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				| /*
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|  * Asterisk -- An open source telephony toolkit.
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|  *
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|  * Copyright (C) 1999 - 2007, Digium, Inc.
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|  *
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|  * Joshua Colp <jcolp@digium.com>
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|  *
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|  * 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
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|  * channels for your use.
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|  *
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|  * This program is free software, distributed under the terms of
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|  * the GNU General Public License Version 2. See the LICENSE file
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|  * at the top of the source tree.
 | |
|  */
 | |
| 
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| /*! \file
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|  *
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|  * \brief Audiohooks Architecture
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|  *
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|  * \author Joshua Colp <jcolp@digium.com>
 | |
|  */
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| 
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| #include "asterisk.h"
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| 
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| ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
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| 
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| #include <signal.h>
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| 
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| #include "asterisk/channel.h"
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| #include "asterisk/utils.h"
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| #include "asterisk/lock.h"
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| #include "asterisk/linkedlists.h"
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| #include "asterisk/audiohook.h"
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| #include "asterisk/slinfactory.h"
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| #include "asterisk/frame.h"
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| #include "asterisk/translate.h"
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| 
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| struct ast_audiohook_translate {
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| 	struct ast_trans_pvt *trans_pvt;
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| 	int format;
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| };
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| 
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| struct ast_audiohook_list {
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| 	struct ast_audiohook_translate in_translate[2];
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| 	struct ast_audiohook_translate out_translate[2];
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| 	AST_LIST_HEAD_NOLOCK(, ast_audiohook) spy_list;
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| 	AST_LIST_HEAD_NOLOCK(, ast_audiohook) whisper_list;
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| 	AST_LIST_HEAD_NOLOCK(, ast_audiohook) manipulate_list;
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| };
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| 
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| /*! \brief Initialize an audiohook structure
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|  * \param audiohook Audiohook structure
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|  * \param type
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|  * \param source
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|  * \return Returns 0 on success, -1 on failure
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|  */
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| int ast_audiohook_init(struct ast_audiohook *audiohook, enum ast_audiohook_type type, const char *source)
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| {
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| 	/* Need to keep the type and source */
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| 	audiohook->type = type;
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| 	audiohook->source = source;
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| 
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| 	/* Initialize lock that protects our audiohook */
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| 	ast_mutex_init(&audiohook->lock);
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| 	ast_cond_init(&audiohook->trigger, NULL);
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| 
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| 	/* Setup the factories that are needed for this audiohook type */
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| 	switch (type) {
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| 	case AST_AUDIOHOOK_TYPE_SPY:
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| 		ast_slinfactory_init(&audiohook->read_factory);
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| 	case AST_AUDIOHOOK_TYPE_WHISPER:
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| 		ast_slinfactory_init(&audiohook->write_factory);
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| 		break;
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| 	default:
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| 		break;
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| 	}
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| 
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| 	/* Since we are just starting out... this audiohook is new */
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| 	audiohook->status = AST_AUDIOHOOK_STATUS_NEW;
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| 
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| 	return 0;
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| }
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| 
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| /*! \brief Destroys an audiohook structure
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|  * \param audiohook Audiohook structure
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|  * \return Returns 0 on success, -1 on failure
 | |
|  */
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| int ast_audiohook_destroy(struct ast_audiohook *audiohook)
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| {
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| 	/* Drop the factories used by this audiohook type */
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| 	switch (audiohook->type) {
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| 	case AST_AUDIOHOOK_TYPE_SPY:
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| 		ast_slinfactory_destroy(&audiohook->read_factory);
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| 	case AST_AUDIOHOOK_TYPE_WHISPER:
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| 		ast_slinfactory_destroy(&audiohook->write_factory);
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| 		break;
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| 	default:
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| 		break;
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| 	}
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| 
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| 	/* Destroy translation path if present */
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| 	if (audiohook->trans_pvt)
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| 		ast_translator_free_path(audiohook->trans_pvt);
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| 
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| 	/* Lock and trigger be gone! */
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| 	ast_cond_destroy(&audiohook->trigger);
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| 	ast_mutex_destroy(&audiohook->lock);
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| 
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| 	return 0;
 | |
| }
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| 
 | |
| /*! \brief Writes a frame into the audiohook structure
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|  * \param audiohook Audiohook structure
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|  * \param direction Direction the audio frame came from
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|  * \param frame Frame to write in
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|  * \return Returns 0 on success, -1 on failure
 | |
|  */
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| int ast_audiohook_write_frame(struct ast_audiohook *audiohook, enum ast_audiohook_direction direction, struct ast_frame *frame)
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| {
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| 	struct ast_slinfactory *factory = (direction == AST_AUDIOHOOK_DIRECTION_READ ? &audiohook->read_factory : &audiohook->write_factory);
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| 
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| 	/* Write frame out to respective factory */
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| 	ast_slinfactory_feed(factory, frame);
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| 
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| 	/* If we need to notify the respective handler of this audiohook, do so */
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| 	switch (ast_test_flag(audiohook, AST_AUDIOHOOK_TRIGGER_MODE)) {
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| 	case AST_AUDIOHOOK_TRIGGER_READ:
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| 		if (direction == AST_AUDIOHOOK_DIRECTION_READ)
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| 			ast_cond_signal(&audiohook->trigger);
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| 		break;
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| 	case AST_AUDIOHOOK_TRIGGER_WRITE:
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| 		if (direction == AST_AUDIOHOOK_DIRECTION_WRITE)
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| 			ast_cond_signal(&audiohook->trigger);
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| 		break;
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| 	default:
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| 		break;
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| 	}
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| 
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| 	return 0;
 | |
| }
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| 
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| static struct ast_frame *audiohook_read_frame_single(struct ast_audiohook *audiohook, size_t samples, enum ast_audiohook_direction direction)
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| {
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| 	struct ast_slinfactory *factory = (direction == AST_AUDIOHOOK_DIRECTION_READ ? &audiohook->read_factory : &audiohook->write_factory);
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| 	int vol = (direction == AST_AUDIOHOOK_DIRECTION_READ ? audiohook->options.read_volume : audiohook->options.write_volume);
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| 	short buf[samples];
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| 	struct ast_frame frame = {
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| 		.frametype = AST_FRAME_VOICE,
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| 		.subclass = AST_FORMAT_SLINEAR,
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| 		.data = buf,
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| 		.datalen = sizeof(buf),
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| 		.samples = samples,
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| 	};
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| 
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| 	/* Ensure the factory is able to give us the samples we want */
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| 	if (samples > ast_slinfactory_available(factory))
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| 		return NULL;
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| 	
 | |
| 	/* Read data in from factory */
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| 	if (!ast_slinfactory_read(factory, buf, samples))
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| 		return NULL;
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| 
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| 	/* If a volume adjustment needs to be applied apply it */
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| 	if (vol)
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| 		ast_frame_adjust_volume(&frame, vol);
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| 
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| 	return ast_frdup(&frame);
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| }
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| 
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| static struct ast_frame *audiohook_read_frame_both(struct ast_audiohook *audiohook, size_t samples)
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| {
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| 	int i = 0;
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| 	short buf1[samples], buf2[samples], *read_buf = NULL, *write_buf = NULL, *final_buf = NULL, *data1 = NULL, *data2 = NULL;
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| 	struct ast_frame frame = {
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| 		.frametype = AST_FRAME_VOICE,
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| 		.subclass = AST_FORMAT_SLINEAR,
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| 		.data = NULL,
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| 		.datalen = sizeof(buf1),
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| 		.samples = samples,
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| 	};
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| 
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| 	/* Start with the read factory... if there are enough samples, read them in */
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| 	if (ast_slinfactory_available(&audiohook->read_factory) >= samples) {
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| 		if (ast_slinfactory_read(&audiohook->read_factory, buf1, samples)) {
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| 			read_buf = buf1;
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| 			/* Adjust read volume if need be */
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| 			if (audiohook->options.read_volume) {
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| 				int count = 0;
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| 				short adjust_value = abs(audiohook->options.read_volume);
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| 				for (count = 0; count < samples; count++) {
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| 					if (audiohook->options.read_volume > 0)
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| 						ast_slinear_saturated_multiply(&buf1[count], &adjust_value);
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| 					else if (audiohook->options.read_volume < 0)
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| 						ast_slinear_saturated_divide(&buf1[count], &adjust_value);
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| 				}
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| 			}
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| 		}
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| 	} else if (option_debug)
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| 		ast_log(LOG_DEBUG, "Failed to get %d samples from read factory %p\n", (int)samples, &audiohook->read_factory);
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| 
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| 	/* Move on to the write factory... if there are enough samples, read them in */
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| 	if (ast_slinfactory_available(&audiohook->write_factory) >= samples) {
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| 		if (ast_slinfactory_read(&audiohook->write_factory, buf2, samples)) {
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| 			write_buf = buf2;
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| 			/* Adjust write volume if need be */
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| 			if (audiohook->options.write_volume) {
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| 				int count = 0;
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| 				short adjust_value = abs(audiohook->options.write_volume);
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| 				for (count = 0; count < samples; count++) {
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| 					if (audiohook->options.write_volume > 0)
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| 						ast_slinear_saturated_multiply(&buf2[count], &adjust_value);
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| 					else if (audiohook->options.write_volume < 0)
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| 						ast_slinear_saturated_divide(&buf2[count], &adjust_value);
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| 				}
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| 			}
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| 		}
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| 	} else if (option_debug)
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| 		ast_log(LOG_DEBUG, "Failed to get %d samples from write factory %p\n", (int)samples, &audiohook->write_factory);
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| 
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| 	/* Basically we figure out which buffer to use... and if mixing can be done here */
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| 	if (!read_buf && !write_buf)
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| 		return NULL;
 | |
| 	else if (read_buf && write_buf) {
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| 		for (i = 0, data1 = read_buf, data2 = write_buf; i < samples; i++, data1++, data2++)
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| 			ast_slinear_saturated_add(data1, data2);
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| 		final_buf = buf1;
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| 	} else if (read_buf)
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| 		final_buf = buf1;
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| 	else if (write_buf)
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| 		final_buf = buf2;
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| 
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| 	/* Make the final buffer part of the frame, so it gets duplicated fine */
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| 	frame.data = final_buf;
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| 
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| 	/* Yahoo, a combined copy of the audio! */
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| 	return ast_frdup(&frame);
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| }
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| 
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| /*! \brief Reads a frame in from the audiohook structure
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|  * \param audiohook Audiohook structure
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|  * \param samples Number of samples wanted
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|  * \param direction Direction the audio frame came from
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|  * \param format Format of frame remote side wants back
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|  * \return Returns frame on success, NULL on failure
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|  */
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| struct ast_frame *ast_audiohook_read_frame(struct ast_audiohook *audiohook, size_t samples, enum ast_audiohook_direction direction, int format)
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| {
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| 	struct ast_frame *read_frame = NULL, *final_frame = NULL;
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| 
 | |
| 	if (!(read_frame = (direction == AST_AUDIOHOOK_DIRECTION_BOTH ? audiohook_read_frame_both(audiohook, samples) : audiohook_read_frame_single(audiohook, samples, direction))))
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| 		return NULL;
 | |
| 
 | |
| 	/* If they don't want signed linear back out, we'll have to send it through the translation path */
 | |
| 	if (format != AST_FORMAT_SLINEAR) {
 | |
| 		/* Rebuild translation path if different format then previously */
 | |
| 		if (audiohook->format != format) {
 | |
| 			if (audiohook->trans_pvt) {
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| 				ast_translator_free_path(audiohook->trans_pvt);
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| 				audiohook->trans_pvt = NULL;
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| 			}
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| 			/* Setup new translation path for this format... if we fail we can't very well return signed linear so free the frame and return nothing */
 | |
| 			if (!(audiohook->trans_pvt = ast_translator_build_path(format, AST_FORMAT_SLINEAR))) {
 | |
| 				ast_frfree(read_frame);
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| 				return NULL;
 | |
| 			}
 | |
| 		}
 | |
| 		/* Convert to requested format, and allow the read in frame to be freed */
 | |
| 		final_frame = ast_translate(audiohook->trans_pvt, read_frame, 1);
 | |
| 	} else {
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| 		final_frame = read_frame;
 | |
| 	}
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| 
 | |
| 	return final_frame;
 | |
| }
 | |
| 
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| /*! \brief Attach audiohook to channel
 | |
|  * \param chan Channel
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|  * \param audiohook Audiohook structure
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|  * \return Returns 0 on success, -1 on failure
 | |
|  */
 | |
| int ast_audiohook_attach(struct ast_channel *chan, struct ast_audiohook *audiohook)
 | |
| {
 | |
| 	ast_channel_lock(chan);
 | |
| 
 | |
| 	if (!chan->audiohooks) {
 | |
| 		/* Whoops... allocate a new structure */
 | |
| 		if (!(chan->audiohooks = ast_calloc(1, sizeof(*chan->audiohooks)))) {
 | |
| 			ast_channel_unlock(chan);
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| 			return -1;
 | |
| 		}
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| 		AST_LIST_HEAD_INIT_NOLOCK(&chan->audiohooks->spy_list);
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| 		AST_LIST_HEAD_INIT_NOLOCK(&chan->audiohooks->whisper_list);
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| 		AST_LIST_HEAD_INIT_NOLOCK(&chan->audiohooks->manipulate_list);
 | |
| 	}
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| 
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| 	/* Drop into respective list */
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| 	if (audiohook->type == AST_AUDIOHOOK_TYPE_SPY)
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| 		AST_LIST_INSERT_TAIL(&chan->audiohooks->spy_list, audiohook, list);
 | |
| 	else if (audiohook->type == AST_AUDIOHOOK_TYPE_WHISPER)
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| 		AST_LIST_INSERT_TAIL(&chan->audiohooks->whisper_list, audiohook, list);
 | |
| 	else if (audiohook->type == AST_AUDIOHOOK_TYPE_MANIPULATE)
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| 		AST_LIST_INSERT_TAIL(&chan->audiohooks->manipulate_list, audiohook, list);
 | |
| 
 | |
| 	/* Change status over to running since it is now attached */
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| 	audiohook->status = AST_AUDIOHOOK_STATUS_RUNNING;
 | |
| 
 | |
| 	ast_channel_unlock(chan);
 | |
| 
 | |
| 	return 0;
 | |
| }
 | |
| 
 | |
| /*! \brief Detach audiohook from channel
 | |
|  * \param audiohook Audiohook structure
 | |
|  * \return Returns 0 on success, -1 on failure
 | |
|  */
 | |
| int ast_audiohook_detach(struct ast_audiohook *audiohook)
 | |
| {
 | |
| 	if (audiohook->status == AST_AUDIOHOOK_STATUS_DONE)
 | |
| 		return 0;
 | |
| 
 | |
| 	audiohook->status = AST_AUDIOHOOK_STATUS_SHUTDOWN;
 | |
| 
 | |
| 	while (audiohook->status != AST_AUDIOHOOK_STATUS_DONE)
 | |
| 		ast_audiohook_trigger_wait(audiohook);
 | |
| 
 | |
| 	return 0;
 | |
| }
 | |
| 
 | |
| /*! \brief Detach audiohooks from list and destroy said list
 | |
|  * \param audiohook_list List of audiohooks
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|  * \return Returns 0 on success, -1 on failure
 | |
|  */
 | |
| int ast_audiohook_detach_list(struct ast_audiohook_list *audiohook_list)
 | |
| {
 | |
| 	int i = 0;
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| 	struct ast_audiohook *audiohook = NULL;
 | |
| 
 | |
| 	/* Drop any spies */
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| 	while ((audiohook = AST_LIST_REMOVE_HEAD(&audiohook_list->spy_list, list))) {
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| 		ast_audiohook_lock(audiohook);
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| 		audiohook->status = AST_AUDIOHOOK_STATUS_DONE;
 | |
| 		ast_cond_signal(&audiohook->trigger);
 | |
| 		ast_audiohook_unlock(audiohook);
 | |
| 	}
 | |
| 
 | |
| 	/* Drop any whispering sources */
 | |
| 	while ((audiohook = AST_LIST_REMOVE_HEAD(&audiohook_list->whisper_list, list))) {
 | |
| 		ast_audiohook_lock(audiohook);
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| 		audiohook->status = AST_AUDIOHOOK_STATUS_DONE;
 | |
| 		ast_cond_signal(&audiohook->trigger);
 | |
| 		ast_audiohook_unlock(audiohook);
 | |
| 	}
 | |
| 
 | |
| 	/* Drop any manipulaters */
 | |
| 	while ((audiohook = AST_LIST_REMOVE_HEAD(&audiohook_list->manipulate_list, list))) {
 | |
| 		ast_audiohook_lock(audiohook);
 | |
| 		ast_mutex_lock(&audiohook->lock);
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| 		audiohook->status = AST_AUDIOHOOK_STATUS_DONE;
 | |
| 		ast_audiohook_unlock(audiohook);
 | |
| 		audiohook->manipulate_callback(audiohook, NULL, NULL, 0);
 | |
| 	}
 | |
| 
 | |
| 	/* Drop translation paths if present */
 | |
| 	for (i = 0; i < 2; i++) {
 | |
| 		if (audiohook_list->in_translate[i].trans_pvt)
 | |
| 			ast_translator_free_path(audiohook_list->in_translate[i].trans_pvt);
 | |
| 		if (audiohook_list->out_translate[i].trans_pvt)
 | |
| 			ast_translator_free_path(audiohook_list->out_translate[i].trans_pvt);
 | |
| 	}
 | |
| 	
 | |
| 	/* Free ourselves */
 | |
| 	ast_free(audiohook_list);
 | |
| 
 | |
| 	return 0;
 | |
| }
 | |
| 
 | |
| static struct ast_audiohook *find_audiohook_by_source(struct ast_audiohook_list *audiohook_list, const char *source)
 | |
| {
 | |
| 	struct ast_audiohook *audiohook = NULL;
 | |
| 
 | |
| 	AST_LIST_TRAVERSE(&audiohook_list->spy_list, audiohook, list) {
 | |
| 		if (!strcasecmp(audiohook->source, source))
 | |
| 			return audiohook;
 | |
| 	}
 | |
| 
 | |
| 	AST_LIST_TRAVERSE(&audiohook_list->whisper_list, audiohook, list) {
 | |
| 		if (!strcasecmp(audiohook->source, source))
 | |
| 			return audiohook;
 | |
| 	}
 | |
| 
 | |
| 	AST_LIST_TRAVERSE(&audiohook_list->manipulate_list, audiohook, list) {
 | |
| 		if (!strcasecmp(audiohook->source, source))
 | |
| 			return audiohook;
 | |
| 	}
 | |
| 
 | |
| 	return NULL;
 | |
| }
 | |
| 
 | |
| /*! \brief Detach specified source audiohook from channel
 | |
|  * \param chan Channel to detach from
 | |
|  * \param source Name of source to detach
 | |
|  * \return Returns 0 on success, -1 on failure
 | |
|  */
 | |
| int ast_audiohook_detach_source(struct ast_channel *chan, const char *source)
 | |
| {
 | |
| 	struct ast_audiohook *audiohook = NULL;
 | |
| 
 | |
| 	ast_channel_lock(chan);
 | |
| 
 | |
| 	/* Ensure the channel has audiohooks on it */
 | |
| 	if (!chan->audiohooks) {
 | |
| 		ast_channel_unlock(chan);
 | |
| 		return -1;
 | |
| 	}
 | |
| 
 | |
| 	audiohook = find_audiohook_by_source(chan->audiohooks, source);
 | |
| 
 | |
| 	ast_channel_unlock(chan);
 | |
| 
 | |
| 	if (audiohook && audiohook->status != AST_AUDIOHOOK_STATUS_DONE)
 | |
| 		audiohook->status = AST_AUDIOHOOK_STATUS_SHUTDOWN;
 | |
| 
 | |
| 	return (audiohook ? 0 : -1);
 | |
| }
 | |
| 
 | |
| /*! \brief Pass a DTMF frame off to be handled by the audiohook core
 | |
|  * \param chan Channel that the list is coming off of
 | |
|  * \param audiohook_list List of audiohooks
 | |
|  * \param direction Direction frame is coming in from
 | |
|  * \param frame The frame itself
 | |
|  * \return Return frame on success, NULL on failure
 | |
|  */
 | |
| static struct ast_frame *dtmf_audiohook_write_list(struct ast_channel *chan, struct ast_audiohook_list *audiohook_list, enum ast_audiohook_direction direction, struct ast_frame *frame)
 | |
| {
 | |
| 	struct ast_audiohook *audiohook = NULL;
 | |
| 
 | |
| 	AST_LIST_TRAVERSE_SAFE_BEGIN(&audiohook_list->manipulate_list, audiohook, list) {
 | |
| 		ast_audiohook_lock(audiohook);
 | |
| 		if (audiohook->status != AST_AUDIOHOOK_STATUS_RUNNING) {
 | |
| 			AST_LIST_REMOVE_CURRENT(list);
 | |
| 			audiohook->status = AST_AUDIOHOOK_STATUS_DONE;
 | |
| 			ast_audiohook_unlock(audiohook);
 | |
| 			audiohook->manipulate_callback(audiohook, NULL, NULL, 0);
 | |
| 			continue;
 | |
| 		}
 | |
| 		if (ast_test_flag(audiohook, AST_AUDIOHOOK_WANTS_DTMF))
 | |
| 			audiohook->manipulate_callback(audiohook, chan, frame, direction);
 | |
| 		ast_audiohook_unlock(audiohook);
 | |
| 	}
 | |
| 	AST_LIST_TRAVERSE_SAFE_END;
 | |
| 
 | |
| 	return frame;
 | |
| }
 | |
| 
 | |
| /*! \brief Pass an AUDIO frame off to be handled by the audiohook core
 | |
|  * \param chan Channel that the list is coming off of
 | |
|  * \param audiohook_list List of audiohooks
 | |
|  * \param direction Direction frame is coming in from
 | |
|  * \param frame The frame itself
 | |
|  * \return Return frame on success, NULL on failure
 | |
|  */
 | |
| static struct ast_frame *audio_audiohook_write_list(struct ast_channel *chan, struct ast_audiohook_list *audiohook_list, enum ast_audiohook_direction direction, struct ast_frame *frame)
 | |
| {
 | |
| 	struct ast_audiohook_translate *in_translate = (direction == AST_AUDIOHOOK_DIRECTION_READ ? &audiohook_list->in_translate[0] : &audiohook_list->in_translate[1]);
 | |
| 	struct ast_audiohook_translate *out_translate = (direction == AST_AUDIOHOOK_DIRECTION_READ ? &audiohook_list->out_translate[0] : &audiohook_list->out_translate[1]);
 | |
| 	struct ast_frame *start_frame = frame, *middle_frame = frame, *end_frame = frame;
 | |
| 	struct ast_audiohook *audiohook = NULL;
 | |
| 	int samples = frame->samples;
 | |
| 	
 | |
| 	/* If the frame coming in is not signed linear we have to send it through the in_translate path */
 | |
| 	if (frame->subclass != AST_FORMAT_SLINEAR) {
 | |
| 		if (in_translate->format != frame->subclass) {
 | |
| 			if (in_translate->trans_pvt)
 | |
| 				ast_translator_free_path(in_translate->trans_pvt);
 | |
| 			if (!(in_translate->trans_pvt = ast_translator_build_path(AST_FORMAT_SLINEAR, frame->subclass)))
 | |
| 				return frame;
 | |
| 			in_translate->format = frame->subclass;
 | |
| 		}
 | |
| 		if (!(middle_frame = ast_translate(in_translate->trans_pvt, frame, 0)))
 | |
| 			return frame;
 | |
| 	}
 | |
| 
 | |
| 	/* Queue up signed linear frame to each spy */
 | |
| 	AST_LIST_TRAVERSE_SAFE_BEGIN(&audiohook_list->spy_list, audiohook, list) {
 | |
| 		ast_audiohook_lock(audiohook);
 | |
| 		if (audiohook->status != AST_AUDIOHOOK_STATUS_RUNNING) {
 | |
| 			AST_LIST_REMOVE_CURRENT(list);
 | |
| 			audiohook->status = AST_AUDIOHOOK_STATUS_DONE;
 | |
| 			ast_cond_signal(&audiohook->trigger);
 | |
| 			ast_audiohook_unlock(audiohook);
 | |
| 			continue;
 | |
| 		}
 | |
| 		ast_audiohook_write_frame(audiohook, direction, middle_frame);
 | |
| 		ast_audiohook_unlock(audiohook);
 | |
| 	}
 | |
| 	AST_LIST_TRAVERSE_SAFE_END
 | |
| 
 | |
| 	/* If this frame is being written out to the channel then we need to use whisper sources */
 | |
| 	if (direction == AST_AUDIOHOOK_DIRECTION_WRITE && !AST_LIST_EMPTY(&audiohook_list->whisper_list)) {
 | |
| 		int i = 0;
 | |
| 		short read_buf[samples], combine_buf[samples], *data1 = NULL, *data2 = NULL;
 | |
| 		memset(&combine_buf, 0, sizeof(combine_buf));
 | |
| 		AST_LIST_TRAVERSE_SAFE_BEGIN(&audiohook_list->whisper_list, audiohook, list) {
 | |
| 			ast_audiohook_lock(audiohook);
 | |
| 			if (audiohook->status != AST_AUDIOHOOK_STATUS_RUNNING) {
 | |
| 				AST_LIST_REMOVE_CURRENT(list);
 | |
| 				audiohook->status = AST_AUDIOHOOK_STATUS_DONE;
 | |
| 				ast_cond_signal(&audiohook->trigger);
 | |
| 				ast_audiohook_unlock(audiohook);
 | |
| 				continue;
 | |
| 			}
 | |
| 			if (ast_slinfactory_available(&audiohook->write_factory) >= samples && ast_slinfactory_read(&audiohook->write_factory, read_buf, samples)) {
 | |
| 				/* Take audio from this whisper source and combine it into our main buffer */
 | |
| 				for (i = 0, data1 = combine_buf, data2 = read_buf; i < samples; i++, data1++, data2++)
 | |
| 					ast_slinear_saturated_add(data1, data2);
 | |
| 			}
 | |
| 			ast_audiohook_unlock(audiohook);
 | |
| 		}
 | |
| 		AST_LIST_TRAVERSE_SAFE_END
 | |
| 		/* We take all of the combined whisper sources and combine them into the audio being written out */
 | |
| 		for (i = 0, data1 = middle_frame->data, data2 = combine_buf; i < samples; i++, data1++, data2++)
 | |
| 			ast_slinear_saturated_add(data1, data2);
 | |
| 		end_frame = middle_frame;
 | |
| 	}
 | |
| 
 | |
| 	/* Pass off frame to manipulate audiohooks */
 | |
| 	if (!AST_LIST_EMPTY(&audiohook_list->manipulate_list)) {
 | |
| 		AST_LIST_TRAVERSE_SAFE_BEGIN(&audiohook_list->manipulate_list, audiohook, list) {
 | |
| 			ast_audiohook_lock(audiohook);
 | |
| 			if (audiohook->status != AST_AUDIOHOOK_STATUS_RUNNING) {
 | |
| 				AST_LIST_REMOVE_CURRENT(list);
 | |
| 				audiohook->status = AST_AUDIOHOOK_STATUS_DONE;
 | |
| 				ast_audiohook_unlock(audiohook);
 | |
| 				/* We basically drop all of our links to the manipulate audiohook and prod it to do it's own destructive things */
 | |
| 				audiohook->manipulate_callback(audiohook, chan, NULL, direction);
 | |
| 				continue;
 | |
| 			}
 | |
| 			/* Feed in frame to manipulation */
 | |
| 			audiohook->manipulate_callback(audiohook, chan, middle_frame, direction);
 | |
| 			ast_audiohook_unlock(audiohook);
 | |
| 		}
 | |
| 		AST_LIST_TRAVERSE_SAFE_END
 | |
| 		end_frame = middle_frame;
 | |
| 	}
 | |
| 
 | |
| 	/* Now we figure out what to do with our end frame (whether to transcode or not) */
 | |
| 	if (middle_frame == end_frame) {
 | |
| 		/* Middle frame was modified and became the end frame... let's see if we need to transcode */
 | |
| 		if (end_frame->subclass != start_frame->subclass) {
 | |
| 			if (out_translate->format != start_frame->subclass) {
 | |
| 				if (out_translate->trans_pvt)
 | |
| 					ast_translator_free_path(out_translate->trans_pvt);
 | |
| 				if (!(out_translate->trans_pvt = ast_translator_build_path(start_frame->subclass, AST_FORMAT_SLINEAR))) {
 | |
| 					/* We can't transcode this... drop our middle frame and return the original */
 | |
| 					ast_frfree(middle_frame);
 | |
| 					return start_frame;
 | |
| 				}
 | |
| 				out_translate->format = start_frame->subclass;
 | |
| 			}
 | |
| 			/* Transcode from our middle (signed linear) frame to new format of the frame that came in */
 | |
| 			if (!(end_frame = ast_translate(out_translate->trans_pvt, middle_frame, 0))) {
 | |
| 				/* Failed to transcode the frame... drop it and return the original */
 | |
| 				ast_frfree(middle_frame);
 | |
| 				return start_frame;
 | |
| 			}
 | |
| 			/* Here's the scoop... middle frame is no longer of use to us */
 | |
| 			ast_frfree(middle_frame);
 | |
| 		}
 | |
| 	} else {
 | |
| 		/* No frame was modified, we can just drop our middle frame and pass the frame we got in out */
 | |
| 		ast_frfree(middle_frame);
 | |
| 	}
 | |
| 
 | |
| 	return end_frame;
 | |
| }
 | |
| 
 | |
| /*! \brief Pass a frame off to be handled by the audiohook core
 | |
|  * \param chan Channel that the list is coming off of
 | |
|  * \param audiohook_list List of audiohooks
 | |
|  * \param direction Direction frame is coming in from
 | |
|  * \param frame The frame itself
 | |
|  * \return Return frame on success, NULL on failure
 | |
|  */
 | |
| struct ast_frame *ast_audiohook_write_list(struct ast_channel *chan, struct ast_audiohook_list *audiohook_list, enum ast_audiohook_direction direction, struct ast_frame *frame)
 | |
| {
 | |
| 	/* Pass off frame to it's respective list write function */
 | |
| 	if (frame->frametype == AST_FRAME_VOICE)
 | |
| 		return audio_audiohook_write_list(chan, audiohook_list, direction, frame);
 | |
| 	else if (frame->frametype == AST_FRAME_DTMF)
 | |
| 		return dtmf_audiohook_write_list(chan, audiohook_list, direction, frame);
 | |
| 	else
 | |
| 		return frame;
 | |
| }
 | |
| 			
 | |
| 
 | |
| /*! \brief Wait for audiohook trigger to be triggered
 | |
|  * \param audiohook Audiohook to wait on
 | |
|  */
 | |
| void ast_audiohook_trigger_wait(struct ast_audiohook *audiohook)
 | |
| {
 | |
| 	struct timeval tv;
 | |
| 	struct timespec ts;
 | |
| 
 | |
| 	tv = ast_tvadd(ast_tvnow(), ast_samp2tv(50000, 1000));
 | |
| 	ts.tv_sec = tv.tv_sec;
 | |
| 	ts.tv_nsec = tv.tv_usec * 1000;
 | |
| 	
 | |
| 	ast_cond_timedwait(&audiohook->trigger, &audiohook->lock, &ts);
 | |
| 	
 | |
| 	return;
 | |
| }
 | |
| 
 | |
| /* Count number of channel audiohooks by type, regardless of type */
 | |
| int ast_channel_audiohook_count_by_source(struct ast_channel *chan, const char *source, enum ast_audiohook_type type)
 | |
| {
 | |
| 	int count = 0;
 | |
| 	struct ast_audiohook *ah = NULL;
 | |
| 
 | |
| 	if (!chan->audiohooks)
 | |
| 		return -1;
 | |
| 
 | |
| 	switch (type) {
 | |
| 		case AST_AUDIOHOOK_TYPE_SPY:
 | |
| 			AST_LIST_TRAVERSE_SAFE_BEGIN(&chan->audiohooks->spy_list, ah, list) {
 | |
| 				if (!strcmp(ah->source, source)) {
 | |
| 					count++;
 | |
| 				}
 | |
| 			}
 | |
| 			AST_LIST_TRAVERSE_SAFE_END;
 | |
| 			break;
 | |
| 		case AST_AUDIOHOOK_TYPE_WHISPER:
 | |
| 			AST_LIST_TRAVERSE_SAFE_BEGIN(&chan->audiohooks->whisper_list, ah, list) {
 | |
| 				if (!strcmp(ah->source, source)) {
 | |
| 					count++;
 | |
| 				}
 | |
| 			}
 | |
| 			AST_LIST_TRAVERSE_SAFE_END;
 | |
| 			break;
 | |
| 		case AST_AUDIOHOOK_TYPE_MANIPULATE:
 | |
| 			AST_LIST_TRAVERSE_SAFE_BEGIN(&chan->audiohooks->manipulate_list, ah, list) {
 | |
| 				if (!strcmp(ah->source, source)) {
 | |
| 					count++;
 | |
| 				}
 | |
| 			}
 | |
| 			AST_LIST_TRAVERSE_SAFE_END;
 | |
| 			break;
 | |
| 		default:
 | |
| 			ast_log(LOG_DEBUG, "Invalid audiohook type supplied, (%d)\n", type);
 | |
| 			return -1;
 | |
| 	}
 | |
| 
 | |
| 	return count;
 | |
| }
 | |
| 
 | |
| /* Count number of channel audiohooks by type that are running */
 | |
| int ast_channel_audiohook_count_by_source_running(struct ast_channel *chan, const char *source, enum ast_audiohook_type type)
 | |
| {
 | |
| 	int count = 0;
 | |
| 	struct ast_audiohook *ah = NULL;
 | |
| 	if (!chan->audiohooks)
 | |
| 		return -1;
 | |
| 
 | |
| 	switch (type) {
 | |
| 		case AST_AUDIOHOOK_TYPE_SPY:
 | |
| 			AST_LIST_TRAVERSE_SAFE_BEGIN(&chan->audiohooks->spy_list, ah, list) {
 | |
| 				if ((!strcmp(ah->source, source)) && (ah->status == AST_AUDIOHOOK_STATUS_RUNNING))
 | |
| 					count++;
 | |
| 			}
 | |
| 			AST_LIST_TRAVERSE_SAFE_END;
 | |
| 			break;
 | |
| 		case AST_AUDIOHOOK_TYPE_WHISPER:
 | |
| 			AST_LIST_TRAVERSE_SAFE_BEGIN(&chan->audiohooks->whisper_list, ah, list) {
 | |
| 				if ((!strcmp(ah->source, source)) && (ah->status == AST_AUDIOHOOK_STATUS_RUNNING))
 | |
| 					count++;
 | |
| 			}
 | |
| 			AST_LIST_TRAVERSE_SAFE_END;
 | |
| 			break;
 | |
| 		case AST_AUDIOHOOK_TYPE_MANIPULATE:
 | |
| 			AST_LIST_TRAVERSE_SAFE_BEGIN(&chan->audiohooks->manipulate_list, ah, list) {
 | |
| 				if ((!strcmp(ah->source, source)) && (ah->status == AST_AUDIOHOOK_STATUS_RUNNING))
 | |
| 					count++;
 | |
| 			}
 | |
| 			AST_LIST_TRAVERSE_SAFE_END;
 | |
| 			break;
 | |
| 		default:
 | |
| 			ast_log(LOG_DEBUG, "Invalid audiohook type supplied, (%d)\n", type);
 | |
| 			return -1;
 | |
| 	}
 | |
| 	return count;
 | |
| }
 | |
| 
 |