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;
}

--
View this message in context: http://gstreamer-devel.966125.n4.nabble.com/scaletempo-again-tp4626422.html
Sent from the GStreamer-devel mailing list archive at Nabble.com.