Gstreamer FLAC pipeline creation Error - c++

Question

I keep getting this error:
$ ./test recit24bit.flac
Now playing: recit24bit.flac
Running...
Error: Internal data flow error.
Returned, stopping playback
Deleting pipeline
When compiling this code:
#include <gst/gst.h>
#include <glib.h>
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);
}
*/
int main (int argc,
char *argv[])
{
GMainLoop *loop;
GstElement *pipeline,
*source,
//*demuxer,
*decoder,
*conv,
*sink;
GstBus *bus;
guint bus_watch_id;
/* Initialisation */
gst_init (&argc, &argv);
loop = g_main_loop_new (NULL, FALSE);
/* Check input arguments */
if (argc != 2)
{
g_printerr ("Usage: %s <Flac 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 ("oggdemux", "ogg-demuxer");
decoder = gst_element_factory_make ("flacdec", "flac-decoder");
conv = gst_element_factory_make ("audioconvert", "converter");
sink = gst_element_factory_make ("alsasink", "audio-output");
if (!pipeline || !source ||/* !demuxer ||*/ !decoder ||/* !conv ||*/ !sink)
{
g_printerr ("One element could not be created. Exiting.\n");
return -1;
}
/* Set up the pipeline */
/* we set the input filename to the source element */
g_object_set (G_OBJECT (source), "location", argv[1], NULL);
/* we add a message handler */
bus = gst_pipeline_get_bus (GST_PIPELINE (pipeline));
bus_watch_id = gst_bus_add_watch (bus, bus_call, loop);
gst_object_unref (bus);
/* we add all elements into the pipeline */
/* file-source | ogg-demuxer | vorbis-decoder | converter | alsa-output */
gst_bin_add_many (GST_BIN (pipeline), source,/* demuxer,*/ decoder, conv, sink, NULL);
/* we link the elements together */
/* file-source -> ogg-demuxer ~> vorbis-decoder -> converter -> alsa-output */
//gst_element_link (source, demuxer);
gst_element_link_many (source, decoder, conv, sink, NULL);
// g_signal_connect (demuxer, "pad-added", G_CALLBACK (on_pad_added), decoder);
/* note that the demuxer will be linked to the decoder dynamically.
The reason is that Ogg may contain various streams (for example
audio and video). The source pad(s) will be created at run time,
by the demuxer when it detects the amount and nature of streams.
Therefore we connect a callback function which will be executed
when the "pad-added" is emitted.*/
/* Set the pipeline to "playing" state*/
g_print ("Now playing: %s\n", argv[1]);
gst_element_set_state (pipeline, GST_STATE_PLAYING);
/* Iterate */
g_print ("Running...\n");
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));
g_source_remove (bus_watch_id);
g_main_loop_unref (loop);
return 0;
I'm using this to compile it successfully:
g++ -Wall test-flac.cc -o test $(pkg-config --cflags --libs gstreamer-1.0)
I'm using Arch, if that means anything. Does anybody have some advice? I'm a pretty big noob, but I don't understand what I'm not doing right because it seems like it should work.

Answer 1

I just needed to replace the demuxer with a parsar, which is (apparently) necessary. Derp. I used flacparse, of course.