gstreamer/gst-libs/gst/audio/gstaudiosrc.h
Stefan Kost 1a2642a1d2 Fix broken GObject macros
Original commit message from CVS:
* ext/pango/gsttextrender.h:
* gst-libs/gst/audio/gstaudiosink.h:
* gst-libs/gst/audio/gstaudiosrc.h:
* gst-libs/gst/audio/gstbaseaudiosink.h:
* gst-libs/gst/audio/gstbaseaudiosrc.h:
* gst-libs/gst/audio/gstringbuffer.h:
* gst-libs/gst/rtp/gstbasertpdepayload.h:
* gst-libs/gst/rtp/gstbasertppayload.h:
* gst-libs/gst/video/gstvideofilter.h:
* gst-libs/gst/video/gstvideosink.h:
* gst/playback/gstplaybasebin.h:
* gst/tcp/gstmultifdsink.h:
* sys/v4l/gstv4lelement.h:
Fix broken GObject macros
2006-04-08 18:09:17 +00:00

94 lines
3.2 KiB
C

/* GStreamer
* Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
* 2005 Wim Taymans <wim@fluendo.com>
*
* gstaudiosrc.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.
*/
/* a base class for simple audio srcs.
*
* This base class only requires subclasses to implement a set
* of simple functions.
*
* - open: open the device with the specified caps
* - read: read samples to the audio device
* - close: close the device
* - delay: the number of samples queued in the device
* - reset: unblock a read to the device and reset.
*
* All scheduling of samples and timestamps is done in this
* base class.
*/
#ifndef __GST_AUDIO_SRC_H__
#define __GST_AUDIO_SRC_H__
#include <gst/gst.h>
#include <gst/audio/gstbaseaudiosrc.h>
G_BEGIN_DECLS
#define GST_TYPE_AUDIO_SRC (gst_audio_src_get_type())
#define GST_AUDIO_SRC(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIO_SRC,GstAudioSrc))
#define GST_AUDIO_SRC_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_AUDIO_SRC,GstAudioSrcClass))
#define GST_AUDIO_SRC_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj),GST_TYPE_AUDIO_SRC,GstAudioSrcClass))
#define GST_IS_AUDIO_SRC(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIO_SRC))
#define GST_IS_AUDIO_SRC_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_AUDIO_SRC))
typedef struct _GstAudioSrc GstAudioSrc;
typedef struct _GstAudioSrcClass GstAudioSrcClass;
struct _GstAudioSrc {
GstBaseAudioSrc element;
/*< private >*/ /* with LOCK */
GThread *thread;
/*< private >*/
gpointer _gst_reserved[GST_PADDING];
};
struct _GstAudioSrcClass {
GstBaseAudioSrcClass parent_class;
/* vtable */
/* open the device with given specs */
gboolean (*open) (GstAudioSrc *src);
/* prepare resources and state to operate with the given specs */
gboolean (*prepare) (GstAudioSrc *src, GstRingBufferSpec *spec);
/* undo anything that was done in prepare() */
gboolean (*unprepare) (GstAudioSrc *src);
/* close the device */
gboolean (*close) (GstAudioSrc *src);
/* read samples from the device */
guint (*read) (GstAudioSrc *src, gpointer data, guint length);
/* get number of samples queued in the device */
guint (*delay) (GstAudioSrc *src);
/* reset the audio device, unblock from a write */
void (*reset) (GstAudioSrc *src);
/*< private >*/
gpointer _gst_reserved[GST_PADDING];
};
GType gst_audio_src_get_type(void);
G_END_DECLS
#endif /* __GST_AUDIO_SRC_H__ */