How to fix Gstreamer to capture microphone audio and buffer or dump as raw file, when i am talking it does not save anything

I am trying to capture the microphone audio and save it as a file. But its not working, i can only play the file while assign. How can i enable the microphone and buffer it or save or dump as raw .odd/vorbis ?

#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;

  /* Initialisation */
  gst_init (&argc, &argv);

  loop = g_main_loop_new (NULL, FALSE);


  /* Check input arguments */
  if (argc != 2) {
    g_printerr ("Usage: %s <开发者_开发百科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 ("oggdemux",      "ogg-demuxer");
  decoder  = gst_element_factory_make ("vorbisdec",     "vorbis-decoder");
  conv     = gst_element_factory_make ("audioconvert",  "converter");
  sink     = gst_element_factory_make ("autoaudiosink", "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));
  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 (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));

  return 0;
}

what's the question actually?

on linux with pulseaudio it's as simple as

$ gst-launch pulsesrc ! filesink location=dump.raw
$ gst-launch pulsesrc ! audioconvert ! vorbisenc ! oggmux ! filesink location=dump.ogg

you can also use the following pipeline :

gst-launch osssrc device=<mic i/p dev> ! audioconvert ! vorbisenc ! oggmux ! filesink location=dump.ogg