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rtspsrc: refactor collecting the transport info

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Make a method to collect the ports and destination address.
This commit is contained in:
Wim Taymans 2010-05-07 12:24:51 +02:00
parent 05352d7ea8
commit 4e1ced0a77
1 changed files with 68 additions and 42 deletions
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110
gst/rtsp/gstrtspsrc.c
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@ -2377,12 +2377,56 @@ gst_rtspsrc_stream_configure_tcp (GstRTSPSrc * src, GstRTSPStream * stream,
return TRUE;
}
static void
gst_rtspsrc_get_transport_info (GstRTSPSrc * src, GstRTSPStream * stream,
GstRTSPTransport * transport, const gchar ** destination, gint * min,
gint * max, guint * ttl)
{
if (transport->lower_transport == GST_RTSP_LOWER_TRANS_UDP_MCAST) {
if (destination) {
if (!(*destination = transport->destination))
*destination = stream->destination;
}
if (min && max) {
/* transport first */
*min = transport->port.min;
*max = transport->port.max;
if (*min == -1 && *max == -1) {
/* then try from SDP */
}
if (*min == -1 && *max == -1) {
/* some bad servers use the server_port attribute for multicast, try to handle
* those cases too here */
*min = transport->server_port.min;
*max = transport->server_port.max;
}
}
if (ttl) {
if (!(*ttl = transport->ttl))
*ttl = stream->ttl;
}
} else {
if (destination) {
/* first take the source, then the endpoint to figure out where to send
* the RTCP. */
if (!(*destination = transport->source))
*destination = gst_rtsp_connection_get_ip (src->connection);
}
if (min && max) {
*min = transport->server_port.min;
*max = transport->server_port.max;
}
}
}
/* For multicast create UDP sources and join the multicast group. */
static gboolean
gst_rtspsrc_stream_configure_mcast (GstRTSPSrc * src, GstRTSPStream * stream,
GstRTSPTransport * transport, GstPad ** outpad)
{
gchar *uri, *destination;
gchar *uri;
const gchar *destination;
gint min, max;
GST_DEBUG_OBJECT (src, "creating UDP sources for multicast");
@ -2390,29 +2434,20 @@ gst_rtspsrc_stream_configure_mcast (GstRTSPSrc * src, GstRTSPStream * stream,
/* we can remove the allocated UDP ports now */
gst_rtspsrc_stream_free_udp (stream);
gst_rtspsrc_get_transport_info (src, stream, transport, &destination, &min,
&max, NULL);
/* we need a destination now */
if (!(destination = transport->destination)) {
if (!(destination = stream->destination))
goto no_destination;
}
min = transport->port.min;
max = transport->port.max;
/* some bad servers use the server_port attribute for multicast, try to handle
* those cases too here */
if (min == -1 && max == -1) {
GST_DEBUG_OBJECT (src, "no port attribute set, fallback to server_port");
min = transport->server_port.min;
max = transport->server_port.max;
}
GST_DEBUG_OBJECT (src, "have destination '%s' and ports (%d)-(%d)",
destination, min, max);
if (destination == NULL)
goto no_destination;
/* we really need ports now or we won't be able to receive anything at all */
if (min == -1 && max == -1)
goto no_ports;
GST_DEBUG_OBJECT (src, "have destination '%s' and ports (%d)-(%d)",
destination, min, max);
/* creating UDP source for RTP */
if (min != -1) {
uri = g_strdup_printf ("udp://%s:%d", destination, min);
@ -2532,35 +2567,28 @@ gst_rtspsrc_stream_configure_udp_sinks (GstRTSPSrc * src,
{
GstPad *pad;
gint rtp_port, rtcp_port, sockfd = -1;
gboolean do_rtp, do_rtcp;
const gchar *destination;
gchar *uri, *name;
guint ttl = 0;
/* get host and port */
if (transport->lower_transport == GST_RTSP_LOWER_TRANS_UDP_MCAST) {
rtp_port = transport->port.min;
rtcp_port = transport->port.max;
/* multicast destination */
if (!(destination = transport->destination))
destination = stream->destination;
/* get ttl */
if (!(ttl = transport->ttl))
ttl = stream->ttl;
} else {
rtp_port = transport->server_port.min;
rtcp_port = transport->server_port.max;
/* first take the source, then the endpoint to figure out where to send
* the RTCP. */
destination = transport->source;
if (destination == NULL)
destination = gst_rtsp_connection_get_ip (src->connection);
}
if (destination == NULL)
/* get transport info */
gst_rtspsrc_get_transport_info (src, stream, transport, &destination,
&rtp_port, &rtcp_port, &ttl);
/* see what we need to do */
do_rtp = (rtp_port != -1);
/* it's possible that the server does not want us to send RTCP in which case
* the port is -1 */
do_rtcp = (rtcp_port != -1 && src->session != NULL && src->do_rtcp);
/* we need a destination when we have RTP or RTCP ports */
if (destination == NULL && (do_rtp || do_rtcp))
goto no_destination;
/* try to construct the fakesrc to the RTP port of the server to open up any
* NAT firewalls */
if (rtp_port != -1) {
if (do_rtp) {
GST_DEBUG_OBJECT (src, "configure RTP UDP sink for %s:%d", destination,
rtp_port);
@ -2615,9 +2643,7 @@ gst_rtspsrc_stream_configure_udp_sinks (GstRTSPSrc * src,
gst_element_link (stream->fakesrc, stream->udpsink[0]);
}
/* it's possible that the server does not want us to send RTCP in which case
* the port is -1 */
if (rtcp_port != -1 && src->session != NULL && src->do_rtcp) {
if (do_rtcp) {
GST_DEBUG_OBJECT (src, "configure RTCP UDP sink for %s:%d", destination,
rtcp_port);