1 | /* GStreamer |
2 | * Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu> |
3 | * 2005 Wim Taymans <wim@fluendo.com> |
4 | * |
5 | * gstaudiobasesrc.h: |
6 | * |
7 | * This library is free software; you can redistribute it and/or |
8 | * modify it under the terms of the GNU Library General Public |
9 | * License as published by the Free Software Foundation; either |
10 | * version 2 of the License, or (at your option) any later version. |
11 | * |
12 | * This library is distributed in the hope that it will be useful, |
13 | * but WITHOUT ANY WARRANTY; without even the implied warranty of |
14 | * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU |
15 | * Library General Public License for more details. |
16 | * |
17 | * You should have received a copy of the GNU Library General Public |
18 | * License along with this library; if not, write to the |
19 | * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor, |
20 | * Boston, MA 02110-1301, USA. |
21 | */ |
22 | |
23 | /* a base class for audio sources. |
24 | */ |
25 | |
26 | #ifndef __GST_AUDIO_AUDIO_H__ |
27 | #include <gst/audio/audio.h> |
28 | #endif |
29 | |
30 | #ifndef __GST_AUDIO_BASE_SRC_H__ |
31 | #define __GST_AUDIO_BASE_SRC_H__ |
32 | |
33 | #include <gst/gst.h> |
34 | #include <gst/base/gstpushsrc.h> |
35 | |
36 | G_BEGIN_DECLS |
37 | |
38 | #define GST_TYPE_AUDIO_BASE_SRC (gst_audio_base_src_get_type()) |
39 | #define GST_AUDIO_BASE_SRC(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIO_BASE_SRC,GstAudioBaseSrc)) |
40 | #define GST_AUDIO_BASE_SRC_CAST(obj) ((GstAudioBaseSrc*)obj) |
41 | #define GST_AUDIO_BASE_SRC_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_AUDIO_BASE_SRC,GstAudioBaseSrcClass)) |
42 | #define GST_AUDIO_BASE_SRC_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_AUDIO_BASE_SRC, GstAudioBaseSrcClass)) |
43 | #define GST_IS_AUDIO_BASE_SRC(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIO_BASE_SRC)) |
44 | #define GST_IS_AUDIO_BASE_SRC_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_AUDIO_BASE_SRC)) |
45 | |
46 | /** |
47 | * GST_AUDIO_BASE_SRC_CLOCK: |
48 | * @obj: a #GstAudioBaseSrc |
49 | * |
50 | * Get the #GstClock of @obj. |
51 | */ |
52 | #define GST_AUDIO_BASE_SRC_CLOCK(obj) (GST_AUDIO_BASE_SRC (obj)->clock) |
53 | /** |
54 | * GST_AUDIO_BASE_SRC_PAD: |
55 | * @obj: a #GstAudioBaseSrc |
56 | * |
57 | * Get the source #GstPad of @obj. |
58 | */ |
59 | #define GST_AUDIO_BASE_SRC_PAD(obj) (GST_BASE_SRC (obj)->srcpad) |
60 | |
61 | typedef struct _GstAudioBaseSrc GstAudioBaseSrc; |
62 | typedef struct _GstAudioBaseSrcClass GstAudioBaseSrcClass; |
63 | typedef struct _GstAudioBaseSrcPrivate GstAudioBaseSrcPrivate; |
64 | |
65 | /* FIXME 2.0: Should be "retimestamp" not "re-timestamp" */ |
66 | |
67 | /** |
68 | * GstAudioBaseSrcSlaveMethod: |
69 | * @GST_AUDIO_BASE_SRC_SLAVE_RESAMPLE: Resample to match the master clock. |
70 | * @GST_AUDIO_BASE_SRC_SLAVE_RE_TIMESTAMP: Retimestamp output buffers with master |
71 | * clock time. |
72 | * @GST_AUDIO_BASE_SRC_SLAVE_SKEW: Adjust capture pointer when master clock |
73 | * drifts too much. |
74 | * @GST_AUDIO_BASE_SRC_SLAVE_NONE: No adjustment is done. |
75 | * |
76 | * Different possible clock slaving algorithms when the internal audio clock was |
77 | * not selected as the pipeline clock. |
78 | */ |
79 | typedef enum |
80 | { |
81 | GST_AUDIO_BASE_SRC_SLAVE_RESAMPLE, |
82 | GST_AUDIO_BASE_SRC_SLAVE_RE_TIMESTAMP, |
83 | GST_AUDIO_BASE_SRC_SLAVE_SKEW, |
84 | GST_AUDIO_BASE_SRC_SLAVE_NONE |
85 | } GstAudioBaseSrcSlaveMethod; |
86 | |
87 | #define GST_AUDIO_BASE_SRC_SLAVE_RETIMESTAMP GST_AUDIO_BASE_SRC_SLAVE_RE_TIMESTAMP |
88 | |
89 | /** |
90 | * GstAudioBaseSrc: |
91 | * |
92 | * Opaque #GstAudioBaseSrc. |
93 | */ |
94 | struct _GstAudioBaseSrc { |
95 | GstPushSrc element; |
96 | |
97 | /*< protected >*/ /* with LOCK */ |
98 | /* our ringbuffer */ |
99 | GstAudioRingBuffer *ringbuffer; |
100 | |
101 | /* required buffer and latency */ |
102 | GstClockTime buffer_time; |
103 | GstClockTime latency_time; |
104 | |
105 | /* the next sample to write */ |
106 | guint64 next_sample; |
107 | |
108 | /* clock */ |
109 | GstClock *clock; |
110 | |
111 | /*< private >*/ |
112 | GstAudioBaseSrcPrivate *priv; |
113 | |
114 | gpointer _gst_reserved[GST_PADDING]; |
115 | }; |
116 | |
117 | /** |
118 | * GstAudioBaseSrcClass: |
119 | * @parent_class: the parent class. |
120 | * @create_ringbuffer: create and return a #GstAudioRingBuffer to read from. |
121 | * |
122 | * #GstAudioBaseSrc class. Override the vmethod to implement |
123 | * functionality. |
124 | */ |
125 | struct _GstAudioBaseSrcClass { |
126 | GstPushSrcClass parent_class; |
127 | |
128 | /* subclass ringbuffer allocation */ |
129 | GstAudioRingBuffer* (*create_ringbuffer) (GstAudioBaseSrc *src); |
130 | |
131 | /*< private >*/ |
132 | gpointer _gst_reserved[GST_PADDING]; |
133 | }; |
134 | |
135 | GST_AUDIO_API |
136 | GType gst_audio_base_src_get_type(void); |
137 | |
138 | GST_AUDIO_API |
139 | GstAudioRingBuffer * |
140 | gst_audio_base_src_create_ringbuffer (GstAudioBaseSrc *src); |
141 | |
142 | GST_AUDIO_API |
143 | void gst_audio_base_src_set_provide_clock (GstAudioBaseSrc *src, gboolean provide); |
144 | |
145 | GST_AUDIO_API |
146 | gboolean gst_audio_base_src_get_provide_clock (GstAudioBaseSrc *src); |
147 | |
148 | GST_AUDIO_API |
149 | void gst_audio_base_src_set_slave_method (GstAudioBaseSrc *src, |
150 | GstAudioBaseSrcSlaveMethod method); |
151 | GST_AUDIO_API |
152 | GstAudioBaseSrcSlaveMethod |
153 | gst_audio_base_src_get_slave_method (GstAudioBaseSrc *src); |
154 | |
155 | |
156 | G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstAudioBaseSrc, gst_object_unref) |
157 | |
158 | G_END_DECLS |
159 | |
160 | #endif /* __GST_AUDIO_BASE_SRC_H__ */ |
161 | |