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
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
155
156
157
158
159
160
161
162
163
164
165
166
167
168
169
170
171
172
173
174
175
176
177
178
179
180
181
182
183
184
185
186
187
188
189
190
191
192
193
194
195
196
197
198
199
200
201
202
203
204
205
206
207
208
209
210
211
212
213
214
215
216
217
218
219
220
221
222
223
224
225
226
227
228
229
230
231
232
233
234
235
236
237
238
239
240
241
242
243
244
245
246
247
248
249
250
251
252
253
254
255
256
257
258
259
260
261
262
263
264
265
266
267
|
/* gstreamer-audio-0.10.vapi generated by lt-vapigen, do not modify. */
[CCode (cprefix = "Gst", lower_case_cprefix = "gst_")]
namespace Gst {
[CCode (cprefix = "GST_AUDIO_CHANNEL_POSITION_", cheader_filename = "gst/audio/multichannel.h")]
public enum AudioChannelPosition {
INVALID,
FRONT_MONO,
FRONT_LEFT,
FRONT_RIGHT,
REAR_CENTER,
REAR_LEFT,
REAR_RIGHT,
LFE,
FRONT_CENTER,
FRONT_LEFT_OF_CENTER,
FRONT_RIGHT_OF_CENTER,
SIDE_LEFT,
SIDE_RIGHT,
NONE,
NUM,
}
[CCode (cprefix = "GST_AUDIO_FIELD_", cheader_filename = "gst/audio/audio.h")]
public enum AudioFieldFlag {
RATE,
CHANNELS,
ENDIANNESS,
WIDTH,
DEPTH,
SIGNED,
}
[CCode (cprefix = "GST_BASE_AUDIO_SINK_SLAVE_", cheader_filename = "gst/audio/gstbaseaudiosink.h")]
public enum BaseAudioSinkSlaveMethod {
RESAMPLE,
SKEW,
NONE,
}
[CCode (cprefix = "GST_", cheader_filename = "gst/audio/gstringbuffer.h")]
public enum BufferFormat {
UNKNOWN,
S8,
U8,
S16_LE,
S16_BE,
U16_LE,
U16_BE,
S24_LE,
S24_BE,
U24_LE,
U24_BE,
S32_LE,
S32_BE,
U32_LE,
U32_BE,
S24_3LE,
S24_3BE,
U24_3LE,
U24_3BE,
S20_3LE,
S20_3BE,
U20_3LE,
U20_3BE,
S18_3LE,
S18_3BE,
U18_3LE,
U18_3BE,
FLOAT32_LE,
FLOAT32_BE,
FLOAT64_LE,
FLOAT64_BE,
MU_LAW,
A_LAW,
IMA_ADPCM,
MPEG,
GSM,
}
[CCode (cprefix = "GST_BUFTYPE_", cheader_filename = "gst/audio/gstringbuffer.h")]
public enum BufferFormatType {
LINEAR,
FLOAT,
MU_LAW,
A_LAW,
IMA_ADPCM,
MPEG,
GSM,
}
[CCode (cprefix = "GST_SEGSTATE_", cheader_filename = "gst/audio/gstringbuffer.h")]
public enum RingBufferSegState {
INVALID,
EMPTY,
FILLED,
PARTIAL,
}
[CCode (cprefix = "GST_RING_BUFFER_STATE_", cheader_filename = "gst/audio/gstringbuffer.h")]
public enum RingBufferState {
STOPPED,
PAUSED,
STARTED,
}
[CCode (cheader_filename = "gst/audio/gstaudiofilter.h")]
public class RingBufferSpec {
public weak Gst.Caps caps;
public Gst.BufferFormatType type;
public Gst.BufferFormat format;
public bool sign;
public bool bigend;
public int width;
public int depth;
public int rate;
public int channels;
public uint64 latency_time;
public uint64 buffer_time;
public int segsize;
public int segtotal;
public int bytes_per_sample;
[NoArrayLength]
public weak uchar[] silence_sample;
[NoArrayLength]
public weak pointer[] _gst_reserved;
}
[CCode (cheader_filename = "gst/audio/gstaudioclock.h")]
public class AudioClock : Gst.SystemClock {
public Gst.AudioClockGetTimeFunc func;
public pointer user_data;
public weak Gst.ClockTime last_time;
public AudioClock (string name, Gst.AudioClockGetTimeFunc func, pointer user_data);
}
[CCode (cheader_filename = "gst/audio/gstaudiofilter.h")]
public class AudioFilter : Gst.BaseTransform {
public weak Gst.RingBufferSpec format;
public static void class_add_pad_templates (pointer klass, Gst.Caps allowed_caps);
[NoWrapper]
public virtual bool setup (Gst.RingBufferSpec format);
}
[CCode (cheader_filename = "gst/audio/gstaudiosink.h")]
public class AudioSink : Gst.BaseAudioSink {
public weak GLib.Thread thread;
[NoWrapper]
public virtual bool close ();
[NoWrapper]
public virtual uint delay ();
[NoWrapper]
public virtual bool open ();
[NoWrapper]
public virtual bool prepare (Gst.RingBufferSpec spec);
[NoWrapper]
public virtual void reset ();
[NoWrapper]
public virtual bool unprepare ();
[NoWrapper]
public virtual uint write (pointer data, uint length);
}
[CCode (cheader_filename = "gst/audio/gstaudiosrc.h")]
public class AudioSrc : Gst.BaseAudioSrc {
public weak GLib.Thread thread;
[NoWrapper]
public virtual bool close ();
[NoWrapper]
public virtual uint delay ();
[NoWrapper]
public virtual bool open ();
[NoWrapper]
public virtual bool prepare (Gst.RingBufferSpec spec);
[NoWrapper]
public virtual uint read (pointer data, uint length);
[NoWrapper]
public virtual void reset ();
[NoWrapper]
public virtual bool unprepare ();
}
[CCode (cheader_filename = "gst/audio/gstaudiosink.h")]
public class BaseAudioSink : Gst.BaseSink {
public weak Gst.RingBuffer ringbuffer;
public uint64 next_sample;
public weak Gst.Clock provided_clock;
public bool get_provide_clock ();
public Gst.BaseAudioSinkSlaveMethod get_slave_method ();
public void set_provide_clock (bool provide);
public void set_slave_method (Gst.BaseAudioSinkSlaveMethod method);
public virtual weak Gst.RingBuffer create_ringbuffer ();
[NoAccessorMethod]
public weak int64 buffer_time { get; set; }
[NoAccessorMethod]
public weak int64 latency_time { get; set; }
public weak bool provide_clock { get; set; }
public weak Gst.BaseAudioSinkSlaveMethod slave_method { get; set; }
}
[CCode (cheader_filename = "gst/audio/gstaudiosrc.h")]
public class BaseAudioSrc : Gst.PushSrc {
public weak Gst.RingBuffer ringbuffer;
public uint64 next_sample;
public weak Gst.Clock clock;
public bool get_provide_clock ();
public void set_provide_clock (bool provide);
public virtual weak Gst.RingBuffer create_ringbuffer ();
[NoAccessorMethod]
public weak int64 buffer_time { get; set; }
[NoAccessorMethod]
public weak int64 latency_time { get; set; }
public weak bool provide_clock { get; set; }
}
[CCode (cheader_filename = "gst/audio/gstaudiofilter.h")]
public class RingBuffer : Gst.Object {
public weak GLib.Cond cond;
public bool open;
public bool acquired;
public weak Gst.Buffer data;
public weak Gst.RingBufferSpec spec;
public Gst.RingBufferSegState segstate;
public int samples_per_seg;
public uchar empty_seg;
public int state;
public int segdone;
public int segbase;
public int waiting;
public Gst.RingBufferCallback callback;
public pointer cb_data;
public pointer abidata;
public void advance (uint advance);
public void clear (int segment);
public void clear_all ();
public uint commit (uint64 sample, uchar[] data, uint len);
public uint commit_full (uint64 sample, uchar[] data, int in_samples, int out_samples, int accum);
public static void debug_spec_buff (Gst.RingBufferSpec spec);
public static void debug_spec_caps (Gst.RingBufferSpec spec);
public bool device_is_open ();
public bool is_acquired ();
public void may_start (bool allowed);
public static bool parse_caps (Gst.RingBufferSpec spec, Gst.Caps caps);
public bool prepare_read (int segment, uchar readptr, int len);
public uint read (uint64 sample, uchar[] data, uint len);
public uint64 samples_done ();
public void set_callback (Gst.RingBufferCallback cb, pointer user_data);
public void set_flushing (bool flushing);
public void set_sample (uint64 sample);
public virtual bool acquire (Gst.RingBufferSpec spec);
public virtual bool close_device ();
public virtual uint delay ();
public virtual bool open_device ();
public virtual bool pause ();
public virtual bool release ();
[NoWrapper]
public virtual bool resume ();
public virtual bool start ();
public virtual bool stop ();
}
public static delegate weak Gst.ClockTime AudioClockGetTimeFunc (Gst.Clock clock, pointer user_data);
public static delegate bool AudioMixerFilterFunc (Gst.Mixer mixer, pointer user_data);
public static delegate void RingBufferCallback (Gst.RingBuffer rbuf, uchar data, uint len, pointer user_data);
public const int AUDIO_DEF_RATE;
public const string AUDIO_FLOAT_PAD_TEMPLATE_CAPS;
public const string AUDIO_FLOAT_STANDARD_PAD_TEMPLATE_CAPS;
public const string AUDIO_INT_PAD_TEMPLATE_CAPS;
public const string AUDIO_INT_STANDARD_PAD_TEMPLATE_CAPS;
public static weak Gst.Buffer audio_buffer_clip (Gst.Buffer buffer, Gst.Segment segment, int rate, int frame_size);
public static weak GLib.List audio_default_registry_mixer_filter (Gst.AudioMixerFilterFunc filter_func, bool first, pointer user_data);
public static weak Gst.ClockTime audio_duration_from_pad_buffer (Gst.Pad pad, Gst.Buffer buf);
public static Gst.AudioChannelPosition audio_fixate_channel_positions (Gst.Structure str);
public static int audio_frame_byte_size (Gst.Pad pad);
public static long audio_frame_length (Gst.Pad pad, Gst.Buffer buf);
public static Gst.AudioChannelPosition audio_get_channel_positions (Gst.Structure str);
public static bool audio_is_buffer_framed (Gst.Pad pad, Gst.Buffer buf);
public static void audio_set_caps_channel_positions_list (Gst.Caps caps, Gst.AudioChannelPosition pos, int num_positions);
public static void audio_set_channel_positions (Gst.Structure str, Gst.AudioChannelPosition pos);
public static void audio_set_structure_channel_positions_list (Gst.Structure str, Gst.AudioChannelPosition pos, int num_positions);
public static void audio_structure_set_int (Gst.Structure structure, Gst.AudioFieldFlag flag);
}
|
