/*
 * Asterisk -- An open source telephony toolkit.
 *
 * Copyright (C) 1999 - 2005, Digium, Inc.
 *
 * Mark Spencer <markster@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 Convenient Signal Processing routines
 */

#ifndef _ASTERISK_DSP_H
#define _ASTERISK_DSP_H

#define DSP_FEATURE_SILENCE_SUPPRESS	(1 << 0)
#define DSP_FEATURE_BUSY_DETECT		(1 << 1)
#define DSP_FEATURE_DTMF_DETECT		(1 << 3)
#define DSP_FEATURE_FAX_DETECT		(1 << 4)

#define	DSP_DIGITMODE_DTMF			0				/* Detect DTMF digits */
#define DSP_DIGITMODE_MF			1				/* Detect MF digits */

#define DSP_DIGITMODE_NOQUELCH		(1 << 8)		/* Do not quelch DTMF from in-band */
#define DSP_DIGITMODE_MUTECONF		(1 << 9)		/* Mute conference */
#define DSP_DIGITMODE_MUTEMAX		(1 << 10)		/* Delay audio by a frame to try to extra quelch */
#define DSP_DIGITMODE_RELAXDTMF		(1 << 11)		/* "Radio" mode (relaxed DTMF) */

#define DSP_PROGRESS_TALK		(1 << 16)		/* Enable talk detection */
#define DSP_PROGRESS_RINGING		(1 << 17)		/* Enable calling tone detection */
#define DSP_PROGRESS_BUSY		(1 << 18)		/* Enable busy tone detection */
#define DSP_PROGRESS_CONGESTION		(1 << 19)		/* Enable congestion tone detection */
#define DSP_FEATURE_CALL_PROGRESS	(DSP_PROGRESS_TALK | DSP_PROGRESS_RINGING | DSP_PROGRESS_BUSY | DSP_PROGRESS_CONGESTION)

#define DSP_TONE_STATE_SILENCE  0
#define DSP_TONE_STATE_RINGING  1 
#define DSP_TONE_STATE_DIALTONE 2
#define DSP_TONE_STATE_TALKING  3
#define DSP_TONE_STATE_BUSY     4
#define DSP_TONE_STATE_SPECIAL1	5
#define DSP_TONE_STATE_SPECIAL2 6
#define DSP_TONE_STATE_SPECIAL3 7
#define DSP_TONE_STATE_HUNGUP 	8

struct ast_dsp;

struct ast_dsp *ast_dsp_new(void);
void ast_dsp_free(struct ast_dsp *dsp);

/*! \brief Set threshold value for silence */
void ast_dsp_set_threshold(struct ast_dsp *dsp, int threshold);

/*! \brief Set number of required cadences for busy */
void ast_dsp_set_busy_count(struct ast_dsp *dsp, int cadences);

/*! \brief Set expected lengths of the busy tone */
void ast_dsp_set_busy_pattern(struct ast_dsp *dsp, int tonelength, int quietlength);

/*! \brief Scans for progress indication in audio */
int ast_dsp_call_progress(struct ast_dsp *dsp, struct ast_frame *inf);

/*! \brief Set zone for doing progress detection */
int ast_dsp_set_call_progress_zone(struct ast_dsp *dsp, char *zone);

/*! \brief Return AST_FRAME_NULL frames when there is silence, AST_FRAME_BUSY on 
   busies, and call progress, all dependent upon which features are enabled */
struct ast_frame *ast_dsp_process(struct ast_channel *chan, struct ast_dsp *dsp, struct ast_frame *inf);

/*! \brief Return non-zero if this is silence.  Updates "totalsilence" with the total
   number of seconds of silence  */
int ast_dsp_silence(struct ast_dsp *dsp, struct ast_frame *f, int *totalsilence);

/*! \brief Return non-zero if historically this should be a busy, request that
  ast_dsp_silence has already been called */
int ast_dsp_busydetect(struct ast_dsp *dsp);

/*! \brief Return non-zero if DTMF hit was found */
int ast_dsp_digitdetect(struct ast_dsp *dsp, struct ast_frame *f);

/*! \brief Reset total silence count */
void ast_dsp_reset(struct ast_dsp *dsp);

/*! \brief Reset DTMF detector */
void ast_dsp_digitreset(struct ast_dsp *dsp);

/*! \brief Select feature set */
void ast_dsp_set_features(struct ast_dsp *dsp, int features);

/*! \brief Get pending DTMF/MF digits */
int ast_dsp_getdigits(struct ast_dsp *dsp, char *buf, int max);

/*! \brief Set digit mode */
int ast_dsp_digitmode(struct ast_dsp *dsp, int digitmode);

/*! \brief Get tstate (Tone State) */
int ast_dsp_get_tstate(struct ast_dsp *dsp);

/*! \brief Get tcount (Threshold counter) */
int ast_dsp_get_tcount(struct ast_dsp *dsp);

#endif /* _ASTERISK_DSP_H */