コード例 #1
0
ファイル: test-launch.c プロジェクト: rlauss/rtsp_server
int
main (int argc, char *argv[])
{
  GMainLoop *loop;
  GstRTSPServer *server;
  GstRTSPMountPoints *mounts;
  GstRTSPMediaFactory *factory;

  gst_init (&argc, &argv);

  if (argc < 2) {
    g_print ("usage: %s <launch line> \n"
        "example: %s \"( videotestsrc ! x264enc ! rtph264pay name=pay0 pt=96 )\"\n",
        argv[0], argv[0]);
    return -1;
  }

  loop = g_main_loop_new (NULL, FALSE);

  /* create a server instance */
  server = gst_rtsp_server_new ();

  /* get the mount points for this server, every server has a default object
   * that be used to map uri mount points to media factories */
  mounts = gst_rtsp_server_get_mount_points (server);

  /* make a media factory for a test stream. The default media factory can use
   * gst-launch syntax to create pipelines.
   * any launch line works as long as it contains elements named pay%d. Each
   * element with pay%d names will be a stream */
  factory = gst_rtsp_media_factory_new ();
  gst_rtsp_media_factory_set_launch (factory, argv[1]);

  gst_rtsp_media_factory_set_shared (factory, TRUE);

  /* attach the test factory to the /test url */
  gst_rtsp_mount_points_add_factory (mounts, "/test", factory);

  /* don't need the ref to the mapper anymore */
  g_object_unref (mounts);

  /* attach the server to the default maincontext */
  gst_rtsp_server_attach (server, NULL);

  /* start serving */
  g_print ("stream ready at rtsp://127.0.0.1:8554/test\n");
  g_main_loop_run (loop);

  return 0;
}
コード例 #2
0
ファイル: ReStream.cpp プロジェクト: ashu1402/NASRepository
//void ReStream::AddVODStream(char* fileName,char * mountUrl, int CID, int DID, char* width,char* height, char* frameRate,char* bitrate, char * outputRtspUrl, bool isResize )
void ReStream::AddVODStream( char* fileName,char * mountUrl, char * outputRtspUrl )
{
	try
	{
		if(isServerAttached){
			GstRTSPMediaFactory *factory;
			char desc[SIZEOFCHARARRAYBIGGER];
			char localUrl[SIZEOFCHARARRAYBIGGER];

			factory = gst_rtsp_media_factory_new ();


			sprintf(desc,"( filesrc location=%s ! decodebin !  x264enc ! rtph264pay name=pay0 pt=96 )",fileName);

			gst_rtsp_media_factory_set_launch (factory,desc);

			gst_rtsp_media_factory_set_shared (factory, FALSE);


			cout<<"mountRl before = "<<mountUrl<<endl;
			sprintf(localUrl,"/%s",mountUrl);
			cout<<"mountRl after = "<<localUrl<<endl;

			gst_rtsp_mount_points_add_factory (mounts, localUrl, factory);
			gst_rtsp_server_set_backlog (server, 10);
			if( isLocalStream ){
				sprintf( outputRtspUrl,"rtsp://%s:%s%s","127.0.0.1",portNo,localUrl);
			}else{
				sprintf( outputRtspUrl,"rtsp://%s:%s%s",ipAddress,portNo,localUrl);
			}
			if ( ISDEBUG ){
				cout<<"\nVODDesc---"<<outputRtspUrl<<endl;
				cout<<"VOD Add Successful\n";
		}
	}
		else{
			if ( ISDEBUG )
				cout<<"Server is Not Attached. Hence cannot add channel\n";
		}
	}
	catch(Exception &e){
		commclass.PrintException("ReStream","CV::AddChannelStream",e);
	}
	catch(exception &e){
		commclass.PrintException("ReStream","STD::AddChannelStream",e);
	}
}
コード例 #3
0
ファイル: client.c プロジェクト: alleen/gst-rtsp-server-wfd
static GstRTSPClient *
setup_multicast_client (void)
{
  GstRTSPClient *client;
  GstRTSPSessionPool *session_pool;
  GstRTSPMountPoints *mount_points;
  GstRTSPMediaFactory *factory;
  GstRTSPAddressPool *address_pool;
  GstRTSPThreadPool *thread_pool;

  client = gst_rtsp_client_new ();

  session_pool = gst_rtsp_session_pool_new ();
  gst_rtsp_client_set_session_pool (client, session_pool);

  mount_points = gst_rtsp_mount_points_new ();
  factory = gst_rtsp_media_factory_new ();
  gst_rtsp_media_factory_set_launch (factory,
      "audiotestsrc ! audio/x-raw,rate=44100 ! audioconvert ! rtpL16pay name=pay0");
  address_pool = gst_rtsp_address_pool_new ();
  fail_unless (gst_rtsp_address_pool_add_range (address_pool,
          "233.252.0.1", "233.252.0.1", 5000, 5010, 1));
  gst_rtsp_media_factory_set_address_pool (factory, address_pool);
  gst_rtsp_media_factory_add_role (factory, "user",
      "media.factory.access", G_TYPE_BOOLEAN, TRUE,
      "media.factory.construct", G_TYPE_BOOLEAN, TRUE, NULL);
  gst_rtsp_mount_points_add_factory (mount_points, "/test", factory);
  gst_rtsp_client_set_mount_points (client, mount_points);

  thread_pool = gst_rtsp_thread_pool_new ();
  gst_rtsp_client_set_thread_pool (client, thread_pool);

  g_object_unref (mount_points);
  g_object_unref (session_pool);
  g_object_unref (address_pool);
  g_object_unref (thread_pool);

  return client;
}
コード例 #4
0
int
main (int argc, char *argv[])
{
  GMainLoop *loop;
  GstRTSPServer *server;
  GstRTSPMountPoints *mounts;
  GstRTSPMediaFactory *factory;

  gst_init (&argc, &argv);

  loop = g_main_loop_new (NULL, FALSE);

  /* create a server instance */
  server = gst_rtsp_server_new ();

  /* get the mount points for this server, every server has a default object
   * that be used to map uri mount points to media factories */
  mounts = gst_rtsp_server_get_mount_points (server);

  /* make a media factory for a test stream. The default media factory can use
   * gst-launch syntax to create pipelines.
   * any launch line works as long as it contains elements named pay%d. Each
   * element with pay%d names will be a stream */
  factory = gst_rtsp_media_factory_new ();
  gst_rtsp_media_factory_set_launch (factory, "( "
      "videotestsrc ! video/x-raw,width=352,height=288,framerate=15/1 ! "
      "x264enc ! rtph264pay name=pay0 pt=96 "
      "audiotestsrc ! audio/x-raw,rate=8000 ! "
      "alawenc ! rtppcmapay name=pay1 pt=8 " ")");

  gst_rtsp_media_factory_set_profiles (factory, GST_RTSP_PROFILE_AVPF);

  /* store up to 0.4 seconds of retransmission data */
  gst_rtsp_media_factory_set_retransmission_time (factory, 400 * GST_MSECOND);

  /* attach the test factory to the /test url */
  gst_rtsp_mount_points_add_factory (mounts, "/test", factory);

  /* don't need the ref to the mapper anymore */
  g_object_unref (mounts);

  /* attach the server to the default maincontext */
  if (gst_rtsp_server_attach (server, NULL) == 0)
    goto failed;

  /* add a timeout for the session cleanup */
  g_timeout_add_seconds (2, (GSourceFunc) timeout, server);

  /* start serving, this never stops */
  g_print ("stream ready at rtsp://127.0.0.1:8554/test\n");

  g_main_loop_run (loop);

  return 0;

  /* ERRORS */
failed:
  {
    g_print ("failed to attach the server\n");
    return -1;
  }
}
コード例 #5
0
int
main (int argc, char *argv[])
{
  GMainLoop *loop;
  GstRTSPServer *server;
  GstRTSPMountPoints *mounts;
  GstRTSPMediaFactory *factory;

  gst_init (&argc, &argv);

  if (argc < 1) {
    g_print ("usage: %s [portnum] \n"
        "example: %s 8555 \n"
        "Pipeline is fixed. Default port 8554\n",
        argv[0], argv[0]);
    return -1;
  }

  loop = g_main_loop_new (NULL, FALSE);

  //global_clock = gst_system_clock_obtain ();
  //gst_net_time_provider_new (global_clock, "0.0.0.0", 8554);

  
  /* create a server instance */
  server = gst_rtsp_server_new ();
  
  gint * portnum;
  portnum = malloc(sizeof(gint));
  *portnum = 8554;
  if (argc == 2)
  {
    /* set server listening port*/
    gst_rtsp_server_set_service (server,argv[1]);
    *portnum = atoi(argv[1]);  
  }
  
  
  /* callback for clients */
  g_signal_connect (server, "client-connected", G_CALLBACK (client_connection), portnum);
  

  /* get the mount points for this server, every server has a default object
   * that be used to map uri mount points to media factories */
  mounts = gst_rtsp_server_get_mount_points (server);

  /* make a media factory for a test stream. The default media factory can use
   * gst-launch syntax to create pipelines.
   * any launch line works as long as it contains elements named pay%d. Each
   * element with pay%d names will be a stream */
  factory = test_rtsp_media_factory_new ();
  gst_rtsp_media_factory_set_launch (factory,
      "appsrc name=mysrc ! videoconvert ! x264enc tune=\"zerolatency\" ! rtph264pay pt=96 ! rtpatimetimestamp name=pay0 ntp-offset=0");

  g_signal_connect (factory, "media-configure", (GCallback) media_configure,
      NULL);      
      
  gst_rtsp_media_factory_set_shared (GST_RTSP_MEDIA_FACTORY (factory), TRUE);
  gst_rtsp_media_factory_set_media_gtype (GST_RTSP_MEDIA_FACTORY (factory),
      TEST_TYPE_RTSP_MEDIA);

  /* attach the test factory to the /test url */
  gst_rtsp_mount_points_add_factory (mounts, "/test", factory);

  /* don't need the ref to the mapper anymore */
  g_object_unref (mounts);

  /* attach the server to the default maincontext */
  gst_rtsp_server_attach (server, NULL);

  /* start serving */
  if (argc < 2)
    g_print ("stream ready at rtsp://127.0.0.1:8554/test\n");
  else
  {
    g_print ("stream ready at rtsp://127.0.0.1:");
    g_print (argv[1]);
    g_print ("/test\n");
  }
  g_main_loop_run (loop);

  return 0;
}
コード例 #6
0
int
main (int argc, char *argv[])
{
  GMainLoop *loop;
  GstRTSPServer *server;
  GstRTSPMountPoints *mounts;
  GstRTSPMediaFactory *factory;
  GOptionContext *optctx;
  GError *error = NULL;
  gchar *str;

  optctx = g_option_context_new ("<filename.ogg> - Test RTSP Server, OGG");
  g_option_context_add_main_entries (optctx, entries, NULL);
  g_option_context_add_group (optctx, gst_init_get_option_group ());
  if (!g_option_context_parse (optctx, &argc, &argv, &error)) {
    g_printerr ("Error parsing options: %s\n", error->message);
    return -1;
  }
  g_option_context_free (optctx);

  loop = g_main_loop_new (NULL, FALSE);

  /* create a server instance */
  server = gst_rtsp_server_new ();
  g_object_set (server, "service", port, NULL);

  /* get the mount points for this server, every server has a default object
   * that be used to map uri mount points to media factories */
  mounts = gst_rtsp_server_get_mount_points (server);

// filesrc location=amy.wav ! wavparse ! audioconvert ! audioresample ! alawenc !  rtppcmapay  audioconvert  ! audioresample ! vorbisenc
// /home/dhruv/Music/

str = "( "
      "flitesrc location=%s ! wavparse ! audioconvert ! audioresample ! alawenc ! rtppcmapay name=pay0 pt=96 "
	 ")";


  /* make a media factory for a test stream. The default media factory can use
   * gst-launch syntax to create pipelines. 
   * any launch line works as long as it contains elements named pay%d. Each
   * element with pay%d names will be a stream */
  factory = gst_rtsp_media_factory_new ();
//klass = GST_RTSP_MEDIA_FACTORY_GET_CLASS (factory);
//klass->create_element = my_default_create_element;

  gst_rtsp_media_factory_set_launch (factory, str);
  //g_free (str);

  /* attach the test factory to the /test url */
  gst_rtsp_mount_points_add_factory (mounts, "/test", factory);

  /* don't need the ref to the mapper anymore */
  g_object_unref (mounts);

  /* attach the server to the default maincontext */
  gst_rtsp_server_attach (server, NULL);

  /* start serving */
  g_print ("stream ready at rtsp://127.0.0.1:%s/test\n", port);
  g_main_loop_run (loop);

  return 0;
}
コード例 #7
0
int
main (int argc, char *argv[])
{
  GMainLoop *loop;
  GstRTSPServer *server;
  GstRTSPMountPoints *mounts;
  GstRTSPMediaFactory *factory;
  GOptionContext *optctx;
  GError *error = NULL;
  gchar *str;

  optctx = g_option_context_new ("<filename.mp4> - Test RTSP Server, MP4");
  g_option_context_add_main_entries (optctx, entries, NULL);
  g_option_context_add_group (optctx, gst_init_get_option_group ());
  if (!g_option_context_parse (optctx, &argc, &argv, &error)) {
    g_printerr ("Error parsing options: %s\n", error->message);
    return -1;
  }

  if (argc < 2) {
    g_print ("%s\n", g_option_context_get_help (optctx, TRUE, NULL));
    return 1;
  }
  g_option_context_free (optctx);

  loop = g_main_loop_new (NULL, FALSE);

  /* create a server instance */
  server = gst_rtsp_server_new ();
  g_object_set (server, "service", port, NULL);

  /* get the mount points for this server, every server has a default object
   * that be used to map uri mount points to media factories */
  mounts = gst_rtsp_server_get_mount_points (server);

  str = g_strdup_printf ("( "
      "filesrc location=\"%s\" ! qtdemux name=d "
      "d. ! queue ! rtph264pay pt=96 name=pay0 "
      "d. ! queue ! rtpmp4apay pt=97 name=pay1 " ")", argv[1]);

  /* make a media factory for a test stream. The default media factory can use
   * gst-launch syntax to create pipelines. 
   * any launch line works as long as it contains elements named pay%d. Each
   * element with pay%d names will be a stream */
  factory = gst_rtsp_media_factory_new ();
  gst_rtsp_media_factory_set_launch (factory, str);
  g_signal_connect (factory, "media-configure", (GCallback) media_configure_cb,
      factory);
  g_free (str);

  /* attach the test factory to the /test url */
  gst_rtsp_mount_points_add_factory (mounts, "/test", factory);

  /* don't need the ref to the mapper anymore */
  g_object_unref (mounts);

  /* attach the server to the default maincontext */
  gst_rtsp_server_attach (server, NULL);

  /* start serving */
  g_print ("stream ready at rtsp://127.0.0.1:%s/test\n", port);
  g_main_loop_run (loop);

  return 0;
}
コード例 #8
0
int
main (int argc, char *argv[])
{
  GMainLoop *loop;
  GstRTSPServer *server;
  GstRTSPMountPoints *mounts;
  GstRTSPMediaFactory *factory;
  GOptionContext *optctx;
  GError *error = NULL;
  GstRTSPAuth *auth;
  GstRTSPToken *token;
  gchar *basic;
#ifdef WITH_TLS
  GTlsCertificate *cert;
#endif

  optctx = g_option_context_new ("<launch line> - Test RTSP Server, Launch\n\n"
      "Example: \"( decodebin name=depay0 ! autovideosink )\"");
  g_option_context_add_main_entries (optctx, entries, NULL);
  g_option_context_add_group (optctx, gst_init_get_option_group ());
  if (!g_option_context_parse (optctx, &argc, &argv, &error)) {
    g_printerr ("Error parsing options: %s\n", error->message);
    return -1;
  }

  if (argc < 2) {
    g_print ("%s\n", g_option_context_get_help (optctx, TRUE, NULL));
    return 1;
  }
  g_option_context_free (optctx);

  loop = g_main_loop_new (NULL, FALSE);

  /* create a server instance */
  server = gst_rtsp_server_new ();
  g_object_set (server, "service", port, NULL);

  /* get the mount points for this server, every server has a default object
   * that be used to map uri mount points to media factories */
  mounts = gst_rtsp_server_get_mount_points (server);

  /* make a media factory for a test stream. The default media factory can use
   * gst-launch syntax to create pipelines.
   * any launch line works as long as it contains elements named depay%d. Each
   * element with depay%d names will be a stream */
  factory = gst_rtsp_media_factory_new ();
  gst_rtsp_media_factory_set_transport_mode (factory,
      GST_RTSP_TRANSPORT_MODE_RECORD);
  gst_rtsp_media_factory_set_launch (factory, argv[1]);
  gst_rtsp_media_factory_set_latency (factory, 2000);
#ifdef WITH_TLS
  gst_rtsp_media_factory_set_profiles (factory,
      GST_RTSP_PROFILE_SAVP | GST_RTSP_PROFILE_SAVPF);
#else
  gst_rtsp_media_factory_set_profiles (factory,
      GST_RTSP_PROFILE_AVP | GST_RTSP_PROFILE_AVPF);
#endif

  /* allow user to access this resource */
  gst_rtsp_media_factory_add_role (factory, "user",
      GST_RTSP_PERM_MEDIA_FACTORY_ACCESS, G_TYPE_BOOLEAN, TRUE,
      GST_RTSP_PERM_MEDIA_FACTORY_CONSTRUCT, G_TYPE_BOOLEAN, TRUE, NULL);
  /* Anonymous users can see but not construct, so get UNAUTHORIZED */
  gst_rtsp_media_factory_add_role (factory, "anonymous",
      GST_RTSP_PERM_MEDIA_FACTORY_ACCESS, G_TYPE_BOOLEAN, TRUE,
      GST_RTSP_PERM_MEDIA_FACTORY_CONSTRUCT, G_TYPE_BOOLEAN, FALSE, NULL);

  /* attach the test factory to the /test url */
  gst_rtsp_mount_points_add_factory (mounts, "/test", factory);

  /* don't need the ref to the mapper anymore */
  g_object_unref (mounts);

  /* Set up the auth for user account */
  /* make a new authentication manager */
  auth = gst_rtsp_auth_new ();
#ifdef WITH_TLS
  cert = g_tls_certificate_new_from_pem ("-----BEGIN CERTIFICATE-----"
      "MIICJjCCAY+gAwIBAgIBBzANBgkqhkiG9w0BAQUFADCBhjETMBEGCgmSJomT8ixk"
      "ARkWA0NPTTEXMBUGCgmSJomT8ixkARkWB0VYQU1QTEUxHjAcBgNVBAsTFUNlcnRp"
      "ZmljYXRlIEF1dGhvcml0eTEXMBUGA1UEAxMOY2EuZXhhbXBsZS5jb20xHTAbBgkq"
      "hkiG9w0BCQEWDmNhQGV4YW1wbGUuY29tMB4XDTExMDExNzE5NDcxN1oXDTIxMDEx"
      "NDE5NDcxN1owSzETMBEGCgmSJomT8ixkARkWA0NPTTEXMBUGCgmSJomT8ixkARkW"
      "B0VYQU1QTEUxGzAZBgNVBAMTEnNlcnZlci5leGFtcGxlLmNvbTBcMA0GCSqGSIb3"
      "DQEBAQUAA0sAMEgCQQDYScTxk55XBmbDM9zzwO+grVySE4rudWuzH2PpObIonqbf"
      "hRoAalKVluG9jvbHI81eXxCdSObv1KBP1sbN5RzpAgMBAAGjIjAgMAkGA1UdEwQC"
      "MAAwEwYDVR0lBAwwCgYIKwYBBQUHAwEwDQYJKoZIhvcNAQEFBQADgYEAYx6fMqT1"
      "Gvo0jq88E8mc+bmp4LfXD4wJ7KxYeadQxt75HFRpj4FhFO3DOpVRFgzHlOEo3Fwk"
      "PZOKjvkT0cbcoEq5whLH25dHoQxGoVQgFyAP5s+7Vp5AlHh8Y/vAoXeEVyy/RCIH"
      "QkhUlAflfDMcrrYjsmwoOPSjhx6Mm/AopX4="
      "-----END CERTIFICATE-----"
      "-----BEGIN PRIVATE KEY-----"
      "MIIBVAIBADANBgkqhkiG9w0BAQEFAASCAT4wggE6AgEAAkEA2EnE8ZOeVwZmwzPc"
      "88DvoK1ckhOK7nVrsx9j6TmyKJ6m34UaAGpSlZbhvY72xyPNXl8QnUjm79SgT9bG"
      "zeUc6QIDAQABAkBRFJZ32VbqWMP9OVwDJLiwC01AlYLnka0mIQZbT/2xq9dUc9GW"
      "U3kiVw4lL8v/+sPjtTPCYYdzHHOyDen6znVhAiEA9qJT7BtQvRxCvGrAhr9MS022"
      "tTdPbW829BoUtIeH64cCIQDggG5i48v7HPacPBIH1RaSVhXl8qHCpQD3qrIw3FMw"
      "DwIga8PqH5Sf5sHedy2+CiK0V4MRfoU4c3zQ6kArI+bEgSkCIQCLA1vXBiE31B5s"
      "bdHoYa1BXebfZVd+1Hd95IfEM5mbRwIgSkDuQwV55BBlvWph3U8wVIMIb4GStaH8"
      "W535W8UBbEg=" "-----END PRIVATE KEY-----", -1, &error);
  if (cert == NULL) {
    g_printerr ("failed to parse PEM: %s\n", error->message);
    return -1;
  }
  gst_rtsp_auth_set_tls_certificate (auth, cert);
  g_object_unref (cert);
#endif

  /* make default token - anonymous unauthenticated access */
  token =
      gst_rtsp_token_new (GST_RTSP_TOKEN_MEDIA_FACTORY_ROLE, G_TYPE_STRING,
      "anonymous", NULL);
  gst_rtsp_auth_set_default_token (auth, token);
  gst_rtsp_token_unref (token);

  /* make user token */
  token =
      gst_rtsp_token_new (GST_RTSP_TOKEN_MEDIA_FACTORY_ROLE, G_TYPE_STRING,
      "user", NULL);
  basic = gst_rtsp_auth_make_basic ("user", "password");
  gst_rtsp_auth_add_basic (auth, basic, token);
  g_free (basic);
  gst_rtsp_token_unref (token);

  /* set as the server authentication manager */
  gst_rtsp_server_set_auth (server, auth);
  g_object_unref (auth);

  /* attach the server to the default maincontext */
  gst_rtsp_server_attach (server, NULL);

  /* start serving */
#ifdef WITH_TLS
  g_print ("stream ready at rtsps://127.0.0.1:%s/test\n", port);
#else
  g_print ("stream ready at rtsp://127.0.0.1:%s/test\n", port);
#endif
  g_main_loop_run (loop);

  return 0;
}