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rtspsrc: parse connection information

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Parse the connection information from the SDP and use it to figure out if we are
dealing with ipv4 or ipv6 connections.
This commit is contained in:
Wim Taymans 2010-03-10 11:19:46 +01:00
parent 8eb5c2c794
commit 821096c4f1
1 changed files with 48 additions and 1 deletions
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49
gst/rtsp/gstrtspsrc.c
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@ -804,6 +804,49 @@ gst_rtspsrc_collect_bandwidth (GstRTSPSrc * src, const GstSDPMessage * sdp,
stream->rs_bandwidth = -1;
}
static void
gst_rtspsrc_do_stream_connection (GstRTSPSrc * src, GstRTSPStream * stream,
const GstSDPConnection * conn)
{
if (strcmp (conn->nettype, "IN") != 0)
return;
/* check for IPV6 */
if (strcmp (conn->addrtype, "IP4") == 0)
stream->is_ipv6 = FALSE;
else if (strcmp (conn->addrtype, "IP6") == 0)
stream->is_ipv6 = TRUE;
else
return;
/* FIXME check for multicast */
}
/* Go over the connections for a stream.
* - If we are dealing with IPV6, we will setup IPV6 sockets for sending and
* receiving.
* - If we are dealing with a localhost address, we disable multicast
*/
static void
gst_rtspsrc_collect_connections (GstRTSPSrc * src, const GstSDPMessage * sdp,
const GstSDPMedia * media, GstRTSPStream * stream)
{
const GstSDPConnection *conn;
guint i, len;
/* first look in the media specific section */
len = gst_sdp_media_connections_len (media);
for (i = 0; i < len; i++) {
conn = gst_sdp_media_get_connection (media, i);
gst_rtspsrc_do_stream_connection (src, stream, conn);
}
/* then look in the message specific section */
if ((conn = gst_sdp_message_get_connection (sdp))) {
gst_rtspsrc_do_stream_connection (src, stream, conn);
}
}
static GstRTSPStream *
gst_rtspsrc_create_stream (GstRTSPSrc * src, GstSDPMessage * sdp, gint idx)
{
@ -830,9 +873,13 @@ gst_rtspsrc_create_stream (GstRTSPSrc * src, GstSDPMessage * sdp, gint idx)
stream->seqbase = -1;
stream->timebase = -1;
/* collect bandwidth information for this steam */
/* collect bandwidth information for this steam. FIXME, configure in the RTP
* session manager to scale RTCP. */
gst_rtspsrc_collect_bandwidth (src, sdp, media, stream);
/* collect connection info */
gst_rtspsrc_collect_connections (src, sdp, media, stream);
/* we must have a payload. No payload means we cannot create caps */
/* FIXME, handle multiple formats. The problem here is that we just want to
* take the first available format that we can handle but in order to do that