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
25G_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
30typedef 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 */
50struct _GstAudioDownmixMeta {
51 GstMeta meta;
52
53 GstAudioChannelPosition *from_position;
54 GstAudioChannelPosition *to_position;
55 gint from_channels, to_channels;
56 gfloat **matrix;
57};
58
59GST_AUDIO_API
60GType gst_audio_downmix_meta_api_get_type (void);
61
62GST_AUDIO_API
63const 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))
66GST_AUDIO_API
67GstAudioDownmixMeta * gst_buffer_get_audio_downmix_meta_for_channels (GstBuffer *buffer,
68 const GstAudioChannelPosition *to_position,
69 gint to_channels);
70
71GST_AUDIO_API
72GstAudioDownmixMeta * 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
83typedef 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 */
106struct _GstAudioClippingMeta {
107 GstMeta meta;
108
109 GstFormat format;
110 guint64 start;
111 guint64 end;
112};
113
114GST_AUDIO_API
115GType gst_audio_clipping_meta_api_get_type (void);
116
117GST_AUDIO_API
118const 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
122GST_AUDIO_API
123GstAudioClippingMeta * 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
132typedef 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 */
175struct _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
187GST_AUDIO_API
188GType gst_audio_meta_api_get_type (void);
189
190GST_AUDIO_API
191const 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
196GST_AUDIO_API
197GstAudioMeta * 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())
217typedef 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 */
229struct _GstAudioLevelMeta
230{
231 GstMeta meta;
232
233 guint8 level;
234 gboolean voice_activity;
235};
236
237GST_AUDIO_API
238GType gst_audio_level_meta_api_get_type (void);
239
240GST_AUDIO_API
241const GstMetaInfo * gst_audio_level_meta_get_info (void);
242
243GST_AUDIO_API
244GstAudioLevelMeta * gst_buffer_add_audio_level_meta (GstBuffer * buffer,
245 guint8 level,
246 gboolean voice_activity);
247GST_AUDIO_API
248GstAudioLevelMeta * gst_buffer_get_audio_level_meta (GstBuffer * buffer);
249
250G_END_DECLS
251
252#endif /* __GST_AUDIO_META_H__ */
253

source code of include/gstreamer-1.0/gst/audio/gstaudiometa.h