GST rtspsrc example, bug on windows

rtspsrc2.c

/***
https://bugzilla.gnome.org/show_bug.cgi?id=756818
***/
#include <gst/gst.h>
#include <gst/app/gstappsink.h>
#include <gst/gstpad.h>
#include <gst/rtsp/gstrtsp.h>


typedef struct myDataTag {
	GstElement *pipeline;
	GstElement *rtspsrc;
	GstElement *depayloader;
	GstElement *decoder;
	GstElement *sink;
} myData_t;

myData_t appData;

static void pad_added_handler(GstElement *src, GstPad *new_pad, myData_t *pThis) 
{
	GstPad *sink_pad = gst_element_get_static_pad(pThis->depayloader, "sink");
	GstPadLinkReturn ret;
	GstCaps *new_pad_caps = NULL;
	GstStructure *new_pad_struct = NULL;
	const gchar *new_pad_type = NULL;

	g_print("Received new pad '%s' from '%s':\n", GST_PAD_NAME(new_pad), GST_ELEMENT_NAME(src));

	/* Check the new pad's name */
	if (!g_str_has_prefix(GST_PAD_NAME(new_pad), "recv_rtp_src_")) {
		g_print("  It is not the right pad.  Need recv_rtp_src_. Ignoring.\n");
		goto exit;
	}

	/* If our converter is already linked, we have nothing to do here */
	if (gst_pad_is_linked(sink_pad)) {
		g_print(" Sink pad from %s already linked. Ignoring.\n", GST_ELEMENT_NAME(src));
		goto exit;
	}

	/* Check the new pad's type */
	new_pad_caps = gst_pad_query_caps(new_pad, NULL);
	new_pad_struct = gst_caps_get_structure(new_pad_caps, 0);
	new_pad_type = gst_structure_get_name(new_pad_struct);

	/* Attempt the link */
	ret = gst_pad_link(new_pad, sink_pad);
	if (GST_PAD_LINK_FAILED(ret)) {
		g_print("  Type is '%s' but link failed.\n", new_pad_type);
	}
	else {
		g_print("  Link succeeded (type '%s').\n", new_pad_type);
	}

exit:
	/* Unreference the new pad's caps, if we got them */
	if (new_pad_caps != NULL)
		gst_caps_unref(new_pad_caps);

	/* Unreference the sink pad */
	gst_object_unref(sink_pad);
}

static gboolean select_stream_callback(
	GstElement *rtspsrc, guint num,
	GstCaps *caps, gpointer udata)
{
	g_print("connected stream %d from %s \n", 
		num, GST_ELEMENT_NAME(rtspsrc));

	int size = gst_caps_get_size(caps);
	g_print("caps size = %d \n", size);

	if (size > 0)
	{
		const GstStructure* structure = gst_caps_get_structure(caps, 0);

		g_print("parse first sturcture %s has %d fields \n", 
			gst_structure_get_name(structure), gst_structure_n_fields(structure));

		gchar* rstructure = gst_structure_to_string(structure);
		g_print("%s \n", rstructure);
		g_print("\n");
#if 0
		for (int i = 0; i < gst_structure_n_fields(structure); i++)
		{
			const char* fieldName = gst_structure_nth_field_name(structure, i);
			const GValue* gval = gst_structure_get_value();

			g_print("%d) %s \n", i, fieldName);
		}
#endif
	}

	return true;
}

int main(int argc, char *argv[])
{
	GstBus *bus;
	GstMessage *msg;
	GstStateChangeReturn ret;

	gst_init(NULL, NULL);

	appData.pipeline = gst_pipeline_new("videoclient");
	appData.rtspsrc = gst_element_factory_make("rtspsrc", "rtspsrc");
	g_object_set(
		G_OBJECT(appData.rtspsrc), 
		"location", 
		"rtsp://192.168.0.3:554/Streaming/Channels/101?transportmode=unicast&profile=Profile_1",
		NULL);

	g_object_set(G_OBJECT(appData.rtspsrc), "user-id", "admin", "user-pw", "passwd", NULL);

	g_object_set(G_OBJECT(appData.rtspsrc), "udp-buffer-size", 2097152, NULL);

	g_object_set(G_OBJECT(appData.rtspsrc), "udp-reconnect", false, NULL);

	/***
	rsj: 
	UDP bug... Invalid pointer address
	https://bugzilla.gnome.org/show_bug.cgi?id=756818
	***/
	g_object_set(G_OBJECT(appData.rtspsrc), "protocols", GST_RTSP_LOWER_TRANS_TCP, NULL);

	g_object_set(G_OBJECT(appData.rtspsrc), "timeout", 500000, NULL);

	appData.depayloader = gst_element_factory_make("rtph264depay", "depayloader");
	appData.decoder = gst_element_factory_make("avdec_h264", "h264-decoder");
	appData.sink = gst_element_factory_make("autovideosink", "sink");

	//then add all elements together
	gst_bin_add_many(GST_BIN(appData.pipeline), 
		appData.rtspsrc, appData.depayloader, appData.decoder, appData.sink, NULL);

	//link everythink after source
	gst_element_link_many(appData.depayloader, appData.decoder, appData.sink, NULL);

	//just for debug
	g_signal_connect(appData.rtspsrc, "select-stream", G_CALLBACK(select_stream_callback), NULL);
	/*
	 * Connect to the pad-added signal for the rtpbin.  This allows us to link
	 * the dynamic RTP source pad to the depayloader when it is created.
	 */
	g_signal_connect(appData.rtspsrc, "pad-added", G_CALLBACK(pad_added_handler), &appData);

	/* Set the pipeline to "playing" state*/
	ret = gst_element_set_state(appData.pipeline, GST_STATE_PLAYING);
	if (ret == GST_STATE_CHANGE_FAILURE) {
		g_printerr("Unable to set the pipeline to the playing state.\n");
		gst_object_unref(appData.pipeline);
		return -1;
	}

	/* Wait until error or EOS */
	bus = gst_element_get_bus(appData.pipeline);
	msg = gst_bus_timed_pop_filtered(bus, GST_CLOCK_TIME_NONE, (GstMessageType)(GST_MESSAGE_ERROR | GST_MESSAGE_EOS));

	/* Parse message */
	if (msg != NULL) {
		GError *err;
		gchar *debug_info;

		switch (GST_MESSAGE_TYPE(msg)) {
		case GST_MESSAGE_ERROR:
			gst_message_parse_error(msg, &err, &debug_info);
			g_printerr("Error received from element %s: %s\n", GST_OBJECT_NAME(msg->src), err->message);
			g_printerr("Debugging information: %s\n", debug_info ? debug_info : "none");
			g_clear_error(&err);
			g_free(debug_info);
			break;
		case GST_MESSAGE_EOS:
			g_print("End-Of-Stream reached.\n");
			break;
		default:
			/* We should not reach here because we only asked for ERRORs and EOS */
			g_printerr("Unexpected message received.\n");
			break;
		}
		gst_message_unref(msg);
	}

	/* Free resources */
	gst_object_unref(bus);
	gst_element_set_state(appData.pipeline, GST_STATE_NULL);
	gst_object_unref(appData.pipeline);
	return 0;
}