/usr/include/gstreamer-1.0/gst/rtp/gstrtpbaseaudiopayload.h is in libgstreamer-plugins-base1.0-dev 1.2.4-1~ubuntu2.1.
This file is owned by root:root, with mode 0o644.
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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 | /* 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);
G_END_DECLS
#endif /* __GST_RTP_BASE_AUDIO_PAYLOAD_H__ */
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