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Diffstat (limited to 'include/gst/rtsp-server/rtsp-client.h')
| -rw-r--r-- | include/gst/rtsp-server/rtsp-client.h | 294 |
1 files changed, 294 insertions, 0 deletions
diff --git a/include/gst/rtsp-server/rtsp-client.h b/include/gst/rtsp-server/rtsp-client.h new file mode 100644 index 0000000000..604a042399 --- /dev/null +++ b/include/gst/rtsp-server/rtsp-client.h @@ -0,0 +1,294 @@ +/* GStreamer + * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com> + * + * This library is free software; you can redistribute it and/or + * modify it under the terms of the GNU Library General Public + * License as published by the Free Software Foundation; either + * version 2 of the License, or (at your option) any later version. + * + * This library is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Library General Public License for more details. + * + * You should have received a copy of the GNU Library General Public + * License along with this library; if not, write to the + * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor, + * Boston, MA 02110-1301, USA. + */ + +#include <gst/gst.h> +#include <gst/rtsp/gstrtspconnection.h> + +#ifndef __GST_RTSP_CLIENT_H__ +#define __GST_RTSP_CLIENT_H__ + +G_BEGIN_DECLS + +typedef struct _GstRTSPClient GstRTSPClient; +typedef struct _GstRTSPClientClass GstRTSPClientClass; +typedef struct _GstRTSPClientPrivate GstRTSPClientPrivate; + +#include "rtsp-server-prelude.h" +#include "rtsp-context.h" +#include "rtsp-mount-points.h" +#include "rtsp-sdp.h" +#include "rtsp-auth.h" + +#define GST_TYPE_RTSP_CLIENT (gst_rtsp_client_get_type ()) +#define GST_IS_RTSP_CLIENT(obj) (G_TYPE_CHECK_INSTANCE_TYPE ((obj), GST_TYPE_RTSP_CLIENT)) +#define GST_IS_RTSP_CLIENT_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE ((klass), GST_TYPE_RTSP_CLIENT)) +#define GST_RTSP_CLIENT_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_RTSP_CLIENT, GstRTSPClientClass)) +#define GST_RTSP_CLIENT(obj) (G_TYPE_CHECK_INSTANCE_CAST ((obj), GST_TYPE_RTSP_CLIENT, GstRTSPClient)) +#define GST_RTSP_CLIENT_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST ((klass), GST_TYPE_RTSP_CLIENT, GstRTSPClientClass)) +#define GST_RTSP_CLIENT_CAST(obj) ((GstRTSPClient*)(obj)) +#define GST_RTSP_CLIENT_CLASS_CAST(klass) ((GstRTSPClientClass*)(klass)) + +/** + * GstRTSPClientSendFunc: + * @client: a #GstRTSPClient + * @message: a #GstRTSPMessage + * @close: close the connection + * @user_data: user data when registering the callback + * + * This callback is called when @client wants to send @message. When @close is + * %TRUE, the connection should be closed when the message has been sent. + * + * Returns: %TRUE on success. + */ +typedef gboolean (*GstRTSPClientSendFunc) (GstRTSPClient *client, + GstRTSPMessage *message, + gboolean close, + gpointer user_data); + +/** + * GstRTSPClientSendMessagesFunc: + * @client: a #GstRTSPClient + * @messages: #GstRTSPMessage + * @n_messages: number of messages + * @close: close the connection + * @user_data: user data when registering the callback + * + * This callback is called when @client wants to send @messages. When @close is + * %TRUE, the connection should be closed when the message has been sent. + * + * Returns: %TRUE on success. + * + * Since: 1.16 + */ +typedef gboolean (*GstRTSPClientSendMessagesFunc) (GstRTSPClient *client, + GstRTSPMessage *messages, + guint n_messages, + gboolean close, + gpointer user_data); + +/** + * GstRTSPClient: + * + * The client object represents the connection and its state with a client. + */ +struct _GstRTSPClient { + GObject parent; + + /*< private >*/ + GstRTSPClientPrivate *priv; + gpointer _gst_reserved[GST_PADDING]; +}; + +/** + * GstRTSPClientClass: + * @create_sdp: called when the SDP needs to be created for media. + * @configure_client_media: called when the stream in media needs to be configured. + * The default implementation will configure the blocksize on the payloader when + * spcified in the request headers. + * @configure_client_transport: called when the client transport needs to be + * configured. + * @params_set: set parameters. This function should also initialize the + * RTSP response(ctx->response) via a call to gst_rtsp_message_init_response() + * @params_get: get parameters. This function should also initialize the + * RTSP response(ctx->response) via a call to gst_rtsp_message_init_response() + * @make_path_from_uri: called to create path from uri. + * @adjust_play_mode: called to give the application the possibility to adjust + * the range, seek flags, rate and rate-control. Since 1.18 + * @adjust_play_response: called to give the implementation the possibility to + * adjust the response to a play request, for example if extra headers were + * parsed when #GstRTSPClientClass.adjust_play_mode was called. Since 1.18 + * @tunnel_http_response: called when a response to the GET request is about to + * be sent for a tunneled connection. The response can be modified. Since: 1.4 + * + * The client class structure. + */ +struct _GstRTSPClientClass { + GObjectClass parent_class; + + GstSDPMessage * (*create_sdp) (GstRTSPClient *client, GstRTSPMedia *media); + gboolean (*configure_client_media) (GstRTSPClient * client, + GstRTSPMedia * media, GstRTSPStream * stream, + GstRTSPContext * ctx); + gboolean (*configure_client_transport) (GstRTSPClient * client, + GstRTSPContext * ctx, + GstRTSPTransport * ct); + GstRTSPResult (*params_set) (GstRTSPClient *client, GstRTSPContext *ctx); + GstRTSPResult (*params_get) (GstRTSPClient *client, GstRTSPContext *ctx); + gchar * (*make_path_from_uri) (GstRTSPClient *client, const GstRTSPUrl *uri); + GstRTSPStatusCode (*adjust_play_mode) (GstRTSPClient * client, + GstRTSPContext * context, + GstRTSPTimeRange ** range, + GstSeekFlags * flags, + gdouble * rate, + GstClockTime * trickmode_interval, + gboolean * enable_rate_control); + GstRTSPStatusCode (*adjust_play_response) (GstRTSPClient * client, + GstRTSPContext * context); + + /* signals */ + void (*closed) (GstRTSPClient *client); + void (*new_session) (GstRTSPClient *client, GstRTSPSession *session); + void (*options_request) (GstRTSPClient *client, GstRTSPContext *ctx); + void (*describe_request) (GstRTSPClient *client, GstRTSPContext *ctx); + void (*setup_request) (GstRTSPClient *client, GstRTSPContext *ctx); + void (*play_request) (GstRTSPClient *client, GstRTSPContext *ctx); + void (*pause_request) (GstRTSPClient *client, GstRTSPContext *ctx); + void (*teardown_request) (GstRTSPClient *client, GstRTSPContext *ctx); + void (*set_parameter_request) (GstRTSPClient *client, GstRTSPContext *ctx); + void (*get_parameter_request) (GstRTSPClient *client, GstRTSPContext *ctx); + void (*handle_response) (GstRTSPClient *client, GstRTSPContext *ctx); + + void (*tunnel_http_response) (GstRTSPClient * client, GstRTSPMessage * request, + GstRTSPMessage * response); + void (*send_message) (GstRTSPClient * client, GstRTSPContext *ctx, + GstRTSPMessage * response); + + gboolean (*handle_sdp) (GstRTSPClient *client, GstRTSPContext *ctx, GstRTSPMedia *media, GstSDPMessage *sdp); + + void (*announce_request) (GstRTSPClient *client, GstRTSPContext *ctx); + void (*record_request) (GstRTSPClient *client, GstRTSPContext *ctx); + gchar* (*check_requirements) (GstRTSPClient *client, GstRTSPContext *ctx, gchar ** arr); + + GstRTSPStatusCode (*pre_options_request) (GstRTSPClient *client, GstRTSPContext *ctx); + GstRTSPStatusCode (*pre_describe_request) (GstRTSPClient *client, GstRTSPContext *ctx); + GstRTSPStatusCode (*pre_setup_request) (GstRTSPClient *client, GstRTSPContext *ctx); + GstRTSPStatusCode (*pre_play_request) (GstRTSPClient *client, GstRTSPContext *ctx); + GstRTSPStatusCode (*pre_pause_request) (GstRTSPClient *client, GstRTSPContext *ctx); + GstRTSPStatusCode (*pre_teardown_request) (GstRTSPClient *client, GstRTSPContext *ctx); + GstRTSPStatusCode (*pre_set_parameter_request) (GstRTSPClient *client, GstRTSPContext *ctx); + GstRTSPStatusCode (*pre_get_parameter_request) (GstRTSPClient *client, GstRTSPContext *ctx); + GstRTSPStatusCode (*pre_announce_request) (GstRTSPClient *client, GstRTSPContext *ctx); + GstRTSPStatusCode (*pre_record_request) (GstRTSPClient *client, GstRTSPContext *ctx); + + /*< private >*/ + gpointer _gst_reserved[GST_PADDING_LARGE-18]; +}; + +GST_RTSP_SERVER_API +GType gst_rtsp_client_get_type (void); + +GST_RTSP_SERVER_API +GstRTSPClient * gst_rtsp_client_new (void); + +GST_RTSP_SERVER_API +void gst_rtsp_client_set_session_pool (GstRTSPClient *client, + GstRTSPSessionPool *pool); + +GST_RTSP_SERVER_API +GstRTSPSessionPool * gst_rtsp_client_get_session_pool (GstRTSPClient *client); + +GST_RTSP_SERVER_API +void gst_rtsp_client_set_mount_points (GstRTSPClient *client, + GstRTSPMountPoints *mounts); + +GST_RTSP_SERVER_API +GstRTSPMountPoints * gst_rtsp_client_get_mount_points (GstRTSPClient *client); + +GST_RTSP_SERVER_API +void gst_rtsp_client_set_content_length_limit (GstRTSPClient *client, guint limit); + +GST_RTSP_SERVER_API +guint gst_rtsp_client_get_content_length_limit (GstRTSPClient *client); + +GST_RTSP_SERVER_API +void gst_rtsp_client_set_auth (GstRTSPClient *client, GstRTSPAuth *auth); + +GST_RTSP_SERVER_API +GstRTSPAuth * gst_rtsp_client_get_auth (GstRTSPClient *client); + +GST_RTSP_SERVER_API +void gst_rtsp_client_set_thread_pool (GstRTSPClient *client, GstRTSPThreadPool *pool); + +GST_RTSP_SERVER_API +GstRTSPThreadPool * gst_rtsp_client_get_thread_pool (GstRTSPClient *client); + +GST_RTSP_SERVER_API +gboolean gst_rtsp_client_set_connection (GstRTSPClient *client, GstRTSPConnection *conn); + +GST_RTSP_SERVER_API +GstRTSPConnection * gst_rtsp_client_get_connection (GstRTSPClient *client); + +GST_RTSP_SERVER_API +guint gst_rtsp_client_attach (GstRTSPClient *client, + GMainContext *context); + +GST_RTSP_SERVER_API +void gst_rtsp_client_close (GstRTSPClient * client); + +GST_RTSP_SERVER_API +void gst_rtsp_client_set_send_func (GstRTSPClient *client, + GstRTSPClientSendFunc func, + gpointer user_data, + GDestroyNotify notify); + +GST_RTSP_SERVER_API +void gst_rtsp_client_set_send_messages_func (GstRTSPClient *client, + GstRTSPClientSendMessagesFunc func, + gpointer user_data, + GDestroyNotify notify); + +GST_RTSP_SERVER_API +GstRTSPResult gst_rtsp_client_handle_message (GstRTSPClient *client, + GstRTSPMessage *message); + +GST_RTSP_SERVER_API +GstRTSPResult gst_rtsp_client_send_message (GstRTSPClient * client, + GstRTSPSession *session, + GstRTSPMessage *message); +/** + * GstRTSPClientSessionFilterFunc: + * @client: a #GstRTSPClient object + * @sess: a #GstRTSPSession in @client + * @user_data: user data that has been given to gst_rtsp_client_session_filter() + * + * This function will be called by the gst_rtsp_client_session_filter(). An + * implementation should return a value of #GstRTSPFilterResult. + * + * When this function returns #GST_RTSP_FILTER_REMOVE, @sess will be removed + * from @client. + * + * A return value of #GST_RTSP_FILTER_KEEP will leave @sess untouched in + * @client. + * + * A value of #GST_RTSP_FILTER_REF will add @sess to the result #GList of + * gst_rtsp_client_session_filter(). + * + * Returns: a #GstRTSPFilterResult. + */ +typedef GstRTSPFilterResult (*GstRTSPClientSessionFilterFunc) (GstRTSPClient *client, + GstRTSPSession *sess, + gpointer user_data); + +GST_RTSP_SERVER_API +GList * gst_rtsp_client_session_filter (GstRTSPClient *client, + GstRTSPClientSessionFilterFunc func, + gpointer user_data); + +GST_RTSP_SERVER_API +GstRTSPStreamTransport * gst_rtsp_client_get_stream_transport (GstRTSPClient *client, + guint8 channel); + + +#ifdef G_DEFINE_AUTOPTR_CLEANUP_FUNC +G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstRTSPClient, gst_object_unref) +#endif + +G_END_DECLS + +#endif /* __GST_RTSP_CLIENT_H__ */ |