gstreamer/gst-libs/gst/rtp/gstbasertpaudiopayload.h
Stefan Kost cade791150 docs/libs/: add remaining symbols into correct setions
Original commit message from CVS:
* docs/libs/Makefile.am:
* docs/libs/gst-plugins-base-libs-docs.sgml:
* docs/libs/gst-plugins-base-libs-sections.txt:
* docs/libs/gst-plugins-base-libs.types:
add remaining symbols into correct setions
* gst-libs/gst/audio/gstringbuffer.c:
fix incomplete docs
* gst-libs/gst/audio/gstringbuffer.h:
comment out not yet implemented function
* gst-libs/gst/floatcast/floatcast.h:
* gst-libs/gst/netbuffer/gstnetbuffer.c:
add short descriptions
* gst-libs/gst/interfaces/propertyprobe.c:
fix return value docs
* gst-libs/gst/riff/riff-read.c: (gst_riff_read_chunk):
simplify debug logging
* gst-libs/gst/riff/riff-read.h:
sync function prototype and docs
* gst-libs/gst/rtp/gstbasertpaudiopayload.h:
remove left over symbol
2006-06-16 10:02:25 +00:00

92 lines
2.7 KiB
C

/* GStreamer
* Copyright (C) <2006> Philippe Khalaf <burger@speedy.org>
*
* 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>
G_BEGIN_DECLS
typedef struct _GstBaseRTPAudioPayload GstBaseRTPAudioPayload;
typedef struct _GstBaseRTPAudioPayloadClass GstBaseRTPAudioPayloadClass;
#define GST_TYPE_BASE_RTP_AUDIO_PAYLOAD \
(gst_basertpaudiopayload_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))
typedef enum {
AUDIO_CODEC_TYPE_NONE,
AUDIO_CODEC_TYPE_FRAME_BASED,
AUDIO_CODEC_TYPE_SAMPLE_BASED
} AudioCodecType;
struct _GstBaseRTPAudioPayload
{
GstBaseRTPPayload payload;
GstClockTime base_ts;
gint frame_size;
gint frame_duration;
gint sample_size;
AudioCodecType type;
gpointer _gst_reserved[GST_PADDING];
};
struct _GstBaseRTPAudioPayloadClass
{
GstBaseRTPPayloadClass parent_class;
gpointer _gst_reserved[GST_PADDING];
};
GType gst_basertpaudiopayload_get_type (void);
void
gst_basertpaudiopayload_set_frame_based (GstBaseRTPAudioPayload
*basertpaudiopayload);
void
gst_basertpaudiopayload_set_sample_based (GstBaseRTPAudioPayload
*basertpaudiopayload);
void
gst_basertpaudiopayload_set_frame_options (GstBaseRTPAudioPayload
*basertpaudiopayload, gint frame_duration, gint frame_size);
void
gst_basertpaudiopayload_set_sample_options (GstBaseRTPAudioPayload
*basertpaudiopayload, gint sample_size);
G_END_DECLS
#endif /* __GST_BASE_RTP_AUDIO_PAYLOAD_H__ */