gnunet_gst_test.c File Reference

FIXME. More...

#include "platform.h"
#include "gnunet_gst_def.h"
#include "gnunet_gst.h"

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Functions
int	main (int argc, char *argv[])

Detailed Description

FIXME.

Author

Hark

Definition in file gnunet_gst_test.c.

Function Documentation

main()

int main	(	int	argc,
		char *	argv[]
	)

Definition at line 31 of file gnunet_gst_test.c.

{
  struct GNUNET_gstData *gst;
  // GstBus *bus;
  GstElement *gnunetsrc, *gnunetsink, *source, *sink, *encoder, *decoder;
 
 
  // audio_message = GNUNET_malloc (UINT16_MAX);
  // audio_message->header.type = htons (GNUNET_MESSAGE_TYPE_CONVERSATION_AUDIO);
 
 
  // GstPipeline *pipeline;
 
  gst = (GNUNET_gstData *) malloc (sizeof(struct GNUNET_gstData));
 
  // gst->audio_message.header.type = htons (GNUNET_MESSAGE_TYPE_CONVERSATION_AUDIO);
 
 
  gg_load_configuration (gst);
/*
   gst->audiobackend = JACK;
   gst->dropsilence = TRUE;
   gst->usertp = FALSE;
 *//* Initialize GStreamer */gst_init (&argc, &argv);
 
  gst->pipeline = GST_PIPELINE (gst_pipeline_new ("gnunet-media-helper"));
 
#ifdef IS_SPEAKER
  int type = SPEAKER;
  printf ("this is the speaker \n");
#endif
#ifdef IS_MIC
  int type = MICROPHONE;
  printf ("this is the microphone \n");
#endif
  if (type == SPEAKER)
  {
    gnunetsrc = GST_ELEMENT (get_app (gst, SOURCE));
 
    sink = GST_ELEMENT (get_audiobin (gst, SINK));
    decoder = GST_ELEMENT (get_coder (gst, DECODER));
    gst_bin_add_many (GST_BIN (gst->pipeline), gnunetsrc, decoder, sink, NULL);
    gst_element_link_many (gnunetsrc, decoder, sink, NULL);
  }
  if (type == MICROPHONE)
  {
    source = GST_ELEMENT (get_audiobin (gst, SOURCE));
 
    encoder = GST_ELEMENT (get_coder (gst, ENCODER));
 
    gnunetsink = GST_ELEMENT (get_app (gst, SINK));
 
    gst_bin_add_many (GST_BIN (gst->pipeline), source, encoder, gnunetsink,
                      NULL);
    gst_element_link_many (source, encoder, gnunetsink, NULL);
  }
  /*
     gst_bin_add_many( GST_BIN(gst->pipeline), appsource, appsink, source, encoder, decoder, sink, NULL);
     gst_element_link_many( source, encoder, decoder, sink , NULL);
   */
  pl_graph (gst->pipeline);
  /* Start playing */
  gst_element_set_state (GST_ELEMENT (gst->pipeline), GST_STATE_PLAYING);
 
  // pl_graph(gst->pipeline);
 
  /* Wait until error or EOS */
  // bus = gst_element_get_bus (GST_ELEMENT(gst->pipeline));
  // bus_watch_id = gst_bus_add_watch (bus, gnunet_gst_bus_call, pipeline);
 
  gg_setup_gst_bus (gst);
// g_print ("Running...\n");
 
 
  // start pushing buffers
  if (type == MICROPHONE)
  {
    GMainLoop *loop;
    loop = g_main_loop_new (NULL, FALSE);
 
    g_main_loop_run (loop);
 
/*
   while ( 1 )
     {
         GstFlowReturn flow;
         flow = on_appsink_new_sample (gst->appsink, gst);
    }
 */}
  if (type == SPEAKER)
  {
    while (1)
    {
//      printf("read.. \n");
      gnunet_read (gst);
    }
  }
  g_print ("Returned, stopping playback\n");
 
  // gst_object_unref (bus);
  gst_element_set_state (GST_ELEMENT (gst->pipeline), GST_STATE_NULL);
  gst_object_unref (gst->pipeline);
 
  return 0;
}

References decoder, DECODER, ENCODER, get_app(), get_audiobin(), get_coder(), gg_load_configuration(), gg_setup_gst_bus(), gnunet_read(), MICROPHONE, GNUNET_gstData::pipeline, pl_graph(), sink, SINK, source, SOURCE, SPEAKER, and type.

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