Merge "app_agent_spool: Fix typo in dtmf features usage desctiption"
[asterisk/asterisk.git] / codecs / codec_resample.c
1 /*
2  * Asterisk -- An open source telephony toolkit.
3  *
4  * Copyright (C) 2011, Digium, Inc.
5  *
6  * Russell Bryant <russell@digium.com>
7  * David Vossel <dvossel@digium.com>
8  *
9  * See http://www.asterisk.org for more information about
10  * the Asterisk project. Please do not directly contact
11  * any of the maintainers of this project for assistance;
12  * the project provides a web site, mailing lists and IRC
13  * channels for your use.
14  *
15  * This program is free software, distributed under the terms of
16  * the GNU General Public License Version 2. See the LICENSE file
17  * at the top of the source tree.
18  */
19
20 /*! 
21  * \file
22  *
23  * \brief Resample slinear audio
24  * 
25  * \ingroup codecs
26  */
27
28 /*** MODULEINFO
29         <support_level>core</support_level>
30  ***/
31
32 #include "asterisk.h"
33 #include "speex/speex_resampler.h"
34
35 #include "asterisk/module.h"
36 #include "asterisk/translate.h"
37 #include "asterisk/slin.h"
38
39 #define OUTBUF_SAMPLES   11520
40
41 static struct ast_translator *translators;
42 static int trans_size;
43 static struct ast_codec codec_list[] = {
44         {
45                 .name = "slin",
46                 .type = AST_MEDIA_TYPE_AUDIO,
47                 .sample_rate = 8000,
48         },
49         {
50                 .name = "slin",
51                 .type = AST_MEDIA_TYPE_AUDIO,
52                 .sample_rate = 12000,
53         },
54         {
55                 .name = "slin",
56                 .type = AST_MEDIA_TYPE_AUDIO,
57                 .sample_rate = 16000,
58         },
59         {
60                 .name = "slin",
61                 .type = AST_MEDIA_TYPE_AUDIO,
62                 .sample_rate = 24000,
63         },
64         {
65                 .name = "slin",
66                 .type = AST_MEDIA_TYPE_AUDIO,
67                 .sample_rate = 32000,
68         },
69         {
70                 .name = "slin",
71                 .type = AST_MEDIA_TYPE_AUDIO,
72                 .sample_rate = 44100,
73         },
74         {
75                 .name = "slin",
76                 .type = AST_MEDIA_TYPE_AUDIO,
77                 .sample_rate = 48000,
78         },
79         {
80                 .name = "slin",
81                 .type = AST_MEDIA_TYPE_AUDIO,
82                 .sample_rate = 96000,
83         },
84         {
85                 .name = "slin",
86                 .type = AST_MEDIA_TYPE_AUDIO,
87                 .sample_rate = 192000,
88         },
89 };
90
91 static int resamp_new(struct ast_trans_pvt *pvt)
92 {
93         int err;
94
95         if (!(pvt->pvt = speex_resampler_init(1, pvt->t->src_codec.sample_rate, pvt->t->dst_codec.sample_rate, 5, &err))) {
96                 return -1;
97         }
98
99         ast_assert(pvt->f.subclass.format == NULL);
100         pvt->f.subclass.format = ao2_bump(ast_format_cache_get_slin_by_rate(pvt->t->dst_codec.sample_rate));
101
102         return 0;
103 }
104
105 static void resamp_destroy(struct ast_trans_pvt *pvt)
106 {
107         SpeexResamplerState *resamp_pvt = pvt->pvt;
108
109         speex_resampler_destroy(resamp_pvt);
110 }
111
112 static int resamp_framein(struct ast_trans_pvt *pvt, struct ast_frame *f)
113 {
114         SpeexResamplerState *resamp_pvt = pvt->pvt;
115         unsigned int out_samples = OUTBUF_SAMPLES - pvt->samples;
116         unsigned int in_samples;
117
118         if (!f->datalen) {
119                 return -1;
120         }
121         in_samples = f->datalen / 2;
122
123         speex_resampler_process_int(resamp_pvt,
124                 0,
125                 f->data.ptr,
126                 &in_samples,
127                 pvt->outbuf.i16 + pvt->samples,
128                 &out_samples);
129
130         pvt->samples += out_samples;
131         pvt->datalen += out_samples * 2;
132
133         return 0;
134 }
135
136 static int unload_module(void)
137 {
138         int res = 0;
139         int idx;
140
141         for (idx = 0; idx < trans_size; idx++) {
142                 res |= ast_unregister_translator(&translators[idx]);
143         }
144         ast_free(translators);
145
146         return res;
147 }
148
149 static int load_module(void)
150 {
151         int res = 0;
152         int x, y, idx = 0;
153
154         trans_size = ARRAY_LEN(codec_list) * (ARRAY_LEN(codec_list) - 1);
155         if (!(translators = ast_calloc(1, sizeof(struct ast_translator) * trans_size))) {
156                 return AST_MODULE_LOAD_DECLINE;
157         }
158
159         for (x = 0; x < ARRAY_LEN(codec_list); x++) {
160                 for (y = 0; y < ARRAY_LEN(codec_list); y++) {
161                         if (x == y) {
162                                 continue;
163                         }
164                         translators[idx].newpvt = resamp_new;
165                         translators[idx].destroy = resamp_destroy;
166                         translators[idx].framein = resamp_framein;
167                         translators[idx].desc_size = 0;
168                         translators[idx].buffer_samples = OUTBUF_SAMPLES;
169                         translators[idx].buf_size = (OUTBUF_SAMPLES * sizeof(int16_t));
170                         memcpy(&translators[idx].src_codec, &codec_list[x], sizeof(struct ast_codec));
171                         memcpy(&translators[idx].dst_codec, &codec_list[y], sizeof(struct ast_codec));
172                         snprintf(translators[idx].name, sizeof(translators[idx].name), "slin %ukhz -> %ukhz",
173                                 translators[idx].src_codec.sample_rate, translators[idx].dst_codec.sample_rate);
174                         res |= ast_register_translator(&translators[idx]);
175                         idx++;
176                 }
177
178         }
179         /* in case ast_register_translator() failed, we call unload_module() and
180         ast_unregister_translator won't fail.*/
181         if (res) {
182                 unload_module();
183                 return AST_MODULE_LOAD_DECLINE;
184         }
185
186         return AST_MODULE_LOAD_SUCCESS;
187 }
188
189 AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "SLIN Resampling Codec");