1 /* GStreamer
2  * Copyright (C) <2006> Philippe Khalaf <philippe.kalaf@collabora.co.uk>
3  *
4  * This library is free software; you can redistribute it and/or
5  * modify it under the terms of the GNU Library General Public
6  * License as published by the Free Software Foundation; either
7  * version 2 of the License, or (at your option) any later version.
8  *
9  * This library is distributed in the hope that it will be useful,
10  * but WITHOUT ANY WARRANTY; without even the implied warranty of
11  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
12  * Library General Public License for more details.
13  *
14  * You should have received a copy of the GNU Library General Public
15  * License along with this library; if not, write to the
16  * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
17  * Boston, MA 02110-1301, USA.
18  */
19 
20 #ifndef __GST_RTP_BASE_AUDIO_PAYLOAD_H__
21 #define __GST_RTP_BASE_AUDIO_PAYLOAD_H__
22 
23 #include <gst/gst.h>
24 #include <gst/rtp/gstrtpbasepayload.h>
25 #include <gst/base/gstadapter.h>
26 
27 G_BEGIN_DECLS
28 
29 typedef struct _GstRTPBaseAudioPayload GstRTPBaseAudioPayload;
30 typedef struct _GstRTPBaseAudioPayloadClass GstRTPBaseAudioPayloadClass;
31 
32 typedef struct _GstRTPBaseAudioPayloadPrivate GstRTPBaseAudioPayloadPrivate;
33 
34 #define GST_TYPE_RTP_BASE_AUDIO_PAYLOAD \
35   (gst_rtp_base_audio_payload_get_type())
36 #define GST_RTP_BASE_AUDIO_PAYLOAD(obj) \
37   (G_TYPE_CHECK_INSTANCE_CAST((obj), \
38   GST_TYPE_RTP_BASE_AUDIO_PAYLOAD,GstRTPBaseAudioPayload))
39 #define GST_RTP_BASE_AUDIO_PAYLOAD_CLASS(klass) \
40   (G_TYPE_CHECK_CLASS_CAST((klass), \
41   GST_TYPE_RTP_BASE_AUDIO_PAYLOAD,GstRTPBaseAudioPayloadClass))
42 #define GST_IS_RTP_BASE_AUDIO_PAYLOAD(obj) \
43   (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTP_BASE_AUDIO_PAYLOAD))
44 #define GST_IS_RTP_BASE_AUDIO_PAYLOAD_CLASS(klass) \
45   (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RTP_BASE_AUDIO_PAYLOAD))
46 #define GST_RTP_BASE_AUDIO_PAYLOAD_CAST(obj) \
47   ((GstRTPBaseAudioPayload *) (obj))
48 
49 struct _GstRTPBaseAudioPayload
50 {
51   GstRTPBasePayload payload;
52 
53   GstRTPBaseAudioPayloadPrivate *priv;
54 
55   GstClockTime base_ts;
56   gint frame_size;
57   gint frame_duration;
58 
59   gint sample_size;
60 
61   /*< private >*/
62   gpointer _gst_reserved[GST_PADDING];
63 };
64 
65 /**
66  * GstRTPBaseAudioPayloadClass:
67  * @parent_class: the parent class
68  *
69  * Base class for audio RTP payloader.
70  */
71 struct _GstRTPBaseAudioPayloadClass
72 {
73   GstRTPBasePayloadClass parent_class;
74 
75   /*< private >*/
76   gpointer _gst_reserved[GST_PADDING];
77 };
78 
79 GST_RTP_API
80 GType gst_rtp_base_audio_payload_get_type (void);
81 
82 /* configure frame based */
83 
84 GST_RTP_API
85 void            gst_rtp_base_audio_payload_set_frame_based        (GstRTPBaseAudioPayload *rtpbaseaudiopayload);
86 
87 GST_RTP_API
88 void            gst_rtp_base_audio_payload_set_frame_options      (GstRTPBaseAudioPayload *rtpbaseaudiopayload,
89                                                                    gint frame_duration, gint frame_size);
90 
91 /* configure sample based */
92 
93 GST_RTP_API
94 void            gst_rtp_base_audio_payload_set_sample_based       (GstRTPBaseAudioPayload *rtpbaseaudiopayload);
95 
96 GST_RTP_API
97 void            gst_rtp_base_audio_payload_set_sample_options     (GstRTPBaseAudioPayload *rtpbaseaudiopayload,
98                                                                    gint sample_size);
99 
100 GST_RTP_API
101 void            gst_rtp_base_audio_payload_set_samplebits_options (GstRTPBaseAudioPayload *rtpbaseaudiopayload,
102                                                                    gint sample_size);
103 
104 /* get the internal adapter */
105 
106 GST_RTP_API
107 GstAdapter*     gst_rtp_base_audio_payload_get_adapter            (GstRTPBaseAudioPayload *rtpbaseaudiopayload);
108 
109 /* push and flushing data */
110 
111 GST_RTP_API
112 GstFlowReturn   gst_rtp_base_audio_payload_push                   (GstRTPBaseAudioPayload * baseaudiopayload,
113                                                                    const guint8 * data, guint payload_len,
114                                                                    GstClockTime timestamp);
115 
116 GST_RTP_API
117 GstFlowReturn   gst_rtp_base_audio_payload_flush                  (GstRTPBaseAudioPayload * baseaudiopayload,
118                                                                    guint payload_len, GstClockTime timestamp);
119 
120 #ifdef G_DEFINE_AUTOPTR_CLEANUP_FUNC
121 G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstRTPBaseAudioPayload, gst_object_unref)
122 #endif
123 
124 G_END_DECLS
125 
126 #endif /* __GST_RTP_BASE_AUDIO_PAYLOAD_H__ */
127