#include "asterisk.h"
#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"
Go to the source code of this file.
Data Structures | |
struct | ast_dsp |
struct | digit_detect_state_t |
struct | dtmf_detect_state_t |
struct | fragment_t |
struct | goertzel_result_t |
struct | goertzel_state_t |
struct | mf_detect_state_t |
struct | progalias |
struct | progress |
struct | tone_detect_state_t |
Defines | |
#define | BELL_MF_RELATIVE_PEAK 12.6 |
#define | BELL_MF_THRESHOLD 1.6e9 |
#define | BELL_MF_TWIST 4.0 |
#define | CONFIG_FILE_NAME "dsp.conf" |
#define | DEFAULT_THRESHOLD 512 |
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 | DSP_HISTORY 15 |
#define | DTMF_2ND_HARMONIC_COL 63.1 |
#define | DTMF_2ND_HARMONIC_ROW (relax ? 1.7 : 2.5) |
#define | DTMF_GSIZE 102 |
#define | DTMF_HITS_TO_BEGIN 4 |
#define | DTMF_MISSES_TO_END 4 |
#define | DTMF_NORMAL_TWIST 6.3 |
#define | DTMF_RELATIVE_PEAK_COL 6.3 |
#define | DTMF_RELATIVE_PEAK_ROW 6.3 |
#define | DTMF_REVERSE_TWIST (relax ? 4.0 : 2.5) |
#define | DTMF_THRESHOLD 8.0e7 |
#define | DTMF_TO_TOTAL_ENERGY 42.0 |
#define | FAX_2ND_HARMONIC 2.0 |
#define | FAX_THRESHOLD 8.0e7 |
#define | FAX_TONE_CED_DB 16 |
#define | FAX_TONE_CED_DURATION 2600 |
#define | FAX_TONE_CED_FREQ 2100 |
#define | FAX_TONE_CNG_DB 16 |
#define | FAX_TONE_CNG_DURATION 500 |
#define | FAX_TONE_CNG_FREQ 1100 |
#define | MAX_DTMF_DIGITS 128 |
#define | MF_GSIZE 120 |
#define | SAMPLE_RATE 8000 |
#define | SAMPLES_IN_FRAME 160 |
#define | TONE_MIN_THRESH 1e8 |
#define | TONE_THRESH 10.0 |
Enumerations | |
enum | busy_detect { BUSY_PERCENT = 10, BUSY_PAT_PERCENT = 8, BUSY_THRESHOLD = 100, BUSY_MIN = 150, BUSY_MAX = 600 } |
enum | freq_index { HZ_350 = 0, HZ_440, HZ_480, HZ_620, HZ_950, HZ_1400, HZ_1800, HZ_425 = 0, HZ_350UK = 0, HZ_400UK, HZ_440UK } |
enum | gsamp_size { GSAMP_SIZE_NA = 183, GSAMP_SIZE_CR = 188, GSAMP_SIZE_UK = 160 } |
enum | gsamp_thresh { THRESH_RING = 8, THRESH_TALK = 2, THRESH_BUSY = 4, THRESH_CONGESTION = 4, THRESH_HANGUP = 60, THRESH_RING2ANSWER = 300 } |
enum | prog_mode { PROG_MODE_NA = 0, PROG_MODE_CR, PROG_MODE_UK } |
Functions | |
static int | __ast_dsp_call_progress (struct ast_dsp *dsp, short *s, int len) |
static int | __ast_dsp_silence_noise (struct ast_dsp *dsp, short *s, int len, int *totalsilence, int *totalnoise) |
static int | _dsp_init (int reload) |
static void | ast_digit_detect_init (digit_detect_state_t *s, int mf) |
int | ast_dsp_busydetect (struct ast_dsp *dsp) |
Return non-zero if historically this should be a busy, request that ast_dsp_silence has already been called. | |
int | ast_dsp_call_progress (struct ast_dsp *dsp, struct ast_frame *inf) |
Scans for progress indication in audio. | |
void | ast_dsp_digitreset (struct ast_dsp *dsp) |
Reset DTMF detector. | |
void | ast_dsp_free (struct ast_dsp *dsp) |
int | ast_dsp_get_tcount (struct ast_dsp *dsp) |
Get tcount (Threshold counter). | |
int | ast_dsp_get_threshold_from_settings (enum threshold which) |
Get silence threshold from dsp.conf. | |
int | ast_dsp_get_tstate (struct ast_dsp *dsp) |
Get tstate (Tone State). | |
int | ast_dsp_init (void) |
Load dsp settings from dsp.conf. | |
ast_dsp * | ast_dsp_new (void) |
int | ast_dsp_noise (struct ast_dsp *dsp, struct ast_frame *f, int *totalnoise) |
Return non-zero if this is noise. Updates "totalnoise" with the total number of seconds of noise. | |
ast_frame * | ast_dsp_process (struct ast_channel *chan, struct ast_dsp *dsp, struct ast_frame *af) |
Return AST_FRAME_NULL frames when there is silence, AST_FRAME_BUSY on busies, and call progress, all dependent upon which features are enabled. | |
static void | ast_dsp_prog_reset (struct ast_dsp *dsp) |
int | ast_dsp_reload (void) |
Reloads dsp settings from dsp.conf. | |
void | ast_dsp_reset (struct ast_dsp *dsp) |
Reset total silence count. | |
void | ast_dsp_set_busy_compare (struct ast_dsp *dsp, int compare) |
Set if silence and noice lengths must be compared for busy. | |
void | ast_dsp_set_busy_count (struct ast_dsp *dsp, int cadences) |
Set number of required cadences for busy. | |
void | ast_dsp_set_busy_pattern (struct ast_dsp *dsp, int tonelength, int quietlength, int fuzzy) |
Set expected lengths of the busy tones. | |
int | ast_dsp_set_call_progress_zone (struct ast_dsp *dsp, char *zone) |
Set zone for doing progress detection. | |
int | ast_dsp_set_digitmode (struct ast_dsp *dsp, int digitmode) |
Set digit mode. | |
int | ast_dsp_set_faxmode (struct ast_dsp *dsp, int faxmode) |
Set fax mode. | |
void | ast_dsp_set_features (struct ast_dsp *dsp, int features) |
Select feature set. | |
void | ast_dsp_set_threshold (struct ast_dsp *dsp, int threshold) |
Set threshold value for silence. | |
int | ast_dsp_silence (struct ast_dsp *dsp, struct ast_frame *f, int *totalsilence) |
Return non-zero if this is silence. Updates "totalsilence" with the total number of seconds of silence. | |
int | ast_dsp_was_muted (struct ast_dsp *dsp) |
Returns true if DSP code was muting any fragment of the last processed frame. Muting (squelching) happens when DSP code removes DTMF/MF/generic tones from the audio. | |
static void | ast_dtmf_detect_init (dtmf_detect_state_t *s) |
static void | ast_fax_detect_init (struct ast_dsp *s) |
static void | ast_mf_detect_init (mf_detect_state_t *s) |
static void | ast_tone_detect_init (tone_detect_state_t *s, int freq, int duration, int amp) |
static int | dtmf_detect (struct ast_dsp *dsp, digit_detect_state_t *s, int16_t amp[], int samples, int squelch, int relax) |
static void | goertzel_init (goertzel_state_t *s, float freq, int samples) |
static void | goertzel_reset (goertzel_state_t *s) |
static float | goertzel_result (goertzel_state_t *s) |
static void | goertzel_sample (goertzel_state_t *s, short sample) |
static void | goertzel_update (goertzel_state_t *s, short *samps, int count) |
static int | mf_detect (struct ast_dsp *dsp, digit_detect_state_t *s, int16_t amp[], int samples, int squelch, int relax) |
static void | mute_fragment (struct ast_dsp *dsp, fragment_t *fragment) |
static int | pair_there (float p1, float p2, float i1, float i2, float e) |
static void | store_digit (digit_detect_state_t *s, char digit) |
static int | tone_detect (struct ast_dsp *dsp, tone_detect_state_t *s, int16_t *amp, int samples) |
Variables | |
static struct progalias | aliases [] |
static const char | bell_mf_positions [] = "1247C-358A--69*---0B----#" |
static const int | DEFAULT_SILENCE_THRESHOLD = 256 |
The default silence threshold we will use if an alternate configured value is not present or is invalid. | |
static const float | dtmf_col [] |
static const char | dtmf_positions [] = "123A" "456B" "789C" "*0#D" |
static const float | dtmf_row [] |
static const float | mf_tones [] |
static struct progress | modes [] |
static int | thresholds [THRESHOLD_MAX] |
Steve Underwood <steveu@coppice.org>
Definition in file dsp.c.
#define DEFAULT_THRESHOLD 512 |
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.
Definition at line 117 of file dsp.c.
Referenced by ast_dsp_new().
#define DSP_HISTORY 15 |
Remember last 15 units
Definition at line 128 of file dsp.c.
Referenced by __ast_dsp_silence_noise(), ast_dsp_busydetect(), ast_dsp_new(), and ast_dsp_set_busy_count().
#define DTMF_GSIZE 102 |
#define DTMF_HITS_TO_BEGIN 4 |
#define DTMF_MISSES_TO_END 4 |
#define DTMF_NORMAL_TWIST 6.3 |
#define DTMF_RELATIVE_PEAK_COL 6.3 |
#define DTMF_RELATIVE_PEAK_ROW 6.3 |
#define DTMF_REVERSE_TWIST (relax ? 4.0 : 2.5) |
#define DTMF_THRESHOLD 8.0e7 |
#define DTMF_TO_TOTAL_ENERGY 42.0 |
#define FAX_TONE_CED_DB 16 |
#define FAX_TONE_CED_DURATION 2600 |
#define FAX_TONE_CED_FREQ 2100 |
#define FAX_TONE_CNG_DB 16 |
#define FAX_TONE_CNG_DURATION 500 |
#define FAX_TONE_CNG_FREQ 1100 |
#define MAX_DTMF_DIGITS 128 |
#define MF_GSIZE 120 |
#define SAMPLES_IN_FRAME 160 |
#define TONE_MIN_THRESH 1e8 |
How much tone there should be at least to attempt
Definition at line 131 of file dsp.c.
Referenced by pair_there().
#define TONE_THRESH 10.0 |
How much louder the tone should be than channel energy
Definition at line 130 of file dsp.c.
Referenced by pair_there().
enum busy_detect |
Definition at line 119 of file dsp.c.
00119 { 00120 BUSY_PERCENT = 10, /*!< The percentage difference between the two last tone periods */ 00121 BUSY_PAT_PERCENT = 8, /*!< The percentage difference between measured and actual pattern */ 00122 BUSY_THRESHOLD = 100, /*!< Max number of ms difference between max and min times in busy */ 00123 BUSY_MIN = 150, /*!< Busy must be at least 150 ms in half-cadence */ 00124 BUSY_MAX = 600 /*!< Busy can't be longer than 600 ms in half-cadence */ 00125 };
enum freq_index |
HZ_350 | For US modes { |
HZ_440 | |
HZ_480 | |
HZ_620 | |
HZ_950 | |
HZ_1400 | |
HZ_1800 | } |
HZ_425 | For CR/BR modes |
HZ_350UK | For UK mode |
HZ_400UK | |
HZ_440UK |
Definition at line 71 of file dsp.c.
00071 { 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_350UK = 0, 00086 HZ_400UK, 00087 HZ_440UK 00088 };
enum gsamp_size |
Number of goertzels for progress detect
GSAMP_SIZE_NA | North America - 350, 440, 480, 620, 950, 1400, 1800 Hz |
GSAMP_SIZE_CR | Costa Rica, Brazil - Only care about 425 Hz |
GSAMP_SIZE_UK | UK disconnect goertzel feed - should trigger 400hz |
Definition at line 59 of file dsp.c.
00059 { 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 };
enum gsamp_thresh |
All THRESH_XXX values are in GSAMP_SIZE chunks (us = 22ms)
Definition at line 134 of file dsp.c.
00134 { 00135 THRESH_RING = 8, /*!< Need at least 150ms ring to accept */ 00136 THRESH_TALK = 2, /*!< Talk detection does not work continuously */ 00137 THRESH_BUSY = 4, /*!< Need at least 80ms to accept */ 00138 THRESH_CONGESTION = 4, /*!< Need at least 80ms to accept */ 00139 THRESH_HANGUP = 60, /*!< Need at least 1300ms to accept hangup */ 00140 THRESH_RING2ANSWER = 300 /*!< Timeout from start of ring to answer (about 6600 ms) */ 00141 };
enum prog_mode |
Definition at line 65 of file dsp.c.
00065 { 00066 PROG_MODE_NA = 0, 00067 PROG_MODE_CR, 00068 PROG_MODE_UK 00069 };
static int __ast_dsp_call_progress | ( | struct ast_dsp * | dsp, | |
short * | s, | |||
int | len | |||
) | [static] |
Definition at line 965 of file dsp.c.
References DSP_TONE_STATE_BUSY, DSP_TONE_STATE_RINGING, DSP_TONE_STATE_SILENCE, ast_dsp::freqcount, ast_dsp::freqs, ast_dsp::genergy, goertzel_result(), goertzel_sample(), ast_dsp::gsamp_size, ast_dsp::gsamps, HZ_350, HZ_440, HZ_480, HZ_620, pair_there(), pass, PROG_MODE_NA, and ast_dsp::progmode.
Referenced by ast_dsp_call_progress().
00966 { 00967 int x; 00968 int y; 00969 int pass; 00970 int newstate = DSP_TONE_STATE_SILENCE; 00971 int res = 0; 00972 while (len) { 00973 /* Take the lesser of the number of samples we need and what we have */ 00974 pass = len; 00975 if (pass > dsp->gsamp_size - dsp->gsamps) { 00976 pass = dsp->gsamp_size - dsp->gsamps; 00977 } 00978 for (x = 0; x < pass; x++) { 00979 for (y = 0; y < dsp->freqcount; y++) { 00980 goertzel_sample(&dsp->freqs[y], s[x]); 00981 } 00982 dsp->genergy += s[x] * s[x]; 00983 } 00984 s += pass; 00985 dsp->gsamps += pass; 00986 len -= pass; 00987 if (dsp->gsamps == dsp->gsamp_size) { 00988 float hz[7]; 00989 for (y = 0; y < 7; y++) { 00990 hz[y] = goertzel_result(&dsp->freqs[y]); 00991 } 00992 switch (dsp->progmode) { 00993 case PROG_MODE_NA: 00994 if (pair_there(hz[HZ_480], hz[HZ_620], hz[HZ_350], hz[HZ_440], dsp->genergy)) { 00995 newstate = DSP_TONE_STATE_BUSY; 00996 } else if (pair_there(hz[HZ_440], hz[HZ_480], hz[HZ_350], hz[HZ_620], dsp->genergy)) { 00997 newstate = DSP_TONE_STATE_RINGING; 00998 } else if (pair_there(hz[HZ_350], hz[HZ_440], hz[HZ_480], hz[HZ_620], dsp->genergy)) { 00999 newstate = DSP_TONE_STATE_DIALTONE; 01000 } else if (hz[HZ_950] > TONE_MIN_THRESH * TONE_THRESH) { 01001 newstate = DSP_TONE_STATE_SPECIAL1; 01002 } else if (hz[HZ_1400] > TONE_MIN_THRESH * TONE_THRESH) { 01003 /* End of SPECIAL1 or middle of SPECIAL2 */ 01004 if (dsp->tstate == DSP_TONE_STATE_SPECIAL1 || dsp->tstate == DSP_TONE_STATE_SPECIAL2) { 01005 newstate = DSP_TONE_STATE_SPECIAL2; 01006 } 01007 } else if (hz[HZ_1800] > TONE_MIN_THRESH * TONE_THRESH) { 01008 /* End of SPECIAL2 or middle of SPECIAL3 */ 01009 if (dsp->tstate == DSP_TONE_STATE_SPECIAL2 || dsp->tstate == DSP_TONE_STATE_SPECIAL3) { 01010 newstate = DSP_TONE_STATE_SPECIAL3; 01011 } 01012 } else if (dsp->genergy > TONE_MIN_THRESH * TONE_THRESH) { 01013 newstate = DSP_TONE_STATE_TALKING; 01014 } else { 01015 newstate = DSP_TONE_STATE_SILENCE; 01016 } 01017 break; 01018 case PROG_MODE_CR: 01019 if (hz[HZ_425] > TONE_MIN_THRESH * TONE_THRESH) { 01020 newstate = DSP_TONE_STATE_RINGING; 01021 } else if (dsp->genergy > TONE_MIN_THRESH * TONE_THRESH) { 01022 newstate = DSP_TONE_STATE_TALKING; 01023 } else { 01024 newstate = DSP_TONE_STATE_SILENCE; 01025 } 01026 break; 01027 case PROG_MODE_UK: 01028 if (hz[HZ_400UK] > TONE_MIN_THRESH * TONE_THRESH) { 01029 newstate = DSP_TONE_STATE_HUNGUP; 01030 } else if (pair_there(hz[HZ_350UK], hz[HZ_440UK], hz[HZ_400UK], hz[HZ_400UK], dsp->genergy)) { 01031 newstate = DSP_TONE_STATE_DIALTONE; 01032 } 01033 break; 01034 default: 01035 ast_log(LOG_WARNING, "Can't process in unknown prog mode '%d'\n", dsp->progmode); 01036 } 01037 if (newstate == dsp->tstate) { 01038 dsp->tcount++; 01039 if (dsp->ringtimeout) { 01040 dsp->ringtimeout++; 01041 } 01042 switch (dsp->tstate) { 01043 case DSP_TONE_STATE_RINGING: 01044 if ((dsp->features & DSP_PROGRESS_RINGING) && 01045 (dsp->tcount == THRESH_RING)) { 01046 res = AST_CONTROL_RINGING; 01047 dsp->ringtimeout = 1; 01048 } 01049 break; 01050 case DSP_TONE_STATE_BUSY: 01051 if ((dsp->features & DSP_PROGRESS_BUSY) && 01052 (dsp->tcount == THRESH_BUSY)) { 01053 res = AST_CONTROL_BUSY; 01054 dsp->features &= ~DSP_FEATURE_CALL_PROGRESS; 01055 } 01056 break; 01057 case DSP_TONE_STATE_TALKING: 01058 if ((dsp->features & DSP_PROGRESS_TALK) && 01059 (dsp->tcount == THRESH_TALK)) { 01060 res = AST_CONTROL_ANSWER; 01061 dsp->features &= ~DSP_FEATURE_CALL_PROGRESS; 01062 } 01063 break; 01064 case DSP_TONE_STATE_SPECIAL3: 01065 if ((dsp->features & DSP_PROGRESS_CONGESTION) && 01066 (dsp->tcount == THRESH_CONGESTION)) { 01067 res = AST_CONTROL_CONGESTION; 01068 dsp->features &= ~DSP_FEATURE_CALL_PROGRESS; 01069 } 01070 break; 01071 case DSP_TONE_STATE_HUNGUP: 01072 if ((dsp->features & DSP_FEATURE_CALL_PROGRESS) && 01073 (dsp->tcount == THRESH_HANGUP)) { 01074 res = AST_CONTROL_HANGUP; 01075 dsp->features &= ~DSP_FEATURE_CALL_PROGRESS; 01076 } 01077 break; 01078 } 01079 if (dsp->ringtimeout == THRESH_RING2ANSWER) { 01080 ast_debug(1, "Consider call as answered because of timeout after last ring\n"); 01081 res = AST_CONTROL_ANSWER; 01082 dsp->features &= ~DSP_FEATURE_CALL_PROGRESS; 01083 } 01084 } else { 01085 ast_debug(5, "Stop state %d with duration %d\n", dsp->tstate, dsp->tcount); 01086 ast_debug(5, "Start state %d\n", newstate); 01087 dsp->tstate = newstate; 01088 dsp->tcount = 1; 01089 } 01090 01091 /* Reset goertzel */ 01092 for (x = 0; x < 7; x++) { 01093 dsp->freqs[x].v2 = dsp->freqs[x].v3 = 0.0; 01094 } 01095 dsp->gsamps = 0; 01096 dsp->genergy = 0.0; 01097 } 01098 } 01099 01100 return res; 01101 }
static int __ast_dsp_silence_noise | ( | struct ast_dsp * | dsp, | |
short * | s, | |||
int | len, | |||
int * | totalsilence, | |||
int * | totalnoise | |||
) | [static] |
Definition at line 1116 of file dsp.c.
References BUSY_PERCENT, ast_dsp::busycount, ast_dsp::busymaybe, DSP_HISTORY, ast_dsp::historicnoise, ast_dsp::totalnoise, and ast_dsp::totalsilence.
Referenced by ast_dsp_noise(), ast_dsp_process(), and ast_dsp_silence().
01117 { 01118 int accum; 01119 int x; 01120 int res = 0; 01121 01122 if (!len) { 01123 return 0; 01124 } 01125 accum = 0; 01126 for (x = 0; x < len; x++) { 01127 accum += abs(s[x]); 01128 } 01129 accum /= len; 01130 if (accum < dsp->threshold) { 01131 /* Silent */ 01132 dsp->totalsilence += len / 8; 01133 #ifdef DEBUG_DSP_BUSYDETECT 01134 fprintf(stderr, "SILENCE: len = %d, level = %d\n", dsp->totalsilence, accum); 01135 #endif 01136 if (dsp->totalnoise) { 01137 /* Move and save history */ 01138 memmove(dsp->historicnoise + DSP_HISTORY - dsp->busycount, dsp->historicnoise + DSP_HISTORY - dsp->busycount + 1, (dsp->busycount-1) * sizeof(dsp->historicnoise[0])); 01139 dsp->historicnoise[DSP_HISTORY - 1] = dsp->totalnoise; 01140 /* check if previous tone differs BUSY_PERCENT from the one before it */ 01141 int tone1 = dsp->historicnoise[DSP_HISTORY - 1]; 01142 int tone2 = dsp->historicnoise[DSP_HISTORY - 2]; 01143 if (tone1 < tone2) { 01144 if ((tone1 + tone1*BUSY_PERCENT/100) >= tone2) 01145 dsp->busymaybe = 1; 01146 else 01147 dsp->busymaybe = 0; 01148 } else { 01149 if ((tone1 - tone1*BUSY_PERCENT/100) <= tone2) 01150 dsp->busymaybe = 1; 01151 else 01152 dsp->busymaybe = 0; 01153 } 01154 } 01155 dsp->totalnoise = 0; 01156 res = 1; 01157 } else { 01158 /* Not silent */ 01159 dsp->totalnoise += len / 8; 01160 #ifdef DEBUG_DSP_BUSYDETECT 01161 fprintf(stderr, "NOISE: len = %d, level = %d\n", dsp->totalnoise, accum); 01162 #endif 01163 if (dsp->totalsilence) { 01164 /* Move and save history */ 01165 memmove(dsp->historicsilence + DSP_HISTORY - dsp->busycount, dsp->historicsilence + DSP_HISTORY - dsp->busycount + 1, (dsp->busycount-1) * sizeof(dsp->historicsilence[0])); 01166 dsp->historicsilence[DSP_HISTORY - 1] = dsp->totalsilence; 01167 } 01168 dsp->totalsilence = 0; 01169 } 01170 if (totalsilence) { 01171 *totalsilence = dsp->totalsilence; 01172 } 01173 if (totalnoise) { 01174 *totalnoise = dsp->totalnoise; 01175 } 01176 return res; 01177 }
static int _dsp_init | ( | int | reload | ) | [static] |
Definition at line 1721 of file dsp.c.
References ast_config_destroy(), ast_config_load2(), ast_variable_retrieve(), ast_verb, CONFIG_FILE_NAME, CONFIG_FLAG_FILEUNCHANGED, config_flags, CONFIG_STATUS_FILEINVALID, CONFIG_STATUS_FILEMISSING, CONFIG_STATUS_FILEUNCHANGED, THRESHOLD_SILENCE, and value.
Referenced by ast_dsp_init(), and ast_dsp_reload().
01722 { 01723 struct ast_flags config_flags = { reload ? CONFIG_FLAG_FILEUNCHANGED : 0 }; 01724 struct ast_config *cfg; 01725 01726 cfg = ast_config_load2(CONFIG_FILE_NAME, "dsp", config_flags); 01727 if (cfg == CONFIG_STATUS_FILEMISSING || cfg == CONFIG_STATUS_FILEINVALID) { 01728 ast_verb(5, "Can't find dsp config file %s. Assuming default silencethreshold of %d.\n", CONFIG_FILE_NAME, DEFAULT_SILENCE_THRESHOLD); 01729 thresholds[THRESHOLD_SILENCE] = DEFAULT_SILENCE_THRESHOLD; 01730 return 0; 01731 } 01732 01733 if (cfg == CONFIG_STATUS_FILEUNCHANGED) { 01734 return 0; 01735 } 01736 01737 if (cfg) { 01738 const char *value; 01739 01740 value = ast_variable_retrieve(cfg, "default", "silencethreshold"); 01741 if (value && sscanf(value, "%30d", &thresholds[THRESHOLD_SILENCE]) != 1) { 01742 ast_verb(5, "%s: '%s' is not a valid silencethreshold value\n", CONFIG_FILE_NAME, value); 01743 thresholds[THRESHOLD_SILENCE] = DEFAULT_SILENCE_THRESHOLD; 01744 } else if (!value) { 01745 thresholds[THRESHOLD_SILENCE] = DEFAULT_SILENCE_THRESHOLD; 01746 } 01747 01748 ast_config_destroy(cfg); 01749 } 01750 return 0; 01751 }
static void ast_digit_detect_init | ( | digit_detect_state_t * | s, | |
int | mf | |||
) | [static] |
Definition at line 513 of file dsp.c.
References ast_dtmf_detect_init(), ast_mf_detect_init(), digit_detect_state_t::current_digits, digit_detect_state_t::detected_digits, digit_detect_state_t::digits, digit_detect_state_t::dtmf, digit_detect_state_t::lost_digits, digit_detect_state_t::mf, and digit_detect_state_t::td.
Referenced by ast_dsp_new(), and ast_dsp_set_digitmode().
00514 { 00515 s->current_digits = 0; 00516 s->detected_digits = 0; 00517 s->lost_digits = 0; 00518 s->digits[0] = '\0'; 00519 00520 if (mf) { 00521 ast_mf_detect_init(&s->td.mf); 00522 } else { 00523 ast_dtmf_detect_init(&s->td.dtmf); 00524 } 00525 }
int ast_dsp_busydetect | ( | struct ast_dsp * | dsp | ) |
Return non-zero if historically this should be a busy, request that ast_dsp_silence has already been called.
Definition at line 1179 of file dsp.c.
References ast_debug, ast_log(), BUSY_MAX, BUSY_MIN, ast_dsp::busy_pattern_fuzzy, BUSY_PERCENT, ast_dsp::busy_quietlength, ast_dsp::busy_tonelength, ast_dsp::busycompare, ast_dsp::busycount, ast_dsp::busymaybe, ast_dsp::busytoneonly, DSP_HISTORY, ast_dsp::historicnoise, ast_dsp::historicsilence, LOG_ERROR, LOG_NOTICE, MAX, and option_debug.
01180 { 01181 int res = 0, x; 01182 int avgsilence = 0, hitsilence = 0; 01183 int avgtone = 0, hittone = 0; 01184 #ifdef DEBUG_DSP_BUSYDETECT 01185 char buf[16]; 01186 char silence_list[64]="", tone_list[64]=""; 01187 #endif 01188 01189 if (!dsp->busymaybe) { 01190 return res; 01191 } 01192 dsp->busymaybe = 0; 01193 01194 for (x = DSP_HISTORY - dsp->busycount; x < DSP_HISTORY; x++) { 01195 avgsilence += dsp->historicsilence[x]; 01196 avgtone += dsp->historicnoise[x]; 01197 } 01198 avgsilence /= dsp->busycount; 01199 avgtone /= dsp->busycount; 01200 #ifdef DEBUG_DSP_BUSYDETECT 01201 sprintf(silence_list,"Silences: "); 01202 sprintf(tone_list,"Tones: "); 01203 #endif 01204 for (x=DSP_HISTORY - dsp->busycount; x<DSP_HISTORY; x++) { 01205 #ifdef DEBUG_DSP_BUSYDETECT 01206 snprintf(buf, sizeof(buf), "%5d ", dsp->historicsilence[x]); 01207 strcat(silence_list, buf); 01208 snprintf(buf, sizeof(buf), "%5d ", dsp->historicnoise[x]); 01209 strcat(tone_list, buf); 01210 #endif 01211 if (!dsp->busytoneonly) { 01212 if (avgsilence > dsp->historicsilence[x]) { 01213 if (avgsilence - (avgsilence*BUSY_PERCENT/100) <= dsp->historicsilence[x]) 01214 hitsilence++; 01215 } else { 01216 if (avgsilence + (avgsilence*BUSY_PERCENT/100) >= dsp->historicsilence[x]) 01217 hitsilence++; 01218 } 01219 } 01220 if (avgtone > dsp->historicnoise[x]) { 01221 if (avgtone - (avgtone * BUSY_PERCENT / 100) <= dsp->historicnoise[x]) { 01222 hittone++; 01223 } 01224 } else { 01225 if (avgtone + (avgtone * BUSY_PERCENT / 100) >= dsp->historicnoise[x]) { 01226 hittone++; 01227 } 01228 } 01229 } 01230 #ifdef DEBUG_DSP_BUSYDETECT 01231 fprintf(stderr, "BUSY DETECTOR\n"); 01232 fprintf(stderr, "%s\n", tone_list); 01233 fprintf(stderr, "%s\n", silence_list) 01234 #endif 01235 if ((dsp->busytoneonly || 01236 (hitsilence >= dsp->busycount - 1 && avgsilence >= BUSY_MIN && avgsilence <= BUSY_MAX)) && 01237 (hittone >= dsp->busycount - 1 && avgtone >= BUSY_MIN && avgtone <= BUSY_MAX)) { 01238 if (dsp->busycompare) { 01239 if (dsp->busytoneonly) { 01240 res = 1; 01241 ast_log(LOG_ERROR, "You can't use busytoneonly together with busycompare"); 01242 } else { 01243 if (avgtone > avgsilence) { 01244 if (avgtone - avgtone*BUSY_PERCENT/100 <= avgsilence) 01245 res = 1; 01246 } else { 01247 if (avgtone + avgtone*BUSY_PERCENT/100 >= avgsilence) 01248 res = 1; 01249 } 01250 } 01251 } else { 01252 res = 1; 01253 } 01254 } 01255 /* If we know the expected busy tone length, check we are in the range */ 01256 if (res && (dsp->busy_tonelength > 0)) { 01257 if (abs(avgtone - dsp->busy_tonelength) > MAX(dsp->busy_tonelength*dsp->busy_pattern_fuzzy/100, 20)) { 01258 #ifdef BUSYDETECT_DEBUG 01259 ast_debug(5, "busy detector: avgtone of %d not close enough to desired %d\n", 01260 avgtone, dsp->busy_tonelength); 01261 #endif 01262 res = 0; 01263 } 01264 } 01265 /* If we know the expected busy tone silent-period length, check we are in the range */ 01266 if (res && (!dsp->busytoneonly) && (dsp->busy_quietlength > 0)) { 01267 if (abs(avgsilence - dsp->busy_quietlength) > MAX(dsp->busy_quietlength*dsp->busy_pattern_fuzzy/100, 20)) { 01268 #ifdef BUSYDETECT_DEBUG 01269 ast_debug(5, "busy detector: avgsilence of %d not close enough to desired %d\n", 01270 avgsilence, dsp->busy_quietlength); 01271 #endif 01272 res = 0; 01273 } 01274 } 01275 if (res) { 01276 if (option_debug) 01277 ast_log(LOG_NOTICE, "ast_dsp_busydetect detected busy sequence, avgtone: %d, avgsilence %d\n", avgtone, avgsilence); 01278 } else { 01279 ast_debug(5, "busy detector: FAILED with avgtone: %d, avgsilence %d\n", avgtone, avgsilence); 01280 } 01281 return res; 01282 }
Scans for progress indication in audio.
Definition at line 1103 of file dsp.c.
References __ast_dsp_call_progress(), AST_FORMAT_SLINEAR, AST_FRAME_VOICE, ast_log(), ast_frame_subclass::codec, ast_frame::data, ast_frame::datalen, ast_frame::frametype, LOG_WARNING, ast_frame::ptr, and ast_frame::subclass.
01104 { 01105 if (inf->frametype != AST_FRAME_VOICE) { 01106 ast_log(LOG_WARNING, "Can't check call progress of non-voice frames\n"); 01107 return 0; 01108 } 01109 if (inf->subclass.codec != AST_FORMAT_SLINEAR) { 01110 ast_log(LOG_WARNING, "Can only check call progress in signed-linear frames\n"); 01111 return 0; 01112 } 01113 return __ast_dsp_call_progress(dsp, inf->data.ptr, inf->datalen / 2); 01114 }
void ast_dsp_digitreset | ( | struct ast_dsp * | dsp | ) |
Reset DTMF detector.
Definition at line 1623 of file dsp.c.
References dtmf_detect_state_t::col_out, digit_detect_state_t::current_digits, dtmf_detect_state_t::current_hit, mf_detect_state_t::current_hit, dtmf_detect_state_t::current_sample, mf_detect_state_t::current_sample, ast_dsp::digit_state, ast_dsp::digitmode, digit_detect_state_t::digits, digit_detect_state_t::dtmf, ast_dsp::dtmf_began, dtmf_detect_state_t::energy, goertzel_reset(), dtmf_detect_state_t::hits, mf_detect_state_t::hits, dtmf_detect_state_t::lasthit, digit_detect_state_t::mf, dtmf_detect_state_t::misses, dtmf_detect_state_t::row_out, digit_detect_state_t::td, and mf_detect_state_t::tone_out.
Referenced by analog_ss_thread(), and my_dsp_reset_and_flush_digits().
01624 { 01625 int i; 01626 01627 dsp->dtmf_began = 0; 01628 if (dsp->digitmode & DSP_DIGITMODE_MF) { 01629 mf_detect_state_t *s = &dsp->digit_state.td.mf; 01630 /* Reinitialise the detector for the next block */ 01631 for (i = 0; i < 6; i++) { 01632 goertzel_reset(&s->tone_out[i]); 01633 } 01634 s->hits[4] = s->hits[3] = s->hits[2] = s->hits[1] = s->hits[0] = s->current_hit = 0; 01635 s->current_sample = 0; 01636 } else { 01637 dtmf_detect_state_t *s = &dsp->digit_state.td.dtmf; 01638 /* Reinitialise the detector for the next block */ 01639 for (i = 0; i < 4; i++) { 01640 goertzel_reset(&s->row_out[i]); 01641 goertzel_reset(&s->col_out[i]); 01642 } 01643 s->lasthit = s->current_hit = 0; 01644 s->energy = 0.0; 01645 s->current_sample = 0; 01646 s->hits = 0; 01647 s->misses = 0; 01648 } 01649 01650 dsp->digit_state.digits[0] = '\0'; 01651 dsp->digit_state.current_digits = 0; 01652 }
void ast_dsp_free | ( | struct ast_dsp * | dsp | ) |
Definition at line 1579 of file dsp.c.
References ast_free.
Referenced by __ast_play_and_record(), __oh323_destroy(), analog_ss_thread(), background_detect_exec(), chan_list_destructor(), cleanup_connection(), conf_run(), dahdi_hangup(), destroy_session(), disable_dsp_detect(), do_waiting(), handle_recordfile(), isAnsweringMachine(), mgcp_hangup(), my_all_subchannels_hungup(), my_dsp_set_digitmode(), sip_rtp_read(), and unload_module().
01580 { 01581 ast_free(dsp); 01582 }
int ast_dsp_get_tcount | ( | struct ast_dsp * | dsp | ) |
Get tcount (Threshold counter).
Definition at line 1716 of file dsp.c.
References ast_dsp::tcount.
Referenced by dahdi_read().
01717 { 01718 return dsp->tcount; 01719 }
int ast_dsp_get_threshold_from_settings | ( | enum threshold | which | ) |
Get silence threshold from dsp.conf.
Definition at line 1753 of file dsp.c.
Referenced by actual_load_config(), app_exec(), ast_record_review(), conf_run(), do_waiting(), handle_recordfile(), load_config(), and setup_privacy_args().
01754 { 01755 return thresholds[which]; 01756 }
int ast_dsp_get_tstate | ( | struct ast_dsp * | dsp | ) |
Get tstate (Tone State).
Definition at line 1711 of file dsp.c.
References ast_dsp::tstate.
Referenced by dahdi_read().
01712 { 01713 return dsp->tstate; 01714 }
int ast_dsp_init | ( | void | ) |
Load dsp settings from dsp.conf.
Definition at line 1758 of file dsp.c.
References _dsp_init().
Referenced by main().
01759 { 01760 return _dsp_init(0); 01761 }
struct ast_dsp* ast_dsp_new | ( | void | ) |
Definition at line 1536 of file dsp.c.
References ast_calloc, ast_digit_detect_init(), ast_dsp_prog_reset(), ast_fax_detect_init(), BUSY_PAT_PERCENT, DEFAULT_THRESHOLD, DSP_DIGITMODE_DTMF, DSP_HISTORY, and ast_dsp::threshold.
Referenced by __ast_play_and_record(), __oh323_new(), background_detect_exec(), conf_run(), dahdi_new(), do_waiting(), enable_dsp_detect(), fax_session_new(), handle_recordfile(), isAnsweringMachine(), mgcp_new(), misdn_set_opt_exec(), my_dsp_set_digitmode(), read_config(), and store_config().
01537 { 01538 struct ast_dsp *dsp; 01539 01540 if ((dsp = ast_calloc(1, sizeof(*dsp)))) { 01541 dsp->threshold = DEFAULT_THRESHOLD; 01542 dsp->features = DSP_FEATURE_SILENCE_SUPPRESS; 01543 dsp->busycount = DSP_HISTORY; 01544 dsp->digitmode = DSP_DIGITMODE_DTMF; 01545 dsp->faxmode = DSP_FAXMODE_DETECT_CNG; 01546 dsp->busy_pattern_fuzzy = BUSY_PAT_PERCENT; 01547 #ifdef BUSYDETECT_TONEONLY 01548 dsp->busytoneonly = 1; 01549 #ifdef BUSYDETECT_COMPARE_TONE_AND_SILENCE 01550 #error "You can't use BUSYDETECT_TONEONLY together with BUSYDETECT_COMPARE_TONE_AND_SILENCE"); 01551 #endif 01552 #else 01553 dsp->busytoneonly = 0; 01554 #ifdef BUSYDETECT_COMPARE_TONE_AND_SILENCE 01555 dsp->busycompare = 1; 01556 #else 01557 dsp->busycompare = 0; 01558 #endif 01559 #endif 01560 /* Initialize digit detector */ 01561 ast_digit_detect_init(&dsp->digit_state, dsp->digitmode & DSP_DIGITMODE_MF); 01562 dsp->display_inband_dtmf_warning = 1; 01563 /* Initialize initial DSP progress detect parameters */ 01564 ast_dsp_prog_reset(dsp); 01565 /* Initialize fax detector */ 01566 ast_fax_detect_init(dsp); 01567 } 01568 return dsp; 01569 }
Return non-zero if this is noise. Updates "totalnoise" with the total number of seconds of noise.
Definition at line 1302 of file dsp.c.
References __ast_dsp_silence_noise(), AST_FORMAT_SLINEAR, AST_FRAME_VOICE, ast_log(), f, len(), and LOG_WARNING.
Referenced by do_waiting().
01303 { 01304 short *s; 01305 int len; 01306 01307 if (f->frametype != AST_FRAME_VOICE) { 01308 ast_log(LOG_WARNING, "Can't calculate noise on a non-voice frame\n"); 01309 return 0; 01310 } 01311 if (f->subclass.codec != AST_FORMAT_SLINEAR) { 01312 ast_log(LOG_WARNING, "Can only calculate noise on signed-linear frames :(\n"); 01313 return 0; 01314 } 01315 s = f->data.ptr; 01316 len = f->datalen/2; 01317 return __ast_dsp_silence_noise(dsp, s, len, NULL, totalnoise); 01318 }
struct ast_frame* ast_dsp_process | ( | struct ast_channel * | chan, | |
struct ast_dsp * | dsp, | |||
struct ast_frame * | af | |||
) |
Return AST_FRAME_NULL frames when there is silence, AST_FRAME_BUSY on busies, and call progress, all dependent upon which features are enabled.
Definition at line 1321 of file dsp.c.
References __ast_dsp_silence_noise(), AST_ALAW, AST_FORMAT_ALAW, AST_FORMAT_SLINEAR, AST_FORMAT_TESTLAW, AST_FORMAT_ULAW, AST_FRAME_VOICE, ast_getformatname(), ast_log(), AST_MULAW, ast_frame_subclass::codec, ast_frame::data, ast_frame::datalen, ast_dsp::display_inband_dtmf_warning, DSP_FEATURE_BUSY_DETECT, DSP_FEATURE_SILENCE_SUPPRESS, ast_dsp::features, ast_frame::frametype, len(), LOG_WARNING, ast_dsp::mute_fragments, ast_frame::ptr, and ast_frame::subclass.
Referenced by dahdi_read(), mgcp_rtp_read(), oh323_rtp_read(), process_ast_dsp(), sip_rtp_read(), and usbradio_read().
01322 { 01323 int silence; 01324 int res; 01325 int digit = 0, fax_digit = 0; 01326 int x; 01327 short *shortdata; 01328 unsigned char *odata; 01329 int len; 01330 struct ast_frame *outf = NULL; 01331 01332 if (!af) { 01333 return NULL; 01334 } 01335 if (af->frametype != AST_FRAME_VOICE) { 01336 return af; 01337 } 01338 01339 odata = af->data.ptr; 01340 len = af->datalen; 01341 /* Make sure we have short data */ 01342 switch (af->subclass.codec) { 01343 case AST_FORMAT_SLINEAR: 01344 shortdata = af->data.ptr; 01345 len = af->datalen / 2; 01346 break; 01347 case AST_FORMAT_ULAW: 01348 case AST_FORMAT_TESTLAW: 01349 shortdata = alloca(af->datalen * 2); 01350 for (x = 0;x < len; x++) { 01351 shortdata[x] = AST_MULAW(odata[x]); 01352 } 01353 break; 01354 case AST_FORMAT_ALAW: 01355 shortdata = alloca(af->datalen * 2); 01356 for (x = 0; x < len; x++) { 01357 shortdata[x] = AST_ALAW(odata[x]); 01358 } 01359 break; 01360 default: 01361 /*Display warning only once. Otherwise you would get hundreds of warnings every second */ 01362 if (dsp->display_inband_dtmf_warning) 01363 ast_log(LOG_WARNING, "Inband DTMF is not supported on codec %s. Use RFC2833\n", ast_getformatname(af->subclass.codec)); 01364 dsp->display_inband_dtmf_warning = 0; 01365 return af; 01366 } 01367 01368 /* Initially we do not want to mute anything */ 01369 dsp->mute_fragments = 0; 01370 01371 /* Need to run the silence detection stuff for silence suppression and busy detection */ 01372 if ((dsp->features & DSP_FEATURE_SILENCE_SUPPRESS) || (dsp->features & DSP_FEATURE_BUSY_DETECT)) { 01373 res = __ast_dsp_silence_noise(dsp, shortdata, len, &silence, NULL); 01374 } 01375 01376 if ((dsp->features & DSP_FEATURE_SILENCE_SUPPRESS) && silence) { 01377 memset(&dsp->f, 0, sizeof(dsp->f)); 01378 dsp->f.frametype = AST_FRAME_NULL; 01379 ast_frfree(af); 01380 return ast_frisolate(&dsp->f); 01381 } 01382 if ((dsp->features & DSP_FEATURE_BUSY_DETECT) && ast_dsp_busydetect(dsp)) { 01383 chan->_softhangup |= AST_SOFTHANGUP_DEV; 01384 memset(&dsp->f, 0, sizeof(dsp->f)); 01385 dsp->f.frametype = AST_FRAME_CONTROL; 01386 /* Signal this as it was a channel hangup, to avoid msg "channel.c:3473 ast_waitfordigit_full: Unexpected control subclass '5'" */ 01387 dsp->f.subclass.integer = AST_CONTROL_HANGUP; 01388 ast_frfree(af); 01389 ast_debug(1, "Requesting Hangup because the busy tone was detected on channel %s\n", chan->name); 01390 return ast_frisolate(&dsp->f); 01391 } 01392 01393 if ((dsp->features & DSP_FEATURE_FAX_DETECT)) { 01394 if ((dsp->faxmode & DSP_FAXMODE_DETECT_CNG) && tone_detect(dsp, &dsp->cng_tone_state, shortdata, len)) { 01395 fax_digit = 'f'; 01396 } 01397 01398 if ((dsp->faxmode & DSP_FAXMODE_DETECT_CED) && tone_detect(dsp, &dsp->ced_tone_state, shortdata, len)) { 01399 fax_digit = 'e'; 01400 } 01401 } 01402 01403 if (dsp->features & (DSP_FEATURE_DIGIT_DETECT | DSP_FEATURE_BUSY_DETECT)) { 01404 if (dsp->digitmode & DSP_DIGITMODE_MF) 01405 digit = mf_detect(dsp, &dsp->digit_state, shortdata, len, (dsp->digitmode & DSP_DIGITMODE_NOQUELCH) == 0, (dsp->digitmode & DSP_DIGITMODE_RELAXDTMF)); 01406 else 01407 digit = dtmf_detect(dsp, &dsp->digit_state, shortdata, len, (dsp->digitmode & DSP_DIGITMODE_NOQUELCH) == 0, (dsp->digitmode & DSP_DIGITMODE_RELAXDTMF)); 01408 01409 if (dsp->digit_state.current_digits) { 01410 int event = 0, event_len = 0; 01411 char event_digit = 0; 01412 01413 if (!dsp->dtmf_began) { 01414 /* We have not reported DTMF_BEGIN for anything yet */ 01415 01416 if (dsp->features & DSP_FEATURE_DIGIT_DETECT) { 01417 event = AST_FRAME_DTMF_BEGIN; 01418 event_digit = dsp->digit_state.digits[0]; 01419 } 01420 dsp->dtmf_began = 1; 01421 01422 } else if (dsp->digit_state.current_digits > 1 || digit != dsp->digit_state.digits[0]) { 01423 /* Digit changed. This means digit we have reported with DTMF_BEGIN ended */ 01424 if (dsp->features & DSP_FEATURE_DIGIT_DETECT) { 01425 event = AST_FRAME_DTMF_END; 01426 event_digit = dsp->digit_state.digits[0]; 01427 event_len = dsp->digit_state.digitlen[0] * 1000 / SAMPLE_RATE; 01428 } 01429 memmove(&dsp->digit_state.digits[0], &dsp->digit_state.digits[1], dsp->digit_state.current_digits); 01430 memmove(&dsp->digit_state.digitlen[0], &dsp->digit_state.digitlen[1], dsp->digit_state.current_digits * sizeof(dsp->digit_state.digitlen[0])); 01431 dsp->digit_state.current_digits--; 01432 dsp->dtmf_began = 0; 01433 01434 if (dsp->features & DSP_FEATURE_BUSY_DETECT) { 01435 /* Reset Busy Detector as we have some confirmed activity */ 01436 memset(dsp->historicsilence, 0, sizeof(dsp->historicsilence)); 01437 memset(dsp->historicnoise, 0, sizeof(dsp->historicnoise)); 01438 ast_debug(1, "DTMF Detected - Reset busydetector\n"); 01439 } 01440 } 01441 01442 if (event) { 01443 memset(&dsp->f, 0, sizeof(dsp->f)); 01444 dsp->f.frametype = event; 01445 dsp->f.subclass.integer = event_digit; 01446 dsp->f.len = event_len; 01447 outf = &dsp->f; 01448 goto done; 01449 } 01450 } 01451 } 01452 01453 if (fax_digit) { 01454 /* Fax was detected - digit is either 'f' or 'e' */ 01455 01456 memset(&dsp->f, 0, sizeof(dsp->f)); 01457 dsp->f.frametype = AST_FRAME_DTMF; 01458 dsp->f.subclass.integer = fax_digit; 01459 outf = &dsp->f; 01460 goto done; 01461 } 01462 01463 if ((dsp->features & DSP_FEATURE_CALL_PROGRESS)) { 01464 res = __ast_dsp_call_progress(dsp, shortdata, len); 01465 if (res) { 01466 switch (res) { 01467 case AST_CONTROL_ANSWER: 01468 case AST_CONTROL_BUSY: 01469 case AST_CONTROL_RINGING: 01470 case AST_CONTROL_CONGESTION: 01471 case AST_CONTROL_HANGUP: 01472 memset(&dsp->f, 0, sizeof(dsp->f)); 01473 dsp->f.frametype = AST_FRAME_CONTROL; 01474 dsp->f.subclass.integer = res; 01475 dsp->f.src = "dsp_progress"; 01476 if (chan) 01477 ast_queue_frame(chan, &dsp->f); 01478 break; 01479 default: 01480 ast_log(LOG_WARNING, "Don't know how to represent call progress message %d\n", res); 01481 } 01482 } 01483 } else if ((dsp->features & DSP_FEATURE_WAITDIALTONE)) { 01484 res = __ast_dsp_call_progress(dsp, shortdata, len); 01485 } 01486 01487 done: 01488 /* Mute fragment of the frame */ 01489 for (x = 0; x < dsp->mute_fragments; x++) { 01490 memset(shortdata + dsp->mute_data[x].start, 0, sizeof(int16_t) * (dsp->mute_data[x].end - dsp->mute_data[x].start)); 01491 } 01492 01493 switch (af->subclass.codec) { 01494 case AST_FORMAT_SLINEAR: 01495 break; 01496 case AST_FORMAT_ULAW: 01497 for (x = 0; x < len; x++) { 01498 odata[x] = AST_LIN2MU((unsigned short) shortdata[x]); 01499 } 01500 break; 01501 case AST_FORMAT_ALAW: 01502 for (x = 0; x < len; x++) { 01503 odata[x] = AST_LIN2A((unsigned short) shortdata[x]); 01504 } 01505 break; 01506 } 01507 01508 if (outf) { 01509 if (chan) { 01510 ast_queue_frame(chan, af); 01511 } 01512 ast_frfree(af); 01513 return ast_frisolate(outf); 01514 } else { 01515 return af; 01516 } 01517 }
static void ast_dsp_prog_reset | ( | struct ast_dsp * | dsp | ) | [static] |
Definition at line 1519 of file dsp.c.
References ARRAY_LEN, ast_dsp::freqcount, ast_dsp::freqs, goertzel_init(), ast_dsp::gsamp_size, ast_dsp::gsamps, modes, ast_dsp::progmode, ast_dsp::ringtimeout, and progress::size.
Referenced by ast_dsp_new(), and ast_dsp_set_call_progress_zone().
01520 { 01521 int max = 0; 01522 int x; 01523 01524 dsp->gsamp_size = modes[dsp->progmode].size; 01525 dsp->gsamps = 0; 01526 for (x = 0; x < ARRAY_LEN(modes[dsp->progmode].freqs); x++) { 01527 if (modes[dsp->progmode].freqs[x]) { 01528 goertzel_init(&dsp->freqs[x], (float)modes[dsp->progmode].freqs[x], dsp->gsamp_size); 01529 max = x + 1; 01530 } 01531 } 01532 dsp->freqcount = max; 01533 dsp->ringtimeout= 0; 01534 }
int ast_dsp_reload | ( | void | ) |
Reloads dsp settings from dsp.conf.
Definition at line 1763 of file dsp.c.
References _dsp_init().
01764 { 01765 return _dsp_init(1); 01766 }
void ast_dsp_reset | ( | struct ast_dsp * | dsp | ) |
Reset total silence count.
Definition at line 1654 of file dsp.c.
References ast_dsp::freqs, ast_dsp::gsamps, ast_dsp::historicnoise, ast_dsp::historicsilence, ast_dsp::ringtimeout, ast_dsp::totalsilence, goertzel_state_t::v2, and goertzel_state_t::v3.
Referenced by debug_check_frame_for_silence().
01655 { 01656 int x; 01657 01658 dsp->totalsilence = 0; 01659 dsp->gsamps = 0; 01660 for (x = 0; x < 4; x++) { 01661 dsp->freqs[x].v2 = dsp->freqs[x].v3 = 0.0; 01662 } 01663 memset(dsp->historicsilence, 0, sizeof(dsp->historicsilence)); 01664 memset(dsp->historicnoise, 0, sizeof(dsp->historicnoise)); 01665 dsp->ringtimeout= 0; 01666 }
void ast_dsp_set_busy_compare | ( | struct ast_dsp * | dsp, | |
int | compare | |||
) |
Set if silence and noice lengths must be compared for busy.
Definition at line 1603 of file dsp.c.
References ast_dsp::busycompare.
Referenced by dahdi_new().
01604 { 01605 if (compare > 0) 01606 dsp->busycompare = 1; 01607 else 01608 dsp->busycompare = 0; 01609 }
void ast_dsp_set_busy_count | ( | struct ast_dsp * | dsp, | |
int | cadences | |||
) |
Set number of required cadences for busy.
Definition at line 1592 of file dsp.c.
References ast_dsp::busycount, and DSP_HISTORY.
Referenced by dahdi_new().
01593 { 01594 if (cadences < 4) { 01595 cadences = 4; 01596 } 01597 if (cadences > DSP_HISTORY) { 01598 cadences = DSP_HISTORY; 01599 } 01600 dsp->busycount = cadences; 01601 }
void ast_dsp_set_busy_pattern | ( | struct ast_dsp * | dsp, | |
int | tonelength, | |||
int | quietlength, | |||
int | fuzzy | |||
) |
Set expected lengths of the busy tones.
Definition at line 1611 of file dsp.c.
References ast_debug, ast_dsp::busy_pattern_fuzzy, ast_dsp::busy_quietlength, ast_dsp::busy_tonelength, and ast_dsp::busytoneonly.
Referenced by dahdi_new().
01612 { 01613 dsp->busy_tonelength = tonelength; 01614 if (quietlength > 0) 01615 dsp->busy_quietlength = quietlength; 01616 else 01617 dsp->busytoneonly = 1; 01618 ast_debug(1, "dsp busy pattern set to %d,%d\n", tonelength, quietlength); 01619 if( fuzzy > 0 && fuzzy < 50 ) 01620 dsp->busy_pattern_fuzzy = fuzzy; 01621 }
int ast_dsp_set_call_progress_zone | ( | struct ast_dsp * | dsp, | |
char * | zone | |||
) |
Set zone for doing progress detection.
Definition at line 1692 of file dsp.c.
References aliases, ARRAY_LEN, ast_dsp_prog_reset(), progalias::mode, name, and ast_dsp::progmode.
Referenced by dahdi_new().
01693 { 01694 int x; 01695 01696 for (x = 0; x < ARRAY_LEN(aliases); x++) { 01697 if (!strcasecmp(aliases[x].name, zone)) { 01698 dsp->progmode = aliases[x].mode; 01699 ast_dsp_prog_reset(dsp); 01700 return 0; 01701 } 01702 } 01703 return -1; 01704 }
int ast_dsp_set_digitmode | ( | struct ast_dsp * | dsp, | |
int | digitmode | |||
) |
Set digit mode.
Definition at line 1668 of file dsp.c.
References ast_digit_detect_init(), ast_dsp::digit_state, ast_dsp::digitmode, DSP_DIGITMODE_DTMF, DSP_DIGITMODE_MUTECONF, and DSP_DIGITMODE_MUTEMAX.
Referenced by analog_ss_thread(), dahdi_hangup(), dahdi_new(), dahdi_setoption(), enable_dsp_detect(), mgcp_new(), my_dsp_set_digitmode(), and store_config().
01669 { 01670 int new; 01671 int old; 01672 01673 old = dsp->digitmode & (DSP_DIGITMODE_DTMF | DSP_DIGITMODE_MF | DSP_DIGITMODE_MUTECONF | DSP_DIGITMODE_MUTEMAX); 01674 new = digitmode & (DSP_DIGITMODE_DTMF | DSP_DIGITMODE_MF | DSP_DIGITMODE_MUTECONF | DSP_DIGITMODE_MUTEMAX); 01675 if (old != new) { 01676 /* Must initialize structures if switching from MF to DTMF or vice-versa */ 01677 ast_digit_detect_init(&dsp->digit_state, new & DSP_DIGITMODE_MF); 01678 } 01679 dsp->digitmode = digitmode; 01680 return 0; 01681 }
int ast_dsp_set_faxmode | ( | struct ast_dsp * | dsp, | |
int | faxmode | |||
) |
Set fax mode.
Definition at line 1683 of file dsp.c.
References ast_fax_detect_init(), and ast_dsp::faxmode.
01684 { 01685 if (dsp->faxmode != faxmode) { 01686 ast_fax_detect_init(dsp); 01687 } 01688 dsp->faxmode = faxmode; 01689 return 0; 01690 }
void ast_dsp_set_features | ( | struct ast_dsp * | dsp, | |
int | features | |||
) |
Select feature set.
Definition at line 1571 of file dsp.c.
References ast_dsp::display_inband_dtmf_warning, DSP_FEATURE_DIGIT_DETECT, and ast_dsp::features.
Referenced by __oh323_new(), dahdi_handle_dtmf(), dahdi_new(), dahdi_read(), dahdi_setoption(), disable_dtmf_detect(), enable_dsp_detect(), enable_dtmf_detect(), mgcp_new(), misdn_set_opt_exec(), my_handle_dtmf(), my_pri_open_media(), read_config(), sip_rtp_read(), and store_config().
01572 { 01573 dsp->features = features; 01574 if (!(features & DSP_FEATURE_DIGIT_DETECT)) { 01575 dsp->display_inband_dtmf_warning = 0; 01576 } 01577 }
void ast_dsp_set_threshold | ( | struct ast_dsp * | dsp, | |
int | threshold | |||
) |
Set threshold value for silence.
Definition at line 1584 of file dsp.c.
References ast_dsp::threshold.
Referenced by __ast_play_and_record(), dahdi_new(), do_waiting(), fax_session_new(), handle_recordfile(), and isAnsweringMachine().
01585 { 01586 if (threshold < 256) 01587 dsp->threshold = 256; 01588 else 01589 dsp->threshold = threshold; 01590 }
Return non-zero if this is silence. Updates "totalsilence" with the total number of seconds of silence.
Definition at line 1284 of file dsp.c.
References __ast_dsp_silence_noise(), AST_FORMAT_SLINEAR, AST_FRAME_VOICE, ast_log(), f, len(), and LOG_WARNING.
Referenced by __ast_play_and_record(), background_detect_exec(), conf_run(), debug_check_frame_for_silence(), do_waiting(), handle_recordfile(), and isAnsweringMachine().
01285 { 01286 short *s; 01287 int len; 01288 01289 if (f->frametype != AST_FRAME_VOICE) { 01290 ast_log(LOG_WARNING, "Can't calculate silence on a non-voice frame\n"); 01291 return 0; 01292 } 01293 if (f->subclass.codec != AST_FORMAT_SLINEAR) { 01294 ast_log(LOG_WARNING, "Can only calculate silence on signed-linear frames :(\n"); 01295 return 0; 01296 } 01297 s = f->data.ptr; 01298 len = f->datalen/2; 01299 return __ast_dsp_silence_noise(dsp, s, len, totalsilence, NULL); 01300 }
int ast_dsp_was_muted | ( | struct ast_dsp * | dsp | ) |
Returns true if DSP code was muting any fragment of the last processed frame. Muting (squelching) happens when DSP code removes DTMF/MF/generic tones from the audio.
Definition at line 1706 of file dsp.c.
References ast_dsp::mute_fragments.
Referenced by dahdi_read().
01707 { 01708 return (dsp->mute_fragments > 0); 01709 }
static void ast_dtmf_detect_init | ( | dtmf_detect_state_t * | s | ) | [static] |
Definition at line 483 of file dsp.c.
References dtmf_detect_state_t::col_out, dtmf_detect_state_t::current_hit, dtmf_detect_state_t::current_sample, DTMF_GSIZE, DTMF_HITS_TO_BEGIN, DTMF_MISSES_TO_END, dtmf_detect_state_t::energy, goertzel_init(), dtmf_detect_state_t::hits, dtmf_detect_state_t::hits_to_begin, dtmf_detect_state_t::lasthit, dtmf_detect_state_t::misses, dtmf_detect_state_t::misses_to_end, and dtmf_detect_state_t::row_out.
Referenced by ast_digit_detect_init().
00484 { 00485 int i; 00486 00487 s->lasthit = 0; 00488 s->current_hit = 0; 00489 for (i = 0; i < 4; i++) { 00490 goertzel_init(&s->row_out[i], dtmf_row[i], DTMF_GSIZE); 00491 goertzel_init(&s->col_out[i], dtmf_col[i], DTMF_GSIZE); 00492 s->energy = 0.0; 00493 } 00494 s->current_sample = 0; 00495 s->hits = 0; 00496 s->misses = 0; 00497 00498 s->hits_to_begin = DTMF_HITS_TO_BEGIN; 00499 s->misses_to_end = DTMF_MISSES_TO_END; 00500 }
static void ast_fax_detect_init | ( | struct ast_dsp * | s | ) | [static] |
Definition at line 477 of file dsp.c.
References ast_tone_detect_init(), ast_dsp::ced_tone_state, ast_dsp::cng_tone_state, FAX_TONE_CED_DB, FAX_TONE_CED_DURATION, FAX_TONE_CED_FREQ, FAX_TONE_CNG_DB, FAX_TONE_CNG_DURATION, and FAX_TONE_CNG_FREQ.
Referenced by ast_dsp_new(), and ast_dsp_set_faxmode().
00478 { 00479 ast_tone_detect_init(&s->cng_tone_state, FAX_TONE_CNG_FREQ, FAX_TONE_CNG_DURATION, FAX_TONE_CNG_DB); 00480 ast_tone_detect_init(&s->ced_tone_state, FAX_TONE_CED_FREQ, FAX_TONE_CED_DURATION, FAX_TONE_CED_DB); 00481 }
static void ast_mf_detect_init | ( | mf_detect_state_t * | s | ) | [static] |
Definition at line 502 of file dsp.c.
References mf_detect_state_t::current_hit, mf_detect_state_t::current_sample, goertzel_init(), mf_detect_state_t::hits, and mf_detect_state_t::tone_out.
Referenced by ast_digit_detect_init().
00503 { 00504 int i; 00505 s->hits[0] = s->hits[1] = s->hits[2] = s->hits[3] = s->hits[4] = 0; 00506 for (i = 0; i < 6; i++) { 00507 goertzel_init (&s->tone_out[i], mf_tones[i], 160); 00508 } 00509 s->current_sample = 0; 00510 s->current_hit = 0; 00511 }
static void ast_tone_detect_init | ( | tone_detect_state_t * | s, | |
int | freq, | |||
int | duration, | |||
int | amp | |||
) | [static] |
Definition at line 417 of file dsp.c.
References ast_debug, tone_detect_state_t::block_size, tone_detect_state_t::energy, tone_detect_state_t::freq, goertzel_init(), tone_detect_state_t::hit_count, tone_detect_state_t::hits_required, tone_detect_state_t::last_hit, SAMPLE_RATE, SAMPLES_IN_FRAME, tone_detect_state_t::samples_pending, tone_detect_state_t::squelch, tone_detect_state_t::threshold, and tone_detect_state_t::tone.
Referenced by ast_fax_detect_init().
00418 { 00419 int duration_samples; 00420 float x; 00421 int periods_in_block; 00422 00423 s->freq = freq; 00424 00425 /* Desired tone duration in samples */ 00426 duration_samples = duration * SAMPLE_RATE / 1000; 00427 /* We want to allow 10% deviation of tone duration */ 00428 duration_samples = duration_samples * 9 / 10; 00429 00430 /* If we want to remove tone, it is important to have block size not 00431 to exceed frame size. Otherwise by the moment tone is detected it is too late 00432 to squelch it from previous frames */ 00433 s->block_size = SAMPLES_IN_FRAME; 00434 00435 periods_in_block = s->block_size * freq / SAMPLE_RATE; 00436 00437 /* Make sure we will have at least 5 periods at target frequency for analisys. 00438 This may make block larger than expected packet and will make squelching impossible 00439 but at least we will be detecting the tone */ 00440 if (periods_in_block < 5) 00441 periods_in_block = 5; 00442 00443 /* Now calculate final block size. It will contain integer number of periods */ 00444 s->block_size = periods_in_block * SAMPLE_RATE / freq; 00445 00446 /* tone_detect is currently only used to detect fax tones and we 00447 do not need suqlching the fax tones */ 00448 s->squelch = 0; 00449 00450 /* Account for the first and the last block to be incomplete 00451 and thus no tone will be detected in them */ 00452 s->hits_required = (duration_samples - (s->block_size - 1)) / s->block_size; 00453 00454 goertzel_init(&s->tone, freq, s->block_size); 00455 00456 s->samples_pending = s->block_size; 00457 s->hit_count = 0; 00458 s->last_hit = 0; 00459 s->energy = 0.0; 00460 00461 /* We want tone energy to be amp decibels above the rest of the signal (the noise). 00462 According to Parseval's theorem the energy computed in time domain equals to energy 00463 computed in frequency domain. So subtracting energy in the frequency domain (Goertzel result) 00464 from the energy in the time domain we will get energy of the remaining signal (without the tone 00465 we are detecting). We will be checking that 00466 10*log(Ew / (Et - Ew)) > amp 00467 Calculate threshold so that we will be actually checking 00468 Ew > Et * threshold 00469 */ 00470 00471 x = pow(10.0, amp / 10.0); 00472 s->threshold = x / (x + 1); 00473 00474 ast_debug(1, "Setup tone %d Hz, %d ms, block_size=%d, hits_required=%d\n", freq, duration, s->block_size, s->hits_required); 00475 }
static int dtmf_detect | ( | struct ast_dsp * | dsp, | |
digit_detect_state_t * | s, | |||
int16_t | amp[], | |||
int | samples, | |||
int | squelch, | |||
int | relax | |||
) | [static] |
Definition at line 644 of file dsp.c.
References dtmf_detect_state_t::col_out, dtmf_detect_state_t::current_sample, digit_detect_state_t::dtmf, DTMF_GSIZE, DTMF_NORMAL_TWIST, DTMF_RELATIVE_PEAK_COL, DTMF_RELATIVE_PEAK_ROW, DTMF_REVERSE_TWIST, DTMF_THRESHOLD, DTMF_TO_TOTAL_ENERGY, dtmf_detect_state_t::energy, goertzel_result(), goertzel_sample(), mute, dtmf_detect_state_t::mute_samples, dtmf_detect_state_t::row_out, and digit_detect_state_t::td.
00645 { 00646 float row_energy[4]; 00647 float col_energy[4]; 00648 float famp; 00649 int i; 00650 int j; 00651 int sample; 00652 int best_row; 00653 int best_col; 00654 int hit; 00655 int limit; 00656 fragment_t mute = {0, 0}; 00657 00658 if (squelch && s->td.dtmf.mute_samples > 0) { 00659 mute.end = (s->td.dtmf.mute_samples < samples) ? s->td.dtmf.mute_samples : samples; 00660 s->td.dtmf.mute_samples -= mute.end; 00661 } 00662 00663 hit = 0; 00664 for (sample = 0; sample < samples; sample = limit) { 00665 /* DTMF_GSIZE is optimised to meet the DTMF specs. */ 00666 if ((samples - sample) >= (DTMF_GSIZE - s->td.dtmf.current_sample)) { 00667 limit = sample + (DTMF_GSIZE - s->td.dtmf.current_sample); 00668 } else { 00669 limit = samples; 00670 } 00671 /* The following unrolled loop takes only 35% (rough estimate) of the 00672 time of a rolled loop on the machine on which it was developed */ 00673 for (j = sample; j < limit; j++) { 00674 famp = amp[j]; 00675 s->td.dtmf.energy += famp*famp; 00676 /* With GCC 2.95, the following unrolled code seems to take about 35% 00677 (rough estimate) as long as a neat little 0-3 loop */ 00678 goertzel_sample(s->td.dtmf.row_out, amp[j]); 00679 goertzel_sample(s->td.dtmf.col_out, amp[j]); 00680 goertzel_sample(s->td.dtmf.row_out + 1, amp[j]); 00681 goertzel_sample(s->td.dtmf.col_out + 1, amp[j]); 00682 goertzel_sample(s->td.dtmf.row_out + 2, amp[j]); 00683 goertzel_sample(s->td.dtmf.col_out + 2, amp[j]); 00684 goertzel_sample(s->td.dtmf.row_out + 3, amp[j]); 00685 goertzel_sample(s->td.dtmf.col_out + 3, amp[j]); 00686 } 00687 s->td.dtmf.current_sample += (limit - sample); 00688 if (s->td.dtmf.current_sample < DTMF_GSIZE) { 00689 continue; 00690 } 00691 /* We are at the end of a DTMF detection block */ 00692 /* Find the peak row and the peak column */ 00693 row_energy[0] = goertzel_result (&s->td.dtmf.row_out[0]); 00694 col_energy[0] = goertzel_result (&s->td.dtmf.col_out[0]); 00695 00696 for (best_row = best_col = 0, i = 1; i < 4; i++) { 00697 row_energy[i] = goertzel_result (&s->td.dtmf.row_out[i]); 00698 if (row_energy[i] > row_energy[best_row]) { 00699 best_row = i; 00700 } 00701 col_energy[i] = goertzel_result (&s->td.dtmf.col_out[i]); 00702 if (col_energy[i] > col_energy[best_col]) { 00703 best_col = i; 00704 } 00705 } 00706 hit = 0; 00707 /* Basic signal level test and the twist test */ 00708 if (row_energy[best_row] >= DTMF_THRESHOLD && 00709 col_energy[best_col] >= DTMF_THRESHOLD && 00710 col_energy[best_col] < row_energy[best_row] * DTMF_REVERSE_TWIST && 00711 col_energy[best_col] * DTMF_NORMAL_TWIST > row_energy[best_row]) { 00712 /* Relative peak test */ 00713 for (i = 0; i < 4; i++) { 00714 if ((i != best_col && 00715 col_energy[i] * DTMF_RELATIVE_PEAK_COL > col_energy[best_col]) || 00716 (i != best_row 00717 && row_energy[i] * DTMF_RELATIVE_PEAK_ROW > row_energy[best_row])) { 00718 break; 00719 } 00720 } 00721 /* ... and fraction of total energy test */ 00722 if (i >= 4 && 00723 (row_energy[best_row] + col_energy[best_col]) > DTMF_TO_TOTAL_ENERGY * s->td.dtmf.energy) { 00724 /* Got a hit */ 00725 hit = dtmf_positions[(best_row << 2) + best_col]; 00726 } 00727 } 00728 00729 if (hit == s->td.dtmf.lasthit) { 00730 if (s->td.dtmf.current_hit) { 00731 /* We are in the middle of a digit already */ 00732 if (hit) { 00733 if (hit != s->td.dtmf.current_hit) { 00734 /* Look for a start of a new digit. 00735 This is because hits_to_begin may be smaller than misses_to_end 00736 and we may find the beginning of new digit before we consider last one ended. */ 00737 s->td.dtmf.current_hit = 0; 00738 } else { 00739 /* Current hit was same as last, so increment digit duration (of last digit) */ 00740 s->digitlen[s->current_digits - 1] += DTMF_GSIZE; 00741 } 00742 } else { 00743 /* No Digit */ 00744 s->td.dtmf.misses++; 00745 if (s->td.dtmf.misses == s->td.dtmf.misses_to_end) { 00746 /* There were enough misses to consider digit ended */ 00747 s->td.dtmf.current_hit = 0; 00748 } 00749 } 00750 } else if (hit) { 00751 /* Detecting new digit */ 00752 s->td.dtmf.hits++; 00753 if (s->td.dtmf.hits == s->td.dtmf.hits_to_begin) { 00754 store_digit(s, hit); 00755 s->td.dtmf.current_hit = hit; 00756 } 00757 } 00758 } else { 00759 s->td.dtmf.hits = 1; 00760 s->td.dtmf.misses = 1; 00761 s->td.dtmf.lasthit = hit; 00762 } 00763 00764 /* If we had a hit in this block, include it into mute fragment */ 00765 if (squelch && hit) { 00766 if (mute.end < sample - DTMF_GSIZE) { 00767 /* There is a gap between fragments */ 00768 mute_fragment(dsp, &mute); 00769 mute.start = (sample > DTMF_GSIZE) ? (sample - DTMF_GSIZE) : 0; 00770 } 00771 mute.end = limit + DTMF_GSIZE; 00772 } 00773 00774 /* Reinitialise the detector for the next block */ 00775 for (i = 0; i < 4; i++) { 00776 goertzel_reset(&s->td.dtmf.row_out[i]); 00777 goertzel_reset(&s->td.dtmf.col_out[i]); 00778 } 00779 s->td.dtmf.energy = 0.0; 00780 s->td.dtmf.current_sample = 0; 00781 } 00782 00783 if (squelch && mute.end) { 00784 if (mute.end > samples) { 00785 s->td.dtmf.mute_samples = mute.end - samples; 00786 mute.end = samples; 00787 } 00788 mute_fragment(dsp, &mute); 00789 } 00790 00791 return (s->td.dtmf.current_hit); /* return the debounced hit */ 00792 }
static void goertzel_init | ( | goertzel_state_t * | s, | |
float | freq, | |||
int | samples | |||
) | [inline, static] |
Definition at line 342 of file dsp.c.
References goertzel_state_t::chunky, cos, goertzel_state_t::fac, M_PI, SAMPLE_RATE, goertzel_state_t::samples, goertzel_state_t::v2, and goertzel_state_t::v3.
Referenced by ast_dsp_prog_reset(), ast_dtmf_detect_init(), ast_mf_detect_init(), and ast_tone_detect_init().
00343 { 00344 s->v2 = s->v3 = s->chunky = 0.0; 00345 s->fac = (int)(32768.0 * 2.0 * cos(2.0 * M_PI * freq / SAMPLE_RATE)); 00346 s->samples = samples; 00347 }
static void goertzel_reset | ( | goertzel_state_t * | s | ) | [inline, static] |
Definition at line 349 of file dsp.c.
References goertzel_state_t::chunky, goertzel_state_t::v2, and goertzel_state_t::v3.
Referenced by ast_dsp_digitreset().
static float goertzel_result | ( | goertzel_state_t * | s | ) | [inline, static] |
Definition at line 333 of file dsp.c.
References goertzel_state_t::chunky, goertzel_state_t::fac, goertzel_result_t::power, goertzel_state_t::v2, goertzel_state_t::v3, and goertzel_result_t::value.
Referenced by __ast_dsp_call_progress(), and dtmf_detect().
00334 { 00335 goertzel_result_t r; 00336 r.value = (s->v3 * s->v3) + (s->v2 * s->v2); 00337 r.value -= ((s->v2 * s->v3) >> 15) * s->fac; 00338 r.power = s->chunky * 2; 00339 return (float)r.value * (float)(1 << r.power); 00340 }
static void goertzel_sample | ( | goertzel_state_t * | s, | |
short | sample | |||
) | [inline, static] |
Definition at line 306 of file dsp.c.
References goertzel_state_t::chunky, goertzel_state_t::fac, goertzel_state_t::v2, and goertzel_state_t::v3.
Referenced by __ast_dsp_call_progress(), dtmf_detect(), goertzel_update(), mf_detect(), and tone_detect().
00307 { 00308 int v1; 00309 00310 v1 = s->v2; 00311 s->v2 = s->v3; 00312 00313 s->v3 = (s->fac * s->v2) >> 15; 00314 s->v3 = s->v3 - v1 + (sample >> s->chunky); 00315 if (abs(s->v3) > 32768) { 00316 s->chunky++; 00317 s->v3 = s->v3 >> 1; 00318 s->v2 = s->v2 >> 1; 00319 v1 = v1 >> 1; 00320 } 00321 }
static void goertzel_update | ( | goertzel_state_t * | s, | |
short * | samps, | |||
int | count | |||
) | [inline, static] |
Definition at line 323 of file dsp.c.
References goertzel_sample().
00324 { 00325 int i; 00326 00327 for (i = 0; i < count; i++) { 00328 goertzel_sample(s, samps[i]); 00329 } 00330 }
static int mf_detect | ( | struct ast_dsp * | dsp, | |
digit_detect_state_t * | s, | |||
int16_t | amp[], | |||
int | samples, | |||
int | squelch, | |||
int | relax | |||
) | [static] |
Definition at line 794 of file dsp.c.
References mf_detect_state_t::current_sample, goertzel_sample(), digit_detect_state_t::mf, MF_GSIZE, mute, mf_detect_state_t::mute_samples, digit_detect_state_t::td, and mf_detect_state_t::tone_out.
00796 { 00797 float energy[6]; 00798 int best; 00799 int second_best; 00800 int i; 00801 int j; 00802 int sample; 00803 int hit; 00804 int limit; 00805 fragment_t mute = {0, 0}; 00806 00807 if (squelch && s->td.mf.mute_samples > 0) { 00808 mute.end = (s->td.mf.mute_samples < samples) ? s->td.mf.mute_samples : samples; 00809 s->td.mf.mute_samples -= mute.end; 00810 } 00811 00812 hit = 0; 00813 for (sample = 0; sample < samples; sample = limit) { 00814 /* 80 is optimised to meet the MF specs. */ 00815 /* XXX So then why is MF_GSIZE defined as 120? */ 00816 if ((samples - sample) >= (MF_GSIZE - s->td.mf.current_sample)) { 00817 limit = sample + (MF_GSIZE - s->td.mf.current_sample); 00818 } else { 00819 limit = samples; 00820 } 00821 /* The following unrolled loop takes only 35% (rough estimate) of the 00822 time of a rolled loop on the machine on which it was developed */ 00823 for (j = sample; j < limit; j++) { 00824 /* With GCC 2.95, the following unrolled code seems to take about 35% 00825 (rough estimate) as long as a neat little 0-3 loop */ 00826 goertzel_sample(s->td.mf.tone_out, amp[j]); 00827 goertzel_sample(s->td.mf.tone_out + 1, amp[j]); 00828 goertzel_sample(s->td.mf.tone_out + 2, amp[j]); 00829 goertzel_sample(s->td.mf.tone_out + 3, amp[j]); 00830 goertzel_sample(s->td.mf.tone_out + 4, amp[j]); 00831 goertzel_sample(s->td.mf.tone_out + 5, amp[j]); 00832 } 00833 s->td.mf.current_sample += (limit - sample); 00834 if (s->td.mf.current_sample < MF_GSIZE) { 00835 continue; 00836 } 00837 /* We're at the end of an MF detection block. */ 00838 /* Find the two highest energies. The spec says to look for 00839 two tones and two tones only. Taking this literally -ie 00840 only two tones pass the minimum threshold - doesn't work 00841 well. The sinc function mess, due to rectangular windowing 00842 ensure that! Find the two highest energies and ensure they 00843 are considerably stronger than any of the others. */ 00844 energy[0] = goertzel_result(&s->td.mf.tone_out[0]); 00845 energy[1] = goertzel_result(&s->td.mf.tone_out[1]); 00846 if (energy[0] > energy[1]) { 00847 best = 0; 00848 second_best = 1; 00849 } else { 00850 best = 1; 00851 second_best = 0; 00852 } 00853 /*endif*/ 00854 for (i = 2; i < 6; i++) { 00855 energy[i] = goertzel_result(&s->td.mf.tone_out[i]); 00856 if (energy[i] >= energy[best]) { 00857 second_best = best; 00858 best = i; 00859 } else if (energy[i] >= energy[second_best]) { 00860 second_best = i; 00861 } 00862 } 00863 /* Basic signal level and twist tests */ 00864 hit = 0; 00865 if (energy[best] >= BELL_MF_THRESHOLD && energy[second_best] >= BELL_MF_THRESHOLD 00866 && energy[best] < energy[second_best]*BELL_MF_TWIST 00867 && energy[best] * BELL_MF_TWIST > energy[second_best]) { 00868 /* Relative peak test */ 00869 hit = -1; 00870 for (i = 0; i < 6; i++) { 00871 if (i != best && i != second_best) { 00872 if (energy[i]*BELL_MF_RELATIVE_PEAK >= energy[second_best]) { 00873 /* The best two are not clearly the best */ 00874 hit = 0; 00875 break; 00876 } 00877 } 00878 } 00879 } 00880 if (hit) { 00881 /* Get the values into ascending order */ 00882 if (second_best < best) { 00883 i = best; 00884 best = second_best; 00885 second_best = i; 00886 } 00887 best = best * 5 + second_best - 1; 00888 hit = bell_mf_positions[best]; 00889 /* Look for two successive similar results */ 00890 /* The logic in the next test is: 00891 For KP we need 4 successive identical clean detects, with 00892 two blocks of something different preceeding it. For anything 00893 else we need two successive identical clean detects, with 00894 two blocks of something different preceeding it. */ 00895 if (hit == s->td.mf.hits[4] && hit == s->td.mf.hits[3] && 00896 ((hit != '*' && hit != s->td.mf.hits[2] && hit != s->td.mf.hits[1])|| 00897 (hit == '*' && hit == s->td.mf.hits[2] && hit != s->td.mf.hits[1] && 00898 hit != s->td.mf.hits[0]))) { 00899 store_digit(s, hit); 00900 } 00901 } 00902 00903 00904 if (hit != s->td.mf.hits[4] && hit != s->td.mf.hits[3]) { 00905 /* Two successive block without a hit terminate current digit */ 00906 s->td.mf.current_hit = 0; 00907 } 00908 00909 s->td.mf.hits[0] = s->td.mf.hits[1]; 00910 s->td.mf.hits[1] = s->td.mf.hits[2]; 00911 s->td.mf.hits[2] = s->td.mf.hits[3]; 00912 s->td.mf.hits[3] = s->td.mf.hits[4]; 00913 s->td.mf.hits[4] = hit; 00914 00915 /* If we had a hit in this block, include it into mute fragment */ 00916 if (squelch && hit) { 00917 if (mute.end < sample - MF_GSIZE) { 00918 /* There is a gap between fragments */ 00919 mute_fragment(dsp, &mute); 00920 mute.start = (sample > MF_GSIZE) ? (sample - MF_GSIZE) : 0; 00921 } 00922 mute.end = limit + DTMF_GSIZE; 00923 } 00924 00925 /* Reinitialise the detector for the next block */ 00926 for (i = 0; i < 6; i++) 00927 goertzel_reset(&s->td.mf.tone_out[i]); 00928 s->td.mf.current_sample = 0; 00929 } 00930 00931 if (squelch && mute.end) { 00932 if (mute.end > samples) { 00933 s->td.mf.mute_samples = mute.end - samples; 00934 mute.end = samples; 00935 } 00936 mute_fragment(dsp, &mute); 00937 } 00938 00939 return (s->td.mf.current_hit); /* return the debounced hit */ 00940 }
static void mute_fragment | ( | struct ast_dsp * | dsp, | |
fragment_t * | fragment | |||
) | [static] |
Definition at line 407 of file dsp.c.
References ARRAY_LEN, ast_log(), LOG_ERROR, ast_dsp::mute_data, and ast_dsp::mute_fragments.
00408 { 00409 if (dsp->mute_fragments >= ARRAY_LEN(dsp->mute_data)) { 00410 ast_log(LOG_ERROR, "Too many fragments to mute. Ignoring\n"); 00411 return; 00412 } 00413 00414 dsp->mute_data[dsp->mute_fragments++] = *fragment; 00415 }
static int pair_there | ( | float | p1, | |
float | p2, | |||
float | i1, | |||
float | i2, | |||
float | e | |||
) | [inline, static] |
Definition at line 942 of file dsp.c.
References TONE_MIN_THRESH, and TONE_THRESH.
Referenced by __ast_dsp_call_progress().
00943 { 00944 /* See if p1 and p2 are there, relative to i1 and i2 and total energy */ 00945 /* Make sure absolute levels are high enough */ 00946 if ((p1 < TONE_MIN_THRESH) || (p2 < TONE_MIN_THRESH)) { 00947 return 0; 00948 } 00949 /* Amplify ignored stuff */ 00950 i2 *= TONE_THRESH; 00951 i1 *= TONE_THRESH; 00952 e *= TONE_THRESH; 00953 /* Check first tone */ 00954 if ((p1 < i1) || (p1 < i2) || (p1 < e)) { 00955 return 0; 00956 } 00957 /* And second */ 00958 if ((p2 < i1) || (p2 < i2) || (p2 < e)) { 00959 return 0; 00960 } 00961 /* Guess it's there... */ 00962 return 1; 00963 }
static void store_digit | ( | digit_detect_state_t * | s, | |
char | digit | |||
) | [static] |
Definition at line 631 of file dsp.c.
References ast_log(), digit_detect_state_t::current_digits, digit_detect_state_t::detected_digits, digit_detect_state_t::digitlen, digit_detect_state_t::digits, LOG_WARNING, digit_detect_state_t::lost_digits, and MAX_DTMF_DIGITS.
00632 { 00633 s->detected_digits++; 00634 if (s->current_digits < MAX_DTMF_DIGITS) { 00635 s->digitlen[s->current_digits] = 0; 00636 s->digits[s->current_digits++] = digit; 00637 s->digits[s->current_digits] = '\0'; 00638 } else { 00639 ast_log(LOG_WARNING, "Digit lost due to full buffer\n"); 00640 s->lost_digits++; 00641 } 00642 }
static int tone_detect | ( | struct ast_dsp * | dsp, | |
tone_detect_state_t * | s, | |||
int16_t * | amp, | |||
int | samples | |||
) | [static] |
Definition at line 527 of file dsp.c.
References tone_detect_state_t::energy, goertzel_sample(), mute, tone_detect_state_t::mute_samples, ast_frame::ptr, tone_detect_state_t::samples_pending, tone_detect_state_t::squelch, and tone_detect_state_t::tone.
00528 { 00529 float tone_energy; 00530 int i; 00531 int hit = 0; 00532 int limit; 00533 int res = 0; 00534 int16_t *ptr; 00535 int start, end; 00536 fragment_t mute = {0, 0}; 00537 00538 if (s->squelch && s->mute_samples > 0) { 00539 mute.end = (s->mute_samples < samples) ? s->mute_samples : samples; 00540 s->mute_samples -= mute.end; 00541 } 00542 00543 for (start = 0; start < samples; start = end) { 00544 /* Process in blocks. */ 00545 limit = samples - start; 00546 if (limit > s->samples_pending) { 00547 limit = s->samples_pending; 00548 } 00549 end = start + limit; 00550 00551 for (i = limit, ptr = amp ; i > 0; i--, ptr++) { 00552 /* signed 32 bit int should be enough to suqare any possible signed 16 bit value */ 00553 s->energy += (int32_t) *ptr * (int32_t) *ptr; 00554 00555 goertzel_sample(&s->tone, *ptr); 00556 } 00557 00558 s->samples_pending -= limit; 00559 00560 if (s->samples_pending) { 00561 /* Finished incomplete (last) block */ 00562 break; 00563 } 00564 00565 tone_energy = goertzel_result(&s->tone); 00566 00567 /* Scale to make comparable */ 00568 tone_energy *= 2.0; 00569 s->energy *= s->block_size; 00570 00571 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)); 00572 hit = 0; 00573 if (tone_energy > s->energy * s->threshold) { 00574 ast_debug(10, "Hit! count=%d\n", s->hit_count); 00575 hit = 1; 00576 } 00577 00578 if (s->hit_count) { 00579 s->hit_count++; 00580 } 00581 00582 if (hit == s->last_hit) { 00583 if (!hit) { 00584 /* Two successive misses. Tone ended */ 00585 s->hit_count = 0; 00586 } else if (!s->hit_count) { 00587 s->hit_count++; 00588 } 00589 00590 } 00591 00592 if (s->hit_count == s->hits_required) { 00593 ast_debug(1, "%d Hz done detected\n", s->freq); 00594 res = 1; 00595 } 00596 00597 s->last_hit = hit; 00598 00599 /* If we had a hit in this block, include it into mute fragment */ 00600 if (s->squelch && hit) { 00601 if (mute.end < start - s->block_size) { 00602 /* There is a gap between fragments */ 00603 mute_fragment(dsp, &mute); 00604 mute.start = (start > s->block_size) ? (start - s->block_size) : 0; 00605 } 00606 mute.end = end + s->block_size; 00607 } 00608 00609 /* Reinitialise the detector for the next block */ 00610 /* Reset for the next block */ 00611 goertzel_reset(&s->tone); 00612 00613 /* Advance to the next block */ 00614 s->energy = 0.0; 00615 s->samples_pending = s->block_size; 00616 00617 amp += limit; 00618 } 00619 00620 if (s->squelch && mute.end) { 00621 if (mute.end > samples) { 00622 s->mute_samples = mute.end - samples; 00623 mute.end = samples; 00624 } 00625 mute_fragment(dsp, &mute); 00626 } 00627 00628 return res; 00629 }
Referenced by ast_dsp_set_call_progress_zone(), and find_alias().
const char bell_mf_positions[] = "1247C-358A--69*---0B----#" [static] |
const int DEFAULT_SILENCE_THRESHOLD = 256 [static] |
const float dtmf_col[] [static] |
const char dtmf_positions[] = "123A" "456B" "789C" "*0#D" [static] |
const float dtmf_row[] [static] |
const float mf_tones[] [static] |
Referenced by ast_dsp_prog_reset(), and setrem().
int thresholds[THRESHOLD_MAX] [static] |