/usr/include/gstreamer-0.10/gst/rtp/gstbasertpaudiopayload.h is in libgstreamer-plugins-base0.10-dev 0.10.36-1ubuntu0.2.
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* 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))
#define GST_BASE_RTP_AUDIO_PAYLOAD_CAST(obj) \
((GstBaseRTPAudioPayload *) (obj))
struct _GstBaseRTPAudioPayload
{
GstBaseRTPPayload payload;
GstBaseRTPAudioPayloadPrivate *priv;
GstClockTime base_ts;
gint frame_size;
gint frame_duration;
gint sample_size;
gpointer _gst_reserved[GST_PADDING];
};
/**
* GstBaseRTPAudioPayloadClass:
* @parent_class: the parent class
*
* Base class for audio RTP payloader.
*/
struct _GstBaseRTPAudioPayloadClass
{
GstBaseRTPPayloadClass parent_class;
/*< private >*/
gpointer _gst_reserved[GST_PADDING];
};
GType gst_base_rtp_audio_payload_get_type (void);
/* configure frame based */
void gst_base_rtp_audio_payload_set_frame_based (GstBaseRTPAudioPayload *basertpaudiopayload);
void gst_base_rtp_audio_payload_set_frame_options (GstBaseRTPAudioPayload *basertpaudiopayload,
gint frame_duration, gint frame_size);
/* configure sample based */
void gst_base_rtp_audio_payload_set_sample_based (GstBaseRTPAudioPayload *basertpaudiopayload);
void gst_base_rtp_audio_payload_set_sample_options (GstBaseRTPAudioPayload *basertpaudiopayload,
gint sample_size);
void gst_base_rtp_audio_payload_set_samplebits_options (GstBaseRTPAudioPayload *basertpaudiopayload,
gint sample_size);
/* get the internal adapter */
GstAdapter* gst_base_rtp_audio_payload_get_adapter (GstBaseRTPAudioPayload *basertpaudiopayload);
/* push and flushing data */
GstFlowReturn gst_base_rtp_audio_payload_push (GstBaseRTPAudioPayload * baseaudiopayload,
const guint8 * data, guint payload_len,
GstClockTime timestamp);
GstFlowReturn gst_base_rtp_audio_payload_flush (GstBaseRTPAudioPayload * baseaudiopayload,
guint payload_len, GstClockTime timestamp);
G_END_DECLS
#endif /* __GST_BASE_RTP_AUDIO_PAYLOAD_H__ */
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