mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-07-05 14:16:06 +00:00
rtsp-media-factory: expose API to disable RTCP
This is supported by the RFC, and can be useful on systems where allocating two consecutive ports is problematic, and RTCP is not necessary. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/159>
This commit is contained in:
Mathieu Duponchelle
parent
5029335dcb
commit
1730940abd
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@ -22,12 +22,16 @@
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#include <gst/rtsp-server/rtsp-server.h>
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#define DEFAULT_RTSP_PORT "8554"
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#define DEFAULT_DISABLE_RTCP FALSE
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static char *port = (char *) DEFAULT_RTSP_PORT;
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static gboolean disable_rtcp = DEFAULT_DISABLE_RTCP;
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static GOptionEntry entries[] = {
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{"port", 'p', 0, G_OPTION_ARG_STRING, &port,
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"Port to listen on (default: " DEFAULT_RTSP_PORT ")", "PORT"},
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{"disable-rtcp", '\0', 0, G_OPTION_ARG_NONE, &disable_rtcp,
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"Whether RTCP should be disabled (default false)", NULL},
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{NULL}
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};
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@ -70,6 +74,7 @@ main (int argc, char *argv[])
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factory = gst_rtsp_media_factory_new ();
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gst_rtsp_media_factory_set_launch (factory, argv[1]);
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gst_rtsp_media_factory_set_shared (factory, TRUE);
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gst_rtsp_media_factory_set_enable_rtcp (factory, !disable_rtcp);
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/* attach the test factory to the /test url */
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gst_rtsp_mount_points_add_factory (mounts, "/test", factory);
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@ -41,6 +41,7 @@
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#include "config.h"
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#endif
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#include "rtsp-server-internal.h"
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#include "rtsp-media-factory.h"
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#define GST_RTSP_MEDIA_FACTORY_GET_LOCK(f) (&(GST_RTSP_MEDIA_FACTORY_CAST(f)->priv->lock))
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@ -65,6 +66,7 @@ struct _GstRTSPMediaFactoryPrivate
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gchar *multicast_iface;
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guint max_mcast_ttl;
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gboolean bind_mcast_address;
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gboolean enable_rtcp;
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GstClockTime rtx_time;
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guint latency;
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@ -95,6 +97,7 @@ struct _GstRTSPMediaFactoryPrivate
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#define DEFAULT_STOP_ON_DISCONNECT TRUE
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#define DEFAULT_DO_RETRANSMISSION FALSE
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#define DEFAULT_DSCP_QOS (-1)
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#define DEFAULT_ENABLE_RTCP TRUE
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enum
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{
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@ -113,6 +116,7 @@ enum
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PROP_MAX_MCAST_TTL,
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PROP_BIND_MCAST_ADDRESS,
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PROP_DSCP_QOS,
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PROP_ENABLE_RTCP,
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PROP_LAST
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};
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@ -247,6 +251,18 @@ gst_rtsp_media_factory_class_init (GstRTSPMediaFactoryClass * klass)
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DEFAULT_BIND_MCAST_ADDRESS,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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/**
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* GstRTSPMediaFactory:enable-rtcp:
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*
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* Whether the created media should send and receive RTCP
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*
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* Since: 1.20
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*/
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g_object_class_install_property (gobject_class, PROP_ENABLE_RTCP,
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g_param_spec_boolean ("enable-rtcp", "Enable RTCP",
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"Whether the created media should send and receive RTCP",
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DEFAULT_ENABLE_RTCP, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, PROP_DSCP_QOS,
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g_param_spec_int ("dscp-qos", "DSCP QoS",
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"The IP DSCP field to use", -1, 63,
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@ -295,6 +311,7 @@ gst_rtsp_media_factory_init (GstRTSPMediaFactory * factory)
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priv->do_retransmission = DEFAULT_DO_RETRANSMISSION;
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priv->max_mcast_ttl = DEFAULT_MAX_MCAST_TTL;
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priv->bind_mcast_address = DEFAULT_BIND_MCAST_ADDRESS;
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priv->enable_rtcp = DEFAULT_ENABLE_RTCP;
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priv->dscp_qos = DEFAULT_DSCP_QOS;
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g_mutex_init (&priv->lock);
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@ -381,6 +398,10 @@ gst_rtsp_media_factory_get_property (GObject * object, guint propid,
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case PROP_DSCP_QOS:
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g_value_set_int (value, gst_rtsp_media_factory_get_dscp_qos (factory));
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break;
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case PROP_ENABLE_RTCP:
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g_value_set_boolean (value,
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gst_rtsp_media_factory_is_enable_rtcp (factory));
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
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}
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@ -442,6 +463,10 @@ gst_rtsp_media_factory_set_property (GObject * object, guint propid,
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case PROP_DSCP_QOS:
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gst_rtsp_media_factory_set_dscp_qos (factory, g_value_get_int (value));
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break;
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case PROP_ENABLE_RTCP:
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gst_rtsp_media_factory_set_enable_rtcp (factory,
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g_value_get_boolean (value));
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
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}
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@ -1686,6 +1711,57 @@ gst_rtsp_media_factory_is_bind_mcast_address (GstRTSPMediaFactory * factory)
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return result;
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}
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/**
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* gst_rtsp_media_factory_set_enable_rtcp:
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* @factory: a #GstRTSPMediaFactory
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* @enable: the new value
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*
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* Decide whether the created media should send and receive RTCP
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*
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* Since: 1.20
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*/
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void
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gst_rtsp_media_factory_set_enable_rtcp (GstRTSPMediaFactory * factory,
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gboolean enable)
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{
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GstRTSPMediaFactoryPrivate *priv;
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g_return_if_fail (GST_IS_RTSP_MEDIA_FACTORY (factory));
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priv = factory->priv;
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GST_RTSP_MEDIA_FACTORY_LOCK (factory);
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priv->enable_rtcp = enable;
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GST_RTSP_MEDIA_FACTORY_UNLOCK (factory);
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}
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/**
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* gst_rtsp_media_factory_is_enable_rtcp:
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* @factory: a #GstRTSPMediaFactory
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*
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* Check if created media will send and receive RTCP
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*
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* Returns: %TRUE if created media will send and receive RTCP
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*
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* Since: 1.20
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*/
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gboolean
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gst_rtsp_media_factory_is_enable_rtcp (GstRTSPMediaFactory * factory)
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{
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GstRTSPMediaFactoryPrivate *priv;
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gboolean result;
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g_return_val_if_fail (GST_IS_RTSP_MEDIA_FACTORY (factory), FALSE);
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priv = factory->priv;
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GST_RTSP_MEDIA_FACTORY_LOCK (factory);
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result = priv->enable_rtcp;
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GST_RTSP_MEDIA_FACTORY_UNLOCK (factory);
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return result;
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}
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static gchar *
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default_gen_key (GstRTSPMediaFactory * factory, const GstRTSPUrl * url)
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{
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@ -1757,6 +1833,7 @@ default_construct (GstRTSPMediaFactory * factory, const GstRTSPUrl * url)
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GstElement *element, *pipeline;
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GstRTSPMediaFactoryClass *klass;
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GType media_gtype;
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gboolean enable_rtcp;
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klass = GST_RTSP_MEDIA_FACTORY_GET_CLASS (factory);
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@ -1769,6 +1846,7 @@ default_construct (GstRTSPMediaFactory * factory, const GstRTSPUrl * url)
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GST_RTSP_MEDIA_FACTORY_LOCK (factory);
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media_gtype = factory->priv->media_gtype;
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enable_rtcp = factory->priv->enable_rtcp;
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GST_RTSP_MEDIA_FACTORY_UNLOCK (factory);
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/* create a new empty media */
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@ -1776,6 +1854,9 @@ default_construct (GstRTSPMediaFactory * factory, const GstRTSPUrl * url)
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g_object_new (media_gtype, "element", element, "transport-mode",
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factory->priv->transport_mode, NULL);
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/* We need to call this prior to collecting streams */
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gst_rtsp_media_set_enable_rtcp (media, enable_rtcp);
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gst_rtsp_media_collect_streams (media);
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pipeline = klass->create_pipeline (factory, media);
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@ -258,6 +258,13 @@ void gst_rtsp_media_factory_set_dscp_qos (GstRTSPMediaFactory *
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GST_RTSP_SERVER_API
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gint gst_rtsp_media_factory_get_dscp_qos (GstRTSPMediaFactory * factory);
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GST_RTSP_SERVER_API
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void gst_rtsp_media_factory_set_enable_rtcp (GstRTSPMediaFactory * factory,
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gboolean enable);
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GST_RTSP_SERVER_API
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gboolean gst_rtsp_media_factory_is_enable_rtcp (GstRTSPMediaFactory * factory);
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/* creating the media from the factory and a url */
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GST_RTSP_SERVER_API
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@ -112,6 +112,7 @@ struct _GstRTSPMediaPrivate
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gchar *multicast_iface;
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guint max_mcast_ttl;
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gboolean bind_mcast_address;
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gboolean enable_rtcp;
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gboolean blocked;
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GstRTSPTransportMode transport_mode;
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gboolean stop_on_disconnect;
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@ -180,6 +181,7 @@ struct _GstRTSPMediaPrivate
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#define DEFAULT_MAX_MCAST_TTL 255
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#define DEFAULT_BIND_MCAST_ADDRESS FALSE
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#define DEFAULT_DO_RATE_CONTROL TRUE
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#define DEFAULT_ENABLE_RTCP TRUE
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#define DEFAULT_DO_RETRANSMISSION FALSE
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@ -492,6 +494,7 @@ gst_rtsp_media_init (GstRTSPMedia * media)
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priv->do_retransmission = DEFAULT_DO_RETRANSMISSION;
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priv->max_mcast_ttl = DEFAULT_MAX_MCAST_TTL;
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priv->bind_mcast_address = DEFAULT_BIND_MCAST_ADDRESS;
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priv->enable_rtcp = DEFAULT_ENABLE_RTCP;
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priv->do_rate_control = DEFAULT_DO_RATE_CONTROL;
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priv->dscp_qos = DEFAULT_DSCP_QOS;
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priv->expected_async_done = FALSE;
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@ -2104,6 +2107,20 @@ gst_rtsp_media_is_bind_mcast_address (GstRTSPMedia * media)
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return result;
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}
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void
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gst_rtsp_media_set_enable_rtcp (GstRTSPMedia * media, gboolean enable)
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{
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GstRTSPMediaPrivate *priv;
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g_return_if_fail (GST_IS_RTSP_MEDIA (media));
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priv = media->priv;
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g_mutex_lock (&priv->lock);
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priv->enable_rtcp = enable;
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g_mutex_unlock (&priv->lock);
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}
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static GList *
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_find_payload_types (GstRTSPMedia * media)
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{
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@ -2425,6 +2442,7 @@ gst_rtsp_media_create_stream (GstRTSPMedia * media, GstElement * payloader,
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gst_rtsp_stream_set_multicast_iface (stream, priv->multicast_iface);
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gst_rtsp_stream_set_max_mcast_ttl (stream, priv->max_mcast_ttl);
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gst_rtsp_stream_set_bind_mcast_address (stream, priv->bind_mcast_address);
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gst_rtsp_stream_set_enable_rtcp (stream, priv->enable_rtcp);
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gst_rtsp_stream_set_profiles (stream, priv->profiles);
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gst_rtsp_stream_set_protocols (stream, priv->protocols);
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gst_rtsp_stream_set_retransmission_time (stream, priv->rtx_time);
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@ -58,6 +58,9 @@ gboolean gst_rtsp_stream_transport_check_back_pressure (GstRTSPS
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gboolean gst_rtsp_stream_is_tcp_receiver (GstRTSPStream * stream);
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void gst_rtsp_media_set_enable_rtcp (GstRTSPMedia *media, gboolean enable);
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void gst_rtsp_stream_set_enable_rtcp (GstRTSPStream *stream, gboolean enable);
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G_END_DECLS
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#endif /* __GST_RTSP_SERVER_INTERNAL_H__ */
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@ -220,6 +220,9 @@ struct _GstRTSPStreamPrivate
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guint32 blocked_seqnum;
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guint32 blocked_rtptime;
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GstClockTime blocked_running_time;
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/* Whether we should send and receive RTCP */
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gboolean enable_rtcp;
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};
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#define DEFAULT_CONTROL NULL
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@ -229,6 +232,7 @@ struct _GstRTSPStreamPrivate
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#define DEFAULT_MAX_MCAST_TTL 255
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#define DEFAULT_BIND_MCAST_ADDRESS FALSE
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#define DEFAULT_DO_RATE_CONTROL TRUE
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#define DEFAULT_ENABLE_RTCP TRUE
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enum
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{
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@ -328,6 +332,7 @@ gst_rtsp_stream_init (GstRTSPStream * stream)
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priv->max_mcast_ttl = DEFAULT_MAX_MCAST_TTL;
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priv->bind_mcast_address = DEFAULT_BIND_MCAST_ADDRESS;
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priv->do_rate_control = DEFAULT_DO_RATE_CONTROL;
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priv->enable_rtcp = DEFAULT_ENABLE_RTCP;
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g_mutex_init (&priv->lock);
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@ -1422,6 +1427,7 @@ alloc_ports_one_family (GstRTSPStream * stream, GSocketFamily family,
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/* Start with random port */
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tmp_rtp = 0;
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tmp_rtcp = 0;
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if (use_transport_settings) {
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if (!multicast)
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@ -1453,14 +1459,16 @@ alloc_ports_one_family (GstRTSPStream * stream, GSocketFamily family,
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}
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}
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rtcp_socket = g_socket_new (family, G_SOCKET_TYPE_DATAGRAM,
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G_SOCKET_PROTOCOL_UDP, NULL);
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if (!rtcp_socket)
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goto no_udp_protocol;
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g_socket_set_multicast_loopback (rtcp_socket, FALSE);
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if (priv->enable_rtcp) {
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rtcp_socket = g_socket_new (family, G_SOCKET_TYPE_DATAGRAM,
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G_SOCKET_PROTOCOL_UDP, NULL);
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if (!rtcp_socket)
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goto no_udp_protocol;
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g_socket_set_multicast_loopback (rtcp_socket, FALSE);
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}
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/* try to allocate 2 UDP ports, the RTP port should be an even
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* number and the RTCP port should be the next (uneven) port */
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/* try to allocate UDP ports, the RTP port should be an even
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* number and the RTCP port (if enabled) should be the next (uneven) port */
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again:
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if (rtp_socket == NULL) {
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@ -1496,7 +1504,8 @@ again:
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if (*server_addr_out)
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addr = *server_addr_out;
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else
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addr = gst_rtsp_address_pool_acquire_address (pool, flags, 2);
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addr = gst_rtsp_address_pool_acquire_address (pool, flags,
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priv->enable_rtcp ? 2 : 1);
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if (addr == NULL)
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goto no_address;
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@ -1556,18 +1565,20 @@ again:
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g_object_unref (rtp_sockaddr);
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/* set port */
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tmp_rtcp = tmp_rtp + 1;
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if (priv->enable_rtcp) {
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tmp_rtcp = tmp_rtp + 1;
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rtcp_sockaddr = g_inet_socket_address_new (inetaddr, tmp_rtcp);
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if (!g_socket_bind (rtcp_socket, rtcp_sockaddr, FALSE, NULL)) {
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GST_DEBUG_OBJECT (stream, "rctp bind() failed, will try again");
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rtcp_sockaddr = g_inet_socket_address_new (inetaddr, tmp_rtcp);
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if (!g_socket_bind (rtcp_socket, rtcp_sockaddr, FALSE, NULL)) {
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GST_DEBUG_OBJECT (stream, "rctp bind() failed, will try again");
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g_object_unref (rtcp_sockaddr);
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g_clear_object (&rtp_socket);
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if (transport_settings_defined)
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goto transport_settings_error;
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goto again;
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}
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g_object_unref (rtcp_sockaddr);
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g_clear_object (&rtp_socket);
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if (transport_settings_defined)
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goto transport_settings_error;
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goto again;
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}
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g_object_unref (rtcp_sockaddr);
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if (!addr) {
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addr = g_slice_new0 (GstRTSPAddress);
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@ -1585,15 +1596,21 @@ again:
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if (multicast && (ct->ttl > 0) && (ct->ttl <= priv->max_mcast_ttl)) {
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GST_DEBUG ("setting mcast ttl to %d", ct->ttl);
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g_socket_set_multicast_ttl (rtp_socket, ct->ttl);
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g_socket_set_multicast_ttl (rtcp_socket, ct->ttl);
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if (rtcp_socket)
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g_socket_set_multicast_ttl (rtcp_socket, ct->ttl);
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}
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socket_out[0] = rtp_socket;
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socket_out[1] = rtcp_socket;
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*server_addr_out = addr;
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GST_DEBUG_OBJECT (stream, "allocated address: %s and ports: %d, %d",
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addr->address, tmp_rtp, tmp_rtcp);
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if (priv->enable_rtcp) {
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GST_DEBUG_OBJECT (stream, "allocated address: %s and ports: %d, %d",
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addr->address, tmp_rtp, tmp_rtcp);
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} else {
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GST_DEBUG_OBJECT (stream, "allocated address: %s and port: %d",
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addr->address, tmp_rtp);
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}
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g_list_free_full (rejected_addresses, (GDestroyNotify) gst_rtsp_address_free);
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@ -1922,14 +1939,18 @@ gst_rtsp_stream_get_server_port (GstRTSPStream * stream,
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if (family == G_SOCKET_FAMILY_IPV4) {
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if (server_port && priv->server_addr_v4) {
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server_port->min = priv->server_addr_v4->port;
|
||||
server_port->max =
|
||||
priv->server_addr_v4->port + priv->server_addr_v4->n_ports - 1;
|
||||
if (priv->enable_rtcp) {
|
||||
server_port->max =
|
||||
priv->server_addr_v4->port + priv->server_addr_v4->n_ports - 1;
|
||||
}
|
||||
}
|
||||
} else {
|
||||
if (server_port && priv->server_addr_v6) {
|
||||
server_port->min = priv->server_addr_v6->port;
|
||||
server_port->max =
|
||||
priv->server_addr_v6->port + priv->server_addr_v6->n_ports - 1;
|
||||
if (priv->enable_rtcp) {
|
||||
server_port->max =
|
||||
priv->server_addr_v6->port + priv->server_addr_v6->n_ports - 1;
|
||||
}
|
||||
}
|
||||
}
|
||||
g_mutex_unlock (&priv->lock);
|
||||
|
|
@ -2262,6 +2283,16 @@ gst_rtsp_stream_is_bind_mcast_address (GstRTSPStream * stream)
|
|||
return result;
|
||||
}
|
||||
|
||||
void
|
||||
gst_rtsp_stream_set_enable_rtcp (GstRTSPStream * stream, gboolean enable)
|
||||
{
|
||||
g_return_if_fail (GST_IS_RTSP_STREAM (stream));
|
||||
|
||||
g_mutex_lock (&stream->priv->lock);
|
||||
stream->priv->enable_rtcp = enable;
|
||||
g_mutex_unlock (&stream->priv->lock);
|
||||
}
|
||||
|
||||
/* executed from streaming thread */
|
||||
static void
|
||||
caps_notify (GstPad * pad, GParamSpec * unused, GstRTSPStream * stream)
|
||||
|
|
@ -3518,10 +3549,10 @@ create_sender_part (GstRTSPStream * stream, const GstRTSPTransport * transport)
|
|||
g_object_set (priv->payloader, "onvif-no-rate-control",
|
||||
!priv->do_rate_control, NULL);
|
||||
|
||||
for (i = 0; i < 2; i++) {
|
||||
for (i = 0; i < (priv->enable_rtcp ? 2 : 1); i++) {
|
||||
gboolean link_tee = FALSE;
|
||||
/* For the sender we create this bit of pipeline for both
|
||||
* RTP and RTCP.
|
||||
* RTP and RTCP (when enabled).
|
||||
* Initially there will be only one active transport for
|
||||
* the stream, so the pipeline will look like this:
|
||||
*
|
||||
|
|
@ -3674,9 +3705,9 @@ create_receiver_part (GstRTSPStream * stream, const GstRTSPTransport *
|
|||
"RTP caps: %" GST_PTR_FORMAT " RTCP caps: %" GST_PTR_FORMAT, rtp_caps,
|
||||
rtcp_caps);
|
||||
|
||||
for (i = 0; i < 2; i++) {
|
||||
for (i = 0; i < (priv->enable_rtcp ? 2 : 1); i++) {
|
||||
/* For the receiver we create this bit of pipeline for both
|
||||
* RTP and RTCP. We receive RTP/RTCP on appsrc and udpsrc
|
||||
* RTP and RTCP (when enabled). We receive RTP/RTCP on appsrc and udpsrc
|
||||
* and it is all funneled into the rtpbin receive pad.
|
||||
*
|
||||
*
|
||||
|
|
@ -3938,12 +3969,15 @@ gst_rtsp_stream_join_bin (GstRTSPStream * stream, GstBin * bin,
|
|||
g_free (name);
|
||||
}
|
||||
|
||||
name = g_strdup_printf ("send_rtcp_src_%u", idx);
|
||||
priv->send_src[1] = gst_element_get_request_pad (rtpbin, name);
|
||||
g_free (name);
|
||||
name = g_strdup_printf ("recv_rtcp_sink_%u", idx);
|
||||
priv->recv_sink[1] = gst_element_get_request_pad (rtpbin, name);
|
||||
g_free (name);
|
||||
if (priv->enable_rtcp) {
|
||||
name = g_strdup_printf ("send_rtcp_src_%u", idx);
|
||||
priv->send_src[1] = gst_element_get_request_pad (rtpbin, name);
|
||||
g_free (name);
|
||||
|
||||
name = g_strdup_printf ("recv_rtcp_sink_%u", idx);
|
||||
priv->recv_sink[1] = gst_element_get_request_pad (rtpbin, name);
|
||||
g_free (name);
|
||||
}
|
||||
|
||||
/* get the session */
|
||||
g_signal_emit_by_name (rtpbin, "get-internal-session", idx, &priv->session);
|
||||
|
|
@ -4085,7 +4119,7 @@ gst_rtsp_stream_leave_bin (GstRTSPStream * stream, GstBin * bin,
|
|||
priv->recv_rtp_src = NULL;
|
||||
}
|
||||
|
||||
for (i = 0; i < 2; i++) {
|
||||
for (i = 0; i < (priv->enable_rtcp ? 2 : 1); i++) {
|
||||
clear_element (bin, &priv->udpsrc_v4[i]);
|
||||
clear_element (bin, &priv->udpsrc_v6[i]);
|
||||
clear_element (bin, &priv->udpqueue[i]);
|
||||
|
|
@ -4115,9 +4149,11 @@ gst_rtsp_stream_leave_bin (GstRTSPStream * stream, GstBin * bin,
|
|||
priv->send_src[0] = NULL;
|
||||
}
|
||||
|
||||
gst_element_release_request_pad (rtpbin, priv->send_src[1]);
|
||||
gst_object_unref (priv->send_src[1]);
|
||||
priv->send_src[1] = NULL;
|
||||
if (priv->enable_rtcp) {
|
||||
gst_element_release_request_pad (rtpbin, priv->send_src[1]);
|
||||
gst_object_unref (priv->send_src[1]);
|
||||
priv->send_src[1] = NULL;
|
||||
}
|
||||
|
||||
g_object_unref (priv->session);
|
||||
priv->session = NULL;
|
||||
|
|
@ -6206,8 +6242,8 @@ gst_rtsp_stream_set_rate_control (GstRTSPStream * stream, gboolean enabled)
|
|||
if (stream->priv->appsink[0])
|
||||
g_object_set (stream->priv->appsink[0], "sync", enabled, NULL);
|
||||
if (stream->priv->payloader
|
||||
&& g_object_class_find_property (G_OBJECT_GET_CLASS (stream->
|
||||
priv->payloader), "onvif-no-rate-control"))
|
||||
&& g_object_class_find_property (G_OBJECT_GET_CLASS (stream->priv->
|
||||
payloader), "onvif-no-rate-control"))
|
||||
g_object_set (stream->priv->payloader, "onvif-no-rate-control", !enabled,
|
||||
NULL);
|
||||
if (stream->priv->session) {
|
||||
|
|
|
|||
|
|
@ -206,7 +206,7 @@ create_connection (GstRTSPConnection ** conn)
|
|||
}
|
||||
|
||||
static GstRTSPClient *
|
||||
setup_client (const gchar * launch_line)
|
||||
setup_client (const gchar * launch_line, gboolean enable_rtcp)
|
||||
{
|
||||
GstRTSPClient *client;
|
||||
GstRTSPSessionPool *session_pool;
|
||||
|
|
@ -227,6 +227,8 @@ setup_client (const gchar * launch_line)
|
|||
else
|
||||
gst_rtsp_media_factory_set_launch (factory, launch_line);
|
||||
|
||||
gst_rtsp_media_factory_set_enable_rtcp (factory, enable_rtcp);
|
||||
|
||||
gst_rtsp_mount_points_add_factory (mount_points, "/test", factory);
|
||||
gst_rtsp_client_set_mount_points (client, mount_points);
|
||||
|
||||
|
|
@ -515,7 +517,7 @@ GST_START_TEST (test_describe)
|
|||
g_object_unref (client);
|
||||
|
||||
/* simple DESCRIBE for an existing url */
|
||||
client = setup_client (NULL);
|
||||
client = setup_client (NULL, TRUE);
|
||||
fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_DESCRIBE,
|
||||
"rtsp://localhost/test") == GST_RTSP_OK);
|
||||
str = g_strdup_printf ("%d", cseq);
|
||||
|
|
@ -688,7 +690,7 @@ GST_START_TEST (test_setup_tcp)
|
|||
GstRTSPMessage request = { 0, };
|
||||
gchar *str;
|
||||
|
||||
client = setup_client (NULL);
|
||||
client = setup_client (NULL, TRUE);
|
||||
create_connection (&conn);
|
||||
fail_unless (gst_rtsp_client_set_connection (client, conn));
|
||||
|
||||
|
|
@ -714,6 +716,40 @@ GST_START_TEST (test_setup_tcp)
|
|||
|
||||
GST_END_TEST;
|
||||
|
||||
GST_START_TEST (test_setup_no_rtcp)
|
||||
{
|
||||
GstRTSPClient *client;
|
||||
GstRTSPConnection *conn;
|
||||
GstRTSPMessage request = { 0, };
|
||||
gchar *str;
|
||||
|
||||
client = setup_client (NULL, FALSE);
|
||||
create_connection (&conn);
|
||||
fail_unless (gst_rtsp_client_set_connection (client, conn));
|
||||
|
||||
fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_SETUP,
|
||||
"rtsp://localhost/test/stream=0") == GST_RTSP_OK);
|
||||
str = g_strdup_printf ("%d", cseq);
|
||||
gst_rtsp_message_add_header (&request, GST_RTSP_HDR_CSEQ, str);
|
||||
g_free (str);
|
||||
gst_rtsp_message_add_header (&request, GST_RTSP_HDR_TRANSPORT,
|
||||
"RTP/AVP;unicast;client_port=5000-5001");
|
||||
|
||||
gst_rtsp_client_set_send_func (client, test_setup_response_200, NULL, NULL);
|
||||
/* We want to verify that server_port holds a single number, not a range */
|
||||
expected_transport =
|
||||
"RTP/AVP;unicast;client_port=5000-5001;server_port=[0-9]+;ssrc=.*;mode=\"PLAY\"";
|
||||
fail_unless (gst_rtsp_client_handle_message (client,
|
||||
&request) == GST_RTSP_OK);
|
||||
|
||||
gst_rtsp_message_unset (&request);
|
||||
|
||||
send_teardown (client);
|
||||
teardown_client (client);
|
||||
}
|
||||
|
||||
GST_END_TEST;
|
||||
|
||||
GST_START_TEST (test_setup_tcp_two_streams_same_channels)
|
||||
{
|
||||
GstRTSPClient *client;
|
||||
|
|
@ -721,7 +757,7 @@ GST_START_TEST (test_setup_tcp_two_streams_same_channels)
|
|||
GstRTSPMessage request = { 0, };
|
||||
gchar *str;
|
||||
|
||||
client = setup_client (NULL);
|
||||
client = setup_client (NULL, TRUE);
|
||||
create_connection (&conn);
|
||||
fail_unless (gst_rtsp_client_set_connection (client, conn));
|
||||
|
||||
|
|
@ -1044,7 +1080,7 @@ test_client_sdp (const gchar * launch_line, guint * bandwidth_val)
|
|||
gchar *str;
|
||||
|
||||
/* simple DESCRIBE for an existing url */
|
||||
client = setup_client (launch_line);
|
||||
client = setup_client (launch_line, TRUE);
|
||||
fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_DESCRIBE,
|
||||
"rtsp://localhost/test") == GST_RTSP_OK);
|
||||
str = g_strdup_printf ("%d", cseq);
|
||||
|
|
@ -2044,6 +2080,7 @@ rtspclient_suite (void)
|
|||
tcase_add_test (tc, test_options);
|
||||
tcase_add_test (tc, test_describe);
|
||||
tcase_add_test (tc, test_setup_tcp);
|
||||
tcase_add_test (tc, test_setup_no_rtcp);
|
||||
tcase_add_test (tc, test_setup_tcp_two_streams_same_channels);
|
||||
tcase_add_test (tc, test_client_multicast_transport_404);
|
||||
tcase_add_test (tc, test_client_multicast_transport);
|
||||
|
|
|
|||
Loading…
Reference in a new issue