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 #ifdef G_DEFINE_AUTOPTR_CLEANUP_FUNC
157 G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstAudioBaseSrc, gst_object_unref)
158 #endif
159 
160 G_END_DECLS
161 
162 #endif /* __GST_AUDIO_BASE_SRC_H__ */
163