dsp.c File Reference

Convenience Signal Processing routines. More...

#include "asterisk.h"
#include <math.h>
#include "asterisk/module.h"
#include "asterisk/frame.h"
#include "asterisk/format_cache.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"
#include "asterisk/test.h"

Include dependency graph for dsp.c:

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

Macros
#define	BELL_MF_RELATIVE_PEAK   12.6 /* 11dB */
#define	BELL_MF_THRESHOLD   1.6e9
#define	BELL_MF_TWIST   4.0 /* 6dB */
#define	CONFIG_FILE_NAME   "dsp.conf"
#define	DEF_DTMF_HITS_TO_BEGIN   2
#define	DEF_DTMF_MISSES_TO_END   3
#define	DEF_DTMF_NORMAL_TWIST   6.31 /* 8.0dB */
#define	DEF_DTMF_REVERSE_TWIST   2.51 /* 4.01dB */
#define	DEF_RELAX_DTMF_NORMAL_TWIST   6.31 /* 8.0dB */
#define	DEF_RELAX_DTMF_REVERSE_TWIST   3.98 /* 6.0dB */
#define	DEFAULT_SAMPLE_RATE   8000
#define	DEFAULT_THRESHOLD   512
	Default minimum average magnitude threshold to determine talking/noise by the DSP.
#define	DSP_HISTORY   15
#define	DTMF_GSIZE   102
#define	DTMF_MATRIX_SIZE   4
#define	DTMF_RELATIVE_PEAK_COL   6.3 /* 8dB */
#define	DTMF_RELATIVE_PEAK_ROW   6.3 /* 8dB */
#define	DTMF_THRESHOLD   8.0e7
#define	DTMF_TO_TOTAL_ENERGY   42.0
#define	FAX_TONE_CED_DB   16
#define	FAX_TONE_CED_DURATION   2600 /* ms */
#define	FAX_TONE_CED_FREQ   2100
#define	FAX_TONE_CNG_DB   16
#define	FAX_TONE_CNG_DURATION   500 /* ms */
#define	FAX_TONE_CNG_FREQ   1100
#define	FREQ_ARRAY_SIZE   7
#define	MAX_DTMF_DIGITS   128
#define	MF_GSIZE   120
#define	TONE_MIN_THRESH   1e8
#define	TONE_THRESH   10.0
#define	TONE_THRESHOLD   7.8e7

Enumerations
enum	busy_detect {   BUSY_PERCENT = 10 , BUSY_PAT_PERCENT = 7 , BUSY_THRESHOLD = 100 , BUSY_MIN = 75 ,   BUSY_MAX = 3100 }
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 struct ast_dsp *	__ast_dsp_new (unsigned int sample_rate)
static int	__ast_dsp_silence_noise (struct ast_dsp *dsp, short *s, int len, int *totalsilence, int *totalnoise, int *frames_energy)
static void	__reg_module (void)
static void	__unreg_module (void)
static int	_dsp_init (int reload)
static void	ast_digit_detect_init (digit_detect_state_t *s, int mf, unsigned int sample_rate)
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_features (struct ast_dsp *dsp)
	Get features.
unsigned int	ast_dsp_get_sample_rate (const struct ast_dsp *dsp)
	Retrieve the sample rate this DSP structure was created with.
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)
struct ast_dsp *	ast_dsp_new (void)
	Allocates a new dsp, assumes 8khz for internal sample rate.
struct ast_dsp *	ast_dsp_new_with_rate (unsigned int sample_rate)
	Allocates a new dsp with a specific internal sample rate used during processing.
int	ast_dsp_noise (struct ast_dsp *dsp, struct ast_frame *f, int *totalnoise)
	Process the audio frame for noise.
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.
static void	ast_dsp_prog_reset (struct ast_dsp *dsp)
void	ast_dsp_reset (struct ast_dsp *dsp)
	Reset total silence count.
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, const struct ast_dsp_busy_pattern *cadence)
	Set expected lengths of the busy tone.
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.
int	ast_dsp_set_freqmode (struct ast_dsp *dsp, int freq, int dur, int db, int squelch)
	Set arbitrary frequency detection mode.
void	ast_dsp_set_threshold (struct ast_dsp *dsp, int threshold)
	Set the minimum average magnitude threshold to determine talking by the DSP.
int	ast_dsp_silence (struct ast_dsp *dsp, struct ast_frame *f, int *totalsilence)
	Process the audio frame for silence.
static int	ast_dsp_silence_noise_with_energy (struct ast_dsp *dsp, struct ast_frame *f, int *total, int *frames_energy, int noise)
int	ast_dsp_silence_with_energy (struct ast_dsp *dsp, struct ast_frame *f, int *totalsilence, int *frames_energy)
	Process the audio frame for 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, unsigned int sample_rate)
static void	ast_fax_detect_init (struct ast_dsp *s)
static void	ast_freq_detect_init (struct ast_dsp *s, int freq, int dur, int db, int squelch)
static void	ast_mf_detect_init (mf_detect_state_t *s, unsigned int sample_rate)
struct ast_module *	AST_MODULE_SELF_SYM (void)
static void	ast_tone_detect_init (tone_detect_state_t *s, int freq, int duration, int amp, unsigned int sample_rate)
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, unsigned int sample_rate)
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 int	load_module (void)
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 int	reload_module (void)
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)
static int	unload_module (void)

Variables
static struct ast_module_info	__mod_info = { .name = AST_MODULE, .flags = AST_MODFLAG_GLOBAL_SYMBOLS | AST_MODFLAG_LOAD_ORDER , .description = "DSP" , .key = ASTERISK_GPL_KEY , .buildopt_sum = AST_BUILDOPT_SUM, .support_level = AST_MODULE_SUPPORT_CORE, .load = load_module, .unload = unload_module, .reload = reload_module, .load_pri = AST_MODPRI_CORE, .requires = "extconfig", }
static struct progalias	aliases []
static const struct ast_module_info *	ast_module_info = &__mod_info
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 int	dtmf_hits_to_begin
static int	dtmf_misses_to_end
static float	dtmf_normal_twist
static const char	dtmf_positions [] = "123A" "456B" "789C" "*0#D"
static float	dtmf_reverse_twist
static const float	dtmf_row []
static const float	mf_tones []
static struct progress	modes []
static float	relax_dtmf_normal_twist
static float	relax_dtmf_reverse_twist
static int	thresholds [THRESHOLD_MAX]

Detailed Description

Convenience Signal Processing routines.

Author

Mark Spencer marks.nosp@m.ter@.nosp@m.digiu.nosp@m.m.co.nosp@m.m

Steve Underwood steve.nosp@m.u@co.nosp@m.ppice.nosp@m..org

Definition in file dsp.c.

Macro Definition Documentation

BELL_MF_RELATIVE_PEAK

#define BELL_MF_RELATIVE_PEAK   12.6 /* 11dB */

Definition at line 202 of file dsp.c.

BELL_MF_THRESHOLD

#define BELL_MF_THRESHOLD   1.6e9

Definition at line 200 of file dsp.c.

BELL_MF_TWIST

#define BELL_MF_TWIST   4.0 /* 6dB */

Definition at line 201 of file dsp.c.

CONFIG_FILE_NAME

#define CONFIG_FILE_NAME   "dsp.conf"

Definition at line 247 of file dsp.c.

DEF_DTMF_HITS_TO_BEGIN

#define DEF_DTMF_HITS_TO_BEGIN   2

Definition at line 234 of file dsp.c.

DEF_DTMF_MISSES_TO_END

#define DEF_DTMF_MISSES_TO_END   3

Definition at line 239 of file dsp.c.

DEF_DTMF_NORMAL_TWIST

#define DEF_DTMF_NORMAL_TWIST   6.31 /* 8.0dB */

Definition at line 185 of file dsp.c.

DEF_DTMF_REVERSE_TWIST

#define DEF_DTMF_REVERSE_TWIST   2.51 /* 4.01dB */

Definition at line 192 of file dsp.c.

DEF_RELAX_DTMF_NORMAL_TWIST

#define DEF_RELAX_DTMF_NORMAL_TWIST   6.31 /* 8.0dB */

Definition at line 186 of file dsp.c.

DEF_RELAX_DTMF_REVERSE_TWIST

#define DEF_RELAX_DTMF_REVERSE_TWIST   3.98 /* 6.0dB */

Definition at line 193 of file dsp.c.

DEFAULT_SAMPLE_RATE

#define DEFAULT_SAMPLE_RATE   8000

Definition at line 223 of file dsp.c.

DEFAULT_THRESHOLD

#define DEFAULT_THRESHOLD   512

Default minimum average magnitude threshold to determine talking/noise by the DSP.

The magnitude calculated for this threshold is determined by averaging the absolute value of all samples within a frame.

This value is the threshold for which a frame's average magnitude is determined to either be silence (below the threshold) or noise/talking (at or 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 work for any value between 1 and 2^15.

Definition at line 140 of file dsp.c.

DSP_HISTORY

#define DSP_HISTORY   15

Remember last 15 units

Definition at line 151 of file dsp.c.

DTMF_GSIZE

#define DTMF_GSIZE   102

Definition at line 229 of file dsp.c.

DTMF_MATRIX_SIZE

#define DTMF_MATRIX_SIZE   4

Definition at line 168 of file dsp.c.

DTMF_RELATIVE_PEAK_COL

#define DTMF_RELATIVE_PEAK_COL   6.3 /* 8dB */

Definition at line 197 of file dsp.c.

DTMF_RELATIVE_PEAK_ROW

#define DTMF_RELATIVE_PEAK_ROW   6.3 /* 8dB */

Definition at line 196 of file dsp.c.

DTMF_THRESHOLD

#define DTMF_THRESHOLD   8.0e7

Definition at line 182 of file dsp.c.

DTMF_TO_TOTAL_ENERGY

#define DTMF_TO_TOTAL_ENERGY   42.0

Definition at line 198 of file dsp.c.

FAX_TONE_CED_DB

#define FAX_TONE_CED_DB   16

Definition at line 221 of file dsp.c.

FAX_TONE_CED_DURATION

#define FAX_TONE_CED_DURATION   2600 /* ms */

Definition at line 220 of file dsp.c.

FAX_TONE_CED_FREQ

#define FAX_TONE_CED_FREQ   2100

Definition at line 219 of file dsp.c.

FAX_TONE_CNG_DB

#define FAX_TONE_CNG_DB   16

Definition at line 213 of file dsp.c.

FAX_TONE_CNG_DURATION

#define FAX_TONE_CNG_DURATION   500 /* ms */

Definition at line 212 of file dsp.c.

FAX_TONE_CNG_FREQ

#define FAX_TONE_CNG_FREQ   1100

Definition at line 211 of file dsp.c.

FREQ_ARRAY_SIZE

#define FREQ_ARRAY_SIZE   7

Definition at line 115 of file dsp.c.

MAX_DTMF_DIGITS

#define MAX_DTMF_DIGITS   128

Definition at line 166 of file dsp.c.

MF_GSIZE

#define MF_GSIZE   120

Definition at line 226 of file dsp.c.

TONE_MIN_THRESH

#define TONE_MIN_THRESH   1e8

How much tone there should be at least to attempt

Definition at line 154 of file dsp.c.

TONE_THRESH

#define TONE_THRESH   10.0

How much louder the tone should be than channel energy

Definition at line 153 of file dsp.c.

TONE_THRESHOLD

#define TONE_THRESHOLD   7.8e7

Definition at line 183 of file dsp.c.

Enumeration Type Documentation

busy_detect

enum busy_detect

Enumerator
BUSY_PERCENT	The percentage difference between the two last silence periods
BUSY_PAT_PERCENT	The percentage difference between measured and actual pattern
BUSY_THRESHOLD	Max number of ms difference between max and min times in busy
BUSY_MIN	Busy must be at least 80 ms in half-cadence
BUSY_MAX	Busy can't be longer than 3100 ms in half-cadence

Definition at line 142 of file dsp.c.

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

freq_index

enum freq_index

Enumerator
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 85 of file dsp.c.

                {
    /*! For US modes { */
    HZ_350 = 0,
    HZ_440,
    HZ_480,
    HZ_620,
    HZ_950,
    HZ_1400,
    HZ_1800, /*!< } */
 
    /*! For CR/BR modes */
    HZ_425 = 0,
 
    /*! For UK mode */
    HZ_350UK = 0,
    HZ_400UK,
    HZ_440UK
};

gsamp_size

enum gsamp_size

Number of goertzels for progress detect

Enumerator
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 73 of file dsp.c.

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

gsamp_thresh

enum gsamp_thresh

All THRESH_XXX values are in GSAMP_SIZE chunks (us = 22ms)

Enumerator
THRESH_RING	Need at least 150ms ring to accept
THRESH_TALK	Talk detection does not work continuously
THRESH_BUSY	Need at least 80ms to accept
THRESH_CONGESTION	Need at least 80ms to accept
THRESH_HANGUP	Need at least 1300ms to accept hangup
THRESH_RING2ANSWER	Timeout from start of ring to answer (about 6600 ms)

Definition at line 157 of file dsp.c.

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

prog_mode

enum prog_mode

Enumerator
PROG_MODE_NA
PROG_MODE_CR
PROG_MODE_UK

Definition at line 79 of file dsp.c.

               {
    PROG_MODE_NA = 0,
    PROG_MODE_CR,
    PROG_MODE_UK
};

Function Documentation

__ast_dsp_call_progress()

static int __ast_dsp_call_progress ( struct ast_dsp *  dsp, short *  s, int  len  )

static

Definition at line 1081 of file dsp.c.

{
    short samp;
    int x;
    int y;
    int pass;
    int newstate = DSP_TONE_STATE_SILENCE;
    int res = 0;
    int freqcount = dsp->freqcount > FREQ_ARRAY_SIZE ? FREQ_ARRAY_SIZE : dsp->freqcount;
 
    while (len) {
        /* Take the lesser of the number of samples we need and what we have */
        pass = len;
        if (pass > dsp->gsamp_size - dsp->gsamps) {
            pass = dsp->gsamp_size - dsp->gsamps;
        }
        for (x = 0; x < pass; x++) {
            samp = s[x];
            dsp->genergy += (int32_t) samp * (int32_t) samp;
            for (y = 0; y < freqcount; y++) {
                goertzel_sample(&dsp->freqs[y], samp);
            }
        }
        s += pass;
        dsp->gsamps += pass;
        len -= pass;
        if (dsp->gsamps == dsp->gsamp_size) {
            float hz[FREQ_ARRAY_SIZE];
            for (y = 0; y < FREQ_ARRAY_SIZE; y++) {
                hz[y] = goertzel_result(&dsp->freqs[y]);
            }
            switch (dsp->progmode) {
            case PROG_MODE_NA:
                if (pair_there(hz[HZ_480], hz[HZ_620], hz[HZ_350], hz[HZ_440], dsp->genergy)) {
                    newstate = DSP_TONE_STATE_BUSY;
                } else if (pair_there(hz[HZ_440], hz[HZ_480], hz[HZ_350], hz[HZ_620], dsp->genergy)) {
                    newstate = DSP_TONE_STATE_RINGING;
                } else if (pair_there(hz[HZ_350], hz[HZ_440], hz[HZ_480], hz[HZ_620], dsp->genergy)) {
                    newstate = DSP_TONE_STATE_DIALTONE;
                } else if (hz[HZ_950] > TONE_MIN_THRESH * TONE_THRESH) {
                    newstate = DSP_TONE_STATE_SPECIAL1;
                } else if (hz[HZ_1400] > TONE_MIN_THRESH * TONE_THRESH) {
                    /* End of SPECIAL1 or middle of SPECIAL2 */
                    if (dsp->tstate == DSP_TONE_STATE_SPECIAL1 || dsp->tstate == DSP_TONE_STATE_SPECIAL2) {
                        newstate = DSP_TONE_STATE_SPECIAL2;
                    }
                } else if (hz[HZ_1800] > TONE_MIN_THRESH * TONE_THRESH) {
                    /* End of SPECIAL2 or middle of SPECIAL3 */
                    if (dsp->tstate == DSP_TONE_STATE_SPECIAL2 || dsp->tstate == DSP_TONE_STATE_SPECIAL3) {
                        newstate = DSP_TONE_STATE_SPECIAL3;
                    }
                } else if (dsp->genergy > TONE_MIN_THRESH * TONE_THRESH) {
                    newstate = DSP_TONE_STATE_TALKING;
                } else {
                    newstate = DSP_TONE_STATE_SILENCE;
                }
                break;
            case PROG_MODE_CR:
                if (hz[HZ_425] > TONE_MIN_THRESH * TONE_THRESH) {
                    newstate = DSP_TONE_STATE_RINGING;
                } else if (dsp->genergy > TONE_MIN_THRESH * TONE_THRESH) {
                    newstate = DSP_TONE_STATE_TALKING;
                } else {
                    newstate = DSP_TONE_STATE_SILENCE;
                }
                break;
            case PROG_MODE_UK:
                if (hz[HZ_400UK] > TONE_MIN_THRESH * TONE_THRESH) {
                    newstate = DSP_TONE_STATE_HUNGUP;
                } else if (pair_there(hz[HZ_350UK], hz[HZ_440UK], hz[HZ_400UK], hz[HZ_400UK], dsp->genergy)) {
                    newstate = DSP_TONE_STATE_DIALTONE;
                }
                break;
            default:
                ast_log(LOG_WARNING, "Can't process in unknown prog mode '%u'\n", dsp->progmode);
            }
            if (newstate == dsp->tstate) {
                dsp->tcount++;
                if (dsp->ringtimeout) {
                    dsp->ringtimeout++;
                }
                switch (dsp->tstate) {
                case DSP_TONE_STATE_RINGING:
                    if ((dsp->features & DSP_PROGRESS_RINGING) &&
                        (dsp->tcount == THRESH_RING)) {
                        res = AST_CONTROL_RINGING;
                        dsp->ringtimeout = 1;
                    }
                    break;
                case DSP_TONE_STATE_BUSY:
                    if ((dsp->features & DSP_PROGRESS_BUSY) &&
                        (dsp->tcount == THRESH_BUSY)) {
                        res = AST_CONTROL_BUSY;
                        dsp->features &= ~DSP_FEATURE_CALL_PROGRESS;
                    }
                    break;
                case DSP_TONE_STATE_TALKING:
                    if ((dsp->features & DSP_PROGRESS_TALK) &&
                        (dsp->tcount == THRESH_TALK)) {
                        res = AST_CONTROL_ANSWER;
                        dsp->features &= ~DSP_FEATURE_CALL_PROGRESS;
                    }
                    break;
                case DSP_TONE_STATE_SPECIAL3:
                    if ((dsp->features & DSP_PROGRESS_CONGESTION) &&
                        (dsp->tcount == THRESH_CONGESTION)) {
                        res = AST_CONTROL_CONGESTION;
                        dsp->features &= ~DSP_FEATURE_CALL_PROGRESS;
                    }
                    break;
                case DSP_TONE_STATE_HUNGUP:
                    if ((dsp->features & DSP_FEATURE_CALL_PROGRESS) &&
                        (dsp->tcount == THRESH_HANGUP)) {
                        res = AST_CONTROL_HANGUP;
                        dsp->features &= ~DSP_FEATURE_CALL_PROGRESS;
                    }
                    break;
                }
                if (dsp->ringtimeout == THRESH_RING2ANSWER) {
                    ast_debug(1, "Consider call as answered because of timeout after last ring\n");
                    res = AST_CONTROL_ANSWER;
                    dsp->features &= ~DSP_FEATURE_CALL_PROGRESS;
                }
            } else {
                ast_debug(5, "Stop state %d with duration %d\n", dsp->tstate, dsp->tcount);
                ast_debug(5, "Start state %d\n", newstate);
                dsp->tstate = newstate;
                dsp->tcount = 1;
            }
 
            /* Reset goertzel */
            for (x = 0; x < 7; x++) {
                dsp->freqs[x].v2 = dsp->freqs[x].v3 = 0.0;
            }
            dsp->gsamps = 0;
            dsp->genergy = 0.0;
        }
    }
 
    return res;
}

References AST_CONTROL_ANSWER, AST_CONTROL_BUSY, AST_CONTROL_CONGESTION, AST_CONTROL_HANGUP, AST_CONTROL_RINGING, ast_debug, ast_log, DSP_FEATURE_CALL_PROGRESS, DSP_PROGRESS_BUSY, DSP_PROGRESS_CONGESTION, DSP_PROGRESS_RINGING, DSP_PROGRESS_TALK, DSP_TONE_STATE_BUSY, DSP_TONE_STATE_DIALTONE, DSP_TONE_STATE_HUNGUP, DSP_TONE_STATE_RINGING, DSP_TONE_STATE_SILENCE, DSP_TONE_STATE_SPECIAL1, DSP_TONE_STATE_SPECIAL2, DSP_TONE_STATE_SPECIAL3, DSP_TONE_STATE_TALKING, ast_dsp::features, FREQ_ARRAY_SIZE, ast_dsp::freqcount, ast_dsp::freqs, ast_dsp::genergy, goertzel_result(), goertzel_sample(), ast_dsp::gsamp_size, ast_dsp::gsamps, HZ_1400, HZ_1800, HZ_350, HZ_350UK, HZ_400UK, HZ_425, HZ_440, HZ_440UK, HZ_480, HZ_620, HZ_950, len(), LOG_WARNING, pair_there(), pass, PROG_MODE_CR, PROG_MODE_NA, PROG_MODE_UK, ast_dsp::progmode, ast_dsp::ringtimeout, ast_dsp::tcount, THRESH_BUSY, THRESH_CONGESTION, THRESH_HANGUP, THRESH_RING, THRESH_RING2ANSWER, THRESH_TALK, TONE_MIN_THRESH, TONE_THRESH, ast_dsp::tstate, goertzel_state_t::v2, and goertzel_state_t::v3.

Referenced by ast_dsp_call_progress(), and ast_dsp_process().

__ast_dsp_new()

static struct ast_dsp * __ast_dsp_new ( unsigned int  sample_rate )

static

Definition at line 1739 of file dsp.c.

{
    struct ast_dsp *dsp;
 
    if ((dsp = ast_calloc(1, sizeof(*dsp)))) {
        dsp->threshold = DEFAULT_THRESHOLD;
        dsp->features = DSP_FEATURE_SILENCE_SUPPRESS;
        dsp->busycount = DSP_HISTORY;
        dsp->digitmode = DSP_DIGITMODE_DTMF;
        dsp->faxmode = DSP_FAXMODE_DETECT_CNG;
        dsp->sample_rate = sample_rate;
        dsp->freqcount = 0;
        /* Initialize digit detector */
        ast_digit_detect_init(&dsp->digit_state, dsp->digitmode & DSP_DIGITMODE_MF, dsp->sample_rate);
        dsp->display_inband_dtmf_warning = 1;
        /* Initialize initial DSP progress detect parameters */
        ast_dsp_prog_reset(dsp);
        /* Initialize fax detector */
        ast_fax_detect_init(dsp);
    }
    return dsp;
}

References ast_calloc, ast_digit_detect_init(), ast_dsp_prog_reset(), ast_fax_detect_init(), ast_dsp::busycount, DEFAULT_THRESHOLD, ast_dsp::digit_state, ast_dsp::digitmode, ast_dsp::display_inband_dtmf_warning, DSP_DIGITMODE_DTMF, DSP_DIGITMODE_MF, DSP_FAXMODE_DETECT_CNG, DSP_FEATURE_SILENCE_SUPPRESS, DSP_HISTORY, ast_dsp::faxmode, ast_dsp::features, ast_dsp::freqcount, ast_dsp::sample_rate, and ast_dsp::threshold.

Referenced by ast_dsp_new(), and ast_dsp_new_with_rate().

__ast_dsp_silence_noise()

static int __ast_dsp_silence_noise ( struct ast_dsp *  dsp, short *  s, int  len, int *  totalsilence, int *  totalnoise, int *  frames_energy  )

static

Definition at line 1237 of file dsp.c.

{
    int accum;
    int x;
    int res = 0;
 
    if (!len) {
        return 0;
    }
    accum = 0;
    for (x = 0; x < len; x++) {
        accum += abs(s[x]);
    }
    accum /= len;
    if (accum < dsp->threshold) {
        /* Silent */
        dsp->totalsilence += len / (dsp->sample_rate / 1000);
        if (dsp->totalnoise) {
            /* Move and save history */
            memmove(dsp->historicnoise + DSP_HISTORY - dsp->busycount, dsp->historicnoise + DSP_HISTORY - dsp->busycount + 1, dsp->busycount * sizeof(dsp->historicnoise[0]));
            dsp->historicnoise[DSP_HISTORY - 1] = dsp->totalnoise;
/* we don't want to check for busydetect that frequently */
#if 0
            dsp->busymaybe = 1;
#endif
        }
        dsp->totalnoise = 0;
        res = 1;
    } else {
        /* Not silent */
        dsp->totalnoise += len / (dsp->sample_rate / 1000);
        if (dsp->totalsilence) {
            int silence1 = dsp->historicsilence[DSP_HISTORY - 1];
            int silence2 = dsp->historicsilence[DSP_HISTORY - 2];
            /* Move and save history */
            memmove(dsp->historicsilence + DSP_HISTORY - dsp->busycount, dsp->historicsilence + DSP_HISTORY - dsp->busycount + 1, dsp->busycount * sizeof(dsp->historicsilence[0]));
            dsp->historicsilence[DSP_HISTORY - 1] = dsp->totalsilence;
            /* check if the previous sample differs only by BUSY_PERCENT from the one before it */
            if (silence1 < silence2) {
                if (silence1 + silence1 * BUSY_PERCENT / 100 >= silence2) {
                    dsp->busymaybe = 1;
                } else {
                    dsp->busymaybe = 0;
                }
            } else {
                if (silence1 - silence1 * BUSY_PERCENT / 100 <= silence2) {
                    dsp->busymaybe = 1;
                } else {
                    dsp->busymaybe = 0;
                }
            }
        }
        dsp->totalsilence = 0;
    }
    if (totalsilence) {
        *totalsilence = dsp->totalsilence;
    }
    if (totalnoise) {
        *totalnoise = dsp->totalnoise;
    }
    if (frames_energy) {
        *frames_energy = accum;
    }
    return res;
}

References abs, BUSY_PERCENT, ast_dsp::busycount, ast_dsp::busymaybe, DSP_HISTORY, ast_dsp::historicnoise, ast_dsp::historicsilence, len(), ast_dsp::sample_rate, ast_dsp::totalnoise, and ast_dsp::totalsilence.

Referenced by ast_dsp_process(), and ast_dsp_silence_noise_with_energy().

__reg_module()

static void __reg_module ( void  )

static

Definition at line 2483 of file dsp.c.

__unreg_module()

static void __unreg_module ( void  )

static

Definition at line 2483 of file dsp.c.

_dsp_init()

static int _dsp_init ( int  reload )

static

Definition at line 1925 of file dsp.c.

{
    struct ast_config *cfg;
    struct ast_variable *v;
    struct ast_flags config_flags = { reload ? CONFIG_FLAG_FILEUNCHANGED : 0 };
    int cfg_threshold;
    float cfg_twist;
 
    if ((cfg = ast_config_load2(CONFIG_FILE_NAME, "dsp", config_flags)) == CONFIG_STATUS_FILEUNCHANGED) {
        return 0;
    }
 
    thresholds[THRESHOLD_SILENCE] = DEFAULT_SILENCE_THRESHOLD;
    dtmf_normal_twist = DEF_DTMF_NORMAL_TWIST;
    dtmf_reverse_twist = DEF_DTMF_REVERSE_TWIST;
    relax_dtmf_normal_twist = DEF_RELAX_DTMF_NORMAL_TWIST;
    relax_dtmf_reverse_twist = DEF_RELAX_DTMF_REVERSE_TWIST;
    dtmf_hits_to_begin = DEF_DTMF_HITS_TO_BEGIN;
    dtmf_misses_to_end = DEF_DTMF_MISSES_TO_END;
 
    if (cfg == CONFIG_STATUS_FILEMISSING || cfg == CONFIG_STATUS_FILEINVALID) {
        return 0;
    }
 
    for (v = ast_variable_browse(cfg, "default"); v; v = v->next) {
        if (!strcasecmp(v->name, "silencethreshold")) {
            if (sscanf(v->value, "%30d", &cfg_threshold) < 1) {
                ast_log(LOG_WARNING, "Unable to convert '%s' to a numeric value.\n", v->value);
            } else if (cfg_threshold < 0) {
                ast_log(LOG_WARNING, "Invalid silence threshold '%d' specified, using default\n", cfg_threshold);
            } else {
                thresholds[THRESHOLD_SILENCE] = cfg_threshold;
            }
        } else if (!strcasecmp(v->name, "dtmf_normal_twist")) {
            if (sscanf(v->value, "%30f", &cfg_twist) < 1) {
                ast_log(LOG_WARNING, "Unable to convert '%s' to a numeric value.\n", v->value);
            } else if ((cfg_twist < 2.0) || (cfg_twist > 100.0)) {      /* < 3.0dB or > 20dB */
                ast_log(LOG_WARNING, "Invalid dtmf_normal_twist value '%.2f' specified, using default of %.2f\n", cfg_twist, dtmf_normal_twist);
            } else {
                dtmf_normal_twist = cfg_twist;
            }
        } else if (!strcasecmp(v->name, "dtmf_reverse_twist")) {
            if (sscanf(v->value, "%30f", &cfg_twist) < 1) {
                ast_log(LOG_WARNING, "Unable to convert '%s' to a numeric value.\n", v->value);
            } else if ((cfg_twist < 2.0) || (cfg_twist > 100.0)) {      /* < 3.0dB or > 20dB */
                ast_log(LOG_WARNING, "Invalid dtmf_reverse_twist value '%.2f' specified, using default of %.2f\n", cfg_twist, dtmf_reverse_twist);
            } else {
                dtmf_reverse_twist = cfg_twist;
            }
        } else if (!strcasecmp(v->name, "relax_dtmf_normal_twist")) {
            if (sscanf(v->value, "%30f", &cfg_twist) < 1) {
                ast_log(LOG_WARNING, "Unable to convert '%s' to a numeric value.\n", v->value);
            } else if ((cfg_twist < 2.0) || (cfg_twist > 100.0)) {      /* < 3.0dB or > 20dB */
                ast_log(LOG_WARNING, "Invalid relax_dtmf_normal_twist value '%.2f' specified, using default of %.2f\n", cfg_twist, relax_dtmf_normal_twist);
            } else {
                relax_dtmf_normal_twist = cfg_twist;
            }
        } else if (!strcasecmp(v->name, "relax_dtmf_reverse_twist")) {
            if (sscanf(v->value, "%30f", &cfg_twist) < 1) {
                ast_log(LOG_WARNING, "Unable to convert '%s' to a numeric value.\n", v->value);
            } else if ((cfg_twist < 2.0) || (cfg_twist > 100.0)) {      /* < 3.0dB or > 20dB */
                ast_log(LOG_WARNING, "Invalid relax_dtmf_reverse_twist value '%.2f' specified, using default of %.2f\n", cfg_twist, relax_dtmf_reverse_twist);
            } else {
                relax_dtmf_reverse_twist = cfg_twist;
            }
        } else if (!strcasecmp(v->name, "dtmf_hits_to_begin")) {
            if (sscanf(v->value, "%30d", &cfg_threshold) < 1) {
                ast_log(LOG_WARNING, "Unable to convert '%s' to a numeric value.\n", v->value);
            } else if (cfg_threshold < 1) {     /* must be 1 or greater */
                ast_log(LOG_WARNING, "Invalid dtmf_hits_to_begin value '%d' specified, using default of %d\n", cfg_threshold, dtmf_hits_to_begin);
            } else {
                dtmf_hits_to_begin = cfg_threshold;
            }
        } else if (!strcasecmp(v->name, "dtmf_misses_to_end")) {
            if (sscanf(v->value, "%30d", &cfg_threshold) < 1) {
                ast_log(LOG_WARNING, "Unable to convert '%s' to a numeric value.\n", v->value);
            } else if (cfg_threshold < 1) {     /* must be 1 or greater */
                ast_log(LOG_WARNING, "Invalid dtmf_misses_to_end value '%d' specified, using default of %d\n", cfg_threshold, dtmf_misses_to_end);
            } else {
                dtmf_misses_to_end = cfg_threshold;
            }
        }
    }
    ast_config_destroy(cfg);
 
    return 0;
}

References ast_config_destroy(), ast_config_load2(), ast_log, ast_variable_browse(), CONFIG_FILE_NAME, CONFIG_FLAG_FILEUNCHANGED, CONFIG_STATUS_FILEINVALID, CONFIG_STATUS_FILEMISSING, CONFIG_STATUS_FILEUNCHANGED, DEF_DTMF_HITS_TO_BEGIN, DEF_DTMF_MISSES_TO_END, DEF_DTMF_NORMAL_TWIST, DEF_DTMF_REVERSE_TWIST, DEF_RELAX_DTMF_NORMAL_TWIST, DEF_RELAX_DTMF_REVERSE_TWIST, DEFAULT_SILENCE_THRESHOLD, dtmf_hits_to_begin, dtmf_misses_to_end, dtmf_normal_twist, dtmf_reverse_twist, LOG_WARNING, ast_variable::name, ast_variable::next, relax_dtmf_normal_twist, relax_dtmf_reverse_twist, reload(), THRESHOLD_SILENCE, thresholds, and ast_variable::value.

Referenced by load_module(), and reload_module().

ast_digit_detect_init()

static void ast_digit_detect_init ( digit_detect_state_t *  s, int  mf, unsigned int  sample_rate  )

static

Definition at line 561 of file dsp.c.

{
    s->current_digits = 0;
    s->detected_digits = 0;
    s->lost_digits = 0;
    s->digits[0] = '\0';
 
    if (mf) {
        ast_mf_detect_init(&s->td.mf, sample_rate);
    } else {
        ast_dtmf_detect_init(&s->td.dtmf, sample_rate);
    }
}

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().

ast_dsp_busydetect()

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 1303 of file dsp.c.

{
    int res = 0, x;
#ifndef BUSYDETECT_TONEONLY
    int avgsilence = 0, hitsilence = 0;
#endif
    int avgtone = 0, hittone = 0;
 
    /* if we have a 4 length pattern, the way busymaybe is set doesn't help us. */
    if (dsp->busy_cadence.length != 4) {
        if (!dsp->busymaybe) {
            return res;
        }
    }
 
    for (x = DSP_HISTORY - dsp->busycount; x < DSP_HISTORY; x++) {
#ifndef BUSYDETECT_TONEONLY
        avgsilence += dsp->historicsilence[x];
#endif
        avgtone += dsp->historicnoise[x];
    }
#ifndef BUSYDETECT_TONEONLY
    avgsilence /= dsp->busycount;
#endif
    avgtone /= dsp->busycount;
    for (x = DSP_HISTORY - dsp->busycount; x < DSP_HISTORY; x++) {
#ifndef BUSYDETECT_TONEONLY
        if (avgsilence > dsp->historicsilence[x]) {
            if (avgsilence - (avgsilence * BUSY_PERCENT / 100) <= dsp->historicsilence[x]) {
                hitsilence++;
            }
        } else {
            if (avgsilence + (avgsilence * BUSY_PERCENT / 100) >= dsp->historicsilence[x]) {
                hitsilence++;
            }
        }
#endif
        if (avgtone > dsp->historicnoise[x]) {
            if (avgtone - (avgtone * BUSY_PERCENT / 100) <= dsp->historicnoise[x]) {
                hittone++;
            }
        } else {
            if (avgtone + (avgtone * BUSY_PERCENT / 100) >= dsp->historicnoise[x]) {
                hittone++;
            }
        }
    }
#ifndef BUSYDETECT_TONEONLY
    if ((hittone >= dsp->busycount - 1) && (hitsilence >= dsp->busycount - 1) &&
        (avgtone >= BUSY_MIN && avgtone <= BUSY_MAX) &&
        (avgsilence >= BUSY_MIN && avgsilence <= BUSY_MAX))
#else
    if ((hittone >= dsp->busycount - 1) && (avgtone >= BUSY_MIN && avgtone <= BUSY_MAX))
#endif
    {
#ifdef BUSYDETECT_COMPARE_TONE_AND_SILENCE
        if (avgtone > avgsilence) {
            if (avgtone - avgtone*BUSY_PERCENT/100 <= avgsilence) {
                res = 1;
            }
        } else {
            if (avgtone + avgtone*BUSY_PERCENT/100 >= avgsilence) {
                res = 1;
            }
        }
#else
        res = 1;
#endif
    }
 
    /* If we have a 4-length pattern, we can go ahead and just check it in a different way. */
    if (dsp->busy_cadence.length == 4) {
        int x;
        int errors = 0;
        int errors_max = ((4 * dsp->busycount) / 100.0) * BUSY_PAT_PERCENT;
 
        for (x = DSP_HISTORY - (dsp->busycount); x < DSP_HISTORY; x += 2) {
            int temp_error;
            temp_error = abs(dsp->historicnoise[x] - dsp->busy_cadence.pattern[0]);
            if ((temp_error * 100) / dsp->busy_cadence.pattern[0] > BUSY_PERCENT) {
                errors++;
            }
 
            temp_error = abs(dsp->historicnoise[x + 1] - dsp->busy_cadence.pattern[2]);
            if ((temp_error * 100) / dsp->busy_cadence.pattern[2] > BUSY_PERCENT) {
                errors++;
            }
 
            temp_error = abs(dsp->historicsilence[x] - dsp->busy_cadence.pattern[1]);
            if ((temp_error * 100) / dsp->busy_cadence.pattern[1] > BUSY_PERCENT) {
                errors++;
            }
 
            temp_error = abs(dsp->historicsilence[x + 1] - dsp->busy_cadence.pattern[3]);
            if ((temp_error * 100) / dsp->busy_cadence.pattern[3] > BUSY_PERCENT) {
                errors++;
            }
        }
 
        ast_debug(5, "errors = %d  max = %d\n", errors, errors_max);
 
        if (errors <= errors_max) {
            return 1;
        }
    }
 
    /* If we know the expected busy tone length, check we are in the range */
    if (res && (dsp->busy_cadence.pattern[0] > 0)) {
        if (abs(avgtone - dsp->busy_cadence.pattern[0]) > MAX(dsp->busy_cadence.pattern[0]*BUSY_PAT_PERCENT/100, 20)) {
#ifdef BUSYDETECT_DEBUG
            ast_debug(5, "busy detector: avgtone of %d not close enough to desired %d\n",
                avgtone, dsp->busy_cadence.pattern[0]);
#endif
            res = 0;
        }
    }
#ifndef BUSYDETECT_TONEONLY
    /* If we know the expected busy tone silent-period length, check we are in the range */
    if (res && (dsp->busy_cadence.pattern[1] > 0)) {
        if (abs(avgsilence - dsp->busy_cadence.pattern[1]) > MAX(dsp->busy_cadence.pattern[1]*BUSY_PAT_PERCENT/100, 20)) {
#ifdef BUSYDETECT_DEBUG
        ast_debug(5, "busy detector: avgsilence of %d not close enough to desired %d\n",
            avgsilence, dsp->busy_cadence.pattern[1]);
#endif
            res = 0;
        }
    }
#endif
#if !defined(BUSYDETECT_TONEONLY) && defined(BUSYDETECT_DEBUG)
    if (res) {
        ast_debug(5, "ast_dsp_busydetect detected busy, avgtone: %d, avgsilence %d\n", avgtone, avgsilence);
    } else {
        ast_debug(5, "busy detector: FAILED with avgtone: %d, avgsilence %d\n", avgtone, avgsilence);
    }
#endif
    return res;
}

References abs, ast_debug, ast_dsp::busy_cadence, BUSY_MAX, BUSY_MIN, BUSY_PAT_PERCENT, BUSY_PERCENT, ast_dsp::busycount, ast_dsp::busymaybe, DSP_HISTORY, ast_dsp::historicnoise, ast_dsp::historicsilence, ast_dsp_busy_pattern::length, MAX, and ast_dsp_busy_pattern::pattern.

Referenced by ast_dsp_process().

ast_dsp_call_progress()

int ast_dsp_call_progress	(	struct ast_dsp *	dsp,
		struct ast_frame *	inf
	)

Scans for progress indication in audio.

Definition at line 1223 of file dsp.c.

{
    if (inf->frametype != AST_FRAME_VOICE) {
        ast_log(LOG_WARNING, "Can't check call progress of non-voice frames\n");
        return 0;
    }
    if (!ast_format_cache_is_slinear(inf->subclass.format)) {
        ast_log(LOG_WARNING, "Can only check call progress in signed-linear frames, %s not supported\n",
            ast_format_get_name(inf->subclass.format));
        return 0;
    }
    return __ast_dsp_call_progress(dsp, inf->data.ptr, inf->datalen / 2);
}

References __ast_dsp_call_progress(), ast_format_cache_is_slinear(), ast_format_get_name(), AST_FRAME_VOICE, ast_log, ast_frame::data, ast_frame::datalen, ast_frame_subclass::format, ast_frame::frametype, LOG_WARNING, ast_frame::ptr, and ast_frame::subclass.

ast_dsp_digitreset()

void ast_dsp_digitreset

(

struct ast_dsp *

dsp

)

Reset DTMF detector.

Definition at line 1814 of file dsp.c.

{
    int i;
 
    dsp->dtmf_began = 0;
    if (dsp->digitmode & DSP_DIGITMODE_MF) {
        mf_detect_state_t *s = &dsp->digit_state.td.mf;
        /* Reinitialise the detector for the next block */
        for (i = 0; i < 6; i++) {
            goertzel_reset(&s->tone_out[i]);
        }
        s->hits[4] = s->hits[3] = s->hits[2] = s->hits[1] = s->hits[0] = 0;
        s->current_hit = 0;
        s->current_sample = 0;
    } else {
        dtmf_detect_state_t *s = &dsp->digit_state.td.dtmf;
        /* Reinitialise the detector for the next block */
        for (i = 0; i < 4; i++) {
            goertzel_reset(&s->row_out[i]);
            goertzel_reset(&s->col_out[i]);
        }
        s->lasthit = 0;
        s->current_hit = 0;
        s->energy = 0.0;
        s->current_sample = 0;
        s->hits = 0;
        s->misses = 0;
    }
 
    dsp->digit_state.digits[0] = '\0';
    dsp->digit_state.current_digits = 0;
}

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, DSP_DIGITMODE_MF, 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(), mbl_new(), and my_dsp_reset_and_flush_digits().

ast_dsp_free()

void ast_dsp_free

(

struct ast_dsp *

dsp

)

Definition at line 1787 of file dsp.c.

{
    ast_free(dsp);
}

References ast_free.

Referenced by __ast_play_and_record(), analog_ss_thread(), background_detect_exec(), chan_pjsip_cng_tone_detected(), chan_pjsip_read_stream(), conf_run(), dahdi_hangup(), datastore_destroy_cb(), destroy_callback(), destroy_faxdetect(), destroy_session(), do_waiting(), handle_recordfile(), isAnsweringMachine(), mbl_load_device(), my_all_subchannels_hungup(), my_dsp_set_digitmode(), ooh323_destroy(), pjsip_acf_dtmf_mode_write(), read_mf_digits(), read_sf_digits(), record_exec(), scan_exec(), session_destructor(), set_caps(), set_softmix_bridge_data(), softmix_bridge_leave(), unload_module(), and wait_exec().

ast_dsp_get_features()

int ast_dsp_get_features

(

struct ast_dsp *

dsp

)

Get features.

Definition at line 1781 of file dsp.c.

{
        return (dsp->features);
}

References ast_dsp::features.

Referenced by chan_pjsip_cng_tone_detected(), chan_pjsip_read_stream(), pjsip_acf_dtmf_mode_write(), and set_caps().

ast_dsp_get_sample_rate()

unsigned int ast_dsp_get_sample_rate

(

const struct ast_dsp *

dsp

)

Retrieve the sample rate this DSP structure was created with.

Definition at line 1734 of file dsp.c.

{
    return dsp->sample_rate;
}

References ast_dsp::sample_rate.

ast_dsp_get_tcount()

int ast_dsp_get_tcount

(

struct ast_dsp *

dsp

)

Get tcount (Threshold counter)

Definition at line 1920 of file dsp.c.

{
    return dsp->tcount;
}

References ast_dsp::tcount.

Referenced by dahdi_read(), detect_callback(), and scan_exec().

ast_dsp_get_threshold_from_settings()

int ast_dsp_get_threshold_from_settings

(

enum threshold

which

)

Get silence threshold from dsp.conf.

Since

1.6.1

Definition at line 2013 of file dsp.c.

{
    return thresholds[which];
}

References thresholds.

Referenced by actual_load_config(), app_exec(), ast_record_review(), conf_rec_name(), conf_run(), do_waiting(), handle_recordfile(), load_config(), load_config(), record_exec(), setup_privacy_args(), and talk_detect_fn_write().

ast_dsp_get_tstate()

int ast_dsp_get_tstate

(

struct ast_dsp *

dsp

)

Get tstate (Tone State)

Definition at line 1915 of file dsp.c.

{
    return dsp->tstate;
}

References ast_dsp::tstate.

Referenced by dahdi_read(), detect_callback(), and scan_exec().

ast_dsp_new()

struct ast_dsp * ast_dsp_new

(

void

)

Allocates a new dsp, assumes 8khz for internal sample rate.

Definition at line 1762 of file dsp.c.

{
    return __ast_dsp_new(DEFAULT_SAMPLE_RATE);
}

References __ast_dsp_new(), and DEFAULT_SAMPLE_RATE.

Referenced by __ast_play_and_record(), ast_sip_session_alloc(), background_detect_exec(), conf_run(), dahdi_new(), detect_write(), do_waiting(), fax_detect_new(), fax_session_new(), handle_recordfile(), isAnsweringMachine(), mbl_load_device(), my_dsp_set_digitmode(), my_set_waitingfordt(), ooh323_new(), pjsip_acf_dtmf_mode_write(), read_mf_digits(), read_sf_digits(), record_exec(), scan_exec(), and wait_exec().

ast_dsp_new_with_rate()

struct ast_dsp * ast_dsp_new_with_rate

(

unsigned int

sample_rate

)

Allocates a new dsp with a specific internal sample rate used during processing.

Definition at line 1767 of file dsp.c.

{
    return __ast_dsp_new(sample_rate);
}

References __ast_dsp_new(), and ast_dsp::sample_rate.

Referenced by set_softmix_bridge_data(), and set_talk_detect().

ast_dsp_noise()

int ast_dsp_noise	(	struct ast_dsp *	dsp,
		struct ast_frame *	f,
		int *	totalnoise
	)

Process the audio frame for noise.

Since

1.6.1

Parameters

dsp	DSP processing audio media.
f	Audio frame to process.
totalnoise	Variable to set to the total accumulated noise in ms seen by the DSP since the last silence.

Returns

Non-zero if the frame is silence.

Definition at line 1497 of file dsp.c.

{
    return ast_dsp_silence_noise_with_energy(dsp, f, totalnoise, NULL, 1);
}

References ast_dsp_silence_noise_with_energy(), and NULL.

ast_dsp_process()

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 1503 of file dsp.c.

{
    int silence;
    int res;
    int digit = 0, fax_digit = 0, custom_freq_digit = 0;
    int x;
    short *shortdata;
    unsigned char *odata;
    int len;
    struct ast_frame *outf = NULL;
 
    if (!af) {
        return NULL;
    }
    if (af->frametype != AST_FRAME_VOICE) {
        return af;
    }
 
    odata = af->data.ptr;
    len = af->datalen;
    /* Make sure we have short data */
    if (ast_format_cache_is_slinear(af->subclass.format)) {
        shortdata = af->data.ptr;
        len = af->datalen / 2;
    } else if (ast_format_cmp(af->subclass.format, ast_format_ulaw) == AST_FORMAT_CMP_EQUAL) {
        shortdata = ast_alloca(af->datalen * 2);
        for (x = 0; x < len; x++) {
            shortdata[x] = AST_MULAW(odata[x]);
        }
    } else if (ast_format_cmp(af->subclass.format, ast_format_alaw) == AST_FORMAT_CMP_EQUAL) {
        shortdata = ast_alloca(af->datalen * 2);
        for (x = 0; x < len; x++) {
            shortdata[x] = AST_ALAW(odata[x]);
        }
    } else {
        /* Display warning only once. Otherwise you would get hundreds of warnings every second */
        if (dsp->display_inband_dtmf_warning) {
            /* If DTMF is enabled for the DSP, try to be helpful and warn about that specifically,
             * otherwise emit a more generic message that covers all other cases. */
            if ((dsp->features & DSP_FEATURE_DIGIT_DETECT) && (dsp->digitmode & DSP_DIGITMODE_DTMF)) {
                ast_log(LOG_WARNING, "Inband DTMF is not supported on codec %s. Use RFC2833\n",
                    ast_format_get_name(af->subclass.format));
            } else {
                ast_log(LOG_WARNING, "Can only do DSP on signed-linear, alaw or ulaw frames (%s not supported)\n",
                    ast_format_get_name(af->subclass.format));
            }
        }
        dsp->display_inband_dtmf_warning = 0;
        return af;
    }
 
    /* Initially we do not want to mute anything */
    dsp->mute_fragments = 0;
 
    /* Need to run the silence detection stuff for silence suppression and busy detection */
    if ((dsp->features & DSP_FEATURE_SILENCE_SUPPRESS) || (dsp->features & DSP_FEATURE_BUSY_DETECT)) {
        res = __ast_dsp_silence_noise(dsp, shortdata, len, &silence, NULL, NULL);
    }
 
    if ((dsp->features & DSP_FEATURE_SILENCE_SUPPRESS) && silence) {
        memset(&dsp->f, 0, sizeof(dsp->f));
        dsp->f.frametype = AST_FRAME_NULL;
        ast_frfree(af);
        return ast_frisolate(&dsp->f);
    }
    if ((dsp->features & DSP_FEATURE_BUSY_DETECT) && ast_dsp_busydetect(dsp)) {
        ast_channel_softhangup_internal_flag_add(chan, AST_SOFTHANGUP_DEV);
        memset(&dsp->f, 0, sizeof(dsp->f));
        dsp->f.frametype = AST_FRAME_CONTROL;
        dsp->f.subclass.integer = AST_CONTROL_BUSY;
        ast_frfree(af);
        ast_debug(1, "Requesting Hangup because the busy tone was detected on channel %s\n", ast_channel_name(chan));
        return ast_frisolate(&dsp->f);
    }
 
    if ((dsp->features & DSP_FEATURE_FAX_DETECT)) {
        if ((dsp->faxmode & DSP_FAXMODE_DETECT_CNG) && tone_detect(dsp, &dsp->cng_tone_state, shortdata, len)) {
            fax_digit = 'f';
        }
 
        if ((dsp->faxmode & DSP_FAXMODE_DETECT_CED) && tone_detect(dsp, &dsp->ced_tone_state, shortdata, len)) {
            fax_digit = 'e';
        }
    }
 
    if ((dsp->features & DSP_FEATURE_FREQ_DETECT)) {
        if ((dsp->freqmode) && tone_detect(dsp, &dsp->cng_tone_state, shortdata, len)) {
            custom_freq_digit = 'q';
        }
    }
 
    if (dsp->features & (DSP_FEATURE_DIGIT_DETECT | DSP_FEATURE_BUSY_DETECT)) {
        if (dsp->digitmode & DSP_DIGITMODE_MF) {
            digit = mf_detect(dsp, &dsp->digit_state, shortdata, len, (dsp->digitmode & DSP_DIGITMODE_NOQUELCH) == 0, (dsp->digitmode & DSP_DIGITMODE_RELAXDTMF));
        } else {
            digit = dtmf_detect(dsp, &dsp->digit_state, shortdata, len, (dsp->digitmode & DSP_DIGITMODE_NOQUELCH) == 0, (dsp->digitmode & DSP_DIGITMODE_RELAXDTMF));
        }
 
        if (dsp->digit_state.current_digits) {
            int event = 0, event_len = 0;
            char event_digit = 0;
 
            if (!dsp->dtmf_began) {
                /* We have not reported DTMF_BEGIN for anything yet */
 
                if (dsp->features & DSP_FEATURE_DIGIT_DETECT) {
                    event = AST_FRAME_DTMF_BEGIN;
                    event_digit = dsp->digit_state.digits[0];
                }
                dsp->dtmf_began = 1;
 
            } else if (dsp->digit_state.current_digits > 1 || digit != dsp->digit_state.digits[0]) {
                /* Digit changed. This means digit we have reported with DTMF_BEGIN ended */
                if (dsp->features & DSP_FEATURE_DIGIT_DETECT) {
                    event = AST_FRAME_DTMF_END;
                    event_digit = dsp->digit_state.digits[0];
                    event_len = dsp->digit_state.digitlen[0] * 1000 / dsp->sample_rate;
                }
                memmove(&dsp->digit_state.digits[0], &dsp->digit_state.digits[1], dsp->digit_state.current_digits);
                memmove(&dsp->digit_state.digitlen[0], &dsp->digit_state.digitlen[1], dsp->digit_state.current_digits * sizeof(dsp->digit_state.digitlen[0]));
                dsp->digit_state.current_digits--;
                dsp->dtmf_began = 0;
 
                if (dsp->features & DSP_FEATURE_BUSY_DETECT) {
                    /* Reset Busy Detector as we have some confirmed activity */
                    memset(dsp->historicsilence, 0, sizeof(dsp->historicsilence));
                    memset(dsp->historicnoise, 0, sizeof(dsp->historicnoise));
                    ast_debug(1, "DTMF Detected - Reset busydetector\n");
                }
            }
 
            if (event) {
                memset(&dsp->f, 0, sizeof(dsp->f));
                dsp->f.frametype = event;
                dsp->f.subclass.integer = event_digit;
                dsp->f.len = event_len;
                outf = &dsp->f;
                goto done;
            }
        }
    }
 
    if (fax_digit) {
        /* Fax was detected - digit is either 'f' or 'e' */
 
        memset(&dsp->f, 0, sizeof(dsp->f));
        dsp->f.frametype = AST_FRAME_DTMF;
        dsp->f.subclass.integer = fax_digit;
        outf = &dsp->f;
        goto done;
    }
 
    if (custom_freq_digit) {
        /* Custom frequency was detected - digit is 'q' */
 
        memset(&dsp->f, 0, sizeof(dsp->f));
        dsp->f.frametype = AST_FRAME_DTMF;
        dsp->f.subclass.integer = custom_freq_digit;
        outf = &dsp->f;
        goto done;
    }
 
    if ((dsp->features & DSP_FEATURE_CALL_PROGRESS)) {
        res = __ast_dsp_call_progress(dsp, shortdata, len);
        if (res) {
            switch (res) {
            case AST_CONTROL_ANSWER:
            case AST_CONTROL_BUSY:
            case AST_CONTROL_RINGING:
            case AST_CONTROL_CONGESTION:
            case AST_CONTROL_HANGUP:
                memset(&dsp->f, 0, sizeof(dsp->f));
                dsp->f.frametype = AST_FRAME_CONTROL;
                dsp->f.subclass.integer = res;
                dsp->f.src = "dsp_progress";
                if (chan) {
                    ast_queue_frame(chan, &dsp->f);
                }
                break;
            default:
                ast_log(LOG_WARNING, "Don't know how to represent call progress message %d\n", res);
            }
        }
    } else if ((dsp->features & DSP_FEATURE_WAITDIALTONE)) {
        res = __ast_dsp_call_progress(dsp, shortdata, len);
    }
 
done:
    /* Mute fragment of the frame */
    for (x = 0; x < dsp->mute_fragments; x++) {
        memset(shortdata + dsp->mute_data[x].start, 0, sizeof(int16_t) * (dsp->mute_data[x].end - dsp->mute_data[x].start));
    }
 
    if (ast_format_cmp(af->subclass.format, ast_format_ulaw) == AST_FORMAT_CMP_EQUAL) {
        for (x = 0; x < len; x++) {
            odata[x] = AST_LIN2MU((unsigned short) shortdata[x]);
        }
    } else if (ast_format_cmp(af->subclass.format, ast_format_alaw) == AST_FORMAT_CMP_EQUAL) {
        for (x = 0; x < len; x++) {
            odata[x] = AST_LIN2A((unsigned short) shortdata[x]);
        }
    }
 
    if (outf) {
        if (chan) {
            ast_queue_frame(chan, af);
        }
        ast_frfree(af);
        return ast_frisolate(outf);
    } else {
        return af;
    }
}

References __ast_dsp_call_progress(), __ast_dsp_silence_noise(), AST_ALAW, ast_alloca, ast_channel_name(), ast_channel_softhangup_internal_flag_add(), AST_CONTROL_ANSWER, AST_CONTROL_BUSY, AST_CONTROL_CONGESTION, AST_CONTROL_HANGUP, AST_CONTROL_RINGING, ast_debug, ast_dsp_busydetect(), ast_format_alaw, ast_format_cache_is_slinear(), ast_format_cmp(), AST_FORMAT_CMP_EQUAL, ast_format_get_name(), ast_format_ulaw, AST_FRAME_CONTROL, AST_FRAME_DTMF, AST_FRAME_DTMF_BEGIN, AST_FRAME_DTMF_END, AST_FRAME_NULL, AST_FRAME_VOICE, ast_frfree, ast_frisolate, AST_LIN2A, AST_LIN2MU, ast_log, AST_MULAW, ast_queue_frame(), AST_SOFTHANGUP_DEV, ast_dsp::ced_tone_state, ast_dsp::cng_tone_state, digit_detect_state_t::current_digits, ast_frame::data, ast_frame::datalen, digit, ast_dsp::digit_state, digit_detect_state_t::digitlen, ast_dsp::digitmode, digit_detect_state_t::digits, ast_dsp::display_inband_dtmf_warning, done, DSP_DIGITMODE_DTMF, DSP_DIGITMODE_MF, DSP_DIGITMODE_NOQUELCH, DSP_DIGITMODE_RELAXDTMF, DSP_FAXMODE_DETECT_CED, DSP_FAXMODE_DETECT_CNG, DSP_FEATURE_BUSY_DETECT, DSP_FEATURE_CALL_PROGRESS, DSP_FEATURE_DIGIT_DETECT, DSP_FEATURE_FAX_DETECT, DSP_FEATURE_FREQ_DETECT, DSP_FEATURE_SILENCE_SUPPRESS, DSP_FEATURE_WAITDIALTONE, ast_dsp::dtmf_began, dtmf_detect(), fragment_t::end, ast_dsp::f, ast_dsp::faxmode, ast_dsp::features, ast_frame_subclass::format, ast_frame::frametype, ast_dsp::freqmode, ast_dsp::historicnoise, ast_dsp::historicsilence, ast_frame_subclass::integer, len(), ast_frame::len, LOG_WARNING, mf_detect(), ast_dsp::mute_data, ast_dsp::mute_fragments, NULL, ast_frame::ptr, ast_dsp::sample_rate, ast_frame::src, fragment_t::start, ast_frame::subclass, and tone_detect().

Referenced by chan_pjsip_read_stream(), dahdi_read(), detect_callback(), fax_detect_framehook(), mbl_read(), ooh323_rtp_read(), read_mf_digits(), read_sf_digits(), scan_exec(), and wait_exec().

ast_dsp_prog_reset()

static void ast_dsp_prog_reset ( struct ast_dsp *  dsp )

static

Definition at line 1717 of file dsp.c.

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

References FREQ_ARRAY_SIZE, ast_dsp::freqcount, progress::freqs, ast_dsp::freqs, goertzel_init(), ast_dsp::gsamp_size, ast_dsp::gsamps, max, modes, ast_dsp::progmode, ast_dsp::ringtimeout, ast_dsp::sample_rate, and progress::size.

Referenced by __ast_dsp_new(), and ast_dsp_set_call_progress_zone().

ast_dsp_reset()

void ast_dsp_reset

(

struct ast_dsp *

dsp

)

Reset total silence count.

Definition at line 1847 of file dsp.c.

{
    int x;
 
    dsp->totalsilence = 0;
    dsp->gsamps = 0;
    for (x = 0; x < 4; x++) {
        dsp->freqs[x].v2 = dsp->freqs[x].v3 = 0.0;
    }
    memset(dsp->historicsilence, 0, sizeof(dsp->historicsilence));
    memset(dsp->historicnoise, 0, sizeof(dsp->historicnoise));
    dsp->ringtimeout = 0;
}

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().

ast_dsp_set_busy_count()

void ast_dsp_set_busy_count	(	struct ast_dsp *	dsp,
		int	cadences
	)

Set number of required cadences for busy.

Definition at line 1797 of file dsp.c.

{
    if (cadences < 4) {
        cadences = 4;
    }
    if (cadences > DSP_HISTORY) {
        cadences = DSP_HISTORY;
    }
    dsp->busycount = cadences;
}

References ast_dsp::busycount, cadences, and DSP_HISTORY.

Referenced by dahdi_new().

ast_dsp_set_busy_pattern()

void ast_dsp_set_busy_pattern	(	struct ast_dsp *	dsp,
		const struct ast_dsp_busy_pattern *	cadence
	)

Set expected lengths of the busy tone.

Definition at line 1808 of file dsp.c.

{
    dsp->busy_cadence = *cadence;
    ast_debug(1, "dsp busy pattern set to %d,%d,%d,%d\n", cadence->pattern[0], cadence->pattern[1], (cadence->length == 4) ? cadence->pattern[2] : 0, (cadence->length == 4) ? cadence->pattern[3] : 0);
}

References ast_debug, ast_dsp::busy_cadence, ast_dsp_busy_pattern::length, and ast_dsp_busy_pattern::pattern.

Referenced by dahdi_new().

ast_dsp_set_call_progress_zone()

int ast_dsp_set_call_progress_zone	(	struct ast_dsp *	dsp,
		char *	zone
	)

Set zone for doing progress detection.

Definition at line 1896 of file dsp.c.

{
    int x;
 
    for (x = 0; x < ARRAY_LEN(aliases); x++) {
        if (!strcasecmp(aliases[x].name, zone)) {
            dsp->progmode = aliases[x].mode;
            ast_dsp_prog_reset(dsp);
            return 0;
        }
    }
    return -1;
}

References aliases, ARRAY_LEN, ast_dsp_prog_reset(), progalias::mode, name, and ast_dsp::progmode.

Referenced by dahdi_new(), and scan_exec().

ast_dsp_set_digitmode()

int ast_dsp_set_digitmode	(	struct ast_dsp *	dsp,
		int	digitmode
	)

Set digit mode.

Version

1.6.1 renamed from ast_dsp_digitmode to ast_dsp_set_digitmode

Definition at line 1861 of file dsp.c.

{
    int new;
    int old;
 
    old = dsp->digitmode & (DSP_DIGITMODE_DTMF | DSP_DIGITMODE_MF | DSP_DIGITMODE_MUTECONF | DSP_DIGITMODE_MUTEMAX);
    new = digitmode & (DSP_DIGITMODE_DTMF | DSP_DIGITMODE_MF | DSP_DIGITMODE_MUTECONF | DSP_DIGITMODE_MUTEMAX);
    if (old != new) {
        /* Must initialize structures if switching from MF to DTMF or vice-versa */
        ast_digit_detect_init(&dsp->digit_state, new & DSP_DIGITMODE_MF, dsp->sample_rate);
    }
    dsp->digitmode = digitmode;
    return 0;
}

References ast_digit_detect_init(), ast_dsp::digit_state, ast_dsp::digitmode, DSP_DIGITMODE_DTMF, DSP_DIGITMODE_MF, DSP_DIGITMODE_MUTECONF, DSP_DIGITMODE_MUTEMAX, and ast_dsp::sample_rate.

Referenced by analog_ss_thread(), dahdi_hangup(), dahdi_new(), dahdi_setoption(), mbl_load_device(), mkintf(), my_dsp_set_digitmode(), ooh323_new(), and read_mf_digits().

ast_dsp_set_faxmode()

int ast_dsp_set_faxmode	(	struct ast_dsp *	dsp,
		int	faxmode
	)

Set fax mode.

Definition at line 1887 of file dsp.c.

{
    if (dsp->faxmode != faxmode) {
        dsp->faxmode = faxmode;
        ast_fax_detect_init(dsp);
    }
    return 0;
}

References ast_fax_detect_init(), and ast_dsp::faxmode.

Referenced by fax_detect_new(), ooh323_new(), and scan_exec().

ast_dsp_set_features()

void ast_dsp_set_features	(	struct ast_dsp *	dsp,
		int	features
	)

Select feature set.

Definition at line 1772 of file dsp.c.

{
    dsp->features = features;
    if (!(features & DSP_FEATURE_DIGIT_DETECT)) {
        dsp->display_inband_dtmf_warning = 0;
    }
}

References ast_dsp::display_inband_dtmf_warning, DSP_FEATURE_DIGIT_DETECT, and ast_dsp::features.

Referenced by ast_sip_session_alloc(), chan_pjsip_cng_tone_detected(), chan_pjsip_read_stream(), dahdi_dtmf_detect_disable(), dahdi_dtmf_detect_enable(), dahdi_handle_dtmf(), dahdi_new(), dahdi_read(), dahdi_setoption(), detect_write(), fax_detect_new(), mbl_load_device(), my_handle_dtmf(), my_set_waitingfordt(), ooh323_new(), pjsip_acf_dtmf_mode_write(), read_mf_digits(), read_sf_digits(), scan_exec(), set_caps(), and wait_exec().

ast_dsp_set_freqmode()

int ast_dsp_set_freqmode	(	struct ast_dsp *	dsp,
		int	freq,
		int	dur,
		int	db,
		int	squelch
	)

Set arbitrary frequency detection mode.

Definition at line 1876 of file dsp.c.

{
    if (freq > 0) {
        dsp->freqmode = 1;
        ast_freq_detect_init(dsp, freq, dur, db, squelch);
    } else {
        dsp->freqmode = 0;
    }
    return 0;
}

References ast_freq_detect_init(), db, and ast_dsp::freqmode.

Referenced by detect_write(), read_sf_digits(), scan_exec(), and wait_exec().

ast_dsp_set_threshold()

void ast_dsp_set_threshold	(	struct ast_dsp *	dsp,
		int	threshold
	)

Set the minimum average magnitude threshold to determine talking by the DSP.

Definition at line 1792 of file dsp.c.

{
    dsp->threshold = threshold;
}

References ast_dsp::threshold.

Referenced by __ast_play_and_record(), do_waiting(), fax_session_new(), handle_recordfile(), isAnsweringMachine(), record_exec(), set_softmix_bridge_data(), and set_talk_detect().

ast_dsp_silence()

int ast_dsp_silence	(	struct ast_dsp *	dsp,
		struct ast_frame *	f,
		int *	totalsilence
	)

Process the audio frame for silence.

Parameters

dsp	DSP processing audio media.
f	Audio frame to process.
totalsilence	Variable to set to the total accumulated silence in ms seen by the DSP since the last noise.

Returns

Non-zero if the frame is silence.

Definition at line 1492 of file dsp.c.

{
    return ast_dsp_silence_noise_with_energy(dsp, f, totalsilence, NULL, 0);
}

References ast_dsp_silence_noise_with_energy(), and NULL.

Referenced by __ast_play_and_record(), background_detect_exec(), conf_run(), debug_check_frame_for_silence(), handle_recordfile(), isAnsweringMachine(), record_exec(), and talk_detect_audiohook_cb().

ast_dsp_silence_noise_with_energy()

static int ast_dsp_silence_noise_with_energy ( struct ast_dsp *  dsp, struct ast_frame *  f, int *  total, int *  frames_energy, int  noise  )

static

Definition at line 1441 of file dsp.c.

{
    short *s;
    int len;
    int x;
    unsigned char *odata;
 
    if (!f) {
        return 0;
    }
 
    if (f->frametype != AST_FRAME_VOICE) {
        ast_log(LOG_WARNING, "Can't calculate silence on a non-voice frame\n");
        return 0;
    }
 
    if (ast_format_cache_is_slinear(f->subclass.format)) {
        s = f->data.ptr;
        len = f->datalen/2;
    } else {
        odata = f->data.ptr;
        len = f->datalen;
        if (ast_format_cmp(f->subclass.format, ast_format_ulaw) == AST_FORMAT_CMP_EQUAL) {
            s = ast_alloca(len * 2);
            for (x = 0; x < len; x++) {
                s[x] = AST_MULAW(odata[x]);
            }
        } else if (ast_format_cmp(f->subclass.format, ast_format_alaw) == AST_FORMAT_CMP_EQUAL) {
            s = ast_alloca(len * 2);
            for (x = 0; x < len; x++) {
                s[x] = AST_ALAW(odata[x]);
            }
        } else {
            ast_log(LOG_WARNING, "Can only calculate silence on signed-linear, alaw or ulaw frames, %s not supported\n",
                ast_format_get_name(f->subclass.format));
            return 0;
        }
    }
 
    if (noise) {
        return __ast_dsp_silence_noise(dsp, s, len, NULL, total, frames_energy);
    } else {
        return __ast_dsp_silence_noise(dsp, s, len, total, NULL, frames_energy);
    }
}

References __ast_dsp_silence_noise(), AST_ALAW, ast_alloca, ast_format_alaw, ast_format_cache_is_slinear(), ast_format_cmp(), AST_FORMAT_CMP_EQUAL, ast_format_get_name(), ast_format_ulaw, AST_FRAME_VOICE, ast_log, AST_MULAW, ast_frame::data, ast_frame::datalen, ast_frame_subclass::format, ast_frame::frametype, len(), LOG_WARNING, NULL, ast_frame::ptr, ast_frame::subclass, and total.

Referenced by ast_dsp_noise(), ast_dsp_silence(), and ast_dsp_silence_with_energy().

ast_dsp_silence_with_energy()

int ast_dsp_silence_with_energy	(	struct ast_dsp *	dsp,
		struct ast_frame *	f,
		int *	totalsilence,
		int *	frames_energy
	)

Process the audio frame for silence.

Parameters

dsp	DSP processing audio media.
f	Audio frame to process.
totalsilence	Variable to set to the total accumulated silence in ms seen by the DSP since the last noise.
frames_energy	Variable to set to the average energy of the samples in the frame.

Returns

Non-zero if the frame is silence.

Definition at line 1487 of file dsp.c.

{
    return ast_dsp_silence_noise_with_energy(dsp, f, totalsilence, frames_energy, 0);
}

References ast_dsp_silence_noise_with_energy().

Referenced by softmix_bridge_write_voice().

ast_dsp_was_muted()

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.

Since

1.6.1

Definition at line 1910 of file dsp.c.

{
    return (dsp->mute_fragments > 0);
}

References ast_dsp::mute_fragments.

Referenced by dahdi_read().

ast_dtmf_detect_init()

static void ast_dtmf_detect_init ( dtmf_detect_state_t *  s, unsigned int  sample_rate  )

static

Definition at line 533 of file dsp.c.

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

References dtmf_detect_state_t::col_out, dtmf_detect_state_t::current_hit, dtmf_detect_state_t::current_sample, dtmf_col, DTMF_MATRIX_SIZE, dtmf_row, dtmf_detect_state_t::energy, goertzel_init(), dtmf_detect_state_t::hits, dtmf_detect_state_t::lasthit, dtmf_detect_state_t::misses, and dtmf_detect_state_t::row_out.

Referenced by ast_digit_detect_init().

ast_fax_detect_init()

static void ast_fax_detect_init ( struct ast_dsp *  s )

static

Definition at line 513 of file dsp.c.

{
    ast_tone_detect_init(&s->cng_tone_state, FAX_TONE_CNG_FREQ, FAX_TONE_CNG_DURATION, FAX_TONE_CNG_DB, s->sample_rate);
    ast_tone_detect_init(&s->ced_tone_state, FAX_TONE_CED_FREQ, FAX_TONE_CED_DURATION, FAX_TONE_CED_DB, s->sample_rate);
    if (s->faxmode & DSP_FAXMODE_DETECT_SQUELCH) {
        s->cng_tone_state.squelch = 1;
        s->ced_tone_state.squelch = 1;
    }
 
}

References ast_tone_detect_init(), ast_dsp::ced_tone_state, ast_dsp::cng_tone_state, DSP_FAXMODE_DETECT_SQUELCH, FAX_TONE_CED_DB, FAX_TONE_CED_DURATION, FAX_TONE_CED_FREQ, FAX_TONE_CNG_DB, FAX_TONE_CNG_DURATION, FAX_TONE_CNG_FREQ, ast_dsp::faxmode, ast_dsp::sample_rate, and tone_detect_state_t::squelch.

Referenced by __ast_dsp_new(), and ast_dsp_set_faxmode().

ast_freq_detect_init()

static void ast_freq_detect_init ( struct ast_dsp *  s, int  freq, int  dur, int  db, int  squelch  )

static

Definition at line 524 of file dsp.c.

{
    /* we can conveniently just use one of the two fax tone states */
    ast_tone_detect_init(&s->cng_tone_state, freq, dur, db, s->sample_rate);
    if (s->freqmode & squelch) {
        s->cng_tone_state.squelch = 1;
    }
}

References ast_tone_detect_init(), ast_dsp::cng_tone_state, db, ast_dsp::freqmode, ast_dsp::sample_rate, and tone_detect_state_t::squelch.

Referenced by ast_dsp_set_freqmode().

ast_mf_detect_init()

static void ast_mf_detect_init ( mf_detect_state_t *  s, unsigned int  sample_rate  )

static

Definition at line 549 of file dsp.c.

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

References mf_detect_state_t::current_hit, mf_detect_state_t::current_sample, goertzel_init(), mf_detect_state_t::hits, mf_tones, and mf_detect_state_t::tone_out.

Referenced by ast_digit_detect_init().

AST_MODULE_SELF_SYM()

struct ast_module * AST_MODULE_SELF_SYM

(

void

)

Definition at line 2483 of file dsp.c.

ast_tone_detect_init()

static void ast_tone_detect_init ( tone_detect_state_t *  s, int  freq, int  duration, int  amp, unsigned int  sample_rate  )

static

Definition at line 452 of file dsp.c.

{
    int duration_samples;
    float x;
    int periods_in_block;
 
    s->freq = freq;
 
    /* Desired tone duration in samples */
    duration_samples = duration * sample_rate / 1000;
    /* We want to allow 10% deviation of tone duration */
    duration_samples = duration_samples * 9 / 10;
 
    /* If we want to remove tone, it is important to have block size not
       to exceed frame size. Otherwise by the moment tone is detected it is too late
       to squelch it from previous frames. Block size is 20ms at the given sample rate.*/
    s->block_size = (20 * sample_rate) / 1000;
 
    periods_in_block = s->block_size * freq / sample_rate;
 
    /* Make sure we will have at least 5 periods at target frequency for analysis.
       This may make block larger than expected packet and will make squelching impossible
       but at least we will be detecting the tone */
    if (periods_in_block < 5) {
        periods_in_block = 5;
    }
 
    /* Now calculate final block size. It will contain integer number of periods */
    s->block_size = periods_in_block * sample_rate / freq;
 
    /* tone_detect is generally only used to detect fax tones and we
       do not need squelching the fax tones */
    s->squelch = 0;
 
    /* Account for the first and the last block to be incomplete
       and thus no tone will be detected in them */
    s->hits_required = (duration_samples - (s->block_size - 1)) / s->block_size;
 
    goertzel_init(&s->tone, freq, sample_rate);
 
    s->samples_pending = s->block_size;
    s->hit_count = 0;
    s->last_hit = 0;
    s->energy = 0.0;
 
    /* We want tone energy to be amp decibels above the rest of the signal (the noise).
       According to Parseval's theorem the energy computed in time domain equals to energy
       computed in frequency domain. So subtracting energy in the frequency domain (Goertzel result)
       from the energy in the time domain we will get energy of the remaining signal (without the tone
       we are detecting). We will be checking that
        10*log(Ew / (Et - Ew)) > amp
       Calculate threshold so that we will be actually checking
        Ew > Et * threshold
    */
 
    x = pow(10.0, amp / 10.0);
    s->threshold = x / (x + 1);
 
    ast_debug(1, "Setup tone %d Hz, %d ms, block_size=%d, hits_required=%d\n", freq, duration, s->block_size, s->hits_required);
}

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, 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(), and ast_freq_detect_init().

dtmf_detect()

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 699 of file dsp.c.

{
    float row_energy[DTMF_MATRIX_SIZE];
    float col_energy[DTMF_MATRIX_SIZE];
    int i;
    int j;
    int sample;
    short samp;
    int best_row;
    int best_col;
    int hit;
    int limit;
    fragment_t mute = {0, 0};
 
    if (squelch && s->td.dtmf.mute_samples > 0) {
        mute.end = (s->td.dtmf.mute_samples < samples) ? s->td.dtmf.mute_samples : samples;
        s->td.dtmf.mute_samples -= mute.end;
    }
 
    hit = 0;
    for (sample = 0; sample < samples; sample = limit) {
        /* DTMF_GSIZE is optimised to meet the DTMF specs. */
        if ((samples - sample) >= (DTMF_GSIZE - s->td.dtmf.current_sample)) {
            limit = sample + (DTMF_GSIZE - s->td.dtmf.current_sample);
        } else {
            limit = samples;
        }
        /* The following unrolled loop takes only 35% (rough estimate) of the
           time of a rolled loop on the machine on which it was developed */
        for (j = sample; j < limit; j++) {
            samp = amp[j];
            s->td.dtmf.energy += (int32_t) samp * (int32_t) samp;
            /* With GCC 2.95, the following unrolled code seems to take about 35%
               (rough estimate) as long as a neat little 0-3 loop */
            goertzel_sample(s->td.dtmf.row_out, samp);
            goertzel_sample(s->td.dtmf.col_out, samp);
            goertzel_sample(s->td.dtmf.row_out + 1, samp);
            goertzel_sample(s->td.dtmf.col_out + 1, samp);
            goertzel_sample(s->td.dtmf.row_out + 2, samp);
            goertzel_sample(s->td.dtmf.col_out + 2, samp);
            goertzel_sample(s->td.dtmf.row_out + 3, samp);
            goertzel_sample(s->td.dtmf.col_out + 3, samp);
            /* go up to DTMF_MATRIX_SIZE - 1 */
        }
        s->td.dtmf.current_sample += (limit - sample);
        if (s->td.dtmf.current_sample < DTMF_GSIZE) {
            continue;
        }
        /* We are at the end of a DTMF detection block */
        /* Find the peak row and the peak column */
        row_energy[0] = goertzel_result(&s->td.dtmf.row_out[0]);
        col_energy[0] = goertzel_result(&s->td.dtmf.col_out[0]);
 
        for (best_row = best_col = 0, i = 1; i < DTMF_MATRIX_SIZE; i++) {
            row_energy[i] = goertzel_result(&s->td.dtmf.row_out[i]);
            if (row_energy[i] > row_energy[best_row]) {
                best_row = i;
            }
            col_energy[i] = goertzel_result(&s->td.dtmf.col_out[i]);
            if (col_energy[i] > col_energy[best_col]) {
                best_col = i;
            }
        }
        ast_debug(10, "DTMF best '%c' Erow=%.4E Ecol=%.4E Erc=%.4E Et=%.4E\n",
            dtmf_positions[(best_row << 2) + best_col],
            row_energy[best_row], col_energy[best_col],
            row_energy[best_row] + col_energy[best_col], s->td.dtmf.energy);
        hit = 0;
        /* Basic signal level test and the twist test */
        if (row_energy[best_row] >= DTMF_THRESHOLD &&
            col_energy[best_col] >= DTMF_THRESHOLD &&
            col_energy[best_col] < row_energy[best_row] * (relax ? relax_dtmf_reverse_twist : dtmf_reverse_twist) &&
            row_energy[best_row] < col_energy[best_col] * (relax ? relax_dtmf_normal_twist : dtmf_normal_twist)) {
            /* Relative peak test */
            for (i = 0; i < DTMF_MATRIX_SIZE; i++) {
                if ((i != best_col &&
                    col_energy[i] * DTMF_RELATIVE_PEAK_COL > col_energy[best_col]) ||
                    (i != best_row
                     && row_energy[i] * DTMF_RELATIVE_PEAK_ROW > row_energy[best_row])) {
                    break;
                }
            }
            /* ... and fraction of total energy test */
            if (i >= DTMF_MATRIX_SIZE &&
                (row_energy[best_row] + col_energy[best_col]) > DTMF_TO_TOTAL_ENERGY * s->td.dtmf.energy) {
                /* Got a hit */
                hit = dtmf_positions[(best_row << 2) + best_col];
                ast_debug(10, "DTMF hit '%c'\n", hit);
            }
        }
 
/*
 * Adapted from ETSI ES 201 235-3 V1.3.1 (2006-03)
 * (40ms reference is tunable with hits_to_begin and misses_to_end)
 * each hit/miss is 12.75ms with DTMF_GSIZE at 102
 *
 * Character recognition: When not DRC *(1) and then
 *      Shall exist VSC > 40 ms (hits_to_begin)
 *      May exist 20 ms <= VSC <= 40 ms
 *      Shall not exist VSC < 20 ms
 *
 * Character recognition: When DRC and then
 *      Shall cease Not VSC > 40 ms (misses_to_end)
 *      May cease 20 ms >= Not VSC >= 40 ms
 *      Shall not cease Not VSC < 20 ms
 *
 * *(1) or optionally a different digit recognition condition
 *
 * Legend: VSC The continuous existence of a valid signal condition.
 *      Not VSC The continuous non-existence of valid signal condition.
 *      DRC The existence of digit recognition condition.
 *      Not DRC The non-existence of digit recognition condition.
 */
 
/*
 * Example: hits_to_begin=2 misses_to_end=3
 * -------A last_hit=A hits=0&1
 * ------AA hits=2 current_hit=A misses=0       BEGIN A
 * -----AA- misses=1 last_hit=' ' hits=0
 * ----AA-- misses=2
 * ---AA--- misses=3 current_hit=' '            END A
 * --AA---B last_hit=B hits=0&1
 * -AA---BC last_hit=C hits=0&1
 * AA---BCC hits=2 current_hit=C misses=0       BEGIN C
 * A---BCC- misses=1 last_hit=' ' hits=0
 * ---BCC-C misses=0 last_hit=C hits=0&1
 * --BCC-CC misses=0
 *
 * Example: hits_to_begin=3 misses_to_end=2
 * -------A last_hit=A hits=0&1
 * ------AA hits=2
 * -----AAA hits=3 current_hit=A misses=0       BEGIN A
 * ----AAAB misses=1 last_hit=B hits=0&1
 * ---AAABB misses=2 current_hit=' ' hits=2     END A
 * --AAABBB hits=3 current_hit=B misses=0       BEGIN B
 * -AAABBBB misses=0
 *
 * Example: hits_to_begin=2 misses_to_end=2
 * -------A last_hit=A hits=0&1
 * ------AA hits=2 current_hit=A misses=0       BEGIN A
 * -----AAB misses=1 hits=0&1
 * ----AABB misses=2 current_hit=' ' hits=2 current_hit=B misses=0 BEGIN B
 * ---AABBB misses=0
 */
 
        if (s->td.dtmf.current_hit) {
            /* We are in the middle of a digit already */
            if (hit != s->td.dtmf.current_hit) {
                s->td.dtmf.misses++;
                if (s->td.dtmf.misses == dtmf_misses_to_end) {
                    /* There were enough misses to consider digit ended */
                    s->td.dtmf.current_hit = 0;
                }
            } else {
                s->td.dtmf.misses = 0;
                /* Current hit was same as last, so increment digit duration (of last digit) */
                s->digitlen[s->current_digits - 1] += DTMF_GSIZE;
            }
        }
 
        /* Look for a start of a new digit no matter if we are already in the middle of some
           digit or not. This is because hits_to_begin may be smaller than misses_to_end
           and we may find begin of new digit before we consider last one ended. */
 
        if (hit != s->td.dtmf.lasthit) {
            s->td.dtmf.lasthit = hit;
            s->td.dtmf.hits = 0;
        }
        if (hit && hit != s->td.dtmf.current_hit) {
            s->td.dtmf.hits++;
            if (s->td.dtmf.hits == dtmf_hits_to_begin) {
                store_digit(s, hit);
                s->digitlen[s->current_digits - 1] = dtmf_hits_to_begin * DTMF_GSIZE;
                s->td.dtmf.current_hit = hit;
                s->td.dtmf.misses = 0;
            }
        }
 
        /* If we had a hit in this block, include it into mute fragment */
        if (squelch && hit) {
            if (mute.end < sample - DTMF_GSIZE) {
                /* There is a gap between fragments */
                mute_fragment(dsp, &mute);
                mute.start = (sample > DTMF_GSIZE) ? (sample - DTMF_GSIZE) : 0;
            }
            mute.end = limit + DTMF_GSIZE;
        }
 
        /* Reinitialise the detector for the next block */
        for (i = 0; i < DTMF_MATRIX_SIZE; i++) {
            goertzel_reset(&s->td.dtmf.row_out[i]);
            goertzel_reset(&s->td.dtmf.col_out[i]);
        }
        s->td.dtmf.energy = 0.0;
        s->td.dtmf.current_sample = 0;
    }
 
    if (squelch && mute.end) {
        if (mute.end > samples) {
            s->td.dtmf.mute_samples = mute.end - samples;
            mute.end = samples;
        }
        mute_fragment(dsp, &mute);
    }
 
    return (s->td.dtmf.current_hit);    /* return the debounced hit */
}

References ast_debug, dtmf_detect_state_t::col_out, digit_detect_state_t::current_digits, dtmf_detect_state_t::current_hit, dtmf_detect_state_t::current_sample, digit_detect_state_t::digitlen, digit_detect_state_t::dtmf, DTMF_GSIZE, dtmf_hits_to_begin, DTMF_MATRIX_SIZE, dtmf_misses_to_end, dtmf_normal_twist, dtmf_positions, DTMF_RELATIVE_PEAK_COL, DTMF_RELATIVE_PEAK_ROW, dtmf_reverse_twist, DTMF_THRESHOLD, DTMF_TO_TOTAL_ENERGY, fragment_t::end, dtmf_detect_state_t::energy, goertzel_reset(), goertzel_result(), goertzel_sample(), dtmf_detect_state_t::hits, dtmf_detect_state_t::lasthit, dtmf_detect_state_t::misses, mute_fragment(), dtmf_detect_state_t::mute_samples, relax_dtmf_normal_twist, relax_dtmf_reverse_twist, dtmf_detect_state_t::row_out, fragment_t::start, store_digit(), and digit_detect_state_t::td.

Referenced by ast_dsp_process().

goertzel_init()

static void goertzel_init ( goertzel_state_t *  s, float  freq, unsigned int  sample_rate  )

inlinestatic

Definition at line 378 of file dsp.c.

{
    s->v2 = s->v3 = s->chunky = 0;
    s->fac = (int)(32768.0 * 2.0 * cos(2.0 * M_PI * freq / sample_rate));
}

References goertzel_state_t::chunky, cos, goertzel_state_t::fac, M_PI, 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().

goertzel_reset()

static void goertzel_reset ( goertzel_state_t *  s )

inlinestatic

Definition at line 384 of file dsp.c.

{
    s->v2 = s->v3 = s->chunky = 0;
}

References goertzel_state_t::chunky, goertzel_state_t::v2, and goertzel_state_t::v3.

Referenced by ast_dsp_digitreset(), dtmf_detect(), mf_detect(), and tone_detect().

goertzel_result()

static float goertzel_result ( goertzel_state_t *  s )

inlinestatic

Definition at line 364 of file dsp.c.

{
    goertzel_result_t r;
 
    r.value = (s->v3 * s->v3) + (s->v2 * s->v2);
    r.value -= ((s->v2 * s->v3) >> 15) * s->fac;
    /*
     * We have to double the exponent because we multiplied the
     * previous sample calculation values together.
     */
    r.power = s->chunky * 2;
    return (float)r.value * (float)(1 << r.power);
}

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(), dtmf_detect(), mf_detect(), and tone_detect().

goertzel_sample()

static void goertzel_sample ( goertzel_state_t *  s, short  sample  )

inlinestatic

Definition at line 338 of file dsp.c.

{
    int v1;
 
    /*
     * Shift previous values so
     * v1 is previous previous value
     * v2 is previous value
     * until the new v3 is calculated.
     */
    v1 = s->v2;
    s->v2 = s->v3;
 
    /* Discard the binary fraction introduced by s->fac */
    s->v3 = (s->fac * s->v2) >> 15;
    /* Scale sample to match previous values */
    s->v3 = s->v3 - v1 + (sample >> s->chunky);
 
    if (abs(s->v3) > (1 << 15)) {
        /* The result is now too large so increase the chunky power. */
        s->chunky++;
        s->v3 = s->v3 >> 1;
        s->v2 = s->v2 >> 1;
    }
}

References abs, 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(), mf_detect(), and tone_detect().

load_module()

static int load_module ( void  )

static

Definition at line 2459 of file dsp.c.

{
    if (_dsp_init(0)) {
        return AST_MODULE_LOAD_FAILURE;
    }
 
    AST_TEST_REGISTER(test_dsp_fax_detect);
    AST_TEST_REGISTER(test_dsp_dtmf_detect);
 
    return AST_MODULE_LOAD_SUCCESS;
}

References _dsp_init(), AST_MODULE_LOAD_FAILURE, AST_MODULE_LOAD_SUCCESS, and AST_TEST_REGISTER.

mf_detect()

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 907 of file dsp.c.

{
    float energy[6];
    int best;
    int second_best;
    int i;
    int j;
    int sample;
    short samp;
    int hit;
    int limit;
    fragment_t mute = {0, 0};
 
    if (squelch && s->td.mf.mute_samples > 0) {
        mute.end = (s->td.mf.mute_samples < samples) ? s->td.mf.mute_samples : samples;
        s->td.mf.mute_samples -= mute.end;
    }
 
    hit = 0;
    for (sample = 0; sample < samples; sample = limit) {
        /* 80 is optimised to meet the MF specs. */
        /* XXX So then why is MF_GSIZE defined as 120? */
        if ((samples - sample) >= (MF_GSIZE - s->td.mf.current_sample)) {
            limit = sample + (MF_GSIZE - s->td.mf.current_sample);
        } else {
            limit = samples;
        }
        /* The following unrolled loop takes only 35% (rough estimate) of the
           time of a rolled loop on the machine on which it was developed */
        for (j = sample; j < limit; j++) {
            /* With GCC 2.95, the following unrolled code seems to take about 35%
               (rough estimate) as long as a neat little 0-3 loop */
            samp = amp[j];
            goertzel_sample(s->td.mf.tone_out, samp);
            goertzel_sample(s->td.mf.tone_out + 1, samp);
            goertzel_sample(s->td.mf.tone_out + 2, samp);
            goertzel_sample(s->td.mf.tone_out + 3, samp);
            goertzel_sample(s->td.mf.tone_out + 4, samp);
            goertzel_sample(s->td.mf.tone_out + 5, samp);
        }
        s->td.mf.current_sample += (limit - sample);
        if (s->td.mf.current_sample < MF_GSIZE) {
            continue;
        }
        /* We're at the end of an MF detection block.  */
        /* Find the two highest energies. The spec says to look for
           two tones and two tones only. Taking this literally -ie
           only two tones pass the minimum threshold - doesn't work
           well. The sinc function mess, due to rectangular windowing
           ensure that! Find the two highest energies and ensure they
           are considerably stronger than any of the others. */
        energy[0] = goertzel_result(&s->td.mf.tone_out[0]);
        energy[1] = goertzel_result(&s->td.mf.tone_out[1]);
        if (energy[0] > energy[1]) {
            best = 0;
            second_best = 1;
        } else {
            best = 1;
            second_best = 0;
        }
        /*endif*/
        for (i = 2; i < 6; i++) {
            energy[i] = goertzel_result(&s->td.mf.tone_out[i]);
            if (energy[i] >= energy[best]) {
                second_best = best;
                best = i;
            } else if (energy[i] >= energy[second_best]) {
                second_best = i;
            }
        }
        /* Basic signal level and twist tests */
        hit = 0;
        if (energy[best] >= BELL_MF_THRESHOLD && energy[second_best] >= BELL_MF_THRESHOLD
            && energy[best] < energy[second_best]*BELL_MF_TWIST
            && energy[best] * BELL_MF_TWIST > energy[second_best]) {
            /* Relative peak test */
            hit = -1;
            for (i = 0; i < 6; i++) {
                if (i != best && i != second_best) {
                    if (energy[i]*BELL_MF_RELATIVE_PEAK >= energy[second_best]) {
                        /* The best two are not clearly the best */
                        hit = 0;
                        break;
                    }
                }
            }
        }
        if (hit) {
            /* Get the values into ascending order */
            if (second_best < best) {
                i = best;
                best = second_best;
                second_best = i;
            }
            best = best * 5 + second_best - 1;
            hit = bell_mf_positions[best];
            /* Look for two successive similar results */
            /* The logic in the next test is:
               For KP we need 4 successive identical clean detects, with
               two blocks of something different preceeding it. For anything
               else we need two successive identical clean detects, with
               two blocks of something different preceeding it. */
            if (hit == s->td.mf.hits[4] && hit == s->td.mf.hits[3] &&
               ((hit != '*' && hit != s->td.mf.hits[2] && hit != s->td.mf.hits[1])||
                (hit == '*' && hit == s->td.mf.hits[2] && hit != s->td.mf.hits[1] &&
                hit != s->td.mf.hits[0]))) {
                store_digit(s, hit);
            }
        }
 
 
        if (hit != s->td.mf.hits[4] && hit != s->td.mf.hits[3]) {
            /* Two successive block without a hit terminate current digit */
            s->td.mf.current_hit = 0;
        }
 
        s->td.mf.hits[0] = s->td.mf.hits[1];
        s->td.mf.hits[1] = s->td.mf.hits[2];
        s->td.mf.hits[2] = s->td.mf.hits[3];
        s->td.mf.hits[3] = s->td.mf.hits[4];
        s->td.mf.hits[4] = hit;
 
        /* If we had a hit in this block, include it into mute fragment */
        if (squelch && hit) {
            if (mute.end < sample - MF_GSIZE) {
                /* There is a gap between fragments */
                mute_fragment(dsp, &mute);
                mute.start = (sample > MF_GSIZE) ? (sample - MF_GSIZE) : 0;
            }
            mute.end = limit + MF_GSIZE;
        }
 
        /* Reinitialise the detector for the next block */
        for (i = 0; i < 6; i++) {
            goertzel_reset(&s->td.mf.tone_out[i]);
        }
        s->td.mf.current_sample = 0;
    }
 
    if (squelch && mute.end) {
        if (mute.end > samples) {
            s->td.mf.mute_samples = mute.end - samples;
            mute.end = samples;
        }
        mute_fragment(dsp, &mute);
    }
 
    return (s->td.mf.current_hit); /* return the debounced hit */
}

References bell_mf_positions, BELL_MF_RELATIVE_PEAK, BELL_MF_THRESHOLD, BELL_MF_TWIST, mf_detect_state_t::current_hit, mf_detect_state_t::current_sample, fragment_t::end, goertzel_reset(), goertzel_result(), goertzel_sample(), mf_detect_state_t::hits, digit_detect_state_t::mf, MF_GSIZE, mute_fragment(), mf_detect_state_t::mute_samples, fragment_t::start, store_digit(), digit_detect_state_t::td, and mf_detect_state_t::tone_out.

Referenced by ast_dsp_process().

mute_fragment()

static void mute_fragment ( struct ast_dsp *  dsp, fragment_t *  fragment  )

static

Definition at line 442 of file dsp.c.

{
    if (dsp->mute_fragments >= ARRAY_LEN(dsp->mute_data)) {
        ast_log(LOG_ERROR, "Too many fragments to mute. Ignoring\n");
        return;
    }
 
    dsp->mute_data[dsp->mute_fragments++] = *fragment;
}

References ARRAY_LEN, ast_log, LOG_ERROR, ast_dsp::mute_data, and ast_dsp::mute_fragments.

Referenced by dtmf_detect(), mf_detect(), and tone_detect().

pair_there()

static int pair_there ( float  p1, float  p2, float  i1, float  i2, float  e  )

inlinestatic

Definition at line 1058 of file dsp.c.

{
    /* See if p1 and p2 are there, relative to i1 and i2 and total energy */
    /* Make sure absolute levels are high enough */
    if ((p1 < TONE_MIN_THRESH) || (p2 < TONE_MIN_THRESH)) {
        return 0;
    }
    /* Amplify ignored stuff */
    i2 *= TONE_THRESH;
    i1 *= TONE_THRESH;
    e *= TONE_THRESH;
    /* Check first tone */
    if ((p1 < i1) || (p1 < i2) || (p1 < e)) {
        return 0;
    }
    /* And second */
    if ((p2 < i1) || (p2 < i2) || (p2 < e)) {
        return 0;
    }
    /* Guess it's there... */
    return 1;
}

References TONE_MIN_THRESH, and TONE_THRESH.

Referenced by __ast_dsp_call_progress().

reload_module()

static int reload_module ( void  )

static

Definition at line 2471 of file dsp.c.

{
    return _dsp_init(1);
}

References _dsp_init().

store_digit()

static void store_digit ( digit_detect_state_t *  s, char  digit  )

static

Definition at line 686 of file dsp.c.

{
    s->detected_digits++;
    if (s->current_digits < MAX_DTMF_DIGITS) {
        s->digitlen[s->current_digits] = 0;
        s->digits[s->current_digits++] = digit;
        s->digits[s->current_digits] = '\0';
    } else {
        ast_log(LOG_WARNING, "Digit lost due to full buffer\n");
        s->lost_digits++;
    }
}

References ast_log, digit_detect_state_t::current_digits, digit_detect_state_t::detected_digits, digit, digit_detect_state_t::digitlen, digit_detect_state_t::digits, LOG_WARNING, digit_detect_state_t::lost_digits, and MAX_DTMF_DIGITS.

Referenced by dtmf_detect(), and mf_detect().

tone_detect()

static int tone_detect ( struct ast_dsp *  dsp, tone_detect_state_t *  s, int16_t *  amp, int  samples  )

static

Definition at line 575 of file dsp.c.

{
    float tone_energy;
    int i;
    int hit = 0;
    int limit;
    int res = 0;
    int16_t *ptr;
    short samp;
    int start, end;
    fragment_t mute = {0, 0};
 
    if (s->squelch && s->mute_samples > 0) {
        mute.end = (s->mute_samples < samples) ? s->mute_samples : samples;
        s->mute_samples -= mute.end;
    }
 
    for (start = 0; start < samples; start = end) {
        /* Process in blocks. */
        limit = samples - start;
        if (limit > s->samples_pending) {
            limit = s->samples_pending;
        }
        end = start + limit;
 
        for (i = limit, ptr = amp ; i > 0; i--, ptr++) {
            samp = *ptr;
            /* signed 32 bit int should be enough to square any possible signed 16 bit value */
            s->energy += (int32_t) samp * (int32_t) samp;
 
            goertzel_sample(&s->tone, samp);
        }
 
        s->samples_pending -= limit;
 
        if (s->samples_pending) {
            /* Finished incomplete (last) block */
            break;
        }
 
        tone_energy = goertzel_result(&s->tone);
 
        /* Scale to make comparable */
        tone_energy *= 2.0;
        s->energy *= s->block_size;
 
        /* Add 1 to hit_count when logging, since it doesn't get incremented until later in the loop. */
        if (TONE_THRESHOLD <= tone_energy
            && tone_energy > s->energy * s->threshold) {
            ast_debug(9, "%d Hz tone Hit! [%d/%d] Ew=%.4E, Et=%.4E, s/n=%10.2f\n", s->freq, s->hit_count + 1, s->hits_required, tone_energy, s->energy, tone_energy / (s->energy - tone_energy));
            hit = 1;
        } else {
            ast_debug(10, "%d Hz tone [%d/%d] Ew=%.4E, Et=%.4E, s/n=%10.2f\n", s->freq, s->hit_count, s->hits_required, tone_energy, s->energy, tone_energy / (s->energy - tone_energy));
            hit = 0;
        }
 
        if (s->hit_count) {
            s->hit_count++;
        }
 
        if (hit == s->last_hit) {
            if (!hit) {
                /* Two successive misses. Tone ended */
                if (s->hit_count) {
                    ast_debug(9, "Partial detect expired after %d/%d hits\n", s->hit_count, s->hits_required);
                }
                s->hit_count = 0;
            } else if (!s->hit_count) {
                s->hit_count++;
            }
        }
 
        if (s->hit_count == s->hits_required) {
            ast_debug(1, "%d Hz tone detected\n", s->freq);
            res = 1;
        }
 
        s->last_hit = hit;
 
        /* If we had a hit in this block, include it into mute fragment */
        if (s->squelch && hit) {
            if (mute.end < start - s->block_size) {
                /* There is a gap between fragments */
                mute_fragment(dsp, &mute);
                mute.start = (start > s->block_size) ? (start - s->block_size) : 0;
            }
            mute.end = end + s->block_size;
        }
 
        /* Reinitialise the detector for the next block */
        /* Reset for the next block */
        goertzel_reset(&s->tone);
 
        /* Advance to the next block */
        s->energy = 0.0;
        s->samples_pending = s->block_size;
 
        amp += limit;
    }
 
    if (s->squelch && mute.end) {
        if (mute.end > samples) {
            s->mute_samples = mute.end - samples;
            mute.end = samples;
        }
        mute_fragment(dsp, &mute);
    }
 
    return res;
}

References ast_debug, tone_detect_state_t::block_size, end, fragment_t::end, tone_detect_state_t::energy, tone_detect_state_t::freq, goertzel_reset(), goertzel_result(), goertzel_sample(), tone_detect_state_t::hit_count, tone_detect_state_t::hits_required, tone_detect_state_t::last_hit, mute_fragment(), tone_detect_state_t::mute_samples, tone_detect_state_t::samples_pending, tone_detect_state_t::squelch, fragment_t::start, tone_detect_state_t::threshold, tone_detect_state_t::tone, and TONE_THRESHOLD.

Referenced by ast_dsp_process().

unload_module()

static int unload_module ( void  )

static

Definition at line 2451 of file dsp.c.

{
    AST_TEST_UNREGISTER(test_dsp_fax_detect);
    AST_TEST_UNREGISTER(test_dsp_dtmf_detect);
 
    return 0;
}

References AST_TEST_UNREGISTER.

Variable Documentation

__mod_info

struct ast_module_info __mod_info = { .name = AST_MODULE, .flags = AST_MODFLAG_GLOBAL_SYMBOLS | AST_MODFLAG_LOAD_ORDER , .description = "DSP" , .key = ASTERISK_GPL_KEY , .buildopt_sum = AST_BUILDOPT_SUM, .support_level = AST_MODULE_SUPPORT_CORE, .load = load_module, .unload = unload_module, .reload = reload_module, .load_pri = AST_MODPRI_CORE, .requires = "extconfig", }

static

Definition at line 2483 of file dsp.c.

aliases

struct progalias aliases[]

static

Referenced by ast_dsp_set_call_progress_zone(), handle_voicemail_show_aliases(), ooh323c_set_aliases(), and queue_mwi_event().

ast_module_info

const struct ast_module_info* ast_module_info = &__mod_info

static

Definition at line 2483 of file dsp.c.

bell_mf_positions

const char bell_mf_positions[] = "1247C-358A--69*---0B----#"

static

Definition at line 329 of file dsp.c.

Referenced by mf_detect().

DEFAULT_SILENCE_THRESHOLD

const int DEFAULT_SILENCE_THRESHOLD = 256

static

The default silence threshold we will use if an alternate configured value is not present or is invalid.

Definition at line 245 of file dsp.c.

Referenced by _dsp_init().

dtmf_col

const float dtmf_col[]

static

Initial value:

= {
    1209.0, 1336.0, 1477.0, 1633.0
}

Definition at line 322 of file dsp.c.

                                {
    1209.0, 1336.0, 1477.0, 1633.0
};

Referenced by ast_dtmf_detect_init().

dtmf_hits_to_begin

int dtmf_hits_to_begin

static

Definition at line 335 of file dsp.c.

Referenced by _dsp_init(), and dtmf_detect().

dtmf_misses_to_end

int dtmf_misses_to_end

static

Definition at line 336 of file dsp.c.

Referenced by _dsp_init(), and dtmf_detect().

dtmf_normal_twist

float dtmf_normal_twist

static

Definition at line 331 of file dsp.c.

Referenced by _dsp_init(), and dtmf_detect().

dtmf_positions

const char dtmf_positions[] = "123A" "456B" "789C" "*0#D"

static

Definition at line 328 of file dsp.c.

Referenced by dtmf_detect().

dtmf_reverse_twist

float dtmf_reverse_twist

static

Definition at line 332 of file dsp.c.

Referenced by _dsp_init(), and dtmf_detect().

dtmf_row

const float dtmf_row[]

static

Initial value:

= {
    697.0,  770.0,  852.0,  941.0
}

Definition at line 319 of file dsp.c.

                                {
    697.0,  770.0,  852.0,  941.0
};

Referenced by ast_dtmf_detect_init().

mf_tones

const float mf_tones[]

static

Initial value:

= {
    700.0, 900.0, 1100.0, 1300.0, 1500.0, 1700.0
}

Definition at line 325 of file dsp.c.

                                {
    700.0, 900.0, 1100.0, 1300.0, 1500.0, 1700.0
};

Referenced by ast_mf_detect_init(), and ast_senddigit_mf_begin().

modes

struct progress modes[]

static

Initial value:

= {
    { GSAMP_SIZE_NA, { 350, 440, 480, 620, 950, 1400, 1800 } }, 
    { GSAMP_SIZE_CR, { 425 } },                 
    { GSAMP_SIZE_UK, { 350, 400, 440 } },               
}

Referenced by ast_dsp_prog_reset().

relax_dtmf_normal_twist

float relax_dtmf_normal_twist

static

Definition at line 333 of file dsp.c.

Referenced by _dsp_init(), and dtmf_detect().

relax_dtmf_reverse_twist

float relax_dtmf_reverse_twist

static

Definition at line 334 of file dsp.c.

Referenced by _dsp_init(), and dtmf_detect().

thresholds

int thresholds[THRESHOLD_MAX]

static

Definition at line 330 of file dsp.c.

Referenced by _dsp_init(), and ast_dsp_get_threshold_from_settings().