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/* GStreamer
* Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
* 2005 Wim Taymans <wim@fluendo.com>
*
* gstaudiobasesrc.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., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
/* a base class for audio sources.
*/
#ifndef __GST_AUDIO_AUDIO_H__
#include <gst/audio/audio.h>
#endif
#ifndef __GST_AUDIO_BASE_SRC_H__
#define __GST_AUDIO_BASE_SRC_H__
#include <gst/gst.h>
#include <gst/base/gstpushsrc.h>
G_BEGIN_DECLS
#define GST_TYPE_AUDIO_BASE_SRC (gst_audio_base_src_get_type())
#define GST_AUDIO_BASE_SRC(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIO_BASE_SRC,GstAudioBaseSrc))
#define GST_AUDIO_BASE_SRC_CAST(obj) ((GstAudioBaseSrc*)obj)
#define GST_AUDIO_BASE_SRC_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_AUDIO_BASE_SRC,GstAudioBaseSrcClass))
#define GST_AUDIO_BASE_SRC_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_AUDIO_BASE_SRC, GstAudioBaseSrcClass))
#define GST_IS_AUDIO_BASE_SRC(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIO_BASE_SRC))
#define GST_IS_AUDIO_BASE_SRC_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_AUDIO_BASE_SRC))
/**
* GST_AUDIO_BASE_SRC_CLOCK:
* @obj: a #GstAudioBaseSrc
*
* Get the #GstClock of @obj.
*/
#define GST_AUDIO_BASE_SRC_CLOCK(obj) (GST_AUDIO_BASE_SRC (obj)->clock)
/**
* GST_AUDIO_BASE_SRC_PAD:
* @obj: a #GstAudioBaseSrc
*
* Get the source #GstPad of @obj.
*/
#define GST_AUDIO_BASE_SRC_PAD(obj) (GST_BASE_SRC (obj)->srcpad)
typedef struct _GstAudioBaseSrc GstAudioBaseSrc;
typedef struct _GstAudioBaseSrcClass GstAudioBaseSrcClass;
typedef struct _GstAudioBaseSrcPrivate GstAudioBaseSrcPrivate;
/* FIXME 2.0: Should be "retimestamp" not "re-timestamp" */
/**
* GstAudioBaseSrcSlaveMethod:
* @GST_AUDIO_BASE_SRC_SLAVE_RESAMPLE: Resample to match the master clock.
* @GST_AUDIO_BASE_SRC_SLAVE_RE_TIMESTAMP: Retimestamp output buffers with master
* clock time.
* @GST_AUDIO_BASE_SRC_SLAVE_SKEW: Adjust capture pointer when master clock
* drifts too much.
* @GST_AUDIO_BASE_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_AUDIO_BASE_SRC_SLAVE_RESAMPLE,
GST_AUDIO_BASE_SRC_SLAVE_RE_TIMESTAMP,
GST_AUDIO_BASE_SRC_SLAVE_SKEW,
GST_AUDIO_BASE_SRC_SLAVE_NONE
} GstAudioBaseSrcSlaveMethod;
#define GST_AUDIO_BASE_SRC_SLAVE_RETIMESTAMP GST_AUDIO_BASE_SRC_SLAVE_RE_TIMESTAMP
/**
* GstAudioBaseSrc:
*
* Opaque #GstAudioBaseSrc.
*/
struct _GstAudioBaseSrc {
GstPushSrc element;
/*< protected >*/ /* with LOCK */
/* our ringbuffer */
GstAudioRingBuffer *ringbuffer;
/* required buffer and latency */
GstClockTime buffer_time;
GstClockTime latency_time;
/* the next sample to write */
guint64 next_sample;
/* clock */
GstClock *clock;
/*< private >*/
GstAudioBaseSrcPrivate *priv;
gpointer _gst_reserved[GST_PADDING];
};
/**
* GstAudioBaseSrcClass:
* @parent_class: the parent class.
* @create_ringbuffer: create and return a #GstAudioRingBuffer to read from.
*
* #GstAudioBaseSrc class. Override the vmethod to implement
* functionality.
*/
struct _GstAudioBaseSrcClass {
GstPushSrcClass parent_class;
/* subclass ringbuffer allocation */
GstAudioRingBuffer* (*create_ringbuffer) (GstAudioBaseSrc *src);
/*< private >*/
gpointer _gst_reserved[GST_PADDING];
};
GST_AUDIO_API
GType gst_audio_base_src_get_type(void);
GST_AUDIO_API
GstAudioRingBuffer *
gst_audio_base_src_create_ringbuffer (GstAudioBaseSrc *src);
GST_AUDIO_API
void gst_audio_base_src_set_provide_clock (GstAudioBaseSrc *src, gboolean provide);
GST_AUDIO_API
gboolean gst_audio_base_src_get_provide_clock (GstAudioBaseSrc *src);
GST_AUDIO_API
void gst_audio_base_src_set_slave_method (GstAudioBaseSrc *src,
GstAudioBaseSrcSlaveMethod method);
GST_AUDIO_API
GstAudioBaseSrcSlaveMethod
gst_audio_base_src_get_slave_method (GstAudioBaseSrc *src);
G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstAudioBaseSrc, gst_object_unref)
G_END_DECLS
#endif /* __GST_AUDIO_BASE_SRC_H__ */
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