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/*
* Asterisk -- An open source telephony toolkit.
*
* Copyright (C) 2011, Digium, Inc.
*
* 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 SILK Format Attributes
*
* \author David Vossel <dvossel@digium.com>
*/
#ifndef _AST_FORMAT_SILK_H_
#define _AST_FORMAT_SILK_H_
/*! SILK format attribute key value pairs, all are accessible through ast_format_get_value()*/
enum silk_attr_keys {
SILK_ATTR_KEY_SAMP_RATE, /*!< value is silk_attr_vals enum */
SILK_ATTR_KEY_DTX, /*!< value is an int, 1 dtx is enabled, 0 dtx not enabled. */
SILK_ATTR_KEY_FEC, /*!< value is an int, 1 encode with FEC, 0 do not use FEC. */
SILK_ATTR_KEY_PACKETLOSS_PERCENTAGE, /*!< value is an int (0-100), Represents estimated packetloss in uplink direction.*/
SILK_ATTR_KEY_MAX_BITRATE, /*!< value is an int */
};
enum silk_attr_vals {
SILK_ATTR_VAL_SAMP_8KHZ = (1 << 0),
SILK_ATTR_VAL_SAMP_12KHZ = (1 << 1),
SILK_ATTR_VAL_SAMP_16KHZ = (1 << 2),
SILK_ATTR_VAL_SAMP_24KHZ = (1 << 3),
};
#endif /* _AST_FORMAT_SILK_H */
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