X-Git-Url: http://review.tizen.org/git/?a=blobdiff_plain;f=gst%2Frtsp-server%2Frtsp-stream.c;h=fb200901a7d9a70dad0cb7c44aa5ba86136e7031;hb=48b6b8e3e60369347972298aa96546912970ada5;hp=8f2cd91044a3675df36f26824c479ba370bf4a0c;hpb=0ddd98bfa6420e87d33078a6ae22b73752e625d5;p=platform%2Fupstream%2Fgstreamer.git diff --git a/gst/rtsp-server/rtsp-stream.c b/gst/rtsp-server/rtsp-stream.c index 8f2cd91..fb20090 100644 --- a/gst/rtsp-server/rtsp-stream.c +++ b/gst/rtsp-server/rtsp-stream.c @@ -16,9 +16,37 @@ * 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 #include +#include +#include #include @@ -36,9 +64,12 @@ struct _GstRTSPStreamPrivate guint idx; GstPad *srcpad; GstElement *payloader; - gboolean is_ipv6; guint buffer_size; gboolean is_joined; + gchar *control; + + GstRTSPProfile profiles; + GstRTSPLowerTrans protocols; /* pads on the rtpbin */ GstPad *send_rtp_sink; @@ -48,10 +79,16 @@ struct _GstRTSPStreamPrivate /* the RTPSession object */ GObject *session; - /* sinks used for sending and receiving RTP and RTCP, they share + /* 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[2]; + GstElement *udpsrc_v6[2]; + GstElement *udpsink[2]; + /* for TCP transport */ GstElement *appsrc[2]; GstElement *appqueue[2]; @@ -60,13 +97,20 @@ struct _GstRTSPStreamPrivate GstElement *tee[2]; GstElement *funnel[2]; - /* server ports for sending/receiving */ - GstRTSPRange server_port; - GstRTSPAddress *server_addr; + /* 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; + GstRTSPAddress *addr_v4; + GstRTSPAddress *addr_v6; /* the caps of the stream */ gulong caps_sig; @@ -75,12 +119,25 @@ struct _GstRTSPStreamPrivate /* 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 }; @@ -89,6 +146,11 @@ GST_DEBUG_CATEGORY_STATIC (rtsp_stream_debug); 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); @@ -102,8 +164,25 @@ gst_rtsp_stream_class_init (GstRTSPStreamClass * klass) 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"); @@ -118,6 +197,11 @@ gst_rtsp_stream_init (GstRTSPStream * 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); } @@ -135,19 +219,66 @@ gst_rtsp_stream_finalize (GObject * obj) /* 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->server_addr) - gst_rtsp_address_free (priv->server_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) + 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 @@ -194,13 +325,36 @@ gst_rtsp_stream_get_index (GstRTSPStream * stream) 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. * - * Return: the srcpad. Unref after usage. + * Returns: (transfer full): the srcpad. Unref after usage. */ GstPad * gst_rtsp_stream_get_srcpad (GstRTSPStream * stream) @@ -211,6 +365,89 @@ 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; + + 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 @@ -254,6 +491,223 @@ gst_rtsp_stream_get_mtu (GstRTSPStream * stream) 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 @@ -313,35 +767,48 @@ gst_rtsp_stream_get_address_pool (GstRTSPStream * stream) } /** - * 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; @@ -364,11 +831,15 @@ no_address: /** * gst_rtsp_stream_reserve_address: * @stream: a #GstRTSPStream + * @address: an address + * @port: a port + * @n_ports: n_ports + * @ttl: a TTL * - * Get a specific multicast address of @stream. + * Reserve @address and @port as the address and port of @stream. * - * Returns: the #GstRTSPAddress of @stream or %NULL when no address could be - * allocated. gst_rtsp_address_free() after usage. + * 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, @@ -376,6 +847,9 @@ gst_rtsp_stream_reserve_address (GstRTSPStream * stream, { 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); @@ -385,22 +859,38 @@ gst_rtsp_stream_reserve_address (GstRTSPStream * stream, 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; @@ -428,11 +918,12 @@ different_address: } } -/* 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) { - GstRTSPStreamPrivate *priv = stream->priv; GstStateChangeReturn ret; GstElement *udpsrc0, *udpsrc1; GstElement *udpsink0, *udpsink1; @@ -443,10 +934,15 @@ alloc_ports (GstRTSPStream * stream) gint rtpport, rtcpport; GList *rejected_addresses = NULL; GstRTSPAddress *addr = NULL; - GSocketFamily family; 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; @@ -457,19 +953,13 @@ alloc_ports (GstRTSPStream * stream) /* Start with random port */ tmp_rtp = 0; - if (priv->is_ipv6) { - family = G_SOCKET_FAMILY_IPV6; - } else { - family = G_SOCKET_FAMILY_IPV4; - } - rtcp_socket = g_socket_new (family, G_SOCKET_TYPE_DATAGRAM, G_SOCKET_PROTOCOL_UDP, NULL); if (!rtcp_socket) goto no_udp_protocol; - if (priv->server_addr) - gst_rtsp_address_free (priv->server_addr); + 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 */ @@ -482,19 +972,19 @@ again: goto no_udp_protocol; } - if (priv->pool && gst_rtsp_address_pool_has_unicast_addresses (priv->pool)) { + 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 (priv->is_ipv6) + 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 (priv->pool, flags, 2); + addr = gst_rtsp_address_pool_acquire_address (pool, flags, 2); if (addr == NULL) goto no_ports; @@ -576,23 +1066,31 @@ again: 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_set (G_OBJECT (udpsink0), "close-socket", FALSE, NULL); - g_object_set (G_OBJECT (udpsink0), "socket", rtp_socket, 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; 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", priv->buffer_size, NULL); + g_object_set (G_OBJECT (udpsink0), "buffer-size", buffer_size, NULL); g_object_set (G_OBJECT (udpsink1), "close-socket", FALSE, NULL); - g_object_set (G_OBJECT (udpsink1), "socket", rtcp_socket, 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); @@ -602,14 +1100,14 @@ again: /* we keep these elements, we will further configure them when the * client told us to really use the UDP ports. */ - priv->udpsrc[0] = udpsrc0; - priv->udpsrc[1] = udpsrc1; - priv->udpsink[0] = udpsink0; - priv->udpsink[1] = udpsink1; - priv->server_port.min = rtpport; - priv->server_port.max = rtcpport; - - priv->server_addr = addr; + 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); @@ -652,10 +1150,6 @@ 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, @@ -670,17 +1164,35 @@ cleanup: } } +/* 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) + GstRTSPRange * server_port, GSocketFamily family) { GstRTSPStreamPrivate *priv; @@ -689,12 +1201,43 @@ gst_rtsp_stream_get_server_port (GstRTSPStream * stream, g_return_if_fail (priv->is_joined); g_mutex_lock (&priv->lock); - if (server_port) - *server_port = priv->server_port; + 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 @@ -787,6 +1330,8 @@ find_transport (GstRTSPStream * stream, const gchar * rtcp_from) break; } } + if (result) + g_object_ref (result); g_mutex_unlock (&priv->lock); g_free (dest); @@ -813,7 +1358,8 @@ check_transport (GObject * source, GstRTSPStream * stream) if ((trans = find_transport (stream, rtcp_from))) { GST_INFO ("%p: found transport %p for source %p", stream, trans, 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); } @@ -879,6 +1425,7 @@ on_bye_timeout (GObject * session, GObject * source, GstRTSPStream * stream) if ((trans = g_object_get_qdata (source, ssrc_stream_map_key))) { gst_rtsp_stream_transport_set_timed_out (trans, TRUE); + g_object_set_qdata (source, ssrc_stream_map_key, NULL); } } @@ -891,6 +1438,7 @@ on_timeout (GObject * session, GObject * source, GstRTSPStream * stream) if ((trans = g_object_get_qdata (source, ssrc_stream_map_key))) { gst_rtsp_stream_transport_set_timed_out (trans, TRUE); + g_object_set_qdata (source, ssrc_stream_map_key, NULL); } } @@ -941,7 +1489,7 @@ static GstAppSinkCallbacks sink_cb = { * @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. @@ -953,9 +1501,10 @@ gst_rtsp_stream_join_bin (GstRTSPStream * stream, GstBin * bin, 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); @@ -971,11 +1520,14 @@ gst_rtsp_stream_join_bin (GstRTSPStream * stream, GstBin * bin, /* 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; + /* 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); priv->send_rtp_sink = gst_element_get_request_pad (rtpbin, name); @@ -1018,6 +1570,7 @@ gst_rtsp_stream_join_bin (GstRTSPStream * stream, GstBin * bin, 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 @@ -1032,49 +1585,57 @@ gst_rtsp_stream_join_bin (GstRTSPStream * stream, GstBin * bin, * | | | 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]); - - /* 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); + 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 @@ -1097,37 +1658,61 @@ gst_rtsp_stream_join_bin (GstRTSPStream * stream, GstBin * bin, gst_pad_link (pad, priv->recv_sink[i]); gst_object_unref (pad); - /* we set and keep these to playing so that they don't cause NO_PREROLL return - * values */ - gst_element_set_state (priv->udpsrc[i], GST_STATE_PLAYING); - gst_element_set_locked_state (priv->udpsrc[i], TRUE); - /* add udpsrc */ - gst_bin_add (bin, priv->udpsrc[i]); - /* and link to the funnel */ - selpad = gst_element_get_request_pad (priv->funnel[i], "sink_%u"); - pad = gst_element_get_static_pad (priv->udpsrc[i], "src"); - gst_pad_link (pad, selpad); - gst_object_unref (pad); - 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->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 (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); } } @@ -1149,12 +1734,12 @@ was_joined: 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); @@ -1201,31 +1786,50 @@ gst_rtsp_stream_leave_bin (GstRTSPStream * stream, GstBin * bin, 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); - /* and set udpsrc to NULL now before removing */ - gst_element_set_locked_state (priv->udpsrc[i], FALSE); - gst_element_set_state (priv->udpsrc[i], GST_STATE_NULL); - - /* removing them should also nicely release the request - * pads when they finalize */ - gst_bin_remove (bin, priv->udpsrc[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_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[i] = NULL; + priv->udpsrc_v4[i] = NULL; + priv->udpsrc_v6[i] = NULL; priv->udpsink[i] = NULL; priv->appsrc[i] = NULL; priv->appsink[i] = NULL; @@ -1253,6 +1857,7 @@ gst_rtsp_stream_leave_bin (GstRTSPStream * stream, GstBin * bin, was_not_joined: { + g_mutex_unlock (&priv->lock); return TRUE; } } @@ -1260,36 +1865,78 @@ was_not_joined: /** * 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); - g_object_get (priv->payloader, "seqnum", seq, "timestamp", rtptime, NULL); + 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); + + 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; + } } /** @@ -1344,13 +1991,18 @@ gst_rtsp_stream_recv_rtp (GstRTSPStream * stream, GstBuffer * buffer) 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; } @@ -1379,13 +2031,18 @@ gst_rtsp_stream_recv_rtcp (GstRTSPStream * stream, GstBuffer * buffer) 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; } @@ -1418,14 +2075,14 @@ update_transport (GstRTSPStream * stream, GstRTSPStreamTransport * trans, } 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); @@ -1523,3 +2180,224 @@ gst_rtsp_stream_remove_transport (GstRTSPStream * stream, 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; +}