--- /dev/null
+/* GStreamer
+ * Copyright (C) 2006 Wim Taymans <wim@fluendo.com>
+ *
+ * gstjack.h:
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, USA.
+ */
+
+#ifndef _GST_JACK_H_
+#define _GST_JACK_H_
+
+
+/**
+ * GstJackConnect:
+ * @GST_JACK_CONNECT_NONE: Don't automatically connect to physical ports.
+ * In this mode, the element will accept any number of input channels and will
+ * create (but not connect) an output port for each channel.
+ * @GST_JACK_CONNECT_AUTO: In this mode, the element will try to connect each
+ * output port to a random physical jack input pin. The sink will
+ * expose the number of physical channels on its pad caps.
+ *
+ * Specify how the output ports will be connected.
+ */
+
+typedef enum {
+ GST_JACK_CONNECT_NONE,
+ GST_JACK_CONNECT_AUTO
+} GstJackConnect;
+
+typedef jack_default_audio_sample_t sample_t;
+
+#define GST_TYPE_JACK_CONNECT (gst_jack_connect_get_type())
+GType gst_jack_connect_get_type();
+
+#endif // _GST_JACK_H_
#include "gstjackaudiosrc.h"
#include "gstjackringbuffer.h"
-GST_DEBUG_CATEGORY_STATIC (gst_jackaudiosrc_debug);
+GST_DEBUG_CATEGORY_STATIC (gst_jack_audio_src_debug);
#define GST_CAT_DEFAULT gst_jackaudiosrc_debug
static gboolean
);
#define _do_init(bla) \
- GST_DEBUG_CATEGORY_INIT(gst_jackaudiosrc_debug, "jacksrc", 0, "jacksrc element");
+ GST_DEBUG_CATEGORY_INIT(gst_jack_audio_src_debug, "jacksrc", 0, "jacksrc element");
GST_BOILERPLATE_FULL (GstJackAudioSrc, gst_jackaudiosrc, GstBaseAudioSrc,
GST_TYPE_BASE_AUDIO_SRC, _do_init);
gstbaseaudiosrc_class = (GstBaseAudioSrcClass *) klass;
gobject_class->set_property =
- GST_DEBUG_FUNCPTR (gst_jackaudiosrc_set_property);
+ GST_DEBUG_FUNCPTR (gst_jack_audio_src_set_property);
gobject_class->get_property =
- GST_DEBUG_FUNCPTR (gst_jackaudiosrc_get_property);
+ GST_DEBUG_FUNCPTR (gst_jack_audio_src_get_property);
g_object_class_install_property (gobject_class, PROP_CONNECT,
g_param_spec_enum ("connect", "Connect",
"The Jack server to connect to (NULL = default)",
DEFAULT_PROP_SERVER, G_PARAM_READWRITE));
- gstbasesrc_class->get_caps = GST_DEBUG_FUNCPTR (gst_jackaudiosrc_getcaps);
+ gstbasesrc_class->get_caps = GST_DEBUG_FUNCPTR (gst_jack_audio_src_getcaps);
gstbaseaudiosrc_class->create_ringbuffer =
- GST_DEBUG_FUNCPTR (gst_jackaudiosrc_create_ringbuffer);
+ GST_DEBUG_FUNCPTR (gst_jack_audio_src_create_ringbuffer);
/* ref class from a thread-safe context to work around missing bit of
* thread-safety in GObject */
G_BEGIN_DECLS
-#define GST_TYPE_JACK_AUDIO_SRC (gst_jackaudiosrc_get_type())
+#define GST_TYPE_JACK_AUDIO_SRC (gst_jack_audio_src_get_type())
#define GST_JACK_AUDIO_SRC(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_JACK_AUDIO_SRC,GstJackAudioSrc))
#define GST_JACK_AUDIO_SRC_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_JACK_AUDIO_SRC,GstJackAudioSrcClass))
+#define GST_JACK_AUDIO_SRC_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj),GST_TYPE_JACK_AUDIO_SRC,GstJackAudioSrcClass))
#define GST_IS_JACK_AUDIO_SRC(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_JACK_AUDIO_SRC))
#define GST_IS_JACK_AUDIO_SRC_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_JACK_AUDIO_SRC))
GstBaseAudioSrcClass parent_class;
};
-GType gst_jackaudiosrc_get_type (void);
+GType gst_jack_audio_src_get_type (void);
G_END_DECLS