GST_PAD_SOMETIMES,
GST_STATIC_CAPS ("application/x-rtp; application/x-rdt"));
+/* template used internally */
+static GstStaticPadTemplate anytemplate = GST_STATIC_PAD_TEMPLATE ("internal%d",
+ GST_PAD_SRC,
+ GST_PAD_SOMETIMES,
+ GST_STATIC_CAPS_ANY);
+
enum
{
/* FILL ME */
g_free (rtspsrc->req_location);
g_free (rtspsrc->content_base);
rtsp_url_free (rtspsrc->url);
- g_free (rtspsrc->addr);
g_static_rec_mutex_free (rtspsrc->state_rec_lock);
g_free (rtspsrc->state_rec_lock);
g_object_get (G_OBJECT (udpsrc0), "port", rtpport, NULL);
g_object_get (G_OBJECT (udpsrc1), "port", rtcpport, NULL);
- /* this should not happen */
+ /* this should not happen... */
if (*rtpport != tmp_rtp || *rtcpport != tmp_rtcp)
goto port_error;
}
}
+static gboolean
+gst_rtspsrc_handle_src_event (GstPad * pad, GstEvent * event)
+{
+ GstRTSPSrc *src;
+ gboolean res = TRUE;
+
+ src = GST_RTSPSRC_CAST (gst_pad_get_element_private (pad));
+
+ GST_DEBUG_OBJECT (src, "pad %s:%s received event %s",
+ GST_DEBUG_PAD_NAME (pad), GST_EVENT_TYPE_NAME (event));
+
+ switch (GST_EVENT_TYPE (event)) {
+ case GST_EVENT_QOS:
+ break;
+ case GST_EVENT_SEEK:
+ break;
+ case GST_EVENT_NAVIGATION:
+ break;
+ case GST_EVENT_LATENCY:
+ break;
+ default:
+ break;
+ }
+
+ return res;
+}
+
+static gboolean
+gst_rtspsrc_handle_src_query (GstPad * pad, GstQuery * query)
+{
+ GstRTSPSrc *src;
+ gboolean res = TRUE;
+
+ src = GST_RTSPSRC_CAST (gst_pad_get_element_private (pad));
+
+ GST_DEBUG_OBJECT (src, "pad %s:%s received query %s",
+ GST_DEBUG_PAD_NAME (pad), GST_QUERY_TYPE_NAME (query));
+
+ switch (GST_QUERY_TYPE (query)) {
+ case GST_QUERY_POSITION:
+ {
+ break;
+ }
+ case GST_QUERY_DURATION:
+ {
+ GstFormat format;
+
+ gst_query_parse_duration (query, &format, NULL);
+
+ switch (format) {
+ case GST_FORMAT_TIME:
+ break;
+ default:
+ res = FALSE;
+ break;
+ }
+ break;
+ }
+ case GST_QUERY_LATENCY:
+ {
+ /* we are live with a min latency of 0 and unlimted max latency */
+ gst_query_set_latency (query, TRUE, 0, -1);
+ break;
+ }
+ default:
+ break;
+ }
+
+ return res;
+}
+
static void
pad_unblocked (GstPad * pad, gboolean blocked, GstRTSPSrc * src)
{
}
}
-/* sets up all elements needed for streaming over the specified transport.
- * Does not yet expose the element pads, this will be done when there is actuall
- * dataflow detected, which might never happen when UDP is blocked in a
- * firewall, for example.
- */
+/* try to get and configure a manager */
static gboolean
-gst_rtspsrc_stream_configure_transport (GstRTSPStream * stream,
+gst_rtspsrc_stream_configure_manager (GstRTSPSrc * src, GstRTSPStream * stream,
RTSPTransport * transport)
{
- GstRTSPSrc *src;
- GstPad *outpad = NULL;
- GstPadTemplate *template;
- GstStateChangeReturn ret;
+ const gchar *manager;
gchar *name;
- GstStructure *s;
- const gchar *mime, *manager;
RTSPResult res;
-
- src = stream->parent;
-
- GST_DEBUG_OBJECT (src, "configuring transport for stream %p", stream);
-
- s = gst_caps_get_structure (stream->caps, 0);
-
- /* get the proper mime type for this stream now */
- if ((res = rtsp_transport_get_mime (transport->trans, &mime)) < 0)
- goto no_mime;
- if (!mime)
- goto no_mime;
-
- /* configure the final mime type */
- GST_DEBUG_OBJECT (src, "setting mime to %s", mime);
- gst_structure_set_name (s, mime);
+ GstStateChangeReturn ret;
/* find a manager */
if ((res = rtsp_transport_get_manager (transport->trans, &manager, 0)) < 0)
goto start_session_failure;
g_object_set (src->session, "latency", src->latency, NULL);
+
+ /* connect to signals if we did not already do so */
+ GST_DEBUG_OBJECT (src, "connect to signals on session manager");
+ src->session_sig_id =
+ g_signal_connect (src->session, "pad-added",
+ (GCallback) new_session_pad, src);
+ src->session_ptmap_id =
+ g_signal_connect (src->session, "request-pt-map",
+ (GCallback) request_pt_map, src);
}
/* we stream directly to the manager, get some pads. Each RTSP stream goes
stream->channelpad[1] = gst_element_get_request_pad (src->session, name);
g_free (name);
}
+
+
use_no_manager:
+ return TRUE;
- 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
- * 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;
- }
+ /* ERRORS */
+no_manager:
+ {
+ GST_DEBUG_OBJECT (src, "cannot get a session manager");
+ return FALSE;
+ }
+manager_failed:
+ {
+ GST_DEBUG_OBJECT (src, "no session manager element %s found", manager);
+ return FALSE;
+ }
+start_session_failure:
+ {
+ GST_DEBUG_OBJECT (src, "could not start session");
+ return FALSE;
+ }
+}
+
+/* free the UDP sources allocated when negotiating a transport.
+ * This function is called when the server negotiated to a transport where the
+ * UDP sources are not needed anymore, such as TCP or multicast. */
+static void
+gst_rtspsrc_stream_free_udp (GstRTSPStream * stream)
+{
+ gint i;
+
+ 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");
+/* for TCP, create pads to send and receive data to and from the manager and to
+ * intercept various events and queries
+ */
+static gboolean
+gst_rtspsrc_stream_configure_tcp (GstRTSPSrc * src, GstRTSPStream * stream,
+ RTSPTransport * transport, GstPad ** outpad)
+{
+ gchar *name;
+ GstPadTemplate *template;
+ GstPad *pad0, *pad1;
- /* create a new pad we will use to stream to */
- name = g_strdup_printf ("stream%d", stream->id);
- template = gst_static_pad_template_get (&rtptemplate);
- stream->channelpad[0] = gst_pad_new_from_template (template, name);
- gst_object_unref (template);
- g_free (name);
+ /* configure for interleaved delivery, nothing needs to be done
+ * here, the loop function will call the chain functions of the
+ * 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]);
- /* set caps and activate */
- gst_pad_use_fixed_caps (stream->channelpad[0]);
- gst_pad_set_active (stream->channelpad[0], TRUE);
+ /* we can remove the allocated UDP ports now */
+ gst_rtspsrc_stream_free_udp (stream);
- outpad = gst_object_ref (stream->channelpad[0]);
- } else {
- GST_DEBUG_OBJECT (src, "using manager source pad");
- /* we connected to pad-added signal to get pads from the manager */
- }
- } else {
- /* multicast was selected, create UDP sources and join the multicast
- * group. */
- if (transport->lower_transport == RTSP_LOWER_TRANS_UDP_MCAST) {
- gchar *uri;
-
- 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;
-
- /* take ownership */
- gst_object_ref (stream->udpsrc[0]);
- gst_object_sink (stream->udpsrc[0]);
-
- /* change state */
- gst_element_set_state (stream->udpsrc[0], GST_STATE_READY);
- }
+ /* no session manager, send data to srcpad directly */
+ if (!stream->channelpad[0]) {
+ GST_DEBUG_OBJECT (src, "no manager, creating pad");
- /* 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;
+ /* create a new pad we will use to stream to */
+ name = g_strdup_printf ("stream%d", stream->id);
+ template = gst_static_pad_template_get (&rtptemplate);
+ stream->channelpad[0] = gst_pad_new_from_template (template, name);
+ gst_object_unref (template);
+ g_free (name);
- /* take ownership */
- gst_object_ref (stream->udpsrc[1]);
- gst_object_sink (stream->udpsrc[1]);
+ /* set caps and activate */
+ gst_pad_use_fixed_caps (stream->channelpad[0]);
+ gst_pad_set_active (stream->channelpad[0], TRUE);
- gst_element_set_state (stream->udpsrc[1], GST_STATE_READY);
- }
+ *outpad = gst_object_ref (stream->channelpad[0]);
+ } else {
+ GstPadTemplate *template;
+
+ GST_DEBUG_OBJECT (src, "using manager source pad");
+
+ template = gst_static_pad_template_get (&anytemplate);
+
+ /* allocate pads for sending the channel data into the manager */
+ pad0 = gst_pad_new_from_template (template, "internal0");
+ gst_pad_set_event_function (pad0, gst_rtspsrc_handle_src_event);
+ gst_pad_set_query_function (pad0, gst_rtspsrc_handle_src_query);
+ gst_pad_link (pad0, stream->channelpad[0]);
+ stream->channelpad[0] = pad0;
+ gst_pad_set_element_private (pad0, src);
+
+ if (stream->channelpad[1]) {
+ /* if we have a sinkpad for the other channel, create a pad and link to the
+ * manager. */
+ pad1 = gst_pad_new_from_template (template, "internal1");
+ gst_pad_link (pad1, stream->channelpad[1]);
+ stream->channelpad[1] = pad1;
}
+ gst_object_unref (template);
+ }
+ return TRUE;
+}
- /* 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]) {
- gst_bin_add (GST_BIN_CAST (src), stream->udpsrc[0]);
-
- GST_DEBUG_OBJECT (src, "setting up UDP source");
-
- /* configure a timeout on the UDP port. When the timeout message is
- * posted, we assume UDP transport is not possible. We reconnect using TCP
- * if we can. */
- g_object_set (G_OBJECT (stream->udpsrc[0]), "timeout", src->timeout,
- NULL);
-
- /* get output pad of the UDP source. */
- outpad = gst_element_get_pad (stream->udpsrc[0], "src");
-
- /* save it so we can unblock */
- stream->blockedpad = outpad;
-
- /* configure pad block on the pad. As soon as there is dataflow on the
- * UDP source, we know that UDP is not blocked by a firewall and we can
- * configure all the streams to let the application autoplug decoders. */
- gst_pad_set_blocked_async (outpad, TRUE,
- (GstPadBlockCallback) pad_blocked, src);
-
- 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. */
- gst_pad_link (outpad, stream->channelpad[0]);
- gst_object_unref (outpad);
- outpad = NULL;
- /* we connected to pad-added signal to get pads from the manager */
- } else {
- GST_DEBUG_OBJECT (src, "using UDP src pad as output");
- }
- }
+/* For multicast create UDP sources and join the multicast group. */
+static gboolean
+gst_rtspsrc_stream_configure_mcast (GstRTSPSrc * src, GstRTSPStream * stream,
+ RTSPTransport * transport, GstPad ** outpad)
+{
+ gchar *uri;
- if (stream->udpsrc[1]) {
- gst_bin_add (GST_BIN_CAST (src), stream->udpsrc[1]);
+ GST_DEBUG_OBJECT (src, "creating UDP sources for multicast");
- if (stream->channelpad[1]) {
- GstPad *pad;
+ /* we can remove the allocated UDP ports now */
+ gst_rtspsrc_stream_free_udp (stream);
- GST_DEBUG_OBJECT (src, "connecting UDP source 1 to manager");
+ /* 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;
- pad = gst_element_get_pad (stream->udpsrc[1], "src");
- gst_pad_link (pad, stream->channelpad[1]);
- gst_object_unref (pad);
- }
+ /* take ownership */
+ gst_object_ref (stream->udpsrc[0]);
+ gst_object_sink (stream->udpsrc[0]);
+
+ /* change state */
+ gst_element_set_state (stream->udpsrc[0], GST_STATE_READY);
+ }
+
+ /* 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;
+
+ /* take ownership */
+ gst_object_ref (stream->udpsrc[1]);
+ gst_object_sink (stream->udpsrc[1]);
+
+ gst_element_set_state (stream->udpsrc[1], GST_STATE_READY);
+ }
+ return TRUE;
+
+ /* ERRORS */
+no_element:
+ {
+ GST_DEBUG_OBJECT (src, "no UDP source element found");
+ return FALSE;
+ }
+}
+
+/* configure the remainder of the UDP ports */
+static gboolean
+gst_rtspsrc_stream_configure_udp (GstRTSPSrc * src, GstRTSPStream * stream,
+ RTSPTransport * transport, GstPad ** outpad)
+{
+ /* 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]) {
+ gst_bin_add (GST_BIN_CAST (src), stream->udpsrc[0]);
+
+ GST_DEBUG_OBJECT (src, "setting up UDP source");
+
+ /* configure a timeout on the UDP port. When the timeout message is
+ * posted, we assume UDP transport is not possible. We reconnect using TCP
+ * if we can. */
+ g_object_set (G_OBJECT (stream->udpsrc[0]), "timeout", src->timeout, NULL);
+
+ /* get output pad of the UDP source. */
+ *outpad = gst_element_get_pad (stream->udpsrc[0], "src");
+
+ /* save it so we can unblock */
+ stream->blockedpad = *outpad;
+
+ /* configure pad block on the pad. As soon as there is dataflow on the
+ * UDP source, we know that UDP is not blocked by a firewall and we can
+ * configure all the streams to let the application autoplug decoders. */
+ gst_pad_set_blocked_async (stream->blockedpad, TRUE,
+ (GstPadBlockCallback) pad_blocked, src);
+
+ 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. */
+ gst_pad_link (*outpad, stream->channelpad[0]);
+ gst_object_unref (*outpad);
+ *outpad = NULL;
+ /* we connected to pad-added signal to get pads from the manager */
+ } else {
+ GST_DEBUG_OBJECT (src, "using UDP src pad as output");
}
- /* configure udpsink back to the server for RTCP messages. */
- {
+ }
+
+ /* RTCP port */
+ if (stream->udpsrc[1]) {
+ gst_bin_add (GST_BIN_CAST (src), stream->udpsrc[1]);
+
+ if (stream->channelpad[1]) {
GstPad *pad;
- gint port;
- gchar *destination, *uri;
- /* get host and port */
- if (transport->lower_transport == RTSP_LOWER_TRANS_UDP_MCAST)
- port = transport->port.max;
- else
- port = transport->server_port.max;
+ 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);
+ } else {
+ /* leave unlinked */
+ }
+ }
+ return TRUE;
+}
+
+/* configure the UDP sink back to the server for status reports */
+static gboolean
+gst_rtspsrc_stream_configure_udp_sink (GstRTSPSrc * src, GstRTSPStream * stream,
+ RTSPTransport * transport)
+{
+ GstPad *pad;
+ gint port;
+ gchar *destination, *uri, *name;
- /* first take the source, then the endpoint to figure out where to send
- * the RTCP. */
- destination = transport->source;
- if (destination == NULL)
- destination = src->connection->ip;
+ /* get host and port */
+ if (transport->lower_transport == RTSP_LOWER_TRANS_UDP_MCAST)
+ port = transport->port.max;
+ else
+ port = transport->server_port.max;
- GST_DEBUG_OBJECT (src, "configure UDP sink for %s:%d", destination, port);
+ /* first take the source, then the endpoint to figure out where to send
+ * the RTCP. */
+ destination = transport->source;
+ if (destination == NULL)
+ destination = src->connection->ip;
- uri = g_strdup_printf ("udp://%s:%d", destination, port);
- stream->udpsink = gst_element_make_from_uri (GST_URI_SINK, uri, NULL);
- g_free (uri);
- if (stream->udpsink == NULL)
- goto no_sink_element;
+ GST_DEBUG_OBJECT (src, "configure UDP sink for %s:%d", destination, port);
- /* we keep this playing always */
- gst_element_set_locked_state (stream->udpsink, TRUE);
- gst_element_set_state (stream->udpsink, GST_STATE_PLAYING);
+ uri = g_strdup_printf ("udp://%s:%d", destination, port);
+ stream->udpsink = gst_element_make_from_uri (GST_URI_SINK, uri, NULL);
+ g_free (uri);
+ if (stream->udpsink == NULL)
+ goto no_sink_element;
- /* no sync needed */
- g_object_set (G_OBJECT (stream->udpsink), "sync", FALSE, NULL);
+ /* we keep this playing always */
+ gst_element_set_locked_state (stream->udpsink, TRUE);
+ gst_element_set_state (stream->udpsink, GST_STATE_PLAYING);
- gst_object_ref (stream->udpsink);
- gst_bin_add (GST_BIN_CAST (src), stream->udpsink);
+ /* no sync needed */
+ g_object_set (G_OBJECT (stream->udpsink), "sync", FALSE, NULL);
- stream->rtcppad = gst_element_get_pad (stream->udpsink, "sink");
+ gst_object_ref (stream->udpsink);
+ gst_bin_add (GST_BIN_CAST (src), stream->udpsink);
- /* get session RTCP pad */
- name = g_strdup_printf ("send_rtcp_src_%d", stream->id);
- pad = gst_element_get_request_pad (src->session, name);
- g_free (name);
+ stream->rtcppad = gst_element_get_pad (stream->udpsink, "sink");
- /* and link */
- gst_pad_link (pad, stream->rtcppad);
- }
+ /* get session RTCP pad */
+ name = g_strdup_printf ("send_rtcp_src_%d", stream->id);
+ pad = gst_element_get_request_pad (src->session, name);
+ g_free (name);
+
+ /* and link */
+ gst_pad_link (pad, stream->rtcppad);
+
+ return TRUE;
+
+ /* ERRORS */
+no_sink_element:
+ {
+ GST_DEBUG_OBJECT (src, "no UDP sink element found");
+ return FALSE;
}
+}
- if (src->session && !src->session_sig_id) {
- GST_DEBUG_OBJECT (src, "connect to signals on session manager");
- src->session_sig_id =
- g_signal_connect (src->session, "pad-added",
- (GCallback) new_session_pad, src);
- src->session_ptmap_id =
- g_signal_connect (src->session, "request-pt-map",
- (GCallback) request_pt_map, src);
+/* sets up all elements needed for streaming over the specified transport.
+ * Does not yet expose the element pads, this will be done when there is actuall
+ * dataflow detected, which might never happen when UDP is blocked in a
+ * firewall, for example.
+ */
+static gboolean
+gst_rtspsrc_stream_configure_transport (GstRTSPStream * stream,
+ RTSPTransport * transport)
+{
+ GstRTSPSrc *src;
+ GstPad *outpad = NULL;
+ GstPadTemplate *template;
+ gchar *name;
+ GstStructure *s;
+ const gchar *mime;
+ RTSPResult res;
+
+ src = stream->parent;
+
+ GST_DEBUG_OBJECT (src, "configuring transport for stream %p", stream);
+
+ s = gst_caps_get_structure (stream->caps, 0);
+
+ /* get the proper mime type for this stream now */
+ if ((res = rtsp_transport_get_mime (transport->trans, &mime)) < 0)
+ goto unknown_transport;
+ if (!mime)
+ goto unknown_transport;
+
+ /* configure the final mime type */
+ GST_DEBUG_OBJECT (src, "setting mime to %s", mime);
+ gst_structure_set_name (s, mime);
+
+ /* try to get and configure a manager, channelpad[0-1] will be configured with
+ * the pads for the manager, or NULL when no manager is needed. */
+ if (!gst_rtspsrc_stream_configure_manager (src, stream, transport))
+ goto no_manager;
+
+ switch (transport->lower_transport) {
+ case RTSP_LOWER_TRANS_TCP:
+ if (!gst_rtspsrc_stream_configure_tcp (src, stream, transport, &outpad))
+ goto transport_failed;
+ break;
+ case RTSP_LOWER_TRANS_UDP_MCAST:
+ if (!gst_rtspsrc_stream_configure_mcast (src, stream, transport, &outpad))
+ goto transport_failed;
+ /* fallthrough, the rest is the same for UDP and MCAST */
+ case RTSP_LOWER_TRANS_UDP:
+ if (!gst_rtspsrc_stream_configure_udp (src, stream, transport, &outpad))
+ goto transport_failed;
+ /* configure udpsink back to the server for RTCP messages. */
+ if (!gst_rtspsrc_stream_configure_udp_sink (src, stream, transport))
+ goto transport_failed;
+ break;
+ default:
+ goto unknown_transport;
}
if (outpad) {
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;
- }
-manager_failed:
- {
- GST_DEBUG_OBJECT (src, "no session manager element %s found", manager);
- return FALSE;
- }
-no_element:
+transport_failed:
{
- GST_DEBUG_OBJECT (src, "no UDP source element found");
+ GST_DEBUG_OBJECT (src, "failed to configure transport");
return FALSE;
}
-no_sink_element:
+unknown_transport:
{
- GST_DEBUG_OBJECT (src, "no UDP sink element found");
+ GST_DEBUG_OBJECT (src, "unknown transport");
return FALSE;
}
-start_session_failure:
+no_manager:
{
- GST_DEBUG_OBJECT (src, "could not start session");
+ GST_DEBUG_OBJECT (src, "cannot get a session manager");
return FALSE;
}
}
for (streams = src->streams; streams; streams = g_list_next (streams)) {
GstRTSPStream *ostream = (GstRTSPStream *) streams->data;
- /* only pads that have a connection to the outside world */
+ /* only streams that have a connection to the outside world */
if (ostream->srcpad == NULL)
continue;
rtsp_connection_flush (src->connection, FALSE);
goto interrupt;
case RTSP_ETIMEOUT:
- /* ignore result, a warning was posted */
+ /* send keep-alive, ignore the result, a warning will be posted. */
GST_DEBUG_OBJECT (src, "timout, sending keep-alive");
res = gst_rtspsrc_send_keep_alive (src);
continue;
if (src->extension && src->extension->parse_sdp)
src->extension->parse_sdp (src->extension, &sdp);
- /* parse address */
+ /* parse range */
{
- SDPOrigin *origin;
+ gchar *range;
- origin = sdp_message_get_origin (&sdp);
+ range = sdp_message_get_attribute_val (&sdp, "range");
- g_free (src->addr);
- src->addr = g_strdup (origin->addr);
+ GST_DEBUG_OBJECT (src, "got range: %s", GST_STR_NULL (range));
}
/* create streams */
timebase = atol (fields[j] + 8);
}
}
+ g_strfreev (fields);
/* now we need to store the values in the caps of the stream and make sure
* that the UDP elements have the same caps property set before they receive
* the first buffer. */
/* parse the RTP-Info header field (if ANY) to get the base seqnum and timestamp
* for the RTP packets. If this is not present, we assume all starts from 0...
- * FIXME, this is info for the RTP session manager ideally. */
+ * This is info for the RTP session manager that we pass to it in caps. */
rtsp_message_get_header (&response, RTSP_HDR_RTP_INFO, &rtpinfo);
if (rtpinfo)
gst_rtspsrc_parse_rtpinfo (src, rtpinfo);
if (src->state == RTSP_STATE_READY)
goto was_paused;
+ /* wait for streaming to finish */
+ GST_RTSP_STREAM_LOCK (src);
+ GST_RTSP_STREAM_UNLOCK (src);
+
+ rtsp_connection_flush (src->connection, FALSE);
+
/* do pause */
res = rtsp_message_init_request (&request, RTSP_PAUSE, src->req_location);
if (res < 0)
projects / platform / upstream / gst-plugins-good.git / commitdiff