/* 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];

};

#ifdef __SYMBIAN32__

IMPORT_C

#endif

GType gst_base_audio_src_get_type(void);

#ifdef __SYMBIAN32__

IMPORT_C

#endif

GType gst_base_audio_src_slave_method_get_type (void);

#ifdef __SYMBIAN32__

IMPORT_C

#endif

GstRingBuffer *gst_base_audio_src_create_ringbuffer (GstBaseAudioSrc *src);

#ifdef __SYMBIAN32__

IMPORT_C

#endif

void       gst_base_audio_src_set_provide_clock        (GstBaseAudioSrc *src, gboolean provide);

#ifdef __SYMBIAN32__

IMPORT_C

#endif

gboolean   gst_base_audio_src_get_provide_clock        (GstBaseAudioSrc *src);

#ifdef __SYMBIAN32__

IMPORT_C

#endif

void       gst_base_audio_src_set_slave_method         (GstBaseAudioSrc *src,

                                                        GstBaseAudioSrcSlaveMethod method);

#ifdef __SYMBIAN32__

IMPORT_C

#endif

GstBaseAudioSrcSlaveMethod

           gst_base_audio_src_get_slave_method         (GstBaseAudioSrc *src);

G_END_DECLS

#endif /* __GST_BASE_AUDIO_SRC_H__ */