mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-10-03 01:02:19 +00:00
41dd6399a6
Added various other test server examples Move the SDP message generation to a separate helper. Refactor common code for finding the session. Add content-base for realplayer compatibility Clean up request uris before processing for better vlc compatibility. Move prerolling and pipeline construction to the RTSPMedia object. Use multiudpsink for future pipeline reuse.
846 lines
22 KiB
C
846 lines
22 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., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, USA.
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*/
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#include <sys/ioctl.h>
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#include "rtsp-client.h"
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#include "rtsp-sdp.h"
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#undef DEBUG
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static void gst_rtsp_client_finalize (GObject * obj);
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G_DEFINE_TYPE (GstRTSPClient, gst_rtsp_client, G_TYPE_OBJECT);
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static void
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gst_rtsp_client_class_init (GstRTSPClientClass * klass)
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{
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GObjectClass *gobject_class;
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gobject_class = G_OBJECT_CLASS (klass);
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gobject_class->finalize = gst_rtsp_client_finalize;
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}
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static void
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gst_rtsp_client_init (GstRTSPClient * client)
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{
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}
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static void
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gst_rtsp_client_finalize (GObject * obj)
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{
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G_OBJECT_CLASS (gst_rtsp_client_parent_class)->finalize (obj);
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}
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/**
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* gst_rtsp_client_new:
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*
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* Create a new #GstRTSPClient instance.
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*/
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GstRTSPClient *
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gst_rtsp_client_new (void)
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{
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GstRTSPClient *result;
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result = g_object_new (GST_TYPE_RTSP_CLIENT, NULL);
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return result;
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}
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static void
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handle_response (GstRTSPClient *client, GstRTSPMessage *response)
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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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gst_rtsp_connection_send (client->connection, response, NULL);
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}
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static void
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handle_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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handle_response (client, &response);
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}
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static GstRTSPMedia *
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find_media (GstRTSPClient *client, const GstRTSPUrl *uri, GstRTSPMessage *request)
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{
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GstRTSPMediaFactory *factory;
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GstRTSPMedia *media;
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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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goto no_factory;
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/* prepare the media and add it to the pipeline */
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if (!(media = gst_rtsp_media_factory_construct (factory, uri)))
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goto no_media;
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/* prepare the media */
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if (!(gst_rtsp_media_prepare (media)))
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goto no_prepare;
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return media;
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/* ERRORS */
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no_factory:
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{
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handle_generic_response (client, GST_RTSP_STS_NOT_FOUND, request);
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return NULL;
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}
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no_media:
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{
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handle_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, request);
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g_object_unref (factory);
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return NULL;
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}
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no_prepare:
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{
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handle_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, request);
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g_object_unref (media);
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g_object_unref (factory);
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return NULL;
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}
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}
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/* Get the session or NULL when there was no session */
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static GstRTSPSession *
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ensure_session (GstRTSPClient *client, GstRTSPMessage *request)
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{
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GstRTSPResult res;
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GstRTSPSession *session;
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gchar *sessid;
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res = gst_rtsp_message_get_header (request, GST_RTSP_HDR_SESSION, &sessid, 0);
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if (res == GST_RTSP_OK) {
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/* we had a session in the request, find it again */
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if (!(session = gst_rtsp_session_pool_find (client->session_pool, sessid)))
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goto session_not_found;
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}
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else
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goto service_unavailable;
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return session;
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/* ERRORS */
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session_not_found:
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{
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handle_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, request);
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return NULL;
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}
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service_unavailable:
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{
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handle_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, request);
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return NULL;
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}
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}
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static gboolean
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handle_teardown_response (GstRTSPClient *client, GstRTSPUrl *uri, GstRTSPMessage *request)
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{
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GstRTSPSessionMedia *media;
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GstRTSPSession *session;
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GstRTSPMessage response = { 0 };
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GstRTSPStatusCode code;
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if (!(session = ensure_session (client, request)))
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goto no_session;
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/* get a handle to the configuration of the media in the session */
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media = gst_rtsp_session_get_media (session, uri);
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if (!media)
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goto not_found;
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gst_rtsp_session_media_stop (media);
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gst_rtsp_session_pool_remove (client->session_pool, session);
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g_object_unref (session);
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/* remove the session id from the request, which will also remove it from the
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* response */
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gst_rtsp_message_remove_header (request, GST_RTSP_HDR_SESSION, -1);
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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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handle_response (client, &response);
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return FALSE;
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/* ERRORS */
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no_session:
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{
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/* error was sent already */
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return FALSE;
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}
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not_found:
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{
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handle_generic_response (client, GST_RTSP_STS_NOT_FOUND, 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_pause_response (GstRTSPClient *client, GstRTSPUrl *uri, GstRTSPMessage *request)
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{
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GstRTSPSessionMedia *media;
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GstRTSPSession *session;
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GstRTSPMessage response = { 0 };
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GstRTSPStatusCode code;
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if (!(session = ensure_session (client, request)))
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goto no_session;
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/* get a handle to the configuration of the media in the session */
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media = gst_rtsp_session_get_media (session, uri);
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if (!media)
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goto not_found;
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gst_rtsp_session_media_pause (media);
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g_object_unref (session);
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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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handle_response (client, &response);
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return FALSE;
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/* ERRORS */
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no_session:
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{
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return FALSE;
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}
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not_found:
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{
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handle_generic_response (client, GST_RTSP_STS_NOT_FOUND, 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_response (GstRTSPClient *client, GstRTSPUrl *uri, GstRTSPMessage *request)
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{
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GstRTSPSessionMedia *media;
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GstRTSPSession *session;
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GstRTSPMessage response = { 0 };
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GstRTSPStatusCode code;
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GString *rtpinfo;
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guint n_streams, i;
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guint timestamp, seqnum;
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if (!(session = ensure_session (client, request)))
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goto no_session;
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/* get a handle to the configuration of the media in the session */
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media = gst_rtsp_session_get_media (session, uri);
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if (!media)
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goto not_found;
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/* grab RTPInfo from the payloaders now */
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rtpinfo = g_string_new ("");
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n_streams = gst_rtsp_media_n_streams (media->media);
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for (i = 0; i < n_streams; i++) {
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GstRTSPMediaStream *stream;
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gchar *uristr;
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stream = gst_rtsp_media_get_stream (media->media, i);
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g_object_get (G_OBJECT (stream->payloader), "seqnum", &seqnum, NULL);
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g_object_get (G_OBJECT (stream->payloader), "timestamp", ×tamp, NULL);
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if (i > 0)
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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_free (uristr);
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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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/* add the RTP-Info header */
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gst_rtsp_message_add_header (&response, GST_RTSP_HDR_RTP_INFO, rtpinfo->str);
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g_string_free (rtpinfo, TRUE);
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handle_response (client, &response);
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/* start playing after sending the request */
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gst_rtsp_session_media_play (media);
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g_object_unref (session);
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return FALSE;
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/* ERRORS */
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no_session:
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{
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/* error was sent */
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return FALSE;
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}
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not_found:
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{
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handle_generic_response (client, GST_RTSP_STS_NOT_FOUND, 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_setup_response (GstRTSPClient *client, GstRTSPUrl *uri, GstRTSPMessage *request)
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{
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GstRTSPResult res;
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gchar *sessid;
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gchar *transport;
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gchar **transports;
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gboolean have_transport;
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GstRTSPTransport *ct, *st;
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GstRTSPSession *session;
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gint i;
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GstRTSPLowerTrans supported;
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GstRTSPMessage response = { 0 };
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GstRTSPStatusCode code;
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GstRTSPSessionStream *stream;
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gchar *trans_str, *pos;
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guint streamid;
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GstRTSPSessionMedia *media;
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gboolean need_session;
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/* the uri contains the stream number we added in the SDP config, which is
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* always /stream=%d so we need to strip that off
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* parse the stream we need to configure, look for the stream in the abspath
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* first and then in the query. */
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if (!(pos = strstr (uri->abspath, "/stream="))) {
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if (!(pos = strstr (uri->query, "/stream=")))
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goto bad_request;
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}
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/* we can mofify the parse uri in place */
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*pos = '\0';
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pos += strlen ("/stream=");
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if (sscanf (pos, "%u", &streamid) != 1)
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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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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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/* 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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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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break;
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}
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}
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g_strfreev (transports);
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g_free (ct->destination);
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ct->destination = g_strdup (inet_ntoa (client->address.sin_addr));
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/* we have not found anything usable, error out */
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if (!have_transport) {
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goto unsupported_transports;
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}
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/* we have a valid transport, check if we can handle it */
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if (ct->trans != GST_RTSP_TRANS_RTP)
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goto unsupported_transports;
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if (ct->profile != GST_RTSP_PROFILE_AVP)
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goto unsupported_transports;
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supported = GST_RTSP_LOWER_TRANS_UDP |
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GST_RTSP_LOWER_TRANS_UDP_MCAST | GST_RTSP_LOWER_TRANS_TCP;
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if (!(ct->lower_transport & supported))
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goto unsupported_transports;
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/* a setup request creates a session for a client, check if the client already
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* sent a session id to us */
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res = gst_rtsp_message_get_header (request, GST_RTSP_HDR_SESSION, &sessid, 0);
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if (res == GST_RTSP_OK) {
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/* we had a session in the request, find it again */
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if (!(session = gst_rtsp_session_pool_find (client->session_pool, sessid)))
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goto session_not_found;
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need_session = FALSE;
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}
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else {
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/* create a session if this fails we probably reached our session limit or
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* something. */
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if (!(session = gst_rtsp_session_pool_create (client->session_pool)))
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goto service_unavailable;
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need_session = TRUE;
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}
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if (need_session) {
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GstRTSPMedia *m;
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/* get a handle to the configuration of the media in the session */
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if ((m = find_media (client, uri, request))) {
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media = gst_rtsp_session_manage_media (session, uri, m);
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}
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}
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/* get a handle to the configuration of the media in the session */
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if (!(media = gst_rtsp_session_get_media (session, uri)))
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goto not_found;
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/* get a handle to the stream in the media */
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if (!(stream = gst_rtsp_session_media_get_stream (media, streamid)))
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goto no_stream;
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/* setup the server transport from the client transport */
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st = gst_rtsp_session_stream_set_transport (stream, ct);
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/* serialize the server transport */
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trans_str = gst_rtsp_transport_as_text (st);
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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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if (need_session)
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gst_rtsp_message_add_header (&response, GST_RTSP_HDR_SESSION, session->sessionid);
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gst_rtsp_message_add_header (&response, GST_RTSP_HDR_TRANSPORT, trans_str);
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g_free (trans_str);
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g_object_unref (session);
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handle_response (client, &response);
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return TRUE;
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/* ERRORS */
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bad_request:
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{
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handle_generic_response (client, GST_RTSP_STS_BAD_REQUEST, request);
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return FALSE;
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}
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not_found:
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{
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handle_generic_response (client, GST_RTSP_STS_NOT_FOUND, request);
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return FALSE;
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}
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no_stream:
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{
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handle_generic_response (client, GST_RTSP_STS_NOT_FOUND, request);
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return FALSE;
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}
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session_not_found:
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{
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handle_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, request);
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return FALSE;
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}
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no_transport:
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{
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handle_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, request);
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return FALSE;
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}
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unsupported_transports:
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{
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handle_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, request);
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gst_rtsp_transport_free (ct);
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return FALSE;
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}
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service_unavailable:
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{
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handle_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, request);
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return FALSE;
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}
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}
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/* for the describe we must generate an SDP */
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static gboolean
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handle_describe_response (GstRTSPClient *client, GstRTSPUrl *uri, GstRTSPMessage *request)
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{
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GstRTSPMessage response = { 0 };
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GstRTSPResult res;
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GstSDPMessage *sdp;
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guint i;
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gchar *str;
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GstRTSPMedia *media;
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/* check what kind of format is accepted, we don't really do anything with it
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* and always return SDP for now. */
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for (i = 0; i++; ) {
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gchar *accept;
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res = gst_rtsp_message_get_header (request, GST_RTSP_HDR_ACCEPT, &accept, i);
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if (res == GST_RTSP_ENOTIMPL)
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break;
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if (g_ascii_strcasecmp (accept, "application/sdp") == 0)
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break;
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}
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/* find the media object for the uri */
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if (!(media = find_media (client, uri, request)))
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goto no_media;
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/* create an SDP for the media object */
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if (!(sdp = gst_rtsp_sdp_from_media (media)))
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goto no_sdp;
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gst_rtsp_message_init_response (&response, GST_RTSP_STS_OK,
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gst_rtsp_status_as_text (GST_RTSP_STS_OK), request);
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|
|
gst_rtsp_message_add_header (&response, GST_RTSP_HDR_CONTENT_TYPE, "application/sdp");
|
|
|
|
str = g_strdup_printf ("rtsp://%s:%u%s/", uri->host, uri->port, uri->abspath);
|
|
gst_rtsp_message_add_header (&response, GST_RTSP_HDR_CONTENT_BASE, str);
|
|
g_free (str);
|
|
|
|
/* add SDP to the response body */
|
|
str = gst_sdp_message_as_text (sdp);
|
|
gst_rtsp_message_take_body (&response, (guint8 *)str, strlen (str));
|
|
gst_sdp_message_free (sdp);
|
|
|
|
handle_response (client, &response);
|
|
|
|
return TRUE;
|
|
|
|
/* ERRORS */
|
|
no_media:
|
|
{
|
|
/* error reply is already sent */
|
|
return FALSE;
|
|
}
|
|
no_sdp:
|
|
{
|
|
handle_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, request);
|
|
g_object_unref (media);
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
static void
|
|
handle_options_response (GstRTSPClient *client, GstRTSPUrl *uri, 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_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_add_header (&response, GST_RTSP_HDR_PUBLIC, str);
|
|
g_free (str);
|
|
|
|
handle_response (client, &response);
|
|
}
|
|
|
|
/* remove duplicate and trailing '/' */
|
|
static void
|
|
santize_uri (GstRTSPUrl *uri)
|
|
{
|
|
gint i, len;
|
|
gchar *s, *d;
|
|
gboolean have_slash, prev_slash;
|
|
|
|
s = d = uri->abspath;
|
|
len = strlen (uri->abspath);
|
|
|
|
prev_slash = FALSE;
|
|
|
|
for (i = 0; i < len; i++) {
|
|
have_slash = s[i] == '/';
|
|
*d = s[i];
|
|
if (!have_slash || !prev_slash)
|
|
d++;
|
|
prev_slash = have_slash;
|
|
}
|
|
len = d - uri->abspath;
|
|
/* don't remove the first slash if that's the only thing left */
|
|
if (len > 1 && *(d-1) == '/')
|
|
d--;
|
|
*d = '\0';
|
|
}
|
|
|
|
/* this function runs in a client specific thread and handles all rtsp messages
|
|
* with the client */
|
|
static gpointer
|
|
handle_client (GstRTSPClient *client)
|
|
{
|
|
GstRTSPMessage request = { 0 };
|
|
GstRTSPResult res;
|
|
GstRTSPMethod method;
|
|
const gchar *uristr;
|
|
GstRTSPUrl *uri;
|
|
GstRTSPVersion version;
|
|
|
|
while (TRUE) {
|
|
/* start by waiting for a message from the client */
|
|
res = gst_rtsp_connection_receive (client->connection, &request, NULL);
|
|
if (res < 0)
|
|
goto receive_failed;
|
|
|
|
#ifdef DEBUG
|
|
gst_rtsp_message_dump (&request);
|
|
#endif
|
|
|
|
gst_rtsp_message_parse_request (&request, &method, &uristr, &version);
|
|
|
|
if (version != GST_RTSP_VERSION_1_0) {
|
|
/* we can only handle 1.0 requests */
|
|
handle_generic_response (client, GST_RTSP_STS_RTSP_VERSION_NOT_SUPPORTED, &request);
|
|
continue;
|
|
}
|
|
|
|
/* we always try to parse the url first */
|
|
if ((res = gst_rtsp_url_parse (uristr, &uri)) != GST_RTSP_OK) {
|
|
handle_generic_response (client, GST_RTSP_STS_BAD_REQUEST, &request);
|
|
continue;
|
|
}
|
|
|
|
/* sanitize the uri */
|
|
santize_uri (uri);
|
|
|
|
/* now see what is asked and dispatch to a dedicated handler */
|
|
switch (method) {
|
|
case GST_RTSP_OPTIONS:
|
|
handle_options_response (client, uri, &request);
|
|
break;
|
|
case GST_RTSP_DESCRIBE:
|
|
handle_describe_response (client, uri, &request);
|
|
break;
|
|
case GST_RTSP_SETUP:
|
|
handle_setup_response (client, uri, &request);
|
|
break;
|
|
case GST_RTSP_PLAY:
|
|
handle_play_response (client, uri, &request);
|
|
break;
|
|
case GST_RTSP_PAUSE:
|
|
handle_pause_response (client, uri, &request);
|
|
break;
|
|
case GST_RTSP_TEARDOWN:
|
|
handle_teardown_response (client, uri, &request);
|
|
break;
|
|
case GST_RTSP_ANNOUNCE:
|
|
case GST_RTSP_GET_PARAMETER:
|
|
case GST_RTSP_RECORD:
|
|
case GST_RTSP_REDIRECT:
|
|
case GST_RTSP_SET_PARAMETER:
|
|
handle_generic_response (client, GST_RTSP_STS_NOT_IMPLEMENTED, &request);
|
|
break;
|
|
case GST_RTSP_INVALID:
|
|
default:
|
|
handle_generic_response (client, GST_RTSP_STS_BAD_REQUEST, &request);
|
|
break;
|
|
}
|
|
gst_rtsp_url_free (uri);
|
|
}
|
|
g_object_unref (client);
|
|
return NULL;
|
|
|
|
/* ERRORS */
|
|
receive_failed:
|
|
{
|
|
g_message ("receive failed %d (%s), disconnect client %p", res,
|
|
gst_rtsp_strresult (res), client);
|
|
gst_rtsp_connection_close (client->connection);
|
|
g_object_unref (client);
|
|
return NULL;
|
|
}
|
|
}
|
|
|
|
/* called when we need to accept a new request from a client */
|
|
static gboolean
|
|
client_accept (GstRTSPClient *client, GIOChannel *channel)
|
|
{
|
|
/* a new client connected. */
|
|
int server_sock_fd, fd;
|
|
unsigned int address_len;
|
|
GstRTSPConnection *conn;
|
|
|
|
server_sock_fd = g_io_channel_unix_get_fd (channel);
|
|
|
|
address_len = sizeof (client->address);
|
|
memset (&client->address, 0, address_len);
|
|
|
|
fd = accept (server_sock_fd, (struct sockaddr *) &client->address,
|
|
&address_len);
|
|
if (fd == -1)
|
|
goto accept_failed;
|
|
|
|
/* now create the connection object */
|
|
gst_rtsp_connection_create (NULL, &conn);
|
|
conn->fd.fd = fd;
|
|
|
|
/* FIXME some hackery, we need to have a connection method to accept server
|
|
* connections */
|
|
gst_poll_add_fd (conn->fdset, &conn->fd);
|
|
|
|
g_message ("added new client %p ip %s with fd %d", client,
|
|
inet_ntoa (client->address.sin_addr), conn->fd.fd);
|
|
|
|
client->connection = conn;
|
|
|
|
return TRUE;
|
|
|
|
/* ERRORS */
|
|
accept_failed:
|
|
{
|
|
g_error ("Could not accept client on server socket %d: %s (%d)",
|
|
server_sock_fd, g_strerror (errno), errno);
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_client_set_session_pool:
|
|
* @client: a #GstRTSPClient
|
|
* @pool: a #GstRTSPSessionPool
|
|
*
|
|
* Set @pool as the sessionpool for @client which it will use to find
|
|
* or allocate sessions. the sessionpool is usually inherited from the server
|
|
* that created the client but can be overridden later.
|
|
*/
|
|
void
|
|
gst_rtsp_client_set_session_pool (GstRTSPClient *client, GstRTSPSessionPool *pool)
|
|
{
|
|
GstRTSPSessionPool *old;
|
|
|
|
old = client->session_pool;
|
|
if (old != pool) {
|
|
if (pool)
|
|
g_object_ref (pool);
|
|
client->session_pool = pool;
|
|
if (old)
|
|
g_object_unref (old);
|
|
}
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_client_get_session_pool:
|
|
* @client: a #GstRTSPClient
|
|
*
|
|
* Get the #GstRTSPSessionPool object that @client uses to manage its sessions.
|
|
*
|
|
* Returns: a #GstRTSPSessionPool, unref after usage.
|
|
*/
|
|
GstRTSPSessionPool *
|
|
gst_rtsp_client_get_session_pool (GstRTSPClient *client)
|
|
{
|
|
GstRTSPSessionPool *result;
|
|
|
|
if ((result = client->session_pool))
|
|
g_object_ref (result);
|
|
|
|
return result;
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_client_set_media_mapping:
|
|
* @client: a #GstRTSPClient
|
|
* @mapping: a #GstRTSPMediaMapping
|
|
*
|
|
* Set @mapping as the media mapping for @client which it will use to map urls
|
|
* to media streams. These mapping is usually inherited from the server that
|
|
* created the client but can be overriden later.
|
|
*/
|
|
void
|
|
gst_rtsp_client_set_media_mapping (GstRTSPClient *client, GstRTSPMediaMapping *mapping)
|
|
{
|
|
GstRTSPMediaMapping *old;
|
|
|
|
old = client->media_mapping;
|
|
|
|
if (old != mapping) {
|
|
if (mapping)
|
|
g_object_ref (mapping);
|
|
client->media_mapping = mapping;
|
|
if (old)
|
|
g_object_unref (old);
|
|
}
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_client_get_media_mapping:
|
|
* @client: a #GstRTSPClient
|
|
*
|
|
* Get the #GstRTSPMediaMapping object that @client uses to manage its sessions.
|
|
*
|
|
* Returns: a #GstRTSPMediaMapping, unref after usage.
|
|
*/
|
|
GstRTSPMediaMapping *
|
|
gst_rtsp_client_get_media_mapping (GstRTSPClient *client)
|
|
{
|
|
GstRTSPMediaMapping *result;
|
|
|
|
if ((result = client->media_mapping))
|
|
g_object_ref (result);
|
|
|
|
return result;
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_client_attach:
|
|
* @client: a #GstRTSPClient
|
|
* @channel: a #GIOChannel
|
|
*
|
|
* Accept a new connection for @client on the socket in @source.
|
|
*
|
|
* This function should be called when the client properties and urls are fully
|
|
* configured and the client is ready to start.
|
|
*
|
|
* Returns: %TRUE if the client could be accepted.
|
|
*/
|
|
gboolean
|
|
gst_rtsp_client_accept (GstRTSPClient *client, GIOChannel *channel)
|
|
{
|
|
if (!client_accept (client, channel))
|
|
goto accept_failed;
|
|
|
|
/* client accepted, spawn a thread for the client */
|
|
g_object_ref (client);
|
|
client->thread = g_thread_create ((GThreadFunc)handle_client, client, TRUE, NULL);
|
|
|
|
return TRUE;
|
|
|
|
/* ERRORS */
|
|
accept_failed:
|
|
{
|
|
return FALSE;
|
|
}
|
|
}
|