This file is indexed.

/usr/include/gstreamer-1.0/gst/rtp/gstrtpbaseaudiopayload.h is in libgstreamer-plugins-base1.0-dev 1.8.0-1ubuntu1.

This file is owned by root:root, with mode 0o644.

The actual contents of the file can be viewed below.

  1
  2
  3
  4
  5
  6
  7
  8
  9
 10
 11
 12
 13
 14
 15
 16
 17
 18
 19
 20
 21
 22
 23
 24
 25
 26
 27
 28
 29
 30
 31
 32
 33
 34
 35
 36
 37
 38
 39
 40
 41
 42
 43
 44
 45
 46
 47
 48
 49
 50
 51
 52
 53
 54
 55
 56
 57
 58
 59
 60
 61
 62
 63
 64
 65
 66
 67
 68
 69
 70
 71
 72
 73
 74
 75
 76
 77
 78
 79
 80
 81
 82
 83
 84
 85
 86
 87
 88
 89
 90
 91
 92
 93
 94
 95
 96
 97
 98
 99
100
101
102
103
104
105
106
107
108
109
110
/* 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., 51 Franklin St, Fifth Floor,
 * Boston, MA 02110-1301, USA.
 */

#ifndef __GST_RTP_BASE_AUDIO_PAYLOAD_H__
#define __GST_RTP_BASE_AUDIO_PAYLOAD_H__

#include <gst/gst.h>
#include <gst/rtp/gstrtpbasepayload.h>
#include <gst/base/gstadapter.h>

G_BEGIN_DECLS

typedef struct _GstRTPBaseAudioPayload GstRTPBaseAudioPayload;
typedef struct _GstRTPBaseAudioPayloadClass GstRTPBaseAudioPayloadClass;

typedef struct _GstRTPBaseAudioPayloadPrivate GstRTPBaseAudioPayloadPrivate;

#define GST_TYPE_RTP_BASE_AUDIO_PAYLOAD \
  (gst_rtp_base_audio_payload_get_type())
#define GST_RTP_BASE_AUDIO_PAYLOAD(obj) \
  (G_TYPE_CHECK_INSTANCE_CAST((obj), \
  GST_TYPE_RTP_BASE_AUDIO_PAYLOAD,GstRTPBaseAudioPayload))
#define GST_RTP_BASE_AUDIO_PAYLOAD_CLASS(klass) \
  (G_TYPE_CHECK_CLASS_CAST((klass), \
  GST_TYPE_RTP_BASE_AUDIO_PAYLOAD,GstRTPBaseAudioPayloadClass))
#define GST_IS_RTP_BASE_AUDIO_PAYLOAD(obj) \
  (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTP_BASE_AUDIO_PAYLOAD))
#define GST_IS_RTP_BASE_AUDIO_PAYLOAD_CLASS(klass) \
  (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RTP_BASE_AUDIO_PAYLOAD))
#define GST_RTP_BASE_AUDIO_PAYLOAD_CAST(obj) \
  ((GstRTPBaseAudioPayload *) (obj))

struct _GstRTPBaseAudioPayload
{
  GstRTPBasePayload payload;

  GstRTPBaseAudioPayloadPrivate *priv;

  GstClockTime base_ts;
  gint frame_size;
  gint frame_duration;

  gint sample_size;

  /*< private >*/
  gpointer _gst_reserved[GST_PADDING];
};

/**
 * GstRTPBaseAudioPayloadClass:
 * @parent_class: the parent class
 *
 * Base class for audio RTP payloader.
 */
struct _GstRTPBaseAudioPayloadClass
{
  GstRTPBasePayloadClass parent_class;

  /*< private >*/
  gpointer _gst_reserved[GST_PADDING];
};

GType gst_rtp_base_audio_payload_get_type (void);

/* configure frame based */
void            gst_rtp_base_audio_payload_set_frame_based        (GstRTPBaseAudioPayload *rtpbaseaudiopayload);

void            gst_rtp_base_audio_payload_set_frame_options      (GstRTPBaseAudioPayload *rtpbaseaudiopayload,
                                                                   gint frame_duration, gint frame_size);

/* configure sample based */
void            gst_rtp_base_audio_payload_set_sample_based       (GstRTPBaseAudioPayload *rtpbaseaudiopayload);
void            gst_rtp_base_audio_payload_set_sample_options     (GstRTPBaseAudioPayload *rtpbaseaudiopayload,
                                                                   gint sample_size);
void            gst_rtp_base_audio_payload_set_samplebits_options (GstRTPBaseAudioPayload *rtpbaseaudiopayload,
                                                                   gint sample_size);

/* get the internal adapter */
GstAdapter*     gst_rtp_base_audio_payload_get_adapter            (GstRTPBaseAudioPayload *rtpbaseaudiopayload);

/* push and flushing data */
GstFlowReturn   gst_rtp_base_audio_payload_push                   (GstRTPBaseAudioPayload * baseaudiopayload,
                                                                   const guint8 * data, guint payload_len,
                                                                   GstClockTime timestamp);
GstFlowReturn   gst_rtp_base_audio_payload_flush                  (GstRTPBaseAudioPayload * baseaudiopayload,
                                                                   guint payload_len, GstClockTime timestamp);

#ifdef G_DEFINE_AUTOPTR_CLEANUP_FUNC
G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstRTPBaseAudioPayload, gst_object_unref)
#endif

G_END_DECLS

#endif /* __GST_RTP_BASE_AUDIO_PAYLOAD_H__ */