aboutsummaryrefslogtreecommitdiffstats
path: root/codecs/codec_resample.c
blob: 66ef584bd7b0402aa79a149ee045a444a08b2df0 (plain)
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
/*
 * Asterisk -- An open source telephony toolkit.
 *
 * Copyright (C) 2011, Digium, Inc.
 *
 * Russell Bryant <russell@digium.com>
 * David Vossel <dvossel@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.
 */

/*! 
 * \file
 *
 * \brief Resample slinear audio
 * 
 * \ingroup codecs
 */

#include "asterisk.h"
#include "speex/speex_resampler.h"

ASTERISK_FILE_VERSION(__FILE__, "$Revision$")

#include "asterisk/module.h"
#include "asterisk/translate.h"
#include "asterisk/slin.h"

#define OUTBUF_SIZE   8096

static struct ast_translator *translators;
static int trans_size;
static int id_list[] = {
	AST_FORMAT_SLINEAR,
	AST_FORMAT_SLINEAR12,
	AST_FORMAT_SLINEAR16,
	AST_FORMAT_SLINEAR24,
	AST_FORMAT_SLINEAR32,
	AST_FORMAT_SLINEAR44,
	AST_FORMAT_SLINEAR48,
	AST_FORMAT_SLINEAR96,
	AST_FORMAT_SLINEAR192,
};

static int resamp_new(struct ast_trans_pvt *pvt)
{
	int err;

	if (!(pvt->pvt = speex_resampler_init(1, ast_format_rate(&pvt->t->src_format), ast_format_rate(&pvt->t->dst_format), 5, &err))) {
		return -1;
	}

	return 0;
}

static void resamp_destroy(struct ast_trans_pvt *pvt)
{
	SpeexResamplerState *resamp_pvt = pvt->pvt;
	speex_resampler_destroy(resamp_pvt);
}

static int resamp_framein(struct ast_trans_pvt *pvt, struct ast_frame *f)
{
	SpeexResamplerState *resamp_pvt = pvt->pvt;
	unsigned int out_samples = (OUTBUF_SIZE / sizeof(int16_t)) - pvt->samples;
	unsigned int in_samples;

	if (!f->datalen) {
		return -1;
	}
	in_samples = f->datalen / 2;

	speex_resampler_process_int(resamp_pvt,
		0,
		f->data.ptr,
		&in_samples,
		pvt->outbuf.i16 + pvt->samples,
		&out_samples);

	pvt->samples += out_samples;
	pvt->datalen += out_samples * 2;

	return 0;
}

static int unload_module(void)
{
	int res = 0;
	int idx;

	for (idx = 0; idx < trans_size; idx++) {
		res |= ast_unregister_translator(&translators[idx]);
	}
	ast_free(translators);

	return res;
}

static int load_module(void)
{
	int res = 0;
	int x, y, idx = 0;

	trans_size = ARRAY_LEN(id_list) * ARRAY_LEN(id_list);
	if (!(translators = ast_calloc(1, sizeof(struct ast_translator) * trans_size))) {
		return AST_MODULE_LOAD_FAILURE;
	}

	for (x = 0; x < ARRAY_LEN(id_list); x++) {
		for (y = 0; y < ARRAY_LEN(id_list); y++) {
			if (x == y) {
				continue;
			}
			translators[idx].newpvt = resamp_new;
			translators[idx].destroy = resamp_destroy;
			translators[idx].framein = resamp_framein;
			translators[idx].desc_size = 0;
			translators[idx].buffer_samples = (OUTBUF_SIZE / sizeof(int16_t));
			translators[idx].buf_size = OUTBUF_SIZE;
			ast_format_set(&translators[idx].src_format, id_list[x], 0);
			ast_format_set(&translators[idx].dst_format, id_list[y], 0);
			snprintf(translators[idx].name, sizeof(translators[idx].name), "slin %dkhz -> %dkhz",
				ast_format_rate(&translators[idx].src_format), ast_format_rate(&translators[idx].dst_format));
			res |= ast_register_translator(&translators[idx]);
			idx++;
		}

	}

	return AST_MODULE_LOAD_SUCCESS;
}

AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "SLIN Resampling Codec");