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
|
/*
* Asterisk -- An open source telephony toolkit.
*
* Copyright (C) 2016, Digium, Inc.
*
* Mark Michelson <mmichelson@digium.com>
*
* See http://www.asterisk.org for more information about
* the Asterisk project. Please do not directly contact
* any of the maintainers of this project for assistance;
* the project provides a web site, mailing lists and IRC
* channels for your use.
*
* This program is free software, distributed under the terms of
* the GNU General Public License Version 2. See the LICENSE file
* at the top of the source tree.
*/
/*** MODULEINFO
<depend>opusfile</depend>
<support_level>core</support_level>
***/
#include "asterisk.h"
#include <opus/opus.h>
#include <opus/opusfile.h>
#include "asterisk/mod_format.h"
#include "asterisk/utils.h"
#include "asterisk/module.h"
#include "asterisk/format_cache.h"
/* 120ms of 48KHz audio */
#define SAMPLES_MAX 5760
#define BUF_SIZE (2 * SAMPLES_MAX)
struct ogg_opus_desc {
OggOpusFile *of;
};
static int fread_wrapper(void *_stream, unsigned char *_ptr, int _nbytes)
{
FILE *stream = _stream;
size_t bytes_read;
if (!stream || _nbytes < 0) {
return -1;
}
bytes_read = fread(_ptr, 1, _nbytes, stream);
return bytes_read > 0 || feof(stream) ? (int) bytes_read : OP_EREAD;
}
static int fseek_wrapper(void *_stream, opus_int64 _offset, int _whence)
{
FILE *stream = _stream;
return fseeko(stream, (off_t) _offset, _whence);
}
static opus_int64 ftell_wrapper(void *_stream)
{
FILE *stream = _stream;
return ftello(stream);
}
static int ogg_opus_open(struct ast_filestream *s)
{
struct ogg_opus_desc *desc = (struct ogg_opus_desc *) s->_private;
OpusFileCallbacks cb = {
.read = fread_wrapper,
.seek = fseek_wrapper,
.tell = ftell_wrapper,
.close = NULL,
};
memset(desc, 0, sizeof(*desc));
desc->of = op_open_callbacks(s->f, &cb, NULL, 0, NULL);
if (!desc->of) {
return -1;
}
return 0;
}
static int ogg_opus_rewrite(struct ast_filestream *s, const char *comment)
{
/* XXX Unimplemented. We currently only can read from OGG/Opus streams */
ast_log(LOG_ERROR, "Cannot write OGG/Opus streams. Sorry :(\n");
return -1;
}
static int ogg_opus_write(struct ast_filestream *fs, struct ast_frame *f)
{
/* XXX Unimplemented. We currently only can read from OGG/Opus streams */
ast_log(LOG_ERROR, "Cannot write OGG/Opus streams. Sorry :(\n");
return -1;
}
static int ogg_opus_seek(struct ast_filestream *fs, off_t sample_offset, int whence)
{
int seek_result = -1;
off_t relative_pcm_pos;
struct ogg_opus_desc *desc = fs->_private;
switch (whence) {
case SEEK_SET:
seek_result = op_pcm_seek(desc->of, sample_offset);
break;
case SEEK_CUR:
if ((relative_pcm_pos = op_pcm_tell(desc->of)) < 0) {
seek_result = -1;
break;
}
seek_result = op_pcm_seek(desc->of, relative_pcm_pos + sample_offset);
break;
case SEEK_END:
if ((relative_pcm_pos = op_pcm_total(desc->of, -1)) < 0) {
seek_result = -1;
break;
}
seek_result = op_pcm_seek(desc->of, relative_pcm_pos - sample_offset);
break;
default:
ast_log(LOG_WARNING, "Unknown *whence* to seek on OGG/Opus streams!\n");
break;
}
/* normalize error value to -1,0 */
return (seek_result == 0) ? 0 : -1;
}
static int ogg_opus_trunc(struct ast_filestream *fs)
{
/* XXX Unimplemented. This is only used when recording, and we don't support that right now. */
ast_log(LOG_ERROR, "Truncation is not supported on OGG/Opus streams!\n");
return -1;
}
static off_t ogg_opus_tell(struct ast_filestream *fs)
{
struct ogg_opus_desc *desc = fs->_private;
off_t pos;
pos = (off_t) op_pcm_tell(desc->of);
if (pos < 0) {
return -1;
}
return pos;
}
static struct ast_frame *ogg_opus_read(struct ast_filestream *fs, int *whennext)
{
struct ogg_opus_desc *desc = fs->_private;
int hole = 1;
int samples_read;
opus_int16 *out_buf;
AST_FRAME_SET_BUFFER(&fs->fr, fs->buf, AST_FRIENDLY_OFFSET, BUF_SIZE);
out_buf = (opus_int16 *) fs->fr.data.ptr;
while (hole) {
samples_read = op_read(
desc->of,
out_buf,
SAMPLES_MAX,
NULL);
if (samples_read != OP_HOLE) {
hole = 0;
}
}
if (samples_read <= 0) {
return NULL;
}
fs->fr.datalen = samples_read * 2;
fs->fr.samples = samples_read;
*whennext = fs->fr.samples;
return &fs->fr;
}
static void ogg_opus_close(struct ast_filestream *fs)
{
struct ogg_opus_desc *desc = fs->_private;
op_free(desc->of);
}
static struct ast_format_def opus_f = {
.name = "ogg_opus",
.exts = "opus",
.open = ogg_opus_open,
.rewrite = ogg_opus_rewrite,
.write = ogg_opus_write,
.seek = ogg_opus_seek,
.trunc = ogg_opus_trunc,
.tell = ogg_opus_tell,
.read = ogg_opus_read,
.close = ogg_opus_close,
.buf_size = BUF_SIZE + AST_FRIENDLY_OFFSET,
.desc_size = sizeof(struct ogg_opus_desc),
};
static int load_module(void)
{
opus_f.format = ast_format_slin48;
if (ast_format_def_register(&opus_f)) {
return AST_MODULE_LOAD_FAILURE;
}
return AST_MODULE_LOAD_SUCCESS;
}
static int unload_module(void)
{
return ast_format_def_unregister(opus_f.name);
}
AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "OGG/Opus audio",
.support_level = AST_MODULE_SUPPORT_CORE,
.load = load_module,
.unload = unload_module,
.load_pri = AST_MODPRI_APP_DEPEND
);
|
