scaletempo again

[padam]

Hi, sorry that i ask so many questions.
I wrote that simple app to play audio, but it doesn't want to add scaletempo pad to pipeline... I get errors !gst_pad_is_linked <target> failed and GST_IS_PAD <pad> failed. Another problem is that, that when scaletempo is linking player doesn't want to play. Where i made a mistake? Thanks for your patience.

#include <gst/gst.h>
#include <glib.h>

GstElement *pipeline;

static gboolean bus_call (GstBus *bus, GstMessage *msg, gpointer data)
{
        GMainLoop *loop = (GMainLoop *) data;
        switch (GST_MESSAGE_TYPE (msg))
        {
        case GST_MESSAGE_EOS:
                g_print ("End of stream\n");
                g_main_loop_quit (loop);
                break;
        case GST_MESSAGE_ERROR:
                {
                        gchar *debug;
                        GError *error;
                        gst_message_parse_error (msg, &error, &debug);
                        g_free (debug);
                        g_printerr ("Error: %s\n", error->message);
                        g_error_free (error);
                        g_main_loop_quit (loop);
                        break;
                }
        default:
                break;
        }
        return TRUE;
}

static void on_pad_added (GstElement *element, GstPad *pad, gpointer data)
{
        GstPad *sinkpad;
        GstElement *decoder = (GstElement *) data;
        /* We can now link this pad with the vorbis-decoder sink pad */
        g_print ("Dynamic pad created, linking demuxer/decoder\n");
        sinkpad = gst_element_get_static_pad (decoder, "sink");
        gst_pad_link (pad, sinkpad);
        gst_object_unref (sinkpad);
}

static gboolean cb_print_position (GstElement *pipeline)
{
        GstFormat fmt = GST_FORMAT_TIME;
        gint64 pos, len;
        if (gst_element_query_position (pipeline, &fmt, &pos) && gst_element_query_duration (pipeline, &fmt, &len))
        {
                g_print ("Time: %" GST_TIME_FORMAT " / %" GST_TIME_FORMAT "\r",
                GST_TIME_ARGS (pos), GST_TIME_ARGS (len));
        }
        /* call me again */
        return TRUE;
}

void SetRate(gint64 sec)
{
        GstFormat fmt = GST_FORMAT_TIME;
        gint64 pos;
        gdouble rate;
        if(sec == 0)
                rate = 1.0;
        else if(sec == 1)
                rate = 1.1;
        else if(sec == 2)
                rate = 1.2;
        else if(sec == 3)
                rate = 1.3;
        else if(sec == 4)
                rate = 1.4;
        else if(sec == 5)
                rate = 1.5;
        gst_element_query_position(GST_ELEMENT(pipeline), &fmt, &pos);
        GST_TIME_ARGS (pos);

        gst_element_seek (pipeline, rate, GST_FORMAT_TIME, GST_SEEK_FLAG_FLUSH,
                         GST_SEEK_TYPE_SET, pos,
                         GST_SEEK_TYPE_NONE, GST_CLOCK_TIME_NONE);
}

int main (int argc, char *argv[])
{
        GMainLoop *loop;
        GstElement *source, *demuxer, *decoder, *conv, *sink, *queue, *audioparser, *audioresample, *scaletempo;
        GstBus *bus;
        GstElement *bin;
        GstPad *ghostpad;
        GstElement *audioline;

        /* Initialisation */
        argc++;
        argv[1] = "D:\\chill.mp3";
        gst_init (&argc, &argv);
        loop = g_main_loop_new (NULL, FALSE);
        /* Check input arguments */
        if (argc != 2)
        {
                g_printerr ("Usage: %s <Ogg/Vorbis filename>\n", argv[0]);
                return -1;
        }

        /* Create gstreamer elements */
        pipeline = gst_pipeline_new ("audio-player");
        source = gst_element_factory_make ("filesrc", "file-source");
        demuxer = gst_element_factory_make ("typefind", "ogg-demuxer");
        queue = gst_element_factory_make("queue", "queue");
        audioparser = gst_element_factory_make("mp3parse","wav-parser");
        decoder = gst_element_factory_make ("ffdec_mp3", "vorbis-decoder");
        conv = gst_element_factory_make ("audioconvert", "converter");
        scaletempo = gst_element_factory_make("scaletempo", "scaletempo");
        audioresample = gst_element_factory_make("audioresample", "audioresample");
        sink = gst_element_factory_make ("autoaudiosink", "audio-output");
        bin = gst_bin_new("mybin");

        if (!pipeline || !source || !demuxer || !decoder || !conv || !sink)
        {
                g_printerr ("One element could not be created. Exiting.\n");
                return -1;
        }

        g_object_set (G_OBJECT (source), "location", argv[1], NULL);

        bus = gst_pipeline_get_bus (GST_PIPELINE (pipeline));
        gst_bus_add_watch (bus, bus_call, loop);
        gst_object_unref (bus);

        gst_bin_add(GST_BIN(pipeline), bin);
        gst_bin_add_many (GST_BIN(bin),
                source, demuxer, queue, audioparser, decoder, conv, audioresample, scaletempo, sink, NULL);

        gst_element_link_many (source, demuxer, queue, audioparser, decoder, conv, audioresample, scaletempo, sink, NULL);
        g_signal_connect (demuxer, "pad-added", G_CALLBACK (on_pad_added), decoder);

        ghostpad = gst_element_get_pad (scaletempo, "sink");
        gst_element_add_pad (pipeline, gst_ghost_pad_new ("sink", ghostpad));
        gst_object_unref (ghostpad);


        g_timeout_add (200, (GSourceFunc) cb_print_position, pipeline);
        g_print ("Now playing: %s\n", argv[1]);
        gst_element_set_state (pipeline, GST_STATE_PLAYING);
        /* Iterate */
        g_print ("Running...\n");
        while(GST_STATE(pipeline) != GST_STATE_PLAYING);
        SetRate(5);
        g_object_set (G_OBJECT (scaletempo), "search", 500, NULL);
        g_main_loop_run (loop);
        /* Out of the main loop, clean up nicely */
        g_print ("Returned, stopping playback\n");
        gst_element_set_state (pipeline, GST_STATE_NULL);
        g_print ("Deleting pipeline\n");
        gst_object_unref (GST_OBJECT (pipeline));
        return 0;
}