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authorChristian Grothoff <christian@grothoff.org>2019-10-05 15:09:28 +0200
committerChristian Grothoff <christian@grothoff.org>2019-10-05 15:09:28 +0200
commitc4e9ba925ffd758aaa3feee2ccfc0b76f26fe207 (patch)
treecac3ce030d77b4cbe7c7dc62ed58cfe6d24f73e1 /src/conversation/gnunet_gst_test.c
parentfbb71d527c7d6babf269a8fefce1db291b9f7068 (diff)
global reindent, now with uncrustify hook enabled
Diffstat (limited to 'src/conversation/gnunet_gst_test.c')
-rw-r--r--src/conversation/gnunet_gst_test.c87
1 files changed, 44 insertions, 43 deletions
diff --git a/src/conversation/gnunet_gst_test.c b/src/conversation/gnunet_gst_test.c
index 60e920f10..884b0fe83 100644
--- a/src/conversation/gnunet_gst_test.c
+++ b/src/conversation/gnunet_gst_test.c
@@ -27,7 +27,7 @@
#include "gnunet_gst.h"
int
-main(int argc, char *argv[])
+main (int argc, char *argv[])
{
struct GNUNET_gstData *gst;
// GstBus *bus;
@@ -36,80 +36,81 @@ main(int argc, char *argv[])
// audio_message = GNUNET_malloc (UINT16_MAX);
- //audio_message->header.type = htons (GNUNET_MESSAGE_TYPE_CONVERSATION_AUDIO);
+ // audio_message->header.type = htons (GNUNET_MESSAGE_TYPE_CONVERSATION_AUDIO);
- //GstPipeline *pipeline;
+ // GstPipeline *pipeline;
- gst = (GNUNET_gstData*)malloc(sizeof(struct GNUNET_gstData));
+ gst = (GNUNET_gstData*) malloc (sizeof(struct GNUNET_gstData));
- //gst->audio_message.header.type = htons (GNUNET_MESSAGE_TYPE_CONVERSATION_AUDIO);
+ // gst->audio_message.header.type = htons (GNUNET_MESSAGE_TYPE_CONVERSATION_AUDIO);
- gg_load_configuration(gst);
+ gg_load_configuration (gst);
/*
gst->audiobackend = JACK;
gst->dropsilence = TRUE;
gst->usertp = FALSE;
*/
/* Initialize GStreamer */
- gst_init(&argc, &argv);
+ gst_init (&argc, &argv);
- gst->pipeline = GST_PIPELINE(gst_pipeline_new("gnunet-media-helper"));
+ gst->pipeline = GST_PIPELINE (gst_pipeline_new ("gnunet-media-helper"));
#ifdef IS_SPEAKER
int type = SPEAKER;
- printf("this is the speaker \n");
+ printf ("this is the speaker \n");
#endif
#ifdef IS_MIC
int type = MICROPHONE;
- printf("this is the microphone \n");
+ 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);
- }
+ {
+ 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));
+ {
+ source = GST_ELEMENT (get_audiobin (gst, SOURCE));
- encoder = GST_ELEMENT(get_coder(gst, ENCODER));
+ encoder = GST_ELEMENT (get_coder (gst, ENCODER));
- gnunetsink = GST_ELEMENT(get_app(gst, SINK));
+ 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), 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);
+ pl_graph (gst->pipeline);
/* Start playing */
- gst_element_set_state(GST_ELEMENT(gst->pipeline), GST_STATE_PLAYING);
+ gst_element_set_state (GST_ELEMENT (gst->pipeline), GST_STATE_PLAYING);
- //pl_graph(gst->pipeline);
+ // 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);
+ // 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);
+ gg_setup_gst_bus (gst);
// g_print ("Running...\n");
// start pushing buffers
if (type == MICROPHONE)
- {
- GMainLoop *loop;
- loop = g_main_loop_new(NULL, FALSE);
+ {
+ GMainLoop *loop;
+ loop = g_main_loop_new (NULL, FALSE);
- g_main_loop_run(loop);
+ g_main_loop_run (loop);
/*
while ( 1 )
@@ -118,20 +119,20 @@ main(int argc, char *argv[])
flow = on_appsink_new_sample (gst->appsink, gst);
}
*/
- }
+ }
if (type == SPEAKER)
+ {
+ while (1)
{
- while (1)
- {
// printf("read.. \n");
- gnunet_read(gst);
- }
+ gnunet_read (gst);
}
- g_print("Returned, stopping playback\n");
+ }
+ 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);
+ gst_element_set_state (GST_ELEMENT (gst->pipeline), GST_STATE_NULL);
+ gst_object_unref (gst->pipeline);
return 0;
}