/* GStreamer

 * Copyright (C) <2006> Philippe Khalaf <philippe.kalaf@collabora.co.uk>

 *

 * 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.

 */

#ifndef __GST_BASE_RTP_AUDIO_PAYLOAD_H__

#define __GST_BASE_RTP_AUDIO_PAYLOAD_H__

#include <gst/gst.h>

#include <gst/rtp/gstbasertppayload.h>

#include <gst/base/gstadapter.h>

G_BEGIN_DECLS

typedef struct _GstBaseRTPAudioPayload GstBaseRTPAudioPayload;

typedef struct _GstBaseRTPAudioPayloadClass GstBaseRTPAudioPayloadClass;

typedef struct _GstBaseRTPAudioPayloadPrivate GstBaseRTPAudioPayloadPrivate;

#define GST_TYPE_BASE_RTP_AUDIO_PAYLOAD \

  (gst_base_rtp_audio_payload_get_type())

#define GST_BASE_RTP_AUDIO_PAYLOAD(obj) \

  (G_TYPE_CHECK_INSTANCE_CAST((obj), \

  GST_TYPE_BASE_RTP_AUDIO_PAYLOAD,GstBaseRTPAudioPayload))

#define GST_BASE_RTP_AUDIO_PAYLOAD_CLASS(klass) \

  (G_TYPE_CHECK_CLASS_CAST((klass), \

  GST_TYPE_BASE_RTP_AUDIO_PAYLOAD,GstBaseRTPAudioPayloadClass))

#define GST_IS_BASE_RTP_AUDIO_PAYLOAD(obj) \

  (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_BASE_RTP_AUDIO_PAYLOAD))

#define GST_IS_BASE_RTP_AUDIO_PAYLOAD_CLASS(klass) \

  (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_BASE_RTP_AUDIO_PAYLOAD))

struct _GstBaseRTPAudioPayload

{

  GstBaseRTPPayload payload;

  GstBaseRTPAudioPayloadPrivate *priv;

  GstClockTime base_ts;

  gint frame_size;

  gint frame_duration;

  gint sample_size;

  gpointer _gst_reserved[GST_PADDING];

};

struct _GstBaseRTPAudioPayloadClass

{

  GstBaseRTPPayloadClass parent_class;

  gpointer _gst_reserved[GST_PADDING];

};

#ifdef __SYMBIAN32__

IMPORT_C

#endif

GType gst_base_rtp_audio_payload_get_type (void);

#ifdef __SYMBIAN32__

IMPORT_C

#endif

void

gst_base_rtp_audio_payload_set_frame_based (GstBaseRTPAudioPayload

    *basertpaudiopayload);

#ifdef __SYMBIAN32__

IMPORT_C

#endif

void

gst_base_rtp_audio_payload_set_sample_based (GstBaseRTPAudioPayload

    *basertpaudiopayload);

#ifdef __SYMBIAN32__

IMPORT_C

#endif

void

gst_base_rtp_audio_payload_set_frame_options (GstBaseRTPAudioPayload

    *basertpaudiopayload, gint frame_duration, gint frame_size);

#ifdef __SYMBIAN32__

IMPORT_C

#endif

void

gst_base_rtp_audio_payload_set_sample_options (GstBaseRTPAudioPayload

    *basertpaudiopayload, gint sample_size);

#ifdef __SYMBIAN32__

IMPORT_C

#endif

void

gst_base_rtp_audio_payload_set_samplebits_options (GstBaseRTPAudioPayload

    *basertpaudiopayload, gint sample_size);

#ifdef __SYMBIAN32__

IMPORT_C

#endif

GstFlowReturn

gst_base_rtp_audio_payload_push (GstBaseRTPAudioPayload * baseaudiopayload, 

    const guint8 * data, guint payload_len, GstClockTime timestamp);

#ifdef __SYMBIAN32__

IMPORT_C

#endif

GstAdapter*

gst_base_rtp_audio_payload_get_adapter (GstBaseRTPAudioPayload 

    *basertpaudiopayload);

G_END_DECLS

#endif /* __GST_BASE_RTP_AUDIO_PAYLOAD_H__ */