* 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 <string.h>
#include <stdlib.h>
+#include <stdio.h>
+#include <string.h>
#include <gio/gio.h>
GstElement *payloader;
guint buffer_size;
gboolean is_joined;
+ gchar *control;
+
+ GstRTSPLowerTrans protocols;
/* pads on the rtpbin */
GstPad *send_rtp_sink;
/* multicast addresses */
GstRTSPAddressPool *pool;
- GstRTSPAddress *addr;
+ GstRTSPAddress *addr_v4;
+ GstRTSPAddress *addr_v6;
/* the caps of the stream */
gulong caps_sig;
gint dscp_qos;
};
+#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
};
static GQuark ssrc_stream_map_key;
+static void gst_rtsp_stream_get_property (GObject * object, guint propid,
+ GValue * value, GParamSpec * pspec);
+static void gst_rtsp_stream_set_property (GObject * object, guint propid,
+ const GValue * value, GParamSpec * pspec);
+
static void gst_rtsp_stream_finalize (GObject * obj);
G_DEFINE_TYPE (GstRTSPStream, gst_rtsp_stream, G_TYPE_OBJECT);
gobject_class = G_OBJECT_CLASS (klass);
+ gobject_class->get_property = gst_rtsp_stream_get_property;
+ gobject_class->set_property = gst_rtsp_stream_set_property;
gobject_class->finalize = gst_rtsp_stream_finalize;
+ g_object_class_install_property (gobject_class, PROP_CONTROL,
+ g_param_spec_string ("control", "Control",
+ "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");
stream->priv = priv;
- stream->priv->dscp_qos = -1;
+ priv->dscp_qos = -1;
+ priv->control = g_strdup (DEFAULT_CONTROL);
+ priv->protocols = DEFAULT_PROTOCOLS;
g_mutex_init (&priv->lock);
}
/* we really need to be unjoined now */
g_return_if_fail (!priv->is_joined);
- if (priv->addr)
- gst_rtsp_address_free (priv->addr);
+ if (priv->addr_v4)
+ gst_rtsp_address_free (priv->addr_v4);
+ if (priv->addr_v6)
+ gst_rtsp_address_free (priv->addr_v6);
if (priv->server_addr_v4)
gst_rtsp_address_free (priv->server_addr_v4);
if (priv->server_addr_v6)
g_object_unref (priv->pool);
gst_object_unref (priv->payloader);
gst_object_unref (priv->srcpad);
+ g_free (priv->control);
g_mutex_clear (&priv->lock);
G_OBJECT_CLASS (gst_rtsp_stream_parent_class)->finalize (obj);
}
+static void
+gst_rtsp_stream_get_property (GObject * object, guint propid,
+ GValue * value, GParamSpec * pspec)
+{
+ GstRTSPStream *stream = GST_RTSP_STREAM (object);
+
+ switch (propid) {
+ 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);
+ }
+}
+
+static void
+gst_rtsp_stream_set_property (GObject * object, guint propid,
+ const GValue * value, GParamSpec * pspec)
+{
+ GstRTSPStream *stream = GST_RTSP_STREAM (object);
+
+ switch (propid) {
+ 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);
+ }
+}
+
/**
* gst_rtsp_stream_new:
* @idx: an index
return stream->priv->idx;
}
- /**
+/**
* 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)
}
/**
+ * gst_rtsp_stream_get_control:
+ * @stream: a #GstRTSPStream
+ *
+ * Get the control string to identify this stream.
+ *
+ * Returns: (transfer full): the control string. free after usage.
+ */
+gchar *
+gst_rtsp_stream_get_control (GstRTSPStream * stream)
+{
+ GstRTSPStreamPrivate *priv;
+ gchar *result;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
+
+ priv = stream->priv;
+
+ g_mutex_lock (&priv->lock);
+ if ((result = g_strdup (priv->control)) == NULL)
+ result = g_strdup_printf ("stream=%u", priv->idx);
+ g_mutex_unlock (&priv->lock);
+
+ return result;
+}
+
+/**
+ * gst_rtsp_stream_set_control:
+ * @stream: a #GstRTSPStream
+ * @control: a control string
+ *
+ * Set the control string in @stream.
+ */
+void
+gst_rtsp_stream_set_control (GstRTSPStream * stream, const gchar * control)
+{
+ GstRTSPStreamPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_STREAM (stream));
+
+ priv = stream->priv;
+
+ g_mutex_lock (&priv->lock);
+ g_free (priv->control);
+ priv->control = g_strdup (control);
+ g_mutex_unlock (&priv->lock);
+}
+
+/**
+ * gst_rtsp_stream_has_control:
+ * @stream: a #GstRTSPStream
+ * @control: a control string
+ *
+ * Check if @stream has the control string @control.
+ *
+ * Returns: %TRUE is @stream has @control as the control string
+ */
+gboolean
+gst_rtsp_stream_has_control (GstRTSPStream * stream, const gchar * control)
+{
+ GstRTSPStreamPrivate *priv;
+ gboolean res;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
+
+ priv = stream->priv;
+
+ g_mutex_lock (&priv->lock);
+ if (priv->control)
+ res = (g_strcmp0 (priv->control, control) == 0);
+ else {
+ guint streamid;
+ sscanf (control, "stream=%u", &streamid);
+ res = (streamid == priv->idx);
+ }
+ g_mutex_unlock (&priv->lock);
+
+ return res;
+}
+
+/**
* gst_rtsp_stream_set_mtu:
* @stream: a #GstRTSPStream
* @mtu: a new MTU
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:
}
/**
- * gst_rtsp_stream_get_address:
+ * gst_rtsp_stream_get_multicast_address:
* @stream: a #GstRTSPStream
+ * @family: the #GSocketFamily
*
- * Get the multicast address of @stream.
+ * Get the multicast address of @stream for @family.
*
* Returns: the #GstRTSPAddress of @stream or %NULL when no address could be
* allocated. gst_rtsp_address_free() after usage.
*/
GstRTSPAddress *
-gst_rtsp_stream_get_address (GstRTSPStream * stream)
+gst_rtsp_stream_get_multicast_address (GstRTSPStream * stream,
+ GSocketFamily family)
{
GstRTSPStreamPrivate *priv;
GstRTSPAddress *result;
+ GstRTSPAddress **addrp;
+ GstRTSPAddressFlags flags;
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
priv = stream->priv;
+ if (family == G_SOCKET_FAMILY_IPV6) {
+ flags = GST_RTSP_ADDRESS_FLAG_IPV6;
+ addrp = &priv->addr_v4;
+ } else {
+ flags = GST_RTSP_ADDRESS_FLAG_IPV4;
+ addrp = &priv->addr_v6;
+ }
+
g_mutex_lock (&priv->lock);
- if (priv->addr == NULL) {
+ if (*addrp == NULL) {
if (priv->pool == NULL)
goto no_pool;
- priv->addr = gst_rtsp_address_pool_acquire_address (priv->pool,
- GST_RTSP_ADDRESS_FLAG_EVEN_PORT | GST_RTSP_ADDRESS_FLAG_MULTICAST, 2);
- if (priv->addr == NULL)
+ flags |= GST_RTSP_ADDRESS_FLAG_EVEN_PORT | GST_RTSP_ADDRESS_FLAG_MULTICAST;
+
+ *addrp = gst_rtsp_address_pool_acquire_address (priv->pool, flags, 2);
+ if (*addrp == NULL)
goto no_address;
}
- result = gst_rtsp_address_copy (priv->addr);
+ result = gst_rtsp_address_copy (*addrp);
g_mutex_unlock (&priv->lock);
return result;
{
GstRTSPStreamPrivate *priv;
GstRTSPAddress *result;
+ GInetAddress *addr;
+ GSocketFamily family;
+ GstRTSPAddress **addrp;
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
g_return_val_if_fail (address != NULL, NULL);
priv = stream->priv;
+ addr = g_inet_address_new_from_string (address);
+ if (!addr) {
+ GST_ERROR ("failed to get inet addr from %s", address);
+ family = G_SOCKET_FAMILY_IPV4;
+ } else {
+ family = g_inet_address_get_family (addr);
+ g_object_unref (addr);
+ }
+
+ if (family == G_SOCKET_FAMILY_IPV6)
+ addrp = &priv->addr_v4;
+ else
+ addrp = &priv->addr_v6;
+
g_mutex_lock (&priv->lock);
- if (priv->addr == NULL) {
+ if (*addrp == NULL) {
+ GstRTSPAddressPoolResult res;
+
if (priv->pool == NULL)
goto no_pool;
- priv->addr = gst_rtsp_address_pool_reserve_address (priv->pool, address,
- port, n_ports, ttl);
- if (priv->addr == 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 (priv->addr->address, address) ||
- priv->addr->port != port || priv->addr->n_ports != n_ports ||
- priv->addr->ttl != ttl)
+ if (strcmp ((*addrp)->address, address) ||
+ (*addrp)->port != port || (*addrp)->n_ports != n_ports ||
+ (*addrp)->ttl != ttl)
goto different_address;
}
- result = gst_rtsp_address_copy (priv->addr);
+ result = gst_rtsp_address_copy (*addrp);
g_mutex_unlock (&priv->lock);
return result;
* gst_rtsp_stream_get_server_port:
* @stream: a #GstRTSPStream
* @server_port: (out): result server port
+ * @family: the port family to get
*
* Fill @server_port with the port pair used by the server. This function can
* only be called when @stream has been joined.
*
* 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)
GstElement * rtpbin, GstState state)
{
GstRTSPStreamPrivate *priv;
- gint i, idx;
+ 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);
/* create a session with the same index as the stream */
idx = priv->idx;
- GST_INFO ("stream %p joining bin as session %d", stream, idx);
+ GST_INFO ("stream %p joining bin as session %u", stream, idx);
if (!alloc_ports (stream))
goto no_ports;
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);
}
}
no_ports:
{
g_mutex_unlock (&priv->lock);
- GST_WARNING ("failed to allocate ports %d", idx);
+ GST_WARNING ("failed to allocate ports %u", idx);
return FALSE;
}
link_failed:
{
- GST_WARNING ("failed to link stream %d", idx);
+ GST_WARNING ("failed to link stream %u", idx);
gst_object_unref (priv->send_rtp_sink);
priv->send_rtp_sink = NULL;
g_mutex_unlock (&priv->lock);
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