aboutsummaryrefslogtreecommitdiffstats
path: root/funcs/func_speex.c
blob: e17e8ca9c00e74cc911d537da13b920ce4773d89 (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
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
230
231
232
233
234
235
236
237
238
239
240
241
242
243
244
245
246
247
248
249
250
251
252
253
254
255
256
257
258
259
260
261
262
263
264
265
266
267
268
269
270
271
272
273
274
275
276
277
278
279
280
281
282
283
284
285
286
287
288
289
290
291
292
293
294
295
296
297
298
299
300
301
302
303
304
305
306
307
308
309
310
311
312
313
314
315
316
317
318
319
320
321
322
323
324
325
326
327
328
329
330
331
332
333
334
335
336
337
338
339
340
341
342
343
344
345
346
347
348
349
350
351
352
353
354
355
356
357
358
359
360
361
362
363
364
365
366
367
368
369
370
371
372
373
374
375
376
377
378
379
380
/*
 * Asterisk -- An open source telephony toolkit.
 *
 * Copyright (C) 2008, Digium, Inc.
 *
 * Brian Degenhardt <bmd@digium.com>
 * Brett Bryant <bbryant@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 Noise reduction and automatic gain control (AGC)
 *
 * \author Brian Degenhardt <bmd@digium.com> 
 * \author Brett Bryant <bbryant@digium.com> 
 *
 * \ingroup functions
 *
 * \extref The Speex 1.2 library - http://www.speex.org
 * \note Requires the 1.2 version of the Speex library (which might not be what you find in Linux packages)
 */

/*** MODULEINFO
	<depend>speex</depend>
	<depend>speex_preprocess</depend>
	<use type="external">speexdsp</use>
 ***/

#include "asterisk.h"

ASTERISK_FILE_VERSION(__FILE__, "$Revision$")

#include <speex/speex_preprocess.h>
#include "asterisk/module.h"
#include "asterisk/channel.h"
#include "asterisk/pbx.h"
#include "asterisk/utils.h"
#include "asterisk/audiohook.h"

#define DEFAULT_AGC_LEVEL 8000.0

/*** DOCUMENTATION
	<function name="AGC" language="en_US">
		<synopsis>
			Apply automatic gain control to audio on a channel.
		</synopsis>
		<syntax>
			<parameter name="channeldirection" required="true">
				<para>This can be either <literal>rx</literal> or <literal>tx</literal></para>
			</parameter>
		</syntax>
		<description>
			<para>The AGC function will apply automatic gain control to the audio on the
			channel that it is executed on. Using <literal>rx</literal> for audio received
			and <literal>tx</literal> for audio transmitted to the channel. When using this
			function you set a target audio level. It is primarily intended for use with
			analog lines, but could be useful for other channels as well. The target volume 
			is set with a number between <literal>1-32768</literal>. The larger the number
			the louder (more gain) the channel will receive.</para>
			<para>Examples:</para>
			<para>exten => 1,1,Set(AGC(rx)=8000)</para>
			<para>exten => 1,2,Set(AGC(tx)=off)</para>
		</description>
	</function>
	<function name="DENOISE" language="en_US">
		<synopsis>
			Apply noise reduction to audio on a channel.
		</synopsis>
		<syntax>
			<parameter name="channeldirection" required="true">
				<para>This can be either <literal>rx</literal> or <literal>tx</literal> 
				the values that can be set to this are either <literal>on</literal> and
				<literal>off</literal></para>
			</parameter>
		</syntax>
		<description>
			<para>The DENOISE function will apply noise reduction to audio on the channel
			that it is executed on. It is very useful for noisy analog lines, especially
			when adjusting gains or using AGC. Use <literal>rx</literal> for audio received from the channel
			and <literal>tx</literal> to apply the filter to the audio being sent to the channel.</para>
			<para>Examples:</para>
			<para>exten => 1,1,Set(DENOISE(rx)=on)</para>
			<para>exten => 1,2,Set(DENOISE(tx)=off)</para>
		</description>
	</function>
 ***/

struct speex_direction_info {
	SpeexPreprocessState *state;	/*!< speex preprocess state object */
	int agc;						/*!< audio gain control is enabled or not */
	int denoise;					/*!< denoise is enabled or not */
	int samples;					/*!< n of 8Khz samples in last frame */
	float agclevel;					/*!< audio gain control level [1.0 - 32768.0] */
};

struct speex_info {
	struct ast_audiohook audiohook;
	int lastrate;
	struct speex_direction_info *tx, *rx;
};

static void destroy_callback(void *data) 
{
	struct speex_info *si = data;

	ast_audiohook_destroy(&si->audiohook);

	if (si->rx && si->rx->state) {
		speex_preprocess_state_destroy(si->rx->state);
	}

	if (si->tx && si->tx->state) {
		speex_preprocess_state_destroy(si->tx->state);
	}

	if (si->rx) {
		ast_free(si->rx);
	}

	if (si->tx) {
		ast_free(si->tx);
	}

	ast_free(data);
};

static const struct ast_datastore_info speex_datastore = {
	.type = "speex",
	.destroy = destroy_callback
};

static int speex_callback(struct ast_audiohook *audiohook, struct ast_channel *chan, struct ast_frame *frame, enum ast_audiohook_direction direction)
{
	struct ast_datastore *datastore = NULL;
	struct speex_direction_info *sdi = NULL;
	struct speex_info *si = NULL;
	char source[80];

	/* If the audiohook is stopping it means the channel is shutting down.... but we let the datastore destroy take care of it */
	if (audiohook->status == AST_AUDIOHOOK_STATUS_DONE || frame->frametype != AST_FRAME_VOICE) {
		return -1;
	}

	/* We are called with chan already locked */
	if (!(datastore = ast_channel_datastore_find(chan, &speex_datastore, NULL))) {
		return -1;
	}

	si = datastore->data;

	sdi = (direction == AST_AUDIOHOOK_DIRECTION_READ) ? si->rx : si->tx;

	if (!sdi) {
		return -1;
	}

	if ((sdi->samples != frame->samples) || (ast_format_rate(&frame->subclass.format) != si->lastrate)) {
		si->lastrate = ast_format_rate(&frame->subclass.format);
		if (sdi->state) {
			speex_preprocess_state_destroy(sdi->state);
		}

		if (!(sdi->state = speex_preprocess_state_init((sdi->samples = frame->samples), si->lastrate))) {
			return -1;
		}

		speex_preprocess_ctl(sdi->state, SPEEX_PREPROCESS_SET_AGC, &sdi->agc);

		if (sdi->agc) {
			speex_preprocess_ctl(sdi->state, SPEEX_PREPROCESS_SET_AGC_LEVEL, &sdi->agclevel);
		}

		speex_preprocess_ctl(sdi->state, SPEEX_PREPROCESS_SET_DENOISE, &sdi->denoise);
	}

	speex_preprocess(sdi->state, frame->data.ptr, NULL);
	snprintf(source, sizeof(source), "%s/speex", frame->src);
	if (frame->mallocd & AST_MALLOCD_SRC) {
		ast_free((char *) frame->src);
	}
	frame->src = ast_strdup(source);
	frame->mallocd |= AST_MALLOCD_SRC;

	return 0;
}

static int speex_write(struct ast_channel *chan, const char *cmd, char *data, const char *value)
{
	struct ast_datastore *datastore = NULL;
	struct speex_info *si = NULL;
	struct speex_direction_info **sdi = NULL;
	int is_new = 0;

	ast_channel_lock(chan);
	if (!(datastore = ast_channel_datastore_find(chan, &speex_datastore, NULL))) {
		ast_channel_unlock(chan);

		if (!(datastore = ast_datastore_alloc(&speex_datastore, NULL))) {
			return 0;
		}

		if (!(si = ast_calloc(1, sizeof(*si)))) {
			ast_datastore_free(datastore);
			return 0;
		}

		ast_audiohook_init(&si->audiohook, AST_AUDIOHOOK_TYPE_MANIPULATE, "speex", AST_AUDIOHOOK_MANIPULATE_ALL_RATES);
		si->audiohook.manipulate_callback = speex_callback;
		si->lastrate = 8000;
		is_new = 1;
	} else {
		ast_channel_unlock(chan);
		si = datastore->data;
	}

	if (!strcasecmp(data, "rx")) {
		sdi = &si->rx;
	} else if (!strcasecmp(data, "tx")) {
		sdi = &si->tx;
	} else {
		ast_log(LOG_ERROR, "Invalid argument provided to the %s function\n", cmd);

		if (is_new) {
			ast_datastore_free(datastore);
			return -1;
		}
	}

	if (!*sdi) {
		if (!(*sdi = ast_calloc(1, sizeof(**sdi)))) {
			return 0;
		}
		/* Right now, the audiohooks API will _only_ provide us 8 kHz slinear
		 * audio.  When it supports 16 kHz (or any other sample rates, we will
		 * have to take that into account here. */
		(*sdi)->samples = -1;
	}

	if (!strcasecmp(cmd, "agc")) {
		if (!sscanf(value, "%30f", &(*sdi)->agclevel))
			(*sdi)->agclevel = ast_true(value) ? DEFAULT_AGC_LEVEL : 0.0;
	
		if ((*sdi)->agclevel > 32768.0) {
			ast_log(LOG_WARNING, "AGC(%s)=%.01f is greater than 32768... setting to 32768 instead\n", 
					((*sdi == si->rx) ? "rx" : "tx"), (*sdi)->agclevel);
			(*sdi)->agclevel = 32768.0;
		}
	
		(*sdi)->agc = !!((*sdi)->agclevel);

		if ((*sdi)->state) {
			speex_preprocess_ctl((*sdi)->state, SPEEX_PREPROCESS_SET_AGC, &(*sdi)->agc);
			if ((*sdi)->agc) {
				speex_preprocess_ctl((*sdi)->state, SPEEX_PREPROCESS_SET_AGC_LEVEL, &(*sdi)->agclevel);
			}
		}
	} else if (!strcasecmp(cmd, "denoise")) {
		(*sdi)->denoise = (ast_true(value) != 0);

		if ((*sdi)->state) {
			speex_preprocess_ctl((*sdi)->state, SPEEX_PREPROCESS_SET_DENOISE, &(*sdi)->denoise);
		}
	}

	if (!(*sdi)->agc && !(*sdi)->denoise) {
		if ((*sdi)->state)
			speex_preprocess_state_destroy((*sdi)->state);

		ast_free(*sdi);
		*sdi = NULL;
	}

	if (!si->rx && !si->tx) {
		if (is_new) {
			is_new = 0;
		} else {
			ast_channel_lock(chan);
			ast_channel_datastore_remove(chan, datastore);
			ast_channel_unlock(chan);
			ast_audiohook_remove(chan, &si->audiohook);
			ast_audiohook_detach(&si->audiohook);
		}
		
		ast_datastore_free(datastore);
	}

	if (is_new) { 
		datastore->data = si;
		ast_channel_lock(chan);
		ast_channel_datastore_add(chan, datastore);
		ast_channel_unlock(chan);
		ast_audiohook_attach(chan, &si->audiohook);
	}

	return 0;
}

static int speex_read(struct ast_channel *chan, const char *cmd, char *data, char *buf, size_t len)
{
	struct ast_datastore *datastore = NULL;
	struct speex_info *si = NULL;
	struct speex_direction_info *sdi = NULL;

	if (!chan) {
		ast_log(LOG_ERROR, "%s cannot be used without a channel!\n", cmd);
		return -1;
	}

	ast_channel_lock(chan);
	if (!(datastore = ast_channel_datastore_find(chan, &speex_datastore, NULL))) {
		ast_channel_unlock(chan);
		return -1;
	}
	ast_channel_unlock(chan);

	si = datastore->data;

	if (!strcasecmp(data, "tx"))
		sdi = si->tx;
	else if (!strcasecmp(data, "rx"))
		sdi = si->rx;
	else {
		ast_log(LOG_ERROR, "%s(%s) must either \"tx\" or \"rx\"\n", cmd, data);
		return -1;
	}

	if (!strcasecmp(cmd, "agc"))
		snprintf(buf, len, "%.01f", sdi ? sdi->agclevel : 0.0);
	else
		snprintf(buf, len, "%d", sdi ? sdi->denoise : 0);

	return 0;
}

static struct ast_custom_function agc_function = {
	.name = "AGC",
	.write = speex_write,
	.read = speex_read,
	.read_max = 22,
};

static struct ast_custom_function denoise_function = {
	.name = "DENOISE",
	.write = speex_write,
	.read = speex_read,
	.read_max = 22,
};

static int unload_module(void)
{
	ast_custom_function_unregister(&agc_function);
	ast_custom_function_unregister(&denoise_function);
	return 0;
}

static int load_module(void)
{
	if (ast_custom_function_register(&agc_function)) {
		return AST_MODULE_LOAD_DECLINE;
	}

	if (ast_custom_function_register(&denoise_function)) {
		ast_custom_function_unregister(&agc_function);
		return AST_MODULE_LOAD_DECLINE;
	}

	return AST_MODULE_LOAD_SUCCESS;
}

AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Noise reduction and Automatic Gain Control (AGC)");