gstreamer/gst-libs/gst/audio/gstbaseaudiosrc.h
Sebastian Dröge 04d9ff9a24 gst-libs/gst/audio/: API: Add GST_TYPE_BASE_AUDIO_(SRC|SINK)_SLAVE_METHOD to the public API. This is needed for the C...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_slave_method_get_type),
(gst_base_audio_sink_class_init):
* gst-libs/gst/audio/gstbaseaudiosink.h:
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_slave_method_get_type),
(gst_base_audio_src_class_init):
* gst-libs/gst/audio/gstbaseaudiosrc.h:
API: Add GST_TYPE_BASE_AUDIO_(SRC|SINK)_SLAVE_METHOD to the
public API. This is needed for the C++ bindings to be able
to use this base classes. Fixes bug #564200, #564206.
2008-12-13 06:57:09 +00:00

145 lines
4.6 KiB
C

/* GStreamer
* Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
* 2005 Wim Taymans <wim@fluendo.com>
*
* gstbaseaudiosrc.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 audio sources.
*/
#ifndef __GST_BASE_AUDIO_SRC_H__
#define __GST_BASE_AUDIO_SRC_H__
#include <gst/gst.h>
#include <gst/base/gstpushsrc.h>
#include "gstringbuffer.h"
#include "gstaudioclock.h"
G_BEGIN_DECLS
#define GST_TYPE_BASE_AUDIO_SRC (gst_base_audio_src_get_type())
#define GST_BASE_AUDIO_SRC(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_BASE_AUDIO_SRC,GstBaseAudioSrc))
#define GST_BASE_AUDIO_SRC_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_BASE_AUDIO_SRC,GstBaseAudioSrcClass))
#define GST_BASE_AUDIO_SRC_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_BASE_AUDIO_SRC, GstBaseAudioSrcClass))
#define GST_IS_BASE_AUDIO_SRC(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_BASE_AUDIO_SRC))
#define GST_IS_BASE_AUDIO_SRC_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_BASE_AUDIO_SRC))
/**
* GST_BASE_AUDIO_SRC_CLOCK:
* @obj: a #GstBaseAudioSrc
*
* Get the #GstClock of @obj.
*/
#define GST_BASE_AUDIO_SRC_CLOCK(obj) (GST_BASE_AUDIO_SRC (obj)->clock)
/**
* GST_BASE_AUDIO_SRC_PAD:
* @obj: a #GstBaseAudioSrc
*
* Get the source #GstPad of @obj.
*/
#define GST_BASE_AUDIO_SRC_PAD(obj) (GST_BASE_SRC (obj)->srcpad)
typedef struct _GstBaseAudioSrc GstBaseAudioSrc;
typedef struct _GstBaseAudioSrcClass GstBaseAudioSrcClass;
typedef struct _GstBaseAudioSrcPrivate GstBaseAudioSrcPrivate;
/**
* GstBaseAudioSrcSlaveMethod:
* @GST_BASE_AUDIO_SRC_SLAVE_RESAMPLE: Resample to match the master clock.
* @GST_BASE_AUDIO_SRC_SLAVE_RETIMESTAMP: Retimestamp output buffers with master
* clock time.
* @GST_BASE_AUDIO_SRC_SLAVE_SKEW: Adjust capture pointer when master clock
* drifts too much.
* @GST_BASE_AUDIO_SRC_SLAVE_NONE: No adjustment is done.
*
* Different possible clock slaving algorithms when the internal audio clock was
* not selected as the pipeline clock.
*/
typedef enum
{
GST_BASE_AUDIO_SRC_SLAVE_RESAMPLE,
GST_BASE_AUDIO_SRC_SLAVE_RETIMESTAMP,
GST_BASE_AUDIO_SRC_SLAVE_SKEW,
GST_BASE_AUDIO_SRC_SLAVE_NONE
} GstBaseAudioSrcSlaveMethod;
#define GST_TYPE_BASE_AUDIO_SRC_SLAVE_METHOD (gst_base_audio_src_slave_method_get_type ())
/**
* GstBaseAudioSrc:
*
* Opaque #GstBaseAudioSrc.
*/
struct _GstBaseAudioSrc {
GstPushSrc element;
/*< protected >*/ /* with LOCK */
/* our ringbuffer */
GstRingBuffer *ringbuffer;
/* required buffer and latency */
GstClockTime buffer_time;
GstClockTime latency_time;
/* the next sample to write */
guint64 next_sample;
/* clock */
GstClock *clock;
/*< private >*/
GstBaseAudioSrcPrivate *priv;
gpointer _gst_reserved[GST_PADDING - 1];
};
/**
* GstBaseAudioSrcClass:
* @parent_class: the parent class.
* @create_ringbuffer: create and return a #GstRingBuffer to read from.
*
* #GstBaseAudioSrc class. Override the vmethod to implement
* functionality.
*/
struct _GstBaseAudioSrcClass {
GstPushSrcClass parent_class;
/* subclass ringbuffer allocation */
GstRingBuffer* (*create_ringbuffer) (GstBaseAudioSrc *src);
/*< private >*/
gpointer _gst_reserved[GST_PADDING];
};
GType gst_base_audio_src_get_type(void);
GstRingBuffer *gst_base_audio_src_create_ringbuffer (GstBaseAudioSrc *src);
void gst_base_audio_src_set_provide_clock (GstBaseAudioSrc *src, gboolean provide);
gboolean gst_base_audio_src_get_provide_clock (GstBaseAudioSrc *src);
void gst_base_audio_src_set_slave_method (GstBaseAudioSrc *src,
GstBaseAudioSrcSlaveMethod method);
GstBaseAudioSrcSlaveMethod
gst_base_audio_src_get_slave_method (GstBaseAudioSrc *src);
G_END_DECLS
#endif /* __GST_BASE_AUDIO_SRC_H__ */