/* 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);

#ifdef G_DEFINE_AUTOPTR_CLEANUP_FUNC

G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstAudioBaseSrc, gst_object_unref)

#endif

G_END_DECLS

#endif /* __GST_AUDIO_BASE_SRC_H__ */