #349894.
2006-09-17 Stefan Kost <ensonic@users.sf.net> 2006-09-17 Stefan Kost <ensonic@users.sf.net>
* ext/flac/gstflactag.c: * ext/flac/gstflactag.c:
@ -20,7 +35,7 @@
* gst/videofilter/gstvideotemplate.c: * gst/videofilter/gstvideotemplate.c:
* gst/videomixer/videomixer.c: * gst/videomixer/videomixer.c:
* sys/sunaudio/gstsunaudiosrc.h: * sys/sunaudio/gstsunaudiosrc.h:
More G_OBJECT macro fixing. More G_OBJECT macro fixing.
2006-09-16 Wim Taymans <wim@fluendo.com> 2006-09-16 Wim Taymans <wim@fluendo.com>

193
gst/rtsp/gstrtspsrc.c
View file

@ -35,7 +35,7 @@
* rtspsrc currently understands SDP as the format of the session description. * rtspsrc currently understands SDP as the format of the session description.
* For each stream listed in the SDP a new rtp_stream%d pad will be created * For each stream listed in the SDP a new rtp_stream%d pad will be created
* with caps derived from the SDP media description. This is a caps of mime type * with caps derived from the SDP media description. This is a caps of mime type
* "application/x-rtp" that can be connected to any available rtp depayloader * "application/x-rtp" that can be connected to any available RTP depayloader
* element. * element.
* </para> * </para>
* <para> * <para>
@ -53,7 +53,7 @@
* <programlisting> * <programlisting>
* gst-launch rtspsrc location=rtsp://some.server/url ! fakesink * gst-launch rtspsrc location=rtsp://some.server/url ! fakesink
* </programlisting> * </programlisting>
* Establish a connection to an RTSP server and send the stream to a fakesink. * Establish a connection to an RTSP server and send the raw RTP packets to a fakesink.
* </para> * </para>
* </refsect2> * </refsect2>
* *
@ -370,26 +370,22 @@ gst_rtspsrc_set_state (GstRTSPSrc * src, GstState state)
stream = (GstRTSPStream *) streams->data; stream = (GstRTSPStream *) streams->data;
/* first our rtp session manager */ /* first our RTP session manager */
if (stream->rtpdec) { if (stream->rtpdec) {
if ((ret = ret = gst_element_set_state (stream->rtpdec, state);
gst_element_set_state (stream->rtpdec, if (ret == GST_STATE_CHANGE_FAILURE)
state)) == GST_STATE_CHANGE_FAILURE)
goto done; goto done;
} }
/* then our sources */ /* then our sources */
if (stream->rtpsrc) { if (stream->rtpsrc) {
if ((ret = ret = gst_element_set_state (stream->rtpsrc, state);
gst_element_set_state (stream->rtpsrc, if (ret == GST_STATE_CHANGE_FAILURE)
state)) == GST_STATE_CHANGE_FAILURE)
goto done; goto done;
} }
if (stream->rtcpsrc) { if (stream->rtcpsrc) {
if ((ret = ret = gst_element_set_state (stream->rtcpsrc, state);
gst_element_set_state (stream->rtcpsrc, if (ret == GST_STATE_CHANGE_FAILURE)
state)) == GST_STATE_CHANGE_FAILURE)
goto done; goto done;
} }
} }
@ -469,7 +465,7 @@ gst_rtspsrc_parse_rtpmap (gchar * rtpmap, gint * payload, gchar ** name,
/* /*
* Mapping of caps to and from SDP fields: * Mapping of caps to and from SDP fields:
* *
* m=<media> <udp port> RTP/AVP <payload> * m=<media> <UDP port> RTP/AVP <payload>
* a=rtpmap:<payload> <encoding_name>/<clock_rate>[/<encoding_params>] * a=rtpmap:<payload> <encoding_name>/<clock_rate>[/<encoding_params>]
* a=fmtp:<payload> <param>[=<value>];... * a=fmtp:<payload> <param>[=<value>];...
*/ */
@ -493,14 +489,14 @@ gst_rtspsrc_media_to_caps (SDPMedia * media)
} }
pt = atoi (payload); pt = atoi (payload);
/* dynamic payloads need rtpmap */
if (pt >= 96) { if (pt >= 96) {
gint payload = 0; gint payload = 0;
gboolean ret; gboolean ret;
if ((rtpmap = sdp_media_get_attribute_val (media, "rtpmap"))) { if ((rtpmap = sdp_media_get_attribute_val (media, "rtpmap"))) {
if ((ret = ret = gst_rtspsrc_parse_rtpmap (rtpmap, &payload, &name, &rate, &params);
gst_rtspsrc_parse_rtpmap (rtpmap, &payload, &name, &rate, if (ret) {
&params))) {
if (payload != pt) { if (payload != pt) {
g_warning ("rtpmap of wrong payload type"); g_warning ("rtpmap of wrong payload type");
name = NULL; name = NULL;
@ -511,7 +507,7 @@ gst_rtspsrc_media_to_caps (SDPMedia * media)
g_warning ("error parsing rtpmap"); g_warning ("error parsing rtpmap");
} }
} else { } else {
g_warning ("rtpmap type not given fot dynamic payload %d", pt); g_warning ("rtpmap type not given for dynamic payload %d", pt);
return NULL; return NULL;
} }
} }
@ -576,30 +572,29 @@ gst_rtspsrc_stream_setup_rtp (GstRTSPStream * stream, SDPMedia * media,
{ {
GstStateChangeReturn ret; GstStateChangeReturn ret;
GstRTSPSrc *src; GstRTSPSrc *src;
GstCaps *caps; GstElement *tmp, *rtpsrc, *rtcpsrc;
GstElement *tmp, *rtp, *rtcp;
gint tmp_rtp, tmp_rtcp; gint tmp_rtp, tmp_rtcp;
guint count; guint count;
src = stream->parent; src = stream->parent;
tmp = NULL; tmp = NULL;
rtp = NULL; rtpsrc = NULL;
rtcp = NULL; rtcpsrc = NULL;
count = 0; count = 0;
/* try to allocate 2 udp ports, the RTP port should be an even /* try to allocate 2 UDP ports, the RTP port should be an even
* number and the RTCP port should be the next (uneven) port */ * number and the RTCP port should be the next (uneven) port */
again: again:
rtp = gst_element_make_from_uri (GST_URI_SRC, "udp://0.0.0.0:0", NULL); rtpsrc = gst_element_make_from_uri (GST_URI_SRC, "udp://0.0.0.0:0", NULL);
if (rtp == NULL) if (rtpsrc == NULL)
goto no_udp_rtp_protocol; goto no_udp_rtp_protocol;
ret = gst_element_set_state (rtp, GST_STATE_PAUSED); ret = gst_element_set_state (rtpsrc, GST_STATE_PAUSED);
if (ret == GST_STATE_CHANGE_FAILURE) if (ret == GST_STATE_CHANGE_FAILURE)
goto start_rtp_failure; goto start_rtp_failure;
g_object_get (G_OBJECT (rtp), "port", &tmp_rtp, NULL); g_object_get (G_OBJECT (rtpsrc), "port", &tmp_rtp, NULL);
GST_DEBUG_OBJECT (src, "got RTP port %d", tmp_rtp); GST_DEBUG_OBJECT (src, "got RTP port %d", tmp_rtp);
/* check if port is even */ /* check if port is even */
@ -616,7 +611,7 @@ again:
gst_element_set_state (tmp, GST_STATE_NULL); gst_element_set_state (tmp, GST_STATE_NULL);
gst_object_unref (tmp); gst_object_unref (tmp);
} }
tmp = rtp; tmp = rtpsrc;
GST_DEBUG_OBJECT (src, "retry %d", count); GST_DEBUG_OBJECT (src, "retry %d", count);
goto again; goto again;
} }
@ -628,40 +623,35 @@ again:
} }
/* allocate port+1 for RTCP now */ /* allocate port+1 for RTCP now */
rtcp = gst_element_make_from_uri (GST_URI_SRC, "udp://0.0.0.0", NULL); rtcpsrc = gst_element_make_from_uri (GST_URI_SRC, "udp://0.0.0.0", NULL);
if (rtcp == NULL) if (rtcpsrc == NULL)
goto no_udp_rtcp_protocol; goto no_udp_rtcp_protocol;
/* set port */ /* set port */
tmp_rtcp = tmp_rtp + 1; tmp_rtcp = tmp_rtp + 1;
g_object_set (G_OBJECT (rtcp), "port", tmp_rtcp, NULL); g_object_set (G_OBJECT (rtcpsrc), "port", tmp_rtcp, NULL);
GST_DEBUG_OBJECT (src, "starting RTCP on port %d", tmp_rtcp); GST_DEBUG_OBJECT (src, "starting RTCP on port %d", tmp_rtcp);
ret = gst_element_set_state (rtcp, GST_STATE_PAUSED); ret = gst_element_set_state (rtcpsrc, GST_STATE_PAUSED);
/* FIXME, this could fail if the next port is not free, we /* FIXME, this could fail if the next port is not free, we
* should retry with another port then */ * should retry with another port then */
if (ret == GST_STATE_CHANGE_FAILURE) if (ret == GST_STATE_CHANGE_FAILURE)
goto start_rtcp_failure; goto start_rtcp_failure;
/* all fine, do port check */ /* all fine, do port check */
g_object_get (G_OBJECT (rtp), "port", rtpport, NULL); g_object_get (G_OBJECT (rtpsrc), "port", rtpport, NULL);
g_object_get (G_OBJECT (rtcp), "port", rtcpport, NULL); g_object_get (G_OBJECT (rtcpsrc), "port", rtcpport, NULL);
/* this should not happen */ /* this should not happen */
if (*rtpport != tmp_rtp || *rtcpport != tmp_rtcp) if (*rtpport != tmp_rtp || *rtcpport != tmp_rtcp)
goto port_error; goto port_error;
/* we manage these elements */ /* we manage these elements, we set the caps in configure_transport */
stream->rtpsrc = rtp; stream->rtpsrc = rtpsrc;
gst_rtspsrc_add_element (src, stream->rtpsrc); gst_rtspsrc_add_element (src, stream->rtpsrc);
stream->rtcpsrc = rtcp; stream->rtcpsrc = rtcpsrc;
gst_rtspsrc_add_element (src, stream->rtcpsrc); gst_rtspsrc_add_element (src, stream->rtcpsrc);
caps = gst_rtspsrc_media_to_caps (media);
/* set caps */
g_object_set (G_OBJECT (stream->rtpsrc), "caps", caps, NULL);
return TRUE; return TRUE;
/* ERRORS */ /* ERRORS */
@ -703,13 +693,13 @@ cleanup:
gst_element_set_state (tmp, GST_STATE_NULL); gst_element_set_state (tmp, GST_STATE_NULL);
gst_object_unref (tmp); gst_object_unref (tmp);
} }
if (rtp) { if (rtpsrc) {
gst_element_set_state (rtp, GST_STATE_NULL); gst_element_set_state (rtpsrc, GST_STATE_NULL);
gst_object_unref (rtp); gst_object_unref (rtpsrc);
} }
if (rtcp) { if (rtcpsrc) {
gst_element_set_state (rtcp, GST_STATE_NULL); gst_element_set_state (rtcpsrc, GST_STATE_NULL);
gst_object_unref (rtcp); gst_object_unref (rtcpsrc);
} }
return FALSE; return FALSE;
} }
@ -734,9 +724,8 @@ gst_rtspsrc_stream_configure_transport (GstRTSPStream * stream,
/* we manage this element */ /* we manage this element */
gst_rtspsrc_add_element (src, stream->rtpdec); gst_rtspsrc_add_element (src, stream->rtpdec);
if ((ret = ret = gst_element_set_state (stream->rtpdec, GST_STATE_PAUSED);
gst_element_set_state (stream->rtpdec, if (ret != GST_STATE_CHANGE_SUCCESS)
GST_STATE_PAUSED)) != GST_STATE_CHANGE_SUCCESS)
goto start_rtpdec_failure; goto start_rtpdec_failure;
stream->rtpdecrtp = gst_element_get_pad (stream->rtpdec, "sinkrtp"); stream->rtpdecrtp = gst_element_get_pad (stream->rtpdec, "sinkrtp");
@ -745,17 +734,55 @@ gst_rtspsrc_stream_configure_transport (GstRTSPStream * stream,
if (transport->lower_transport == RTSP_LOWER_TRANS_TCP) { if (transport->lower_transport == RTSP_LOWER_TRANS_TCP) {
/* configure for interleaved delivery, nothing needs to be done /* configure for interleaved delivery, nothing needs to be done
* here, the loop function will call the chain functions of the * here, the loop function will call the chain functions of the
* rtp session manager. */ * RTP session manager. */
stream->rtpchannel = transport->interleaved.min; stream->rtpchannel = transport->interleaved.min;
stream->rtcpchannel = transport->interleaved.max; stream->rtcpchannel = transport->interleaved.max;
GST_DEBUG ("stream %p on channels %d-%d", stream, GST_DEBUG ("stream %p on channels %d-%d", stream,
stream->rtpchannel, stream->rtcpchannel); stream->rtpchannel, stream->rtcpchannel);
/* also store the caps in the stream */ /* also store the caps in the stream, we need this when setting caps on
* outgoing buffers */
stream->caps = gst_rtspsrc_media_to_caps (media); stream->caps = gst_rtspsrc_media_to_caps (media);
} else { } else {
/* configure for UDP delivery, we need to connect the udp pads to /* multicast was selected, create UDP sources and connect to the multicast
* the rtp session plugin. */ * group. */
if (transport->multicast) {
gchar *uri;
/* creating RTP source */
uri =
g_strdup_printf ("udp://%s:%d", transport->destination,
transport->port.min);
stream->rtpsrc = gst_element_make_from_uri (GST_URI_SRC, uri, NULL);
g_free (uri);
if (stream->rtpsrc == NULL)
goto no_element;
/* creating RTCP source */
uri =
g_strdup_printf ("udp://%s:%d", transport->destination,
transport->port.max);
stream->rtcpsrc = gst_element_make_from_uri (GST_URI_SRC, uri, NULL);
g_free (uri);
if (stream->rtcpsrc == NULL)
goto no_element;
/* change state */
gst_element_set_state (stream->rtpsrc, GST_STATE_PAUSED);
gst_element_set_state (stream->rtcpsrc, GST_STATE_PAUSED);
/* we manage these elements */
gst_rtspsrc_add_element (src, stream->rtpsrc);
gst_rtspsrc_add_element (src, stream->rtcpsrc);
}
/* configure caps on the RTP source element */
stream->caps = gst_rtspsrc_media_to_caps (media);
g_object_set (G_OBJECT (stream->rtpsrc), "caps", stream->caps, NULL);
/* configure for UDP delivery, we need to connect the UDP pads to
* the RTP session plugin. */
pad = gst_element_get_pad (stream->rtpsrc, "src"); pad = gst_element_get_pad (stream->rtpsrc, "src");
gst_pad_link (pad, stream->rtpdecrtp); gst_pad_link (pad, stream->rtpdecrtp);
gst_object_unref (pad); gst_object_unref (pad);
@ -1008,9 +1035,8 @@ gst_rtspsrc_open (GstRTSPSrc * src)
/* create OPTIONS */ /* create OPTIONS */
GST_DEBUG_OBJECT (src, "create options..."); GST_DEBUG_OBJECT (src, "create options...");
if ((res = res = rtsp_message_init_request (RTSP_OPTIONS, src->location, &request);
rtsp_message_init_request (RTSP_OPTIONS, src->location, if (res < 0)
&request)) < 0)
goto create_request_failed; goto create_request_failed;
/* send OPTIONS */ /* send OPTIONS */
@ -1067,11 +1093,11 @@ gst_rtspsrc_open (GstRTSPSrc * src)
/* create DESCRIBE */ /* create DESCRIBE */
GST_DEBUG_OBJECT (src, "create describe..."); GST_DEBUG_OBJECT (src, "create describe...");
if ((res = res = rtsp_message_init_request (RTSP_DESCRIBE, src->location, &request);
rtsp_message_init_request (RTSP_DESCRIBE, src->location, if (res < 0)
&request)) < 0)
goto create_request_failed; goto create_request_failed;
/* we accept SDP for now */
/* we only accept SDP for now */
rtsp_message_add_header (&request, RTSP_HDR_ACCEPT, "application/sdp"); rtsp_message_add_header (&request, RTSP_HDR_ACCEPT, "application/sdp");
/* send DESCRIBE */ /* send DESCRIBE */
@ -1092,7 +1118,7 @@ gst_rtspsrc_open (GstRTSPSrc * src)
} }
} }
/* parse SDP */ /* get message body and parse as SDP */
rtsp_message_get_body (&response, &data, &size); rtsp_message_get_body (&response, &data, &size);
GST_DEBUG_OBJECT (src, "parse sdp..."); GST_DEBUG_OBJECT (src, "parse sdp...");
@ -1102,8 +1128,10 @@ gst_rtspsrc_open (GstRTSPSrc * src)
if (src->debug) if (src->debug)
sdp_message_dump (&sdp); sdp_message_dump (&sdp);
/* we allow all configured protocols */ /* we initially allow all configured protocols. based on the replies from the
* server we narrow them down. */
protocols = src->protocols; protocols = src->protocols;
/* setup streams */ /* setup streams */
{ {
gint i; gint i;
@ -1135,14 +1163,12 @@ gst_rtspsrc_open (GstRTSPSrc * src)
} }
GST_DEBUG_OBJECT (src, "setup %s", setup_url); GST_DEBUG_OBJECT (src, "setup %s", setup_url);
/* create SETUP request */ /* create SETUP request */
if ((res = res = rtsp_message_init_request (RTSP_SETUP, setup_url, &request);
rtsp_message_init_request (RTSP_SETUP, setup_url,
&request)) < 0) {
g_free (setup_url);
goto create_request_failed;
}
g_free (setup_url); g_free (setup_url);
if (res < 0)
goto create_request_failed;
transports = g_strdup (""); transports = g_strdup ("");
if (protocols & GST_RTSP_PROTO_UDP_UNICAST) { if (protocols & GST_RTSP_PROTO_UDP_UNICAST) {
@ -1150,7 +1176,7 @@ gst_rtspsrc_open (GstRTSPSrc * src)
gint rtpport, rtcpport; gint rtpport, rtcpport;
gchar *trxparams; gchar *trxparams;
/* allocate two udp ports */ /* allocate two UDP ports */
if (!gst_rtspsrc_stream_setup_rtp (stream, media, &rtpport, &rtcpport)) if (!gst_rtspsrc_stream_setup_rtp (stream, media, &rtpport, &rtcpport))
goto setup_rtp_failed; goto setup_rtp_failed;
@ -1167,6 +1193,9 @@ gst_rtspsrc_open (GstRTSPSrc * src)
GST_DEBUG_OBJECT (src, "setting up MULTICAST"); GST_DEBUG_OBJECT (src, "setting up MULTICAST");
/* we don't hav to allocate any UDP ports yet, if the selected transport
* turns out to be multicast we can create them and join the multicast
* group indicated in the transport reply */
new = new =
g_strconcat (transports, transports[0] ? "," : "", g_strconcat (transports, transports[0] ? "," : "",
"RTP/AVP/UDP;multicast", NULL); "RTP/AVP/UDP;multicast", NULL);
@ -1203,18 +1232,21 @@ gst_rtspsrc_open (GstRTSPSrc * src)
/* parse transport */ /* parse transport */
rtsp_transport_parse (resptrans, &transport); rtsp_transport_parse (resptrans, &transport);
/* update allowed transports for other streams */
/* update allowed transports for other streams. once the transport of
* one stream has been determined, we make sure that all other streams
* are configured in the same way */
if (transport.lower_transport == RTSP_LOWER_TRANS_TCP) { if (transport.lower_transport == RTSP_LOWER_TRANS_TCP) {
GST_DEBUG_OBJECT (src, "stream %d as TCP", i); GST_DEBUG_OBJECT (src, "stream %d as TCP", i);
protocols = GST_RTSP_PROTO_TCP; protocols = GST_RTSP_PROTO_TCP;
src->interleaved = TRUE; src->interleaved = TRUE;
} else { } else {
if (transport.multicast) { if (transport.multicast) {
/* disable unicast */ /* only allow multicast for other streams */
GST_DEBUG_OBJECT (src, "stream %d as MULTICAST", i); GST_DEBUG_OBJECT (src, "stream %d as MULTICAST", i);
protocols = GST_RTSP_PROTO_UDP_MULTICAST; protocols = GST_RTSP_PROTO_UDP_MULTICAST;
} else { } else {
/* disable multicast */ /* only allow unicast for other streams */
GST_DEBUG_OBJECT (src, "stream %d as UNICAST", i); GST_DEBUG_OBJECT (src, "stream %d as UNICAST", i);
protocols = GST_RTSP_PROTO_UDP_UNICAST; protocols = GST_RTSP_PROTO_UDP_UNICAST;
} }
@ -1314,9 +1346,8 @@ gst_rtspsrc_close (GstRTSPSrc * src)
if (src->options & RTSP_PLAY) { if (src->options & RTSP_PLAY) {
/* do TEARDOWN */ /* do TEARDOWN */
if ((res = res = rtsp_message_init_request (RTSP_TEARDOWN, src->location, &request);
rtsp_message_init_request (RTSP_TEARDOWN, src->location, if (res < 0)
&request)) < 0)
goto create_request_failed; goto create_request_failed;
if (!gst_rtspsrc_send (src, &request, &response, NULL)) if (!gst_rtspsrc_send (src, &request, &response, NULL))
@ -1363,8 +1394,8 @@ gst_rtspsrc_play (GstRTSPSrc * src)
GST_DEBUG_OBJECT (src, "PLAY..."); GST_DEBUG_OBJECT (src, "PLAY...");
/* do play */ /* do play */
if ((res = res = rtsp_message_init_request (RTSP_PLAY, src->location, &request);
rtsp_message_init_request (RTSP_PLAY, src->location, &request)) < 0) if (res < 0)
goto create_request_failed; goto create_request_failed;
if (!gst_rtspsrc_send (src, &request, &response, NULL)) if (!gst_rtspsrc_send (src, &request, &response, NULL))
@ -1406,8 +1437,8 @@ gst_rtspsrc_pause (GstRTSPSrc * src)
GST_DEBUG_OBJECT (src, "PAUSE..."); GST_DEBUG_OBJECT (src, "PAUSE...");
/* do pause */ /* do pause */
if ((res = res = rtsp_message_init_request (RTSP_PAUSE, src->location, &request);
rtsp_message_init_request (RTSP_PAUSE, src->location, &request)) < 0) if (res < 0)
goto create_request_failed; goto create_request_failed;
if (!gst_rtspsrc_send (src, &request, &response, NULL)) if (!gst_rtspsrc_send (src, &request, &response, NULL))

6
gst/rtsp/rtspconnection.c
View file

@ -45,12 +45,12 @@
#endif #endif
#ifdef G_OS_WIN32 #ifdef G_OS_WIN32
/* note that inet_aton is deprecated on unix because
* inet_addr returns -1 (INADDR_NONE) for the valid 255.255.255.255
* address. */
static int static int
inet_aton (const char *c, struct in_addr *paddr) inet_aton (const char *c, struct in_addr *paddr)
{ {
/* note that inet_addr is deprecated on unix because
* inet_addr returns -1 (INADDR_NONE) for the valid 255.255.255.255
* address. */
paddr->s_addr = inet_addr (c); paddr->s_addr = inet_addr (c);
if (paddr->s_addr == INADDR_NONE) if (paddr->s_addr == INADDR_NONE)