summaryrefslogtreecommitdiffstats
diff options
context:
space:
mode:
Diffstat
-rw-r--r--ChangeLog38
-rw-r--r--gst/rtsp/gstrtspsrc.c587
-rw-r--r--gst/rtsp/gstrtspsrc.h35
-rw-r--r--gst/rtsp/rtspconnection.c22
-rw-r--r--gst/rtsp/rtspdefs.c1
-rw-r--r--gst/rtsp/rtspdefs.h29
-rw-r--r--gst/rtsp/rtspext.h3
-rw-r--r--gst/rtsp/rtspextwms.c60
-rw-r--r--gst/rtsp/rtsptransport.c43
-rw-r--r--gst/rtsp/rtsptransport.h13
10 files changed, 590 insertions, 241 deletions
diff --git a/ChangeLog b/ChangeLog
index 53478602..5e3acf01 100644
--- a/ChangeLog
+++ b/ChangeLog
@@ -1,3 +1,41 @@
+2006-10-06 Wim Taymans <wim@fluendo.com>
+
+ * gst/rtsp/gstrtspsrc.c: (gst_rtsp_proto_get_type),
+ (gst_rtspsrc_class_init), (gst_rtspsrc_init),
+ (gst_rtspsrc_create_stream), (gst_rtspsrc_media_to_caps),
+ (gst_rtspsrc_alloc_udp_ports),
+ (gst_rtspsrc_stream_configure_transport), (find_stream_by_channel),
+ (gst_rtspsrc_push_event), (gst_rtspsrc_loop_interleaved),
+ (gst_rtspsrc_create_transports_string),
+ (gst_rtspsrc_configure_transports), (gst_rtspsrc_open),
+ (gst_rtspsrc_handle_message), (gst_rtspsrc_change_state):
+ * gst/rtsp/gstrtspsrc.h:
+ Rework how the transport string is constructed, try to share channels
+ and udp ports.
+ Make most of the stuff less dependant on RTP as we are also going to use
+ it for RDT.
+ Add support for transport specific session managers.
+
+ * gst/rtsp/rtspconnection.c: (rtsp_connection_flush):
+ Implement _flush().
+
+ * gst/rtsp/rtspdefs.c: (rtsp_strresult):
+ * gst/rtsp/rtspdefs.h:
+ Add generic error return code.
+
+ * gst/rtsp/rtspext.h:
+ Add support for pluggable tranport strings.
+
+ * gst/rtsp/rtspextwms.c: (rtsp_ext_wms_before_send),
+ (rtsp_ext_wms_after_send), (rtsp_ext_wms_parse_sdp),
+ (rtsp_ext_wms_get_context):
+ Detect WMServer and activate the extension.
+
+ * gst/rtsp/rtsptransport.c: (rtsp_transport_get_mime),
+ (rtsp_transport_get_manager), (rtsp_transport_parse):
+ * gst/rtsp/rtsptransport.h:
+ Added methods to get mime/manager for certain transports.
+
2006-10-05 Tim-Philipp Müller <tim at centricular dot net>
* ext/cairo/gsttimeoverlay.c:
diff --git a/gst/rtsp/gstrtspsrc.c b/gst/rtsp/gstrtspsrc.c
index 0ea88313..70c6aa07 100644
--- a/gst/rtsp/gstrtspsrc.c
+++ b/gst/rtsp/gstrtspsrc.c
@@ -96,6 +96,9 @@
#include "sdp.h"
#include "rtspextwms.h"
+#ifdef WITH_EXT_REAL
+#include "rtspextreal.h"
+#endif
GST_DEBUG_CATEGORY_STATIC (rtspsrc_debug);
#define GST_CAT_DEFAULT (rtspsrc_debug)
@@ -109,11 +112,10 @@ GST_ELEMENT_DETAILS ("RTSP packet receiver",
"Thijs Vermeir <thijs.vermeir@barco.com>\n"
"Lutz Mueller <lutz@topfrose.de>");
-static GstStaticPadTemplate rtptemplate =
-GST_STATIC_PAD_TEMPLATE ("rtp_stream%d",
+static GstStaticPadTemplate rtptemplate = GST_STATIC_PAD_TEMPLATE ("stream%d",
GST_PAD_SRC,
GST_PAD_SOMETIMES,
- GST_STATIC_CAPS ("application/x-rtp"));
+ GST_STATIC_CAPS ("application/x-rtp; application/x-rdt"));
enum
{
@@ -122,7 +124,7 @@ enum
};
#define DEFAULT_LOCATION NULL
-#define DEFAULT_PROTOCOLS GST_RTSP_PROTO_UDP_UNICAST | GST_RTSP_PROTO_UDP_MULTICAST | GST_RTSP_PROTO_TCP
+#define DEFAULT_PROTOCOLS RTSP_LOWER_TRANS_UDP | RTSP_LOWER_TRANS_UDP_MCAST | RTSP_LOWER_TRANS_TCP
#define DEFAULT_DEBUG FALSE
#define DEFAULT_RETRY 20
#define DEFAULT_TIMEOUT 5000000
@@ -143,9 +145,9 @@ gst_rtsp_proto_get_type (void)
{
static GType rtsp_proto_type = 0;
static const GFlagsValue rtsp_proto[] = {
- {GST_RTSP_PROTO_UDP_UNICAST, "UDP Unicast Mode", "udp-unicast"},
- {GST_RTSP_PROTO_UDP_MULTICAST, "UDP Multicast Mode", "udp-multicast"},
- {GST_RTSP_PROTO_TCP, "TCP interleaved mode", "tcp"},
+ {RTSP_LOWER_TRANS_UDP, "UDP Unicast Mode", "udp-unicast"},
+ {RTSP_LOWER_TRANS_UDP_MCAST, "UDP Multicast Mode", "udp-multicast"},
+ {RTSP_LOWER_TRANS_TCP, "TCP interleaved mode", "tcp"},
{0, NULL, NULL},
};
@@ -233,9 +235,9 @@ gst_rtspsrc_class_init (GstRTSPSrcClass * klass)
DEFAULT_LOCATION, G_PARAM_READWRITE));
g_object_class_install_property (gobject_class, PROP_PROTOCOLS,
- g_param_spec_flags ("protocols", "Protocols", "Allowed protocols",
- GST_TYPE_RTSP_PROTO, DEFAULT_PROTOCOLS,
- G_PARAM_READWRITE | G_PARAM_CONSTRUCT));
+ g_param_spec_flags ("protocols", "Protocols",
+ "Allowed lower transport protocols", GST_TYPE_RTSP_PROTO,
+ DEFAULT_PROTOCOLS, G_PARAM_READWRITE | G_PARAM_CONSTRUCT));
g_object_class_install_property (gobject_class, PROP_DEBUG,
g_param_spec_boolean ("debug", "Debug",
@@ -272,6 +274,9 @@ gst_rtspsrc_init (GstRTSPSrc * src, GstRTSPSrcClass * g_class)
/* install WMS extension by default */
src->extension = rtsp_ext_wms_get_context ();
+#ifdef WITH_EXT_REAL
+ src->extension = rtsp_ext_real_get_context ();
+#endif
src->extension->src = (gpointer) src;
}
@@ -414,8 +419,9 @@ gst_rtspsrc_create_stream (GstRTSPSrc * src, SDPMessage * sdp, gint idx)
GST_DEBUG_OBJECT (src, " control: %s", GST_STR_NULL (control_url));
if (control_url != NULL) {
- /* FIXME, what if the control_url starts with a '/' or a non rtsp: protocol? */
- /* check absolute/relative URL */
+ /* If the control_url starts with a '/' or a non rtsp: protocol we will most
+ * likely build a URL that the server will fail to understand, this is ok,
+ * we will fail then. */
if (g_str_has_prefix (control_url, "rtsp://"))
stream->setup_url = g_strdup (control_url);
else if (src->content_base)
@@ -565,7 +571,7 @@ gst_rtspsrc_media_to_caps (gint pt, SDPMedia * media)
goto no_rtpmap;
}
- caps = gst_caps_new_simple ("application/x-rtp",
+ caps = gst_caps_new_simple ("application/x-unknown",
"media", G_TYPE_STRING, media->media, "payload", G_TYPE_INT, pt, NULL);
s = gst_caps_get_structure (caps, 0);
@@ -627,34 +633,34 @@ no_rtpmap:
}
static gboolean
-gst_rtspsrc_stream_setup_rtp (GstRTSPStream * stream,
+gst_rtspsrc_alloc_udp_ports (GstRTSPStream * stream,
gint * rtpport, gint * rtcpport)
{
- GstStateChangeReturn ret;
GstRTSPSrc *src;
- GstElement *tmp, *rtpsrc, *rtcpsrc;
+ GstStateChangeReturn ret;
+ GstElement *tmp, *udpsrc0, *udpsrc1;
gint tmp_rtp, tmp_rtcp;
guint count;
src = stream->parent;
tmp = NULL;
- rtpsrc = NULL;
- rtcpsrc = NULL;
+ udpsrc0 = NULL;
+ udpsrc1 = NULL;
count = 0;
/* try to allocate 2 UDP ports, the RTP port should be an even
* number and the RTCP port should be the next (uneven) port */
again:
- rtpsrc = gst_element_make_from_uri (GST_URI_SRC, "udp://0.0.0.0:0", NULL);
- if (rtpsrc == NULL)
- goto no_udp_rtp_protocol;
+ udpsrc0 = gst_element_make_from_uri (GST_URI_SRC, "udp://0.0.0.0:0", NULL);
+ if (udpsrc0 == NULL)
+ goto no_udp_protocol;
- ret = gst_element_set_state (rtpsrc, GST_STATE_PAUSED);
+ ret = gst_element_set_state (udpsrc0, GST_STATE_PAUSED);
if (ret == GST_STATE_CHANGE_FAILURE)
- goto start_rtp_failure;
+ goto start_udp_failure;
- g_object_get (G_OBJECT (rtpsrc), "port", &tmp_rtp, NULL);
+ g_object_get (G_OBJECT (udpsrc0), "port", &tmp_rtp, NULL);
GST_DEBUG_OBJECT (src, "got RTP port %d", tmp_rtp);
/* check if port is even */
@@ -671,7 +677,7 @@ again:
gst_element_set_state (tmp, GST_STATE_NULL);
gst_object_unref (tmp);
}
- tmp = rtpsrc;
+ tmp = udpsrc0;
GST_DEBUG_OBJECT (src, "retry %d", count);
goto again;
}
@@ -683,50 +689,45 @@ again:
}
/* allocate port+1 for RTCP now */
- rtcpsrc = gst_element_make_from_uri (GST_URI_SRC, "udp://0.0.0.0", NULL);
- if (rtcpsrc == NULL)
+ udpsrc1 = gst_element_make_from_uri (GST_URI_SRC, "udp://0.0.0.0", NULL);
+ if (udpsrc1 == NULL)
goto no_udp_rtcp_protocol;
/* set port */
tmp_rtcp = tmp_rtp + 1;
- g_object_set (G_OBJECT (rtcpsrc), "port", tmp_rtcp, NULL);
+ g_object_set (G_OBJECT (udpsrc1), "port", tmp_rtcp, NULL);
GST_DEBUG_OBJECT (src, "starting RTCP on port %d", tmp_rtcp);
- ret = gst_element_set_state (rtcpsrc, GST_STATE_PAUSED);
+ ret = gst_element_set_state (udpsrc1, GST_STATE_PAUSED);
/* FIXME, this could fail if the next port is not free, we
* should retry with another port then */
if (ret == GST_STATE_CHANGE_FAILURE)
goto start_rtcp_failure;
/* all fine, do port check */
- g_object_get (G_OBJECT (rtpsrc), "port", rtpport, NULL);
- g_object_get (G_OBJECT (rtcpsrc), "port", rtcpport, NULL);
+ g_object_get (G_OBJECT (udpsrc0), "port", rtpport, NULL);
+ g_object_get (G_OBJECT (udpsrc1), "port", rtcpport, NULL);
/* this should not happen */
if (*rtpport != tmp_rtp || *rtcpport != tmp_rtcp)
goto port_error;
- /* configure a timeout */
- g_object_set (G_OBJECT (rtpsrc), "timeout", src->timeout, NULL);
-
- /* we manage these elements, we set the caps in configure_transport */
- stream->rtpsrc = rtpsrc;
- stream->rtcpsrc = rtcpsrc;
-
- gst_bin_add (GST_BIN_CAST (src), stream->rtpsrc);
- gst_bin_add (GST_BIN_CAST (src), stream->rtcpsrc);
+ /* we keep these elements, we configure all in configure_transport when the
+ * server told us to really use the UDP ports. */
+ stream->udpsrc[0] = udpsrc0;
+ stream->udpsrc[1] = udpsrc1;
return TRUE;
/* ERRORS */
-no_udp_rtp_protocol:
+no_udp_protocol:
{
- GST_DEBUG_OBJECT (src, "could not get UDP source for RTP");
+ GST_DEBUG_OBJECT (src, "could not get UDP source");
goto cleanup;
}
-start_rtp_failure:
+start_udp_failure:
{
- GST_DEBUG_OBJECT (src, "could not start UDP source for RTP");
+ GST_DEBUG_OBJECT (src, "could not start UDP source");
goto cleanup;
}
no_ports:
@@ -757,13 +758,13 @@ cleanup:
gst_element_set_state (tmp, GST_STATE_NULL);
gst_object_unref (tmp);
}
- if (rtpsrc) {
- gst_element_set_state (rtpsrc, GST_STATE_NULL);
- gst_object_unref (rtpsrc);
+ if (udpsrc0) {
+ gst_element_set_state (udpsrc0, GST_STATE_NULL);
+ gst_object_unref (udpsrc0);
}
- if (rtcpsrc) {
- gst_element_set_state (rtcpsrc, GST_STATE_NULL);
- gst_object_unref (rtcpsrc);
+ if (udpsrc1) {
+ gst_element_set_state (udpsrc1, GST_STATE_NULL);
+ gst_object_unref (udpsrc1);
}
return FALSE;
}
@@ -774,115 +775,209 @@ gst_rtspsrc_stream_configure_transport (GstRTSPStream * stream,
RTSPTransport * transport)
{
GstRTSPSrc *src;
- GstPad *pad;
+ GstPad *outpad = NULL;
GstPadTemplate *template;
GstStateChangeReturn ret;
gchar *name;
+ GstStructure *s;
+ const gchar *mime, *manager;
+ RTSPResult res;
src = stream->parent;
- GST_DEBUG ("configuring RTP transport for stream %p", stream);
+ GST_DEBUG_OBJECT (src, "configuring transport for stream %p", stream);
+
+ s = gst_caps_get_structure (stream->caps, 0);
+
+ if ((res = rtsp_transport_get_mime (transport->trans, &mime)) < 0)
+ goto no_mime;
+ if (!mime)
+ goto no_mime;
+
+ GST_DEBUG_OBJECT (src, "setting mime to %s", mime);
+ /* configure the final mime type */
+ gst_structure_set_name (s, mime);
+
+ if ((res = rtsp_transport_get_manager (transport->trans, &manager)) < 0)
+ goto no_manager;
- /* FIXME, the session manager needs to be shared with all the streams */
- if (!(stream->rtpdec = gst_element_factory_make ("rtpdec", NULL)))
- goto no_element;
+ if (manager) {
+ GST_DEBUG_OBJECT (src, "using manager %s", manager);
+ /* FIXME, the session manager needs to be shared with all the streams */
+ if (!(stream->sess = gst_element_factory_make (manager, NULL)))
+ goto no_element;
- /* we manage this element */
- gst_bin_add (GST_BIN_CAST (src), stream->rtpdec);
+ /* we manage this element */
+ gst_bin_add (GST_BIN_CAST (src), stream->sess);
- ret = gst_element_set_state (stream->rtpdec, GST_STATE_PAUSED);
- if (ret != GST_STATE_CHANGE_SUCCESS)
- goto start_rtpdec_failure;
+ ret = gst_element_set_state (stream->sess, GST_STATE_PAUSED);
+ if (ret != GST_STATE_CHANGE_SUCCESS)
+ goto start_session_failure;
- stream->rtpdecrtp = gst_element_get_pad (stream->rtpdec, "sinkrtp");
- stream->rtpdecrtcp = gst_element_get_pad (stream->rtpdec, "sinkrtcp");
+ stream->channelpad[0] = gst_element_get_pad (stream->sess, "sinkrtp");
+ stream->channelpad[1] = gst_element_get_pad (stream->sess, "sinkrtcp");
+ }
if (transport->lower_transport == RTSP_LOWER_TRANS_TCP) {
+ gint i;
+
/* configure for interleaved delivery, nothing needs to be done
* here, the loop function will call the chain functions of the
- * RTP session manager. */
- stream->rtpchannel = transport->interleaved.min;
- stream->rtcpchannel = transport->interleaved.max;
- GST_DEBUG ("stream %p on channels %d-%d", stream,
- stream->rtpchannel, stream->rtcpchannel);
+ * session manager. */
+ stream->channel[0] = transport->interleaved.min;
+ stream->channel[1] = transport->interleaved.max;
+ GST_DEBUG_OBJECT (src, "stream %p on channels %d-%d", stream,
+ stream->channel[0], stream->channel[1]);
+
+ /* we can remove the allocated UDP ports now */
+ for (i = 0; i < 2; i++) {
+ if (stream->udpsrc[i]) {
+ gst_element_set_state (stream->udpsrc[i], GST_STATE_NULL);
+ gst_object_unref (stream->udpsrc[i]);
+ stream->udpsrc[i] = NULL;
+ }
+ }
+ /* no session manager, send data to srcpad directly */
+ if (!stream->channelpad[0]) {
+ GST_DEBUG_OBJECT (src, "no manager, creating pad");
+
+ name = g_strdup_printf ("stream%d", stream->id);
+ template = gst_static_pad_template_get (&rtptemplate);
+ stream->srcpad = gst_pad_new_from_template (template, name);
+ gst_object_unref (template);
+ g_free (name);
+
+ gst_pad_use_fixed_caps (stream->srcpad);
+ gst_pad_set_caps (stream->srcpad, stream->caps);
+
+ stream->channelpad[0] = gst_object_ref (stream->srcpad);
+ gst_element_add_pad (GST_ELEMENT_CAST (src), stream->srcpad);
+ } else {
+ GST_DEBUG_OBJECT (src, "using manager source pad");
+ outpad = gst_element_get_pad (stream->sess, "srcrtp");
+ }
} else {
/* multicast was selected, create UDP sources and join the multicast
* group. */
if (transport->lower_transport == RTSP_LOWER_TRANS_UDP_MCAST) {
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_bin_add (GST_BIN_CAST (src), stream->rtpsrc);
- gst_bin_add (GST_BIN_CAST (src), stream->rtcpsrc);
+ GST_DEBUG_OBJECT (src, "creating UDP sources for multicast");
+
+ /* creating UDP source */
+ if (transport->port.min != -1) {
+ uri = g_strdup_printf ("udp://%s:%d", transport->destination,
+ transport->port.min);
+ stream->udpsrc[0] = gst_element_make_from_uri (GST_URI_SRC, uri, NULL);
+ g_free (uri);
+ if (stream->udpsrc[0] == NULL)
+ goto no_element;
+
+ /* change state */
+ gst_element_set_state (stream->udpsrc[0], GST_STATE_PAUSED);
+ }
+
+ /* creating another UDP source */
+ if (transport->port.max != -1) {
+ uri = g_strdup_printf ("udp://%s:%d", transport->destination,
+ transport->port.max);
+ stream->udpsrc[1] = gst_element_make_from_uri (GST_URI_SRC, uri, NULL);
+ g_free (uri);
+ if (stream->udpsrc[1] == NULL)
+ goto no_element;
+
+ gst_element_set_state (stream->udpsrc[1], GST_STATE_PAUSED);
+ }
}
- /* set caps */
- g_object_set (G_OBJECT (stream->rtpsrc), "caps", stream->caps, NULL);
+ /* we manage the UDP elements now. For unicast, the UDP sources where
+ * allocated in the stream when we suggested a transport. */
+ if (stream->udpsrc[0]) {
+ GstPad *pad;
- /* configure for UDP delivery, we need to connect the UDP pads to
- * the RTP session plugin. */
- pad = gst_element_get_pad (stream->rtpsrc, "src");
- gst_pad_link (pad, stream->rtpdecrtp);
- gst_object_unref (pad);
+ gst_bin_add (GST_BIN_CAST (src), stream->udpsrc[0]);
- pad = gst_element_get_pad (stream->rtcpsrc, "src");
- gst_pad_link (pad, stream->rtpdecrtcp);
- gst_object_unref (pad);
- }
+ GST_DEBUG_OBJECT (src, "setting up UDP source");
- pad = gst_element_get_pad (stream->rtpdec, "srcrtp");
- if (stream->caps) {
- gst_pad_use_fixed_caps (pad);
- gst_pad_set_caps (pad, stream->caps);
+ /* set caps */
+ g_object_set (G_OBJECT (stream->udpsrc[0]), "caps", stream->caps, NULL);
+
+ /* configure a timeout on the UDP port */
+ g_object_set (G_OBJECT (stream->udpsrc[0]), "timeout", src->timeout,
+ NULL);
+
+ if (stream->channelpad[0]) {
+ GST_DEBUG_OBJECT (src, "connecting UDP source 0 to manager");
+ /* configure for UDP delivery, we need to connect the UDP pads to
+ * the session plugin. */
+ pad = gst_element_get_pad (stream->udpsrc[0], "src");
+ gst_pad_link (pad, stream->channelpad[0]);
+ gst_object_unref (pad);
+
+ outpad = gst_element_get_pad (stream->sess, "srcrtp");
+ } else {
+ GST_DEBUG_OBJECT (src, "using UDP src pad as output");
+ outpad = gst_element_get_pad (stream->udpsrc[0], "src");
+ }
+ }
+
+ if (stream->udpsrc[1]) {
+ gst_bin_add (GST_BIN_CAST (src), stream->udpsrc[1]);
+
+ if (stream->channelpad[1]) {
+ GstPad *pad;
+
+ GST_DEBUG_OBJECT (src, "connecting UDP source 1 to manager");
+
+ pad = gst_element_get_pad (stream->udpsrc[1], "src");
+ gst_pad_link (pad, stream->channelpad[1]);
+ gst_object_unref (pad);
+ }
+ }
}
- /* create ghostpad */
- name = g_strdup_printf ("rtp_stream%d", stream->id);
- template = gst_static_pad_template_get (&rtptemplate);
- stream->srcpad = gst_ghost_pad_new_from_template (name, pad, template);
- gst_object_unref (template);
- g_free (name);
+ if (outpad && !stream->srcpad) {
+ GST_DEBUG_OBJECT (src, "creating ghostpad");
- gst_object_unref (pad);
+ gst_pad_use_fixed_caps (outpad);
+ gst_pad_set_caps (outpad, stream->caps);
+ /* create ghostpad */
+ name = g_strdup_printf ("stream%d", stream->id);
+ template = gst_static_pad_template_get (&rtptemplate);
+ stream->srcpad = gst_ghost_pad_new_from_template (name, outpad, template);
+ gst_object_unref (template);
+ g_free (name);
+
+ gst_object_unref (outpad);
+
+ /* and add */
+ gst_element_add_pad (GST_ELEMENT_CAST (src), stream->srcpad);
+ }
/* mark pad as ok */
stream->last_ret = GST_FLOW_OK;
- /* and add */
- gst_element_add_pad (GST_ELEMENT_CAST (src), stream->srcpad);
return TRUE;
/* ERRORS */
+no_mime:
+ {
+ GST_DEBUG_OBJECT (src, "unknown transport");
+ return FALSE;
+ }
+no_manager:
+ {
+ GST_DEBUG_OBJECT (src, "cannot get a session manager");
+ return FALSE;
+ }
no_element:
{
GST_DEBUG_OBJECT (src, "no rtpdec element found");
return FALSE;
}
-start_rtpdec_failure:
+start_session_failure:
{
- GST_DEBUG_OBJECT (src, "could not start RTP session");
+ GST_DEBUG_OBJECT (src, "could not start session");
return FALSE;
}
}
@@ -892,7 +987,7 @@ find_stream_by_channel (GstRTSPStream * stream, gconstpointer a)
{
gint channel = GPOINTER_TO_INT (a);
- if (stream->rtpchannel == channel || stream->rtcpchannel == channel)
+ if (stream->channel[0] == channel || stream->channel[1] == channel)
return 0;
return -1;
@@ -944,10 +1039,21 @@ gst_rtspsrc_push_event (GstRTSPSrc * src, GstEvent * event)
if (ostream->srcpad == NULL)
continue;
- gst_event_ref (event);
- gst_pad_push_event (ostream->rtpdecrtp, event);
- gst_event_ref (event);
- gst_pad_push_event (ostream->rtpdecrtcp, event);
+ if (ostream->channelpad[0]) {
+ gst_event_ref (event);
+ if (GST_PAD_IS_SRC (ostream->channelpad[0]))
+ gst_pad_push_event (ostream->channelpad[0], event);
+ else
+ gst_pad_send_event (ostream->channelpad[0], event);
+ }
+
+ if (ostream->channelpad[1]) {
+ gst_event_ref (event);
+ if (GST_PAD_IS_SRC (ostream->channelpad[1]))
+ gst_pad_push_event (ostream->channelpad[1], event);
+ else
+ gst_pad_send_event (ostream->channelpad[1], event);
+ }
}
gst_event_unref (event);
}
@@ -984,11 +1090,11 @@ gst_rtspsrc_loop_interleaved (GstRTSPSrc * src)
goto unknown_stream;
stream = (GstRTSPStream *) lstream->data;
- if (channel == stream->rtpchannel) {
- outpad = stream->rtpdecrtp;
+ if (channel == stream->channel[0]) {
+ outpad = stream->channelpad[0];
caps = stream->caps;
- } else if (channel == stream->rtcpchannel) {
- outpad = stream->rtpdecrtcp;
+ } else if (channel == stream->channel[1]) {
+ outpad = stream->channelpad[1];
}
/* take a look at the body to figure out what we have */
@@ -999,7 +1105,7 @@ gst_rtspsrc_loop_interleaved (GstRTSPSrc * src)
/* channels are not correct on some servers, do extra check */
if (data[1] >= 200 && data[1] <= 204) {
/* hmm RTCP message switch to the RTCP pad of the same stream. */
- outpad = stream->rtpdecrtcp;
+ outpad = stream->channelpad[1];
}
/* we have no clue what this is, just ignore then. */
@@ -1027,7 +1133,10 @@ gst_rtspsrc_loop_interleaved (GstRTSPSrc * src)
channel);
/* chain to the peer pad */
- ret = gst_pad_chain (outpad, buf);
+ if (GST_PAD_IS_SINK (outpad))
+ ret = gst_pad_chain (outpad, buf);
+ else
+ ret = gst_pad_push (outpad, buf);
/* combine all stream flows */
ret = gst_rtspsrc_combine_flows (src, stream, ret);
@@ -1311,6 +1420,136 @@ no_setup:
}
}
+static RTSPResult
+gst_rtspsrc_create_transports_string (GstRTSPSrc * src,
+ RTSPLowerTrans protocols, gchar ** transports)
+{
+ gchar *result;
+ RTSPResult res;
+
+ *transports = NULL;
+ if (src->extension && src->extension->get_transports)
+ if ((res =
+ src->extension->get_transports (src->extension, protocols,
+ transports)) < 0)
+ goto failed;
+
+ /* extension listed transports, use those */
+ if (*transports != NULL)
+ return RTSP_OK;
+
+ result = g_strdup ("");
+ if (protocols & RTSP_LOWER_TRANS_UDP) {
+ gchar *new;
+
+ GST_DEBUG_OBJECT (src, "adding UDP unicast");
+
+ new =
+ g_strconcat (result, "RTP/AVP/UDP;unicast;client_port=%%u1-%%u2", NULL);
+ g_free (result);
+ result = new;
+ }
+ if (protocols & RTSP_LOWER_TRANS_UDP_MCAST) {
+ gchar *new;
+
+ GST_DEBUG_OBJECT (src, "adding UDP multicast");
+
+ /* we don't have 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 = g_strconcat (result, result[0] ? "," : "",
+ "RTP/AVP/UDP;multicast", NULL);
+ g_free (result);
+ result = new;
+ }
+ if (protocols & RTSP_LOWER_TRANS_TCP) {
+ gchar *new;
+
+ GST_DEBUG_OBJECT (src, "adding TCP");
+
+ new = g_strconcat (result, result[0] ? "," : "",
+ "RTP/AVP/TCP;unicast;interleaved=%%i1-%%i2", NULL);
+ g_free (result);
+ result = new;
+ }
+ *transports = result;
+
+ return RTSP_OK;
+
+ /* ERRORS */
+failed:
+ {
+ return res;
+ }
+}
+
+static RTSPResult
+gst_rtspsrc_configure_transports (GstRTSPStream * stream, gchar ** transports)
+{
+ GstRTSPSrc *src;
+ gint nr_udp, nr_int;
+ gchar *next, *p;
+ gint rtpport = 0, rtcpport = 0;
+ GString *str;
+
+ src = stream->parent;
+
+ /* find number of placeholders first */
+ if (strstr (*transports, "%%i2"))
+ nr_int = 2;
+ else if (strstr (*transports, "%%i1"))
+ nr_int = 1;
+ else
+ nr_int = 0;
+
+ if (strstr (*transports, "%%u2"))
+ nr_udp = 2;
+ else if (strstr (*transports, "%%u1"))
+ nr_udp = 1;
+ else
+ nr_udp = 0;
+
+ if (nr_udp == 0 && nr_int == 0)
+ goto done;
+
+ if (nr_udp > 0) {
+ if (!gst_rtspsrc_alloc_udp_ports (stream, &rtpport, &rtcpport))
+ goto failed;
+ }
+
+ str = g_string_new ("");
+ p = *transports;
+ while ((next = strstr (p, "%%"))) {
+ g_string_append_len (str, p, next - p);
+ if (next[2] == 'u') {
+ if (next[3] == '1')
+ g_string_append_printf (str, "%d", rtpport);
+ else if (next[3] == '2')
+ g_string_append_printf (str, "%d", rtcpport);
+ }
+ if (next[2] == 'i') {
+ if (next[3] == '1')
+ g_string_append_printf (str, "%d", src->free_channel);
+ else if (next[3] == '2')
+ g_string_append_printf (str, "%d", src->free_channel + 1);
+ }
+
+ p = next + 4;
+ }
+
+ g_free (*transports);
+ *transports = g_string_free (str, FALSE);
+
+done:
+ return RTSP_OK;
+
+ /* ERRORS */
+failed:
+ {
+ return RTSP_ERROR;
+ }
+}
+
static gboolean
gst_rtspsrc_open (GstRTSPSrc * src)
{
@@ -1321,7 +1560,7 @@ gst_rtspsrc_open (GstRTSPSrc * src)
guint size;
gint i, n_streams;
SDPMessage sdp = { 0 };
- GstRTSPProto protocols;
+ RTSPLowerTrans protocols;
GstRTSPStream *stream = NULL;
gchar *respcont = NULL;
@@ -1396,8 +1635,8 @@ gst_rtspsrc_open (GstRTSPSrc * src)
if (src->extension && src->extension->parse_sdp)
src->extension->parse_sdp (src->extension, &sdp);
- /* we initially allow all configured protocols. based on the replies from the
- * server we narrow them down. */
+ /* we initially allow all configured lower transports. based on the
+ * replies from the server we narrow them down. */
protocols = src->protocols;
/* setup streams */
@@ -1440,54 +1679,15 @@ gst_rtspsrc_open (GstRTSPSrc * src)
if (res < 0)
goto create_request_failed;
- transports = g_strdup ("");
- if (protocols & GST_RTSP_PROTO_UDP_UNICAST) {
- gchar *new;
- gint rtpport, rtcpport;
- gchar *trxparams;
-
- /* allocate two UDP ports */
- if (!gst_rtspsrc_stream_setup_rtp (stream, &rtpport, &rtcpport))
- goto setup_rtp_failed;
-
- GST_DEBUG_OBJECT (src, "setting up RTP ports %d-%d", rtpport, rtcpport);
+ /* create a string with all the transports */
+ res = gst_rtspsrc_create_transports_string (src, protocols, &transports);
+ if (res < 0)
+ goto setup_transport_failed;
- trxparams = g_strdup_printf ("client_port=%d-%d", rtpport, rtcpport);
- new = g_strconcat (transports, "RTP/AVP/UDP;unicast;", trxparams, NULL);
- g_free (trxparams);
- g_free (transports);
- transports = new;
- }
- if (protocols & GST_RTSP_PROTO_UDP_MULTICAST) {
- gchar *new;
-
- 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 =
- g_strconcat (transports, transports[0] ? "," : "",
- "RTP/AVP/UDP;multicast", NULL);
- g_free (transports);
- transports = new;
- }
- if (protocols & GST_RTSP_PROTO_TCP) {
- gchar *new, *interleaved;
- gint channel;
-
- GST_DEBUG_OBJECT (src, "setting up TCP");
-
- /* the channels for this stream is by default the next available number */
- channel = i * 2;
- interleaved = g_strdup_printf ("interleaved=%d-%d", channel, channel + 1);
- new =
- g_strconcat (transports, transports[0] ? "," : "",
- "RTP/AVP/TCP;unicast;", interleaved, NULL);
- g_free (interleaved);
- g_free (transports);
- transports = new;
- }
+ /* replace placeholders with real values */
+ res = gst_rtspsrc_configure_transports (stream, &transports);
+ if (res < 0)
+ goto setup_transport_failed;
/* select transport, copy is made when adding to header */
rtsp_message_add_header (&request, RTSP_HDR_TRANSPORT, transports);
@@ -1506,7 +1706,8 @@ gst_rtspsrc_open (GstRTSPSrc * src)
goto no_transport;
/* parse transport */
- rtsp_transport_parse (resptrans, &transport);
+ if (rtsp_transport_parse (resptrans, &transport) != RTSP_OK)
+ continue;
/* update allowed transports for other streams. once the transport of
* one stream has been determined, we make sure that all other streams
@@ -1514,18 +1715,24 @@ gst_rtspsrc_open (GstRTSPSrc * src)
switch (transport.lower_transport) {
case RTSP_LOWER_TRANS_TCP:
GST_DEBUG_OBJECT (src, "stream %d as TCP interleaved", i);
- protocols = GST_RTSP_PROTO_TCP;
+ protocols = RTSP_LOWER_TRANS_TCP;
src->interleaved = TRUE;
+ /* update free channels */
+ src->free_channel =
+ MAX (transport.interleaved.min, src->free_channel);
+ src->free_channel =
+ MAX (transport.interleaved.max, src->free_channel);
+ src->free_channel++;
break;
case RTSP_LOWER_TRANS_UDP_MCAST:
/* only allow multicast for other streams */
GST_DEBUG_OBJECT (src, "stream %d as UDP multicast", i);
- protocols = GST_RTSP_PROTO_UDP_MULTICAST;
+ protocols = RTSP_LOWER_TRANS_UDP_MCAST;
break;
case RTSP_LOWER_TRANS_UDP:
/* only allow unicast for other streams */
GST_DEBUG_OBJECT (src, "stream %d as UDP unicast", i);
- protocols = GST_RTSP_PROTO_UDP_UNICAST;
+ protocols = RTSP_LOWER_TRANS_UDP;
break;
default:
GST_DEBUG_OBJECT (src, "stream %d unknown transport %d", i,
@@ -1612,10 +1819,10 @@ wrong_content_type:
("Server does not support SDP, got %s.", respcont));
goto cleanup_error;
}
-setup_rtp_failed:
+setup_transport_failed:
{
GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
- ("Could not setup rtp."));
+ ("Could not setup transport."));
goto cleanup_error;
}
no_transport:
@@ -1843,7 +2050,8 @@ gst_rtspsrc_handle_message (GstBin * bin, GstMessage * message)
GST_DEBUG_OBJECT (bin, "timeout on UDP port");
gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_RECONNECT);
/* FIXME, we post an error message now to inform the user
- * that nothing happened. It's most likely a firewall thing. */
+ * that nothing happened. It's most likely a firewall thing. Ideally we
+ * notify the thread and redo the setup with only TCP. */
GST_ELEMENT_ERROR (rtspsrc, RESOURCE, READ, (NULL),
("Could not receive any UDP packets for %.4f seconds, maybe your "
"firewall is blocking it.",
@@ -1870,14 +2078,21 @@ gst_rtspsrc_change_state (GstElement * element, GstStateChange transition)
case GST_STATE_CHANGE_NULL_TO_READY:
break;
case GST_STATE_CHANGE_READY_TO_PAUSED:
+ /* the first free channel for interleaved mode */
+ rtspsrc->free_channel = 0;
rtspsrc->interleaved = FALSE;
gst_segment_init (&rtspsrc->segment, GST_FORMAT_TIME);
if (!gst_rtspsrc_open (rtspsrc))
goto open_failed;
break;
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
+ rtsp_connection_flush (rtspsrc->connection, FALSE);
gst_rtspsrc_play (rtspsrc);
break;
+ case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
+ case GST_STATE_CHANGE_PAUSED_TO_READY:
+ rtsp_connection_flush (rtspsrc->connection, TRUE);
+ break;
default:
break;
}
diff --git a/gst/rtsp/gstrtspsrc.h b/gst/rtsp/gstrtspsrc.h
index 8a9c74fe..94b9ed3d 100644
--- a/gst/rtsp/gstrtspsrc.h
+++ b/gst/rtsp/gstrtspsrc.h
@@ -72,21 +72,6 @@ typedef struct _GstRTSPSrcClass GstRTSPSrcClass;
#define GST_RTSP_LOOP_WAIT(rtsp) (g_cond_wait(GST_RTSP_LOOP_GET_COND (rtsp), GST_OBJECT_GET_LOCK (rtsp)))
#define GST_RTSP_LOOP_SIGNAL(rtsp) (g_cond_signal(GST_RTSP_LOOP_GET_COND (rtsp)))
-/**
- * GstRTSPProto:
- * @GST_RTSP_PROTO_UDP_UNICAST: Use unicast UDP transfer.
- * @GST_RTSP_PROTO_UDP_MULTICAST: Use multicast UDP transfer
- * @GST_RTSP_PROTO_TCP: Use TCP transfer.
- *
- * Flags with allowed protocols for the datatransfer.
- */
-typedef enum
-{
- GST_RTSP_PROTO_UDP_UNICAST = (1 << 0),
- GST_RTSP_PROTO_UDP_MULTICAST = (1 << 1),
- GST_RTSP_PROTO_TCP = (1 << 2),
-} GstRTSPProto;
-
typedef struct _GstRTSPStream GstRTSPStream;
struct _GstRTSPStream {
@@ -99,21 +84,18 @@ struct _GstRTSPStream {
GstFlowReturn last_ret;
/* for interleaved mode */
- gint rtpchannel;
- gint rtcpchannel;
+ gint channel[2];
GstCaps *caps;
+ GstPad *channelpad[2];
- /* our udp sources for RTP */
- GstElement *rtpsrc;
- GstElement *rtcpsrc;
+ /* our udp sources */
+ GstElement *udpsrc[2];
/* our udp sink back to the server */
- GstElement *rtcpsink;
+ GstElement *udpsink;
- /* the RTP decoder */
- GstElement *rtpdec;
- GstPad *rtpdecrtp;
- GstPad *rtpdecrtcp;
+ /* the session manager */
+ GstElement *sess;
/* state */
gint pt;
@@ -132,6 +114,7 @@ struct _GstRTSPSrc {
GStaticRecMutex *stream_rec_lock;
GstSegment segment;
gboolean running;
+ gint free_channel;
/* cond to signal loop */
GCond *loop_cond;
@@ -144,7 +127,7 @@ struct _GstRTSPSrc {
gchar *location;
RTSPUrl *url;
gchar *content_base;
- GstRTSPProto protocols;
+ RTSPLowerTrans protocols;
gboolean debug;
guint retry;
guint64 timeout;
diff --git a/gst/rtsp/rtspconnection.c b/gst/rtsp/rtspconnection.c
index e939a55d..27cc3ebb 100644
--- a/gst/rtsp/rtspconnection.c
+++ b/gst/rtsp/rtspconnection.c
@@ -753,3 +753,25 @@ rtsp_connection_free (RTSPConnection * conn)
return RTSP_OK;
}
+
+RTSPResult
+rtsp_connection_flush (RTSPConnection * conn, gboolean flush)
+{
+ g_return_val_if_fail (conn != NULL, RTSP_EINVAL);
+
+ if (flush) {
+ SEND_COMMAND (conn, CONTROL_STOP);
+ } else {
+ while (TRUE) {
+ gchar command;
+ int res;
+
+ READ_COMMAND (conn, command, res);
+ if (res <= 0) {
+ /* no more commands */
+ break;
+ }
+ }
+ }
+ return RTSP_OK;
+}
diff --git a/gst/rtsp/rtspdefs.c b/gst/rtsp/rtspdefs.c
index 706e5aad..0138d792 100644
--- a/gst/rtsp/rtspdefs.c
+++ b/gst/rtsp/rtspdefs.c
@@ -51,6 +51,7 @@ extern int h_errno;
static const gchar *rtsp_results[] = {
"OK",
/* errors */
+ "Generic error",
"Invalid parameter specified",
"Operation interrupted",
"Out of memory",
diff --git a/gst/rtsp/rtspdefs.h b/gst/rtsp/rtspdefs.h
index 4067488c..229fc42a 100644
--- a/gst/rtsp/rtspdefs.h
+++ b/gst/rtsp/rtspdefs.h
@@ -53,20 +53,21 @@ if (G_UNLIKELY ((res = (stmt)) != RTSP_OK)) goto label
typedef enum {
RTSP_OK = 0,
/* errors */
- RTSP_EINVAL = -1,
- RTSP_EINTR = -2,
- RTSP_ENOMEM = -3,
- RTSP_ERESOLV = -4,
- RTSP_ENOTIMPL = -5,
- RTSP_ESYS = -6,
- RTSP_EPARSE = -7,
- RTSP_EWSASTART = -8,
- RTSP_EWSAVERSION = -9,
- RTSP_EEOF = -10,
- RTSP_ENET = -11,
- RTSP_ENOTIP = -12,
-
- RTSP_ELAST = -13,
+ RTSP_ERROR = -1,
+ RTSP_EINVAL = -2,
+ RTSP_EINTR = -3,
+ RTSP_ENOMEM = -4,
+ RTSP_ERESOLV = -5,
+ RTSP_ENOTIMPL = -6,
+ RTSP_ESYS = -7,
+ RTSP_EPARSE = -8,
+ RTSP_EWSASTART = -9,
+ RTSP_EWSAVERSION = -10,
+ RTSP_EEOF = -11,
+ RTSP_ENET = -12,
+ RTSP_ENOTIP = -13,
+
+ RTSP_ELAST = -14,
} RTSPResult;
typedef enum {
diff --git a/gst/rtsp/rtspext.h b/gst/rtsp/rtspext.h
index 779c109b..3f41f310 100644
--- a/gst/rtsp/rtspext.h
+++ b/gst/rtsp/rtspext.h
@@ -45,6 +45,7 @@
#include <glib.h>
+#include "rtsptransport.h"
#include "sdp.h"
G_BEGIN_DECLS
@@ -65,6 +66,8 @@ struct _RTSPExtensionCtx
RTSPResult (*parse_sdp) (RTSPExtensionCtx *ctx, SDPMessage *sdp);
RTSPResult (*setup_media) (RTSPExtensionCtx *ctx, SDPMedia *media);
+ RTSPResult (*get_transports) (RTSPExtensionCtx *ctx, RTSPLowerTrans protocols, gchar **transport);
+
RTSPResult (*stream_select) (RTSPExtensionCtx *ctx);
};
diff --git a/gst/rtsp/rtspextwms.c b/gst/rtsp/rtspextwms.c
index 3769fcdb..81964e1e 100644
--- a/gst/rtsp/rtspextwms.c
+++ b/gst/rtsp/rtspextwms.c
@@ -45,22 +45,72 @@
#include "gstrtspsrc.h"
#include "rtspextwms.h"
+#define SERVER_PREFIX "WMServer/"
+#define HEADER_PREFIX "data:application/vnd.ms.wms-hdr.asfv1;base64,"
+
typedef struct _RTSPExtWMSCtx RTSPExtWMSCtx;
struct _RTSPExtWMSCtx
{
RTSPExtensionCtx ctx;
+
+ gboolean active;
};
-#define HEADER_PREFIX "data:application/vnd.ms.wms-hdr.asfv1;base64,"
+static RTSPResult
+rtsp_ext_wms_before_send (RTSPExtensionCtx * ctx, RTSPMessage * request)
+{
+ RTSPExtWMSCtx *rext = (RTSPExtWMSCtx *) ctx;
+
+ switch (request->type_data.request.method) {
+ case RTSP_OPTIONS:
+ {
+ /* activate ourselves with the first request */
+ rext->active = TRUE;
+ break;
+ }
+ default:
+ break;
+ }
+ return RTSP_OK;
+}
+
+static RTSPResult
+rtsp_ext_wms_after_send (RTSPExtensionCtx * ctx, RTSPMessage * req,
+ RTSPMessage * resp)
+{
+ RTSPExtWMSCtx *rext = (RTSPExtWMSCtx *) ctx;
+
+ switch (req->type_data.request.method) {
+ case RTSP_OPTIONS:
+ {
+ gchar *server = NULL;
+
+ rtsp_message_get_header (resp, RTSP_HDR_SERVER, &server);
+ if (g_str_has_prefix (server, SERVER_PREFIX))
+ rext->active = TRUE;
+ else
+ rext->active = FALSE;
+ break;
+ }
+ default:
+ break;
+ }
+ return RTSP_OK;
+}
+
static RTSPResult
rtsp_ext_wms_parse_sdp (RTSPExtensionCtx * ctx, SDPMessage * sdp)
{
GstRTSPSrc *src = (GstRTSPSrc *) ctx->src;
+ RTSPExtWMSCtx *rext = (RTSPExtWMSCtx *) ctx;
gchar *config, *maxps;
gint i;
+ if (!rext->active)
+ return RTSP_OK;
+
for (i = 0; (config = sdp_message_get_attribute_val_n (sdp, "pgmpu", i)); i++) {
if (g_str_has_prefix (config, HEADER_PREFIX)) {
config += strlen (HEADER_PREFIX);
@@ -86,9 +136,9 @@ rtsp_ext_wms_parse_sdp (RTSPExtensionCtx * ctx, SDPMessage * sdp)
/* ERRORS */
no_config:
{
- GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
- ("Could not find config SDP field."));
- return RTSP_ENOTIMPL;
+ GST_DEBUG_OBJECT (src, "Could not find config SDP field, deactivating.");
+ rext->active = FALSE;
+ return RTSP_OK;
}
}
@@ -99,6 +149,8 @@ rtsp_ext_wms_get_context (void)
res = g_new0 (RTSPExtWMSCtx, 1);
res->ctx.parse_sdp = rtsp_ext_wms_parse_sdp;
+ res->ctx.before_send = rtsp_ext_wms_before_send;
+ res->ctx.after_send = rtsp_ext_wms_after_send;
return (RTSPExtensionCtx *) res;
}
diff --git a/gst/rtsp/rtsptransport.c b/gst/rtsp/rtsptransport.c
index f9848c48..e833080a 100644
--- a/gst/rtsp/rtsptransport.c
+++ b/gst/rtsp/rtsptransport.c
@@ -50,13 +50,14 @@ typedef struct
const gchar *name;
const RTSPTransMode mode;
const gchar *gst_mime;
+ const gchar *manager;
} RTSPTransMap;
static const RTSPTransMap transports[] = {
- {"rtp", RTSP_TRANS_RTP, "application/x-rtp"},
- {"x-real-rdt", RTSP_TRANS_RDT, "application/x-rdt"},
- {"x-pn-tng", RTSP_TRANS_RDT, "application/x-rdt"},
- {NULL, RTSP_TRANS_UNKNOWN, "application/x-unknown"}
+ {"rtp", RTSP_TRANS_RTP, "application/x-rtp", "rtpdec"},
+ {"x-real-rdt", RTSP_TRANS_RDT, "application/x-rdt", NULL},
+ {"x-pn-tng", RTSP_TRANS_RDT, "application/x-rdt", NULL},
+ {NULL, RTSP_TRANS_UNKNOWN, NULL, NULL}
};
typedef struct
@@ -118,6 +119,36 @@ rtsp_transport_init (RTSPTransport * transport)
return RTSP_OK;
}
+RTSPResult
+rtsp_transport_get_mime (RTSPTransMode trans, const gchar ** mime)
+{
+ gint i;
+
+ g_return_val_if_fail (mime != NULL, RTSP_EINVAL);
+
+ for (i = 0; transports[i].name; i++)
+ if (transports[i].mode == trans)
+ break;
+ *mime = transports[i].gst_mime;
+
+ return RTSP_OK;
+}
+
+RTSPResult
+rtsp_transport_get_manager (RTSPTransMode trans, const gchar ** manager)
+{
+ gint i;
+
+ g_return_val_if_fail (manager != NULL, RTSP_EINVAL);
+
+ for (i = 0; transports[i].name; i++)
+ if (transports[i].mode == trans)
+ break;
+ *manager = transports[i].manager;
+
+ return RTSP_OK;
+}
+
static void
parse_mode (RTSPTransport * transport, gchar * str)
{
@@ -163,11 +194,11 @@ rtsp_transport_parse (const gchar * str, RTSPTransport * transport)
if (strstr (split[0], transports[i].name))
break;
transport->trans = transports[i].mode;
- for (i = 1; profiles[i].name; i++)
+ for (i = 0; profiles[i].name; i++)
if (strstr (split[0], profiles[i].name))
break;
transport->profile = profiles[i].profile;
- for (i = 1; ltrans[i].name; i++)
+ for (i = 0; ltrans[i].name; i++)
if (strstr (split[0], ltrans[i].name))
break;
transport->lower_transport = ltrans[i].ltrans;
diff --git a/gst/rtsp/rtsptransport.h b/gst/rtsp/rtsptransport.h
index 32a347b7..84c0996e 100644
--- a/gst/rtsp/rtsptransport.h
+++ b/gst/rtsp/rtsptransport.h
@@ -85,7 +85,7 @@ typedef struct _RTSPTransport {
gboolean append;
RTSPRange interleaved;
- /* mulitcast specific */
+ /* multicast specific */
gint ttl;
/* UDP specific */
@@ -97,12 +97,15 @@ typedef struct _RTSPTransport {
} RTSPTransport;
-RTSPResult rtsp_transport_new (RTSPTransport **transport);
-RTSPResult rtsp_transport_init (RTSPTransport *transport);
+RTSPResult rtsp_transport_new (RTSPTransport **transport);
+RTSPResult rtsp_transport_init (RTSPTransport *transport);
-RTSPResult rtsp_transport_parse (const gchar *str, RTSPTransport *transport);
+RTSPResult rtsp_transport_parse (const gchar *str, RTSPTransport *transport);
-RTSPResult rtsp_transport_free (RTSPTransport *transport);
+RTSPResult rtsp_transport_get_mime (RTSPTransMode trans, const gchar **mime);
+RTSPResult rtsp_transport_get_manager (RTSPTransMode trans, const gchar **manager);
+
+RTSPResult rtsp_transport_free (RTSPTransport *transport);
G_END_DECLS
generated by cgit (git 2.34.1) at 2025-10-31 17:24:59 +0000