2470 lines
74 KiB
C
2470 lines
74 KiB
C
/*
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* Asterisk -- An open source telephony toolkit.
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*
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* Copyright (C) 1999 - 2005, Digium, Inc.
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*
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* Mark Spencer <markster@digium.com>
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*
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* Goertzel routines are borrowed from Steve Underwood's tremendous work on the
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* DTMF detector.
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*
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* See http://www.asterisk.org for more information about
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* the Asterisk project. Please do not directly contact
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* any of the maintainers of this project for assistance;
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* 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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*/
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/*! \file
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*
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* \brief Convenience Signal Processing routines
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*
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* \author Mark Spencer <markster@digium.com>
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* \author Steve Underwood <steveu@coppice.org>
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*/
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/*! \li \ref dsp.c uses the configuration file \ref dsp.conf
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* \addtogroup configuration_file Configuration Files
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*/
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/*!
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* \page dsp.conf dsp.conf
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* \verbinclude dsp.conf.sample
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*/
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/* Some routines from tone_detect.c by Steven Underwood as published under the zapata library */
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/*
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tone_detect.c - General telephony tone detection, and specific
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detection of DTMF.
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Copyright (C) 2001 Steve Underwood <steveu@coppice.org>
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Despite my general liking of the GPL, I place this code in the
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public domain for the benefit of all mankind - even the slimy
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ones who might try to proprietize my work and use it to my
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detriment.
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*/
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/*** MODULEINFO
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<support_level>core</support_level>
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***/
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#include "asterisk.h"
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#include <math.h>
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#include "asterisk/module.h"
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#include "asterisk/frame.h"
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#include "asterisk/format_cache.h"
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#include "asterisk/channel.h"
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#include "asterisk/dsp.h"
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#include "asterisk/ulaw.h"
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#include "asterisk/alaw.h"
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#include "asterisk/utils.h"
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#include "asterisk/options.h"
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#include "asterisk/config.h"
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#include "asterisk/test.h"
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/*! Number of goertzels for progress detect */
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enum gsamp_size {
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GSAMP_SIZE_NA = 183, /*!< North America - 350, 440, 480, 620, 950, 1400, 1800 Hz */
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GSAMP_SIZE_CR = 188, /*!< Costa Rica, Brazil - Only care about 425 Hz */
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GSAMP_SIZE_UK = 160 /*!< UK disconnect goertzel feed - should trigger 400hz */
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};
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enum prog_mode {
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PROG_MODE_NA = 0,
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PROG_MODE_CR,
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PROG_MODE_UK
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};
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enum freq_index {
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/*! For US modes { */
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HZ_350 = 0,
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HZ_440,
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HZ_480,
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HZ_620,
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HZ_950,
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HZ_1400,
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HZ_1800, /*!< } */
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/*! For CR/BR modes */
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HZ_425 = 0,
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/*! For UK mode */
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HZ_350UK = 0,
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HZ_400UK,
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HZ_440UK
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};
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static struct progalias {
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char *name;
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enum prog_mode mode;
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} aliases[] = {
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{ "us", PROG_MODE_NA },
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{ "ca", PROG_MODE_NA },
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{ "cr", PROG_MODE_CR },
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{ "br", PROG_MODE_CR },
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{ "uk", PROG_MODE_UK },
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};
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#define FREQ_ARRAY_SIZE 7
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static struct progress {
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enum gsamp_size size;
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int freqs[FREQ_ARRAY_SIZE];
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} modes[] = {
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{ GSAMP_SIZE_NA, { 350, 440, 480, 620, 950, 1400, 1800 } }, /*!< North America */
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{ GSAMP_SIZE_CR, { 425 } }, /*!< Costa Rica, Brazil */
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{ GSAMP_SIZE_UK, { 350, 400, 440 } }, /*!< UK */
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};
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/*!
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* \brief Default minimum average magnitude threshold to determine talking/noise by the DSP.
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*
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* \details
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* The magnitude calculated for this threshold is determined by
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* averaging the absolute value of all samples within a frame.
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*
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* This value is the threshold for which a frame's average magnitude
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* is determined to either be silence (below the threshold) or
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* noise/talking (at or above the threshold). Please note that while
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* the default threshold is an even exponent of 2, there is no
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* requirement that it be so. The threshold will work for any value
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* between 1 and 2^15.
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*/
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#define DEFAULT_THRESHOLD 512
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enum busy_detect {
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BUSY_PERCENT = 10, /*!< The percentage difference between the two last silence periods */
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BUSY_PAT_PERCENT = 7, /*!< The percentage difference between measured and actual pattern */
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BUSY_THRESHOLD = 100, /*!< Max number of ms difference between max and min times in busy */
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BUSY_MIN = 75, /*!< Busy must be at least 80 ms in half-cadence */
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BUSY_MAX = 3100 /*!< Busy can't be longer than 3100 ms in half-cadence */
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};
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/*! Remember last 15 units */
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#define DSP_HISTORY 15
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#define TONE_THRESH 10.0 /*!< How much louder the tone should be than channel energy */
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#define TONE_MIN_THRESH 1e8 /*!< How much tone there should be at least to attempt */
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/*! All THRESH_XXX values are in GSAMP_SIZE chunks (us = 22ms) */
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enum gsamp_thresh {
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THRESH_RING = 8, /*!< Need at least 150ms ring to accept */
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THRESH_TALK = 2, /*!< Talk detection does not work continuously */
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THRESH_BUSY = 4, /*!< Need at least 80ms to accept */
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THRESH_CONGESTION = 4, /*!< Need at least 80ms to accept */
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THRESH_HANGUP = 60, /*!< Need at least 1300ms to accept hangup */
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THRESH_RING2ANSWER = 300 /*!< Timeout from start of ring to answer (about 6600 ms) */
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};
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#define MAX_DTMF_DIGITS 128
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#define DTMF_MATRIX_SIZE 4
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/* Basic DTMF (AT&T) specs:
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*
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* Minimum tone on = 40ms
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* Minimum tone off = 50ms
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* Maximum digit rate = 10 per second
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* Normal twist <= 8dB accepted
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* Reverse twist <= 4dB accepted
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* S/N >= 15dB will detect OK
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* Attenuation <= 26dB will detect OK
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* Frequency tolerance +- 1.5% will detect, +-3.5% will reject
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*/
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#define DTMF_THRESHOLD 8.0e7
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#define TONE_THRESHOLD 7.8e7
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#define DEF_DTMF_NORMAL_TWIST 6.31 /* 8.0dB */
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#define DEF_RELAX_DTMF_NORMAL_TWIST 6.31 /* 8.0dB */
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#ifdef RADIO_RELAX
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#define DEF_DTMF_REVERSE_TWIST 2.51 /* 4.01dB */
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#define DEF_RELAX_DTMF_REVERSE_TWIST 6.61 /* 8.2dB */
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#else
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#define DEF_DTMF_REVERSE_TWIST 2.51 /* 4.01dB */
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#define DEF_RELAX_DTMF_REVERSE_TWIST 3.98 /* 6.0dB */
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#endif
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#define DTMF_RELATIVE_PEAK_ROW 6.3 /* 8dB */
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#define DTMF_RELATIVE_PEAK_COL 6.3 /* 8dB */
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#define DTMF_TO_TOTAL_ENERGY 42.0
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#define BELL_MF_THRESHOLD 1.6e9
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#define BELL_MF_TWIST 4.0 /* 6dB */
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#define BELL_MF_RELATIVE_PEAK 12.6 /* 11dB */
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#if defined(BUSYDETECT_TONEONLY) && defined(BUSYDETECT_COMPARE_TONE_AND_SILENCE)
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#error You cant use BUSYDETECT_TONEONLY together with BUSYDETECT_COMPARE_TONE_AND_SILENCE
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#endif
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/* The CNG signal consists of the transmission of 1100 Hz for 1/2 second,
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* followed by a 3 second silent (2100 Hz OFF) period.
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*/
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#define FAX_TONE_CNG_FREQ 1100
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#define FAX_TONE_CNG_DURATION 500 /* ms */
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#define FAX_TONE_CNG_DB 16
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/* This signal may be sent by the Terminating FAX machine anywhere between
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* 1.8 to 2.5 seconds AFTER answering the call. The CED signal consists
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* of a 2100 Hz tone that is from 2.6 to 4 seconds in duration.
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*/
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#define FAX_TONE_CED_FREQ 2100
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#define FAX_TONE_CED_DURATION 2600 /* ms */
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#define FAX_TONE_CED_DB 16
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#define DEFAULT_SAMPLE_RATE 8000
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/* MF goertzel size */
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#define MF_GSIZE 120
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/* DTMF goertzel size */
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#define DTMF_GSIZE 102
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/* How many successive hits needed to consider begin of a digit
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* IE. Override with dtmf_hits_to_begin=4 in dsp.conf
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*/
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#define DEF_DTMF_HITS_TO_BEGIN 2
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/* How many successive misses needed to consider end of a digit
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* IE. Override with dtmf_misses_to_end=4 in dsp.conf
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*/
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#define DEF_DTMF_MISSES_TO_END 3
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/*!
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* \brief The default silence threshold we will use if an alternate
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* configured value is not present or is invalid.
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*/
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static const int DEFAULT_SILENCE_THRESHOLD = 256;
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#define CONFIG_FILE_NAME "dsp.conf"
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typedef struct {
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/*! The previous previous sample calculation (No binary point just plain int) */
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int v2;
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/*! The previous sample calculation (No binary point just plain int) */
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int v3;
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/*! v2 and v3 power of two exponent to keep value in int range */
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int chunky;
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/*! 15 bit fixed point goertzel coefficient = 2 * cos(2 * pi * freq / sample_rate) */
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int fac;
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} goertzel_state_t;
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typedef struct {
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int value;
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int power;
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} goertzel_result_t;
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typedef struct
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{
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int freq;
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int block_size;
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int squelch; /* Remove (squelch) tone */
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goertzel_state_t tone;
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float energy; /* Accumulated energy of the current block */
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int samples_pending; /* Samples remain to complete the current block */
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int mute_samples; /* How many additional samples needs to be muted to suppress already detected tone */
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int hits_required; /* How many successive blocks with tone we are looking for */
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float threshold; /* Energy of the tone relative to energy from all other signals to consider a hit */
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int hit_count; /* How many successive blocks we consider tone present */
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int last_hit; /* Indicates if the last processed block was a hit */
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} tone_detect_state_t;
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typedef struct
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{
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goertzel_state_t row_out[DTMF_MATRIX_SIZE];
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goertzel_state_t col_out[DTMF_MATRIX_SIZE];
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int hits; /* How many successive hits we have seen already */
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int misses; /* How many successive misses we have seen already */
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int lasthit;
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int current_hit;
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float energy;
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int current_sample;
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int mute_samples;
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} dtmf_detect_state_t;
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typedef struct
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{
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goertzel_state_t tone_out[6];
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int current_hit;
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int hits[5];
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int current_sample;
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int mute_samples;
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} mf_detect_state_t;
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typedef struct
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{
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char digits[MAX_DTMF_DIGITS + 1];
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int digitlen[MAX_DTMF_DIGITS + 1];
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int current_digits;
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int detected_digits;
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int lost_digits;
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union {
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dtmf_detect_state_t dtmf;
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mf_detect_state_t mf;
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} td;
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} digit_detect_state_t;
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static const float dtmf_row[] = {
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697.0, 770.0, 852.0, 941.0
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};
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static const float dtmf_col[] = {
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1209.0, 1336.0, 1477.0, 1633.0
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};
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static const float mf_tones[] = {
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700.0, 900.0, 1100.0, 1300.0, 1500.0, 1700.0
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};
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static const char dtmf_positions[] = "123A" "456B" "789C" "*0#D";
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static const char bell_mf_positions[] = "1247C-358A--69*---0B----#";
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static int thresholds[THRESHOLD_MAX];
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static float dtmf_normal_twist; /* AT&T = 8dB */
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static float dtmf_reverse_twist; /* AT&T = 4dB */
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static float relax_dtmf_normal_twist; /* AT&T = 8dB */
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static float relax_dtmf_reverse_twist; /* AT&T = 6dB */
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static int dtmf_hits_to_begin; /* How many successive hits needed to consider begin of a digit */
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static int dtmf_misses_to_end; /* How many successive misses needed to consider end of a digit */
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static inline void goertzel_sample(goertzel_state_t *s, short sample)
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{
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int v1;
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/*
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* Shift previous values so
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* v1 is previous previous value
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* v2 is previous value
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* until the new v3 is calculated.
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*/
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v1 = s->v2;
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s->v2 = s->v3;
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/* Discard the binary fraction introduced by s->fac */
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s->v3 = (s->fac * s->v2) >> 15;
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/* Scale sample to match previous values */
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s->v3 = s->v3 - v1 + (sample >> s->chunky);
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if (abs(s->v3) > (1 << 15)) {
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/* The result is now too large so increase the chunky power. */
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s->chunky++;
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s->v3 = s->v3 >> 1;
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s->v2 = s->v2 >> 1;
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}
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}
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static inline float goertzel_result(goertzel_state_t *s)
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{
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goertzel_result_t r;
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r.value = (s->v3 * s->v3) + (s->v2 * s->v2);
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r.value -= ((s->v2 * s->v3) >> 15) * s->fac;
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/*
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* We have to double the exponent because we multiplied the
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* previous sample calculation values together.
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*/
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r.power = s->chunky * 2;
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return (float)r.value * (float)(1 << r.power);
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}
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static inline void goertzel_init(goertzel_state_t *s, float freq, unsigned int sample_rate)
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{
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s->v2 = s->v3 = s->chunky = 0;
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s->fac = (int)(32768.0 * 2.0 * cos(2.0 * M_PI * freq / sample_rate));
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}
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static inline void goertzel_reset(goertzel_state_t *s)
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{
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s->v2 = s->v3 = s->chunky = 0;
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}
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typedef struct {
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int start;
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int end;
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} fragment_t;
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/* Note on tone suppression (squelching). Individual detectors (DTMF/MF/generic tone)
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* report fragments of the frame in which detected tone resides and which needs
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* to be "muted" in order to suppress the tone. To mark fragment for muting,
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* detectors call mute_fragment passing fragment_t there. Multiple fragments
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* can be marked and ast_dsp_process later will mute all of them.
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*
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* Note: When tone starts in the middle of a Goertzel block, it won't be properly
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* detected in that block, only in the next. If we only mute the next block
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* where tone is actually detected, the user will still hear beginning
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* of the tone in preceeding block. This is why we usually want to mute some amount
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* of samples preceeding and following the block where tone was detected.
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*/
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struct ast_dsp {
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struct ast_frame f;
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int threshold;
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/*! Accumulated total silence in ms since last talking/noise. */
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int totalsilence;
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/*! Accumulated total talking/noise in ms since last silence. */
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int totalnoise;
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int features;
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int ringtimeout;
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int busymaybe;
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int busycount;
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struct ast_dsp_busy_pattern busy_cadence;
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int historicnoise[DSP_HISTORY];
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int historicsilence[DSP_HISTORY];
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goertzel_state_t freqs[FREQ_ARRAY_SIZE];
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int freqcount;
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int gsamps;
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enum gsamp_size gsamp_size;
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enum prog_mode progmode;
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int tstate;
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int tcount;
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int digitmode;
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int faxmode;
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int freqmode;
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int dtmf_began;
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int display_inband_dtmf_warning;
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float genergy;
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int mute_fragments;
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unsigned int sample_rate;
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fragment_t mute_data[5];
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digit_detect_state_t digit_state;
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tone_detect_state_t cng_tone_state;
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tone_detect_state_t ced_tone_state;
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};
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static void mute_fragment(struct ast_dsp *dsp, fragment_t *fragment)
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{
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if (dsp->mute_fragments >= ARRAY_LEN(dsp->mute_data)) {
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ast_log(LOG_ERROR, "Too many fragments to mute. Ignoring\n");
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return;
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}
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dsp->mute_data[dsp->mute_fragments++] = *fragment;
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}
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static void ast_tone_detect_init(tone_detect_state_t *s, int freq, int duration, int amp, unsigned int sample_rate)
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{
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int duration_samples;
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float x;
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int periods_in_block;
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s->freq = freq;
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/* Desired tone duration in samples */
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duration_samples = duration * sample_rate / 1000;
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/* We want to allow 10% deviation of tone duration */
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duration_samples = duration_samples * 9 / 10;
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/* If we want to remove tone, it is important to have block size not
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to exceed frame size. Otherwise by the moment tone is detected it is too late
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to squelch it from previous frames. Block size is 20ms at the given sample rate.*/
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s->block_size = (20 * sample_rate) / 1000;
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periods_in_block = s->block_size * freq / sample_rate;
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/* Make sure we will have at least 5 periods at target frequency for analysis.
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This may make block larger than expected packet and will make squelching impossible
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but at least we will be detecting the tone */
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if (periods_in_block < 5) {
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periods_in_block = 5;
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}
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/* Now calculate final block size. It will contain integer number of periods */
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s->block_size = periods_in_block * sample_rate / freq;
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/* tone_detect is generally only used to detect fax tones and we
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do not need squelching the fax tones */
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s->squelch = 0;
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/* Account for the first and the last block to be incomplete
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and thus no tone will be detected in them */
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s->hits_required = (duration_samples - (s->block_size - 1)) / s->block_size;
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goertzel_init(&s->tone, freq, sample_rate);
|
|
|
|
s->samples_pending = s->block_size;
|
|
s->hit_count = 0;
|
|
s->last_hit = 0;
|
|
s->energy = 0.0;
|
|
|
|
/* We want tone energy to be amp decibels above the rest of the signal (the noise).
|
|
According to Parseval's theorem the energy computed in time domain equals to energy
|
|
computed in frequency domain. So subtracting energy in the frequency domain (Goertzel result)
|
|
from the energy in the time domain we will get energy of the remaining signal (without the tone
|
|
we are detecting). We will be checking that
|
|
10*log(Ew / (Et - Ew)) > amp
|
|
Calculate threshold so that we will be actually checking
|
|
Ew > Et * threshold
|
|
*/
|
|
|
|
x = pow(10.0, amp / 10.0);
|
|
s->threshold = x / (x + 1);
|
|
|
|
ast_debug(1, "Setup tone %d Hz, %d ms, block_size=%d, hits_required=%d\n", freq, duration, s->block_size, s->hits_required);
|
|
}
|
|
|
|
static void ast_fax_detect_init(struct ast_dsp *s)
|
|
{
|
|
ast_tone_detect_init(&s->cng_tone_state, FAX_TONE_CNG_FREQ, FAX_TONE_CNG_DURATION, FAX_TONE_CNG_DB, s->sample_rate);
|
|
ast_tone_detect_init(&s->ced_tone_state, FAX_TONE_CED_FREQ, FAX_TONE_CED_DURATION, FAX_TONE_CED_DB, s->sample_rate);
|
|
if (s->faxmode & DSP_FAXMODE_DETECT_SQUELCH) {
|
|
s->cng_tone_state.squelch = 1;
|
|
s->ced_tone_state.squelch = 1;
|
|
}
|
|
|
|
}
|
|
|
|
static void ast_freq_detect_init(struct ast_dsp *s, int freq, int dur, int db, int squelch)
|
|
{
|
|
/* we can conveniently just use one of the two fax tone states */
|
|
ast_tone_detect_init(&s->cng_tone_state, freq, dur, db, s->sample_rate);
|
|
if (s->freqmode & squelch) {
|
|
s->cng_tone_state.squelch = 1;
|
|
}
|
|
}
|
|
|
|
static void ast_dtmf_detect_init(dtmf_detect_state_t *s, unsigned int sample_rate)
|
|
{
|
|
int i;
|
|
|
|
for (i = 0; i < DTMF_MATRIX_SIZE; i++) {
|
|
goertzel_init(&s->row_out[i], dtmf_row[i], sample_rate);
|
|
goertzel_init(&s->col_out[i], dtmf_col[i], sample_rate);
|
|
}
|
|
s->lasthit = 0;
|
|
s->current_hit = 0;
|
|
s->energy = 0.0;
|
|
s->current_sample = 0;
|
|
s->hits = 0;
|
|
s->misses = 0;
|
|
}
|
|
|
|
static void ast_mf_detect_init(mf_detect_state_t *s, unsigned int sample_rate)
|
|
{
|
|
int i;
|
|
|
|
for (i = 0; i < 6; i++) {
|
|
goertzel_init(&s->tone_out[i], mf_tones[i], sample_rate);
|
|
}
|
|
s->hits[0] = s->hits[1] = s->hits[2] = s->hits[3] = s->hits[4] = 0;
|
|
s->current_sample = 0;
|
|
s->current_hit = 0;
|
|
}
|
|
|
|
static void ast_digit_detect_init(digit_detect_state_t *s, int mf, unsigned int sample_rate)
|
|
{
|
|
s->current_digits = 0;
|
|
s->detected_digits = 0;
|
|
s->lost_digits = 0;
|
|
s->digits[0] = '\0';
|
|
|
|
if (mf) {
|
|
ast_mf_detect_init(&s->td.mf, sample_rate);
|
|
} else {
|
|
ast_dtmf_detect_init(&s->td.dtmf, sample_rate);
|
|
}
|
|
}
|
|
|
|
static int tone_detect(struct ast_dsp *dsp, tone_detect_state_t *s, int16_t *amp, int samples)
|
|
{
|
|
float tone_energy;
|
|
int i;
|
|
int hit = 0;
|
|
int limit;
|
|
int res = 0;
|
|
int16_t *ptr;
|
|
short samp;
|
|
int start, end;
|
|
fragment_t mute = {0, 0};
|
|
|
|
if (s->squelch && s->mute_samples > 0) {
|
|
mute.end = (s->mute_samples < samples) ? s->mute_samples : samples;
|
|
s->mute_samples -= mute.end;
|
|
}
|
|
|
|
for (start = 0; start < samples; start = end) {
|
|
/* Process in blocks. */
|
|
limit = samples - start;
|
|
if (limit > s->samples_pending) {
|
|
limit = s->samples_pending;
|
|
}
|
|
end = start + limit;
|
|
|
|
for (i = limit, ptr = amp ; i > 0; i--, ptr++) {
|
|
samp = *ptr;
|
|
/* signed 32 bit int should be enough to square any possible signed 16 bit value */
|
|
s->energy += (int32_t) samp * (int32_t) samp;
|
|
|
|
goertzel_sample(&s->tone, samp);
|
|
}
|
|
|
|
s->samples_pending -= limit;
|
|
|
|
if (s->samples_pending) {
|
|
/* Finished incomplete (last) block */
|
|
break;
|
|
}
|
|
|
|
tone_energy = goertzel_result(&s->tone);
|
|
|
|
/* Scale to make comparable */
|
|
tone_energy *= 2.0;
|
|
s->energy *= s->block_size;
|
|
|
|
ast_debug(10, "%d Hz tone %2d Ew=%.4E, Et=%.4E, s/n=%10.2f\n", s->freq, s->hit_count, tone_energy, s->energy, tone_energy / (s->energy - tone_energy));
|
|
hit = 0;
|
|
if (TONE_THRESHOLD <= tone_energy
|
|
&& tone_energy > s->energy * s->threshold) {
|
|
ast_debug(10, "%d Hz tone Hit! %2d Ew=%.4E, Et=%.4E, s/n=%10.2f\n", s->freq, s->hit_count, tone_energy, s->energy, tone_energy / (s->energy - tone_energy));
|
|
hit = 1;
|
|
}
|
|
|
|
if (s->hit_count) {
|
|
s->hit_count++;
|
|
}
|
|
|
|
if (hit == s->last_hit) {
|
|
if (!hit) {
|
|
/* Two successive misses. Tone ended */
|
|
s->hit_count = 0;
|
|
} else if (!s->hit_count) {
|
|
s->hit_count++;
|
|
}
|
|
|
|
}
|
|
|
|
if (s->hit_count == s->hits_required) {
|
|
ast_debug(1, "%d Hz tone detected\n", s->freq);
|
|
res = 1;
|
|
}
|
|
|
|
s->last_hit = hit;
|
|
|
|
/* If we had a hit in this block, include it into mute fragment */
|
|
if (s->squelch && hit) {
|
|
if (mute.end < start - s->block_size) {
|
|
/* There is a gap between fragments */
|
|
mute_fragment(dsp, &mute);
|
|
mute.start = (start > s->block_size) ? (start - s->block_size) : 0;
|
|
}
|
|
mute.end = end + s->block_size;
|
|
}
|
|
|
|
/* Reinitialise the detector for the next block */
|
|
/* Reset for the next block */
|
|
goertzel_reset(&s->tone);
|
|
|
|
/* Advance to the next block */
|
|
s->energy = 0.0;
|
|
s->samples_pending = s->block_size;
|
|
|
|
amp += limit;
|
|
}
|
|
|
|
if (s->squelch && mute.end) {
|
|
if (mute.end > samples) {
|
|
s->mute_samples = mute.end - samples;
|
|
mute.end = samples;
|
|
}
|
|
mute_fragment(dsp, &mute);
|
|
}
|
|
|
|
return res;
|
|
}
|
|
|
|
static void store_digit(digit_detect_state_t *s, char digit)
|
|
{
|
|
s->detected_digits++;
|
|
if (s->current_digits < MAX_DTMF_DIGITS) {
|
|
s->digitlen[s->current_digits] = 0;
|
|
s->digits[s->current_digits++] = digit;
|
|
s->digits[s->current_digits] = '\0';
|
|
} else {
|
|
ast_log(LOG_WARNING, "Digit lost due to full buffer\n");
|
|
s->lost_digits++;
|
|
}
|
|
}
|
|
|
|
static int dtmf_detect(struct ast_dsp *dsp, digit_detect_state_t *s, int16_t amp[], int samples, int squelch, int relax)
|
|
{
|
|
float row_energy[DTMF_MATRIX_SIZE];
|
|
float col_energy[DTMF_MATRIX_SIZE];
|
|
int i;
|
|
int j;
|
|
int sample;
|
|
short samp;
|
|
int best_row;
|
|
int best_col;
|
|
int hit;
|
|
int limit;
|
|
fragment_t mute = {0, 0};
|
|
|
|
if (squelch && s->td.dtmf.mute_samples > 0) {
|
|
mute.end = (s->td.dtmf.mute_samples < samples) ? s->td.dtmf.mute_samples : samples;
|
|
s->td.dtmf.mute_samples -= mute.end;
|
|
}
|
|
|
|
hit = 0;
|
|
for (sample = 0; sample < samples; sample = limit) {
|
|
/* DTMF_GSIZE is optimised to meet the DTMF specs. */
|
|
if ((samples - sample) >= (DTMF_GSIZE - s->td.dtmf.current_sample)) {
|
|
limit = sample + (DTMF_GSIZE - s->td.dtmf.current_sample);
|
|
} else {
|
|
limit = samples;
|
|
}
|
|
/* The following unrolled loop takes only 35% (rough estimate) of the
|
|
time of a rolled loop on the machine on which it was developed */
|
|
for (j = sample; j < limit; j++) {
|
|
samp = amp[j];
|
|
s->td.dtmf.energy += (int32_t) samp * (int32_t) samp;
|
|
/* With GCC 2.95, the following unrolled code seems to take about 35%
|
|
(rough estimate) as long as a neat little 0-3 loop */
|
|
goertzel_sample(s->td.dtmf.row_out, samp);
|
|
goertzel_sample(s->td.dtmf.col_out, samp);
|
|
goertzel_sample(s->td.dtmf.row_out + 1, samp);
|
|
goertzel_sample(s->td.dtmf.col_out + 1, samp);
|
|
goertzel_sample(s->td.dtmf.row_out + 2, samp);
|
|
goertzel_sample(s->td.dtmf.col_out + 2, samp);
|
|
goertzel_sample(s->td.dtmf.row_out + 3, samp);
|
|
goertzel_sample(s->td.dtmf.col_out + 3, samp);
|
|
/* go up to DTMF_MATRIX_SIZE - 1 */
|
|
}
|
|
s->td.dtmf.current_sample += (limit - sample);
|
|
if (s->td.dtmf.current_sample < DTMF_GSIZE) {
|
|
continue;
|
|
}
|
|
/* We are at the end of a DTMF detection block */
|
|
/* Find the peak row and the peak column */
|
|
row_energy[0] = goertzel_result(&s->td.dtmf.row_out[0]);
|
|
col_energy[0] = goertzel_result(&s->td.dtmf.col_out[0]);
|
|
|
|
for (best_row = best_col = 0, i = 1; i < DTMF_MATRIX_SIZE; i++) {
|
|
row_energy[i] = goertzel_result(&s->td.dtmf.row_out[i]);
|
|
if (row_energy[i] > row_energy[best_row]) {
|
|
best_row = i;
|
|
}
|
|
col_energy[i] = goertzel_result(&s->td.dtmf.col_out[i]);
|
|
if (col_energy[i] > col_energy[best_col]) {
|
|
best_col = i;
|
|
}
|
|
}
|
|
ast_debug(10, "DTMF best '%c' Erow=%.4E Ecol=%.4E Erc=%.4E Et=%.4E\n",
|
|
dtmf_positions[(best_row << 2) + best_col],
|
|
row_energy[best_row], col_energy[best_col],
|
|
row_energy[best_row] + col_energy[best_col], s->td.dtmf.energy);
|
|
hit = 0;
|
|
/* Basic signal level test and the twist test */
|
|
if (row_energy[best_row] >= DTMF_THRESHOLD &&
|
|
col_energy[best_col] >= DTMF_THRESHOLD &&
|
|
col_energy[best_col] < row_energy[best_row] * (relax ? relax_dtmf_reverse_twist : dtmf_reverse_twist) &&
|
|
row_energy[best_row] < col_energy[best_col] * (relax ? relax_dtmf_normal_twist : dtmf_normal_twist)) {
|
|
/* Relative peak test */
|
|
for (i = 0; i < DTMF_MATRIX_SIZE; i++) {
|
|
if ((i != best_col &&
|
|
col_energy[i] * DTMF_RELATIVE_PEAK_COL > col_energy[best_col]) ||
|
|
(i != best_row
|
|
&& row_energy[i] * DTMF_RELATIVE_PEAK_ROW > row_energy[best_row])) {
|
|
break;
|
|
}
|
|
}
|
|
/* ... and fraction of total energy test */
|
|
if (i >= DTMF_MATRIX_SIZE &&
|
|
(row_energy[best_row] + col_energy[best_col]) > DTMF_TO_TOTAL_ENERGY * s->td.dtmf.energy) {
|
|
/* Got a hit */
|
|
hit = dtmf_positions[(best_row << 2) + best_col];
|
|
ast_debug(10, "DTMF hit '%c'\n", hit);
|
|
}
|
|
}
|
|
|
|
/*
|
|
* Adapted from ETSI ES 201 235-3 V1.3.1 (2006-03)
|
|
* (40ms reference is tunable with hits_to_begin and misses_to_end)
|
|
* each hit/miss is 12.75ms with DTMF_GSIZE at 102
|
|
*
|
|
* Character recognition: When not DRC *(1) and then
|
|
* Shall exist VSC > 40 ms (hits_to_begin)
|
|
* May exist 20 ms <= VSC <= 40 ms
|
|
* Shall not exist VSC < 20 ms
|
|
*
|
|
* Character recognition: When DRC and then
|
|
* Shall cease Not VSC > 40 ms (misses_to_end)
|
|
* May cease 20 ms >= Not VSC >= 40 ms
|
|
* Shall not cease Not VSC < 20 ms
|
|
*
|
|
* *(1) or optionally a different digit recognition condition
|
|
*
|
|
* Legend: VSC The continuous existence of a valid signal condition.
|
|
* Not VSC The continuous non-existence of valid signal condition.
|
|
* DRC The existence of digit recognition condition.
|
|
* Not DRC The non-existence of digit recognition condition.
|
|
*/
|
|
|
|
/*
|
|
* Example: hits_to_begin=2 misses_to_end=3
|
|
* -------A last_hit=A hits=0&1
|
|
* ------AA hits=2 current_hit=A misses=0 BEGIN A
|
|
* -----AA- misses=1 last_hit=' ' hits=0
|
|
* ----AA-- misses=2
|
|
* ---AA--- misses=3 current_hit=' ' END A
|
|
* --AA---B last_hit=B hits=0&1
|
|
* -AA---BC last_hit=C hits=0&1
|
|
* AA---BCC hits=2 current_hit=C misses=0 BEGIN C
|
|
* A---BCC- misses=1 last_hit=' ' hits=0
|
|
* ---BCC-C misses=0 last_hit=C hits=0&1
|
|
* --BCC-CC misses=0
|
|
*
|
|
* Example: hits_to_begin=3 misses_to_end=2
|
|
* -------A last_hit=A hits=0&1
|
|
* ------AA hits=2
|
|
* -----AAA hits=3 current_hit=A misses=0 BEGIN A
|
|
* ----AAAB misses=1 last_hit=B hits=0&1
|
|
* ---AAABB misses=2 current_hit=' ' hits=2 END A
|
|
* --AAABBB hits=3 current_hit=B misses=0 BEGIN B
|
|
* -AAABBBB misses=0
|
|
*
|
|
* Example: hits_to_begin=2 misses_to_end=2
|
|
* -------A last_hit=A hits=0&1
|
|
* ------AA hits=2 current_hit=A misses=0 BEGIN A
|
|
* -----AAB misses=1 hits=0&1
|
|
* ----AABB misses=2 current_hit=' ' hits=2 current_hit=B misses=0 BEGIN B
|
|
* ---AABBB misses=0
|
|
*/
|
|
|
|
if (s->td.dtmf.current_hit) {
|
|
/* We are in the middle of a digit already */
|
|
if (hit != s->td.dtmf.current_hit) {
|
|
s->td.dtmf.misses++;
|
|
if (s->td.dtmf.misses == dtmf_misses_to_end) {
|
|
/* There were enough misses to consider digit ended */
|
|
s->td.dtmf.current_hit = 0;
|
|
}
|
|
} else {
|
|
s->td.dtmf.misses = 0;
|
|
/* Current hit was same as last, so increment digit duration (of last digit) */
|
|
s->digitlen[s->current_digits - 1] += DTMF_GSIZE;
|
|
}
|
|
}
|
|
|
|
/* Look for a start of a new digit no matter if we are already in the middle of some
|
|
digit or not. This is because hits_to_begin may be smaller than misses_to_end
|
|
and we may find begin of new digit before we consider last one ended. */
|
|
|
|
if (hit != s->td.dtmf.lasthit) {
|
|
s->td.dtmf.lasthit = hit;
|
|
s->td.dtmf.hits = 0;
|
|
}
|
|
if (hit && hit != s->td.dtmf.current_hit) {
|
|
s->td.dtmf.hits++;
|
|
if (s->td.dtmf.hits == dtmf_hits_to_begin) {
|
|
store_digit(s, hit);
|
|
s->digitlen[s->current_digits - 1] = dtmf_hits_to_begin * DTMF_GSIZE;
|
|
s->td.dtmf.current_hit = hit;
|
|
s->td.dtmf.misses = 0;
|
|
}
|
|
}
|
|
|
|
/* If we had a hit in this block, include it into mute fragment */
|
|
if (squelch && hit) {
|
|
if (mute.end < sample - DTMF_GSIZE) {
|
|
/* There is a gap between fragments */
|
|
mute_fragment(dsp, &mute);
|
|
mute.start = (sample > DTMF_GSIZE) ? (sample - DTMF_GSIZE) : 0;
|
|
}
|
|
mute.end = limit + DTMF_GSIZE;
|
|
}
|
|
|
|
/* Reinitialise the detector for the next block */
|
|
for (i = 0; i < DTMF_MATRIX_SIZE; i++) {
|
|
goertzel_reset(&s->td.dtmf.row_out[i]);
|
|
goertzel_reset(&s->td.dtmf.col_out[i]);
|
|
}
|
|
s->td.dtmf.energy = 0.0;
|
|
s->td.dtmf.current_sample = 0;
|
|
}
|
|
|
|
if (squelch && mute.end) {
|
|
if (mute.end > samples) {
|
|
s->td.dtmf.mute_samples = mute.end - samples;
|
|
mute.end = samples;
|
|
}
|
|
mute_fragment(dsp, &mute);
|
|
}
|
|
|
|
return (s->td.dtmf.current_hit); /* return the debounced hit */
|
|
}
|
|
|
|
static int mf_detect(struct ast_dsp *dsp, digit_detect_state_t *s, int16_t amp[],
|
|
int samples, int squelch, int relax)
|
|
{
|
|
float energy[6];
|
|
int best;
|
|
int second_best;
|
|
int i;
|
|
int j;
|
|
int sample;
|
|
short samp;
|
|
int hit;
|
|
int limit;
|
|
fragment_t mute = {0, 0};
|
|
|
|
if (squelch && s->td.mf.mute_samples > 0) {
|
|
mute.end = (s->td.mf.mute_samples < samples) ? s->td.mf.mute_samples : samples;
|
|
s->td.mf.mute_samples -= mute.end;
|
|
}
|
|
|
|
hit = 0;
|
|
for (sample = 0; sample < samples; sample = limit) {
|
|
/* 80 is optimised to meet the MF specs. */
|
|
/* XXX So then why is MF_GSIZE defined as 120? */
|
|
if ((samples - sample) >= (MF_GSIZE - s->td.mf.current_sample)) {
|
|
limit = sample + (MF_GSIZE - s->td.mf.current_sample);
|
|
} else {
|
|
limit = samples;
|
|
}
|
|
/* The following unrolled loop takes only 35% (rough estimate) of the
|
|
time of a rolled loop on the machine on which it was developed */
|
|
for (j = sample; j < limit; j++) {
|
|
/* With GCC 2.95, the following unrolled code seems to take about 35%
|
|
(rough estimate) as long as a neat little 0-3 loop */
|
|
samp = amp[j];
|
|
goertzel_sample(s->td.mf.tone_out, samp);
|
|
goertzel_sample(s->td.mf.tone_out + 1, samp);
|
|
goertzel_sample(s->td.mf.tone_out + 2, samp);
|
|
goertzel_sample(s->td.mf.tone_out + 3, samp);
|
|
goertzel_sample(s->td.mf.tone_out + 4, samp);
|
|
goertzel_sample(s->td.mf.tone_out + 5, samp);
|
|
}
|
|
s->td.mf.current_sample += (limit - sample);
|
|
if (s->td.mf.current_sample < MF_GSIZE) {
|
|
continue;
|
|
}
|
|
/* We're at the end of an MF detection block. */
|
|
/* Find the two highest energies. The spec says to look for
|
|
two tones and two tones only. Taking this literally -ie
|
|
only two tones pass the minimum threshold - doesn't work
|
|
well. The sinc function mess, due to rectangular windowing
|
|
ensure that! Find the two highest energies and ensure they
|
|
are considerably stronger than any of the others. */
|
|
energy[0] = goertzel_result(&s->td.mf.tone_out[0]);
|
|
energy[1] = goertzel_result(&s->td.mf.tone_out[1]);
|
|
if (energy[0] > energy[1]) {
|
|
best = 0;
|
|
second_best = 1;
|
|
} else {
|
|
best = 1;
|
|
second_best = 0;
|
|
}
|
|
/*endif*/
|
|
for (i = 2; i < 6; i++) {
|
|
energy[i] = goertzel_result(&s->td.mf.tone_out[i]);
|
|
if (energy[i] >= energy[best]) {
|
|
second_best = best;
|
|
best = i;
|
|
} else if (energy[i] >= energy[second_best]) {
|
|
second_best = i;
|
|
}
|
|
}
|
|
/* Basic signal level and twist tests */
|
|
hit = 0;
|
|
if (energy[best] >= BELL_MF_THRESHOLD && energy[second_best] >= BELL_MF_THRESHOLD
|
|
&& energy[best] < energy[second_best]*BELL_MF_TWIST
|
|
&& energy[best] * BELL_MF_TWIST > energy[second_best]) {
|
|
/* Relative peak test */
|
|
hit = -1;
|
|
for (i = 0; i < 6; i++) {
|
|
if (i != best && i != second_best) {
|
|
if (energy[i]*BELL_MF_RELATIVE_PEAK >= energy[second_best]) {
|
|
/* The best two are not clearly the best */
|
|
hit = 0;
|
|
break;
|
|
}
|
|
}
|
|
}
|
|
}
|
|
if (hit) {
|
|
/* Get the values into ascending order */
|
|
if (second_best < best) {
|
|
i = best;
|
|
best = second_best;
|
|
second_best = i;
|
|
}
|
|
best = best * 5 + second_best - 1;
|
|
hit = bell_mf_positions[best];
|
|
/* Look for two successive similar results */
|
|
/* The logic in the next test is:
|
|
For KP we need 4 successive identical clean detects, with
|
|
two blocks of something different preceeding it. For anything
|
|
else we need two successive identical clean detects, with
|
|
two blocks of something different preceeding it. */
|
|
if (hit == s->td.mf.hits[4] && hit == s->td.mf.hits[3] &&
|
|
((hit != '*' && hit != s->td.mf.hits[2] && hit != s->td.mf.hits[1])||
|
|
(hit == '*' && hit == s->td.mf.hits[2] && hit != s->td.mf.hits[1] &&
|
|
hit != s->td.mf.hits[0]))) {
|
|
store_digit(s, hit);
|
|
}
|
|
}
|
|
|
|
|
|
if (hit != s->td.mf.hits[4] && hit != s->td.mf.hits[3]) {
|
|
/* Two successive block without a hit terminate current digit */
|
|
s->td.mf.current_hit = 0;
|
|
}
|
|
|
|
s->td.mf.hits[0] = s->td.mf.hits[1];
|
|
s->td.mf.hits[1] = s->td.mf.hits[2];
|
|
s->td.mf.hits[2] = s->td.mf.hits[3];
|
|
s->td.mf.hits[3] = s->td.mf.hits[4];
|
|
s->td.mf.hits[4] = hit;
|
|
|
|
/* If we had a hit in this block, include it into mute fragment */
|
|
if (squelch && hit) {
|
|
if (mute.end < sample - MF_GSIZE) {
|
|
/* There is a gap between fragments */
|
|
mute_fragment(dsp, &mute);
|
|
mute.start = (sample > MF_GSIZE) ? (sample - MF_GSIZE) : 0;
|
|
}
|
|
mute.end = limit + MF_GSIZE;
|
|
}
|
|
|
|
/* Reinitialise the detector for the next block */
|
|
for (i = 0; i < 6; i++) {
|
|
goertzel_reset(&s->td.mf.tone_out[i]);
|
|
}
|
|
s->td.mf.current_sample = 0;
|
|
}
|
|
|
|
if (squelch && mute.end) {
|
|
if (mute.end > samples) {
|
|
s->td.mf.mute_samples = mute.end - samples;
|
|
mute.end = samples;
|
|
}
|
|
mute_fragment(dsp, &mute);
|
|
}
|
|
|
|
return (s->td.mf.current_hit); /* return the debounced hit */
|
|
}
|
|
|
|
static inline int pair_there(float p1, float p2, float i1, float i2, float e)
|
|
{
|
|
/* See if p1 and p2 are there, relative to i1 and i2 and total energy */
|
|
/* Make sure absolute levels are high enough */
|
|
if ((p1 < TONE_MIN_THRESH) || (p2 < TONE_MIN_THRESH)) {
|
|
return 0;
|
|
}
|
|
/* Amplify ignored stuff */
|
|
i2 *= TONE_THRESH;
|
|
i1 *= TONE_THRESH;
|
|
e *= TONE_THRESH;
|
|
/* Check first tone */
|
|
if ((p1 < i1) || (p1 < i2) || (p1 < e)) {
|
|
return 0;
|
|
}
|
|
/* And second */
|
|
if ((p2 < i1) || (p2 < i2) || (p2 < e)) {
|
|
return 0;
|
|
}
|
|
/* Guess it's there... */
|
|
return 1;
|
|
}
|
|
|
|
static int __ast_dsp_call_progress(struct ast_dsp *dsp, short *s, int len)
|
|
{
|
|
short samp;
|
|
int x;
|
|
int y;
|
|
int pass;
|
|
int newstate = DSP_TONE_STATE_SILENCE;
|
|
int res = 0;
|
|
int freqcount = dsp->freqcount > FREQ_ARRAY_SIZE ? FREQ_ARRAY_SIZE : dsp->freqcount;
|
|
|
|
while (len) {
|
|
/* Take the lesser of the number of samples we need and what we have */
|
|
pass = len;
|
|
if (pass > dsp->gsamp_size - dsp->gsamps) {
|
|
pass = dsp->gsamp_size - dsp->gsamps;
|
|
}
|
|
for (x = 0; x < pass; x++) {
|
|
samp = s[x];
|
|
dsp->genergy += (int32_t) samp * (int32_t) samp;
|
|
for (y = 0; y < freqcount; y++) {
|
|
goertzel_sample(&dsp->freqs[y], samp);
|
|
}
|
|
}
|
|
s += pass;
|
|
dsp->gsamps += pass;
|
|
len -= pass;
|
|
if (dsp->gsamps == dsp->gsamp_size) {
|
|
float hz[FREQ_ARRAY_SIZE];
|
|
for (y = 0; y < FREQ_ARRAY_SIZE; y++) {
|
|
hz[y] = goertzel_result(&dsp->freqs[y]);
|
|
}
|
|
switch (dsp->progmode) {
|
|
case PROG_MODE_NA:
|
|
if (pair_there(hz[HZ_480], hz[HZ_620], hz[HZ_350], hz[HZ_440], dsp->genergy)) {
|
|
newstate = DSP_TONE_STATE_BUSY;
|
|
} else if (pair_there(hz[HZ_440], hz[HZ_480], hz[HZ_350], hz[HZ_620], dsp->genergy)) {
|
|
newstate = DSP_TONE_STATE_RINGING;
|
|
} else if (pair_there(hz[HZ_350], hz[HZ_440], hz[HZ_480], hz[HZ_620], dsp->genergy)) {
|
|
newstate = DSP_TONE_STATE_DIALTONE;
|
|
} else if (hz[HZ_950] > TONE_MIN_THRESH * TONE_THRESH) {
|
|
newstate = DSP_TONE_STATE_SPECIAL1;
|
|
} else if (hz[HZ_1400] > TONE_MIN_THRESH * TONE_THRESH) {
|
|
/* End of SPECIAL1 or middle of SPECIAL2 */
|
|
if (dsp->tstate == DSP_TONE_STATE_SPECIAL1 || dsp->tstate == DSP_TONE_STATE_SPECIAL2) {
|
|
newstate = DSP_TONE_STATE_SPECIAL2;
|
|
}
|
|
} else if (hz[HZ_1800] > TONE_MIN_THRESH * TONE_THRESH) {
|
|
/* End of SPECIAL2 or middle of SPECIAL3 */
|
|
if (dsp->tstate == DSP_TONE_STATE_SPECIAL2 || dsp->tstate == DSP_TONE_STATE_SPECIAL3) {
|
|
newstate = DSP_TONE_STATE_SPECIAL3;
|
|
}
|
|
} else if (dsp->genergy > TONE_MIN_THRESH * TONE_THRESH) {
|
|
newstate = DSP_TONE_STATE_TALKING;
|
|
} else {
|
|
newstate = DSP_TONE_STATE_SILENCE;
|
|
}
|
|
break;
|
|
case PROG_MODE_CR:
|
|
if (hz[HZ_425] > TONE_MIN_THRESH * TONE_THRESH) {
|
|
newstate = DSP_TONE_STATE_RINGING;
|
|
} else if (dsp->genergy > TONE_MIN_THRESH * TONE_THRESH) {
|
|
newstate = DSP_TONE_STATE_TALKING;
|
|
} else {
|
|
newstate = DSP_TONE_STATE_SILENCE;
|
|
}
|
|
break;
|
|
case PROG_MODE_UK:
|
|
if (hz[HZ_400UK] > TONE_MIN_THRESH * TONE_THRESH) {
|
|
newstate = DSP_TONE_STATE_HUNGUP;
|
|
} else if (pair_there(hz[HZ_350UK], hz[HZ_440UK], hz[HZ_400UK], hz[HZ_400UK], dsp->genergy)) {
|
|
newstate = DSP_TONE_STATE_DIALTONE;
|
|
}
|
|
break;
|
|
default:
|
|
ast_log(LOG_WARNING, "Can't process in unknown prog mode '%u'\n", dsp->progmode);
|
|
}
|
|
if (newstate == dsp->tstate) {
|
|
dsp->tcount++;
|
|
if (dsp->ringtimeout) {
|
|
dsp->ringtimeout++;
|
|
}
|
|
switch (dsp->tstate) {
|
|
case DSP_TONE_STATE_RINGING:
|
|
if ((dsp->features & DSP_PROGRESS_RINGING) &&
|
|
(dsp->tcount == THRESH_RING)) {
|
|
res = AST_CONTROL_RINGING;
|
|
dsp->ringtimeout = 1;
|
|
}
|
|
break;
|
|
case DSP_TONE_STATE_BUSY:
|
|
if ((dsp->features & DSP_PROGRESS_BUSY) &&
|
|
(dsp->tcount == THRESH_BUSY)) {
|
|
res = AST_CONTROL_BUSY;
|
|
dsp->features &= ~DSP_FEATURE_CALL_PROGRESS;
|
|
}
|
|
break;
|
|
case DSP_TONE_STATE_TALKING:
|
|
if ((dsp->features & DSP_PROGRESS_TALK) &&
|
|
(dsp->tcount == THRESH_TALK)) {
|
|
res = AST_CONTROL_ANSWER;
|
|
dsp->features &= ~DSP_FEATURE_CALL_PROGRESS;
|
|
}
|
|
break;
|
|
case DSP_TONE_STATE_SPECIAL3:
|
|
if ((dsp->features & DSP_PROGRESS_CONGESTION) &&
|
|
(dsp->tcount == THRESH_CONGESTION)) {
|
|
res = AST_CONTROL_CONGESTION;
|
|
dsp->features &= ~DSP_FEATURE_CALL_PROGRESS;
|
|
}
|
|
break;
|
|
case DSP_TONE_STATE_HUNGUP:
|
|
if ((dsp->features & DSP_FEATURE_CALL_PROGRESS) &&
|
|
(dsp->tcount == THRESH_HANGUP)) {
|
|
res = AST_CONTROL_HANGUP;
|
|
dsp->features &= ~DSP_FEATURE_CALL_PROGRESS;
|
|
}
|
|
break;
|
|
}
|
|
if (dsp->ringtimeout == THRESH_RING2ANSWER) {
|
|
ast_debug(1, "Consider call as answered because of timeout after last ring\n");
|
|
res = AST_CONTROL_ANSWER;
|
|
dsp->features &= ~DSP_FEATURE_CALL_PROGRESS;
|
|
}
|
|
} else {
|
|
ast_debug(5, "Stop state %d with duration %d\n", dsp->tstate, dsp->tcount);
|
|
ast_debug(5, "Start state %d\n", newstate);
|
|
dsp->tstate = newstate;
|
|
dsp->tcount = 1;
|
|
}
|
|
|
|
/* Reset goertzel */
|
|
for (x = 0; x < 7; x++) {
|
|
dsp->freqs[x].v2 = dsp->freqs[x].v3 = 0.0;
|
|
}
|
|
dsp->gsamps = 0;
|
|
dsp->genergy = 0.0;
|
|
}
|
|
}
|
|
|
|
return res;
|
|
}
|
|
|
|
int ast_dsp_call_progress(struct ast_dsp *dsp, struct ast_frame *inf)
|
|
{
|
|
if (inf->frametype != AST_FRAME_VOICE) {
|
|
ast_log(LOG_WARNING, "Can't check call progress of non-voice frames\n");
|
|
return 0;
|
|
}
|
|
if (!ast_format_cache_is_slinear(inf->subclass.format)) {
|
|
ast_log(LOG_WARNING, "Can only check call progress in signed-linear frames\n");
|
|
return 0;
|
|
}
|
|
return __ast_dsp_call_progress(dsp, inf->data.ptr, inf->datalen / 2);
|
|
}
|
|
|
|
static int __ast_dsp_silence_noise(struct ast_dsp *dsp, short *s, int len, int *totalsilence, int *totalnoise, int *frames_energy)
|
|
{
|
|
int accum;
|
|
int x;
|
|
int res = 0;
|
|
|
|
if (!len) {
|
|
return 0;
|
|
}
|
|
accum = 0;
|
|
for (x = 0; x < len; x++) {
|
|
accum += abs(s[x]);
|
|
}
|
|
accum /= len;
|
|
if (accum < dsp->threshold) {
|
|
/* Silent */
|
|
dsp->totalsilence += len / (dsp->sample_rate / 1000);
|
|
if (dsp->totalnoise) {
|
|
/* Move and save history */
|
|
memmove(dsp->historicnoise + DSP_HISTORY - dsp->busycount, dsp->historicnoise + DSP_HISTORY - dsp->busycount + 1, dsp->busycount * sizeof(dsp->historicnoise[0]));
|
|
dsp->historicnoise[DSP_HISTORY - 1] = dsp->totalnoise;
|
|
/* we don't want to check for busydetect that frequently */
|
|
#if 0
|
|
dsp->busymaybe = 1;
|
|
#endif
|
|
}
|
|
dsp->totalnoise = 0;
|
|
res = 1;
|
|
} else {
|
|
/* Not silent */
|
|
dsp->totalnoise += len / (dsp->sample_rate / 1000);
|
|
if (dsp->totalsilence) {
|
|
int silence1 = dsp->historicsilence[DSP_HISTORY - 1];
|
|
int silence2 = dsp->historicsilence[DSP_HISTORY - 2];
|
|
/* Move and save history */
|
|
memmove(dsp->historicsilence + DSP_HISTORY - dsp->busycount, dsp->historicsilence + DSP_HISTORY - dsp->busycount + 1, dsp->busycount * sizeof(dsp->historicsilence[0]));
|
|
dsp->historicsilence[DSP_HISTORY - 1] = dsp->totalsilence;
|
|
/* check if the previous sample differs only by BUSY_PERCENT from the one before it */
|
|
if (silence1 < silence2) {
|
|
if (silence1 + silence1 * BUSY_PERCENT / 100 >= silence2) {
|
|
dsp->busymaybe = 1;
|
|
} else {
|
|
dsp->busymaybe = 0;
|
|
}
|
|
} else {
|
|
if (silence1 - silence1 * BUSY_PERCENT / 100 <= silence2) {
|
|
dsp->busymaybe = 1;
|
|
} else {
|
|
dsp->busymaybe = 0;
|
|
}
|
|
}
|
|
}
|
|
dsp->totalsilence = 0;
|
|
}
|
|
if (totalsilence) {
|
|
*totalsilence = dsp->totalsilence;
|
|
}
|
|
if (totalnoise) {
|
|
*totalnoise = dsp->totalnoise;
|
|
}
|
|
if (frames_energy) {
|
|
*frames_energy = accum;
|
|
}
|
|
return res;
|
|
}
|
|
|
|
int ast_dsp_busydetect(struct ast_dsp *dsp)
|
|
{
|
|
int res = 0, x;
|
|
#ifndef BUSYDETECT_TONEONLY
|
|
int avgsilence = 0, hitsilence = 0;
|
|
#endif
|
|
int avgtone = 0, hittone = 0;
|
|
|
|
/* if we have a 4 length pattern, the way busymaybe is set doesn't help us. */
|
|
if (dsp->busy_cadence.length != 4) {
|
|
if (!dsp->busymaybe) {
|
|
return res;
|
|
}
|
|
}
|
|
|
|
for (x = DSP_HISTORY - dsp->busycount; x < DSP_HISTORY; x++) {
|
|
#ifndef BUSYDETECT_TONEONLY
|
|
avgsilence += dsp->historicsilence[x];
|
|
#endif
|
|
avgtone += dsp->historicnoise[x];
|
|
}
|
|
#ifndef BUSYDETECT_TONEONLY
|
|
avgsilence /= dsp->busycount;
|
|
#endif
|
|
avgtone /= dsp->busycount;
|
|
for (x = DSP_HISTORY - dsp->busycount; x < DSP_HISTORY; x++) {
|
|
#ifndef BUSYDETECT_TONEONLY
|
|
if (avgsilence > dsp->historicsilence[x]) {
|
|
if (avgsilence - (avgsilence * BUSY_PERCENT / 100) <= dsp->historicsilence[x]) {
|
|
hitsilence++;
|
|
}
|
|
} else {
|
|
if (avgsilence + (avgsilence * BUSY_PERCENT / 100) >= dsp->historicsilence[x]) {
|
|
hitsilence++;
|
|
}
|
|
}
|
|
#endif
|
|
if (avgtone > dsp->historicnoise[x]) {
|
|
if (avgtone - (avgtone * BUSY_PERCENT / 100) <= dsp->historicnoise[x]) {
|
|
hittone++;
|
|
}
|
|
} else {
|
|
if (avgtone + (avgtone * BUSY_PERCENT / 100) >= dsp->historicnoise[x]) {
|
|
hittone++;
|
|
}
|
|
}
|
|
}
|
|
#ifndef BUSYDETECT_TONEONLY
|
|
if ((hittone >= dsp->busycount - 1) && (hitsilence >= dsp->busycount - 1) &&
|
|
(avgtone >= BUSY_MIN && avgtone <= BUSY_MAX) &&
|
|
(avgsilence >= BUSY_MIN && avgsilence <= BUSY_MAX))
|
|
#else
|
|
if ((hittone >= dsp->busycount - 1) && (avgtone >= BUSY_MIN && avgtone <= BUSY_MAX))
|
|
#endif
|
|
{
|
|
#ifdef BUSYDETECT_COMPARE_TONE_AND_SILENCE
|
|
if (avgtone > avgsilence) {
|
|
if (avgtone - avgtone*BUSY_PERCENT/100 <= avgsilence) {
|
|
res = 1;
|
|
}
|
|
} else {
|
|
if (avgtone + avgtone*BUSY_PERCENT/100 >= avgsilence) {
|
|
res = 1;
|
|
}
|
|
}
|
|
#else
|
|
res = 1;
|
|
#endif
|
|
}
|
|
|
|
/* If we have a 4-length pattern, we can go ahead and just check it in a different way. */
|
|
if (dsp->busy_cadence.length == 4) {
|
|
int x;
|
|
int errors = 0;
|
|
int errors_max = ((4 * dsp->busycount) / 100.0) * BUSY_PAT_PERCENT;
|
|
|
|
for (x = DSP_HISTORY - (dsp->busycount); x < DSP_HISTORY; x += 2) {
|
|
int temp_error;
|
|
temp_error = abs(dsp->historicnoise[x] - dsp->busy_cadence.pattern[0]);
|
|
if ((temp_error * 100) / dsp->busy_cadence.pattern[0] > BUSY_PERCENT) {
|
|
errors++;
|
|
}
|
|
|
|
temp_error = abs(dsp->historicnoise[x + 1] - dsp->busy_cadence.pattern[2]);
|
|
if ((temp_error * 100) / dsp->busy_cadence.pattern[2] > BUSY_PERCENT) {
|
|
errors++;
|
|
}
|
|
|
|
temp_error = abs(dsp->historicsilence[x] - dsp->busy_cadence.pattern[1]);
|
|
if ((temp_error * 100) / dsp->busy_cadence.pattern[1] > BUSY_PERCENT) {
|
|
errors++;
|
|
}
|
|
|
|
temp_error = abs(dsp->historicsilence[x + 1] - dsp->busy_cadence.pattern[3]);
|
|
if ((temp_error * 100) / dsp->busy_cadence.pattern[3] > BUSY_PERCENT) {
|
|
errors++;
|
|
}
|
|
}
|
|
|
|
ast_debug(5, "errors = %d max = %d\n", errors, errors_max);
|
|
|
|
if (errors <= errors_max) {
|
|
return 1;
|
|
}
|
|
}
|
|
|
|
/* If we know the expected busy tone length, check we are in the range */
|
|
if (res && (dsp->busy_cadence.pattern[0] > 0)) {
|
|
if (abs(avgtone - dsp->busy_cadence.pattern[0]) > MAX(dsp->busy_cadence.pattern[0]*BUSY_PAT_PERCENT/100, 20)) {
|
|
#ifdef BUSYDETECT_DEBUG
|
|
ast_debug(5, "busy detector: avgtone of %d not close enough to desired %d\n",
|
|
avgtone, dsp->busy_cadence.pattern[0]);
|
|
#endif
|
|
res = 0;
|
|
}
|
|
}
|
|
#ifndef BUSYDETECT_TONEONLY
|
|
/* If we know the expected busy tone silent-period length, check we are in the range */
|
|
if (res && (dsp->busy_cadence.pattern[1] > 0)) {
|
|
if (abs(avgsilence - dsp->busy_cadence.pattern[1]) > MAX(dsp->busy_cadence.pattern[1]*BUSY_PAT_PERCENT/100, 20)) {
|
|
#ifdef BUSYDETECT_DEBUG
|
|
ast_debug(5, "busy detector: avgsilence of %d not close enough to desired %d\n",
|
|
avgsilence, dsp->busy_cadence.pattern[1]);
|
|
#endif
|
|
res = 0;
|
|
}
|
|
}
|
|
#endif
|
|
#if !defined(BUSYDETECT_TONEONLY) && defined(BUSYDETECT_DEBUG)
|
|
if (res) {
|
|
ast_debug(5, "ast_dsp_busydetect detected busy, avgtone: %d, avgsilence %d\n", avgtone, avgsilence);
|
|
} else {
|
|
ast_debug(5, "busy detector: FAILED with avgtone: %d, avgsilence %d\n", avgtone, avgsilence);
|
|
}
|
|
#endif
|
|
return res;
|
|
}
|
|
|
|
static int ast_dsp_silence_noise_with_energy(struct ast_dsp *dsp, struct ast_frame *f, int *total, int *frames_energy, int noise)
|
|
{
|
|
short *s;
|
|
int len;
|
|
int x;
|
|
unsigned char *odata;
|
|
|
|
if (!f) {
|
|
return 0;
|
|
}
|
|
|
|
if (f->frametype != AST_FRAME_VOICE) {
|
|
ast_log(LOG_WARNING, "Can't calculate silence on a non-voice frame\n");
|
|
return 0;
|
|
}
|
|
|
|
if (ast_format_cache_is_slinear(f->subclass.format)) {
|
|
s = f->data.ptr;
|
|
len = f->datalen/2;
|
|
} else {
|
|
odata = f->data.ptr;
|
|
len = f->datalen;
|
|
if (ast_format_cmp(f->subclass.format, ast_format_ulaw) == AST_FORMAT_CMP_EQUAL) {
|
|
s = ast_alloca(len * 2);
|
|
for (x = 0; x < len; x++) {
|
|
s[x] = AST_MULAW(odata[x]);
|
|
}
|
|
} else if (ast_format_cmp(f->subclass.format, ast_format_alaw) == AST_FORMAT_CMP_EQUAL) {
|
|
s = ast_alloca(len * 2);
|
|
for (x = 0; x < len; x++) {
|
|
s[x] = AST_ALAW(odata[x]);
|
|
}
|
|
} else {
|
|
ast_log(LOG_WARNING, "Can only calculate silence on signed-linear, alaw or ulaw frames :(\n");
|
|
return 0;
|
|
}
|
|
}
|
|
|
|
if (noise) {
|
|
return __ast_dsp_silence_noise(dsp, s, len, NULL, total, frames_energy);
|
|
} else {
|
|
return __ast_dsp_silence_noise(dsp, s, len, total, NULL, frames_energy);
|
|
}
|
|
}
|
|
|
|
int ast_dsp_silence_with_energy(struct ast_dsp *dsp, struct ast_frame *f, int *totalsilence, int *frames_energy)
|
|
{
|
|
return ast_dsp_silence_noise_with_energy(dsp, f, totalsilence, frames_energy, 0);
|
|
}
|
|
|
|
int ast_dsp_silence(struct ast_dsp *dsp, struct ast_frame *f, int *totalsilence)
|
|
{
|
|
return ast_dsp_silence_noise_with_energy(dsp, f, totalsilence, NULL, 0);
|
|
}
|
|
|
|
int ast_dsp_noise(struct ast_dsp *dsp, struct ast_frame *f, int *totalnoise)
|
|
{
|
|
return ast_dsp_silence_noise_with_energy(dsp, f, totalnoise, NULL, 1);
|
|
}
|
|
|
|
|
|
struct ast_frame *ast_dsp_process(struct ast_channel *chan, struct ast_dsp *dsp, struct ast_frame *af)
|
|
{
|
|
int silence;
|
|
int res;
|
|
int digit = 0, fax_digit = 0, custom_freq_digit = 0;
|
|
int x;
|
|
short *shortdata;
|
|
unsigned char *odata;
|
|
int len;
|
|
struct ast_frame *outf = NULL;
|
|
|
|
if (!af) {
|
|
return NULL;
|
|
}
|
|
if (af->frametype != AST_FRAME_VOICE) {
|
|
return af;
|
|
}
|
|
|
|
odata = af->data.ptr;
|
|
len = af->datalen;
|
|
/* Make sure we have short data */
|
|
if (ast_format_cache_is_slinear(af->subclass.format)) {
|
|
shortdata = af->data.ptr;
|
|
len = af->datalen / 2;
|
|
} else if (ast_format_cmp(af->subclass.format, ast_format_ulaw) == AST_FORMAT_CMP_EQUAL) {
|
|
shortdata = ast_alloca(af->datalen * 2);
|
|
for (x = 0; x < len; x++) {
|
|
shortdata[x] = AST_MULAW(odata[x]);
|
|
}
|
|
} else if (ast_format_cmp(af->subclass.format, ast_format_alaw) == AST_FORMAT_CMP_EQUAL) {
|
|
shortdata = ast_alloca(af->datalen * 2);
|
|
for (x = 0; x < len; x++) {
|
|
shortdata[x] = AST_ALAW(odata[x]);
|
|
}
|
|
} else {
|
|
/*Display warning only once. Otherwise you would get hundreds of warnings every second */
|
|
if (dsp->display_inband_dtmf_warning) {
|
|
ast_log(LOG_WARNING, "Inband DTMF is not supported on codec %s. Use RFC2833\n", ast_format_get_name(af->subclass.format));
|
|
}
|
|
dsp->display_inband_dtmf_warning = 0;
|
|
return af;
|
|
}
|
|
|
|
/* Initially we do not want to mute anything */
|
|
dsp->mute_fragments = 0;
|
|
|
|
/* Need to run the silence detection stuff for silence suppression and busy detection */
|
|
if ((dsp->features & DSP_FEATURE_SILENCE_SUPPRESS) || (dsp->features & DSP_FEATURE_BUSY_DETECT)) {
|
|
res = __ast_dsp_silence_noise(dsp, shortdata, len, &silence, NULL, NULL);
|
|
}
|
|
|
|
if ((dsp->features & DSP_FEATURE_SILENCE_SUPPRESS) && silence) {
|
|
memset(&dsp->f, 0, sizeof(dsp->f));
|
|
dsp->f.frametype = AST_FRAME_NULL;
|
|
ast_frfree(af);
|
|
return ast_frisolate(&dsp->f);
|
|
}
|
|
if ((dsp->features & DSP_FEATURE_BUSY_DETECT) && ast_dsp_busydetect(dsp)) {
|
|
ast_channel_softhangup_internal_flag_add(chan, AST_SOFTHANGUP_DEV);
|
|
memset(&dsp->f, 0, sizeof(dsp->f));
|
|
dsp->f.frametype = AST_FRAME_CONTROL;
|
|
dsp->f.subclass.integer = AST_CONTROL_BUSY;
|
|
ast_frfree(af);
|
|
ast_debug(1, "Requesting Hangup because the busy tone was detected on channel %s\n", ast_channel_name(chan));
|
|
return ast_frisolate(&dsp->f);
|
|
}
|
|
|
|
if ((dsp->features & DSP_FEATURE_FAX_DETECT)) {
|
|
if ((dsp->faxmode & DSP_FAXMODE_DETECT_CNG) && tone_detect(dsp, &dsp->cng_tone_state, shortdata, len)) {
|
|
fax_digit = 'f';
|
|
}
|
|
|
|
if ((dsp->faxmode & DSP_FAXMODE_DETECT_CED) && tone_detect(dsp, &dsp->ced_tone_state, shortdata, len)) {
|
|
fax_digit = 'e';
|
|
}
|
|
}
|
|
|
|
if ((dsp->features & DSP_FEATURE_FREQ_DETECT)) {
|
|
if ((dsp->freqmode) && tone_detect(dsp, &dsp->cng_tone_state, shortdata, len)) {
|
|
custom_freq_digit = 'q';
|
|
}
|
|
}
|
|
|
|
if (dsp->features & (DSP_FEATURE_DIGIT_DETECT | DSP_FEATURE_BUSY_DETECT)) {
|
|
if (dsp->digitmode & DSP_DIGITMODE_MF) {
|
|
digit = mf_detect(dsp, &dsp->digit_state, shortdata, len, (dsp->digitmode & DSP_DIGITMODE_NOQUELCH) == 0, (dsp->digitmode & DSP_DIGITMODE_RELAXDTMF));
|
|
} else {
|
|
digit = dtmf_detect(dsp, &dsp->digit_state, shortdata, len, (dsp->digitmode & DSP_DIGITMODE_NOQUELCH) == 0, (dsp->digitmode & DSP_DIGITMODE_RELAXDTMF));
|
|
}
|
|
|
|
if (dsp->digit_state.current_digits) {
|
|
int event = 0, event_len = 0;
|
|
char event_digit = 0;
|
|
|
|
if (!dsp->dtmf_began) {
|
|
/* We have not reported DTMF_BEGIN for anything yet */
|
|
|
|
if (dsp->features & DSP_FEATURE_DIGIT_DETECT) {
|
|
event = AST_FRAME_DTMF_BEGIN;
|
|
event_digit = dsp->digit_state.digits[0];
|
|
}
|
|
dsp->dtmf_began = 1;
|
|
|
|
} else if (dsp->digit_state.current_digits > 1 || digit != dsp->digit_state.digits[0]) {
|
|
/* Digit changed. This means digit we have reported with DTMF_BEGIN ended */
|
|
if (dsp->features & DSP_FEATURE_DIGIT_DETECT) {
|
|
event = AST_FRAME_DTMF_END;
|
|
event_digit = dsp->digit_state.digits[0];
|
|
event_len = dsp->digit_state.digitlen[0] * 1000 / dsp->sample_rate;
|
|
}
|
|
memmove(&dsp->digit_state.digits[0], &dsp->digit_state.digits[1], dsp->digit_state.current_digits);
|
|
memmove(&dsp->digit_state.digitlen[0], &dsp->digit_state.digitlen[1], dsp->digit_state.current_digits * sizeof(dsp->digit_state.digitlen[0]));
|
|
dsp->digit_state.current_digits--;
|
|
dsp->dtmf_began = 0;
|
|
|
|
if (dsp->features & DSP_FEATURE_BUSY_DETECT) {
|
|
/* Reset Busy Detector as we have some confirmed activity */
|
|
memset(dsp->historicsilence, 0, sizeof(dsp->historicsilence));
|
|
memset(dsp->historicnoise, 0, sizeof(dsp->historicnoise));
|
|
ast_debug(1, "DTMF Detected - Reset busydetector\n");
|
|
}
|
|
}
|
|
|
|
if (event) {
|
|
memset(&dsp->f, 0, sizeof(dsp->f));
|
|
dsp->f.frametype = event;
|
|
dsp->f.subclass.integer = event_digit;
|
|
dsp->f.len = event_len;
|
|
outf = &dsp->f;
|
|
goto done;
|
|
}
|
|
}
|
|
}
|
|
|
|
if (fax_digit) {
|
|
/* Fax was detected - digit is either 'f' or 'e' */
|
|
|
|
memset(&dsp->f, 0, sizeof(dsp->f));
|
|
dsp->f.frametype = AST_FRAME_DTMF;
|
|
dsp->f.subclass.integer = fax_digit;
|
|
outf = &dsp->f;
|
|
goto done;
|
|
}
|
|
|
|
if (custom_freq_digit) {
|
|
/* Custom frequency was detected - digit is 'q' */
|
|
|
|
memset(&dsp->f, 0, sizeof(dsp->f));
|
|
dsp->f.frametype = AST_FRAME_DTMF;
|
|
dsp->f.subclass.integer = custom_freq_digit;
|
|
outf = &dsp->f;
|
|
goto done;
|
|
}
|
|
|
|
if ((dsp->features & DSP_FEATURE_CALL_PROGRESS)) {
|
|
res = __ast_dsp_call_progress(dsp, shortdata, len);
|
|
if (res) {
|
|
switch (res) {
|
|
case AST_CONTROL_ANSWER:
|
|
case AST_CONTROL_BUSY:
|
|
case AST_CONTROL_RINGING:
|
|
case AST_CONTROL_CONGESTION:
|
|
case AST_CONTROL_HANGUP:
|
|
memset(&dsp->f, 0, sizeof(dsp->f));
|
|
dsp->f.frametype = AST_FRAME_CONTROL;
|
|
dsp->f.subclass.integer = res;
|
|
dsp->f.src = "dsp_progress";
|
|
if (chan) {
|
|
ast_queue_frame(chan, &dsp->f);
|
|
}
|
|
break;
|
|
default:
|
|
ast_log(LOG_WARNING, "Don't know how to represent call progress message %d\n", res);
|
|
}
|
|
}
|
|
} else if ((dsp->features & DSP_FEATURE_WAITDIALTONE)) {
|
|
res = __ast_dsp_call_progress(dsp, shortdata, len);
|
|
}
|
|
|
|
done:
|
|
/* Mute fragment of the frame */
|
|
for (x = 0; x < dsp->mute_fragments; x++) {
|
|
memset(shortdata + dsp->mute_data[x].start, 0, sizeof(int16_t) * (dsp->mute_data[x].end - dsp->mute_data[x].start));
|
|
}
|
|
|
|
if (ast_format_cmp(af->subclass.format, ast_format_ulaw) == AST_FORMAT_CMP_EQUAL) {
|
|
for (x = 0; x < len; x++) {
|
|
odata[x] = AST_LIN2MU((unsigned short) shortdata[x]);
|
|
}
|
|
} else if (ast_format_cmp(af->subclass.format, ast_format_alaw) == AST_FORMAT_CMP_EQUAL) {
|
|
for (x = 0; x < len; x++) {
|
|
odata[x] = AST_LIN2A((unsigned short) shortdata[x]);
|
|
}
|
|
}
|
|
|
|
if (outf) {
|
|
if (chan) {
|
|
ast_queue_frame(chan, af);
|
|
}
|
|
ast_frfree(af);
|
|
return ast_frisolate(outf);
|
|
} else {
|
|
return af;
|
|
}
|
|
}
|
|
|
|
static void ast_dsp_prog_reset(struct ast_dsp *dsp)
|
|
{
|
|
int max = 0;
|
|
int x;
|
|
|
|
dsp->gsamp_size = modes[dsp->progmode].size;
|
|
dsp->gsamps = 0;
|
|
for (x = 0; x < FREQ_ARRAY_SIZE; x++) {
|
|
if (modes[dsp->progmode].freqs[x]) {
|
|
goertzel_init(&dsp->freqs[x], (float)modes[dsp->progmode].freqs[x], dsp->sample_rate);
|
|
max = x + 1;
|
|
}
|
|
}
|
|
dsp->freqcount = max;
|
|
dsp->ringtimeout = 0;
|
|
}
|
|
|
|
unsigned int ast_dsp_get_sample_rate(const struct ast_dsp *dsp)
|
|
{
|
|
return dsp->sample_rate;
|
|
}
|
|
|
|
static struct ast_dsp *__ast_dsp_new(unsigned int sample_rate)
|
|
{
|
|
struct ast_dsp *dsp;
|
|
|
|
if ((dsp = ast_calloc(1, sizeof(*dsp)))) {
|
|
dsp->threshold = DEFAULT_THRESHOLD;
|
|
dsp->features = DSP_FEATURE_SILENCE_SUPPRESS;
|
|
dsp->busycount = DSP_HISTORY;
|
|
dsp->digitmode = DSP_DIGITMODE_DTMF;
|
|
dsp->faxmode = DSP_FAXMODE_DETECT_CNG;
|
|
dsp->sample_rate = sample_rate;
|
|
dsp->freqcount = 0;
|
|
/* Initialize digit detector */
|
|
ast_digit_detect_init(&dsp->digit_state, dsp->digitmode & DSP_DIGITMODE_MF, dsp->sample_rate);
|
|
dsp->display_inband_dtmf_warning = 1;
|
|
/* Initialize initial DSP progress detect parameters */
|
|
ast_dsp_prog_reset(dsp);
|
|
/* Initialize fax detector */
|
|
ast_fax_detect_init(dsp);
|
|
}
|
|
return dsp;
|
|
}
|
|
|
|
struct ast_dsp *ast_dsp_new(void)
|
|
{
|
|
return __ast_dsp_new(DEFAULT_SAMPLE_RATE);
|
|
}
|
|
|
|
struct ast_dsp *ast_dsp_new_with_rate(unsigned int sample_rate)
|
|
{
|
|
return __ast_dsp_new(sample_rate);
|
|
}
|
|
|
|
void ast_dsp_set_features(struct ast_dsp *dsp, int features)
|
|
{
|
|
dsp->features = features;
|
|
if (!(features & DSP_FEATURE_DIGIT_DETECT)) {
|
|
dsp->display_inband_dtmf_warning = 0;
|
|
}
|
|
}
|
|
|
|
|
|
int ast_dsp_get_features(struct ast_dsp *dsp)
|
|
{
|
|
return (dsp->features);
|
|
}
|
|
|
|
|
|
void ast_dsp_free(struct ast_dsp *dsp)
|
|
{
|
|
ast_free(dsp);
|
|
}
|
|
|
|
void ast_dsp_set_threshold(struct ast_dsp *dsp, int threshold)
|
|
{
|
|
dsp->threshold = threshold;
|
|
}
|
|
|
|
void ast_dsp_set_busy_count(struct ast_dsp *dsp, int cadences)
|
|
{
|
|
if (cadences < 4) {
|
|
cadences = 4;
|
|
}
|
|
if (cadences > DSP_HISTORY) {
|
|
cadences = DSP_HISTORY;
|
|
}
|
|
dsp->busycount = cadences;
|
|
}
|
|
|
|
void ast_dsp_set_busy_pattern(struct ast_dsp *dsp, const struct ast_dsp_busy_pattern *cadence)
|
|
{
|
|
dsp->busy_cadence = *cadence;
|
|
ast_debug(1, "dsp busy pattern set to %d,%d,%d,%d\n", cadence->pattern[0], cadence->pattern[1], (cadence->length == 4) ? cadence->pattern[2] : 0, (cadence->length == 4) ? cadence->pattern[3] : 0);
|
|
}
|
|
|
|
void ast_dsp_digitreset(struct ast_dsp *dsp)
|
|
{
|
|
int i;
|
|
|
|
dsp->dtmf_began = 0;
|
|
if (dsp->digitmode & DSP_DIGITMODE_MF) {
|
|
mf_detect_state_t *s = &dsp->digit_state.td.mf;
|
|
/* Reinitialise the detector for the next block */
|
|
for (i = 0; i < 6; i++) {
|
|
goertzel_reset(&s->tone_out[i]);
|
|
}
|
|
s->hits[4] = s->hits[3] = s->hits[2] = s->hits[1] = s->hits[0] = 0;
|
|
s->current_hit = 0;
|
|
s->current_sample = 0;
|
|
} else {
|
|
dtmf_detect_state_t *s = &dsp->digit_state.td.dtmf;
|
|
/* Reinitialise the detector for the next block */
|
|
for (i = 0; i < 4; i++) {
|
|
goertzel_reset(&s->row_out[i]);
|
|
goertzel_reset(&s->col_out[i]);
|
|
}
|
|
s->lasthit = 0;
|
|
s->current_hit = 0;
|
|
s->energy = 0.0;
|
|
s->current_sample = 0;
|
|
s->hits = 0;
|
|
s->misses = 0;
|
|
}
|
|
|
|
dsp->digit_state.digits[0] = '\0';
|
|
dsp->digit_state.current_digits = 0;
|
|
}
|
|
|
|
void ast_dsp_reset(struct ast_dsp *dsp)
|
|
{
|
|
int x;
|
|
|
|
dsp->totalsilence = 0;
|
|
dsp->gsamps = 0;
|
|
for (x = 0; x < 4; x++) {
|
|
dsp->freqs[x].v2 = dsp->freqs[x].v3 = 0.0;
|
|
}
|
|
memset(dsp->historicsilence, 0, sizeof(dsp->historicsilence));
|
|
memset(dsp->historicnoise, 0, sizeof(dsp->historicnoise));
|
|
dsp->ringtimeout = 0;
|
|
}
|
|
|
|
int ast_dsp_set_digitmode(struct ast_dsp *dsp, int digitmode)
|
|
{
|
|
int new;
|
|
int old;
|
|
|
|
old = dsp->digitmode & (DSP_DIGITMODE_DTMF | DSP_DIGITMODE_MF | DSP_DIGITMODE_MUTECONF | DSP_DIGITMODE_MUTEMAX);
|
|
new = digitmode & (DSP_DIGITMODE_DTMF | DSP_DIGITMODE_MF | DSP_DIGITMODE_MUTECONF | DSP_DIGITMODE_MUTEMAX);
|
|
if (old != new) {
|
|
/* Must initialize structures if switching from MF to DTMF or vice-versa */
|
|
ast_digit_detect_init(&dsp->digit_state, new & DSP_DIGITMODE_MF, dsp->sample_rate);
|
|
}
|
|
dsp->digitmode = digitmode;
|
|
return 0;
|
|
}
|
|
|
|
int ast_dsp_set_freqmode(struct ast_dsp *dsp, int freq, int dur, int db, int squelch)
|
|
{
|
|
if (freq > 0) {
|
|
dsp->freqmode = 1;
|
|
ast_freq_detect_init(dsp, freq, dur, db, squelch);
|
|
} else {
|
|
dsp->freqmode = 0;
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
int ast_dsp_set_faxmode(struct ast_dsp *dsp, int faxmode)
|
|
{
|
|
if (dsp->faxmode != faxmode) {
|
|
dsp->faxmode = faxmode;
|
|
ast_fax_detect_init(dsp);
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
int ast_dsp_set_call_progress_zone(struct ast_dsp *dsp, char *zone)
|
|
{
|
|
int x;
|
|
|
|
for (x = 0; x < ARRAY_LEN(aliases); x++) {
|
|
if (!strcasecmp(aliases[x].name, zone)) {
|
|
dsp->progmode = aliases[x].mode;
|
|
ast_dsp_prog_reset(dsp);
|
|
return 0;
|
|
}
|
|
}
|
|
return -1;
|
|
}
|
|
|
|
int ast_dsp_was_muted(struct ast_dsp *dsp)
|
|
{
|
|
return (dsp->mute_fragments > 0);
|
|
}
|
|
|
|
int ast_dsp_get_tstate(struct ast_dsp *dsp)
|
|
{
|
|
return dsp->tstate;
|
|
}
|
|
|
|
int ast_dsp_get_tcount(struct ast_dsp *dsp)
|
|
{
|
|
return dsp->tcount;
|
|
}
|
|
|
|
static int _dsp_init(int reload)
|
|
{
|
|
struct ast_config *cfg;
|
|
struct ast_variable *v;
|
|
struct ast_flags config_flags = { reload ? CONFIG_FLAG_FILEUNCHANGED : 0 };
|
|
int cfg_threshold;
|
|
float cfg_twist;
|
|
|
|
if ((cfg = ast_config_load2(CONFIG_FILE_NAME, "dsp", config_flags)) == CONFIG_STATUS_FILEUNCHANGED) {
|
|
return 0;
|
|
}
|
|
|
|
thresholds[THRESHOLD_SILENCE] = DEFAULT_SILENCE_THRESHOLD;
|
|
dtmf_normal_twist = DEF_DTMF_NORMAL_TWIST;
|
|
dtmf_reverse_twist = DEF_DTMF_REVERSE_TWIST;
|
|
relax_dtmf_normal_twist = DEF_RELAX_DTMF_NORMAL_TWIST;
|
|
relax_dtmf_reverse_twist = DEF_RELAX_DTMF_REVERSE_TWIST;
|
|
dtmf_hits_to_begin = DEF_DTMF_HITS_TO_BEGIN;
|
|
dtmf_misses_to_end = DEF_DTMF_MISSES_TO_END;
|
|
|
|
if (cfg == CONFIG_STATUS_FILEMISSING || cfg == CONFIG_STATUS_FILEINVALID) {
|
|
return 0;
|
|
}
|
|
|
|
for (v = ast_variable_browse(cfg, "default"); v; v = v->next) {
|
|
if (!strcasecmp(v->name, "silencethreshold")) {
|
|
if (sscanf(v->value, "%30d", &cfg_threshold) < 1) {
|
|
ast_log(LOG_WARNING, "Unable to convert '%s' to a numeric value.\n", v->value);
|
|
} else if (cfg_threshold < 0) {
|
|
ast_log(LOG_WARNING, "Invalid silence threshold '%d' specified, using default\n", cfg_threshold);
|
|
} else {
|
|
thresholds[THRESHOLD_SILENCE] = cfg_threshold;
|
|
}
|
|
} else if (!strcasecmp(v->name, "dtmf_normal_twist")) {
|
|
if (sscanf(v->value, "%30f", &cfg_twist) < 1) {
|
|
ast_log(LOG_WARNING, "Unable to convert '%s' to a numeric value.\n", v->value);
|
|
} else if ((cfg_twist < 2.0) || (cfg_twist > 100.0)) { /* < 3.0dB or > 20dB */
|
|
ast_log(LOG_WARNING, "Invalid dtmf_normal_twist value '%.2f' specified, using default of %.2f\n", cfg_twist, dtmf_normal_twist);
|
|
} else {
|
|
dtmf_normal_twist = cfg_twist;
|
|
}
|
|
} else if (!strcasecmp(v->name, "dtmf_reverse_twist")) {
|
|
if (sscanf(v->value, "%30f", &cfg_twist) < 1) {
|
|
ast_log(LOG_WARNING, "Unable to convert '%s' to a numeric value.\n", v->value);
|
|
} else if ((cfg_twist < 2.0) || (cfg_twist > 100.0)) { /* < 3.0dB or > 20dB */
|
|
ast_log(LOG_WARNING, "Invalid dtmf_reverse_twist value '%.2f' specified, using default of %.2f\n", cfg_twist, dtmf_reverse_twist);
|
|
} else {
|
|
dtmf_reverse_twist = cfg_twist;
|
|
}
|
|
} else if (!strcasecmp(v->name, "relax_dtmf_normal_twist")) {
|
|
if (sscanf(v->value, "%30f", &cfg_twist) < 1) {
|
|
ast_log(LOG_WARNING, "Unable to convert '%s' to a numeric value.\n", v->value);
|
|
} else if ((cfg_twist < 2.0) || (cfg_twist > 100.0)) { /* < 3.0dB or > 20dB */
|
|
ast_log(LOG_WARNING, "Invalid relax_dtmf_normal_twist value '%.2f' specified, using default of %.2f\n", cfg_twist, relax_dtmf_normal_twist);
|
|
} else {
|
|
relax_dtmf_normal_twist = cfg_twist;
|
|
}
|
|
} else if (!strcasecmp(v->name, "relax_dtmf_reverse_twist")) {
|
|
if (sscanf(v->value, "%30f", &cfg_twist) < 1) {
|
|
ast_log(LOG_WARNING, "Unable to convert '%s' to a numeric value.\n", v->value);
|
|
} else if ((cfg_twist < 2.0) || (cfg_twist > 100.0)) { /* < 3.0dB or > 20dB */
|
|
ast_log(LOG_WARNING, "Invalid relax_dtmf_reverse_twist value '%.2f' specified, using default of %.2f\n", cfg_twist, relax_dtmf_reverse_twist);
|
|
} else {
|
|
relax_dtmf_reverse_twist = cfg_twist;
|
|
}
|
|
} else if (!strcasecmp(v->name, "dtmf_hits_to_begin")) {
|
|
if (sscanf(v->value, "%30d", &cfg_threshold) < 1) {
|
|
ast_log(LOG_WARNING, "Unable to convert '%s' to a numeric value.\n", v->value);
|
|
} else if (cfg_threshold < 1) { /* must be 1 or greater */
|
|
ast_log(LOG_WARNING, "Invalid dtmf_hits_to_begin value '%d' specified, using default of %d\n", cfg_threshold, dtmf_hits_to_begin);
|
|
} else {
|
|
dtmf_hits_to_begin = cfg_threshold;
|
|
}
|
|
} else if (!strcasecmp(v->name, "dtmf_misses_to_end")) {
|
|
if (sscanf(v->value, "%30d", &cfg_threshold) < 1) {
|
|
ast_log(LOG_WARNING, "Unable to convert '%s' to a numeric value.\n", v->value);
|
|
} else if (cfg_threshold < 1) { /* must be 1 or greater */
|
|
ast_log(LOG_WARNING, "Invalid dtmf_misses_to_end value '%d' specified, using default of %d\n", cfg_threshold, dtmf_misses_to_end);
|
|
} else {
|
|
dtmf_misses_to_end = cfg_threshold;
|
|
}
|
|
}
|
|
}
|
|
ast_config_destroy(cfg);
|
|
|
|
return 0;
|
|
}
|
|
|
|
int ast_dsp_get_threshold_from_settings(enum threshold which)
|
|
{
|
|
return thresholds[which];
|
|
}
|
|
|
|
#ifdef TEST_FRAMEWORK
|
|
static void test_tone_sample_gen(short *slin_buf, int samples, int rate, int freq, short amplitude)
|
|
{
|
|
int idx;
|
|
double sample_step = 2.0 * M_PI * freq / rate;/* radians per step */
|
|
|
|
for (idx = 0; idx < samples; ++idx) {
|
|
slin_buf[idx] = amplitude * sin(sample_step * idx);
|
|
}
|
|
}
|
|
#endif
|
|
|
|
#ifdef TEST_FRAMEWORK
|
|
static void test_tone_sample_gen_add(short *slin_buf, int samples, int rate, int freq, short amplitude)
|
|
{
|
|
int idx;
|
|
double sample_step = 2.0 * M_PI * freq / rate;/* radians per step */
|
|
|
|
for (idx = 0; idx < samples; ++idx) {
|
|
slin_buf[idx] += amplitude * sin(sample_step * idx);
|
|
}
|
|
}
|
|
#endif
|
|
|
|
#ifdef TEST_FRAMEWORK
|
|
static void test_dual_sample_gen(short *slin_buf, int samples, int rate, int f1, short a1, int f2, short a2)
|
|
{
|
|
test_tone_sample_gen(slin_buf, samples, rate, f1, a1);
|
|
test_tone_sample_gen_add(slin_buf, samples, rate, f2, a2);
|
|
}
|
|
#endif
|
|
|
|
#ifdef TEST_FRAMEWORK
|
|
#define TONE_AMPLITUDE_MAX 0x7fff /* Max signed linear amplitude */
|
|
#define TONE_AMPLITUDE_MIN 80 /* Min signed linear amplitude detectable */
|
|
|
|
static int test_tone_amplitude_sweep(struct ast_test *test, struct ast_dsp *dsp, tone_detect_state_t *tone_state)
|
|
{
|
|
short slin_buf[tone_state->block_size];
|
|
int result;
|
|
int idx;
|
|
struct {
|
|
short amp_val;
|
|
int detect;
|
|
} amp_tests[] = {
|
|
{ .amp_val = TONE_AMPLITUDE_MAX, .detect = 1, },
|
|
{ .amp_val = 10000, .detect = 1, },
|
|
{ .amp_val = 1000, .detect = 1, },
|
|
{ .amp_val = 100, .detect = 1, },
|
|
{ .amp_val = TONE_AMPLITUDE_MIN, .detect = 1, },
|
|
{ .amp_val = 75, .detect = 0, },
|
|
{ .amp_val = 10, .detect = 0, },
|
|
{ .amp_val = 1, .detect = 0, },
|
|
};
|
|
|
|
result = 0;
|
|
|
|
for (idx = 0; idx < ARRAY_LEN(amp_tests); ++idx) {
|
|
int detected;
|
|
int duration;
|
|
|
|
ast_debug(1, "Test %d Hz at amplitude %d\n",
|
|
tone_state->freq, amp_tests[idx].amp_val);
|
|
test_tone_sample_gen(slin_buf, tone_state->block_size, DEFAULT_SAMPLE_RATE,
|
|
tone_state->freq, amp_tests[idx].amp_val);
|
|
|
|
detected = 0;
|
|
for (duration = 0; !detected && duration < tone_state->hits_required + 3; ++duration) {
|
|
detected = tone_detect(dsp, tone_state, slin_buf, tone_state->block_size) ? 1 : 0;
|
|
}
|
|
if (amp_tests[idx].detect != detected) {
|
|
/*
|
|
* Both messages are needed. ast_debug for when figuring out
|
|
* what went wrong and the test update for normal output before
|
|
* you start debugging. The different logging methods are not
|
|
* synchronized.
|
|
*/
|
|
ast_debug(1,
|
|
"Test %d Hz at amplitude %d failed. Detected: %s\n",
|
|
tone_state->freq, amp_tests[idx].amp_val,
|
|
detected ? "yes" : "no");
|
|
ast_test_status_update(test,
|
|
"Test %d Hz at amplitude %d failed. Detected: %s\n",
|
|
tone_state->freq, amp_tests[idx].amp_val,
|
|
detected ? "yes" : "no");
|
|
result = -1;
|
|
}
|
|
tone_state->hit_count = 0;
|
|
}
|
|
|
|
return result;
|
|
}
|
|
#endif
|
|
|
|
#ifdef TEST_FRAMEWORK
|
|
static int test_dtmf_amplitude_sweep(struct ast_test *test, struct ast_dsp *dsp, int digit_index)
|
|
{
|
|
short slin_buf[DTMF_GSIZE];
|
|
int result;
|
|
int row;
|
|
int column;
|
|
int idx;
|
|
struct {
|
|
short amp_val;
|
|
int digit;
|
|
} amp_tests[] = {
|
|
/*
|
|
* XXX Since there is no current DTMF level detection issue. This test
|
|
* just checks the current detection levels.
|
|
*/
|
|
{ .amp_val = TONE_AMPLITUDE_MAX/2, .digit = dtmf_positions[digit_index], },
|
|
{ .amp_val = 10000, .digit = dtmf_positions[digit_index], },
|
|
{ .amp_val = 1000, .digit = dtmf_positions[digit_index], },
|
|
{ .amp_val = 500, .digit = dtmf_positions[digit_index], },
|
|
{ .amp_val = 250, .digit = dtmf_positions[digit_index], },
|
|
{ .amp_val = 200, .digit = dtmf_positions[digit_index], },
|
|
{ .amp_val = 180, .digit = dtmf_positions[digit_index], },
|
|
/* Various digits detect and not detect in this range */
|
|
{ .amp_val = 170, .digit = 0, },
|
|
{ .amp_val = 100, .digit = 0, },
|
|
/*
|
|
* Amplitudes below TONE_AMPLITUDE_MIN start having questionable detection
|
|
* over quantization and background noise.
|
|
*/
|
|
{ .amp_val = TONE_AMPLITUDE_MIN, .digit = 0, },
|
|
{ .amp_val = 75, .digit = 0, },
|
|
{ .amp_val = 10, .digit = 0, },
|
|
{ .amp_val = 1, .digit = 0, },
|
|
};
|
|
|
|
row = (digit_index >> 2) & 0x03;
|
|
column = digit_index & 0x03;
|
|
|
|
result = 0;
|
|
|
|
for (idx = 0; idx < ARRAY_LEN(amp_tests); ++idx) {
|
|
int digit;
|
|
int duration;
|
|
|
|
ast_debug(1, "Test '%c' at amplitude %d\n",
|
|
dtmf_positions[digit_index], amp_tests[idx].amp_val);
|
|
test_dual_sample_gen(slin_buf, ARRAY_LEN(slin_buf), DEFAULT_SAMPLE_RATE,
|
|
(int) dtmf_row[row], amp_tests[idx].amp_val,
|
|
(int) dtmf_col[column], amp_tests[idx].amp_val);
|
|
|
|
digit = 0;
|
|
for (duration = 0; !digit && duration < 3; ++duration) {
|
|
digit = dtmf_detect(dsp, &dsp->digit_state, slin_buf, ARRAY_LEN(slin_buf),
|
|
0, 0);
|
|
}
|
|
if (amp_tests[idx].digit != digit) {
|
|
/*
|
|
* Both messages are needed. ast_debug for when figuring out
|
|
* what went wrong and the test update for normal output before
|
|
* you start debugging. The different logging methods are not
|
|
* synchronized.
|
|
*/
|
|
ast_debug(1,
|
|
"Test '%c' at amplitude %d failed. Detected Digit: '%c'\n",
|
|
dtmf_positions[digit_index], amp_tests[idx].amp_val,
|
|
digit ?: ' ');
|
|
ast_test_status_update(test,
|
|
"Test '%c' at amplitude %d failed. Detected Digit: '%c'\n",
|
|
dtmf_positions[digit_index], amp_tests[idx].amp_val,
|
|
digit ?: ' ');
|
|
result = -1;
|
|
}
|
|
ast_dsp_digitreset(dsp);
|
|
}
|
|
|
|
return result;
|
|
}
|
|
#endif
|
|
|
|
#ifdef TEST_FRAMEWORK
|
|
static int test_dtmf_twist_sweep(struct ast_test *test, struct ast_dsp *dsp, int digit_index)
|
|
{
|
|
short slin_buf[DTMF_GSIZE];
|
|
int result;
|
|
int row;
|
|
int column;
|
|
int idx;
|
|
struct {
|
|
short amp_row;
|
|
short amp_col;
|
|
int digit;
|
|
} twist_tests[] = {
|
|
/*
|
|
* XXX Since there is no current DTMF twist detection issue. This test
|
|
* just checks the current detection levels.
|
|
*
|
|
* Normal twist has the column higher than the row amplitude.
|
|
* Reverse twist is the other way.
|
|
*/
|
|
{ .amp_row = 1000 + 1800, .amp_col = 1000 + 0, .digit = 0, },
|
|
{ .amp_row = 1000 + 1700, .amp_col = 1000 + 0, .digit = 0, },
|
|
/* Various digits detect and not detect in this range */
|
|
{ .amp_row = 1000 + 1400, .amp_col = 1000 + 0, .digit = dtmf_positions[digit_index], },
|
|
{ .amp_row = 1000 + 1300, .amp_col = 1000 + 0, .digit = dtmf_positions[digit_index], },
|
|
{ .amp_row = 1000 + 1200, .amp_col = 1000 + 0, .digit = dtmf_positions[digit_index], },
|
|
{ .amp_row = 1000 + 1100, .amp_col = 1000 + 0, .digit = dtmf_positions[digit_index], },
|
|
{ .amp_row = 1000 + 1000, .amp_col = 1000 + 0, .digit = dtmf_positions[digit_index], },
|
|
{ .amp_row = 1000 + 100, .amp_col = 1000 + 0, .digit = dtmf_positions[digit_index], },
|
|
{ .amp_row = 1000 + 0, .amp_col = 1000 + 100, .digit = dtmf_positions[digit_index], },
|
|
{ .amp_row = 1000 + 0, .amp_col = 1000 + 200, .digit = dtmf_positions[digit_index], },
|
|
{ .amp_row = 1000 + 0, .amp_col = 1000 + 300, .digit = dtmf_positions[digit_index], },
|
|
{ .amp_row = 1000 + 0, .amp_col = 1000 + 400, .digit = dtmf_positions[digit_index], },
|
|
{ .amp_row = 1000 + 0, .amp_col = 1000 + 500, .digit = dtmf_positions[digit_index], },
|
|
{ .amp_row = 1000 + 0, .amp_col = 1000 + 550, .digit = dtmf_positions[digit_index], },
|
|
/* Various digits detect and not detect in this range */
|
|
{ .amp_row = 1000 + 0, .amp_col = 1000 + 650, .digit = 0, },
|
|
{ .amp_row = 1000 + 0, .amp_col = 1000 + 700, .digit = 0, },
|
|
{ .amp_row = 1000 + 0, .amp_col = 1000 + 800, .digit = 0, },
|
|
};
|
|
float save_normal_twist;
|
|
float save_reverse_twist;
|
|
|
|
save_normal_twist = dtmf_normal_twist;
|
|
save_reverse_twist = dtmf_reverse_twist;
|
|
dtmf_normal_twist = DEF_DTMF_NORMAL_TWIST;
|
|
dtmf_reverse_twist = DEF_DTMF_REVERSE_TWIST;
|
|
|
|
row = (digit_index >> 2) & 0x03;
|
|
column = digit_index & 0x03;
|
|
|
|
result = 0;
|
|
|
|
for (idx = 0; idx < ARRAY_LEN(twist_tests); ++idx) {
|
|
int digit;
|
|
int duration;
|
|
|
|
ast_debug(1, "Test '%c' twist row %d col %d amplitudes\n",
|
|
dtmf_positions[digit_index],
|
|
twist_tests[idx].amp_row, twist_tests[idx].amp_col);
|
|
test_dual_sample_gen(slin_buf, ARRAY_LEN(slin_buf), DEFAULT_SAMPLE_RATE,
|
|
(int) dtmf_row[row], twist_tests[idx].amp_row,
|
|
(int) dtmf_col[column], twist_tests[idx].amp_col);
|
|
|
|
digit = 0;
|
|
for (duration = 0; !digit && duration < 3; ++duration) {
|
|
digit = dtmf_detect(dsp, &dsp->digit_state, slin_buf, ARRAY_LEN(slin_buf),
|
|
0, 0);
|
|
}
|
|
if (twist_tests[idx].digit != digit) {
|
|
/*
|
|
* Both messages are needed. ast_debug for when figuring out
|
|
* what went wrong and the test update for normal output before
|
|
* you start debugging. The different logging methods are not
|
|
* synchronized.
|
|
*/
|
|
ast_debug(1,
|
|
"Test '%c' twist row %d col %d amplitudes failed. Detected Digit: '%c'\n",
|
|
dtmf_positions[digit_index],
|
|
twist_tests[idx].amp_row, twist_tests[idx].amp_col,
|
|
digit ?: ' ');
|
|
ast_test_status_update(test,
|
|
"Test '%c' twist row %d col %d amplitudes failed. Detected Digit: '%c'\n",
|
|
dtmf_positions[digit_index],
|
|
twist_tests[idx].amp_row, twist_tests[idx].amp_col,
|
|
digit ?: ' ');
|
|
result = -1;
|
|
}
|
|
ast_dsp_digitreset(dsp);
|
|
}
|
|
|
|
dtmf_normal_twist = save_normal_twist;
|
|
dtmf_reverse_twist = save_reverse_twist;
|
|
|
|
return result;
|
|
}
|
|
#endif
|
|
|
|
#ifdef TEST_FRAMEWORK
|
|
static int test_tone_freq_sweep(struct ast_test *test, struct ast_dsp *dsp, tone_detect_state_t *tone_state, short amplitude)
|
|
{
|
|
short slin_buf[tone_state->block_size];
|
|
int result;
|
|
int freq;
|
|
int lower_freq;
|
|
int upper_freq;
|
|
|
|
/* Calculate detection frequency range */
|
|
lower_freq = tone_state->freq - 4;
|
|
upper_freq = tone_state->freq + 4;
|
|
|
|
result = 0;
|
|
|
|
/* Sweep frequencies loop. */
|
|
for (freq = 100; freq <= 3500; freq += 1) {
|
|
int detected;
|
|
int duration;
|
|
int expect_detection;
|
|
|
|
if (freq == tone_state->freq) {
|
|
/* This case is done by the amplitude sweep. */
|
|
continue;
|
|
}
|
|
|
|
expect_detection = (lower_freq <= freq && freq <= upper_freq) ? 1 : 0;
|
|
|
|
ast_debug(1, "Test %d Hz detection given %d Hz tone at amplitude %d. Range:%d-%d Expect detect: %s\n",
|
|
tone_state->freq, freq, amplitude, lower_freq, upper_freq,
|
|
expect_detection ? "yes" : "no");
|
|
test_tone_sample_gen(slin_buf, tone_state->block_size, DEFAULT_SAMPLE_RATE, freq,
|
|
amplitude);
|
|
|
|
detected = 0;
|
|
for (duration = 0; !detected && duration < tone_state->hits_required + 3; ++duration) {
|
|
detected = tone_detect(dsp, tone_state, slin_buf, tone_state->block_size) ? 1 : 0;
|
|
}
|
|
if (expect_detection != detected) {
|
|
/*
|
|
* Both messages are needed. ast_debug for when figuring out
|
|
* what went wrong and the test update for normal output before
|
|
* you start debugging. The different logging methods are not
|
|
* synchronized.
|
|
*/
|
|
ast_debug(1,
|
|
"Test %d Hz detection given %d Hz tone at amplitude %d failed. Range:%d-%d Detected: %s\n",
|
|
tone_state->freq, freq, amplitude, lower_freq, upper_freq,
|
|
detected ? "yes" : "no");
|
|
ast_test_status_update(test,
|
|
"Test %d Hz detection given %d Hz tone at amplitude %d failed. Range:%d-%d Detected: %s\n",
|
|
tone_state->freq, freq, amplitude, lower_freq, upper_freq,
|
|
detected ? "yes" : "no");
|
|
result = -1;
|
|
}
|
|
tone_state->hit_count = 0;
|
|
}
|
|
|
|
return result;
|
|
}
|
|
#endif
|
|
|
|
#ifdef TEST_FRAMEWORK
|
|
AST_TEST_DEFINE(test_dsp_fax_detect)
|
|
{
|
|
struct ast_dsp *dsp;
|
|
enum ast_test_result_state result;
|
|
|
|
switch (cmd) {
|
|
case TEST_INIT:
|
|
info->name = "fax";
|
|
info->category = "/main/dsp/";
|
|
info->summary = "DSP fax tone detect unit test";
|
|
info->description =
|
|
"Tests fax tone detection code.";
|
|
return AST_TEST_NOT_RUN;
|
|
case TEST_EXECUTE:
|
|
break;
|
|
}
|
|
|
|
dsp = ast_dsp_new();
|
|
if (!dsp) {
|
|
return AST_TEST_FAIL;
|
|
}
|
|
|
|
result = AST_TEST_PASS;
|
|
|
|
/* Test CNG tone amplitude detection */
|
|
if (test_tone_amplitude_sweep(test, dsp, &dsp->cng_tone_state)) {
|
|
result = AST_TEST_FAIL;
|
|
}
|
|
|
|
/* Test CED tone amplitude detection */
|
|
if (test_tone_amplitude_sweep(test, dsp, &dsp->ced_tone_state)) {
|
|
result = AST_TEST_FAIL;
|
|
}
|
|
|
|
/* Test CNG tone frequency detection */
|
|
if (test_tone_freq_sweep(test, dsp, &dsp->cng_tone_state, TONE_AMPLITUDE_MAX)) {
|
|
result = AST_TEST_FAIL;
|
|
}
|
|
if (test_tone_freq_sweep(test, dsp, &dsp->cng_tone_state, TONE_AMPLITUDE_MIN)) {
|
|
result = AST_TEST_FAIL;
|
|
}
|
|
|
|
/* Test CED tone frequency detection */
|
|
if (test_tone_freq_sweep(test, dsp, &dsp->ced_tone_state, TONE_AMPLITUDE_MAX)) {
|
|
result = AST_TEST_FAIL;
|
|
}
|
|
if (test_tone_freq_sweep(test, dsp, &dsp->ced_tone_state, TONE_AMPLITUDE_MIN)) {
|
|
result = AST_TEST_FAIL;
|
|
}
|
|
|
|
ast_dsp_free(dsp);
|
|
return result;
|
|
}
|
|
#endif
|
|
|
|
#ifdef TEST_FRAMEWORK
|
|
AST_TEST_DEFINE(test_dsp_dtmf_detect)
|
|
{
|
|
int idx;
|
|
struct ast_dsp *dsp;
|
|
enum ast_test_result_state result;
|
|
|
|
switch (cmd) {
|
|
case TEST_INIT:
|
|
info->name = "dtmf";
|
|
info->category = "/main/dsp/";
|
|
info->summary = "DSP DTMF detect unit test";
|
|
info->description =
|
|
"Tests DTMF detection code.";
|
|
return AST_TEST_NOT_RUN;
|
|
case TEST_EXECUTE:
|
|
break;
|
|
}
|
|
|
|
dsp = ast_dsp_new();
|
|
if (!dsp) {
|
|
return AST_TEST_FAIL;
|
|
}
|
|
|
|
result = AST_TEST_PASS;
|
|
|
|
for (idx = 0; dtmf_positions[idx]; ++idx) {
|
|
if (test_dtmf_amplitude_sweep(test, dsp, idx)) {
|
|
result = AST_TEST_FAIL;
|
|
}
|
|
}
|
|
|
|
for (idx = 0; dtmf_positions[idx]; ++idx) {
|
|
if (test_dtmf_twist_sweep(test, dsp, idx)) {
|
|
result = AST_TEST_FAIL;
|
|
}
|
|
}
|
|
|
|
ast_dsp_free(dsp);
|
|
return result;
|
|
}
|
|
#endif
|
|
|
|
static int unload_module(void)
|
|
{
|
|
AST_TEST_UNREGISTER(test_dsp_fax_detect);
|
|
AST_TEST_UNREGISTER(test_dsp_dtmf_detect);
|
|
|
|
return 0;
|
|
}
|
|
|
|
static int load_module(void)
|
|
{
|
|
if (_dsp_init(0)) {
|
|
return AST_MODULE_LOAD_FAILURE;
|
|
}
|
|
|
|
AST_TEST_REGISTER(test_dsp_fax_detect);
|
|
AST_TEST_REGISTER(test_dsp_dtmf_detect);
|
|
|
|
return AST_MODULE_LOAD_SUCCESS;
|
|
}
|
|
|
|
static int reload_module(void)
|
|
{
|
|
return _dsp_init(1);
|
|
}
|
|
|
|
AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_GLOBAL_SYMBOLS | AST_MODFLAG_LOAD_ORDER, "DSP",
|
|
.support_level = AST_MODULE_SUPPORT_CORE,
|
|
.load = load_module,
|
|
.unload = unload_module,
|
|
.reload = reload_module,
|
|
.load_pri = AST_MODPRI_CORE,
|
|
.requires = "extconfig",
|
|
);
|