1 | /* GStreamer |
2 | * Copyright (C) <2011> Wim Taymans <wim.taymans@gmail.com> |
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_AUDIO_META_H__ |
21 | #define __GST_AUDIO_META_H__ |
22 | |
23 | #include <gst/audio/audio.h> |
24 | |
25 | G_BEGIN_DECLS |
26 | |
27 | #define GST_AUDIO_DOWNMIX_META_API_TYPE (gst_audio_downmix_meta_api_get_type()) |
28 | #define GST_AUDIO_DOWNMIX_META_INFO (gst_audio_downmix_meta_get_info()) |
29 | |
30 | typedef struct _GstAudioDownmixMeta GstAudioDownmixMeta; |
31 | |
32 | /** |
33 | * GstAudioDownmixMeta: |
34 | * @meta: parent #GstMeta |
35 | * @from_position: the channel positions of the source |
36 | * @to_position: the channel positions of the destination |
37 | * @from_channels: the number of channels of the source |
38 | * @to_channels: the number of channels of the destination |
39 | * @matrix: the matrix coefficients. |
40 | * |
41 | * Extra buffer metadata describing audio downmixing matrix. This metadata is |
42 | * attached to audio buffers and contains a matrix to downmix the buffer number |
43 | * of channels to @channels. |
44 | * |
45 | * @matrix is an two-dimensional array of @to_channels times @from_channels |
46 | * coefficients, i.e. the i-th output channels is constructed by multiplicating |
47 | * the input channels with the coefficients in @matrix[i] and taking the sum |
48 | * of the results. |
49 | */ |
50 | struct _GstAudioDownmixMeta { |
51 | GstMeta meta; |
52 | |
53 | GstAudioChannelPosition *from_position; |
54 | GstAudioChannelPosition *to_position; |
55 | gint from_channels, to_channels; |
56 | gfloat **matrix; |
57 | }; |
58 | |
59 | GST_AUDIO_API |
60 | GType gst_audio_downmix_meta_api_get_type (void); |
61 | |
62 | GST_AUDIO_API |
63 | const GstMetaInfo * gst_audio_downmix_meta_get_info (void); |
64 | |
65 | #define gst_buffer_get_audio_downmix_meta(b) ((GstAudioDownmixMeta*)gst_buffer_get_meta((b), GST_AUDIO_DOWNMIX_META_API_TYPE)) |
66 | GST_AUDIO_API |
67 | GstAudioDownmixMeta * gst_buffer_get_audio_downmix_meta_for_channels (GstBuffer *buffer, |
68 | const GstAudioChannelPosition *to_position, |
69 | gint to_channels); |
70 | |
71 | GST_AUDIO_API |
72 | GstAudioDownmixMeta * gst_buffer_add_audio_downmix_meta (GstBuffer *buffer, |
73 | const GstAudioChannelPosition *from_position, |
74 | gint from_channels, |
75 | const GstAudioChannelPosition *to_position, |
76 | gint to_channels, |
77 | const gfloat **matrix); |
78 | |
79 | |
80 | #define GST_AUDIO_CLIPPING_META_API_TYPE (gst_audio_clipping_meta_api_get_type()) |
81 | #define GST_AUDIO_CLIPPING_META_INFO (gst_audio_clipping_meta_get_info()) |
82 | |
83 | typedef struct _GstAudioClippingMeta GstAudioClippingMeta; |
84 | |
85 | /** |
86 | * GstAudioClippingMeta: |
87 | * @meta: parent #GstMeta |
88 | * @format: GstFormat of @start and @stop, GST_FORMAT_DEFAULT is samples |
89 | * @start: Amount of audio to clip from start of buffer |
90 | * @end: Amount of to clip from end of buffer |
91 | * |
92 | * Extra buffer metadata describing how much audio has to be clipped from |
93 | * the start or end of a buffer. This is used for compressed formats, where |
94 | * the first frame usually has some additional samples due to encoder and |
95 | * decoder delays, and the last frame usually has some additional samples to |
96 | * be able to fill the complete last frame. |
97 | * |
98 | * This is used to ensure that decoded data in the end has the same amount of |
99 | * samples, and multiply decoded streams can be gaplessly concatenated. |
100 | * |
101 | * Note: If clipping of the start is done by adjusting the segment, this meta |
102 | * has to be dropped from buffers as otherwise clipping could happen twice. |
103 | * |
104 | * Since: 1.8 |
105 | */ |
106 | struct _GstAudioClippingMeta { |
107 | GstMeta meta; |
108 | |
109 | GstFormat format; |
110 | guint64 start; |
111 | guint64 end; |
112 | }; |
113 | |
114 | GST_AUDIO_API |
115 | GType gst_audio_clipping_meta_api_get_type (void); |
116 | |
117 | GST_AUDIO_API |
118 | const GstMetaInfo * gst_audio_clipping_meta_get_info (void); |
119 | |
120 | #define gst_buffer_get_audio_clipping_meta(b) ((GstAudioClippingMeta*)gst_buffer_get_meta((b), GST_AUDIO_CLIPPING_META_API_TYPE)) |
121 | |
122 | GST_AUDIO_API |
123 | GstAudioClippingMeta * gst_buffer_add_audio_clipping_meta (GstBuffer *buffer, |
124 | GstFormat format, |
125 | guint64 start, |
126 | guint64 end); |
127 | |
128 | |
129 | #define GST_AUDIO_META_API_TYPE (gst_audio_meta_api_get_type()) |
130 | #define GST_AUDIO_META_INFO (gst_audio_meta_get_info()) |
131 | |
132 | typedef struct _GstAudioMeta GstAudioMeta; |
133 | |
134 | /** |
135 | * GstAudioMeta: |
136 | * @meta: parent #GstMeta |
137 | * @info: the audio properties of the buffer |
138 | * @samples: the number of valid samples in the buffer |
139 | * @offsets: the offsets (in bytes) where each channel plane starts in the |
140 | * buffer or %NULL if the buffer has interleaved layout; if not %NULL, this |
141 | * is guaranteed to be an array of @info.channels elements |
142 | * |
143 | * Buffer metadata describing how data is laid out inside the buffer. This |
144 | * is useful for non-interleaved (planar) buffers, where it is necessary to |
145 | * have a place to store where each plane starts and how long each plane is. |
146 | * |
147 | * It is a requirement for non-interleaved buffers to have this metadata |
148 | * attached and to be mapped with gst_audio_buffer_map() in order to ensure |
149 | * correct handling of clipping and channel reordering. |
150 | * |
151 | * The different channels in @offsets are always in the GStreamer channel order. |
152 | * Zero-copy channel reordering can be implemented by swapping the values in |
153 | * @offsets. |
154 | * |
155 | * It is not allowed for channels to overlap in memory, |
156 | * i.e. for each i in [0, channels), the range |
157 | * [@offsets[i], @offsets[i] + @samples * sample_stride) must not overlap |
158 | * with any other such range. |
159 | * |
160 | * It is, however, allowed to have parts of the buffer memory unused, |
161 | * by using @offsets and @samples in such a way that leave gaps on it. |
162 | * This is used to implement zero-copy clipping in non-interleaved buffers. |
163 | * |
164 | * Obviously, due to the above, it is not safe to infer the |
165 | * number of valid samples from the size of the buffer. You should always |
166 | * use the @samples variable of this metadata. |
167 | * |
168 | * Note that for interleaved audio it is not a requirement to have this |
169 | * metadata attached and at the moment of writing, there is actually no use |
170 | * case to do so. It is, however, allowed to attach it, for some potential |
171 | * future use case. |
172 | * |
173 | * Since: 1.16 |
174 | */ |
175 | struct _GstAudioMeta { |
176 | GstMeta meta; |
177 | |
178 | GstAudioInfo info; |
179 | gsize samples; |
180 | gsize *offsets; |
181 | |
182 | /*< private >*/ |
183 | gsize priv_offsets_arr[8]; |
184 | gpointer _gst_reserved[GST_PADDING]; |
185 | }; |
186 | |
187 | GST_AUDIO_API |
188 | GType gst_audio_meta_api_get_type (void); |
189 | |
190 | GST_AUDIO_API |
191 | const GstMetaInfo * gst_audio_meta_get_info (void); |
192 | |
193 | #define gst_buffer_get_audio_meta(b) \ |
194 | ((GstAudioMeta*)gst_buffer_get_meta((b), GST_AUDIO_META_API_TYPE)) |
195 | |
196 | GST_AUDIO_API |
197 | GstAudioMeta * gst_buffer_add_audio_meta (GstBuffer *buffer, |
198 | const GstAudioInfo *info, |
199 | gsize samples, gsize offsets[]); |
200 | |
201 | /** |
202 | * GST_AUDIO_LEVEL_META_API_TYPE: |
203 | * |
204 | * The #GType associated with #GstAudioLevelMeta. |
205 | * |
206 | * Since: 1.20 |
207 | */ |
208 | #define GST_AUDIO_LEVEL_META_API_TYPE (gst_audio_level_meta_api_get_type()) |
209 | /** |
210 | * GST_AUDIO_LEVEL_META_INFO: |
211 | * |
212 | * The #GstMetaInfo associated with #GstAudioLevelMeta. |
213 | * |
214 | * Since: 1.20 |
215 | */ |
216 | #define GST_AUDIO_LEVEL_META_INFO (gst_audio_level_meta_get_info()) |
217 | typedef struct _GstAudioLevelMeta GstAudioLevelMeta; |
218 | |
219 | /** |
220 | * GstAudioLevelMeta: |
221 | * @meta: parent #GstMeta |
222 | * @level: the -dBov from 0-127 (127 is silence). |
223 | * @voice_activity: whether the buffer contains voice activity |
224 | * |
225 | * Meta containing Audio Level Indication: https://tools.ietf.org/html/rfc6464 |
226 | * |
227 | * Since: 1.20 |
228 | */ |
229 | struct _GstAudioLevelMeta |
230 | { |
231 | GstMeta meta; |
232 | |
233 | guint8 level; |
234 | gboolean voice_activity; |
235 | }; |
236 | |
237 | GST_AUDIO_API |
238 | GType gst_audio_level_meta_api_get_type (void); |
239 | |
240 | GST_AUDIO_API |
241 | const GstMetaInfo * gst_audio_level_meta_get_info (void); |
242 | |
243 | GST_AUDIO_API |
244 | GstAudioLevelMeta * gst_buffer_add_audio_level_meta (GstBuffer * buffer, |
245 | guint8 level, |
246 | gboolean voice_activity); |
247 | GST_AUDIO_API |
248 | GstAudioLevelMeta * gst_buffer_get_audio_level_meta (GstBuffer * buffer); |
249 | |
250 | G_END_DECLS |
251 | |
252 | #endif /* __GST_AUDIO_META_H__ */ |
253 | |