dsp.c

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/*
 * Asterisk -- An open source telephony toolkit.
 *
 * Copyright (C) 1999 - 2005, Digium, Inc.
 *
 * Mark Spencer <markster@digium.com>
 *
 * Goertzel routines are borrowed from Steve Underwood's tremendous work on the
 * DTMF detector.
 *
 * See http://www.asterisk.org for more information about
 * the Asterisk project. Please do not directly contact
 * any of the maintainers of this project for assistance;
 * the project provides a web site, mailing lists and IRC
 * channels for your use.
 *
 * This program is free software, distributed under the terms of
 * the GNU General Public License Version 2. See the LICENSE file
 * at the top of the source tree.
 */

/*! \file
 *
 * \brief Convenience Signal Processing routines
 *
 * \author Mark Spencer <markster@digium.com>
 * \author Steve Underwood <steveu@coppice.org>
 */

/* Some routines from tone_detect.c by Steven Underwood as published under the zapata library */
/*
   tone_detect.c - General telephony tone detection, and specific
               detection of DTMF.

   Copyright (C) 2001  Steve Underwood <steveu@coppice.org>

   Despite my general liking of the GPL, I place this code in the
   public domain for the benefit of all mankind - even the slimy
   ones who might try to proprietize my work and use it to my
   detriment.
*/

#include "asterisk.h"

ASTERISK_FILE_VERSION(__FILE__, "$Revision: 268690 $")

#include <math.h>

#include "asterisk/frame.h"
#include "asterisk/channel.h"
#include "asterisk/dsp.h"
#include "asterisk/ulaw.h"
#include "asterisk/alaw.h"
#include "asterisk/utils.h"
#include "asterisk/options.h"
#include "asterisk/config.h"

/*! Number of goertzels for progress detect */
enum gsamp_size {
   GSAMP_SIZE_NA = 183,       /*!< North America - 350, 440, 480, 620, 950, 1400, 1800 Hz */
   GSAMP_SIZE_CR = 188,       /*!< Costa Rica, Brazil - Only care about 425 Hz */
   GSAMP_SIZE_UK = 160        /*!< UK disconnect goertzel feed - should trigger 400hz */
};

enum prog_mode {
   PROG_MODE_NA = 0,
   PROG_MODE_CR,
   PROG_MODE_UK
};

enum freq_index { 
   /*! For US modes { */
   HZ_350 = 0,
   HZ_440,
   HZ_480,
   HZ_620,
   HZ_950,
   HZ_1400,
   HZ_1800, /*!< } */

   /*! For CR/BR modes */
   HZ_425 = 0,

   /*! For UK mode */
   HZ_350UK = 0,
   HZ_400UK,
   HZ_440UK
};

static struct progalias {
   char *name;
   enum prog_mode mode;
} aliases[] = {
   { "us", PROG_MODE_NA },
   { "ca", PROG_MODE_NA },
   { "cr", PROG_MODE_CR },
   { "br", PROG_MODE_CR },
   { "uk", PROG_MODE_UK },
};

static struct progress {
   enum gsamp_size size;
   int freqs[7];
} modes[] = {
   { GSAMP_SIZE_NA, { 350, 440, 480, 620, 950, 1400, 1800 } }, /*!< North America */
   { GSAMP_SIZE_CR, { 425 } },                                 /*!< Costa Rica, Brazil */
   { GSAMP_SIZE_UK, { 350, 400, 440 } },                                   /*!< UK */
};

/*!\brief This value is the minimum threshold, calculated by averaging all
 * of the samples within a frame, for which a frame is determined to either
 * be silence (below the threshold) or noise (above the threshold).  Please
 * note that while the default threshold is an even exponent of 2, there is
 * no requirement that it be so.  The threshold will accept any value between
 * 0 and 32767.
 */
#define DEFAULT_THRESHOLD  512

enum busy_detect {
   BUSY_PERCENT = 10,      /*!< The percentage difference between the two last silence periods */
   BUSY_PAT_PERCENT = 7,   /*!< The percentage difference between measured and actual pattern */
   BUSY_THRESHOLD = 100,   /*!< Max number of ms difference between max and min times in busy */
   BUSY_MIN = 75,          /*!< Busy must be at least 80 ms in half-cadence */
   BUSY_MAX =3100          /*!< Busy can't be longer than 3100 ms in half-cadence */
};

/*! Remember last 15 units */
#define DSP_HISTORY     15

#define TONE_THRESH     10.0  /*!< How much louder the tone should be than channel energy */
#define TONE_MIN_THRESH    1e8   /*!< How much tone there should be at least to attempt */

/*! All THRESH_XXX values are in GSAMP_SIZE chunks (us = 22ms) */
enum gsamp_thresh {
   THRESH_RING = 8,           /*!< Need at least 150ms ring to accept */
   THRESH_TALK = 2,           /*!< Talk detection does not work continuously */
   THRESH_BUSY = 4,           /*!< Need at least 80ms to accept */
   THRESH_CONGESTION = 4,     /*!< Need at least 80ms to accept */
   THRESH_HANGUP = 60,        /*!< Need at least 1300ms to accept hangup */
   THRESH_RING2ANSWER = 300   /*!< Timeout from start of ring to answer (about 6600 ms) */
};

#define  MAX_DTMF_DIGITS      128

/* Basic DTMF specs:
 *
 * Minimum tone on = 40ms
 * Minimum tone off = 50ms
 * Maximum digit rate = 10 per second
 * Normal twist <= 8dB accepted
 * Reverse twist <= 4dB accepted
 * S/N >= 15dB will detect OK
 * Attenuation <= 26dB will detect OK
 * Frequency tolerance +- 1.5% will detect, +-3.5% will reject
 */

#define DTMF_THRESHOLD     8.0e7
#define FAX_THRESHOLD      8.0e7
#define FAX_2ND_HARMONIC   2.0     /* 4dB */
#define DTMF_NORMAL_TWIST  6.3     /* 8dB */
#ifdef   RADIO_RELAX
#define DTMF_REVERSE_TWIST          (relax ? 6.5 : 2.5)     /* 4dB normal */
#else
#define DTMF_REVERSE_TWIST          (relax ? 4.0 : 2.5)     /* 4dB normal */
#endif
#define DTMF_RELATIVE_PEAK_ROW   6.3     /* 8dB */
#define DTMF_RELATIVE_PEAK_COL   6.3     /* 8dB */
#define DTMF_2ND_HARMONIC_ROW       (relax ? 1.7 : 2.5)     /* 4dB normal */
#define DTMF_2ND_HARMONIC_COL 63.1    /* 18dB */
#define DTMF_TO_TOTAL_ENERGY  42.0

#define BELL_MF_THRESHOLD  1.6e9
#define BELL_MF_TWIST      4.0     /* 6dB */
#define BELL_MF_RELATIVE_PEAK 12.6    /* 11dB */

#if defined(BUSYDETECT_TONEONLY) && defined(BUSYDETECT_COMPARE_TONE_AND_SILENCE)
#error You cant use BUSYDETECT_TONEONLY together with BUSYDETECT_COMPARE_TONE_AND_SILENCE
#endif

/* The CNG signal consists of the transmission of 1100 Hz for 1/2 second,
 * followed by a 3 second silent (2100 Hz OFF) period.
 */
#define FAX_TONE_CNG_FREQ  1100
#define FAX_TONE_CNG_DURATION 500
#define FAX_TONE_CNG_DB    16

/* This signal may be sent by the Terminating FAX machine anywhere between
 * 1.8 to 2.5 seconds AFTER answering the call.  The CED signal consists
 * of a 2100 Hz tone that is from 2.6 to 4 seconds in duration.
*/
#define FAX_TONE_CED_FREQ  2100
#define FAX_TONE_CED_DURATION 2600
#define FAX_TONE_CED_DB    16

#define SAMPLE_RATE     8000

/* How many samples a frame has.  This constant is used when calculating
 * Goertzel block size for tone_detect.  It is only important if we want to
 * remove (squelch) the tone. In this case it is important to have block
 * size not to exceed size of voice frame.  Otherwise by the moment the tone
 * is detected it is too late to squelch it from previous frames.
 */
#define SAMPLES_IN_FRAME   160

/* MF goertzel size */
#define MF_GSIZE     120

/* DTMF goertzel size */
#define DTMF_GSIZE      102

/* How many successive hits needed to consider begin of a digit */
#define DTMF_HITS_TO_BEGIN 2
/* How many successive misses needed to consider end of a digit */
#define DTMF_MISSES_TO_END 3

/*!
 * \brief The default silence threshold we will use if an alternate
 * configured value is not present or is invalid.
 */
static const int DEFAULT_SILENCE_THRESHOLD = 256;

#define CONFIG_FILE_NAME "dsp.conf"

typedef struct {
   int v2;
   int v3;
   int chunky;
   int fac;
   int samples;
} goertzel_state_t;

typedef struct {
   int value;
   int power;
} goertzel_result_t;

typedef struct
{
   int freq;
   int block_size;
   int squelch;      /* Remove (squelch) tone */
   goertzel_state_t tone;
   float energy;     /* Accumulated energy of the current block */
   int samples_pending; /* Samples remain to complete the current block */
   int mute_samples; /* How many additional samples needs to be muted to suppress already detected tone */

   int hits_required;   /* How many successive blocks with tone we are looking for */
   float threshold;  /* Energy of the tone relative to energy from all other signals to consider a hit */

   int hit_count;    /* How many successive blocks we consider tone present */
   int last_hit;     /* Indicates if the last processed block was a hit */

} tone_detect_state_t;

typedef struct
{
   goertzel_state_t row_out[4];
   goertzel_state_t col_out[4];
   int hits_to_begin;      /* How many successive hits needed to consider begin of a digit */
   int misses_to_end;      /* How many successive misses needed to consider end of a digit */
   int hits;         /* How many successive hits we have seen already */
   int misses;       /* How many successive misses we have seen already */
   int lasthit;
   int current_hit;
   float energy;
   int current_sample;
   int mute_samples;
} dtmf_detect_state_t;

typedef struct
{
   goertzel_state_t tone_out[6];
   int current_hit;
   int hits[5];
   int current_sample;
   int mute_samples;
} mf_detect_state_t;

typedef struct
{
   char digits[MAX_DTMF_DIGITS + 1];
   int digitlen[MAX_DTMF_DIGITS + 1];
   int current_digits;
   int detected_digits;
   int lost_digits;

   union {
      dtmf_detect_state_t dtmf;
      mf_detect_state_t mf;
   } td;
} digit_detect_state_t;

static const float dtmf_row[] = {
   697.0,  770.0,  852.0,  941.0
};
static const float dtmf_col[] = {
   1209.0, 1336.0, 1477.0, 1633.0
};
static const float mf_tones[] = {
   700.0, 900.0, 1100.0, 1300.0, 1500.0, 1700.0
};
static const char dtmf_positions[] = "123A" "456B" "789C" "*0#D";
static const char bell_mf_positions[] = "1247C-358A--69*---0B----#";
static int thresholds[THRESHOLD_MAX];

static inline void goertzel_sample(goertzel_state_t *s, short sample)
{
   int v1;
   
   v1 = s->v2;
   s->v2 = s->v3;
   
   s->v3 = (s->fac * s->v2) >> 15;
   s->v3 = s->v3 - v1 + (sample >> s->chunky);
   if (abs(s->v3) > 32768) {
      s->chunky++;
      s->v3 = s->v3 >> 1;
      s->v2 = s->v2 >> 1;
      v1 = v1 >> 1;
   }
}

static inline void goertzel_update(goertzel_state_t *s, short *samps, int count)
{
   int i;
   
   for (i = 0; i < count; i++) {
      goertzel_sample(s, samps[i]);
   }
}


static inline float goertzel_result(goertzel_state_t *s)
{
   goertzel_result_t r;
   r.value = (s->v3 * s->v3) + (s->v2 * s->v2);
   r.value -= ((s->v2 * s->v3) >> 15) * s->fac;
   r.power = s->chunky * 2;
   return (float)r.value * (float)(1 << r.power);
}

static inline void goertzel_init(goertzel_state_t *s, float freq, int samples)
{
   s->v2 = s->v3 = s->chunky = 0.0;
   s->fac = (int)(32768.0 * 2.0 * cos(2.0 * M_PI * freq / SAMPLE_RATE));
   s->samples = samples;
}

static inline void goertzel_reset(goertzel_state_t *s)
{
   s->v2 = s->v3 = s->chunky = 0.0;
}

typedef struct {
   int start;
   int end;
} fragment_t;

/* Note on tone suppression (squelching). Individual detectors (DTMF/MF/generic tone)
 * report fragmens of the frame in which detected tone resides and which needs
 * to be "muted" in order to suppress the tone. To mark fragment for muting,
 * detectors call mute_fragment passing fragment_t there. Multiple fragments
 * can be marked and ast_dsp_process later will mute all of them.
 *
 * Note: When tone starts in the middle of a Goertzel block, it won't be properly
 * detected in that block, only in the next. If we only mute the next block
 * where tone is actually detected, the user will still hear beginning
 * of the tone in preceeding block. This is why we usually want to mute some amount
 * of samples preceeding and following the block where tone was detected.
*/

struct ast_dsp {
   struct ast_frame f;
   int threshold;
   int totalsilence;
   int totalnoise;
   int features;
   int ringtimeout;
   int busymaybe;
   int busycount;
   int busy_tonelength;
   int busy_quietlength;
   int historicnoise[DSP_HISTORY];
   int historicsilence[DSP_HISTORY];
   goertzel_state_t freqs[7];
   int freqcount;
   int gsamps;
   enum gsamp_size gsamp_size;
   enum prog_mode progmode;
   int tstate;
   int tcount;
   int digitmode;
   int faxmode;
   int dtmf_began;
   int display_inband_dtmf_warning;
   float genergy;
   int mute_fragments;
   fragment_t mute_data[5];
   digit_detect_state_t digit_state;
   tone_detect_state_t cng_tone_state;
   tone_detect_state_t ced_tone_state;
};

static void mute_fragment(struct ast_dsp *dsp, fragment_t *fragment)
{
   if (dsp->mute_fragments >= ARRAY_LEN(dsp->mute_data)) {
      ast_log(LOG_ERROR, "Too many fragments to mute. Ignoring\n");
      return;
   }

   dsp->mute_data[dsp->mute_fragments++] = *fragment;
}

static void ast_tone_detect_init(tone_detect_state_t *s, int freq, int duration, int amp)
{
   int duration_samples;
   float x;
   int periods_in_block;

   s->freq = freq;

   /* Desired tone duration in samples */
   duration_samples = duration * SAMPLE_RATE / 1000;
   /* We want to allow 10% deviation of tone duration */
   duration_samples = duration_samples * 9 / 10;

   /* If we want to remove tone, it is important to have block size not
      to exceed frame size. Otherwise by the moment tone is detected it is too late
      to squelch it from previous frames */
   s->block_size = SAMPLES_IN_FRAME;

   periods_in_block = s->block_size * freq / SAMPLE_RATE;

   /* Make sure we will have at least 5 periods at target frequency for analisys.
      This may make block larger than expected packet and will make squelching impossible
      but at least we will be detecting the tone */
   if (periods_in_block < 5)
      periods_in_block = 5;

   /* Now calculate final block size. It will contain integer number of periods */
   s->block_size = periods_in_block * SAMPLE_RATE / freq;

   /* tone_detect is currently only used to detect fax tones and we
      do not need suqlching the fax tones */
   s->squelch = 0;

   /* Account for the first and the last block to be incomplete
      and thus no tone will be detected in them */
   s->hits_required = (duration_samples - (s->block_size - 1)) / s->block_size;

   goertzel_init(&s->tone, freq, s->block_size);

   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);
   ast_tone_detect_init(&s->ced_tone_state, FAX_TONE_CED_FREQ, FAX_TONE_CED_DURATION, FAX_TONE_CED_DB);
}

static void ast_dtmf_detect_init (dtmf_detect_state_t *s)
{
   int i;

   s->lasthit = 0;
   s->current_hit = 0;
   for (i = 0;  i < 4;  i++) {
      goertzel_init(&s->row_out[i], dtmf_row[i], DTMF_GSIZE);
      goertzel_init(&s->col_out[i], dtmf_col[i], DTMF_GSIZE);
      s->energy = 0.0;
   }
   s->current_sample = 0;
   s->hits = 0;
   s->misses = 0;

   s->hits_to_begin = DTMF_HITS_TO_BEGIN;
   s->misses_to_end = DTMF_MISSES_TO_END;
}

static void ast_mf_detect_init (mf_detect_state_t *s)
{
   int i;
   s->hits[0] = s->hits[1] = s->hits[2] = s->hits[3] = s->hits[4] = 0;
   for (i = 0;  i < 6;  i++) {
      goertzel_init (&s->tone_out[i], mf_tones[i], 160);
   }
   s->current_sample = 0;
   s->current_hit = 0;
}

static void ast_digit_detect_init(digit_detect_state_t *s, int mf)
{
   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);
   } else {
      ast_dtmf_detect_init(&s->td.dtmf);
   }
}

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;
   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++) {
         /* signed 32 bit int should be enough to suqare any possible signed 16 bit value */
         s->energy += (int32_t) *ptr * (int32_t) *ptr;

         goertzel_sample(&s->tone, *ptr);
      }

      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, "tone %d, Ew=%.2E, Et=%.2E, s/n=%10.2f\n", s->freq, tone_energy, s->energy, tone_energy / (s->energy - tone_energy));
      hit = 0;
      if (tone_energy > s->energy * s->threshold) {
         ast_debug(10, "Hit! count=%d\n", s->hit_count);
         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 done 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[4];
   float col_energy[4];
   float famp;
   int i;
   int j;
   int sample;
   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++) {
         famp = amp[j];
         s->td.dtmf.energy += famp*famp;
         /* 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, amp[j]);
         goertzel_sample(s->td.dtmf.col_out, amp[j]);
         goertzel_sample(s->td.dtmf.row_out + 1, amp[j]);
         goertzel_sample(s->td.dtmf.col_out + 1, amp[j]);
         goertzel_sample(s->td.dtmf.row_out + 2, amp[j]);
         goertzel_sample(s->td.dtmf.col_out + 2, amp[j]);
         goertzel_sample(s->td.dtmf.row_out + 3, amp[j]);
         goertzel_sample(s->td.dtmf.col_out + 3, amp[j]);
      }
      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 < 4;  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;
         }
      }
      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] * DTMF_REVERSE_TWIST &&
          col_energy[best_col] * DTMF_NORMAL_TWIST > row_energy[best_row]) {
         /* Relative peak test */
         for (i = 0;  i < 4;  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 >= 4 &&
             (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];
         }
      } 

      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 == s->td.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) {
         if (hit == s->td.dtmf.lasthit) {
            s->td.dtmf.hits++;
         } else {
            s->td.dtmf.hits = 1;
         }

         if (s->td.dtmf.hits == s->td.dtmf.hits_to_begin && hit != s->td.dtmf.current_hit) {
            store_digit(s, hit);
            s->td.dtmf.current_hit = hit;
            s->td.dtmf.misses = 0;
         }
      } else {
         s->td.dtmf.hits = 0;
      }

      s->td.dtmf.lasthit = hit;

      /* 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 < 4; 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;
   float famp;
   int i;
   int j;
   int sample;
   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++) {
         famp = amp[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 */
         goertzel_sample(s->td.mf.tone_out, amp[j]);
         goertzel_sample(s->td.mf.tone_out + 1, amp[j]);
         goertzel_sample(s->td.mf.tone_out + 2, amp[j]);
         goertzel_sample(s->td.mf.tone_out + 3, amp[j]);
         goertzel_sample(s->td.mf.tone_out + 4, amp[j]);
         goertzel_sample(s->td.mf.tone_out + 5, amp[j]);
      }
      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 + DTMF_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)
{
   int x;
   int y;
   int pass;
   int newstate = DSP_TONE_STATE_SILENCE;
   int res = 0;
   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++) {
         for (y = 0; y < dsp->freqcount; y++) {
            goertzel_sample(&dsp->freqs[y], s[x]);
         }
         dsp->genergy += s[x] * s[x];
      }
      s += pass;
      dsp->gsamps += pass;
      len -= pass;
      if (dsp->gsamps == dsp->gsamp_size) {
         float hz[7];
         for (y = 0; y < 7; 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 '%d'\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 (inf->subclass.codec != AST_FORMAT_SLINEAR) {
      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 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 / 8;
      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 / 8;
      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;
   }
   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 (!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 know the expected busy tone length, check we are in the range */
   if (res && (dsp->busy_tonelength > 0)) {
      if (abs(avgtone - dsp->busy_tonelength) > (dsp->busy_tonelength*BUSY_PAT_PERCENT/100)) {
#ifdef BUSYDETECT_DEBUG
         ast_debug(5, "busy detector: avgtone of %d not close enough to desired %d\n",
            avgtone, dsp->busy_tonelength);
#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_quietlength > 0)) {
      if (abs(avgsilence - dsp->busy_quietlength) > (dsp->busy_quietlength*BUSY_PAT_PERCENT/100)) {
#ifdef BUSYDETECT_DEBUG
      ast_debug(5, "busy detector: avgsilence of %d not close enough to desired %d\n",
         avgsilence, dsp->busy_quietlength);
#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;
}

int ast_dsp_silence(struct ast_dsp *dsp, struct ast_frame *f, int *totalsilence)
{
   short *s;
   int len;
   
   if (f->frametype != AST_FRAME_VOICE) {
      ast_log(LOG_WARNING, "Can't calculate silence on a non-voice frame\n");
      return 0;
   }
   if (f->subclass.codec != AST_FORMAT_SLINEAR) {
      ast_log(LOG_WARNING, "Can only calculate silence on signed-linear frames :(\n");
      return 0;
   }
   s = f->data.ptr;
   len = f->datalen/2;
   return __ast_dsp_silence_noise(dsp, s, len, totalsilence, NULL);
}

int ast_dsp_noise(struct ast_dsp *dsp, struct ast_frame *f, int *totalnoise)
{
       short *s;
       int len;

       if (f->frametype != AST_FRAME_VOICE) {
               ast_log(LOG_WARNING, "Can't calculate noise on a non-voice frame\n");
               return 0;
       }
       if (f->subclass.codec != AST_FORMAT_SLINEAR) {
               ast_log(LOG_WARNING, "Can only calculate noise on signed-linear frames :(\n");
               return 0;
       }
       s = f->data.ptr;
       len = f->datalen/2;
       return __ast_dsp_silence_noise(dsp, s, len, NULL, totalnoise);
}


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;
   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 */
   switch (af->subclass.codec) {
   case AST_FORMAT_SLINEAR:
      shortdata = af->data.ptr;
      len = af->datalen / 2;
      break;
   case AST_FORMAT_ULAW:
   case AST_FORMAT_TESTLAW:
      shortdata = alloca(af->datalen * 2);
      for (x = 0;x < len; x++) {
         shortdata[x] = AST_MULAW(odata[x]);
      }
      break;
   case AST_FORMAT_ALAW:
      shortdata = alloca(af->datalen * 2);
      for (x = 0; x < len; x++) {
         shortdata[x] = AST_ALAW(odata[x]);
      }
      break;
   default:
      /*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_getformatname(af->subclass.codec));
      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);
   }

   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)) {
      chan->_softhangup |= 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", chan->name);
      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_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 / 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 ((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));
   }

   switch (af->subclass.codec) {
   case AST_FORMAT_SLINEAR:
      break;
   case AST_FORMAT_ULAW:
      for (x = 0; x < len; x++) {
         odata[x] = AST_LIN2MU((unsigned short) shortdata[x]);
      }
      break;
   case AST_FORMAT_ALAW:
      for (x = 0; x < len; x++) {
         odata[x] = AST_LIN2A((unsigned short) shortdata[x]);
      }
      break;
   }

   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 < ARRAY_LEN(modes[dsp->progmode].freqs); x++) {
      if (modes[dsp->progmode].freqs[x]) {
         goertzel_init(&dsp->freqs[x], (float)modes[dsp->progmode].freqs[x], dsp->gsamp_size);
         max = x + 1;
      }
   }
   dsp->freqcount = max;
   dsp->ringtimeout= 0;
}

struct ast_dsp *ast_dsp_new(void)
{
   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;
      /* Initialize digit detector */
      ast_digit_detect_init(&dsp->digit_state, dsp->digitmode & DSP_DIGITMODE_MF);
      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;
}

void ast_dsp_set_features(struct ast_dsp *dsp, int features)
{
   dsp->features = 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, int tonelength, int quietlength)
{
   dsp->busy_tonelength = tonelength;
   dsp->busy_quietlength = quietlength;
   ast_debug(1, "dsp busy pattern set to %d,%d\n", tonelength, quietlength);
}

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] = 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 = 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->digitmode = digitmode;
   return 0;
}

int ast_dsp_set_faxmode(struct ast_dsp *dsp, int faxmode)
{
   if (dsp->faxmode != faxmode) {
      ast_fax_detect_init(dsp);
   }
   dsp->faxmode = faxmode;
   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_flags config_flags = { reload ? CONFIG_FLAG_FILEUNCHANGED : 0 };
   struct ast_config *cfg;

   cfg = ast_config_load2(CONFIG_FILE_NAME, "dsp", config_flags);
   if (cfg == CONFIG_STATUS_FILEMISSING || cfg == CONFIG_STATUS_FILEINVALID) {
      ast_verb(5, "Can't find dsp config file %s. Assuming default silencethreshold of %d.\n", CONFIG_FILE_NAME, DEFAULT_SILENCE_THRESHOLD);
      thresholds[THRESHOLD_SILENCE] = DEFAULT_SILENCE_THRESHOLD;
      return 0;
   }

   if (cfg == CONFIG_STATUS_FILEUNCHANGED) {
      return 0;
   }

   if (cfg) {
      const char *value;

      value = ast_variable_retrieve(cfg, "default", "silencethreshold");
      if (value && sscanf(value, "%30d", &thresholds[THRESHOLD_SILENCE]) != 1) {
         ast_verb(5, "%s: '%s' is not a valid silencethreshold value\n", CONFIG_FILE_NAME, value);
         thresholds[THRESHOLD_SILENCE] = DEFAULT_SILENCE_THRESHOLD;
      } else if (!value) {
         thresholds[THRESHOLD_SILENCE] = DEFAULT_SILENCE_THRESHOLD;
      }

      ast_config_destroy(cfg);
   }
   return 0;
}

int ast_dsp_get_threshold_from_settings(enum threshold which)
{
   return thresholds[which];
}

int ast_dsp_init(void)
{
   return _dsp_init(0);
}

int ast_dsp_reload(void)
{
   return _dsp_init(1);
}