* 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>
#include "rtsp-stream.h"
+#define GST_RTSP_STREAM_GET_PRIVATE(obj) \
+ (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_RTSP_STREAM, GstRTSPStreamPrivate))
+
+struct _GstRTSPStreamPrivate
+{
+ GMutex lock;
+ guint idx;
+ GstPad *srcpad;
+ GstElement *payloader;
+ guint buffer_size;
+ gboolean is_joined;
+ gchar *control;
+
+ GstRTSPProfile profiles;
+ GstRTSPLowerTrans protocols;
+
+ /* pads on the rtpbin */
+ GstPad *send_rtp_sink;
+ GstPad *recv_sink[2];
+ GstPad *send_src[2];
+
+ /* the RTPSession object */
+ GObject *session;
+
+ /* sinks used for sending and receiving RTP and RTCP over ipv4, they share
+ * sockets */
+ GstElement *udpsrc_v4[2];
+
+ /* sinks used for sending and receiving RTP and RTCP over ipv6, they share
+ * sockets */
+ GstElement *udpsrc_v6[2];
+
+ GstElement *udpsink[2];
+
+ /* for TCP transport */
+ GstElement *appsrc[2];
+ GstElement *appqueue[2];
+ GstElement *appsink[2];
+
+ GstElement *tee[2];
+ GstElement *funnel[2];
+
+ /* server ports for sending/receiving over ipv4 */
+ GstRTSPRange server_port_v4;
+ GstRTSPAddress *server_addr_v4;
+ gboolean have_ipv4;
+
+ /* server ports for sending/receiving over ipv6 */
+ GstRTSPRange server_port_v6;
+ GstRTSPAddress *server_addr_v6;
+ gboolean have_ipv6;
+
+ /* multicast addresses */
+ GstRTSPAddressPool *pool;
+ GstRTSPAddress *addr_v4;
+ GstRTSPAddress *addr_v6;
+
+ /* the caps of the stream */
+ gulong caps_sig;
+ GstCaps *caps;
+
+ /* transports we stream to */
+ guint n_active;
+ GList *transports;
+
+ gint dscp_qos;
+
+ /* stream blocking */
+ gulong blocked_id;
+ gboolean blocking;
+};
+
+#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
};
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);
{
GObjectClass *gobject_class;
+ g_type_class_add_private (klass, sizeof (GstRTSPStreamPrivate));
+
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_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");
static void
gst_rtsp_stream_init (GstRTSPStream * stream)
{
- g_mutex_init (&stream->lock);
+ GstRTSPStreamPrivate *priv = GST_RTSP_STREAM_GET_PRIVATE (stream);
+
+ GST_DEBUG ("new stream %p", stream);
+
+ stream->priv = priv;
+
+ priv->dscp_qos = -1;
+ priv->control = g_strdup (DEFAULT_CONTROL);
+ priv->profiles = DEFAULT_PROFILES;
+ priv->protocols = DEFAULT_PROTOCOLS;
+
+ g_mutex_init (&priv->lock);
}
static void
gst_rtsp_stream_finalize (GObject * obj)
{
GstRTSPStream *stream;
+ GstRTSPStreamPrivate *priv;
stream = GST_RTSP_STREAM (obj);
+ priv = stream->priv;
- /* we really need to be unjoined now */
- g_return_if_fail (!stream->is_joined);
+ GST_DEBUG ("finalize stream %p", stream);
- if (stream->addr)
- gst_rtsp_address_free (stream->addr);
- if (stream->pool)
- g_object_unref (stream->pool);
- gst_object_unref (stream->payloader);
- gst_object_unref (stream->srcpad);
- g_mutex_clear (&stream->lock);
+ /* we really need to be unjoined now */
+ g_return_if_fail (!priv->is_joined);
+
+ 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)
+ gst_rtsp_address_free (priv->server_addr_v6);
+ if (priv->pool)
+ 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_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);
+ }
+}
+
+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_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_new:
* @idx: an index
GstRTSPStream *
gst_rtsp_stream_new (guint idx, GstElement * payloader, GstPad * srcpad)
{
+ GstRTSPStreamPrivate *priv;
GstRTSPStream *stream;
g_return_val_if_fail (GST_IS_ELEMENT (payloader), NULL);
g_return_val_if_fail (GST_PAD_IS_SRC (srcpad), NULL);
stream = g_object_new (GST_TYPE_RTSP_STREAM, NULL);
- stream->idx = idx;
- stream->payloader = gst_object_ref (payloader);
- stream->srcpad = gst_object_ref (srcpad);
+ priv = stream->priv;
+ priv->idx = idx;
+ priv->payloader = gst_object_ref (payloader);
+ priv->srcpad = gst_object_ref (srcpad);
return stream;
}
/**
+ * gst_rtsp_stream_get_index:
+ * @stream: a #GstRTSPStream
+ *
+ * Get the stream index.
+ *
+ * Return: the stream index.
+ */
+guint
+gst_rtsp_stream_get_index (GstRTSPStream * stream)
+{
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), -1);
+
+ return stream->priv->idx;
+}
+
+/**
+ * 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.
+ *
+ * Returns: (transfer full): the srcpad. Unref after usage.
+ */
+GstPad *
+gst_rtsp_stream_get_srcpad (GstRTSPStream * stream)
+{
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
+
+ return gst_object_ref (stream->priv->srcpad);
+}
+
+/**
+ * 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;
+
+ if (sscanf (control, "stream=%u", &streamid) > 0)
+ res = (streamid == priv->idx);
+ else
+ res = FALSE;
+ }
+ g_mutex_unlock (&priv->lock);
+
+ return res;
+}
+
+/**
* gst_rtsp_stream_set_mtu:
* @stream: a #GstRTSPStream
* @mtu: a new MTU
void
gst_rtsp_stream_set_mtu (GstRTSPStream * stream, guint mtu)
{
+ GstRTSPStreamPrivate *priv;
+
g_return_if_fail (GST_IS_RTSP_STREAM (stream));
+ priv = stream->priv;
+
GST_LOG_OBJECT (stream, "set MTU %u", mtu);
- g_object_set (G_OBJECT (stream->payloader), "mtu", mtu, NULL);
+ g_object_set (G_OBJECT (priv->payloader), "mtu", mtu, NULL);
}
/**
guint
gst_rtsp_stream_get_mtu (GstRTSPStream * stream)
{
+ GstRTSPStreamPrivate *priv;
guint mtu;
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), 0);
- g_object_get (G_OBJECT (stream->payloader), "mtu", &mtu, NULL);
+ priv = stream->priv;
+
+ g_object_get (G_OBJECT (priv->payloader), "mtu", &mtu, NULL);
return mtu;
}
+/* Update the dscp qos property on the udp sinks */
+static void
+update_dscp_qos (GstRTSPStream * stream)
+{
+ GstRTSPStreamPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_STREAM (stream));
+
+ priv = stream->priv;
+
+ if (priv->udpsink[0]) {
+ g_object_set (G_OBJECT (priv->udpsink[0]), "qos-dscp", priv->dscp_qos,
+ NULL);
+ }
+
+ if (priv->udpsink[1]) {
+ g_object_set (G_OBJECT (priv->udpsink[1]), "qos-dscp", priv->dscp_qos,
+ NULL);
+ }
+}
+
+/**
+ * gst_rtsp_stream_set_dscp_qos:
+ * @stream: a #GstRTSPStream
+ * @dscp_qos: a new dscp qos value (0-63, or -1 to disable)
+ *
+ * Configure the dscp qos of the outgoing sockets to @dscp_qos.
+ */
+void
+gst_rtsp_stream_set_dscp_qos (GstRTSPStream * stream, gint dscp_qos)
+{
+ GstRTSPStreamPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_STREAM (stream));
+
+ priv = stream->priv;
+
+ GST_LOG_OBJECT (stream, "set DSCP QoS %d", dscp_qos);
+
+ if (dscp_qos < -1 || dscp_qos > 63) {
+ GST_WARNING_OBJECT (stream, "trying to set illegal dscp qos %d", dscp_qos);
+ return;
+ }
+
+ priv->dscp_qos = dscp_qos;
+
+ update_dscp_qos (stream);
+}
+
+/**
+ * gst_rtsp_stream_get_dscp_qos:
+ * @stream: a #GstRTSPStream
+ *
+ * Get the configured DSCP QoS in of the outgoing sockets.
+ *
+ * Returns: the DSCP QoS value of the outgoing sockets, or -1 if disbled.
+ */
+gint
+gst_rtsp_stream_get_dscp_qos (GstRTSPStream * stream)
+{
+ GstRTSPStreamPrivate *priv;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), -1);
+
+ priv = stream->priv;
+
+ 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:
* @stream: a #GstRTSPStream
gst_rtsp_stream_set_address_pool (GstRTSPStream * stream,
GstRTSPAddressPool * pool)
{
+ GstRTSPStreamPrivate *priv;
GstRTSPAddressPool *old;
g_return_if_fail (GST_IS_RTSP_STREAM (stream));
+ priv = stream->priv;
+
GST_LOG_OBJECT (stream, "set address pool %p", pool);
- g_mutex_lock (&stream->lock);
- if ((old = stream->pool) != pool)
- stream->pool = pool ? g_object_ref (pool) : NULL;
+ g_mutex_lock (&priv->lock);
+ if ((old = priv->pool) != pool)
+ priv->pool = pool ? g_object_ref (pool) : NULL;
else
old = NULL;
- g_mutex_unlock (&stream->lock);
+ g_mutex_unlock (&priv->lock);
if (old)
g_object_unref (old);
GstRTSPAddressPool *
gst_rtsp_stream_get_address_pool (GstRTSPStream * stream)
{
+ GstRTSPStreamPrivate *priv;
GstRTSPAddressPool *result;
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
- g_mutex_lock (&stream->lock);
- if ((result = stream->pool))
+ priv = stream->priv;
+
+ g_mutex_lock (&priv->lock);
+ if ((result = priv->pool))
g_object_ref (result);
- g_mutex_unlock (&stream->lock);
+ g_mutex_unlock (&priv->lock);
return result;
}
/**
- * 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;
- g_mutex_lock (&stream->lock);
- if (stream->addr == NULL) {
- if (stream->pool == NULL)
+ 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 (*addrp == NULL) {
+ if (priv->pool == NULL)
goto no_pool;
- stream->addr = gst_rtsp_address_pool_acquire_address (stream->pool,
- GST_RTSP_ADDRESS_FLAG_EVEN_PORT, 2);
- if (stream->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 (stream->addr);
- g_mutex_unlock (&stream->lock);
+ result = gst_rtsp_address_copy (*addrp);
+ g_mutex_unlock (&priv->lock);
return result;
no_pool:
{
GST_ERROR_OBJECT (stream, "no address pool specified");
- g_mutex_unlock (&stream->lock);
+ g_mutex_unlock (&priv->lock);
return NULL;
}
no_address:
{
GST_ERROR_OBJECT (stream, "failed to acquire address from pool");
- g_mutex_unlock (&stream->lock);
+ g_mutex_unlock (&priv->lock);
+ return NULL;
+ }
+}
+
+/**
+ * gst_rtsp_stream_reserve_address:
+ * @stream: a #GstRTSPStream
+ * @address: an address
+ * @port: a port
+ * @n_ports: n_ports
+ * @ttl: a TTL
+ *
+ * Reserve @address and @port as the address and port of @stream.
+ *
+ * Returns: the #GstRTSPAddress of @stream or %NULL when the address could be
+ * reserved. gst_rtsp_address_free() after usage.
+ */
+GstRTSPAddress *
+gst_rtsp_stream_reserve_address (GstRTSPStream * stream,
+ const gchar * address, guint port, guint n_ports, guint ttl)
+{
+ 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);
+ g_return_val_if_fail (port > 0, NULL);
+ g_return_val_if_fail (n_ports > 0, NULL);
+ g_return_val_if_fail (ttl > 0, 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 (*addrp == NULL) {
+ GstRTSPAddressPoolResult res;
+
+ if (priv->pool == NULL)
+ goto no_pool;
+
+ 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) ||
+ (*addrp)->port != port || (*addrp)->n_ports != n_ports ||
+ (*addrp)->ttl != ttl)
+ goto different_address;
+ }
+ result = gst_rtsp_address_copy (*addrp);
+ g_mutex_unlock (&priv->lock);
+
+ return result;
+
+ /* ERRORS */
+no_pool:
+ {
+ GST_ERROR_OBJECT (stream, "no address pool specified");
+ g_mutex_unlock (&priv->lock);
+ return NULL;
+ }
+no_address:
+ {
+ GST_ERROR_OBJECT (stream, "failed to acquire address %s from pool",
+ address);
+ g_mutex_unlock (&priv->lock);
+ return NULL;
+ }
+different_address:
+ {
+ GST_ERROR_OBJECT (stream, "address %s is not the same that was already"
+ " reserved", address);
+ g_mutex_unlock (&priv->lock);
return NULL;
}
}
-/* must be called with lock */
static gboolean
-alloc_ports (GstRTSPStream * stream)
+alloc_ports_one_family (GstRTSPAddressPool * pool, gint buffer_size,
+ GSocketFamily family, GstElement * udpsrc_out[2],
+ GstElement * udpsink_out[2], GstRTSPRange * server_port_out,
+ GstRTSPAddress ** server_addr_out)
{
GstStateChangeReturn ret;
GstElement *udpsrc0, *udpsrc1;
GstElement *udpsink0, *udpsink1;
+ GSocket *rtp_socket = NULL;
+ GSocket *rtcp_socket;
gint tmp_rtp, tmp_rtcp;
guint count;
gint rtpport, rtcpport;
- GSocket *socket;
- const gchar *host;
-
- g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
+ GList *rejected_addresses = NULL;
+ GstRTSPAddress *addr = NULL;
+ GInetAddress *inetaddr = NULL;
+ GSocketAddress *rtp_sockaddr = NULL;
+ GSocketAddress *rtcp_sockaddr = NULL;
+ const gchar *multisink_socket;
+
+ if (family == G_SOCKET_FAMILY_IPV6)
+ multisink_socket = "socket-v6";
+ else
+ multisink_socket = "socket";
udpsrc0 = NULL;
udpsrc1 = NULL;
/* Start with random port */
tmp_rtp = 0;
- if (stream->is_ipv6)
- host = "udp://[::0]";
- else
- host = "udp://0.0.0.0";
+ rtcp_socket = g_socket_new (family, G_SOCKET_TYPE_DATAGRAM,
+ G_SOCKET_PROTOCOL_UDP, NULL);
+ if (!rtcp_socket)
+ goto no_udp_protocol;
+
+ if (*server_addr_out)
+ gst_rtsp_address_free (*server_addr_out);
/* 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:
- udpsrc0 = gst_element_make_from_uri (GST_URI_SRC, host, NULL, NULL);
- if (udpsrc0 == NULL)
- goto no_udp_protocol;
- g_object_set (G_OBJECT (udpsrc0), "port", tmp_rtp, NULL);
- ret = gst_element_set_state (udpsrc0, GST_STATE_PAUSED);
- if (ret == GST_STATE_CHANGE_FAILURE) {
+ if (rtp_socket == NULL) {
+ rtp_socket = g_socket_new (family, G_SOCKET_TYPE_DATAGRAM,
+ G_SOCKET_PROTOCOL_UDP, NULL);
+ if (!rtp_socket)
+ goto no_udp_protocol;
+ }
+
+ if (pool && gst_rtsp_address_pool_has_unicast_addresses (pool)) {
+ GstRTSPAddressFlags flags;
+
+ if (addr)
+ rejected_addresses = g_list_prepend (rejected_addresses, addr);
+
+ flags = GST_RTSP_ADDRESS_FLAG_EVEN_PORT | GST_RTSP_ADDRESS_FLAG_UNICAST;
+ if (family == G_SOCKET_FAMILY_IPV6)
+ flags |= GST_RTSP_ADDRESS_FLAG_IPV6;
+ else
+ flags |= GST_RTSP_ADDRESS_FLAG_IPV4;
+
+ addr = gst_rtsp_address_pool_acquire_address (pool, flags, 2);
+
+ if (addr == NULL)
+ goto no_ports;
+
+ tmp_rtp = addr->port;
+
+ g_clear_object (&inetaddr);
+ inetaddr = g_inet_address_new_from_string (addr->address);
+ } else {
if (tmp_rtp != 0) {
tmp_rtp += 2;
if (++count > 20)
goto no_ports;
+ }
- gst_element_set_state (udpsrc0, GST_STATE_NULL);
- gst_object_unref (udpsrc0);
+ if (inetaddr == NULL)
+ inetaddr = g_inet_address_new_any (family);
+ }
- goto again;
- }
- goto no_udp_protocol;
+ rtp_sockaddr = g_inet_socket_address_new (inetaddr, tmp_rtp);
+ if (!g_socket_bind (rtp_socket, rtp_sockaddr, FALSE, NULL)) {
+ g_object_unref (rtp_sockaddr);
+ goto again;
}
+ g_object_unref (rtp_sockaddr);
- g_object_get (G_OBJECT (udpsrc0), "port", &tmp_rtp, NULL);
+ rtp_sockaddr = g_socket_get_local_address (rtp_socket, NULL);
+ if (rtp_sockaddr == NULL || !G_IS_INET_SOCKET_ADDRESS (rtp_sockaddr)) {
+ g_clear_object (&rtp_sockaddr);
+ goto socket_error;
+ }
+
+ tmp_rtp =
+ g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (rtp_sockaddr));
+ g_object_unref (rtp_sockaddr);
/* check if port is even */
if ((tmp_rtp & 1) != 0) {
/* port not even, close and allocate another */
- if (++count > 20)
- goto no_ports;
-
- gst_element_set_state (udpsrc0, GST_STATE_NULL);
- gst_object_unref (udpsrc0);
-
tmp_rtp++;
+ g_clear_object (&rtp_socket);
goto again;
}
- /* allocate port+1 for RTCP now */
- udpsrc1 = gst_element_make_from_uri (GST_URI_SRC, host, NULL, NULL);
- if (udpsrc1 == NULL)
- goto no_udp_rtcp_protocol;
+ /* set port */
+ tmp_rtcp = tmp_rtp + 1;
+
+ rtcp_sockaddr = g_inet_socket_address_new (inetaddr, tmp_rtcp);
+ if (!g_socket_bind (rtcp_socket, rtcp_sockaddr, FALSE, NULL)) {
+ g_object_unref (rtcp_sockaddr);
+ g_clear_object (&rtp_socket);
+ goto again;
+ }
+ g_object_unref (rtcp_sockaddr);
- /* set port */
- tmp_rtcp = tmp_rtp + 1;
- g_object_set (G_OBJECT (udpsrc1), "port", tmp_rtcp, NULL);
+ g_clear_object (&inetaddr);
- ret = gst_element_set_state (udpsrc1, GST_STATE_PAUSED);
- /* tmp_rtcp port is busy already : retry to make rtp/rtcp pair */
- if (ret == GST_STATE_CHANGE_FAILURE) {
+ udpsrc0 = gst_element_factory_make ("udpsrc", NULL);
+ udpsrc1 = gst_element_factory_make ("udpsrc", NULL);
- if (++count > 20)
- goto no_ports;
+ if (udpsrc0 == NULL || udpsrc1 == NULL)
+ goto no_udp_protocol;
- gst_element_set_state (udpsrc0, GST_STATE_NULL);
- gst_object_unref (udpsrc0);
+ g_object_set (G_OBJECT (udpsrc0), "socket", rtp_socket, NULL);
+ g_object_set (G_OBJECT (udpsrc1), "socket", rtcp_socket, NULL);
- gst_element_set_state (udpsrc1, GST_STATE_NULL);
- gst_object_unref (udpsrc1);
+ ret = gst_element_set_state (udpsrc0, GST_STATE_PAUSED);
+ if (ret == GST_STATE_CHANGE_FAILURE)
+ goto element_error;
+ ret = gst_element_set_state (udpsrc1, GST_STATE_PAUSED);
+ if (ret == GST_STATE_CHANGE_FAILURE)
+ goto element_error;
- tmp_rtp += 2;
- goto again;
- }
/* all fine, do port check */
g_object_get (G_OBJECT (udpsrc0), "port", &rtpport, NULL);
g_object_get (G_OBJECT (udpsrc1), "port", &rtcpport, NULL);
if (rtpport != tmp_rtp || rtcpport != tmp_rtcp)
goto port_error;
- udpsink0 = gst_element_factory_make ("multiudpsink", NULL);
+ if (udpsink_out[0])
+ udpsink0 = udpsink_out[0];
+ else
+ udpsink0 = gst_element_factory_make ("multiudpsink", NULL);
+
if (!udpsink0)
goto no_udp_protocol;
- g_object_get (G_OBJECT (udpsrc0), "used-socket", &socket, NULL);
- g_object_set (G_OBJECT (udpsink0), "socket", socket, NULL);
g_object_set (G_OBJECT (udpsink0), "close-socket", FALSE, NULL);
+ g_object_set (G_OBJECT (udpsink0), multisink_socket, rtp_socket, NULL);
+
+ if (udpsink_out[1])
+ udpsink1 = udpsink_out[1];
+ else
+ udpsink1 = gst_element_factory_make ("multiudpsink", NULL);
- udpsink1 = gst_element_factory_make ("multiudpsink", NULL);
if (!udpsink1)
goto no_udp_protocol;
- if (g_object_class_find_property (G_OBJECT_GET_CLASS (udpsink0),
- "send-duplicates")) {
- g_object_set (G_OBJECT (udpsink0), "send-duplicates", FALSE, NULL);
- g_object_set (G_OBJECT (udpsink1), "send-duplicates", FALSE, NULL);
- } else {
- g_warning
- ("old multiudpsink version found without send-duplicates property");
- }
-
- if (g_object_class_find_property (G_OBJECT_GET_CLASS (udpsink0),
- "buffer-size")) {
- g_object_set (G_OBJECT (udpsink0), "buffer-size", stream->buffer_size,
- NULL);
- } else {
- GST_WARNING ("multiudpsink version found without buffer-size property");
- }
+ g_object_set (G_OBJECT (udpsink0), "send-duplicates", FALSE, NULL);
+ g_object_set (G_OBJECT (udpsink1), "send-duplicates", FALSE, NULL);
+ g_object_set (G_OBJECT (udpsink0), "buffer-size", buffer_size, NULL);
- g_object_get (G_OBJECT (udpsrc1), "used-socket", &socket, NULL);
- g_object_set (G_OBJECT (udpsink1), "socket", socket, NULL);
g_object_set (G_OBJECT (udpsink1), "close-socket", FALSE, NULL);
+ g_object_set (G_OBJECT (udpsink1), multisink_socket, rtcp_socket, NULL);
g_object_set (G_OBJECT (udpsink1), "sync", FALSE, NULL);
g_object_set (G_OBJECT (udpsink1), "async", FALSE, NULL);
g_object_set (G_OBJECT (udpsink0), "auto-multicast", FALSE, NULL);
/* we keep these elements, we will further configure them when the
* client told us to really use the UDP ports. */
- stream->udpsrc[0] = udpsrc0;
- stream->udpsrc[1] = udpsrc1;
- stream->udpsink[0] = udpsink0;
- stream->udpsink[1] = udpsink1;
- stream->server_port.min = rtpport;
- stream->server_port.max = rtcpport;
+ udpsrc_out[0] = udpsrc0;
+ udpsrc_out[1] = udpsrc1;
+ udpsink_out[0] = udpsink0;
+ udpsink_out[1] = udpsink1;
+ server_port_out->min = rtpport;
+ server_port_out->max = rtcpport;
+
+ *server_addr_out = addr;
+ g_list_free_full (rejected_addresses, (GDestroyNotify) gst_rtsp_address_free);
+
+ g_object_unref (rtp_socket);
+ g_object_unref (rtcp_socket);
return TRUE;
{
goto cleanup;
}
-no_udp_rtcp_protocol:
+port_error:
{
goto cleanup;
}
-port_error:
+socket_error:
+ {
+ goto cleanup;
+ }
+element_error:
{
goto cleanup;
}
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,
+ (GDestroyNotify) gst_rtsp_address_free);
+ if (addr)
+ gst_rtsp_address_free (addr);
+ if (rtp_socket)
+ g_object_unref (rtp_socket);
+ if (rtcp_socket)
+ g_object_unref (rtcp_socket);
return FALSE;
}
}
+/* must be called with lock */
+static gboolean
+alloc_ports (GstRTSPStream * stream)
+{
+ GstRTSPStreamPrivate *priv = stream->priv;
+
+ priv->have_ipv4 = alloc_ports_one_family (priv->pool, priv->buffer_size,
+ G_SOCKET_FAMILY_IPV4, priv->udpsrc_v4, priv->udpsink,
+ &priv->server_port_v4, &priv->server_addr_v4);
+
+ priv->have_ipv6 = alloc_ports_one_family (priv->pool, priv->buffer_size,
+ G_SOCKET_FAMILY_IPV6, priv->udpsrc_v6, priv->udpsink,
+ &priv->server_port_v6, &priv->server_addr_v6);
+
+ return priv->have_ipv4 || priv->have_ipv6;
+}
+
+/**
+ * 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.
+ */
+void
+gst_rtsp_stream_get_server_port (GstRTSPStream * stream,
+ GstRTSPRange * server_port, GSocketFamily family)
+{
+ GstRTSPStreamPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_STREAM (stream));
+ priv = stream->priv;
+ g_return_if_fail (priv->is_joined);
+
+ g_mutex_lock (&priv->lock);
+ if (family == G_SOCKET_FAMILY_IPV4) {
+ if (server_port)
+ *server_port = priv->server_port_v4;
+ } else {
+ if (server_port)
+ *server_port = priv->server_port_v6;
+ }
+ g_mutex_unlock (&priv->lock);
+}
+
+/**
+ * gst_rtsp_stream_get_rtpsession:
+ * @stream: a #GstRTSPStream
+ *
+ * Get the RTP session of this stream.
+ *
+ * Returns: (transfer full): The RTP session of this stream. Unref after usage.
+ */
+GObject *
+gst_rtsp_stream_get_rtpsession (GstRTSPStream * stream)
+{
+ GstRTSPStreamPrivate *priv;
+ GObject *session;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
+
+ priv = stream->priv;
+
+ g_mutex_lock (&priv->lock);
+ if ((session = priv->session))
+ g_object_ref (session);
+ g_mutex_unlock (&priv->lock);
+
+ return session;
+}
+
+/**
+ * gst_rtsp_stream_get_ssrc:
+ * @stream: a #GstRTSPStream
+ * @ssrc: (out): result ssrc
+ *
+ * Get the SSRC used by the RTP session of this stream. This function can only
+ * be called when @stream has been joined.
+ */
+void
+gst_rtsp_stream_get_ssrc (GstRTSPStream * stream, guint * ssrc)
+{
+ GstRTSPStreamPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_STREAM (stream));
+ priv = stream->priv;
+ g_return_if_fail (priv->is_joined);
+
+ g_mutex_lock (&priv->lock);
+ if (ssrc && priv->session)
+ g_object_get (priv->session, "internal-ssrc", ssrc, NULL);
+ g_mutex_unlock (&priv->lock);
+}
+
/* executed from streaming thread */
static void
caps_notify (GstPad * pad, GParamSpec * unused, GstRTSPStream * stream)
{
+ GstRTSPStreamPrivate *priv = stream->priv;
GstCaps *newcaps, *oldcaps;
newcaps = gst_pad_get_current_caps (pad);
GST_INFO ("stream %p received caps %p, %" GST_PTR_FORMAT, stream, newcaps,
newcaps);
- g_mutex_lock (&stream->lock);
- oldcaps = stream->caps;
- stream->caps = newcaps;
- g_mutex_unlock (&stream->lock);
+ g_mutex_lock (&priv->lock);
+ oldcaps = priv->caps;
+ priv->caps = newcaps;
+ g_mutex_unlock (&priv->lock);
if (oldcaps)
gst_caps_unref (oldcaps);
static GstRTSPStreamTransport *
find_transport (GstRTSPStream * stream, const gchar * rtcp_from)
{
+ GstRTSPStreamPrivate *priv = stream->priv;
GList *walk;
GstRTSPStreamTransport *result = NULL;
const gchar *tmp;
port = atoi (tmp + 1);
dest = g_strndup (rtcp_from, tmp - rtcp_from);
- g_mutex_lock (&stream->lock);
+ g_mutex_lock (&priv->lock);
GST_INFO ("finding %s:%d in %d transports", dest, port,
- g_list_length (stream->transports));
+ g_list_length (priv->transports));
- for (walk = stream->transports; walk; walk = g_list_next (walk)) {
+ for (walk = priv->transports; walk; walk = g_list_next (walk)) {
GstRTSPStreamTransport *trans = walk->data;
+ const GstRTSPTransport *tr;
gint min, max;
- min = trans->transport->client_port.min;
- max = trans->transport->client_port.max;
+ tr = gst_rtsp_stream_transport_get_transport (trans);
+
+ min = tr->client_port.min;
+ max = tr->client_port.max;
- if ((strcmp (trans->transport->destination, dest) == 0) && (min == port
- || max == port)) {
+ if ((strcmp (tr->destination, dest) == 0) && (min == port || max == port)) {
result = trans;
break;
}
}
- g_mutex_unlock (&stream->lock);
+ if (result)
+ g_object_ref (result);
+ g_mutex_unlock (&priv->lock);
g_free (dest);
if ((trans = find_transport (stream, rtcp_from))) {
GST_INFO ("%p: found transport %p for source %p", stream, trans,
source);
-
- /* keep ref to the source */
- trans->rtpsource = source;
-
- g_object_set_qdata (source, ssrc_stream_map_key, trans);
+ g_object_set_qdata_full (source, ssrc_stream_map_key, trans,
+ g_object_unref);
}
gst_structure_free (stats);
}
}
-
return trans;
}
GST_INFO ("%p: source %p bye timeout", stream, source);
if ((trans = g_object_get_qdata (source, ssrc_stream_map_key))) {
- trans->rtpsource = NULL;
- trans->timeout = TRUE;
+ gst_rtsp_stream_transport_set_timed_out (trans, TRUE);
+ g_object_set_qdata (source, ssrc_stream_map_key, NULL);
}
}
GST_INFO ("%p: source %p timeout", stream, source);
if ((trans = g_object_get_qdata (source, ssrc_stream_map_key))) {
- trans->rtpsource = NULL;
- trans->timeout = TRUE;
+ gst_rtsp_stream_transport_set_timed_out (trans, TRUE);
+ g_object_set_qdata (source, ssrc_stream_map_key, NULL);
}
}
static GstFlowReturn
handle_new_sample (GstAppSink * sink, gpointer user_data)
{
+ GstRTSPStreamPrivate *priv;
GList *walk;
GstSample *sample;
GstBuffer *buffer;
return GST_FLOW_OK;
stream = (GstRTSPStream *) user_data;
+ priv = stream->priv;
buffer = gst_sample_get_buffer (sample);
- g_mutex_lock (&stream->lock);
- for (walk = stream->transports; walk; walk = g_list_next (walk)) {
+ g_mutex_lock (&priv->lock);
+ for (walk = priv->transports; walk; walk = g_list_next (walk)) {
GstRTSPStreamTransport *tr = (GstRTSPStreamTransport *) walk->data;
- if (GST_ELEMENT_CAST (sink) == stream->appsink[0]) {
+ if (GST_ELEMENT_CAST (sink) == priv->appsink[0]) {
gst_rtsp_stream_transport_send_rtp (tr, buffer);
} else {
gst_rtsp_stream_transport_send_rtcp (tr, buffer);
}
}
- g_mutex_unlock (&stream->lock);
+ g_mutex_unlock (&priv->lock);
gst_sample_unref (sample);
* @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.
gst_rtsp_stream_join_bin (GstRTSPStream * stream, GstBin * bin,
GstElement * rtpbin, GstState state)
{
- gint i, idx;
+ GstRTSPStreamPrivate *priv;
+ 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);
g_return_val_if_fail (GST_IS_BIN (bin), FALSE);
g_return_val_if_fail (GST_IS_ELEMENT (rtpbin), FALSE);
- g_mutex_lock (&stream->lock);
- if (stream->is_joined)
+ priv = stream->priv;
+
+ g_mutex_lock (&priv->lock);
+ if (priv->is_joined)
goto was_joined;
/* create a session with the same index as the stream */
- idx = stream->idx;
+ 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;
+ /* update the dscp qos field in the sinks */
+ update_dscp_qos (stream);
+
/* get a pad for sending RTP */
name = g_strdup_printf ("send_rtp_sink_%u", idx);
- stream->send_rtp_sink = gst_element_get_request_pad (rtpbin, name);
+ priv->send_rtp_sink = gst_element_get_request_pad (rtpbin, name);
g_free (name);
/* link the RTP pad to the session manager, it should not really fail unless
* this is not really an RTP pad */
- ret = gst_pad_link (stream->srcpad, stream->send_rtp_sink);
+ ret = gst_pad_link (priv->srcpad, priv->send_rtp_sink);
if (ret != GST_PAD_LINK_OK)
goto link_failed;
/* get pads from the RTP session element for sending and receiving
* RTP/RTCP*/
name = g_strdup_printf ("send_rtp_src_%u", idx);
- stream->send_src[0] = gst_element_get_static_pad (rtpbin, name);
+ priv->send_src[0] = gst_element_get_static_pad (rtpbin, name);
g_free (name);
name = g_strdup_printf ("send_rtcp_src_%u", idx);
- stream->send_src[1] = gst_element_get_request_pad (rtpbin, name);
+ priv->send_src[1] = gst_element_get_request_pad (rtpbin, name);
g_free (name);
name = g_strdup_printf ("recv_rtp_sink_%u", idx);
- stream->recv_sink[0] = gst_element_get_request_pad (rtpbin, name);
+ priv->recv_sink[0] = gst_element_get_request_pad (rtpbin, name);
g_free (name);
name = g_strdup_printf ("recv_rtcp_sink_%u", idx);
- stream->recv_sink[1] = gst_element_get_request_pad (rtpbin, name);
+ priv->recv_sink[1] = gst_element_get_request_pad (rtpbin, name);
g_free (name);
/* get the session */
- g_signal_emit_by_name (rtpbin, "get-internal-session", idx, &stream->session);
+ g_signal_emit_by_name (rtpbin, "get-internal-session", idx, &priv->session);
- g_signal_connect (stream->session, "on-new-ssrc", (GCallback) on_new_ssrc,
+ g_signal_connect (priv->session, "on-new-ssrc", (GCallback) on_new_ssrc,
stream);
- g_signal_connect (stream->session, "on-ssrc-sdes", (GCallback) on_ssrc_sdes,
+ g_signal_connect (priv->session, "on-ssrc-sdes", (GCallback) on_ssrc_sdes,
stream);
- g_signal_connect (stream->session, "on-ssrc-active",
+ g_signal_connect (priv->session, "on-ssrc-active",
(GCallback) on_ssrc_active, stream);
- g_signal_connect (stream->session, "on-bye-ssrc", (GCallback) on_bye_ssrc,
+ g_signal_connect (priv->session, "on-bye-ssrc", (GCallback) on_bye_ssrc,
stream);
- g_signal_connect (stream->session, "on-bye-timeout",
+ g_signal_connect (priv->session, "on-bye-timeout",
(GCallback) on_bye_timeout, stream);
- g_signal_connect (stream->session, "on-timeout", (GCallback) on_timeout,
+ g_signal_connect (priv->session, "on-timeout", (GCallback) on_timeout,
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 */
- stream->tee[i] = gst_element_factory_make ("tee", NULL);
- gst_bin_add (bin, stream->tee[i]);
-
- /* and link to rtpbin send pad */
- pad = gst_element_get_static_pad (stream->tee[i], "sink");
- gst_pad_link (stream->send_src[i], pad);
- gst_object_unref (pad);
-
/* add udpsink */
- gst_bin_add (bin, stream->udpsink[i]);
-
- /* link tee to udpsink */
- teepad = gst_element_get_request_pad (stream->tee[i], "src_%u");
- pad = gst_element_get_static_pad (stream->udpsink[i], "sink");
- gst_pad_link (teepad, pad);
- gst_object_unref (pad);
- gst_object_unref (teepad);
-
- /* make queue */
- stream->appqueue[i] = gst_element_factory_make ("queue", NULL);
- gst_bin_add (bin, stream->appqueue[i]);
- /* and link to tee */
- teepad = gst_element_get_request_pad (stream->tee[i], "src_%u");
- pad = gst_element_get_static_pad (stream->appqueue[i], "sink");
- gst_pad_link (teepad, pad);
- gst_object_unref (pad);
- gst_object_unref (teepad);
-
- /* make appsink */
- stream->appsink[i] = gst_element_factory_make ("appsink", NULL);
- g_object_set (stream->appsink[i], "async", FALSE, "sync", FALSE, NULL);
- g_object_set (stream->appsink[i], "emit-signals", FALSE, NULL);
- gst_bin_add (bin, stream->appsink[i]);
- gst_app_sink_set_callbacks (GST_APP_SINK_CAST (stream->appsink[i]),
- &sink_cb, stream, NULL);
- /* and link to queue */
- queuepad = gst_element_get_static_pad (stream->appqueue[i], "src");
- pad = gst_element_get_static_pad (stream->appsink[i], "sink");
- gst_pad_link (queuepad, pad);
- gst_object_unref (pad);
- gst_object_unref (queuepad);
+ gst_bin_add (bin, priv->udpsink[i]);
+ sinkpad = gst_element_get_static_pad (priv->udpsink[i], "sink");
+
+ 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
* '--------' '--------'
*/
/* make funnel for the RTP/RTCP receivers */
- stream->funnel[i] = gst_element_factory_make ("funnel", NULL);
- gst_bin_add (bin, stream->funnel[i]);
+ priv->funnel[i] = gst_element_factory_make ("funnel", NULL);
+ gst_bin_add (bin, priv->funnel[i]);
- pad = gst_element_get_static_pad (stream->funnel[i], "src");
- gst_pad_link (pad, stream->recv_sink[i]);
+ pad = gst_element_get_static_pad (priv->funnel[i], "src");
+ gst_pad_link (pad, priv->recv_sink[i]);
gst_object_unref (pad);
- /* add udpsrc */
- gst_bin_add (bin, stream->udpsrc[i]);
- /* and link to the funnel */
- selpad = gst_element_get_request_pad (stream->funnel[i], "sink_%u");
- pad = gst_element_get_static_pad (stream->udpsrc[i], "src");
- gst_pad_link (pad, selpad);
- gst_object_unref (pad);
- gst_object_unref (selpad);
-
- /* make and add appsrc */
- stream->appsrc[i] = gst_element_factory_make ("appsrc", NULL);
- gst_bin_add (bin, stream->appsrc[i]);
- /* and link to the funnel */
- selpad = gst_element_get_request_pad (stream->funnel[i], "sink_%u");
- pad = gst_element_get_static_pad (stream->appsrc[i], "src");
- gst_pad_link (pad, selpad);
- gst_object_unref (pad);
- gst_object_unref (selpad);
+ if (priv->udpsrc_v4[i]) {
+ /* we set and keep these to playing so that they don't cause NO_PREROLL return
+ * values */
+ gst_element_set_state (priv->udpsrc_v4[i], GST_STATE_PLAYING);
+ gst_element_set_locked_state (priv->udpsrc_v4[i], TRUE);
+ /* add udpsrc */
+ gst_bin_add (bin, priv->udpsrc_v4[i]);
+
+ /* and link to the funnel v4 */
+ selpad = gst_element_get_request_pad (priv->funnel[i], "sink_%u");
+ pad = gst_element_get_static_pad (priv->udpsrc_v4[i], "src");
+ gst_pad_link (pad, selpad);
+ gst_object_unref (pad);
+ gst_object_unref (selpad);
+ }
+
+ if (priv->udpsrc_v6[i]) {
+ gst_element_set_state (priv->udpsrc_v6[i], GST_STATE_PLAYING);
+ gst_element_set_locked_state (priv->udpsrc_v6[i], TRUE);
+ gst_bin_add (bin, priv->udpsrc_v6[i]);
+
+ /* and link to the funnel v6 */
+ selpad = gst_element_get_request_pad (priv->funnel[i], "sink_%u");
+ pad = gst_element_get_static_pad (priv->udpsrc_v6[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 (stream->udpsink[i], state);
- gst_element_set_state (stream->appsink[i], state);
- gst_element_set_state (stream->appqueue[i], state);
- gst_element_set_state (stream->tee[i], state);
- gst_element_set_state (stream->funnel[i], state);
- gst_element_set_state (stream->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);
}
- /* we set and keep these to playing so that they don't cause NO_PREROLL return
- * values */
- gst_element_set_state (stream->udpsrc[i], GST_STATE_PLAYING);
- gst_element_set_locked_state (stream->udpsrc[i], TRUE);
}
/* be notified of caps changes */
- stream->caps_sig = g_signal_connect (stream->send_rtp_sink, "notify::caps",
+ priv->caps_sig = g_signal_connect (priv->send_rtp_sink, "notify::caps",
(GCallback) caps_notify, stream);
- stream->is_joined = TRUE;
- g_mutex_unlock (&stream->lock);
+ priv->is_joined = TRUE;
+ g_mutex_unlock (&priv->lock);
return TRUE;
/* ERRORS */
was_joined:
{
- g_mutex_unlock (&stream->lock);
+ g_mutex_unlock (&priv->lock);
return TRUE;
}
no_ports:
{
- g_mutex_unlock (&stream->lock);
- GST_WARNING ("failed to allocate ports %d", idx);
+ g_mutex_unlock (&priv->lock);
+ GST_WARNING ("failed to allocate ports %u", idx);
return FALSE;
}
link_failed:
{
- GST_WARNING ("failed to link stream %d", idx);
- gst_object_unref (stream->send_rtp_sink);
- stream->send_rtp_sink = NULL;
- g_mutex_unlock (&stream->lock);
+ 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);
return FALSE;
}
}
* @bin: a #GstBin
* @rtpbin: a rtpbin #GstElement
*
- * Remove the elements of @stream from @bin. @bin must be set
- * to the NULL state before calling this.
+ * Remove the elements of @stream from @bin.
*
* Return: %TRUE on success.
*/
gst_rtsp_stream_leave_bin (GstRTSPStream * stream, GstBin * bin,
GstElement * rtpbin)
{
+ GstRTSPStreamPrivate *priv;
gint i;
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
g_return_val_if_fail (GST_IS_BIN (bin), FALSE);
g_return_val_if_fail (GST_IS_ELEMENT (rtpbin), FALSE);
- g_mutex_lock (&stream->lock);
- if (!stream->is_joined)
+ priv = stream->priv;
+
+ g_mutex_lock (&priv->lock);
+ if (!priv->is_joined)
goto was_not_joined;
/* all transports must be removed by now */
- g_return_val_if_fail (stream->transports == NULL, FALSE);
+ g_return_val_if_fail (priv->transports == NULL, FALSE);
GST_INFO ("stream %p leaving bin", stream);
- gst_pad_unlink (stream->srcpad, stream->send_rtp_sink);
- g_signal_handler_disconnect (stream->send_rtp_sink, stream->caps_sig);
- gst_element_release_request_pad (rtpbin, stream->send_rtp_sink);
- gst_object_unref (stream->send_rtp_sink);
- stream->send_rtp_sink = NULL;
+ gst_pad_unlink (priv->srcpad, priv->send_rtp_sink);
+ g_signal_handler_disconnect (priv->send_rtp_sink, priv->caps_sig);
+ gst_element_release_request_pad (rtpbin, priv->send_rtp_sink);
+ gst_object_unref (priv->send_rtp_sink);
+ priv->send_rtp_sink = NULL;
for (i = 0; i < 2; i++) {
- /* and set udpsrc to NULL now before removing */
- gst_element_set_locked_state (stream->udpsrc[i], FALSE);
- gst_element_set_state (stream->udpsrc[i], GST_STATE_NULL);
-
- /* removing them should also nicely release the request
- * pads when they finalize */
- gst_bin_remove (bin, stream->udpsrc[i]);
- gst_bin_remove (bin, stream->udpsink[i]);
- gst_bin_remove (bin, stream->appsrc[i]);
- gst_bin_remove (bin, stream->appsink[i]);
- gst_bin_remove (bin, stream->appqueue[i]);
- gst_bin_remove (bin, stream->tee[i]);
- gst_bin_remove (bin, stream->funnel[i]);
-
- gst_element_release_request_pad (rtpbin, stream->recv_sink[i]);
- gst_object_unref (stream->recv_sink[i]);
- stream->recv_sink[i] = NULL;
-
- stream->udpsrc[i] = NULL;
- stream->udpsink[i] = NULL;
- stream->appsrc[i] = NULL;
- stream->appsink[i] = NULL;
- stream->appqueue[i] = NULL;
- stream->tee[i] = NULL;
- stream->funnel[i] = 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_v4[i], GST_STATE_NULL);
+ /* removing them should also nicely release the request
+ * pads when they finalize */
+ gst_bin_remove (bin, priv->udpsrc_v4[i]);
+ }
+ if (priv->udpsrc_v6[i]) {
+ gst_element_set_locked_state (priv->udpsrc_v6[i], FALSE);
+ gst_element_set_state (priv->udpsrc_v6[i], GST_STATE_NULL);
+ gst_bin_remove (bin, priv->udpsrc_v6[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]);
+ priv->recv_sink[i] = NULL;
+
+ priv->udpsrc_v4[i] = NULL;
+ priv->udpsrc_v6[i] = NULL;
+ priv->udpsink[i] = NULL;
+ priv->appsrc[i] = NULL;
+ priv->appsink[i] = NULL;
+ priv->appqueue[i] = NULL;
+ priv->tee[i] = NULL;
+ priv->funnel[i] = NULL;
}
- gst_object_unref (stream->send_src[0]);
- stream->send_src[0] = NULL;
+ gst_object_unref (priv->send_src[0]);
+ priv->send_src[0] = NULL;
- gst_element_release_request_pad (rtpbin, stream->send_src[1]);
- gst_object_unref (stream->send_src[1]);
- stream->send_src[1] = NULL;
+ gst_element_release_request_pad (rtpbin, priv->send_src[1]);
+ gst_object_unref (priv->send_src[1]);
+ priv->send_src[1] = NULL;
- g_object_unref (stream->session);
- if (stream->caps)
- gst_caps_unref (stream->caps);
+ g_object_unref (priv->session);
+ priv->session = NULL;
+ if (priv->caps)
+ gst_caps_unref (priv->caps);
+ priv->caps = NULL;
- stream->is_joined = FALSE;
- g_mutex_unlock (&stream->lock);
+ priv->is_joined = FALSE;
+ g_mutex_unlock (&priv->lock);
return TRUE;
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;
- payobjclass = G_OBJECT_GET_CLASS (stream->payloader);
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
- if (!g_object_class_find_property (payobjclass, "seqnum") ||
- !g_object_class_find_property (payobjclass, "timestamp"))
- return FALSE;
+ priv = stream->priv;
+
+ payobjclass = G_OBJECT_GET_CLASS (priv->payloader);
+
+ 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);
- g_object_get (stream->payloader, "seqnum", seq, "timestamp", rtptime, NULL);
+ if (rtptime)
+ g_object_get (priv->payloader, "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;
+ }
+}
+
+/**
+ * gst_rtsp_stream_get_caps:
+ * @stream: a #GstRTSPStream
+ *
+ * Retrieve the current caps of @stream.
+ *
+ * Returns: (transfer full): the #GstCaps of @stream. use gst_caps_unref()
+ * after usage.
+ */
+GstCaps *
+gst_rtsp_stream_get_caps (GstRTSPStream * stream)
+{
+ GstRTSPStreamPrivate *priv;
+ GstCaps *result;
+
+ g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
+
+ priv = stream->priv;
+
+ g_mutex_lock (&priv->lock);
+ if ((result = priv->caps))
+ gst_caps_ref (result);
+ g_mutex_unlock (&priv->lock);
+
+ return result;
}
/**
GstFlowReturn
gst_rtsp_stream_recv_rtp (GstRTSPStream * stream, GstBuffer * buffer)
{
+ GstRTSPStreamPrivate *priv;
GstFlowReturn ret;
GstElement *element;
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), GST_FLOW_ERROR);
+ priv = stream->priv;
g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
- g_return_val_if_fail (stream->is_joined, FALSE);
-
- g_mutex_lock (&stream->lock);
- element = gst_object_ref (stream->appsrc[0]);
- g_mutex_unlock (&stream->lock);
+ g_return_val_if_fail (priv->is_joined, FALSE);
- ret = gst_app_src_push_buffer (GST_APP_SRC_CAST (element), buffer);
-
- gst_object_unref (element);
+ g_mutex_lock (&priv->lock);
+ if (priv->appsrc[0])
+ element = gst_object_ref (priv->appsrc[0]);
+ else
+ element = NULL;
+ g_mutex_unlock (&priv->lock);
+ 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;
}
GstFlowReturn
gst_rtsp_stream_recv_rtcp (GstRTSPStream * stream, GstBuffer * buffer)
{
+ GstRTSPStreamPrivate *priv;
GstFlowReturn ret;
GstElement *element;
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), GST_FLOW_ERROR);
+ priv = stream->priv;
g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
- g_return_val_if_fail (stream->is_joined, FALSE);
+ g_return_val_if_fail (priv->is_joined, FALSE);
- g_mutex_lock (&stream->lock);
- element = gst_object_ref (stream->appsrc[1]);
- g_mutex_unlock (&stream->lock);
-
- ret = gst_app_src_push_buffer (GST_APP_SRC_CAST (element), buffer);
-
- gst_object_unref (element);
+ g_mutex_lock (&priv->lock);
+ if (priv->appsrc[1])
+ element = gst_object_ref (priv->appsrc[1]);
+ else
+ element = NULL;
+ g_mutex_unlock (&priv->lock);
+ 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;
}
update_transport (GstRTSPStream * stream, GstRTSPStreamTransport * trans,
gboolean add)
{
- GstRTSPTransport *tr;
- gboolean updated;
-
- updated = FALSE;
+ GstRTSPStreamPrivate *priv = stream->priv;
+ const GstRTSPTransport *tr;
- tr = trans->transport;
+ tr = gst_rtsp_stream_transport_get_transport (trans);
switch (tr->lower_transport) {
case GST_RTSP_LOWER_TRANS_UDP:
max = tr->client_port.max;
}
- if (add && !trans->active) {
- GST_INFO ("adding %s:%d-%d", dest, min, max);
- g_signal_emit_by_name (stream->udpsink[0], "add", dest, min, NULL);
- g_signal_emit_by_name (stream->udpsink[1], "add", dest, max, NULL);
+ if (add) {
if (ttl > 0) {
GST_INFO ("setting ttl-mc %d", ttl);
- g_object_set (G_OBJECT (stream->udpsink[0]), "ttl-mc", ttl, NULL);
- g_object_set (G_OBJECT (stream->udpsink[1]), "ttl-mc", ttl, NULL);
+ g_object_set (G_OBJECT (priv->udpsink[0]), "ttl-mc", ttl, NULL);
+ g_object_set (G_OBJECT (priv->udpsink[1]), "ttl-mc", ttl, NULL);
}
- stream->transports = g_list_prepend (stream->transports, trans);
- trans->active = TRUE;
- updated = TRUE;
- } else if (trans->active) {
+ 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);
- g_signal_emit_by_name (stream->udpsink[0], "remove", dest, min, NULL);
- g_signal_emit_by_name (stream->udpsink[1], "remove", dest, max, NULL);
- stream->transports = g_list_remove (stream->transports, trans);
- trans->active = FALSE;
- updated = TRUE;
+ g_signal_emit_by_name (priv->udpsink[0], "remove", dest, min, NULL);
+ g_signal_emit_by_name (priv->udpsink[1], "remove", dest, max, NULL);
+ priv->transports = g_list_remove (priv->transports, trans);
}
break;
}
case GST_RTSP_LOWER_TRANS_TCP:
- if (add && !trans->active) {
+ if (add) {
GST_INFO ("adding TCP %s", tr->destination);
- stream->transports = g_list_prepend (stream->transports, trans);
- trans->active = TRUE;
- updated = TRUE;
- } else if (trans->active) {
+ priv->transports = g_list_prepend (priv->transports, trans);
+ } else {
GST_INFO ("removing TCP %s", tr->destination);
- stream->transports = g_list_remove (stream->transports, trans);
- trans->active = FALSE;
- updated = TRUE;
+ priv->transports = g_list_remove (priv->transports, trans);
}
break;
default:
- GST_INFO ("Unknown transport %d", tr->lower_transport);
- break;
+ goto unknown_transport;
+ }
+ return TRUE;
+
+ /* ERRORS */
+unknown_transport:
+ {
+ GST_INFO ("Unknown transport %d", tr->lower_transport);
+ return FALSE;
}
- return updated;
}
gst_rtsp_stream_add_transport (GstRTSPStream * stream,
GstRTSPStreamTransport * trans)
{
+ GstRTSPStreamPrivate *priv;
gboolean res;
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
+ priv = stream->priv;
g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), FALSE);
- g_return_val_if_fail (stream->is_joined, FALSE);
- g_return_val_if_fail (trans->transport != NULL, FALSE);
+ g_return_val_if_fail (priv->is_joined, FALSE);
- g_mutex_lock (&stream->lock);
+ g_mutex_lock (&priv->lock);
res = update_transport (stream, trans, TRUE);
- g_mutex_unlock (&stream->lock);
+ g_mutex_unlock (&priv->lock);
return res;
}
gst_rtsp_stream_remove_transport (GstRTSPStream * stream,
GstRTSPStreamTransport * trans)
{
+ GstRTSPStreamPrivate *priv;
gboolean res;
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
+ priv = stream->priv;
g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), FALSE);
- g_return_val_if_fail (stream->is_joined, FALSE);
- g_return_val_if_fail (trans->transport != NULL, FALSE);
+ g_return_val_if_fail (priv->is_joined, FALSE);
- g_mutex_lock (&stream->lock);
+ g_mutex_lock (&priv->lock);
res = update_transport (stream, trans, FALSE);
- g_mutex_unlock (&stream->lock);
+ g_mutex_unlock (&priv->lock);
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