mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-18 07:47:17 +00:00
rtsp: use RTCP to keep the session alive
Use the RTCP rtcp-from stats field to find the associated session and use this to keep the session alive.
This commit is contained in:
Wim Taymans
parent
7bbdf7bf97
commit
9bed89c3b7
5 changed files with 288 additions and 139 deletions
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@ -35,13 +35,14 @@ enum
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PROP_LAST
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};
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static void gst_rtsp_client_get_property (GObject *object, guint propid,
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GValue *value, GParamSpec *pspec);
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static void gst_rtsp_client_set_property (GObject *object, guint propid,
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const GValue *value, GParamSpec *pspec);
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static void gst_rtsp_client_get_property (GObject * object, guint propid,
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GValue * value, GParamSpec * pspec);
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static void gst_rtsp_client_set_property (GObject * object, guint propid,
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const GValue * value, GParamSpec * pspec);
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static void gst_rtsp_client_finalize (GObject * obj);
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static void client_session_finalized (GstRTSPClient *client, GstRTSPSession *session);
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static void client_session_finalized (GstRTSPClient * client,
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GstRTSPSession * session);
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G_DEFINE_TYPE (GstRTSPClient, gst_rtsp_client, G_TYPE_OBJECT);
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@ -59,14 +60,17 @@ gst_rtsp_client_class_init (GstRTSPClientClass * klass)
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g_object_class_install_property (gobject_class, PROP_SESSION_POOL,
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g_param_spec_object ("session-pool", "Session Pool",
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"The session pool to use for client session",
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GST_TYPE_RTSP_SESSION_POOL, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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GST_TYPE_RTSP_SESSION_POOL,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, PROP_MEDIA_MAPPING,
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g_param_spec_object ("media-mapping", "Media Mapping",
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"The media mapping to use for client session",
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GST_TYPE_RTSP_MEDIA_MAPPING, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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GST_TYPE_RTSP_MEDIA_MAPPING,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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tunnels = g_hash_table_new_full (g_str_hash, g_str_equal, g_free, g_object_unref);
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tunnels =
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g_hash_table_new_full (g_str_hash, g_str_equal, g_free, g_object_unref);
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tunnels_lock = g_mutex_new ();
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}
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@ -100,8 +104,8 @@ gst_rtsp_client_finalize (GObject * obj)
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}
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static void
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gst_rtsp_client_get_property (GObject *object, guint propid,
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GValue *value, GParamSpec *pspec)
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gst_rtsp_client_get_property (GObject * object, guint propid,
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GValue * value, GParamSpec * pspec)
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{
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GstRTSPClient *client = GST_RTSP_CLIENT (object);
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@ -118,8 +122,8 @@ gst_rtsp_client_get_property (GObject *object, guint propid,
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}
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static void
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gst_rtsp_client_set_property (GObject *object, guint propid,
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const GValue *value, GParamSpec *pspec)
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gst_rtsp_client_set_property (GObject * object, guint propid,
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const GValue * value, GParamSpec * pspec)
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{
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GstRTSPClient *client = GST_RTSP_CLIENT (object);
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@ -151,9 +155,11 @@ gst_rtsp_client_new (void)
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}
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static void
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send_response (GstRTSPClient *client, GstRTSPSession *session, GstRTSPMessage *response)
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send_response (GstRTSPClient * client, GstRTSPSession * session,
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GstRTSPMessage * response)
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{
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gst_rtsp_message_add_header (response, GST_RTSP_HDR_SERVER, "GStreamer RTSP server");
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gst_rtsp_message_add_header (response, GST_RTSP_HDR_SERVER,
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"GStreamer RTSP server");
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/* remove any previous header */
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gst_rtsp_message_remove_header (response, GST_RTSP_HDR_SESSION, -1);
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@ -163,13 +169,14 @@ send_response (GstRTSPClient *client, GstRTSPSession *session, GstRTSPMessage *r
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gchar *str;
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if (session->timeout != 60)
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str = g_strdup_printf ("%s; timeout=%d", session->sessionid, session->timeout);
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str =
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g_strdup_printf ("%s; timeout=%d", session->sessionid,
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session->timeout);
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else
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str = g_strdup (session->sessionid);
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gst_rtsp_message_take_header (response, GST_RTSP_HDR_SESSION, str);
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}
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#ifdef DEBUG
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gst_rtsp_message_dump (response);
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#endif
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@ -179,19 +186,19 @@ send_response (GstRTSPClient *client, GstRTSPSession *session, GstRTSPMessage *r
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}
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static void
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send_generic_response (GstRTSPClient *client, GstRTSPStatusCode code,
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GstRTSPMessage *request)
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send_generic_response (GstRTSPClient * client, GstRTSPStatusCode code,
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GstRTSPMessage * request)
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{
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GstRTSPMessage response = { 0 };
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gst_rtsp_message_init_response (&response, code,
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gst_rtsp_status_as_text (code), request);
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gst_rtsp_message_init_response (&response, code,
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gst_rtsp_status_as_text (code), request);
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send_response (client, NULL, &response);
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}
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static gboolean
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compare_uri (const GstRTSPUrl *uri1, const GstRTSPUrl *uri2)
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compare_uri (const GstRTSPUrl * uri1, const GstRTSPUrl * uri2)
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{
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if (uri1 == NULL || uri2 == NULL)
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return FALSE;
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@ -206,7 +213,7 @@ compare_uri (const GstRTSPUrl *uri1, const GstRTSPUrl *uri2)
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* but is cached for when the same client (without breaking the connection) is
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* doing a setup for the exact same url. */
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static GstRTSPMedia *
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find_media (GstRTSPClient *client, GstRTSPUrl *uri, GstRTSPMessage *request)
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find_media (GstRTSPClient * client, GstRTSPUrl * uri, GstRTSPMessage * request)
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{
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GstRTSPMediaFactory *factory;
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GstRTSPMedia *media;
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@ -225,7 +232,8 @@ find_media (GstRTSPClient *client, GstRTSPUrl *uri, GstRTSPMessage *request)
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goto no_mapping;
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/* find the factory for the uri first */
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if (!(factory = gst_rtsp_media_mapping_find_factory (client->media_mapping, uri)))
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if (!(factory =
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gst_rtsp_media_mapping_find_factory (client->media_mapping, uri)))
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goto no_factory;
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/* prepare the media and add it to the pipeline */
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@ -239,11 +247,10 @@ find_media (GstRTSPClient *client, GstRTSPUrl *uri, GstRTSPMessage *request)
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/* now keep track of the uri and the media */
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client->uri = gst_rtsp_url_copy (uri);
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client->media = media;
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}
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else {
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} else {
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/* we have seen this uri before, used cached media */
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media = client->media;
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g_message ("reusing cached media %p", media);
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g_message ("reusing cached media %p", media);
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}
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if (media)
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@ -278,7 +285,7 @@ no_prepare:
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}
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static gboolean
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do_send_data (GstBuffer *buffer, guint8 channel, GstRTSPClient *client)
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do_send_data (GstBuffer * buffer, guint8 channel, GstRTSPClient * client)
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{
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GstRTSPMessage message = { 0 };
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guint8 *data;
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@ -299,38 +306,36 @@ do_send_data (GstBuffer *buffer, guint8 channel, GstRTSPClient *client)
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}
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static void
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link_stream (GstRTSPClient *client, GstRTSPSessionStream *stream)
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link_stream (GstRTSPClient * client, GstRTSPSessionStream * stream)
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{
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gst_rtsp_session_stream_set_callbacks (stream, (GstRTSPSendFunc) do_send_data,
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(GstRTSPSendFunc) do_send_data, client, NULL);
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(GstRTSPSendFunc) do_send_data, client, NULL);
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client->streams = g_list_prepend (client->streams, stream);
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}
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static void
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unlink_stream (GstRTSPClient *client, GstRTSPSessionStream *stream)
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unlink_stream (GstRTSPClient * client, GstRTSPSessionStream * stream)
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{
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gst_rtsp_session_stream_set_callbacks (stream, NULL,
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NULL, NULL, NULL);
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gst_rtsp_session_stream_set_callbacks (stream, NULL, NULL, NULL, NULL);
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client->streams = g_list_remove (client->streams, stream);
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}
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static void
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unlink_streams (GstRTSPClient *client)
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unlink_streams (GstRTSPClient * client)
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{
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GList *walk;
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for (walk = client->streams; walk; walk = g_list_next (walk)) {
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GstRTSPSessionStream *stream = (GstRTSPSessionStream *) walk->data;
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gst_rtsp_session_stream_set_callbacks (stream, NULL,
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NULL, NULL, NULL);
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gst_rtsp_session_stream_set_callbacks (stream, NULL, NULL, NULL, NULL);
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}
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g_list_free (client->streams);
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client->streams = NULL;
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}
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static void
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unlink_session_streams (GstRTSPClient *client, GstRTSPSessionMedia *media)
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unlink_session_streams (GstRTSPClient * client, GstRTSPSessionMedia * media)
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{
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guint n_streams, i;
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@ -353,7 +358,8 @@ unlink_session_streams (GstRTSPClient *client, GstRTSPSessionMedia *media)
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}
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static gboolean
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handle_teardown_request (GstRTSPClient *client, GstRTSPUrl *uri, GstRTSPSession *session, GstRTSPMessage *request)
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handle_teardown_request (GstRTSPClient * client, GstRTSPUrl * uri,
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GstRTSPSession * session, GstRTSPMessage * request)
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{
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GstRTSPSessionMedia *media;
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GstRTSPMessage response = { 0 };
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@ -371,7 +377,8 @@ handle_teardown_request (GstRTSPClient *client, GstRTSPUrl *uri, GstRTSPSession
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unlink_session_streams (client, media);
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/* remove the session from the watched sessions */
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g_object_weak_unref (G_OBJECT (session), (GWeakNotify) client_session_finalized, client);
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g_object_weak_unref (G_OBJECT (session),
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(GWeakNotify) client_session_finalized, client);
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client->sessions = g_list_remove (client->sessions, session);
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gst_rtsp_session_media_set_state (media, GST_STATE_NULL);
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@ -384,7 +391,8 @@ handle_teardown_request (GstRTSPClient *client, GstRTSPUrl *uri, GstRTSPSession
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}
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/* construct the response now */
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code = GST_RTSP_STS_OK;
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gst_rtsp_message_init_response (&response, code, gst_rtsp_status_as_text (code), request);
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gst_rtsp_message_init_response (&response, code,
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gst_rtsp_status_as_text (code), request);
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send_response (client, session, &response);
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@ -404,7 +412,8 @@ not_found:
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}
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static gboolean
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handle_pause_request (GstRTSPClient *client, GstRTSPUrl *uri, GstRTSPSession *session, GstRTSPMessage *request)
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handle_pause_request (GstRTSPClient * client, GstRTSPUrl * uri,
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GstRTSPSession * session, GstRTSPMessage * request)
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{
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GstRTSPSessionMedia *media;
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GstRTSPMessage response = { 0 };
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@ -431,7 +440,8 @@ handle_pause_request (GstRTSPClient *client, GstRTSPUrl *uri, GstRTSPSession *se
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/* construct the response now */
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code = GST_RTSP_STS_OK;
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gst_rtsp_message_init_response (&response, code, gst_rtsp_status_as_text (code), request);
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gst_rtsp_message_init_response (&response, code,
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gst_rtsp_status_as_text (code), request);
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send_response (client, session, &response);
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@ -453,13 +463,15 @@ not_found:
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}
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invalid_state:
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{
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send_generic_response (client, GST_RTSP_STS_METHOD_NOT_VALID_IN_THIS_STATE, request);
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send_generic_response (client, GST_RTSP_STS_METHOD_NOT_VALID_IN_THIS_STATE,
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request);
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return FALSE;
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}
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}
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static gboolean
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handle_play_request (GstRTSPClient *client, GstRTSPUrl *uri, GstRTSPSession *session, GstRTSPMessage *request)
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handle_play_request (GstRTSPClient * client, GstRTSPUrl * uri,
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GstRTSPSession * session, GstRTSPMessage * request)
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{
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GstRTSPSessionMedia *media;
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GstRTSPMessage response = { 0 };
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@ -508,8 +520,10 @@ handle_play_request (GstRTSPClient *client, GstRTSPUrl *uri, GstRTSPSession *ses
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/* get the stream as configured in the session */
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sstream = gst_rtsp_session_media_get_stream (media, i);
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/* get the transport, if there is no transport configured, skip this stream */
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if (!(tr = sstream->trans.transport))
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if (!(tr = sstream->trans.transport)) {
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g_message ("stream %d is not configured", i);
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continue;
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}
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if (tr->lower_transport == GST_RTSP_LOWER_TRANS_TCP) {
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/* for TCP, link the stream to the TCP connection of the client */
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@ -521,7 +535,7 @@ handle_play_request (GstRTSPClient *client, GstRTSPUrl *uri, GstRTSPSession *ses
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payobjclass = G_OBJECT_GET_CLASS (stream->payloader);
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if (g_object_class_find_property (payobjclass, "seqnum") &&
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g_object_class_find_property (payobjclass, "timestamp")) {
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g_object_class_find_property (payobjclass, "timestamp")) {
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GObject *payobj;
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payobj = G_OBJECT (stream->payloader);
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@ -533,26 +547,26 @@ handle_play_request (GstRTSPClient *client, GstRTSPUrl *uri, GstRTSPSession *ses
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g_string_append (rtpinfo, ", ");
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uristr = gst_rtsp_url_get_request_uri (uri);
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g_string_append_printf (rtpinfo, "url=%s/stream=%d;seq=%u;rtptime=%u", uristr, i, seqnum, timestamp);
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g_string_append_printf (rtpinfo, "url=%s/stream=%d;seq=%u;rtptime=%u",
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uristr, i, seqnum, timestamp);
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g_free (uristr);
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infocount++;
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}
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else {
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} else {
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g_warning ("RTP-Info cannot be determined for stream %d", i);
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}
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}
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/* construct the response now */
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code = GST_RTSP_STS_OK;
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gst_rtsp_message_init_response (&response, code, gst_rtsp_status_as_text (code), request);
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gst_rtsp_message_init_response (&response, code,
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gst_rtsp_status_as_text (code), request);
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/* add the RTP-Info header */
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if (infocount > 0) {
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str = g_string_free (rtpinfo, FALSE);
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gst_rtsp_message_take_header (&response, GST_RTSP_HDR_RTP_INFO, str);
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}
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else {
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} else {
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g_string_free (rtpinfo, TRUE);
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}
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@ -582,13 +596,22 @@ not_found:
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}
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invalid_state:
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{
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send_generic_response (client, GST_RTSP_STS_METHOD_NOT_VALID_IN_THIS_STATE, request);
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send_generic_response (client, GST_RTSP_STS_METHOD_NOT_VALID_IN_THIS_STATE,
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request);
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return FALSE;
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}
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}
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static void
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do_keepalive (GstRTSPSession * session)
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{
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g_message ("keep session %p alive", session);
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gst_rtsp_session_touch (session);
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}
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static gboolean
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handle_setup_request (GstRTSPClient *client, GstRTSPUrl *uri, GstRTSPSession *session, GstRTSPMessage *request)
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handle_setup_request (GstRTSPClient * client, GstRTSPUrl * uri,
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GstRTSPSession * session, GstRTSPMessage * request)
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{
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GstRTSPResult res;
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gchar *transport;
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@ -623,18 +646,20 @@ handle_setup_request (GstRTSPClient *client, GstRTSPUrl *uri, GstRTSPSession *se
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goto bad_request;
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/* parse the transport */
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res = gst_rtsp_message_get_header (request, GST_RTSP_HDR_TRANSPORT, &transport, 0);
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res =
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gst_rtsp_message_get_header (request, GST_RTSP_HDR_TRANSPORT, &transport,
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0);
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if (res != GST_RTSP_OK)
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goto no_transport;
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transports = g_strsplit (transport, ",", 0);
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gst_rtsp_transport_new (&ct);
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gst_rtsp_transport_new (&ct);
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/* loop through the transports, try to parse */
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have_transport = FALSE;
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for (i = 0; transports[i]; i++) {
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gst_rtsp_transport_init (ct);
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gst_rtsp_transport_init (ct);
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res = gst_rtsp_transport_parse (transports[i], ct);
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if (res == GST_RTSP_OK) {
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have_transport = TRUE;
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@ -644,7 +669,7 @@ handle_setup_request (GstRTSPClient *client, GstRTSPUrl *uri, GstRTSPSession *se
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g_strfreev (transports);
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/* we have not found anything usable, error out */
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if (!have_transport)
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if (!have_transport)
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goto unsupported_transports;
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/* we have a valid transport, check if we can handle it */
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|
|
@ -654,8 +679,7 @@ handle_setup_request (GstRTSPClient *client, GstRTSPUrl *uri, GstRTSPSession *se
|
|||
goto unsupported_transports;
|
||||
|
||||
supported = GST_RTSP_LOWER_TRANS_UDP |
|
||||
GST_RTSP_LOWER_TRANS_UDP_MCAST |
|
||||
GST_RTSP_LOWER_TRANS_TCP;
|
||||
GST_RTSP_LOWER_TRANS_UDP_MCAST | GST_RTSP_LOWER_TRANS_TCP;
|
||||
if (!(ct->lower_transport & supported))
|
||||
goto unsupported_transports;
|
||||
|
||||
|
|
@ -674,8 +698,7 @@ handle_setup_request (GstRTSPClient *client, GstRTSPUrl *uri, GstRTSPSession *se
|
|||
media = gst_rtsp_session_get_media (session, uri);
|
||||
|
||||
need_session = FALSE;
|
||||
}
|
||||
else {
|
||||
} else {
|
||||
/* create a session if this fails we probably reached our session limit or
|
||||
* something. */
|
||||
if (!(session = gst_rtsp_session_pool_create (client->session_pool)))
|
||||
|
|
@ -708,13 +731,18 @@ handle_setup_request (GstRTSPClient *client, GstRTSPUrl *uri, GstRTSPSession *se
|
|||
|
||||
st = gst_rtsp_session_stream_set_transport (stream, ct);
|
||||
|
||||
/* configure keepalive for this transport */
|
||||
gst_rtsp_session_stream_set_keepalive (stream,
|
||||
(GstRTSPKeepAliveFunc) do_keepalive, session, NULL);
|
||||
|
||||
/* serialize the server transport */
|
||||
trans_str = gst_rtsp_transport_as_text (st);
|
||||
gst_rtsp_transport_free (st);
|
||||
|
||||
/* construct the response now */
|
||||
code = GST_RTSP_STS_OK;
|
||||
gst_rtsp_message_init_response (&response, code, gst_rtsp_status_as_text (code), request);
|
||||
gst_rtsp_message_init_response (&response, code,
|
||||
gst_rtsp_status_as_text (code), request);
|
||||
|
||||
gst_rtsp_message_add_header (&response, GST_RTSP_HDR_TRANSPORT, trans_str);
|
||||
g_free (trans_str);
|
||||
|
|
@ -763,7 +791,7 @@ no_transport:
|
|||
unsupported_transports:
|
||||
{
|
||||
send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, request);
|
||||
gst_rtsp_transport_free (ct);
|
||||
gst_rtsp_transport_free (ct);
|
||||
return FALSE;
|
||||
}
|
||||
no_pool:
|
||||
|
|
@ -780,7 +808,8 @@ service_unavailable:
|
|||
|
||||
/* for the describe we must generate an SDP */
|
||||
static gboolean
|
||||
handle_describe_request (GstRTSPClient *client, GstRTSPUrl *uri, GstRTSPSession *session, GstRTSPMessage *request)
|
||||
handle_describe_request (GstRTSPClient * client, GstRTSPUrl * uri,
|
||||
GstRTSPSession * session, GstRTSPMessage * request)
|
||||
{
|
||||
GstRTSPMessage response = { 0 };
|
||||
GstRTSPResult res;
|
||||
|
|
@ -791,10 +820,11 @@ handle_describe_request (GstRTSPClient *client, GstRTSPUrl *uri, GstRTSPSession
|
|||
|
||||
/* check what kind of format is accepted, we don't really do anything with it
|
||||
* and always return SDP for now. */
|
||||
for (i = 0; i++; ) {
|
||||
for (i = 0; i++;) {
|
||||
gchar *accept;
|
||||
|
||||
res = gst_rtsp_message_get_header (request, GST_RTSP_HDR_ACCEPT, &accept, i);
|
||||
res =
|
||||
gst_rtsp_message_get_header (request, GST_RTSP_HDR_ACCEPT, &accept, i);
|
||||
if (res == GST_RTSP_ENOTIMPL)
|
||||
break;
|
||||
|
||||
|
|
@ -812,10 +842,11 @@ handle_describe_request (GstRTSPClient *client, GstRTSPUrl *uri, GstRTSPSession
|
|||
|
||||
g_object_unref (media);
|
||||
|
||||
gst_rtsp_message_init_response (&response, GST_RTSP_STS_OK,
|
||||
gst_rtsp_status_as_text (GST_RTSP_STS_OK), request);
|
||||
gst_rtsp_message_init_response (&response, GST_RTSP_STS_OK,
|
||||
gst_rtsp_status_as_text (GST_RTSP_STS_OK), request);
|
||||
|
||||
gst_rtsp_message_add_header (&response, GST_RTSP_HDR_CONTENT_TYPE, "application/sdp");
|
||||
gst_rtsp_message_add_header (&response, GST_RTSP_HDR_CONTENT_TYPE,
|
||||
"application/sdp");
|
||||
|
||||
/* content base for some clients that might screw up creating the setup uri */
|
||||
str = g_strdup_printf ("rtsp://%s:%u%s/", uri->host, uri->port, uri->abspath);
|
||||
|
|
@ -824,7 +855,7 @@ handle_describe_request (GstRTSPClient *client, GstRTSPUrl *uri, GstRTSPSession
|
|||
|
||||
/* add SDP to the response body */
|
||||
str = gst_sdp_message_as_text (sdp);
|
||||
gst_rtsp_message_take_body (&response, (guint8 *)str, strlen (str));
|
||||
gst_rtsp_message_take_body (&response, (guint8 *) str, strlen (str));
|
||||
gst_sdp_message_free (sdp);
|
||||
|
||||
send_response (client, session, &response);
|
||||
|
|
@ -846,25 +877,24 @@ no_sdp:
|
|||
}
|
||||
|
||||
static void
|
||||
handle_options_request (GstRTSPClient *client, GstRTSPUrl *uri, GstRTSPSession *session, GstRTSPMessage *request)
|
||||
handle_options_request (GstRTSPClient * client, GstRTSPUrl * uri,
|
||||
GstRTSPSession * session, GstRTSPMessage * request)
|
||||
{
|
||||
GstRTSPMessage response = { 0 };
|
||||
GstRTSPMethod options;
|
||||
gchar *str;
|
||||
|
||||
options = GST_RTSP_DESCRIBE |
|
||||
GST_RTSP_OPTIONS |
|
||||
GST_RTSP_PAUSE |
|
||||
GST_RTSP_PLAY |
|
||||
GST_RTSP_SETUP |
|
||||
GST_RTSP_GET_PARAMETER |
|
||||
GST_RTSP_SET_PARAMETER |
|
||||
GST_RTSP_TEARDOWN;
|
||||
GST_RTSP_OPTIONS |
|
||||
GST_RTSP_PAUSE |
|
||||
GST_RTSP_PLAY |
|
||||
GST_RTSP_SETUP |
|
||||
GST_RTSP_GET_PARAMETER | GST_RTSP_SET_PARAMETER | GST_RTSP_TEARDOWN;
|
||||
|
||||
str = gst_rtsp_options_as_text (options);
|
||||
|
||||
gst_rtsp_message_init_response (&response, GST_RTSP_STS_OK,
|
||||
gst_rtsp_status_as_text (GST_RTSP_STS_OK), request);
|
||||
gst_rtsp_message_init_response (&response, GST_RTSP_STS_OK,
|
||||
gst_rtsp_status_as_text (GST_RTSP_STS_OK), request);
|
||||
|
||||
gst_rtsp_message_add_header (&response, GST_RTSP_HDR_PUBLIC, str);
|
||||
g_free (str);
|
||||
|
|
@ -874,7 +904,7 @@ handle_options_request (GstRTSPClient *client, GstRTSPUrl *uri, GstRTSPSession *
|
|||
|
||||
/* remove duplicate and trailing '/' */
|
||||
static void
|
||||
santize_uri (GstRTSPUrl *uri)
|
||||
santize_uri (GstRTSPUrl * uri)
|
||||
{
|
||||
gint i, len;
|
||||
gchar *s, *d;
|
||||
|
|
@ -894,22 +924,22 @@ santize_uri (GstRTSPUrl *uri)
|
|||
}
|
||||
len = d - uri->abspath;
|
||||
/* don't remove the first slash if that's the only thing left */
|
||||
if (len > 1 && *(d-1) == '/')
|
||||
if (len > 1 && *(d - 1) == '/')
|
||||
d--;
|
||||
*d = '\0';
|
||||
}
|
||||
|
||||
static void
|
||||
client_session_finalized (GstRTSPClient *client, GstRTSPSession *session)
|
||||
client_session_finalized (GstRTSPClient * client, GstRTSPSession * session)
|
||||
{
|
||||
if (!(client->sessions = g_list_remove (client->sessions, session))) {
|
||||
g_message ("all sessions finalized, close the connection");
|
||||
g_source_destroy ((GSource*)client->watch);
|
||||
g_source_destroy ((GSource *) client->watch);
|
||||
}
|
||||
}
|
||||
|
||||
static void
|
||||
client_watch_session (GstRTSPClient *client, GstRTSPSession *session)
|
||||
client_watch_session (GstRTSPClient * client, GstRTSPSession * session)
|
||||
{
|
||||
GList *walk;
|
||||
|
||||
|
|
@ -923,12 +953,13 @@ client_watch_session (GstRTSPClient *client, GstRTSPSession *session)
|
|||
|
||||
g_message ("watching session %p", session);
|
||||
|
||||
g_object_weak_ref (G_OBJECT (session), (GWeakNotify) client_session_finalized, client);
|
||||
g_object_weak_ref (G_OBJECT (session), (GWeakNotify) client_session_finalized,
|
||||
client);
|
||||
client->sessions = g_list_prepend (client->sessions, session);
|
||||
}
|
||||
|
||||
static void
|
||||
handle_request (GstRTSPClient *client, GstRTSPMessage *request)
|
||||
handle_request (GstRTSPClient * client, GstRTSPMessage * request)
|
||||
{
|
||||
GstRTSPMethod method;
|
||||
const gchar *uristr;
|
||||
|
|
@ -948,7 +979,8 @@ handle_request (GstRTSPClient *client, GstRTSPMessage *request)
|
|||
|
||||
if (version != GST_RTSP_VERSION_1_0) {
|
||||
/* we can only handle 1.0 requests */
|
||||
send_generic_response (client, GST_RTSP_STS_RTSP_VERSION_NOT_SUPPORTED, request);
|
||||
send_generic_response (client, GST_RTSP_STS_RTSP_VERSION_NOT_SUPPORTED,
|
||||
request);
|
||||
return;
|
||||
}
|
||||
|
||||
|
|
@ -975,8 +1007,7 @@ handle_request (GstRTSPClient *client, GstRTSPMessage *request)
|
|||
* disappears because it times out, we will be notified. If all sessions are
|
||||
* gone, we will close the connection */
|
||||
client_watch_session (client, session);
|
||||
}
|
||||
else
|
||||
} else
|
||||
session = NULL;
|
||||
|
||||
/* now see what is asked and dispatch to a dedicated handler */
|
||||
|
|
@ -1033,7 +1064,7 @@ session_not_found:
|
|||
}
|
||||
|
||||
static void
|
||||
handle_data (GstRTSPClient *client, GstRTSPMessage *message)
|
||||
handle_data (GstRTSPClient * client, GstRTSPMessage * message)
|
||||
{
|
||||
GstRTSPResult res;
|
||||
guint8 channel;
|
||||
|
|
@ -1043,7 +1074,7 @@ handle_data (GstRTSPClient *client, GstRTSPMessage *message)
|
|||
GstBuffer *buffer;
|
||||
gboolean handled;
|
||||
|
||||
/* find the stream for this message */
|
||||
/* find the stream for this message */
|
||||
res = gst_rtsp_message_parse_data (message, &channel);
|
||||
if (res != GST_RTSP_OK)
|
||||
return;
|
||||
|
|
@ -1073,12 +1104,12 @@ handle_data (GstRTSPClient *client, GstRTSPMessage *message)
|
|||
if (tr->lower_transport == GST_RTSP_LOWER_TRANS_TCP) {
|
||||
/* dispatch to the stream based on the channel number */
|
||||
if (tr->interleaved.min == channel) {
|
||||
gst_rtsp_media_stream_rtp (mstream, buffer);
|
||||
handled = TRUE;
|
||||
gst_rtsp_media_stream_rtp (mstream, buffer);
|
||||
handled = TRUE;
|
||||
break;
|
||||
} else if (tr->interleaved.max == channel) {
|
||||
gst_rtsp_media_stream_rtcp (mstream, buffer);
|
||||
handled = TRUE;
|
||||
gst_rtsp_media_stream_rtcp (mstream, buffer);
|
||||
handled = TRUE;
|
||||
break;
|
||||
}
|
||||
}
|
||||
|
|
@ -1097,7 +1128,8 @@ handle_data (GstRTSPClient *client, GstRTSPMessage *message)
|
|||
* that created the client but can be overridden later.
|
||||
*/
|
||||
void
|
||||
gst_rtsp_client_set_session_pool (GstRTSPClient *client, GstRTSPSessionPool *pool)
|
||||
gst_rtsp_client_set_session_pool (GstRTSPClient * client,
|
||||
GstRTSPSessionPool * pool)
|
||||
{
|
||||
GstRTSPSessionPool *old;
|
||||
|
||||
|
|
@ -1120,7 +1152,7 @@ gst_rtsp_client_set_session_pool (GstRTSPClient *client, GstRTSPSessionPool *poo
|
|||
* Returns: a #GstRTSPSessionPool, unref after usage.
|
||||
*/
|
||||
GstRTSPSessionPool *
|
||||
gst_rtsp_client_get_session_pool (GstRTSPClient *client)
|
||||
gst_rtsp_client_get_session_pool (GstRTSPClient * client)
|
||||
{
|
||||
GstRTSPSessionPool *result;
|
||||
|
||||
|
|
@ -1140,7 +1172,8 @@ gst_rtsp_client_get_session_pool (GstRTSPClient *client)
|
|||
* created the client but can be overriden later.
|
||||
*/
|
||||
void
|
||||
gst_rtsp_client_set_media_mapping (GstRTSPClient *client, GstRTSPMediaMapping *mapping)
|
||||
gst_rtsp_client_set_media_mapping (GstRTSPClient * client,
|
||||
GstRTSPMediaMapping * mapping)
|
||||
{
|
||||
GstRTSPMediaMapping *old;
|
||||
|
||||
|
|
@ -1164,7 +1197,7 @@ gst_rtsp_client_set_media_mapping (GstRTSPClient *client, GstRTSPMediaMapping *m
|
|||
* Returns: a #GstRTSPMediaMapping, unref after usage.
|
||||
*/
|
||||
GstRTSPMediaMapping *
|
||||
gst_rtsp_client_get_media_mapping (GstRTSPClient *client)
|
||||
gst_rtsp_client_get_media_mapping (GstRTSPClient * client)
|
||||
{
|
||||
GstRTSPMediaMapping *result;
|
||||
|
||||
|
|
@ -1175,7 +1208,8 @@ gst_rtsp_client_get_media_mapping (GstRTSPClient *client)
|
|||
}
|
||||
|
||||
static GstRTSPResult
|
||||
message_received (GstRTSPWatch *watch, GstRTSPMessage *message, gpointer user_data)
|
||||
message_received (GstRTSPWatch * watch, GstRTSPMessage * message,
|
||||
gpointer user_data)
|
||||
{
|
||||
GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
|
||||
|
||||
|
|
@ -1195,7 +1229,7 @@ message_received (GstRTSPWatch *watch, GstRTSPMessage *message, gpointer user_da
|
|||
}
|
||||
|
||||
static GstRTSPResult
|
||||
message_sent (GstRTSPWatch *watch, guint cseq, gpointer user_data)
|
||||
message_sent (GstRTSPWatch * watch, guint cseq, gpointer user_data)
|
||||
{
|
||||
GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
|
||||
|
||||
|
|
@ -1205,7 +1239,7 @@ message_sent (GstRTSPWatch *watch, guint cseq, gpointer user_data)
|
|||
}
|
||||
|
||||
static GstRTSPResult
|
||||
closed (GstRTSPWatch *watch, gpointer user_data)
|
||||
closed (GstRTSPWatch * watch, gpointer user_data)
|
||||
{
|
||||
GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
|
||||
const gchar *tunnelid;
|
||||
|
|
@ -1225,7 +1259,7 @@ closed (GstRTSPWatch *watch, gpointer user_data)
|
|||
}
|
||||
|
||||
static GstRTSPResult
|
||||
error (GstRTSPWatch *watch, GstRTSPResult result, gpointer user_data)
|
||||
error (GstRTSPWatch * watch, GstRTSPResult result, gpointer user_data)
|
||||
{
|
||||
GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
|
||||
gchar *str;
|
||||
|
|
@ -1238,7 +1272,7 @@ error (GstRTSPWatch *watch, GstRTSPResult result, gpointer user_data)
|
|||
}
|
||||
|
||||
static GstRTSPStatusCode
|
||||
tunnel_start (GstRTSPWatch *watch, gpointer user_data)
|
||||
tunnel_start (GstRTSPWatch * watch, gpointer user_data)
|
||||
{
|
||||
GstRTSPClient *client;
|
||||
const gchar *tunnelid;
|
||||
|
|
@ -1272,7 +1306,7 @@ tunnel_existed:
|
|||
}
|
||||
|
||||
static GstRTSPResult
|
||||
tunnel_complete (GstRTSPWatch *watch, gpointer user_data)
|
||||
tunnel_complete (GstRTSPWatch * watch, gpointer user_data)
|
||||
{
|
||||
const gchar *tunnelid;
|
||||
GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
|
||||
|
|
@ -1336,7 +1370,7 @@ static GstRTSPWatchFuncs watch_funcs = {
|
|||
* Returns: %TRUE if the client could be accepted.
|
||||
*/
|
||||
gboolean
|
||||
gst_rtsp_client_accept (GstRTSPClient *client, GIOChannel *channel)
|
||||
gst_rtsp_client_accept (GstRTSPClient * client, GIOChannel * channel)
|
||||
{
|
||||
int sock;
|
||||
GstRTSPConnection *conn;
|
||||
|
|
@ -1351,14 +1385,13 @@ gst_rtsp_client_accept (GstRTSPClient *client, GIOChannel *channel)
|
|||
GST_RTSP_CHECK (gst_rtsp_connection_accept (sock, &conn), accept_failed);
|
||||
|
||||
url = gst_rtsp_connection_get_url (conn);
|
||||
g_message ("added new client %p ip %s:%d", client,
|
||||
url->host, url->port);
|
||||
g_message ("added new client %p ip %s:%d", client, url->host, url->port);
|
||||
|
||||
client->connection = conn;
|
||||
|
||||
/* create watch for the connection and attach */
|
||||
client->watch = gst_rtsp_watch_new (client->connection, &watch_funcs,
|
||||
g_object_ref (client), g_object_unref);
|
||||
g_object_ref (client), g_object_unref);
|
||||
|
||||
/* find the context to add the watch */
|
||||
if ((source = g_main_current_source ()))
|
||||
|
|
@ -1378,8 +1411,7 @@ accept_failed:
|
|||
{
|
||||
gchar *str = gst_rtsp_strresult (res);
|
||||
|
||||
g_error ("Could not accept client on server socket %d: %s",
|
||||
sock, str);
|
||||
g_error ("Could not accept client on server socket %d: %s", sock, str);
|
||||
g_free (str);
|
||||
return FALSE;
|
||||
}
|
||||
|
|
|
|||
|
|
@ -17,6 +17,8 @@
|
|||
* Boston, MA 02111-1307, USA.
|
||||
*/
|
||||
|
||||
#include <string.h>
|
||||
|
||||
#include <gst/app/gstappsrc.h>
|
||||
#include <gst/app/gstappsink.h>
|
||||
|
||||
|
|
@ -39,6 +41,8 @@ enum
|
|||
SIGNAL_LAST
|
||||
};
|
||||
|
||||
static GQuark ssrc_stream_map_key;
|
||||
|
||||
static void gst_rtsp_media_get_property (GObject *object, guint propid,
|
||||
GValue *value, GParamSpec *pspec);
|
||||
static void gst_rtsp_media_set_property (GObject *object, guint propid,
|
||||
|
|
@ -87,6 +91,8 @@ gst_rtsp_media_class_init (GstRTSPMediaClass * klass)
|
|||
g_critical ("could not start bus thread: %s", error->message);
|
||||
}
|
||||
klass->handle_message = default_handle_message;
|
||||
|
||||
ssrc_stream_map_key = g_quark_from_static_string ("GstRTSPServer.stream");
|
||||
}
|
||||
|
||||
static void
|
||||
|
|
@ -106,6 +112,8 @@ gst_rtsp_media_stream_free (GstRTSPMediaStream *stream)
|
|||
if (stream->caps)
|
||||
gst_caps_unref (stream->caps);
|
||||
|
||||
g_list_free (stream->transports);
|
||||
|
||||
g_free (stream);
|
||||
}
|
||||
|
||||
|
|
@ -700,48 +708,112 @@ dump_structure (const GstStructure *s)
|
|||
g_free (sstr);
|
||||
}
|
||||
|
||||
static void
|
||||
on_new_ssrc (GObject *session, GObject *source, GstRTSPMedia *media)
|
||||
static GstRTSPMediaTrans *
|
||||
find_transport (GstRTSPMediaStream *stream, const gchar *rtcp_from)
|
||||
{
|
||||
g_message ("%p: new source %p", media, source);
|
||||
GList *walk;
|
||||
GstRTSPMediaTrans *result = NULL;
|
||||
const gchar *dest;
|
||||
guint port;
|
||||
|
||||
if (rtcp_from == NULL)
|
||||
return NULL;
|
||||
|
||||
dest = g_strrstr (rtcp_from, ":");
|
||||
if (dest == NULL)
|
||||
return NULL;
|
||||
|
||||
port = atoi (dest + 1);
|
||||
dest = g_strndup (rtcp_from, dest - rtcp_from);
|
||||
|
||||
g_message ("finding %s:%d", dest, port);
|
||||
|
||||
for (walk = stream->transports; walk; walk = g_list_next (walk)) {
|
||||
GstRTSPMediaTrans *trans = walk->data;
|
||||
gint min, max;
|
||||
|
||||
min = trans->transport->client_port.min;
|
||||
max = trans->transport->client_port.max;
|
||||
|
||||
if ((strcmp (trans->transport->destination, dest) == 0) && (min == port || max == port)) {
|
||||
result = trans;
|
||||
break;
|
||||
}
|
||||
}
|
||||
return result;
|
||||
}
|
||||
|
||||
static void
|
||||
on_ssrc_sdes (GObject *session, GObject *source, GstRTSPMedia *media)
|
||||
on_new_ssrc (GObject *session, GObject *source, GstRTSPMediaStream *stream)
|
||||
{
|
||||
GstStructure *stats;
|
||||
GstRTSPMediaTrans *trans;
|
||||
|
||||
g_message ("%p: new source %p", stream, source);
|
||||
|
||||
/* see if we have a stream to match with the origin of the RTCP packet */
|
||||
trans = g_object_get_qdata (source, ssrc_stream_map_key);
|
||||
if (trans == NULL) {
|
||||
g_object_get (source, "stats", &stats, NULL);
|
||||
if (stats) {
|
||||
const gchar *rtcp_from;
|
||||
|
||||
rtcp_from = gst_structure_get_string (stats, "rtcp-from");
|
||||
if ((trans = find_transport (stream, rtcp_from))) {
|
||||
g_message ("%p: found transport %p for source %p", stream, trans, source);
|
||||
g_object_set_qdata (source, ssrc_stream_map_key, trans);
|
||||
}
|
||||
}
|
||||
} else {
|
||||
g_message ("%p: source %p for transport %p", stream, source, trans);
|
||||
}
|
||||
}
|
||||
|
||||
static void
|
||||
on_ssrc_sdes (GObject *session, GObject *source, GstRTSPMediaStream *stream)
|
||||
{
|
||||
GstStructure *sdes;
|
||||
|
||||
g_message ("%p: new SDES %p", media, source);
|
||||
g_message ("%p: new SDES %p", stream, source);
|
||||
g_object_get (source, "sdes", &sdes, NULL);
|
||||
dump_structure (sdes);
|
||||
}
|
||||
|
||||
static void
|
||||
on_ssrc_active (GObject *session, GObject *source, GstRTSPMedia *media)
|
||||
on_ssrc_active (GObject *session, GObject *source, GstRTSPMediaStream *stream)
|
||||
{
|
||||
GstStructure *stats;
|
||||
GstRTSPMediaTrans *trans;
|
||||
|
||||
trans = g_object_get_qdata (source, ssrc_stream_map_key);
|
||||
|
||||
g_message ("%p: source %p in transport %p is active", stream, trans, source);
|
||||
|
||||
if (trans && trans->keep_alive) {
|
||||
trans->keep_alive (trans->ka_user_data);
|
||||
}
|
||||
|
||||
g_message ("%p: source %p is active", media, source);
|
||||
g_object_get (source, "stats", &stats, NULL);
|
||||
dump_structure (stats);
|
||||
gst_structure_free (stats);
|
||||
}
|
||||
|
||||
static void
|
||||
on_bye_ssrc (GObject *session, GObject *source, GstRTSPMedia *media)
|
||||
on_bye_ssrc (GObject *session, GObject *source, GstRTSPMediaStream *stream)
|
||||
{
|
||||
g_message ("%p: source %p bye", media, source);
|
||||
g_message ("%p: source %p bye", stream, source);
|
||||
}
|
||||
|
||||
static void
|
||||
on_bye_timeout (GObject *session, GObject *source, GstRTSPMedia *media)
|
||||
on_bye_timeout (GObject *session, GObject *source, GstRTSPMediaStream *stream)
|
||||
{
|
||||
g_message ("%p: source %p bye timeout", media, source);
|
||||
g_message ("%p: source %p bye timeout", stream, source);
|
||||
}
|
||||
|
||||
static void
|
||||
on_timeout (GObject *session, GObject *source, GstRTSPMedia *media)
|
||||
on_timeout (GObject *session, GObject *source, GstRTSPMediaStream *stream)
|
||||
{
|
||||
g_message ("%p: source %p timeout", media, source);
|
||||
g_message ("%p: source %p timeout", stream, source);
|
||||
}
|
||||
|
||||
static GstFlowReturn
|
||||
|
|
@ -836,17 +908,17 @@ setup_stream (GstRTSPMediaStream *stream, guint idx, GstRTSPMedia *media)
|
|||
&stream->session);
|
||||
|
||||
g_signal_connect (stream->session, "on-new-ssrc", (GCallback) on_new_ssrc,
|
||||
media);
|
||||
stream);
|
||||
g_signal_connect (stream->session, "on-ssrc-sdes", (GCallback) on_ssrc_sdes,
|
||||
media);
|
||||
stream);
|
||||
g_signal_connect (stream->session, "on-ssrc-active", (GCallback) on_ssrc_active,
|
||||
media);
|
||||
stream);
|
||||
g_signal_connect (stream->session, "on-bye-ssrc", (GCallback) on_bye_ssrc,
|
||||
media);
|
||||
stream);
|
||||
g_signal_connect (stream->session, "on-bye-timeout", (GCallback) on_bye_timeout,
|
||||
media);
|
||||
stream);
|
||||
g_signal_connect (stream->session, "on-timeout", (GCallback) on_timeout,
|
||||
media);
|
||||
stream);
|
||||
|
||||
/* link the RTP pad to the session manager */
|
||||
ret = gst_pad_link (stream->srcpad, stream->send_rtp_sink);
|
||||
|
|
@ -1361,12 +1433,14 @@ gst_rtsp_media_set_state (GstRTSPMedia *media, GstState state, GArray *transport
|
|||
g_message ("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);
|
||||
stream->transports = g_list_prepend (stream->transports, tr);
|
||||
tr->active = TRUE;
|
||||
media->active++;
|
||||
} else if (remove && tr->active) {
|
||||
g_message ("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, tr);
|
||||
tr->active = FALSE;
|
||||
media->active--;
|
||||
}
|
||||
|
|
|
|||
|
|
@ -41,7 +41,8 @@ typedef struct _GstRTSPMedia GstRTSPMedia;
|
|||
typedef struct _GstRTSPMediaClass GstRTSPMediaClass;
|
||||
typedef struct _GstRTSPMediaTrans GstRTSPMediaTrans;
|
||||
|
||||
typedef gboolean (*GstRTSPSendFunc) (GstBuffer *buffer, guint8 channel, gpointer user_data);
|
||||
typedef gboolean (*GstRTSPSendFunc) (GstBuffer *buffer, guint8 channel, gpointer user_data);
|
||||
typedef void (*GstRTSPKeepAliveFunc) (gpointer user_data);
|
||||
|
||||
/**
|
||||
* GstRTSPMediaTrans:
|
||||
|
|
@ -50,20 +51,33 @@ typedef gboolean (*GstRTSPSendFunc) (GstBuffer *buffer, guint8 channel, gpointer
|
|||
* @send_rtcp: callback for sending RTCP messages
|
||||
* @user_data: user data passed in the callbacks
|
||||
* @notify: free function for the user_data.
|
||||
* @keep_alive: keep alive callback
|
||||
* @ka_user_data: data passed to @keep_alive
|
||||
* @ka_notify: called when @ka_user_data is freed
|
||||
* @active: if we are actively sending
|
||||
* @timeout: if we timed out
|
||||
* @transport: a transport description
|
||||
* @rtpsource: the receiver rtp source object
|
||||
*
|
||||
* A Transport description for stream @idx
|
||||
*/
|
||||
struct _GstRTSPMediaTrans {
|
||||
guint idx;
|
||||
|
||||
GstRTSPSendFunc send_rtp;
|
||||
GstRTSPSendFunc send_rtcp;
|
||||
gpointer user_data;
|
||||
GDestroyNotify notify;
|
||||
gboolean active;
|
||||
GstRTSPSendFunc send_rtp;
|
||||
GstRTSPSendFunc send_rtcp;
|
||||
gpointer user_data;
|
||||
GDestroyNotify notify;
|
||||
|
||||
GstRTSPTransport *transport;
|
||||
GstRTSPKeepAliveFunc keep_alive;
|
||||
gpointer ka_user_data;
|
||||
GDestroyNotify ka_notify;
|
||||
gboolean active;
|
||||
gboolean timeout;
|
||||
|
||||
GstRTSPTransport *transport;
|
||||
|
||||
GObject *rtpsource;
|
||||
};
|
||||
|
||||
/**
|
||||
|
|
|
|||
|
|
@ -74,6 +74,10 @@ gst_rtsp_session_free_stream (GstRTSPSessionStream *stream)
|
|||
{
|
||||
g_message ("free session stream %p", stream);
|
||||
|
||||
/* remove callbacks now */
|
||||
gst_rtsp_session_stream_set_callbacks (stream, NULL, NULL, NULL, NULL);
|
||||
gst_rtsp_session_stream_set_keepalive (stream, NULL, NULL, NULL);
|
||||
|
||||
if (stream->trans.transport)
|
||||
gst_rtsp_transport_free (stream->trans.transport);
|
||||
|
||||
|
|
@ -308,7 +312,7 @@ gst_rtsp_session_media_get_stream (GstRTSPSessionMedia *media, guint idx)
|
|||
result->trans.transport = NULL;
|
||||
result->media_stream = media_stream;
|
||||
|
||||
g_array_insert_val (media->streams, idx, result);
|
||||
g_array_index (media->streams, GstRTSPSessionStream *, idx) = result;
|
||||
}
|
||||
return result;
|
||||
|
||||
|
|
@ -512,6 +516,27 @@ gst_rtsp_session_stream_set_callbacks (GstRTSPSessionStream *stream,
|
|||
stream->trans.notify = notify;
|
||||
}
|
||||
|
||||
/**
|
||||
* gst_rtsp_session_stream_set_keepalive:
|
||||
* @stream: a #GstRTSPSessionStream
|
||||
* @keep_alive: a callback called when the receiver is active
|
||||
* @user_data: user data passed to callback
|
||||
* @notify: called with the user_data when no longer needed.
|
||||
*
|
||||
* Install callbacks that will be called when RTCP packets are received from the
|
||||
* receiver of @stream.
|
||||
*/
|
||||
void
|
||||
gst_rtsp_session_stream_set_keepalive (GstRTSPSessionStream *stream,
|
||||
GstRTSPKeepAliveFunc keep_alive, gpointer user_data, GDestroyNotify notify)
|
||||
{
|
||||
stream->trans.keep_alive = keep_alive;
|
||||
if (stream->trans.ka_notify)
|
||||
stream->trans.ka_notify (stream->trans.ka_user_data);
|
||||
stream->trans.ka_user_data = user_data;
|
||||
stream->trans.ka_notify = notify;
|
||||
}
|
||||
|
||||
/**
|
||||
* gst_rtsp_session_media_set_state:
|
||||
* @media: a #GstRTSPSessionMedia
|
||||
|
|
|
|||
|
|
@ -146,6 +146,10 @@ void gst_rtsp_session_stream_set_callbacks (GstRTSPSessionStre
|
|||
GstRTSPSendFunc send_rtcp,
|
||||
gpointer user_data,
|
||||
GDestroyNotify notify);
|
||||
void gst_rtsp_session_stream_set_keepalive (GstRTSPSessionStream *stream,
|
||||
GstRTSPKeepAliveFunc keep_alive,
|
||||
gpointer user_data,
|
||||
GDestroyNotify notify);
|
||||
|
||||
G_END_DECLS
|
||||
|
||||
|
|
|
|||
Loading…
Reference in a new issue