/* GStreamer

 * Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>

 *                    2005 Wim Taymans <wim@fluendo.com>

 *

 * gstbaseaudiosink.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 sinks.

 *

 * It uses a ringbuffer to schedule playback of samples. This makes

 * it very easy to drop or insert samples to align incoming

 * buffers to the exact playback timestamp.

 *

 * Subclasses must provide a ringbuffer pointing to either DMA

 * memory or regular memory. A subclass should also call a callback

 * function when it has played N segments in the buffer. The subclass

 * is free to use a thread to signal this callback, use EIO or any

 * other mechanism.

 *

 * The base class is able to operate in push or pull mode. The chain

 * mode will queue the samples in the ringbuffer as much as possible.

 * The available space is calculated in the callback function.

 *

 * The pull mode will pull_range() a new buffer of N samples with a

 * configurable latency. This allows for high-end real time

 * audio processing pipelines driven by the audiosink. The callback

 * function will be used to perform a pull_range() on the sinkpad.

 * The thread scheduling the callback can be a real-time thread.

 *

 * Subclasses must implement a GstRingBuffer in addition to overriding

 * the methods in GstBaseSink and this class.

 */

#ifndef __GST_BASE_AUDIO_SINK_H__

#define __GST_BASE_AUDIO_SINK_H__

#include <gst/gst.h>

#include <gst/base/gstbasesink.h>

#include "gstringbuffer.h"

#include "gstaudioclock.h"

G_BEGIN_DECLS

#define GST_TYPE_BASE_AUDIO_SINK                (gst_base_audio_sink_get_type())

#define GST_BASE_AUDIO_SINK(obj)                (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_BASE_AUDIO_SINK,GstBaseAudioSink))

#define GST_BASE_AUDIO_SINK_CLASS(klass)        (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_BASE_AUDIO_SINK,GstBaseAudioSinkClass))

#define GST_BASE_AUDIO_SINK_GET_CLASS(obj)      (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_BASE_AUDIO_SINK, GstBaseAudioSinkClass))

#define GST_IS_BASE_AUDIO_SINK(obj)             (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_BASE_AUDIO_SINK))

#define GST_IS_BASE_AUDIO_SINK_CLASS(klass)     (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_BASE_AUDIO_SINK))

/**

 * GST_BASE_AUDIO_SINK_CLOCK:

 * @obj: a #GstBaseAudioSink

 *

 * Get the #GstClock of @obj.

 */

#define GST_BASE_AUDIO_SINK_CLOCK(obj)   (GST_BASE_AUDIO_SINK (obj)->clock)

/**

 * GST_BASE_AUDIO_SINK_PAD:

 * @obj: a #GstBaseAudioSink

 *

 * Get the sink #GstPad of @obj.

 */

#define GST_BASE_AUDIO_SINK_PAD(obj)     (GST_BASE_SINK (obj)->sinkpad)

/**

 * GstBaseAudioSinkSlaveMethod:

 * @GST_BASE_AUDIO_SINK_SLAVE_RESAMPLE: Resample to match the master clock

 * @GST_BASE_AUDIO_SINK_SLAVE_SKEW: Adjust playout pointer when master clock

 * drifts too much.

 * @GST_BASE_AUDIO_SINK_SLAVE_NONE: No adjustment is done. 

 *

 */

typedef enum 

{

  GST_BASE_AUDIO_SINK_SLAVE_RESAMPLE,

  GST_BASE_AUDIO_SINK_SLAVE_SKEW,

  GST_BASE_AUDIO_SINK_SLAVE_NONE

} GstBaseAudioSinkSlaveMethod;

typedef struct _GstBaseAudioSink GstBaseAudioSink;

typedef struct _GstBaseAudioSinkClass GstBaseAudioSinkClass;

typedef struct _GstBaseAudioSinkPrivate GstBaseAudioSinkPrivate;

/**

 * GstBaseAudioSink:

 *

 * Opaque #GstBaseAudioSink.

 */

struct _GstBaseAudioSink {

  GstBaseSink    element;

  /*< protected >*/ /* with LOCK */

  /* our ringbuffer */

  GstRingBuffer *ringbuffer;

  /* required buffer and latency in microseconds */

  guint64        buffer_time;

  guint64        latency_time;

  /* the next sample to write */

  guint64        next_sample;

  /* clock */

  gboolean       provide_clock;

  GstClock      *provided_clock;

  /*< private >*/

  GstBaseAudioSinkPrivate *priv;

  gpointer _gst_reserved[GST_PADDING - 1];

};

/**

 * GstBaseAudioSinkClass:

 * @parent_class: the parent class.

 * @create_ringbuffer: create and return a #GstRingBuffer to write to.

 *

 * #GstBaseAudioSink class. Override the vmethod to implement

 * functionality.

 */

struct _GstBaseAudioSinkClass {

  GstBaseSinkClass parent_class;

  /* subclass ringbuffer allocation */

  GstRingBuffer* (*create_ringbuffer)  (GstBaseAudioSink *sink);

  /*< private >*/

  gpointer _gst_reserved[GST_PADDING];

};

#ifdef __SYMBIAN32__

IMPORT_C

#endif

GType gst_base_audio_sink_get_type(void);

#ifdef __SYMBIAN32__

IMPORT_C

#endif

GstRingBuffer *gst_base_audio_sink_create_ringbuffer (GstBaseAudioSink *sink);

#ifdef __SYMBIAN32__

IMPORT_C

#endif

void       gst_base_audio_sink_set_provide_clock      (GstBaseAudioSink *sink, gboolean provide);

#ifdef __SYMBIAN32__

IMPORT_C

#endif

gboolean   gst_base_audio_sink_get_provide_clock      (GstBaseAudioSink *sink);

#ifdef __SYMBIAN32__

IMPORT_C

#endif

void       gst_base_audio_sink_set_slave_method       (GstBaseAudioSink *sink, 

                                                       GstBaseAudioSinkSlaveMethod method);

#ifdef __SYMBIAN32__

IMPORT_C

#endif

GstBaseAudioSinkSlaveMethod

           gst_base_audio_sink_get_slave_method       (GstBaseAudioSink *sink);

G_END_DECLS

#endif /* __GST_BASE_AUDIO_SINK_H__ */