* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
+/**
+ * SECTION:rtsp-stream
+ * @short_description: A media stream
+ * @see_also: #GstRTSPMedia
+ *
+ * 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>
#include <stdio.h>
gboolean is_joined;
gchar *control;
+ GstRTSPProfile profiles;
+ GstRTSPLowerTrans protocols;
+
/* pads on the rtpbin */
GstPad *send_rtp_sink;
GstPad *recv_sink[2];
GList *transports;
gint dscp_qos;
+
+ /* stream blocking */
+ gulong blocked_id;
+ gboolean blocking;
};
-#define DEFAULT_CONTROL NULL
+#define DEFAULT_CONTROL NULL
+#define DEFAULT_PROFILES GST_RTSP_PROFILE_AVP
+#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_PROFILES,
+ 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_PROFILES,
+ g_param_spec_flags ("profiles", "Profiles",
+ "Allowed transfer profiles", GST_TYPE_RTSP_PROFILE,
+ DEFAULT_PROFILES, 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->profiles = DEFAULT_PROFILES;
+ 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_PROFILES:
+ g_value_set_flags (value, gst_rtsp_stream_get_profiles (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_PROFILES:
+ gst_rtsp_stream_set_profiles (stream, g_value_get_flags (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);
}
}
/**
+ * gst_rtsp_stream_get_pt:
+ * @stream: a #GstRTSPStream
+ *
+ * Get the stream payload type.
+ *
+ * Return: the stream payload type.
+ */
+guint
+gst_rtsp_stream_get_pt (GstRTSPStream * stream)
+{
+ GstRTSPStreamPrivate *priv;
+ guint pt;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), -1);
+
+ priv = stream->priv;
+
+ g_object_get (G_OBJECT (priv->payloader), "pt", &pt, NULL);
+
+ return pt;
+}
+
+/**
* gst_rtsp_stream_get_srcpad:
* @stream: a #GstRTSPStream
*
* 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);
- res = (streamid == priv->idx);
+
+ if (sscanf (control, "stream=%u", &streamid) > 0)
+ res = (streamid == priv->idx);
+ else
+ res = FALSE;
}
g_mutex_unlock (&priv->lock);
return priv->dscp_qos;
}
+/**
+ * gst_rtsp_stream_is_transport_supported:
+ * @stream: a #GstRTSPStream
+ * @transport: a #GstRTSPTransport
+ *
+ * Check if @transport can be handled by stream
+ *
+ * Returns: %TRUE if @transport can be handled by @stream.
+ */
+gboolean
+gst_rtsp_stream_is_transport_supported (GstRTSPStream * stream,
+ GstRTSPTransport * transport)
+{
+ GstRTSPStreamPrivate *priv;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
+
+ priv = stream->priv;
+
+ g_mutex_lock (&priv->lock);
+ if (transport->trans != GST_RTSP_TRANS_RTP)
+ goto unsupported_transmode;
+
+ if (!(transport->profile & priv->profiles))
+ goto unsupported_profile;
+
+ if (!(transport->lower_transport & priv->protocols))
+ goto unsupported_ltrans;
+
+ g_mutex_unlock (&priv->lock);
+
+ return TRUE;
+
+ /* ERRORS */
+unsupported_transmode:
+ {
+ GST_DEBUG ("unsupported transport mode %d", transport->trans);
+ g_mutex_unlock (&priv->lock);
+ return FALSE;
+ }
+unsupported_profile:
+ {
+ GST_DEBUG ("unsupported profile %d", transport->profile);
+ g_mutex_unlock (&priv->lock);
+ return FALSE;
+ }
+unsupported_ltrans:
+ {
+ GST_DEBUG ("unsupported lower transport %d", transport->lower_transport);
+ g_mutex_unlock (&priv->lock);
+ return FALSE;
+ }
+}
+
+/**
+ * gst_rtsp_stream_set_profiles:
+ * @stream: a #GstRTSPStream
+ * @profiles: the new profiles
+ *
+ * Configure the allowed profiles for @stream.
+ */
+void
+gst_rtsp_stream_set_profiles (GstRTSPStream * stream, GstRTSPProfile profiles)
+{
+ GstRTSPStreamPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_STREAM (stream));
+
+ priv = stream->priv;
+
+ g_mutex_lock (&priv->lock);
+ priv->profiles = profiles;
+ g_mutex_unlock (&priv->lock);
+}
+
+/**
+ * gst_rtsp_stream_get_profiles:
+ * @stream: a #GstRTSPStream
+ *
+ * Get the allowed profiles of @stream.
+ *
+ * Returns: a #GstRTSPProfile
+ */
+GstRTSPProfile
+gst_rtsp_stream_get_profiles (GstRTSPStream * stream)
+{
+ GstRTSPStreamPrivate *priv;
+ GstRTSPProfile res;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), GST_RTSP_PROFILE_UNKNOWN);
+
+ priv = stream->priv;
+
+ g_mutex_lock (&priv->lock);
+ res = priv->profiles;
+ g_mutex_unlock (&priv->lock);
+
+ return res;
+}
+
+/**
+ * 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) ||
gst_element_set_state (udpsink0, GST_STATE_NULL);
gst_object_unref (udpsink0);
}
- if (udpsink1) {
- gst_element_set_state (udpsink1, GST_STATE_NULL);
- gst_object_unref (udpsink1);
- }
if (inetaddr)
g_object_unref (inetaddr);
g_list_free_full (rejected_addresses,
*
* 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)
* @rtpbin: a rtpbin element in @bin
* @state: the target state of the new elements
*
- * Join the #Gstbin @bin that contains the element @rtpbin.
+ * Join the #GstBin @bin that contains the element @rtpbin.
*
* @stream will link to @rtpbin, which must be inside @bin. The elements
* added to @bin will be set to the state given in @state.
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]);
was_not_joined:
{
+ g_mutex_unlock (&priv->lock);
return TRUE;
}
}
/**
* gst_rtsp_stream_get_rtpinfo:
* @stream: a #GstRTSPStream
- * @rtptime: result RTP timestamp
- * @seq: result RTP seqnum
+ * @rtptime: (allow-none): result RTP timestamp
+ * @seq: (allow-none): result RTP seqnum
+ * @clock_rate: the clock rate
+ * @running_time: (allow-none): result running-time
*
- * Retrieve the current rtptime and seq. This is used to
+ * Retrieve the current rtptime, seq and running-time. This is used to
* construct a RTPInfo reply header.
*
- * Returns: %TRUE when rtptime and seq could be determined.
+ * Returns: %TRUE when rtptime, seq and running-time could be determined.
*/
gboolean
gst_rtsp_stream_get_rtpinfo (GstRTSPStream * stream,
- guint * rtptime, guint * seq)
+ guint * rtptime, guint * seq, guint * clock_rate,
+ GstClockTime * running_time)
{
GstRTSPStreamPrivate *priv;
+ GstStructure *stats;
GObjectClass *payobjclass;
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
- g_return_val_if_fail (rtptime != NULL, FALSE);
- g_return_val_if_fail (seq != NULL, FALSE);
priv = stream->priv;
payobjclass = G_OBJECT_GET_CLASS (priv->payloader);
- if (!g_object_class_find_property (payobjclass, "seqnum") ||
- !g_object_class_find_property (payobjclass, "timestamp"))
- return FALSE;
+ g_mutex_lock (&priv->lock);
+
+ if (g_object_class_find_property (payobjclass, "stats")) {
+ g_object_get (priv->payloader, "stats", &stats, NULL);
+ if (stats == NULL)
+ goto no_stats;
+
+ if (seq)
+ gst_structure_get_uint (stats, "seqnum", seq);
+
+ if (rtptime)
+ gst_structure_get_uint (stats, "timestamp", rtptime);
+
+ if (running_time)
+ gst_structure_get_clock_time (stats, "running-time", running_time);
+
+ if (clock_rate) {
+ gst_structure_get_uint (stats, "clock-rate", clock_rate);
+ if (*clock_rate == 0 && running_time)
+ *running_time = GST_CLOCK_TIME_NONE;
+ }
+ gst_structure_free (stats);
+ } else {
+ if (!g_object_class_find_property (payobjclass, "seqnum") ||
+ !g_object_class_find_property (payobjclass, "timestamp"))
+ goto no_stats;
+
+ if (seq)
+ g_object_get (priv->payloader, "seqnum", seq, NULL);
+
+ if (rtptime)
+ g_object_get (priv->payloader, "timestamp", rtptime, NULL);
- g_object_get (priv->payloader, "seqnum", seq, "timestamp", rtptime, NULL);
+ if (running_time)
+ *running_time = GST_CLOCK_TIME_NONE;
+ }
+ g_mutex_unlock (&priv->lock);
return TRUE;
+
+ /* ERRORS */
+no_stats:
+ {
+ GST_WARNING ("Could not get payloader stats");
+ g_mutex_unlock (&priv->lock);
+ return FALSE;
+ }
}
/**
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;
}
}
if (add) {
- GST_INFO ("adding %s:%d-%d", dest, min, max);
- g_signal_emit_by_name (priv->udpsink[0], "add", dest, min, NULL);
- g_signal_emit_by_name (priv->udpsink[1], "add", dest, max, NULL);
if (ttl > 0) {
GST_INFO ("setting ttl-mc %d", ttl);
g_object_set (G_OBJECT (priv->udpsink[0]), "ttl-mc", ttl, NULL);
g_object_set (G_OBJECT (priv->udpsink[1]), "ttl-mc", ttl, NULL);
}
+ GST_INFO ("adding %s:%d-%d", dest, min, max);
+ g_signal_emit_by_name (priv->udpsink[0], "add", dest, min, NULL);
+ g_signal_emit_by_name (priv->udpsink[1], "add", dest, max, NULL);
priv->transports = g_list_prepend (priv->transports, trans);
} else {
GST_INFO ("removing %s:%d-%d", dest, min, max);
return res;
}
+
+/**
+ * gst_rtsp_stream_get_rtp_socket:
+ * @stream: a #GstRTSPStream
+ * @family: the socket family
+ *
+ * Get the RTP socket from @stream for a @family.
+ *
+ * @stream must be joined to a bin.
+ *
+ * Returns: (transfer full): the RTP socket or %NULL if no socket could be
+ * allocated for @family. Unref after usage
+ */
+GSocket *
+gst_rtsp_stream_get_rtp_socket (GstRTSPStream * stream, GSocketFamily family)
+{
+ GstRTSPStreamPrivate *priv = GST_RTSP_STREAM_GET_PRIVATE (stream);
+ GSocket *socket;
+ const gchar *name;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
+ g_return_val_if_fail (family == G_SOCKET_FAMILY_IPV4 ||
+ family == G_SOCKET_FAMILY_IPV6, NULL);
+ g_return_val_if_fail (priv->udpsink[0], NULL);
+
+ if (family == G_SOCKET_FAMILY_IPV6)
+ name = "socket-v6";
+ else
+ name = "socket";
+
+ g_object_get (priv->udpsink[0], name, &socket, NULL);
+
+ return socket;
+}
+
+/**
+ * gst_rtsp_stream_get_rtcp_socket:
+ * @stream: a #GstRTSPStream
+ * @family: the socket family
+ *
+ * Get the RTCP socket from @stream for a @family.
+ *
+ * @stream must be joined to a bin.
+ *
+ * Returns: (transfer full): the RTCP socket or %NULL if no socket could be
+ * allocated for @family. Unref after usage
+ */
+GSocket *
+gst_rtsp_stream_get_rtcp_socket (GstRTSPStream * stream, GSocketFamily family)
+{
+ GstRTSPStreamPrivate *priv = GST_RTSP_STREAM_GET_PRIVATE (stream);
+ GSocket *socket;
+ const gchar *name;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
+ g_return_val_if_fail (family == G_SOCKET_FAMILY_IPV4 ||
+ family == G_SOCKET_FAMILY_IPV6, NULL);
+ g_return_val_if_fail (priv->udpsink[1], NULL);
+
+ if (family == G_SOCKET_FAMILY_IPV6)
+ name = "socket-v6";
+ else
+ name = "socket";
+
+ g_object_get (priv->udpsink[1], name, &socket, NULL);
+
+ return socket;
+}
+
+/**
+ * gst_rtsp_stream_transport_filter:
+ * @stream: a #GstRTSPStream
+ * @func: (scope call) (allow-none): a callback
+ * @user_data: user data passed to @func
+ *
+ * Call @func for each transport managed by @stream. The result value of @func
+ * determines what happens to the transport. @func will be called with @stream
+ * locked so no further actions on @stream can be performed from @func.
+ *
+ * If @func returns #GST_RTSP_FILTER_REMOVE, the transport will be removed from
+ * @stream.
+ *
+ * If @func returns #GST_RTSP_FILTER_KEEP, the transport will remain in @stream.
+ *
+ * If @func returns #GST_RTSP_FILTER_REF, the transport will remain in @stream but
+ * will also be added with an additional ref to the result #GList of this
+ * function..
+ *
+ * When @func is %NULL, #GST_RTSP_FILTER_REF will be assumed for each transport.
+ *
+ * Returns: (element-type GstRTSPStreamTransport) (transfer full): a #GList with all
+ * transports for which @func returned #GST_RTSP_FILTER_REF. After usage, each
+ * element in the #GList should be unreffed before the list is freed.
+ */
+GList *
+gst_rtsp_stream_transport_filter (GstRTSPStream * stream,
+ GstRTSPStreamTransportFilterFunc func, gpointer user_data)
+{
+ GstRTSPStreamPrivate *priv;
+ GList *result, *walk, *next;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
+
+ priv = stream->priv;
+
+ result = NULL;
+
+ g_mutex_lock (&priv->lock);
+ for (walk = priv->transports; walk; walk = next) {
+ GstRTSPStreamTransport *trans = walk->data;
+ GstRTSPFilterResult res;
+
+ next = g_list_next (walk);
+
+ if (func)
+ res = func (stream, trans, user_data);
+ else
+ res = GST_RTSP_FILTER_REF;
+
+ switch (res) {
+ case GST_RTSP_FILTER_REMOVE:
+ update_transport (stream, trans, FALSE);
+ break;
+ case GST_RTSP_FILTER_REF:
+ result = g_list_prepend (result, g_object_ref (trans));
+ break;
+ case GST_RTSP_FILTER_KEEP:
+ default:
+ break;
+ }
+ }
+ g_mutex_unlock (&priv->lock);
+
+ return result;
+}
+
+static GstPadProbeReturn
+pad_blocking (GstPad * pad, GstPadProbeInfo * info, gpointer user_data)
+{
+ GstRTSPStreamPrivate *priv;
+ GstRTSPStream *stream;
+
+ stream = user_data;
+ priv = stream->priv;
+
+ GST_DEBUG_OBJECT (pad, "now blocking");
+
+ g_mutex_lock (&priv->lock);
+ priv->blocking = TRUE;
+ g_mutex_unlock (&priv->lock);
+
+ gst_element_post_message (priv->payloader,
+ gst_message_new_element (GST_OBJECT_CAST (priv->payloader),
+ gst_structure_new_empty ("GstRTSPStreamBlocking")));
+
+ return GST_PAD_PROBE_OK;
+}
+
+/**
+ * gst_rtsp_stream_set_blocked:
+ * @stream: a #GstRTSPStream
+ * @blocked: boolean indicating we should block or unblock
+ *
+ * Blocks or unblocks the dataflow on @stream.
+ *
+ * Returns: %TRUE on success
+ */
+gboolean
+gst_rtsp_stream_set_blocked (GstRTSPStream * stream, gboolean blocked)
+{
+ GstRTSPStreamPrivate *priv;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
+
+ priv = stream->priv;
+
+ g_mutex_lock (&priv->lock);
+ if (blocked) {
+ priv->blocking = FALSE;
+ if (priv->blocked_id == 0) {
+ priv->blocked_id = gst_pad_add_probe (priv->srcpad,
+ GST_PAD_PROBE_TYPE_BLOCK | GST_PAD_PROBE_TYPE_BUFFER |
+ GST_PAD_PROBE_TYPE_BUFFER_LIST, pad_blocking,
+ g_object_ref (stream), g_object_unref);
+ }
+ } else {
+ if (priv->blocked_id != 0) {
+ gst_pad_remove_probe (priv->srcpad, priv->blocked_id);
+ priv->blocked_id = 0;
+ priv->blocking = FALSE;
+ }
+ }
+ g_mutex_unlock (&priv->lock);
+
+ return TRUE;
+}
+
+/**
+ * gst_rtsp_stream_is_blocking:
+ * @stream: a #GstRTSPStream
+ *
+ * Check if @stream is blocking on a #GstBuffer.
+ *
+ * Returns: %TRUE if @stream is blocking
+ */
+gboolean
+gst_rtsp_stream_is_blocking (GstRTSPStream * stream)
+{
+ GstRTSPStreamPrivate *priv;
+ gboolean result;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
+
+ priv = stream->priv;
+
+ g_mutex_lock (&priv->lock);
+ result = priv->blocking;
+ g_mutex_unlock (&priv->lock);
+
+ return result;
+}
projects / platform / upstream / gstreamer.git / blobdiff