static void g_filter_input_stream_skip_async (GInputStream *stream, gsize count, int io_priority, GCancellable *cancellable, GAsyncReadyCallback callback, gpointer user_data) { GFilterInputStream *filter_stream; GInputStream *base_stream; filter_stream = G_FILTER_INPUT_STREAM (stream); base_stream = filter_stream->base_stream; g_input_stream_skip_async (base_stream, count, io_priority, cancellable, callback, user_data); }
int main (int argc, char **argv) { GMainLoop *loop = NULL; GstElement *pipeline = NULL; GstBus *bus = NULL; GstElement *conf = NULL; FsParticipant *part = NULL; GError *error = NULL; GInputStream *istream = NULL; gchar *send_socket, *recv_socket; TestSession *ses; gst_init (&argc, &argv); if (argc != 3) { g_print ("Usage: %s <send socket> <recv_socket>\n", argv[0]); return 1; } send_socket = argv[1]; recv_socket = argv[2]; if (unlink (send_socket) < 0 && errno != ENOENT) { g_print ("Could not delete send or recv sockets"); return 2; } g_print ("Press ENTER when the other side is ready\n"); loop = g_main_loop_new (NULL, FALSE); pipeline = gst_pipeline_new (NULL); bus = gst_pipeline_get_bus (GST_PIPELINE (pipeline)); gst_bus_add_watch (bus, async_bus_cb, pipeline); gst_object_unref (bus); conf = gst_element_factory_make ("fsrtpconference", NULL); g_assert (conf); part = fs_conference_new_participant (FS_CONFERENCE (conf), &error); print_error (error); g_assert (part); g_assert (gst_bin_add (GST_BIN (pipeline), conf)); istream = g_unix_input_stream_new (0, FALSE); ses = add_audio_session (pipeline, FS_CONFERENCE (conf), 1, part, send_socket, recv_socket); g_input_stream_skip_async (istream, 1, G_PRIORITY_DEFAULT, NULL, skipped_cb, ses); g_assert (gst_element_set_state (pipeline, GST_STATE_PLAYING) != GST_STATE_CHANGE_FAILURE); g_main_loop_run (loop); g_assert (gst_element_set_state (pipeline, GST_STATE_NULL) != GST_STATE_CHANGE_FAILURE); g_object_unref (part); g_object_unref (istream); free_session (ses); gst_object_unref (pipeline); g_main_loop_unref (loop); return 0; }