我需要流式传输RTSP / HTTP流，当我使用source = gst_element_make_from_uri();创建元素源时，它不起作用。有人使用gstreamer来流式传输RTSP / HTTP流吗？

source   = gst_element_make_from_uri (GST_URI_SRC,"http://76.73.90.27:80/" ,NULL);
decoder = gst_element_factory_make ("mad",      "mad-decoder"); 
sink     = gst_element_factory_make ("alsasink", "audio-output");
g_object_set (G_OBJECT (source), "location", argv[1], NULL);
gst_element_set_state (pipeline, GST_STATE_NULL);
bus = gst_pipeline_get_bus (GST_PIPELINE (pipeline));
gst_bus_add_watch (bus, bus_call, loop);
gst_object_unref (bus);
gst_bin_add_many (GST_BIN (pipeline),
                  source, decoder,sink, NULL);
gst_element_link_many (source, decoder, sink, NULL);
gst_element_set_state (pipeline, GST_STATE_PLAYING);
g_main_loop_run (loop);
gst_element_set_state (pipeline, GST_STATE_NULL);
gst_object_unref (GST_OBJECT (pipeline));