* @short_description: A media stream
* @see_also: #GstRTSPMedia
*
- * Last reviewed on 2013-07-11 (1.0.0)
+ * The #GstRTSPStream object manages the data transport for one stream. It
+ * is created from a payloader element and a source pad that produce the RTP
+ * packets for the stream.
+ *
+ * With gst_rtsp_stream_join_bin() the streaming elements are added to the bin
+ * and rtpbin. gst_rtsp_stream_leave_bin() removes the elements again.
+ *
+ * The #GstRTSPStream will use the configured addresspool, as set with
+ * gst_rtsp_stream_set_address_pool(), to allocate multicast addresses for the
+ * stream. With gst_rtsp_stream_get_multicast_address() you can get the
+ * configured address.
+ *
+ * With gst_rtsp_stream_get_server_port () you can get the port that the server
+ * will use to receive RTCP. This is the part that the clients will use to send
+ * RTCP to.
+ *
+ * With gst_rtsp_stream_add_transport() destinations can be added where the
+ * stream should be sent to. Use gst_rtsp_stream_remove_transport() to remove
+ * the destination again.
+ *
+ * Last reviewed on 2013-07-16 (1.0.0)
*/
#include <stdlib.h>
gboolean is_joined;
gchar *control;
+ GstRTSPLowerTrans protocols;
+
/* pads on the rtpbin */
GstPad *send_rtp_sink;
GstPad *recv_sink[2];
gint dscp_qos;
};
-#define DEFAULT_CONTROL NULL
+#define DEFAULT_CONTROL NULL
+#define DEFAULT_PROTOCOLS GST_RTSP_LOWER_TRANS_UDP | GST_RTSP_LOWER_TRANS_UDP_MCAST | \
+ GST_RTSP_LOWER_TRANS_TCP
enum
{
PROP_0,
PROP_CONTROL,
+ PROP_PROTOCOLS,
PROP_LAST
};
"The control string for this stream", DEFAULT_CONTROL,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+ g_object_class_install_property (gobject_class, PROP_PROTOCOLS,
+ g_param_spec_flags ("protocols", "Protocols",
+ "Allowed lower transport protocols", GST_TYPE_RTSP_LOWER_TRANS,
+ DEFAULT_PROTOCOLS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
GST_DEBUG_CATEGORY_INIT (rtsp_stream_debug, "rtspstream", 0, "GstRTSPStream");
ssrc_stream_map_key = g_quark_from_static_string ("GstRTSPServer.stream");
priv->dscp_qos = -1;
priv->control = g_strdup (DEFAULT_CONTROL);
+ priv->protocols = DEFAULT_PROTOCOLS;
g_mutex_init (&priv->lock);
}
case PROP_CONTROL:
g_value_take_string (value, gst_rtsp_stream_get_control (stream));
break;
+ case PROP_PROTOCOLS:
+ g_value_set_flags (value, gst_rtsp_stream_get_protocols (stream));
+ break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
}
case PROP_CONTROL:
gst_rtsp_stream_set_control (stream, g_value_get_string (value));
break;
+ case PROP_PROTOCOLS:
+ gst_rtsp_stream_set_protocols (stream, g_value_get_flags (value));
+ break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
}
*
* Get the srcpad associated with @stream.
*
- * Return: the srcpad. Unref after usage.
+ * Returns: (transfer full): the srcpad. Unref after usage.
*/
GstPad *
gst_rtsp_stream_get_srcpad (GstRTSPStream * stream)
*
* Get the control string to identify this stream.
*
- * Return: the control string. free after usage.
+ * Returns: (transfer full): the control string. free after usage.
*/
gchar *
gst_rtsp_stream_get_control (GstRTSPStream * stream)
g_mutex_lock (&priv->lock);
if (priv->control)
- res = g_strcmp0 (priv->control, control);
+ res = (g_strcmp0 (priv->control, control) == 0);
else {
guint streamid;
sscanf (control, "stream=%u", &streamid);
return priv->dscp_qos;
}
+/**
+ * gst_rtsp_stream_set_protocols:
+ * @stream: a #GstRTSPStream
+ * @protocols: the new flags
+ *
+ * Configure the allowed lower transport for @stream.
+ */
+void
+gst_rtsp_stream_set_protocols (GstRTSPStream * stream,
+ GstRTSPLowerTrans protocols)
+{
+ GstRTSPStreamPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_STREAM (stream));
+
+ priv = stream->priv;
+
+ g_mutex_lock (&priv->lock);
+ priv->protocols = protocols;
+ g_mutex_unlock (&priv->lock);
+}
+
+/**
+ * gst_rtsp_stream_get_protocols:
+ * @stream: a #GstRTSPStream
+ *
+ * Get the allowed protocols of @stream.
+ *
+ * Returns: a #GstRTSPLowerTrans
+ */
+GstRTSPLowerTrans
+gst_rtsp_stream_get_protocols (GstRTSPStream * stream)
+{
+ GstRTSPStreamPrivate *priv;
+ GstRTSPLowerTrans res;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream),
+ GST_RTSP_LOWER_TRANS_UNKNOWN);
+
+ priv = stream->priv;
+
+ g_mutex_lock (&priv->lock);
+ res = priv->protocols;
+ g_mutex_unlock (&priv->lock);
+
+ return res;
+}
/**
* gst_rtsp_stream_set_address_pool:
g_mutex_lock (&priv->lock);
if (*addrp == NULL) {
+ GstRTSPAddressPoolResult res;
+
if (priv->pool == NULL)
goto no_pool;
- *addrp = gst_rtsp_address_pool_reserve_address (priv->pool, address,
- port, n_ports, ttl);
- if (*addrp == NULL)
+ res = gst_rtsp_address_pool_reserve_address (priv->pool, address,
+ port, n_ports, ttl, addrp);
+ if (res != GST_RTSP_ADDRESS_POOL_OK)
goto no_address;
} else {
if (strcmp ((*addrp)->address, address) ||
*
* Get the RTP session of this stream.
*
- * Returns: The RTP session of this stream. Unref after usage.
+ * Returns: (transfer full): The RTP session of this stream. Unref after usage.
*/
GObject *
gst_rtsp_stream_get_rtpsession (GstRTSPStream * stream)
gint i;
guint idx;
gchar *name;
- GstPad *pad, *teepad, *queuepad, *selpad;
+ GstPad *pad, *sinkpad, *selpad;
GstPadLinkReturn ret;
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
stream);
for (i = 0; i < 2; i++) {
+ GstPad *teepad, *queuepad;
/* For the sender we create this bit of pipeline for both
* RTP and RTCP. Sync and preroll are enabled on udpsink so
* we need to add a queue before appsink to make the pipeline
* | | | queue | | appsink |
* | src->sink src->sink |
* '-----' '---------' '---------'
+ *
+ * When only UDP is allowed, we skip the tee, queue and appsink and link the
+ * udpsink directly to the session.
*/
- /* make tee for RTP/RTCP */
- priv->tee[i] = gst_element_factory_make ("tee", NULL);
- gst_bin_add (bin, priv->tee[i]);
-
- /* and link to rtpbin send pad */
- pad = gst_element_get_static_pad (priv->tee[i], "sink");
- gst_pad_link (priv->send_src[i], pad);
- gst_object_unref (pad);
-
/* add udpsink */
gst_bin_add (bin, priv->udpsink[i]);
+ sinkpad = gst_element_get_static_pad (priv->udpsink[i], "sink");
- /* link tee to udpsink */
- teepad = gst_element_get_request_pad (priv->tee[i], "src_%u");
- pad = gst_element_get_static_pad (priv->udpsink[i], "sink");
- gst_pad_link (teepad, pad);
- gst_object_unref (pad);
- gst_object_unref (teepad);
-
- /* make queue */
- priv->appqueue[i] = gst_element_factory_make ("queue", NULL);
- gst_bin_add (bin, priv->appqueue[i]);
- /* and link to tee */
- teepad = gst_element_get_request_pad (priv->tee[i], "src_%u");
- pad = gst_element_get_static_pad (priv->appqueue[i], "sink");
- gst_pad_link (teepad, pad);
- gst_object_unref (pad);
- gst_object_unref (teepad);
-
- /* make appsink */
- priv->appsink[i] = gst_element_factory_make ("appsink", NULL);
- g_object_set (priv->appsink[i], "async", FALSE, "sync", FALSE, NULL);
- g_object_set (priv->appsink[i], "emit-signals", FALSE, NULL);
- gst_bin_add (bin, priv->appsink[i]);
- gst_app_sink_set_callbacks (GST_APP_SINK_CAST (priv->appsink[i]),
- &sink_cb, stream, NULL);
- /* and link to queue */
- queuepad = gst_element_get_static_pad (priv->appqueue[i], "src");
- pad = gst_element_get_static_pad (priv->appsink[i], "sink");
- gst_pad_link (queuepad, pad);
- gst_object_unref (pad);
- gst_object_unref (queuepad);
+ if (priv->protocols & GST_RTSP_LOWER_TRANS_TCP) {
+ /* make tee for RTP/RTCP */
+ priv->tee[i] = gst_element_factory_make ("tee", NULL);
+ gst_bin_add (bin, priv->tee[i]);
+
+ /* and link to rtpbin send pad */
+ pad = gst_element_get_static_pad (priv->tee[i], "sink");
+ gst_pad_link (priv->send_src[i], pad);
+ gst_object_unref (pad);
+
+ /* link tee to udpsink */
+ teepad = gst_element_get_request_pad (priv->tee[i], "src_%u");
+ gst_pad_link (teepad, sinkpad);
+ gst_object_unref (teepad);
+
+ /* make queue */
+ priv->appqueue[i] = gst_element_factory_make ("queue", NULL);
+ gst_bin_add (bin, priv->appqueue[i]);
+ /* and link to tee */
+ teepad = gst_element_get_request_pad (priv->tee[i], "src_%u");
+ pad = gst_element_get_static_pad (priv->appqueue[i], "sink");
+ gst_pad_link (teepad, pad);
+ gst_object_unref (pad);
+ gst_object_unref (teepad);
+
+ /* make appsink */
+ priv->appsink[i] = gst_element_factory_make ("appsink", NULL);
+ g_object_set (priv->appsink[i], "async", FALSE, "sync", FALSE, NULL);
+ g_object_set (priv->appsink[i], "emit-signals", FALSE, NULL);
+ gst_bin_add (bin, priv->appsink[i]);
+ gst_app_sink_set_callbacks (GST_APP_SINK_CAST (priv->appsink[i]),
+ &sink_cb, stream, NULL);
+ /* and link to queue */
+ queuepad = gst_element_get_static_pad (priv->appqueue[i], "src");
+ pad = gst_element_get_static_pad (priv->appsink[i], "sink");
+ gst_pad_link (queuepad, pad);
+ gst_object_unref (pad);
+ gst_object_unref (queuepad);
+ } else {
+ /* else only udpsink needed, link it to the session */
+ gst_pad_link (priv->send_src[i], sinkpad);
+ }
+ gst_object_unref (sinkpad);
/* For the receiver we create this bit of pipeline for both
* RTP and RTCP. We receive RTP/RTCP on appsrc and udpsrc
gst_object_unref (selpad);
}
- /* make and add appsrc */
- priv->appsrc[i] = gst_element_factory_make ("appsrc", NULL);
- gst_bin_add (bin, priv->appsrc[i]);
- /* and link to the funnel */
- selpad = gst_element_get_request_pad (priv->funnel[i], "sink_%u");
- pad = gst_element_get_static_pad (priv->appsrc[i], "src");
- gst_pad_link (pad, selpad);
- gst_object_unref (pad);
- gst_object_unref (selpad);
+ if (priv->protocols & GST_RTSP_LOWER_TRANS_TCP) {
+ /* make and add appsrc */
+ priv->appsrc[i] = gst_element_factory_make ("appsrc", NULL);
+ gst_bin_add (bin, priv->appsrc[i]);
+ /* and link to the funnel */
+ selpad = gst_element_get_request_pad (priv->funnel[i], "sink_%u");
+ pad = gst_element_get_static_pad (priv->appsrc[i], "src");
+ gst_pad_link (pad, selpad);
+ gst_object_unref (pad);
+ gst_object_unref (selpad);
+ }
/* check if we need to set to a special state */
if (state != GST_STATE_NULL) {
- gst_element_set_state (priv->udpsink[i], state);
- gst_element_set_state (priv->appsink[i], state);
- gst_element_set_state (priv->appqueue[i], state);
- gst_element_set_state (priv->tee[i], state);
- gst_element_set_state (priv->funnel[i], state);
- gst_element_set_state (priv->appsrc[i], state);
+ if (priv->udpsink[i])
+ gst_element_set_state (priv->udpsink[i], state);
+ if (priv->appsink[i])
+ gst_element_set_state (priv->appsink[i], state);
+ if (priv->appqueue[i])
+ gst_element_set_state (priv->appqueue[i], state);
+ if (priv->tee[i])
+ gst_element_set_state (priv->tee[i], state);
+ if (priv->funnel[i])
+ gst_element_set_state (priv->funnel[i], state);
+ if (priv->appsrc[i])
+ gst_element_set_state (priv->appsrc[i], state);
}
}
priv->send_rtp_sink = NULL;
for (i = 0; i < 2; i++) {
- gst_element_set_state (priv->udpsink[i], GST_STATE_NULL);
- gst_element_set_state (priv->appsink[i], GST_STATE_NULL);
- gst_element_set_state (priv->appqueue[i], GST_STATE_NULL);
- gst_element_set_state (priv->tee[i], GST_STATE_NULL);
- gst_element_set_state (priv->funnel[i], GST_STATE_NULL);
- gst_element_set_state (priv->appsrc[i], GST_STATE_NULL);
+ if (priv->udpsink[i])
+ gst_element_set_state (priv->udpsink[i], GST_STATE_NULL);
+ if (priv->appsink[i])
+ gst_element_set_state (priv->appsink[i], GST_STATE_NULL);
+ if (priv->appqueue[i])
+ gst_element_set_state (priv->appqueue[i], GST_STATE_NULL);
+ if (priv->tee[i])
+ gst_element_set_state (priv->tee[i], GST_STATE_NULL);
+ if (priv->funnel[i])
+ gst_element_set_state (priv->funnel[i], GST_STATE_NULL);
+ if (priv->appsrc[i])
+ gst_element_set_state (priv->appsrc[i], GST_STATE_NULL);
if (priv->udpsrc_v4[i]) {
/* and set udpsrc to NULL now before removing */
gst_element_set_locked_state (priv->udpsrc_v4[i], FALSE);
gst_element_set_state (priv->udpsrc_v6[i], GST_STATE_NULL);
gst_bin_remove (bin, priv->udpsrc_v6[i]);
}
- gst_bin_remove (bin, priv->udpsink[i]);
- gst_bin_remove (bin, priv->appsrc[i]);
- gst_bin_remove (bin, priv->appsink[i]);
- gst_bin_remove (bin, priv->appqueue[i]);
- gst_bin_remove (bin, priv->tee[i]);
- gst_bin_remove (bin, priv->funnel[i]);
+ if (priv->udpsink[i])
+ gst_bin_remove (bin, priv->udpsink[i]);
+ if (priv->appsrc[i])
+ gst_bin_remove (bin, priv->appsrc[i]);
+ if (priv->appsink[i])
+ gst_bin_remove (bin, priv->appsink[i]);
+ if (priv->appqueue[i])
+ gst_bin_remove (bin, priv->appqueue[i]);
+ if (priv->tee[i])
+ gst_bin_remove (bin, priv->tee[i]);
+ if (priv->funnel[i])
+ gst_bin_remove (bin, priv->funnel[i]);
gst_element_release_request_pad (rtpbin, priv->recv_sink[i]);
gst_object_unref (priv->recv_sink[i]);
g_return_val_if_fail (priv->is_joined, FALSE);
g_mutex_lock (&priv->lock);
- element = gst_object_ref (priv->appsrc[0]);
+ if (priv->appsrc[0])
+ element = gst_object_ref (priv->appsrc[0]);
+ else
+ element = NULL;
g_mutex_unlock (&priv->lock);
- ret = gst_app_src_push_buffer (GST_APP_SRC_CAST (element), buffer);
-
- gst_object_unref (element);
-
+ if (element) {
+ ret = gst_app_src_push_buffer (GST_APP_SRC_CAST (element), buffer);
+ gst_object_unref (element);
+ } else {
+ ret = GST_FLOW_OK;
+ }
return ret;
}
g_return_val_if_fail (priv->is_joined, FALSE);
g_mutex_lock (&priv->lock);
- element = gst_object_ref (priv->appsrc[1]);
+ if (priv->appsrc[1])
+ element = gst_object_ref (priv->appsrc[1]);
+ else
+ element = NULL;
g_mutex_unlock (&priv->lock);
- ret = gst_app_src_push_buffer (GST_APP_SRC_CAST (element), buffer);
-
- gst_object_unref (element);
-
+ if (element) {
+ ret = gst_app_src_push_buffer (GST_APP_SRC_CAST (element), buffer);
+ gst_object_unref (element);
+ } else {
+ ret = GST_FLOW_OK;
+ }
return ret;
}
projects / platform / upstream / gstreamer.git / blobdiff