Thu Apr 8 01:20:58 2010

Asterisk developer's documentation


dsp.c

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00001 /*
00002  * Asterisk -- An open source telephony toolkit.
00003  *
00004  * Copyright (C) 1999 - 2005, Digium, Inc.
00005  *
00006  * Mark Spencer <markster@digium.com>
00007  *
00008  * Goertzel routines are borrowed from Steve Underwood's tremendous work on the
00009  * DTMF detector.
00010  *
00011  * See http://www.asterisk.org for more information about
00012  * the Asterisk project. Please do not directly contact
00013  * any of the maintainers of this project for assistance;
00014  * the project provides a web site, mailing lists and IRC
00015  * channels for your use.
00016  *
00017  * This program is free software, distributed under the terms of
00018  * the GNU General Public License Version 2. See the LICENSE file
00019  * at the top of the source tree.
00020  */
00021 
00022 /*! \file
00023  *
00024  * \brief Convenience Signal Processing routines
00025  *
00026  * \author Mark Spencer <markster@digium.com>
00027  * \author Steve Underwood <steveu@coppice.org>
00028  */
00029 
00030 /* Some routines from tone_detect.c by Steven Underwood as published under the zapata library */
00031 /*
00032    tone_detect.c - General telephony tone detection, and specific
00033                detection of DTMF.
00034 
00035    Copyright (C) 2001  Steve Underwood <steveu@coppice.org>
00036 
00037    Despite my general liking of the GPL, I place this code in the
00038    public domain for the benefit of all mankind - even the slimy
00039    ones who might try to proprietize my work and use it to my
00040    detriment.
00041 */
00042 
00043 #include "asterisk.h"
00044 
00045 ASTERISK_FILE_VERSION(__FILE__, "$Revision: 235776 $")
00046 
00047 #include <math.h>
00048 
00049 #include "asterisk/frame.h"
00050 #include "asterisk/channel.h"
00051 #include "asterisk/dsp.h"
00052 #include "asterisk/ulaw.h"
00053 #include "asterisk/alaw.h"
00054 #include "asterisk/utils.h"
00055 #include "asterisk/options.h"
00056 #include "asterisk/config.h"
00057 
00058 /*! Number of goertzels for progress detect */
00059 enum gsamp_size {
00060    GSAMP_SIZE_NA = 183,       /*!< North America - 350, 440, 480, 620, 950, 1400, 1800 Hz */
00061    GSAMP_SIZE_CR = 188,       /*!< Costa Rica, Brazil - Only care about 425 Hz */
00062    GSAMP_SIZE_UK = 160        /*!< UK disconnect goertzel feed - should trigger 400hz */
00063 };
00064 
00065 enum prog_mode {
00066    PROG_MODE_NA = 0,
00067    PROG_MODE_CR,
00068    PROG_MODE_UK
00069 };
00070 
00071 enum freq_index { 
00072    /*! For US modes { */
00073    HZ_350 = 0,
00074    HZ_440,
00075    HZ_480,
00076    HZ_620,
00077    HZ_950,
00078    HZ_1400,
00079    HZ_1800, /*!< } */
00080 
00081    /*! For CR/BR modes */
00082    HZ_425 = 0,
00083 
00084    /*! For UK mode */
00085    HZ_400 = 0
00086 };
00087 
00088 static struct progalias {
00089    char *name;
00090    enum prog_mode mode;
00091 } aliases[] = {
00092    { "us", PROG_MODE_NA },
00093    { "ca", PROG_MODE_NA },
00094    { "cr", PROG_MODE_CR },
00095    { "br", PROG_MODE_CR },
00096    { "uk", PROG_MODE_UK },
00097 };
00098 
00099 static struct progress {
00100    enum gsamp_size size;
00101    int freqs[7];
00102 } modes[] = {
00103    { GSAMP_SIZE_NA, { 350, 440, 480, 620, 950, 1400, 1800 } }, /*!< North America */
00104    { GSAMP_SIZE_CR, { 425 } },                                 /*!< Costa Rica, Brazil */
00105    { GSAMP_SIZE_UK, { 400 } },                                 /*!< UK */
00106 };
00107 
00108 /*!\brief This value is the minimum threshold, calculated by averaging all
00109  * of the samples within a frame, for which a frame is determined to either
00110  * be silence (below the threshold) or noise (above the threshold).  Please
00111  * note that while the default threshold is an even exponent of 2, there is
00112  * no requirement that it be so.  The threshold will accept any value between
00113  * 0 and 32767.
00114  */
00115 #define DEFAULT_THRESHOLD  512
00116 
00117 enum busy_detect {
00118    BUSY_PERCENT = 10,      /*!< The percentage difference between the two last silence periods */
00119    BUSY_PAT_PERCENT = 7,   /*!< The percentage difference between measured and actual pattern */
00120    BUSY_THRESHOLD = 100,   /*!< Max number of ms difference between max and min times in busy */
00121    BUSY_MIN = 75,          /*!< Busy must be at least 80 ms in half-cadence */
00122    BUSY_MAX =3100          /*!< Busy can't be longer than 3100 ms in half-cadence */
00123 };
00124 
00125 /*! Remember last 15 units */
00126 #define DSP_HISTORY     15
00127 
00128 #define TONE_THRESH     10.0  /*!< How much louder the tone should be than channel energy */
00129 #define TONE_MIN_THRESH    1e8   /*!< How much tone there should be at least to attempt */
00130 
00131 /*! All THRESH_XXX values are in GSAMP_SIZE chunks (us = 22ms) */
00132 enum gsamp_thresh {
00133    THRESH_RING = 8,           /*!< Need at least 150ms ring to accept */
00134    THRESH_TALK = 2,           /*!< Talk detection does not work continuously */
00135    THRESH_BUSY = 4,           /*!< Need at least 80ms to accept */
00136    THRESH_CONGESTION = 4,     /*!< Need at least 80ms to accept */
00137    THRESH_HANGUP = 60,        /*!< Need at least 1300ms to accept hangup */
00138    THRESH_RING2ANSWER = 300   /*!< Timeout from start of ring to answer (about 6600 ms) */
00139 };
00140 
00141 #define  MAX_DTMF_DIGITS      128
00142 
00143 /* Basic DTMF specs:
00144  *
00145  * Minimum tone on = 40ms
00146  * Minimum tone off = 50ms
00147  * Maximum digit rate = 10 per second
00148  * Normal twist <= 8dB accepted
00149  * Reverse twist <= 4dB accepted
00150  * S/N >= 15dB will detect OK
00151  * Attenuation <= 26dB will detect OK
00152  * Frequency tolerance +- 1.5% will detect, +-3.5% will reject
00153  */
00154 
00155 #define DTMF_THRESHOLD     8.0e7
00156 #define FAX_THRESHOLD      8.0e7
00157 #define FAX_2ND_HARMONIC   2.0     /* 4dB */
00158 #define DTMF_NORMAL_TWIST  6.3     /* 8dB */
00159 #ifdef   RADIO_RELAX
00160 #define DTMF_REVERSE_TWIST          (relax ? 6.5 : 2.5)     /* 4dB normal */
00161 #else
00162 #define DTMF_REVERSE_TWIST          (relax ? 4.0 : 2.5)     /* 4dB normal */
00163 #endif
00164 #define DTMF_RELATIVE_PEAK_ROW   6.3     /* 8dB */
00165 #define DTMF_RELATIVE_PEAK_COL   6.3     /* 8dB */
00166 #define DTMF_2ND_HARMONIC_ROW       (relax ? 1.7 : 2.5)     /* 4dB normal */
00167 #define DTMF_2ND_HARMONIC_COL 63.1    /* 18dB */
00168 #define DTMF_TO_TOTAL_ENERGY  42.0
00169 
00170 #define BELL_MF_THRESHOLD  1.6e9
00171 #define BELL_MF_TWIST      4.0     /* 6dB */
00172 #define BELL_MF_RELATIVE_PEAK 12.6    /* 11dB */
00173 
00174 #if defined(BUSYDETECT_TONEONLY) && defined(BUSYDETECT_COMPARE_TONE_AND_SILENCE)
00175 #error You cant use BUSYDETECT_TONEONLY together with BUSYDETECT_COMPARE_TONE_AND_SILENCE
00176 #endif
00177 
00178 /* The CNG signal consists of the transmission of 1100 Hz for 1/2 second,
00179  * followed by a 3 second silent (2100 Hz OFF) period.
00180  */
00181 #define FAX_TONE_CNG_FREQ  1100
00182 #define FAX_TONE_CNG_DURATION 500
00183 #define FAX_TONE_CNG_DB    16
00184 
00185 /* This signal may be sent by the Terminating FAX machine anywhere between
00186  * 1.8 to 2.5 seconds AFTER answering the call.  The CED signal consists
00187  * of a 2100 Hz tone that is from 2.6 to 4 seconds in duration.
00188 */
00189 #define FAX_TONE_CED_FREQ  2100
00190 #define FAX_TONE_CED_DURATION 2600
00191 #define FAX_TONE_CED_DB    16
00192 
00193 #define SAMPLE_RATE     8000
00194 
00195 /* How many samples a frame has.  This constant is used when calculating
00196  * Goertzel block size for tone_detect.  It is only important if we want to
00197  * remove (squelch) the tone. In this case it is important to have block
00198  * size not to exceed size of voice frame.  Otherwise by the moment the tone
00199  * is detected it is too late to squelch it from previous frames.
00200  */
00201 #define SAMPLES_IN_FRAME   160
00202 
00203 /* MF goertzel size */
00204 #define MF_GSIZE     120
00205 
00206 /* DTMF goertzel size */
00207 #define DTMF_GSIZE      102
00208 
00209 /* How many successive hits needed to consider begin of a digit */
00210 #define DTMF_HITS_TO_BEGIN 2
00211 /* How many successive misses needed to consider end of a digit */
00212 #define DTMF_MISSES_TO_END 3
00213 
00214 #define CONFIG_FILE_NAME "dsp.conf"
00215 
00216 typedef struct {
00217    int v2;
00218    int v3;
00219    int chunky;
00220    int fac;
00221    int samples;
00222 } goertzel_state_t;
00223 
00224 typedef struct {
00225    int value;
00226    int power;
00227 } goertzel_result_t;
00228 
00229 typedef struct
00230 {
00231    int freq;
00232    int block_size;
00233    int squelch;      /* Remove (squelch) tone */
00234    goertzel_state_t tone;
00235    float energy;     /* Accumulated energy of the current block */
00236    int samples_pending; /* Samples remain to complete the current block */
00237    int mute_samples; /* How many additional samples needs to be muted to suppress already detected tone */
00238 
00239    int hits_required;   /* How many successive blocks with tone we are looking for */
00240    float threshold;  /* Energy of the tone relative to energy from all other signals to consider a hit */
00241 
00242    int hit_count;    /* How many successive blocks we consider tone present */
00243    int last_hit;     /* Indicates if the last processed block was a hit */
00244 
00245 } tone_detect_state_t;
00246 
00247 typedef struct
00248 {
00249    goertzel_state_t row_out[4];
00250    goertzel_state_t col_out[4];
00251    int hits_to_begin;      /* How many successive hits needed to consider begin of a digit */
00252    int misses_to_end;      /* How many successive misses needed to consider end of a digit */
00253    int hits;         /* How many successive hits we have seen already */
00254    int misses;       /* How many successive misses we have seen already */
00255    int lasthit;
00256    int current_hit;
00257    float energy;
00258    int current_sample;
00259    int mute_samples;
00260 } dtmf_detect_state_t;
00261 
00262 typedef struct
00263 {
00264    goertzel_state_t tone_out[6];
00265    int current_hit;
00266    int hits[5];
00267    int current_sample;
00268    int mute_samples;
00269 } mf_detect_state_t;
00270 
00271 typedef struct
00272 {
00273    char digits[MAX_DTMF_DIGITS + 1];
00274    int current_digits;
00275    int detected_digits;
00276    int lost_digits;
00277 
00278    union {
00279       dtmf_detect_state_t dtmf;
00280       mf_detect_state_t mf;
00281    } td;
00282 } digit_detect_state_t;
00283 
00284 static float dtmf_row[] =
00285 {
00286    697.0,  770.0,  852.0,  941.0
00287 };
00288 static float dtmf_col[] =
00289 {
00290    1209.0, 1336.0, 1477.0, 1633.0
00291 };
00292 
00293 static float mf_tones[] =
00294 {
00295    700.0, 900.0, 1100.0, 1300.0, 1500.0, 1700.0
00296 };
00297 
00298 static char dtmf_positions[] = "123A" "456B" "789C" "*0#D";
00299 
00300 static char bell_mf_positions[] = "1247C-358A--69*---0B----#";
00301 
00302 static int thresholds[THRESHOLD_MAX];
00303 
00304 static inline void goertzel_sample(goertzel_state_t *s, short sample)
00305 {
00306    int v1;
00307    
00308    v1 = s->v2;
00309    s->v2 = s->v3;
00310    
00311    s->v3 = (s->fac * s->v2) >> 15;
00312    s->v3 = s->v3 - v1 + (sample >> s->chunky);
00313    if (abs(s->v3) > 32768) {
00314       s->chunky++;
00315       s->v3 = s->v3 >> 1;
00316       s->v2 = s->v2 >> 1;
00317       v1 = v1 >> 1;
00318    }
00319 }
00320 
00321 static inline void goertzel_update(goertzel_state_t *s, short *samps, int count)
00322 {
00323    int i;
00324    
00325    for (i=0;i<count;i++) 
00326       goertzel_sample(s, samps[i]);
00327 }
00328 
00329 
00330 static inline float goertzel_result(goertzel_state_t *s)
00331 {
00332    goertzel_result_t r;
00333    r.value = (s->v3 * s->v3) + (s->v2 * s->v2);
00334    r.value -= ((s->v2 * s->v3) >> 15) * s->fac;
00335    r.power = s->chunky * 2;
00336    return (float)r.value * (float)(1 << r.power);
00337 }
00338 
00339 static inline void goertzel_init(goertzel_state_t *s, float freq, int samples)
00340 {
00341    s->v2 = s->v3 = s->chunky = 0.0;
00342    s->fac = (int)(32768.0 * 2.0 * cos(2.0 * M_PI * freq / SAMPLE_RATE));
00343    s->samples = samples;
00344 }
00345 
00346 static inline void goertzel_reset(goertzel_state_t *s)
00347 {
00348    s->v2 = s->v3 = s->chunky = 0.0;
00349 }
00350 
00351 typedef struct {
00352    int start;
00353    int end;
00354 } fragment_t;
00355 
00356 /* Note on tone suppression (squelching). Individual detectors (DTMF/MF/generic tone)
00357  * report fragmens of the frame in which detected tone resides and which needs
00358  * to be "muted" in order to suppress the tone. To mark fragment for muting,
00359  * detectors call mute_fragment passing fragment_t there. Multiple fragments
00360  * can be marked and ast_dsp_process later will mute all of them.
00361  *
00362  * Note: When tone starts in the middle of a Goertzel block, it won't be properly
00363  * detected in that block, only in the next. If we only mute the next block
00364  * where tone is actually detected, the user will still hear beginning
00365  * of the tone in preceeding block. This is why we usually want to mute some amount
00366  * of samples preceeding and following the block where tone was detected.
00367 */
00368 
00369 struct ast_dsp {
00370    struct ast_frame f;
00371    int threshold;
00372    int totalsilence;
00373    int totalnoise;
00374    int features;
00375    int ringtimeout;
00376    int busymaybe;
00377    int busycount;
00378    int busy_tonelength;
00379    int busy_quietlength;
00380    int historicnoise[DSP_HISTORY];
00381    int historicsilence[DSP_HISTORY];
00382    goertzel_state_t freqs[7];
00383    int freqcount;
00384    int gsamps;
00385    enum gsamp_size gsamp_size;
00386    enum prog_mode progmode;
00387    int tstate;
00388    int tcount;
00389    int digitmode;
00390    int faxmode;
00391    int dtmf_began;
00392    int display_inband_dtmf_warning;
00393    float genergy;
00394    int mute_fragments;
00395    fragment_t mute_data[5];
00396    digit_detect_state_t digit_state;
00397    tone_detect_state_t cng_tone_state;
00398    tone_detect_state_t ced_tone_state;
00399 };
00400 
00401 static void mute_fragment(struct ast_dsp *dsp, fragment_t *fragment)
00402 {
00403    if (dsp->mute_fragments >= ARRAY_LEN(dsp->mute_data)) {
00404       ast_log(LOG_ERROR, "Too many fragments to mute. Ignoring\n");
00405       return;
00406    }
00407 
00408    dsp->mute_data[dsp->mute_fragments++] = *fragment;
00409 }
00410 
00411 static void ast_tone_detect_init(tone_detect_state_t *s, int freq, int duration, int amp)
00412 {
00413    int duration_samples;
00414    float x;
00415    int periods_in_block;
00416 
00417    s->freq = freq;
00418 
00419    /* Desired tone duration in samples */
00420    duration_samples = duration * SAMPLE_RATE / 1000;
00421    /* We want to allow 10% deviation of tone duration */
00422    duration_samples = duration_samples * 9 / 10;
00423 
00424    /* If we want to remove tone, it is important to have block size not
00425       to exceed frame size. Otherwise by the moment tone is detected it is too late
00426       to squelch it from previous frames */
00427    s->block_size = SAMPLES_IN_FRAME;
00428 
00429    periods_in_block = s->block_size * freq / SAMPLE_RATE;
00430 
00431    /* Make sure we will have at least 5 periods at target frequency for analisys.
00432       This may make block larger than expected packet and will make squelching impossible
00433       but at least we will be detecting the tone */
00434    if (periods_in_block < 5)
00435       periods_in_block = 5;
00436 
00437    /* Now calculate final block size. It will contain integer number of periods */
00438    s->block_size = periods_in_block * SAMPLE_RATE / freq;
00439 
00440    /* tone_detect is currently only used to detect fax tones and we
00441       do not need suqlching the fax tones */
00442    s->squelch = 0;
00443 
00444    /* Account for the first and the last block to be incomplete
00445       and thus no tone will be detected in them */
00446    s->hits_required = (duration_samples - (s->block_size - 1)) / s->block_size;
00447 
00448    goertzel_init(&s->tone, freq, s->block_size);
00449 
00450    s->samples_pending = s->block_size;
00451    s->hit_count = 0;
00452    s->last_hit = 0;
00453    s->energy = 0.0;
00454 
00455    /* We want tone energy to be amp decibels above the rest of the signal (the noise).
00456       According to Parseval's theorem the energy computed in time domain equals to energy
00457       computed in frequency domain. So subtracting energy in the frequency domain (Goertzel result)
00458       from the energy in the time domain we will get energy of the remaining signal (without the tone
00459       we are detecting). We will be checking that
00460       10*log(Ew / (Et - Ew)) > amp
00461       Calculate threshold so that we will be actually checking
00462       Ew > Et * threshold
00463    */
00464 
00465    x = pow(10.0, amp / 10.0);
00466    s->threshold = x / (x + 1);
00467 
00468    ast_debug(1, "Setup tone %d Hz, %d ms, block_size=%d, hits_required=%d\n", freq, duration, s->block_size, s->hits_required);
00469 }
00470 
00471 static void ast_fax_detect_init(struct ast_dsp *s)
00472 {
00473    ast_tone_detect_init(&s->cng_tone_state, FAX_TONE_CNG_FREQ, FAX_TONE_CNG_DURATION, FAX_TONE_CNG_DB);
00474    ast_tone_detect_init(&s->ced_tone_state, FAX_TONE_CED_FREQ, FAX_TONE_CED_DURATION, FAX_TONE_CED_DB);
00475 }
00476 
00477 static void ast_dtmf_detect_init (dtmf_detect_state_t *s)
00478 {
00479    int i;
00480 
00481    s->lasthit = 0;
00482    s->current_hit = 0;
00483    for (i = 0;  i < 4;  i++) {
00484       goertzel_init (&s->row_out[i], dtmf_row[i], DTMF_GSIZE);
00485       goertzel_init (&s->col_out[i], dtmf_col[i], DTMF_GSIZE);
00486       s->energy = 0.0;
00487    }
00488    s->current_sample = 0;
00489    s->hits = 0;
00490    s->misses = 0;
00491 
00492    s->hits_to_begin = DTMF_HITS_TO_BEGIN;
00493    s->misses_to_end = DTMF_MISSES_TO_END;
00494 }
00495 
00496 static void ast_mf_detect_init (mf_detect_state_t *s)
00497 {
00498    int i;
00499    s->hits[0] = s->hits[1] = s->hits[2] = s->hits[3] = s->hits[4] = 0;
00500    for (i = 0;  i < 6;  i++) {
00501       goertzel_init (&s->tone_out[i], mf_tones[i], 160);
00502    }
00503    s->current_sample = 0;
00504    s->current_hit = 0;
00505 }
00506 
00507 static void ast_digit_detect_init(digit_detect_state_t *s, int mf)
00508 {
00509    s->current_digits = 0;
00510    s->detected_digits = 0;
00511    s->lost_digits = 0;
00512    s->digits[0] = '\0';
00513 
00514    if (mf)
00515       ast_mf_detect_init(&s->td.mf);
00516    else
00517       ast_dtmf_detect_init(&s->td.dtmf);
00518 }
00519 
00520 static int tone_detect(struct ast_dsp *dsp, tone_detect_state_t *s, int16_t *amp, int samples)
00521 {
00522    float tone_energy;
00523    int i;
00524    int hit = 0;
00525    int limit;
00526    int res = 0;
00527    int16_t *ptr;
00528    int start, end;
00529    fragment_t mute = {0, 0};
00530 
00531    if (s->squelch && s->mute_samples > 0) {
00532       mute.end = (s->mute_samples < samples) ? s->mute_samples : samples;
00533       s->mute_samples -= mute.end;
00534    }
00535 
00536    for (start = 0;  start < samples;  start = end) {
00537       /* Process in blocks. */
00538       limit = samples - start;
00539       if (limit > s->samples_pending)
00540          limit = s->samples_pending;
00541       end = start + limit;
00542 
00543       for (i = limit, ptr = amp ; i > 0; i--, ptr++) {
00544          /* signed 32 bit int should be enough to suqare any possible signed 16 bit value */
00545          s->energy += (int32_t) *ptr * (int32_t) *ptr;
00546 
00547          goertzel_sample(&s->tone, *ptr);
00548       }
00549 
00550       s->samples_pending -= limit;
00551 
00552       if (s->samples_pending) {
00553          /* Finished incomplete (last) block */
00554          break;
00555       }
00556 
00557       tone_energy = goertzel_result(&s->tone);
00558 
00559       /* Scale to make comparable */
00560       tone_energy *= 2.0;
00561       s->energy *= s->block_size;
00562 
00563       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));
00564       hit = 0;
00565       if (tone_energy > s->energy * s->threshold) {
00566          ast_debug(10, "Hit! count=%d\n", s->hit_count);
00567          hit = 1;
00568       }
00569 
00570       if (s->hit_count)
00571          s->hit_count++;
00572 
00573       if (hit == s->last_hit) {
00574          if (!hit) {
00575             /* Two successive misses. Tone ended */
00576             s->hit_count = 0;
00577          } else if (!s->hit_count) {
00578             s->hit_count++;
00579          }
00580 
00581       }
00582 
00583       if (s->hit_count == s->hits_required) {
00584          ast_debug(1, "%d Hz done detected\n", s->freq);
00585          res = 1;
00586       }
00587 
00588       s->last_hit = hit;
00589 
00590       /* If we had a hit in this block, include it into mute fragment */
00591       if (s->squelch && hit) {
00592          if (mute.end < start - s->block_size) {
00593             /* There is a gap between fragments */
00594             mute_fragment(dsp, &mute);
00595             mute.start = (start > s->block_size) ? (start - s->block_size) : 0;
00596          }
00597          mute.end = end + s->block_size;
00598       }
00599 
00600       /* Reinitialise the detector for the next block */
00601       /* Reset for the next block */
00602       goertzel_reset(&s->tone);
00603 
00604       /* Advance to the next block */
00605       s->energy = 0.0;
00606       s->samples_pending = s->block_size;
00607 
00608       amp += limit;
00609    }
00610 
00611    if (s->squelch && mute.end) {
00612       if (mute.end > samples) {
00613          s->mute_samples = mute.end - samples;
00614          mute.end = samples;
00615       }
00616       mute_fragment(dsp, &mute);
00617    }
00618 
00619    return res;
00620 }
00621 
00622 static void store_digit(digit_detect_state_t *s, char digit)
00623 {
00624    s->detected_digits++;
00625    if (s->current_digits < MAX_DTMF_DIGITS) {
00626       s->digits[s->current_digits++] = digit;
00627       s->digits[s->current_digits] = '\0';
00628    } else {
00629       ast_log(LOG_WARNING, "Digit lost due to full buffer\n");
00630       s->lost_digits++;
00631    }
00632 }
00633 
00634 static int dtmf_detect(struct ast_dsp *dsp, digit_detect_state_t *s, int16_t amp[], int samples, int squelch, int relax)
00635 {
00636    float row_energy[4];
00637    float col_energy[4];
00638    float famp;
00639    int i;
00640    int j;
00641    int sample;
00642    int best_row;
00643    int best_col;
00644    int hit;
00645    int limit;
00646    fragment_t mute = {0, 0};
00647 
00648    if (squelch && s->td.dtmf.mute_samples > 0) {
00649       mute.end = (s->td.dtmf.mute_samples < samples) ? s->td.dtmf.mute_samples : samples;
00650       s->td.dtmf.mute_samples -= mute.end;
00651    }
00652 
00653    hit = 0;
00654    for (sample = 0;  sample < samples;  sample = limit) {
00655       /* DTMF_GSIZE is optimised to meet the DTMF specs. */
00656       if ((samples - sample) >= (DTMF_GSIZE - s->td.dtmf.current_sample))
00657          limit = sample + (DTMF_GSIZE - s->td.dtmf.current_sample);
00658       else
00659          limit = samples;
00660       /* The following unrolled loop takes only 35% (rough estimate) of the 
00661          time of a rolled loop on the machine on which it was developed */
00662       for (j = sample; j < limit; j++) {
00663          famp = amp[j];
00664          s->td.dtmf.energy += famp*famp;
00665          /* With GCC 2.95, the following unrolled code seems to take about 35%
00666             (rough estimate) as long as a neat little 0-3 loop */
00667          goertzel_sample(s->td.dtmf.row_out, amp[j]);
00668          goertzel_sample(s->td.dtmf.col_out, amp[j]);
00669          goertzel_sample(s->td.dtmf.row_out + 1, amp[j]);
00670          goertzel_sample(s->td.dtmf.col_out + 1, amp[j]);
00671          goertzel_sample(s->td.dtmf.row_out + 2, amp[j]);
00672          goertzel_sample(s->td.dtmf.col_out + 2, amp[j]);
00673          goertzel_sample(s->td.dtmf.row_out + 3, amp[j]);
00674          goertzel_sample(s->td.dtmf.col_out + 3, amp[j]);
00675       }
00676       s->td.dtmf.current_sample += (limit - sample);
00677       if (s->td.dtmf.current_sample < DTMF_GSIZE) {
00678          continue;
00679       }
00680       /* We are at the end of a DTMF detection block */
00681       /* Find the peak row and the peak column */
00682       row_energy[0] = goertzel_result (&s->td.dtmf.row_out[0]);
00683       col_energy[0] = goertzel_result (&s->td.dtmf.col_out[0]);
00684 
00685       for (best_row = best_col = 0, i = 1;  i < 4;  i++) {
00686          row_energy[i] = goertzel_result (&s->td.dtmf.row_out[i]);
00687          if (row_energy[i] > row_energy[best_row])
00688             best_row = i;
00689          col_energy[i] = goertzel_result (&s->td.dtmf.col_out[i]);
00690          if (col_energy[i] > col_energy[best_col])
00691             best_col = i;
00692       }
00693       hit = 0;
00694       /* Basic signal level test and the twist test */
00695       if (row_energy[best_row] >= DTMF_THRESHOLD && 
00696           col_energy[best_col] >= DTMF_THRESHOLD &&
00697           col_energy[best_col] < row_energy[best_row]*DTMF_REVERSE_TWIST &&
00698           col_energy[best_col]*DTMF_NORMAL_TWIST > row_energy[best_row]) {
00699          /* Relative peak test */
00700          for (i = 0;  i < 4;  i++) {
00701             if ((i != best_col &&
00702                 col_energy[i]*DTMF_RELATIVE_PEAK_COL > col_energy[best_col]) ||
00703                 (i != best_row 
00704                  && row_energy[i]*DTMF_RELATIVE_PEAK_ROW > row_energy[best_row])) {
00705                break;
00706             }
00707          }
00708          /* ... and fraction of total energy test */
00709          if (i >= 4 &&
00710              (row_energy[best_row] + col_energy[best_col]) > DTMF_TO_TOTAL_ENERGY*s->td.dtmf.energy) {
00711             /* Got a hit */
00712             hit = dtmf_positions[(best_row << 2) + best_col];
00713          }
00714       } 
00715 
00716       if (s->td.dtmf.current_hit) {
00717          /* We are in the middle of a digit already */
00718          if (hit != s->td.dtmf.current_hit) {
00719             s->td.dtmf.misses++;
00720             if (s->td.dtmf.misses == s->td.dtmf.misses_to_end) {
00721                /* There were enough misses to consider digit ended */
00722                s->td.dtmf.current_hit = 0;
00723             }
00724          } else {
00725             s->td.dtmf.misses = 0;
00726          }
00727       }
00728 
00729       /* Look for a start of a new digit no matter if we are already in the middle of some
00730          digit or not. This is because hits_to_begin may be smaller than misses_to_end
00731          and we may find begin of new digit before we consider last one ended. */
00732       if (hit) {
00733          if (hit == s->td.dtmf.lasthit) {
00734             s->td.dtmf.hits++;
00735          } else {
00736             s->td.dtmf.hits = 1;
00737          }
00738 
00739          if (s->td.dtmf.hits == s->td.dtmf.hits_to_begin && hit != s->td.dtmf.current_hit) {
00740             store_digit(s, hit);
00741             s->td.dtmf.current_hit = hit;
00742             s->td.dtmf.misses = 0;
00743          }
00744       } else {
00745          s->td.dtmf.hits = 0;
00746       }
00747 
00748       s->td.dtmf.lasthit = hit;
00749 
00750       /* If we had a hit in this block, include it into mute fragment */
00751       if (squelch && hit) {
00752          if (mute.end < sample - DTMF_GSIZE) {
00753             /* There is a gap between fragments */
00754             mute_fragment(dsp, &mute);
00755             mute.start = (sample > DTMF_GSIZE) ? (sample - DTMF_GSIZE) : 0;
00756          }
00757          mute.end = limit + DTMF_GSIZE;
00758       }
00759 
00760       /* Reinitialise the detector for the next block */
00761       for (i = 0;  i < 4;  i++) {
00762          goertzel_reset(&s->td.dtmf.row_out[i]);
00763          goertzel_reset(&s->td.dtmf.col_out[i]);
00764       }
00765       s->td.dtmf.energy = 0.0;
00766       s->td.dtmf.current_sample = 0;
00767    }
00768 
00769    if (squelch && mute.end) {
00770       if (mute.end > samples) {
00771          s->td.dtmf.mute_samples = mute.end - samples;
00772          mute.end = samples;
00773       }
00774       mute_fragment(dsp, &mute);
00775    }
00776 
00777    return (s->td.dtmf.current_hit); /* return the debounced hit */
00778 }
00779 
00780 static int mf_detect(struct ast_dsp *dsp, digit_detect_state_t *s, int16_t amp[],
00781                  int samples, int squelch, int relax)
00782 {
00783    float energy[6];
00784    int best;
00785    int second_best;
00786    float famp;
00787    int i;
00788    int j;
00789    int sample;
00790    int hit;
00791    int limit;
00792    fragment_t mute = {0, 0};
00793 
00794    if (squelch && s->td.mf.mute_samples > 0) {
00795       mute.end = (s->td.mf.mute_samples < samples) ? s->td.mf.mute_samples : samples;
00796       s->td.mf.mute_samples -= mute.end;
00797    }
00798 
00799    hit = 0;
00800    for (sample = 0;  sample < samples;  sample = limit) {
00801       /* 80 is optimised to meet the MF specs. */
00802       /* XXX So then why is MF_GSIZE defined as 120? */
00803       if ((samples - sample) >= (MF_GSIZE - s->td.mf.current_sample))
00804          limit = sample + (MF_GSIZE - s->td.mf.current_sample);
00805       else
00806          limit = samples;
00807       /* The following unrolled loop takes only 35% (rough estimate) of the 
00808          time of a rolled loop on the machine on which it was developed */
00809       for (j = sample;  j < limit;  j++) {
00810          famp = amp[j];
00811          /* With GCC 2.95, the following unrolled code seems to take about 35%
00812             (rough estimate) as long as a neat little 0-3 loop */
00813          goertzel_sample(s->td.mf.tone_out, amp[j]);
00814          goertzel_sample(s->td.mf.tone_out + 1, amp[j]);
00815          goertzel_sample(s->td.mf.tone_out + 2, amp[j]);
00816          goertzel_sample(s->td.mf.tone_out + 3, amp[j]);
00817          goertzel_sample(s->td.mf.tone_out + 4, amp[j]);
00818          goertzel_sample(s->td.mf.tone_out + 5, amp[j]);
00819       }
00820       s->td.mf.current_sample += (limit - sample);
00821       if (s->td.mf.current_sample < MF_GSIZE) {
00822          continue;
00823       }
00824       /* We're at the end of an MF detection block.  */
00825       /* Find the two highest energies. The spec says to look for
00826          two tones and two tones only. Taking this literally -ie
00827          only two tones pass the minimum threshold - doesn't work
00828          well. The sinc function mess, due to rectangular windowing
00829          ensure that! Find the two highest energies and ensure they
00830          are considerably stronger than any of the others. */
00831       energy[0] = goertzel_result(&s->td.mf.tone_out[0]);
00832       energy[1] = goertzel_result(&s->td.mf.tone_out[1]);
00833       if (energy[0] > energy[1]) {
00834          best = 0;
00835          second_best = 1;
00836       } else {
00837          best = 1;
00838          second_best = 0;
00839       }
00840       /*endif*/
00841       for (i=2;i<6;i++) {
00842          energy[i] = goertzel_result(&s->td.mf.tone_out[i]);
00843          if (energy[i] >= energy[best]) {
00844             second_best = best;
00845             best = i;
00846          } else if (energy[i] >= energy[second_best]) {
00847             second_best = i;
00848          }
00849       }
00850       /* Basic signal level and twist tests */
00851       hit = 0;
00852       if (energy[best] >= BELL_MF_THRESHOLD && energy[second_best] >= BELL_MF_THRESHOLD
00853                && energy[best] < energy[second_best]*BELL_MF_TWIST
00854                && energy[best]*BELL_MF_TWIST > energy[second_best]) {
00855          /* Relative peak test */
00856          hit = -1;
00857          for (i=0;i<6;i++) {
00858             if (i != best && i != second_best) {
00859                if (energy[i]*BELL_MF_RELATIVE_PEAK >= energy[second_best]) {
00860                   /* The best two are not clearly the best */
00861                   hit = 0;
00862                   break;
00863                }
00864             }
00865          }
00866       }
00867       if (hit) {
00868          /* Get the values into ascending order */
00869          if (second_best < best) {
00870             i = best;
00871             best = second_best;
00872             second_best = i;
00873          }
00874          best = best*5 + second_best - 1;
00875          hit = bell_mf_positions[best];
00876          /* Look for two successive similar results */
00877          /* The logic in the next test is:
00878             For KP we need 4 successive identical clean detects, with
00879             two blocks of something different preceeding it. For anything
00880             else we need two successive identical clean detects, with
00881             two blocks of something different preceeding it. */
00882          if (hit == s->td.mf.hits[4] && hit == s->td.mf.hits[3] &&
00883             ((hit != '*' && hit != s->td.mf.hits[2] && hit != s->td.mf.hits[1])||
00884              (hit == '*' && hit == s->td.mf.hits[2] && hit != s->td.mf.hits[1] && 
00885              hit != s->td.mf.hits[0]))) {
00886             store_digit(s, hit);
00887          }
00888       }
00889 
00890 
00891       if (hit != s->td.mf.hits[4] && hit != s->td.mf.hits[3]) {
00892          /* Two successive block without a hit terminate current digit */
00893          s->td.mf.current_hit = 0;
00894       }
00895 
00896       s->td.mf.hits[0] = s->td.mf.hits[1];
00897       s->td.mf.hits[1] = s->td.mf.hits[2];
00898       s->td.mf.hits[2] = s->td.mf.hits[3];
00899       s->td.mf.hits[3] = s->td.mf.hits[4];
00900       s->td.mf.hits[4] = hit;
00901 
00902       /* If we had a hit in this block, include it into mute fragment */
00903       if (squelch && hit) {
00904          if (mute.end < sample - MF_GSIZE) {
00905             /* There is a gap between fragments */
00906             mute_fragment(dsp, &mute);
00907             mute.start = (sample > MF_GSIZE) ? (sample - MF_GSIZE) : 0;
00908          }
00909          mute.end = limit + DTMF_GSIZE;
00910       }
00911 
00912       /* Reinitialise the detector for the next block */
00913       for (i = 0;  i < 6;  i++)
00914          goertzel_reset(&s->td.mf.tone_out[i]);
00915       s->td.mf.current_sample = 0;
00916    }
00917 
00918    if (squelch && mute.end) {
00919       if (mute.end > samples) {
00920          s->td.mf.mute_samples = mute.end - samples;
00921          mute.end = samples;
00922       }
00923       mute_fragment(dsp, &mute);
00924    }
00925 
00926    return (s->td.mf.current_hit); /* return the debounced hit */
00927 }
00928 
00929 static inline int pair_there(float p1, float p2, float i1, float i2, float e)
00930 {
00931    /* See if p1 and p2 are there, relative to i1 and i2 and total energy */
00932    /* Make sure absolute levels are high enough */
00933    if ((p1 < TONE_MIN_THRESH) || (p2 < TONE_MIN_THRESH))
00934       return 0;
00935    /* Amplify ignored stuff */
00936    i2 *= TONE_THRESH;
00937    i1 *= TONE_THRESH;
00938    e *= TONE_THRESH;
00939    /* Check first tone */
00940    if ((p1 < i1) || (p1 < i2) || (p1 < e))
00941       return 0;
00942    /* And second */
00943    if ((p2 < i1) || (p2 < i2) || (p2 < e))
00944       return 0;
00945    /* Guess it's there... */
00946    return 1;
00947 }
00948 
00949 static int __ast_dsp_call_progress(struct ast_dsp *dsp, short *s, int len)
00950 {
00951    int x;
00952    int y;
00953    int pass;
00954    int newstate = DSP_TONE_STATE_SILENCE;
00955    int res = 0;
00956    while (len) {
00957       /* Take the lesser of the number of samples we need and what we have */
00958       pass = len;
00959       if (pass > dsp->gsamp_size - dsp->gsamps) 
00960          pass = dsp->gsamp_size - dsp->gsamps;
00961       for (x=0;x<pass;x++) {
00962          for (y=0;y<dsp->freqcount;y++) 
00963             goertzel_sample(&dsp->freqs[y], s[x]);
00964          dsp->genergy += s[x] * s[x];
00965       }
00966       s += pass;
00967       dsp->gsamps += pass;
00968       len -= pass;
00969       if (dsp->gsamps == dsp->gsamp_size) {
00970          float hz[7];
00971          for (y=0;y<7;y++)
00972             hz[y] = goertzel_result(&dsp->freqs[y]);
00973          switch (dsp->progmode) {
00974          case PROG_MODE_NA:
00975             if (pair_there(hz[HZ_480], hz[HZ_620], hz[HZ_350], hz[HZ_440], dsp->genergy)) {
00976                newstate = DSP_TONE_STATE_BUSY;
00977             } else if (pair_there(hz[HZ_440], hz[HZ_480], hz[HZ_350], hz[HZ_620], dsp->genergy)) {
00978                newstate = DSP_TONE_STATE_RINGING;
00979             } else if (pair_there(hz[HZ_350], hz[HZ_440], hz[HZ_480], hz[HZ_620], dsp->genergy)) {
00980                newstate = DSP_TONE_STATE_DIALTONE;
00981             } else if (hz[HZ_950] > TONE_MIN_THRESH * TONE_THRESH) {
00982                newstate = DSP_TONE_STATE_SPECIAL1;
00983             } else if (hz[HZ_1400] > TONE_MIN_THRESH * TONE_THRESH) {
00984                if (dsp->tstate == DSP_TONE_STATE_SPECIAL1)
00985                   newstate = DSP_TONE_STATE_SPECIAL2;
00986             } else if (hz[HZ_1800] > TONE_MIN_THRESH * TONE_THRESH) {
00987                if (dsp->tstate == DSP_TONE_STATE_SPECIAL2)
00988                   newstate = DSP_TONE_STATE_SPECIAL3;
00989             } else if (dsp->genergy > TONE_MIN_THRESH * TONE_THRESH) {
00990                newstate = DSP_TONE_STATE_TALKING;
00991             } else
00992                newstate = DSP_TONE_STATE_SILENCE;
00993             break;
00994          case PROG_MODE_CR:
00995             if (hz[HZ_425] > TONE_MIN_THRESH * TONE_THRESH) {
00996                newstate = DSP_TONE_STATE_RINGING;
00997             } else if (dsp->genergy > TONE_MIN_THRESH * TONE_THRESH) {
00998                newstate = DSP_TONE_STATE_TALKING;
00999             } else
01000                newstate = DSP_TONE_STATE_SILENCE;
01001             break;
01002          case PROG_MODE_UK:
01003             if (hz[HZ_400] > TONE_MIN_THRESH * TONE_THRESH) {
01004                newstate = DSP_TONE_STATE_HUNGUP;
01005             }
01006             break;
01007          default:
01008             ast_log(LOG_WARNING, "Can't process in unknown prog mode '%d'\n", dsp->progmode);
01009          }
01010          if (newstate == dsp->tstate) {
01011             dsp->tcount++;
01012             if (dsp->ringtimeout)
01013                dsp->ringtimeout++;
01014             switch (dsp->tstate) {
01015                case DSP_TONE_STATE_RINGING:
01016                   if ((dsp->features & DSP_PROGRESS_RINGING) &&
01017                       (dsp->tcount==THRESH_RING)) {
01018                      res = AST_CONTROL_RINGING;
01019                      dsp->ringtimeout= 1;
01020                   }
01021                   break;
01022                case DSP_TONE_STATE_BUSY:
01023                   if ((dsp->features & DSP_PROGRESS_BUSY) &&
01024                       (dsp->tcount==THRESH_BUSY)) {
01025                      res = AST_CONTROL_BUSY;
01026                      dsp->features &= ~DSP_FEATURE_CALL_PROGRESS;
01027                   }
01028                   break;
01029                case DSP_TONE_STATE_TALKING:
01030                   if ((dsp->features & DSP_PROGRESS_TALK) &&
01031                       (dsp->tcount==THRESH_TALK)) {
01032                      res = AST_CONTROL_ANSWER;
01033                      dsp->features &= ~DSP_FEATURE_CALL_PROGRESS;
01034                   }
01035                   break;
01036                case DSP_TONE_STATE_SPECIAL3:
01037                   if ((dsp->features & DSP_PROGRESS_CONGESTION) &&
01038                       (dsp->tcount==THRESH_CONGESTION)) {
01039                      res = AST_CONTROL_CONGESTION;
01040                      dsp->features &= ~DSP_FEATURE_CALL_PROGRESS;
01041                   }
01042                   break;
01043                case DSP_TONE_STATE_HUNGUP:
01044                   if ((dsp->features & DSP_FEATURE_CALL_PROGRESS) &&
01045                       (dsp->tcount==THRESH_HANGUP)) {
01046                      res = AST_CONTROL_HANGUP;
01047                      dsp->features &= ~DSP_FEATURE_CALL_PROGRESS;
01048                   }
01049                   break;
01050             }
01051             if (dsp->ringtimeout==THRESH_RING2ANSWER) {
01052                ast_debug(1, "Consider call as answered because of timeout after last ring\n");
01053                res = AST_CONTROL_ANSWER;
01054                dsp->features &= ~DSP_FEATURE_CALL_PROGRESS;
01055             }
01056          } else {
01057             ast_debug(5, "Stop state %d with duration %d\n", dsp->tstate, dsp->tcount);
01058             ast_debug(5, "Start state %d\n", newstate);
01059             dsp->tstate = newstate;
01060             dsp->tcount = 1;
01061          }
01062          
01063          /* Reset goertzel */                
01064          for (x=0;x<7;x++)
01065             dsp->freqs[x].v2 = dsp->freqs[x].v3 = 0.0;
01066          dsp->gsamps = 0;
01067          dsp->genergy = 0.0;
01068       }
01069    }
01070 
01071    return res;
01072 }
01073 
01074 int ast_dsp_call_progress(struct ast_dsp *dsp, struct ast_frame *inf)
01075 {
01076    if (inf->frametype != AST_FRAME_VOICE) {
01077       ast_log(LOG_WARNING, "Can't check call progress of non-voice frames\n");
01078       return 0;
01079    }
01080    if (inf->subclass != AST_FORMAT_SLINEAR) {
01081       ast_log(LOG_WARNING, "Can only check call progress in signed-linear frames\n");
01082       return 0;
01083    }
01084    return __ast_dsp_call_progress(dsp, inf->data.ptr, inf->datalen / 2);
01085 }
01086 
01087 static int __ast_dsp_silence_noise(struct ast_dsp *dsp, short *s, int len, int *totalsilence, int *totalnoise)
01088 {
01089    int accum;
01090    int x;
01091    int res = 0;
01092 
01093    if (!len)
01094       return 0;
01095    accum = 0;
01096    for (x=0;x<len; x++) 
01097       accum += abs(s[x]);
01098    accum /= len;
01099    if (accum < dsp->threshold) {
01100       /* Silent */
01101       dsp->totalsilence += len/8;
01102       if (dsp->totalnoise) {
01103          /* Move and save history */
01104          memmove(dsp->historicnoise + DSP_HISTORY - dsp->busycount, dsp->historicnoise + DSP_HISTORY - dsp->busycount +1, dsp->busycount*sizeof(dsp->historicnoise[0]));
01105          dsp->historicnoise[DSP_HISTORY - 1] = dsp->totalnoise;
01106 /* we don't want to check for busydetect that frequently */
01107 #if 0
01108          dsp->busymaybe = 1;
01109 #endif
01110       }
01111       dsp->totalnoise = 0;
01112       res = 1;
01113    } else {
01114       /* Not silent */
01115       dsp->totalnoise += len/8;
01116       if (dsp->totalsilence) {
01117          int silence1 = dsp->historicsilence[DSP_HISTORY - 1];
01118          int silence2 = dsp->historicsilence[DSP_HISTORY - 2];
01119          /* Move and save history */
01120          memmove(dsp->historicsilence + DSP_HISTORY - dsp->busycount, dsp->historicsilence + DSP_HISTORY - dsp->busycount + 1, dsp->busycount*sizeof(dsp->historicsilence[0]));
01121          dsp->historicsilence[DSP_HISTORY - 1] = dsp->totalsilence;
01122          /* check if the previous sample differs only by BUSY_PERCENT from the one before it */
01123          if (silence1 < silence2) {
01124             if (silence1 + silence1*BUSY_PERCENT/100 >= silence2)
01125                dsp->busymaybe = 1;
01126             else 
01127                dsp->busymaybe = 0;
01128          } else {
01129             if (silence1 - silence1*BUSY_PERCENT/100 <= silence2)
01130                dsp->busymaybe = 1;
01131             else 
01132                dsp->busymaybe = 0;
01133          }
01134       }
01135       dsp->totalsilence = 0;
01136    }
01137    if (totalsilence)
01138       *totalsilence = dsp->totalsilence;
01139    if (totalnoise)
01140       *totalnoise = dsp->totalnoise;
01141    return res;
01142 }
01143 
01144 int ast_dsp_busydetect(struct ast_dsp *dsp)
01145 {
01146    int res = 0, x;
01147 #ifndef BUSYDETECT_TONEONLY
01148    int avgsilence = 0, hitsilence = 0;
01149 #endif
01150    int avgtone = 0, hittone = 0;
01151    if (!dsp->busymaybe)
01152       return res;
01153    for (x=DSP_HISTORY - dsp->busycount;x<DSP_HISTORY;x++) {
01154 #ifndef BUSYDETECT_TONEONLY
01155       avgsilence += dsp->historicsilence[x];
01156 #endif
01157       avgtone += dsp->historicnoise[x];
01158    }
01159 #ifndef BUSYDETECT_TONEONLY
01160    avgsilence /= dsp->busycount;
01161 #endif
01162    avgtone /= dsp->busycount;
01163    for (x=DSP_HISTORY - dsp->busycount;x<DSP_HISTORY;x++) {
01164 #ifndef BUSYDETECT_TONEONLY
01165       if (avgsilence > dsp->historicsilence[x]) {
01166          if (avgsilence - (avgsilence*BUSY_PERCENT/100) <= dsp->historicsilence[x])
01167             hitsilence++;
01168       } else {
01169          if (avgsilence + (avgsilence*BUSY_PERCENT/100) >= dsp->historicsilence[x])
01170             hitsilence++;
01171       }
01172 #endif
01173       if (avgtone > dsp->historicnoise[x]) {
01174          if (avgtone - (avgtone*BUSY_PERCENT/100) <= dsp->historicnoise[x])
01175             hittone++;
01176       } else {
01177          if (avgtone + (avgtone*BUSY_PERCENT/100) >= dsp->historicnoise[x])
01178             hittone++;
01179       }
01180    }
01181 #ifndef BUSYDETECT_TONEONLY
01182    if ((hittone >= dsp->busycount - 1) && (hitsilence >= dsp->busycount - 1) && 
01183        (avgtone >= BUSY_MIN && avgtone <= BUSY_MAX) && 
01184        (avgsilence >= BUSY_MIN && avgsilence <= BUSY_MAX)) {
01185 #else
01186    if ((hittone >= dsp->busycount - 1) && (avgtone >= BUSY_MIN && avgtone <= BUSY_MAX)) {
01187 #endif
01188 #ifdef BUSYDETECT_COMPARE_TONE_AND_SILENCE
01189       if (avgtone > avgsilence) {
01190          if (avgtone - avgtone*BUSY_PERCENT/100 <= avgsilence)
01191             res = 1;
01192       } else {
01193          if (avgtone + avgtone*BUSY_PERCENT/100 >= avgsilence)
01194             res = 1;
01195       }
01196 #else
01197       res = 1;
01198 #endif
01199    }
01200    /* If we know the expected busy tone length, check we are in the range */
01201    if (res && (dsp->busy_tonelength > 0)) {
01202       if (abs(avgtone - dsp->busy_tonelength) > (dsp->busy_tonelength*BUSY_PAT_PERCENT/100)) {
01203 #ifdef BUSYDETECT_DEBUG
01204          ast_debug(5, "busy detector: avgtone of %d not close enough to desired %d\n",
01205             avgtone, dsp->busy_tonelength);
01206 #endif
01207          res = 0;
01208       }
01209    }
01210 #ifndef BUSYDETECT_TONEONLY
01211    /* If we know the expected busy tone silent-period length, check we are in the range */
01212    if (res && (dsp->busy_quietlength > 0)) {
01213       if (abs(avgsilence - dsp->busy_quietlength) > (dsp->busy_quietlength*BUSY_PAT_PERCENT/100)) {
01214 #ifdef BUSYDETECT_DEBUG
01215       ast_debug(5, "busy detector: avgsilence of %d not close enough to desired %d\n",
01216          avgsilence, dsp->busy_quietlength);
01217 #endif
01218          res = 0;
01219       }
01220    }
01221 #endif
01222 #if !defined(BUSYDETECT_TONEONLY) && defined(BUSYDETECT_DEBUG)
01223    if (res) {
01224       ast_debug(5, "ast_dsp_busydetect detected busy, avgtone: %d, avgsilence %d\n", avgtone, avgsilence);
01225    } else {
01226       ast_debug(5, "busy detector: FAILED with avgtone: %d, avgsilence %d\n", avgtone, avgsilence);
01227    }
01228 #endif
01229    return res;
01230 }
01231 
01232 int ast_dsp_silence(struct ast_dsp *dsp, struct ast_frame *f, int *totalsilence)
01233 {
01234    short *s;
01235    int len;
01236    
01237    if (f->frametype != AST_FRAME_VOICE) {
01238       ast_log(LOG_WARNING, "Can't calculate silence on a non-voice frame\n");
01239       return 0;
01240    }
01241    if (f->subclass != AST_FORMAT_SLINEAR) {
01242       ast_log(LOG_WARNING, "Can only calculate silence on signed-linear frames :(\n");
01243       return 0;
01244    }
01245    s = f->data.ptr;
01246    len = f->datalen/2;
01247    return __ast_dsp_silence_noise(dsp, s, len, totalsilence, NULL);
01248 }
01249 
01250 int ast_dsp_noise(struct ast_dsp *dsp, struct ast_frame *f, int *totalnoise)
01251 {
01252        short *s;
01253        int len;
01254 
01255        if (f->frametype != AST_FRAME_VOICE) {
01256                ast_log(LOG_WARNING, "Can't calculate noise on a non-voice frame\n");
01257                return 0;
01258        }
01259        if (f->subclass != AST_FORMAT_SLINEAR) {
01260                ast_log(LOG_WARNING, "Can only calculate noise on signed-linear frames :(\n");
01261                return 0;
01262        }
01263        s = f->data.ptr;
01264        len = f->datalen/2;
01265        return __ast_dsp_silence_noise(dsp, s, len, NULL, totalnoise);
01266 }
01267 
01268 
01269 struct ast_frame *ast_dsp_process(struct ast_channel *chan, struct ast_dsp *dsp, struct ast_frame *af)
01270 {
01271    int silence;
01272    int res;
01273    int digit = 0, fax_digit = 0;
01274    int x;
01275    short *shortdata;
01276    unsigned char *odata;
01277    int len;
01278    struct ast_frame *outf = NULL;
01279 
01280    if (!af)
01281       return NULL;
01282    if (af->frametype != AST_FRAME_VOICE)
01283       return af;
01284 
01285    odata = af->data.ptr;
01286    len = af->datalen;
01287    /* Make sure we have short data */
01288    switch (af->subclass) {
01289    case AST_FORMAT_SLINEAR:
01290       shortdata = af->data.ptr;
01291       len = af->datalen / 2;
01292       break;
01293    case AST_FORMAT_ULAW:
01294       shortdata = alloca(af->datalen * 2);
01295       for (x = 0;x < len; x++) 
01296          shortdata[x] = AST_MULAW(odata[x]);
01297       break;
01298    case AST_FORMAT_ALAW:
01299       shortdata = alloca(af->datalen * 2);
01300       for (x = 0; x < len; x++) 
01301          shortdata[x] = AST_ALAW(odata[x]);
01302       break;
01303    default:
01304       /*Display warning only once. Otherwise you would get hundreds of warnings every second */
01305       if (dsp->display_inband_dtmf_warning)
01306          ast_log(LOG_WARNING, "Inband DTMF is not supported on codec %s. Use RFC2833\n", ast_getformatname(af->subclass));
01307       dsp->display_inband_dtmf_warning = 0;
01308       return af;
01309    }
01310 
01311    /* Initially we do not want to mute anything */
01312    dsp->mute_fragments = 0;
01313 
01314    /* Need to run the silence detection stuff for silence suppression and busy detection */
01315    if ((dsp->features & DSP_FEATURE_SILENCE_SUPPRESS) || (dsp->features & DSP_FEATURE_BUSY_DETECT)) {
01316       res = __ast_dsp_silence_noise(dsp, shortdata, len, &silence, NULL);
01317    }
01318 
01319    if ((dsp->features & DSP_FEATURE_SILENCE_SUPPRESS) && silence) {
01320       memset(&dsp->f, 0, sizeof(dsp->f));
01321       dsp->f.frametype = AST_FRAME_NULL;
01322       ast_frfree(af);
01323       return ast_frisolate(&dsp->f);
01324    }
01325    if ((dsp->features & DSP_FEATURE_BUSY_DETECT) && ast_dsp_busydetect(dsp)) {
01326       chan->_softhangup |= AST_SOFTHANGUP_DEV;
01327       memset(&dsp->f, 0, sizeof(dsp->f));
01328       dsp->f.frametype = AST_FRAME_CONTROL;
01329       dsp->f.subclass = AST_CONTROL_BUSY;
01330       ast_frfree(af);
01331       ast_debug(1, "Requesting Hangup because the busy tone was detected on channel %s\n", chan->name);
01332       return ast_frisolate(&dsp->f);
01333    }
01334 
01335    if ((dsp->features & DSP_FEATURE_FAX_DETECT)) {
01336       if ((dsp->faxmode & DSP_FAXMODE_DETECT_CNG) && tone_detect(dsp, &dsp->cng_tone_state, shortdata, len)) {
01337          fax_digit = 'f';
01338       }
01339 
01340       if ((dsp->faxmode & DSP_FAXMODE_DETECT_CED) && tone_detect(dsp, &dsp->ced_tone_state, shortdata, len)) {
01341          fax_digit = 'e';
01342       }
01343    }
01344 
01345    if (dsp->features & (DSP_FEATURE_DIGIT_DETECT | DSP_FEATURE_BUSY_DETECT)) {
01346       if (dsp->digitmode & DSP_DIGITMODE_MF)
01347          digit = mf_detect(dsp, &dsp->digit_state, shortdata, len, (dsp->digitmode & DSP_DIGITMODE_NOQUELCH) == 0, (dsp->digitmode & DSP_DIGITMODE_RELAXDTMF));
01348       else
01349          digit = dtmf_detect(dsp, &dsp->digit_state, shortdata, len, (dsp->digitmode & DSP_DIGITMODE_NOQUELCH) == 0, (dsp->digitmode & DSP_DIGITMODE_RELAXDTMF));
01350 
01351       if (dsp->digit_state.current_digits) {
01352          int event = 0;
01353          char event_digit = 0;
01354 
01355          if (!dsp->dtmf_began) {
01356             /* We have not reported DTMF_BEGIN for anything yet */
01357 
01358             if (dsp->features & DSP_FEATURE_DIGIT_DETECT) {
01359                event = AST_FRAME_DTMF_BEGIN;
01360                event_digit = dsp->digit_state.digits[0];
01361             }
01362             dsp->dtmf_began = 1;
01363 
01364          } else if (dsp->digit_state.current_digits > 1 || digit != dsp->digit_state.digits[0]) {
01365             /* Digit changed. This means digit we have reported with DTMF_BEGIN ended */
01366             if (dsp->features & DSP_FEATURE_DIGIT_DETECT) {
01367                event = AST_FRAME_DTMF_END;
01368                event_digit = dsp->digit_state.digits[0];
01369             }
01370             memmove(dsp->digit_state.digits, dsp->digit_state.digits + 1, dsp->digit_state.current_digits);
01371             dsp->digit_state.current_digits--;
01372             dsp->dtmf_began = 0;
01373 
01374             if (dsp->features & DSP_FEATURE_BUSY_DETECT) {
01375                /* Reset Busy Detector as we have some confirmed activity */ 
01376                memset(dsp->historicsilence, 0, sizeof(dsp->historicsilence));
01377                memset(dsp->historicnoise, 0, sizeof(dsp->historicnoise));
01378                ast_debug(1, "DTMF Detected - Reset busydetector\n");
01379             }
01380          }
01381 
01382          if (event) {
01383             memset(&dsp->f, 0, sizeof(dsp->f));
01384             dsp->f.frametype = event;
01385             dsp->f.subclass = event_digit;
01386             outf = &dsp->f;
01387             goto done;
01388          }
01389       }
01390    }
01391 
01392    if (fax_digit) {
01393       /* Fax was detected - digit is either 'f' or 'e' */
01394 
01395       memset(&dsp->f, 0, sizeof(dsp->f));
01396       dsp->f.frametype = AST_FRAME_DTMF;
01397       dsp->f.subclass = fax_digit;
01398       outf = &dsp->f;
01399       goto done;
01400    }
01401 
01402    if ((dsp->features & DSP_FEATURE_CALL_PROGRESS)) {
01403       res = __ast_dsp_call_progress(dsp, shortdata, len);
01404       if (res) {
01405          switch (res) {
01406          case AST_CONTROL_ANSWER:
01407          case AST_CONTROL_BUSY:
01408          case AST_CONTROL_RINGING:
01409          case AST_CONTROL_CONGESTION:
01410          case AST_CONTROL_HANGUP:
01411             memset(&dsp->f, 0, sizeof(dsp->f));
01412             dsp->f.frametype = AST_FRAME_CONTROL;
01413             dsp->f.subclass = res;
01414             dsp->f.src = "dsp_progress";
01415             if (chan) 
01416                ast_queue_frame(chan, &dsp->f);
01417             break;
01418          default:
01419             ast_log(LOG_WARNING, "Don't know how to represent call progress message %d\n", res);
01420          }
01421       }
01422    }
01423 
01424 done:
01425    /* Mute fragment of the frame */
01426    for (x = 0; x < dsp->mute_fragments; x++) {
01427       memset(shortdata + dsp->mute_data[x].start, 0, sizeof(int16_t) * (dsp->mute_data[x].end - dsp->mute_data[x].start));
01428    }
01429 
01430    switch (af->subclass) {
01431    case AST_FORMAT_SLINEAR:
01432       break;
01433    case AST_FORMAT_ULAW:
01434       for (x = 0; x < len; x++)
01435          odata[x] = AST_LIN2MU((unsigned short) shortdata[x]);
01436       break;
01437    case AST_FORMAT_ALAW:
01438       for (x = 0; x < len; x++)
01439          odata[x] = AST_LIN2A((unsigned short) shortdata[x]);
01440       break;
01441    }
01442 
01443    if (outf) {
01444       if (chan) 
01445          ast_queue_frame(chan, af);
01446       ast_frfree(af);
01447       return ast_frisolate(outf);
01448    } else {
01449       return af;
01450    }
01451 }
01452 
01453 static void ast_dsp_prog_reset(struct ast_dsp *dsp)
01454 {
01455    int max = 0;
01456    int x;
01457    
01458    dsp->gsamp_size = modes[dsp->progmode].size;
01459    dsp->gsamps = 0;
01460    for (x = 0; x < ARRAY_LEN(modes[dsp->progmode].freqs); x++) {
01461       if (modes[dsp->progmode].freqs[x]) {
01462          goertzel_init(&dsp->freqs[x], (float)modes[dsp->progmode].freqs[x], dsp->gsamp_size);
01463          max = x + 1;
01464       }
01465    }
01466    dsp->freqcount = max;
01467    dsp->ringtimeout= 0;
01468 }
01469 
01470 struct ast_dsp *ast_dsp_new(void)
01471 {
01472    struct ast_dsp *dsp;
01473    
01474    if ((dsp = ast_calloc(1, sizeof(*dsp)))) {      
01475       dsp->threshold = DEFAULT_THRESHOLD;
01476       dsp->features = DSP_FEATURE_SILENCE_SUPPRESS;
01477       dsp->busycount = DSP_HISTORY;
01478       dsp->digitmode = DSP_DIGITMODE_DTMF;
01479       dsp->faxmode = DSP_FAXMODE_DETECT_CNG;
01480       /* Initialize digit detector */
01481       ast_digit_detect_init(&dsp->digit_state, dsp->digitmode & DSP_DIGITMODE_MF);
01482       dsp->display_inband_dtmf_warning = 1;
01483       /* Initialize initial DSP progress detect parameters */
01484       ast_dsp_prog_reset(dsp);
01485       /* Initialize fax detector */
01486       ast_fax_detect_init(dsp);
01487    }
01488    return dsp;
01489 }
01490 
01491 void ast_dsp_set_features(struct ast_dsp *dsp, int features)
01492 {
01493    dsp->features = features;
01494 }
01495 
01496 void ast_dsp_free(struct ast_dsp *dsp)
01497 {
01498    ast_free(dsp);
01499 }
01500 
01501 void ast_dsp_set_threshold(struct ast_dsp *dsp, int threshold)
01502 {
01503    dsp->threshold = threshold;
01504 }
01505 
01506 void ast_dsp_set_busy_count(struct ast_dsp *dsp, int cadences)
01507 {
01508    if (cadences < 4)
01509       cadences = 4;
01510    if (cadences > DSP_HISTORY)
01511       cadences = DSP_HISTORY;
01512    dsp->busycount = cadences;
01513 }
01514 
01515 void ast_dsp_set_busy_pattern(struct ast_dsp *dsp, int tonelength, int quietlength)
01516 {
01517    dsp->busy_tonelength = tonelength;
01518    dsp->busy_quietlength = quietlength;
01519    ast_debug(1, "dsp busy pattern set to %d,%d\n", tonelength, quietlength);
01520 }
01521 
01522 void ast_dsp_digitreset(struct ast_dsp *dsp)
01523 {
01524    int i;
01525    
01526    dsp->dtmf_began = 0;
01527    if (dsp->digitmode & DSP_DIGITMODE_MF) {
01528       mf_detect_state_t *s = &dsp->digit_state.td.mf;
01529       /* Reinitialise the detector for the next block */
01530       for (i = 0;  i < 6;  i++) {
01531          goertzel_reset(&s->tone_out[i]);
01532       }
01533       s->hits[4] = s->hits[3] = s->hits[2] = s->hits[1] = s->hits[0] = s->current_hit = 0;
01534       s->current_sample = 0;
01535    } else {
01536       dtmf_detect_state_t *s = &dsp->digit_state.td.dtmf;
01537       /* Reinitialise the detector for the next block */
01538       for (i = 0;  i < 4;  i++) {
01539          goertzel_reset(&s->row_out[i]);
01540          goertzel_reset(&s->col_out[i]);
01541       }
01542       s->lasthit = s->current_hit = 0;
01543       s->energy = 0.0;
01544       s->current_sample = 0;
01545       s->hits = 0;
01546       s->misses = 0;
01547    }
01548 
01549    dsp->digit_state.digits[0] = '\0';
01550    dsp->digit_state.current_digits = 0;
01551 }
01552 
01553 void ast_dsp_reset(struct ast_dsp *dsp)
01554 {
01555    int x;
01556    
01557    dsp->totalsilence = 0;
01558    dsp->gsamps = 0;
01559    for (x=0;x<4;x++)
01560       dsp->freqs[x].v2 = dsp->freqs[x].v3 = 0.0;
01561    memset(dsp->historicsilence, 0, sizeof(dsp->historicsilence));
01562    memset(dsp->historicnoise, 0, sizeof(dsp->historicnoise));  
01563    dsp->ringtimeout= 0;
01564 }
01565 
01566 int ast_dsp_set_digitmode(struct ast_dsp *dsp, int digitmode)
01567 {
01568    int new;
01569    int old;
01570    
01571    old = dsp->digitmode & (DSP_DIGITMODE_DTMF | DSP_DIGITMODE_MF | DSP_DIGITMODE_MUTECONF | DSP_DIGITMODE_MUTEMAX);
01572    new = digitmode & (DSP_DIGITMODE_DTMF | DSP_DIGITMODE_MF | DSP_DIGITMODE_MUTECONF | DSP_DIGITMODE_MUTEMAX);
01573    if (old != new) {
01574       /* Must initialize structures if switching from MF to DTMF or vice-versa */
01575       ast_digit_detect_init(&dsp->digit_state, new & DSP_DIGITMODE_MF);
01576    }
01577    dsp->digitmode = digitmode;
01578    return 0;
01579 }
01580 
01581 int ast_dsp_set_faxmode(struct ast_dsp *dsp, int faxmode)
01582 {
01583    if (dsp->faxmode != faxmode) {
01584       ast_fax_detect_init(dsp);
01585    }
01586    dsp->faxmode = faxmode;
01587    return 0;
01588 }
01589 
01590 int ast_dsp_set_call_progress_zone(struct ast_dsp *dsp, char *zone)
01591 {
01592    int x;
01593    
01594    for (x = 0; x < ARRAY_LEN(aliases); x++) {
01595       if (!strcasecmp(aliases[x].name, zone)) {
01596          dsp->progmode = aliases[x].mode;
01597          ast_dsp_prog_reset(dsp);
01598          return 0;
01599       }
01600    }
01601    return -1;
01602 }
01603 
01604 int ast_dsp_was_muted(struct ast_dsp *dsp)
01605 {
01606    return (dsp->mute_fragments > 0);
01607 }
01608 
01609 int ast_dsp_get_tstate(struct ast_dsp *dsp) 
01610 {
01611    return dsp->tstate;
01612 }
01613 
01614 int ast_dsp_get_tcount(struct ast_dsp *dsp) 
01615 {
01616    return dsp->tcount;
01617 }
01618 
01619 static int _dsp_init(int reload)
01620 {
01621    struct ast_flags config_flags = { reload ? CONFIG_FLAG_FILEUNCHANGED : 0 };
01622    struct ast_config *cfg;
01623 
01624    cfg = ast_config_load2(CONFIG_FILE_NAME, "dsp", config_flags);
01625 
01626    if (cfg && cfg != CONFIG_STATUS_FILEUNCHANGED) {
01627       const char *value;
01628 
01629       value = ast_variable_retrieve(cfg, "default", "silencethreshold");
01630       if (value && sscanf(value, "%30d", &thresholds[THRESHOLD_SILENCE]) != 1) {
01631          ast_log(LOG_WARNING, "%s: '%s' is not a valid silencethreshold value\n", CONFIG_FILE_NAME, value);
01632          thresholds[THRESHOLD_SILENCE] = 256;
01633       } else if (!value)
01634          thresholds[THRESHOLD_SILENCE] = 256;
01635 
01636       ast_config_destroy(cfg);
01637    }
01638    return 0;
01639 }
01640 
01641 int ast_dsp_get_threshold_from_settings(enum threshold which)
01642 {
01643    return thresholds[which];
01644 }
01645 
01646 int ast_dsp_init(void)
01647 {
01648    return _dsp_init(0);
01649 }
01650 
01651 int ast_dsp_reload(void)
01652 {
01653    return _dsp_init(1);
01654 }
01655 

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