experimental gstreamer audiobackend (not working)

1 files changed, 120 insertions, 0 deletions

diff --git a/src/conversation/gnunet_gst_test.c b/src/conversation/gnunet_gst_test.c
new file mode 100644
index 000000000..3e1454c5b
--- /dev/null
+++ b/src/conversation/gnunet_gst_test.c
@@ -0,0 +1,120 @@
+#include "gnunet_gst_def.h"
+#include "gnunet_gst.h"
+
+int
+main (int argc, char *argv[])
+{
+  struct GNUNET_gstData *gst;
+  GstBus *bus;
+  GstMessage *msg;
+  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;
+}