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ae8a5e110c
Confuses the documentation builder, since it's documented twice it complains about a missing "Since:" marker whereas it's present in the documentation comment further down Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3180>
310 lines
14 KiB
C
310 lines
14 KiB
C
/* GStreamer
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* Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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#include <gst/gst.h>
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#include <gst/rtsp/gstrtspconnection.h>
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#ifndef __GST_RTSP_CLIENT_H__
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#define __GST_RTSP_CLIENT_H__
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G_BEGIN_DECLS
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typedef struct _GstRTSPClient GstRTSPClient;
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typedef struct _GstRTSPClientClass GstRTSPClientClass;
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typedef struct _GstRTSPClientPrivate GstRTSPClientPrivate;
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#include "rtsp-server-prelude.h"
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#include "rtsp-context.h"
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#include "rtsp-mount-points.h"
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#include "rtsp-sdp.h"
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#include "rtsp-auth.h"
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#define GST_TYPE_RTSP_CLIENT (gst_rtsp_client_get_type ())
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#define GST_IS_RTSP_CLIENT(obj) (G_TYPE_CHECK_INSTANCE_TYPE ((obj), GST_TYPE_RTSP_CLIENT))
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#define GST_IS_RTSP_CLIENT_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE ((klass), GST_TYPE_RTSP_CLIENT))
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#define GST_RTSP_CLIENT_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_RTSP_CLIENT, GstRTSPClientClass))
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#define GST_RTSP_CLIENT(obj) (G_TYPE_CHECK_INSTANCE_CAST ((obj), GST_TYPE_RTSP_CLIENT, GstRTSPClient))
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#define GST_RTSP_CLIENT_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST ((klass), GST_TYPE_RTSP_CLIENT, GstRTSPClientClass))
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#define GST_RTSP_CLIENT_CAST(obj) ((GstRTSPClient*)(obj))
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#define GST_RTSP_CLIENT_CLASS_CAST(klass) ((GstRTSPClientClass*)(klass))
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/**
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* GstRTSPClientSendFunc:
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* @client: a #GstRTSPClient
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* @message: a #GstRTSPMessage
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* @close: close the connection
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* @user_data: user data when registering the callback
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*
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* This callback is called when @client wants to send @message. When @close is
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* %TRUE, the connection should be closed when the message has been sent.
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*
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* Returns: %TRUE on success.
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*/
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typedef gboolean (*GstRTSPClientSendFunc) (GstRTSPClient *client,
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GstRTSPMessage *message,
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gboolean close,
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gpointer user_data);
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/**
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* GstRTSPClientSendMessagesFunc:
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* @client: a #GstRTSPClient
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* @messages: #GstRTSPMessage
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* @n_messages: number of messages
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* @close: close the connection
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* @user_data: user data when registering the callback
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*
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* This callback is called when @client wants to send @messages. When @close is
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* %TRUE, the connection should be closed when the message has been sent.
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*
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* Returns: %TRUE on success.
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*
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* Since: 1.16
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*/
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typedef gboolean (*GstRTSPClientSendMessagesFunc) (GstRTSPClient *client,
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GstRTSPMessage *messages,
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guint n_messages,
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gboolean close,
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gpointer user_data);
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/**
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* GstRTSPClient:
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*
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* The client object represents the connection and its state with a client.
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*/
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struct _GstRTSPClient {
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GObject parent;
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/*< private >*/
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GstRTSPClientPrivate *priv;
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gpointer _gst_reserved[GST_PADDING];
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};
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/**
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* GstRTSPClientClass:
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* @create_sdp: called when the SDP needs to be created for media.
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* @configure_client_media: called when the stream in media needs to be configured.
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* The default implementation will configure the blocksize on the payloader when
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* spcified in the request headers.
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* @configure_client_transport: called when the client transport needs to be
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* configured.
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* @params_set: set parameters. This function should also initialize the
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* RTSP response(ctx->response) via a call to gst_rtsp_message_init_response()
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* @params_get: get parameters. This function should also initialize the
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* RTSP response(ctx->response) via a call to gst_rtsp_message_init_response()
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* @make_path_from_uri: called to create path from uri.
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* @adjust_play_mode: called to give the application the possibility to adjust
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* the range, seek flags, rate and rate-control. Since 1.18
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* @adjust_play_response: called to give the implementation the possibility to
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* adjust the response to a play request, for example if extra headers were
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* parsed when #GstRTSPClientClass.adjust_play_mode was called. Since 1.18
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* @tunnel_http_response: called when a response to the GET request is about to
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* be sent for a tunneled connection. The response can be modified. Since: 1.4
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*
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* The client class structure.
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*/
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struct _GstRTSPClientClass {
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GObjectClass parent_class;
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GstSDPMessage * (*create_sdp) (GstRTSPClient *client, GstRTSPMedia *media);
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gboolean (*configure_client_media) (GstRTSPClient * client,
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GstRTSPMedia * media, GstRTSPStream * stream,
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GstRTSPContext * ctx);
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gboolean (*configure_client_transport) (GstRTSPClient * client,
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GstRTSPContext * ctx,
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GstRTSPTransport * ct);
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GstRTSPResult (*params_set) (GstRTSPClient *client, GstRTSPContext *ctx);
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GstRTSPResult (*params_get) (GstRTSPClient *client, GstRTSPContext *ctx);
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gchar * (*make_path_from_uri) (GstRTSPClient *client, const GstRTSPUrl *uri);
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GstRTSPStatusCode (*adjust_play_mode) (GstRTSPClient * client,
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GstRTSPContext * context,
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GstRTSPTimeRange ** range,
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GstSeekFlags * flags,
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gdouble * rate,
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GstClockTime * trickmode_interval,
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gboolean * enable_rate_control);
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GstRTSPStatusCode (*adjust_play_response) (GstRTSPClient * client,
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GstRTSPContext * context);
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/* signals */
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void (*closed) (GstRTSPClient *client);
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void (*new_session) (GstRTSPClient *client, GstRTSPSession *session);
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void (*options_request) (GstRTSPClient *client, GstRTSPContext *ctx);
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void (*describe_request) (GstRTSPClient *client, GstRTSPContext *ctx);
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void (*setup_request) (GstRTSPClient *client, GstRTSPContext *ctx);
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void (*play_request) (GstRTSPClient *client, GstRTSPContext *ctx);
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void (*pause_request) (GstRTSPClient *client, GstRTSPContext *ctx);
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void (*teardown_request) (GstRTSPClient *client, GstRTSPContext *ctx);
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void (*set_parameter_request) (GstRTSPClient *client, GstRTSPContext *ctx);
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void (*get_parameter_request) (GstRTSPClient *client, GstRTSPContext *ctx);
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void (*handle_response) (GstRTSPClient *client, GstRTSPContext *ctx);
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void (*tunnel_http_response) (GstRTSPClient * client, GstRTSPMessage * request,
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GstRTSPMessage * response);
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void (*send_message) (GstRTSPClient * client, GstRTSPContext *ctx,
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GstRTSPMessage * response);
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gboolean (*handle_sdp) (GstRTSPClient *client, GstRTSPContext *ctx, GstRTSPMedia *media, GstSDPMessage *sdp);
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void (*announce_request) (GstRTSPClient *client, GstRTSPContext *ctx);
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void (*record_request) (GstRTSPClient *client, GstRTSPContext *ctx);
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gchar* (*check_requirements) (GstRTSPClient *client, GstRTSPContext *ctx, gchar ** arr);
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GstRTSPStatusCode (*pre_options_request) (GstRTSPClient *client, GstRTSPContext *ctx);
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GstRTSPStatusCode (*pre_describe_request) (GstRTSPClient *client, GstRTSPContext *ctx);
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GstRTSPStatusCode (*pre_setup_request) (GstRTSPClient *client, GstRTSPContext *ctx);
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GstRTSPStatusCode (*pre_play_request) (GstRTSPClient *client, GstRTSPContext *ctx);
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GstRTSPStatusCode (*pre_pause_request) (GstRTSPClient *client, GstRTSPContext *ctx);
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GstRTSPStatusCode (*pre_teardown_request) (GstRTSPClient *client, GstRTSPContext *ctx);
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GstRTSPStatusCode (*pre_set_parameter_request) (GstRTSPClient *client, GstRTSPContext *ctx);
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GstRTSPStatusCode (*pre_get_parameter_request) (GstRTSPClient *client, GstRTSPContext *ctx);
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GstRTSPStatusCode (*pre_announce_request) (GstRTSPClient *client, GstRTSPContext *ctx);
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GstRTSPStatusCode (*pre_record_request) (GstRTSPClient *client, GstRTSPContext *ctx);
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/**
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* GstRTSPClientClass::adjust_error_code:
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* @client: a #GstRTSPClient
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* @ctx: a #GstRTSPContext
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* @code: a #GstRTSPStatusCode
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*
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* Called before sending error response to give the application the
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* possibility to adjust the error code.
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*
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* Returns: a #GstRTSPStatusCode, containing the adjusted error code.
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*
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* Since: 1.22
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*/
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GstRTSPStatusCode (*adjust_error_code) (GstRTSPClient *client, GstRTSPContext *ctx, GstRTSPStatusCode code);
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/*< private >*/
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gpointer _gst_reserved[GST_PADDING_LARGE-19];
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};
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GST_RTSP_SERVER_API
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GType gst_rtsp_client_get_type (void);
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GST_RTSP_SERVER_API
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GstRTSPClient * gst_rtsp_client_new (void);
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GST_RTSP_SERVER_API
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void gst_rtsp_client_set_session_pool (GstRTSPClient *client,
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GstRTSPSessionPool *pool);
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GST_RTSP_SERVER_API
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GstRTSPSessionPool * gst_rtsp_client_get_session_pool (GstRTSPClient *client);
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GST_RTSP_SERVER_API
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void gst_rtsp_client_set_mount_points (GstRTSPClient *client,
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GstRTSPMountPoints *mounts);
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GST_RTSP_SERVER_API
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GstRTSPMountPoints * gst_rtsp_client_get_mount_points (GstRTSPClient *client);
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GST_RTSP_SERVER_API
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void gst_rtsp_client_set_content_length_limit (GstRTSPClient *client, guint limit);
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GST_RTSP_SERVER_API
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guint gst_rtsp_client_get_content_length_limit (GstRTSPClient *client);
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GST_RTSP_SERVER_API
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void gst_rtsp_client_set_auth (GstRTSPClient *client, GstRTSPAuth *auth);
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GST_RTSP_SERVER_API
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GstRTSPAuth * gst_rtsp_client_get_auth (GstRTSPClient *client);
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GST_RTSP_SERVER_API
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void gst_rtsp_client_set_thread_pool (GstRTSPClient *client, GstRTSPThreadPool *pool);
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GST_RTSP_SERVER_API
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GstRTSPThreadPool * gst_rtsp_client_get_thread_pool (GstRTSPClient *client);
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GST_RTSP_SERVER_API
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gboolean gst_rtsp_client_set_connection (GstRTSPClient *client, GstRTSPConnection *conn);
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GST_RTSP_SERVER_API
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GstRTSPConnection * gst_rtsp_client_get_connection (GstRTSPClient *client);
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GST_RTSP_SERVER_API
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guint gst_rtsp_client_attach (GstRTSPClient *client,
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GMainContext *context);
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GST_RTSP_SERVER_API
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void gst_rtsp_client_close (GstRTSPClient * client);
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GST_RTSP_SERVER_API
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void gst_rtsp_client_set_send_func (GstRTSPClient *client,
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GstRTSPClientSendFunc func,
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gpointer user_data,
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GDestroyNotify notify);
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GST_RTSP_SERVER_API
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void gst_rtsp_client_set_send_messages_func (GstRTSPClient *client,
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GstRTSPClientSendMessagesFunc func,
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gpointer user_data,
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GDestroyNotify notify);
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GST_RTSP_SERVER_API
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GstRTSPResult gst_rtsp_client_handle_message (GstRTSPClient *client,
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GstRTSPMessage *message);
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GST_RTSP_SERVER_API
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GstRTSPResult gst_rtsp_client_send_message (GstRTSPClient * client,
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GstRTSPSession *session,
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GstRTSPMessage *message);
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/**
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* GstRTSPClientSessionFilterFunc:
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* @client: a #GstRTSPClient object
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* @sess: a #GstRTSPSession in @client
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* @user_data: user data that has been given to gst_rtsp_client_session_filter()
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*
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* This function will be called by the gst_rtsp_client_session_filter(). An
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* implementation should return a value of #GstRTSPFilterResult.
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*
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* When this function returns #GST_RTSP_FILTER_REMOVE, @sess will be removed
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* from @client.
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*
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* A return value of #GST_RTSP_FILTER_KEEP will leave @sess untouched in
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* @client.
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*
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* A value of #GST_RTSP_FILTER_REF will add @sess to the result #GList of
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* gst_rtsp_client_session_filter().
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*
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* Returns: a #GstRTSPFilterResult.
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*/
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typedef GstRTSPFilterResult (*GstRTSPClientSessionFilterFunc) (GstRTSPClient *client,
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GstRTSPSession *sess,
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gpointer user_data);
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GST_RTSP_SERVER_API
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GList * gst_rtsp_client_session_filter (GstRTSPClient *client,
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GstRTSPClientSessionFilterFunc func,
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gpointer user_data);
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GST_RTSP_SERVER_API
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GstRTSPStreamTransport * gst_rtsp_client_get_stream_transport (GstRTSPClient *client,
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guint8 channel);
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#ifdef G_DEFINE_AUTOPTR_CLEANUP_FUNC
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G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstRTSPClient, gst_object_unref)
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#endif
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G_END_DECLS
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#endif /* __GST_RTSP_CLIENT_H__ */
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