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. More...

#define	DSP_HISTORY 15

#define	DTMF_GSIZE 102

#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_AMPLITUDE_MAX 0x7fff /* Max signed linear amplitude */

#define	TONE_AMPLITUDE_MIN 80 /* Min signed linear amplitude detectable */

#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. More...

int	ast_dsp_call_progress (struct ast_dsp dsp, struct ast_frame inf)
	Scans for progress indication in audio. More...

void	ast_dsp_digitreset (struct ast_dsp *dsp)
	Reset DTMF detector. More...

void	ast_dsp_free (struct ast_dsp *dsp)

int	ast_dsp_get_features (struct ast_dsp *dsp)
	Get features. More...

unsigned int	ast_dsp_get_sample_rate (const struct ast_dsp *dsp)
	Retrieve the sample rate this DSP structure was created with. More...

int	ast_dsp_get_tcount (struct ast_dsp *dsp)
	Get tcount (Threshold counter) More...

int	ast_dsp_get_threshold_from_settings (enum threshold which)
	Get silence threshold from dsp.conf. More...

int	ast_dsp_get_tstate (struct ast_dsp *dsp)
	Get tstate (Tone State) More...

struct ast_dsp *	ast_dsp_new (void)
	Allocates a new dsp, assumes 8khz for internal sample rate. More...

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. More...

int	ast_dsp_noise (struct ast_dsp dsp, struct ast_frame f, int *totalnoise)
	Process the audio frame for noise. More...

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. More...

static void	ast_dsp_prog_reset (struct ast_dsp *dsp)

void	ast_dsp_reset (struct ast_dsp *dsp)
	Reset total silence count. More...

void	ast_dsp_set_busy_count (struct ast_dsp *dsp, int cadences)
	Set number of required cadences for busy. More...

void	ast_dsp_set_busy_pattern (struct ast_dsp dsp, const struct ast_dsp_busy_pattern cadence)
	Set expected lengths of the busy tone. More...

int	ast_dsp_set_call_progress_zone (struct ast_dsp dsp, char zone)
	Set zone for doing progress detection. More...

int	ast_dsp_set_digitmode (struct ast_dsp *dsp, int digitmode)
	Set digit mode. More...

int	ast_dsp_set_faxmode (struct ast_dsp *dsp, int faxmode)
	Set fax mode. More...

void	ast_dsp_set_features (struct ast_dsp *dsp, int features)
	Select feature set. More...

int	ast_dsp_set_freqmode (struct ast_dsp *dsp, int freq, int dur, int db, int squelch)
	Set arbitrary frequency detection mode. More...

void	ast_dsp_set_threshold (struct ast_dsp *dsp, int threshold)
	Set the minimum average magnitude threshold to determine talking by the DSP. More...

int	ast_dsp_silence (struct ast_dsp dsp, struct ast_frame f, int *totalsilence)
	Process the audio frame for silence. More...

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. More...

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. More...

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)

	AST_TEST_DEFINE (test_dsp_fax_detect)

	AST_TEST_DEFINE (test_dsp_dtmf_detect)

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	test_dtmf_amplitude_sweep (struct ast_test test, struct ast_dsp dsp, int digit_index)

static int	test_dtmf_twist_sweep (struct ast_test test, struct ast_dsp dsp, int digit_index)

static void	test_dual_sample_gen (short *slin_buf, int samples, int rate, int f1, short a1, int f2, short a2)

static int	test_tone_amplitude_sweep (struct ast_test test, struct ast_dsp dsp, tone_detect_state_t *tone_state)

static int	test_tone_freq_sweep (struct ast_test test, struct ast_dsp dsp, tone_detect_state_t *tone_state, short amplitude)

static void	test_tone_sample_gen (short *slin_buf, int samples, int rate, int freq, short amplitude)

static void	test_tone_sample_gen_add (short *slin_buf, int samples, int rate, int freq, short amplitude)

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 = "This paragraph is copyright (c) 2006 by Digium, Inc. \In order for your module to load, it must return this \key via a function called \"key\". Any code which \includes this paragraph must be licensed under the GNU \General Public License version 2 or later (at your \option). In addition to Digium's general reservations \of rights, Digium expressly reserves the right to \allow other parties to license this paragraph under \different terms. Any use of Digium, Inc. trademarks or \logos (including \"Asterisk\" or \"Digium\") without \express written permission of Digium, Inc. is prohibited.\n" , .buildopt_sum = "30ef0c93b36035ec78c9cfd712d36d9b" , .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. More...

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 200 of file dsp.c.

Referenced by mf_detect().

◆ BELL_MF_THRESHOLD

#define BELL_MF_THRESHOLD 1.6e9

Definition at line 198 of file dsp.c.

Referenced by mf_detect().

◆ BELL_MF_TWIST

#define BELL_MF_TWIST 4.0 /* 6dB */

Definition at line 199 of file dsp.c.

Referenced by mf_detect().

◆ CONFIG_FILE_NAME

#define CONFIG_FILE_NAME "dsp.conf"

Definition at line 245 of file dsp.c.

Referenced by _dsp_init().

◆ DEF_DTMF_HITS_TO_BEGIN

#define DEF_DTMF_HITS_TO_BEGIN 2

Definition at line 232 of file dsp.c.

Referenced by _dsp_init().

◆ DEF_DTMF_MISSES_TO_END

#define DEF_DTMF_MISSES_TO_END 3

Definition at line 237 of file dsp.c.

Referenced by _dsp_init().

◆ DEF_DTMF_NORMAL_TWIST

#define DEF_DTMF_NORMAL_TWIST 6.31 /* 8.0dB */

Definition at line 183 of file dsp.c.

Referenced by _dsp_init(), and test_dtmf_twist_sweep().

◆ DEF_DTMF_REVERSE_TWIST

#define DEF_DTMF_REVERSE_TWIST 2.51 /* 4.01dB */

Definition at line 190 of file dsp.c.

Referenced by _dsp_init(), and test_dtmf_twist_sweep().

◆ DEF_RELAX_DTMF_NORMAL_TWIST

#define DEF_RELAX_DTMF_NORMAL_TWIST 6.31 /* 8.0dB */

Definition at line 184 of file dsp.c.

Referenced by _dsp_init().

◆ DEF_RELAX_DTMF_REVERSE_TWIST

#define DEF_RELAX_DTMF_REVERSE_TWIST 3.98 /* 6.0dB */

Definition at line 191 of file dsp.c.

Referenced by _dsp_init().

◆ DEFAULT_SAMPLE_RATE

#define DEFAULT_SAMPLE_RATE 8000

Definition at line 221 of file dsp.c.

Referenced by ast_dsp_new(), test_dtmf_amplitude_sweep(), test_dtmf_twist_sweep(), test_tone_amplitude_sweep(), and test_tone_freq_sweep().

◆ 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.

Referenced by __ast_dsp_new().

◆ DSP_HISTORY

#define DSP_HISTORY 15

Remember last 15 units

Definition at line 151 of file dsp.c.

Referenced by __ast_dsp_new(), __ast_dsp_silence_noise(), ast_dsp_busydetect(), and ast_dsp_set_busy_count().

◆ DTMF_GSIZE

#define DTMF_GSIZE 102

Definition at line 227 of file dsp.c.

Referenced by dtmf_detect(), test_dtmf_amplitude_sweep(), and test_dtmf_twist_sweep().

◆ DTMF_RELATIVE_PEAK_COL

#define DTMF_RELATIVE_PEAK_COL 6.3 /* 8dB */

Definition at line 195 of file dsp.c.

Referenced by dtmf_detect().

◆ DTMF_RELATIVE_PEAK_ROW

#define DTMF_RELATIVE_PEAK_ROW 6.3 /* 8dB */

Definition at line 194 of file dsp.c.

Referenced by dtmf_detect().

◆ DTMF_THRESHOLD

#define DTMF_THRESHOLD 8.0e7

Definition at line 180 of file dsp.c.

Referenced by dtmf_detect().

◆ DTMF_TO_TOTAL_ENERGY

#define DTMF_TO_TOTAL_ENERGY 42.0

Definition at line 196 of file dsp.c.

Referenced by dtmf_detect().

◆ FAX_TONE_CED_DB

#define FAX_TONE_CED_DB 16

Definition at line 219 of file dsp.c.

Referenced by ast_fax_detect_init().

◆ FAX_TONE_CED_DURATION

#define FAX_TONE_CED_DURATION 2600 /* ms */

Definition at line 218 of file dsp.c.

Referenced by ast_fax_detect_init().

◆ FAX_TONE_CED_FREQ

#define FAX_TONE_CED_FREQ 2100

Definition at line 217 of file dsp.c.

Referenced by ast_fax_detect_init().

◆ FAX_TONE_CNG_DB

#define FAX_TONE_CNG_DB 16

Definition at line 211 of file dsp.c.

Referenced by ast_fax_detect_init().

◆ FAX_TONE_CNG_DURATION

#define FAX_TONE_CNG_DURATION 500 /* ms */

Definition at line 210 of file dsp.c.

Referenced by ast_fax_detect_init().

◆ FAX_TONE_CNG_FREQ

#define FAX_TONE_CNG_FREQ 1100

Definition at line 209 of file dsp.c.

Referenced by ast_fax_detect_init().

◆ FREQ_ARRAY_SIZE

#define FREQ_ARRAY_SIZE 7

Definition at line 115 of file dsp.c.

Referenced by __ast_dsp_call_progress(), and ast_dsp_prog_reset().

◆ MAX_DTMF_DIGITS

#define MAX_DTMF_DIGITS 128

Definition at line 166 of file dsp.c.

Referenced by store_digit().

◆ MF_GSIZE

#define MF_GSIZE 120

Definition at line 224 of file dsp.c.

Referenced by mf_detect().

◆ TONE_AMPLITUDE_MAX

#define TONE_AMPLITUDE_MAX 0x7fff /* Max signed linear amplitude */

Definition at line 2034 of file dsp.c.

Referenced by AST_TEST_DEFINE(), test_dtmf_amplitude_sweep(), and test_tone_amplitude_sweep().

◆ TONE_AMPLITUDE_MIN

#define TONE_AMPLITUDE_MIN 80 /* Min signed linear amplitude detectable */

Definition at line 2035 of file dsp.c.

Referenced by AST_TEST_DEFINE(), test_dtmf_amplitude_sweep(), and test_tone_amplitude_sweep().

◆ 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.

Referenced by __ast_dsp_call_progress(), and pair_there().

◆ 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.

Referenced by __ast_dsp_call_progress(), and pair_there().

◆ TONE_THRESHOLD

#define TONE_THRESHOLD 7.8e7

Definition at line 181 of file dsp.c.

Referenced by tone_detect().

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 1074 of file dsp.c.

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().

 {
    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;
 }

◆ __ast_dsp_new()

static struct ast_dsp* __ast_dsp_new ( unsigned int sample_rate )

static

Definition at line 1722 of file dsp.c.

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().

 {
    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;
 }

◆ __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 1229 of file dsp.c.

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().

 {
    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;
 }

◆ __reg_module()

static void __reg_module ( void )

static

Definition at line 2466 of file dsp.c.

◆ __unreg_module()

static void __unreg_module ( void )

static

Definition at line 2466 of file dsp.c.

◆ _dsp_init()

static int _dsp_init ( int reload )

static

Definition at line 1908 of file dsp.c.

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, THRESHOLD_SILENCE, thresholds, and ast_variable::value.

Referenced by load_module(), and reload_module().

 {
    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;
 }

◆ 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 559 of file dsp.c.

References ast_dtmf_detect_init(), ast_mf_detect_init(), digit_detect_state_t::current_digits, digit_detect_state_t::detected_digits, digit_detect_state_t::digits, digit_detect_state_t::dtmf, digit_detect_state_t::lost_digits, digit_detect_state_t::mf, and digit_detect_state_t::td.

Referenced by __ast_dsp_new(), and ast_dsp_set_digitmode().

 {
    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);
    }
 }

◆ 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 1295 of file dsp.c.

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().

 {
    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;
 }

◆ 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 1216 of file dsp.c.

References __ast_dsp_call_progress(), ast_format_cache_is_slinear(), 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.

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

◆ ast_dsp_digitreset()

void ast_dsp_digitreset ( struct ast_dsp * dsp )

Reset DTMF detector.

Definition at line 1797 of file dsp.c.

Referenced by analog_ss_thread(), mbl_new(), my_dsp_reset_and_flush_digits(), test_dtmf_amplitude_sweep(), and test_dtmf_twist_sweep().

 {
    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;
 }

◆ ast_dsp_free()

void ast_dsp_free ( struct ast_dsp * dsp )

Definition at line 1770 of file dsp.c.

References ast_free.

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

 {
    ast_free(dsp);
 }

◆ ast_dsp_get_features()

int ast_dsp_get_features ( struct ast_dsp * dsp )

Get features.

Definition at line 1764 of file dsp.c.

References ast_dsp::features.

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

 {
         return (dsp->features);
 }

◆ 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 1717 of file dsp.c.

References ast_dsp::sample_rate.

 {
    return dsp->sample_rate;
 }

◆ ast_dsp_get_tcount()

int ast_dsp_get_tcount ( struct ast_dsp * dsp )

Get tcount (Threshold counter)

Definition at line 1903 of file dsp.c.

References ast_dsp::tcount.

Referenced by dahdi_read().

 {
    return dsp->tcount;
 }

◆ 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 1996 of file dsp.c.

References thresholds.

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

 {
    return thresholds[which];
 }

◆ ast_dsp_get_tstate()

int ast_dsp_get_tstate ( struct ast_dsp * dsp )

Get tstate (Tone State)

Definition at line 1898 of file dsp.c.

References ast_dsp::tstate.

Referenced by dahdi_read().

 {
    return dsp->tstate;
 }

◆ ast_dsp_new()

struct ast_dsp* ast_dsp_new ( void )

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

Definition at line 1745 of file dsp.c.

References __ast_dsp_new(), and DEFAULT_SAMPLE_RATE.

Referenced by __ast_play_and_record(), ast_sip_session_alloc(), AST_TEST_DEFINE(), background_detect_exec(), conf_run(), dahdi_new(), detect_write(), do_waiting(), enable_dsp_detect(), fax_detect_new(), fax_session_new(), handle_recordfile(), isAnsweringMachine(), mbl_load_device(), mgcp_new(), misdn_set_opt_exec(), my_dsp_set_digitmode(), ooh323_new(), pjsip_acf_dtmf_mode_write(), read_config(), read_mf_digits(), read_sf_digits(), record_exec(), and wait_exec().

 {
    return __ast_dsp_new(DEFAULT_SAMPLE_RATE);
 }

◆ 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 1750 of file dsp.c.

References __ast_dsp_new().

Referenced by set_softmix_bridge_data(), and set_talk_detect().

 {
    return __ast_dsp_new(sample_rate);
 }

◆ 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 1488 of file dsp.c.

References ast_dsp_silence_noise_with_energy(), and NULL.

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

◆ 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 1494 of file dsp.c.

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_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, ast_frame::len, 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(), mgcp_rtp_read(), ooh323_rtp_read(), process_ast_dsp(), read_mf_digits(), read_sf_digits(), sip_rtp_read(), and wait_exec().

 {
    int silence;
    int res;
    int digit = 0, fax_digit = 0, custom_freq_digit = 0;
    int x;
    short *shortdata;
    unsigned char *odata;
    int len;
    struct ast_frame *outf = NULL;
 
    if (!af) {
       return NULL;
    }
    if (af->frametype != AST_FRAME_VOICE) {
       return af;
    }
 
    odata = af->data.ptr;
    len = af->datalen;
    /* Make sure we have short data */
    if (ast_format_cache_is_slinear(af->subclass.format)) {
       shortdata = af->data.ptr;
       len = af->datalen / 2;
    } else if (ast_format_cmp(af->subclass.format, ast_format_ulaw) == AST_FORMAT_CMP_EQUAL) {
       shortdata = ast_alloca(af->datalen * 2);
       for (x = 0; x < len; x++) {
          shortdata[x] = AST_MULAW(odata[x]);
       }
    } else if (ast_format_cmp(af->subclass.format, ast_format_alaw) == AST_FORMAT_CMP_EQUAL) {
       shortdata = ast_alloca(af->datalen * 2);
       for (x = 0; x < len; x++) {
          shortdata[x] = AST_ALAW(odata[x]);
       }
    } else {
       /*Display warning only once. Otherwise you would get hundreds of warnings every second */
       if (dsp->display_inband_dtmf_warning) {
          ast_log(LOG_WARNING, "Inband DTMF is not supported on codec %s. Use RFC2833\n", ast_format_get_name(af->subclass.format));
       }
       dsp->display_inband_dtmf_warning = 0;
       return af;
    }
 
    /* Initially we do not want to mute anything */
    dsp->mute_fragments = 0;
 
    /* Need to run the silence detection stuff for silence suppression and busy detection */
    if ((dsp->features & DSP_FEATURE_SILENCE_SUPPRESS) || (dsp->features & DSP_FEATURE_BUSY_DETECT)) {
       res = __ast_dsp_silence_noise(dsp, shortdata, len, &silence, NULL, NULL);
    }
 
    if ((dsp->features & DSP_FEATURE_SILENCE_SUPPRESS) && silence) {
       memset(&dsp->f, 0, sizeof(dsp->f));
       dsp->f.frametype = AST_FRAME_NULL;
       ast_frfree(af);
       return ast_frisolate(&dsp->f);
    }
    if ((dsp->features & DSP_FEATURE_BUSY_DETECT) && ast_dsp_busydetect(dsp)) {
       ast_channel_softhangup_internal_flag_add(chan, AST_SOFTHANGUP_DEV);
       memset(&dsp->f, 0, sizeof(dsp->f));
       dsp->f.frametype = AST_FRAME_CONTROL;
       dsp->f.subclass.integer = AST_CONTROL_BUSY;
       ast_frfree(af);
       ast_debug(1, "Requesting Hangup because the busy tone was detected on channel %s\n", ast_channel_name(chan));
       return ast_frisolate(&dsp->f);
    }
 
    if ((dsp->features & DSP_FEATURE_FAX_DETECT)) {
       if ((dsp->faxmode & DSP_FAXMODE_DETECT_CNG) && tone_detect(dsp, &dsp->cng_tone_state, shortdata, len)) {
          fax_digit = 'f';
       }
 
       if ((dsp->faxmode & DSP_FAXMODE_DETECT_CED) && tone_detect(dsp, &dsp->ced_tone_state, shortdata, len)) {
          fax_digit = 'e';
       }
    }
 
    if ((dsp->features & DSP_FEATURE_FREQ_DETECT)) {
       if ((dsp->freqmode) && tone_detect(dsp, &dsp->cng_tone_state, shortdata, len)) {
          custom_freq_digit = 'q';
       }
    }
 
    if (dsp->features & (DSP_FEATURE_DIGIT_DETECT | DSP_FEATURE_BUSY_DETECT)) {
       if (dsp->digitmode & DSP_DIGITMODE_MF) {
          digit = mf_detect(dsp, &dsp->digit_state, shortdata, len, (dsp->digitmode & DSP_DIGITMODE_NOQUELCH) == 0, (dsp->digitmode & DSP_DIGITMODE_RELAXDTMF));
       } else {
          digit = dtmf_detect(dsp, &dsp->digit_state, shortdata, len, (dsp->digitmode & DSP_DIGITMODE_NOQUELCH) == 0, (dsp->digitmode & DSP_DIGITMODE_RELAXDTMF));
       }
 
       if (dsp->digit_state.current_digits) {
          int event = 0, event_len = 0;
          char event_digit = 0;
 
          if (!dsp->dtmf_began) {
             /* We have not reported DTMF_BEGIN for anything yet */
 
             if (dsp->features & DSP_FEATURE_DIGIT_DETECT) {
                event = AST_FRAME_DTMF_BEGIN;
                event_digit = dsp->digit_state.digits[0];
             }
             dsp->dtmf_began = 1;
 
          } else if (dsp->digit_state.current_digits > 1 || digit != dsp->digit_state.digits[0]) {
             /* Digit changed. This means digit we have reported with DTMF_BEGIN ended */
             if (dsp->features & DSP_FEATURE_DIGIT_DETECT) {
                event = AST_FRAME_DTMF_END;
                event_digit = dsp->digit_state.digits[0];
                event_len = dsp->digit_state.digitlen[0] * 1000 / dsp->sample_rate;
             }
             memmove(&dsp->digit_state.digits[0], &dsp->digit_state.digits[1], dsp->digit_state.current_digits);
             memmove(&dsp->digit_state.digitlen[0], &dsp->digit_state.digitlen[1], dsp->digit_state.current_digits * sizeof(dsp->digit_state.digitlen[0]));
             dsp->digit_state.current_digits--;
             dsp->dtmf_began = 0;
 
             if (dsp->features & DSP_FEATURE_BUSY_DETECT) {
                /* Reset Busy Detector as we have some confirmed activity */
                memset(dsp->historicsilence, 0, sizeof(dsp->historicsilence));
                memset(dsp->historicnoise, 0, sizeof(dsp->historicnoise));
                ast_debug(1, "DTMF Detected - Reset busydetector\n");
             }
          }
 
          if (event) {
             memset(&dsp->f, 0, sizeof(dsp->f));
             dsp->f.frametype = event;
             dsp->f.subclass.integer = event_digit;
             dsp->f.len = event_len;
             outf = &dsp->f;
             goto done;
          }
       }
    }
 
    if (fax_digit) {
       /* Fax was detected - digit is either 'q' or 'r' */
 
       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;
    }
 }

◆ ast_dsp_prog_reset()

static void ast_dsp_prog_reset ( struct ast_dsp * dsp )

static

Definition at line 1700 of file dsp.c.

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().

 {
    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;
 }

◆ ast_dsp_reset()

void ast_dsp_reset ( struct ast_dsp * dsp )

Reset total silence count.

Definition at line 1830 of file dsp.c.

References ast_dsp::freqs, ast_dsp::gsamps, ast_dsp::historicnoise, ast_dsp::historicsilence, ast_dsp::ringtimeout, ast_dsp::totalsilence, goertzel_state_t::v2, and goertzel_state_t::v3.

Referenced by debug_check_frame_for_silence().

 {
    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;
 }

◆ 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 1780 of file dsp.c.

References ast_dsp::busycount, cadences, and DSP_HISTORY.

Referenced by dahdi_new().

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

◆ 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 1791 of file dsp.c.

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

Referenced by dahdi_new().

 {
    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);
 }

◆ 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 1879 of file dsp.c.

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

Referenced by dahdi_new().

 {
    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;
 }

◆ 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 1844 of file dsp.c.

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(), enable_dsp_detect(), mbl_load_device(), mgcp_new(), mkintf(), my_dsp_set_digitmode(), ooh323_new(), and read_mf_digits().

 {
    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;
 }

◆ ast_dsp_set_faxmode()

int ast_dsp_set_faxmode	(	struct ast_dsp *	dsp,
		int	faxmode
	)

Set fax mode.

Definition at line 1870 of file dsp.c.

References ast_fax_detect_init(), and ast_dsp::faxmode.

Referenced by fax_detect_new(), and ooh323_new().

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

◆ ast_dsp_set_features()

void ast_dsp_set_features	(	struct ast_dsp *	dsp,
		int	features
	)

Select feature set.

Definition at line 1755 of file dsp.c.

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(), enable_dsp_detect(), fax_detect_new(), mbl_load_device(), mgcp_new(), misdn_set_opt_exec(), my_handle_dtmf(), my_new_analog_ast_channel(), ooh323_new(), pjsip_acf_dtmf_mode_write(), read_config(), read_mf_digits(), read_sf_digits(), set_caps(), sip_rtp_read(), and wait_exec().

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

◆ 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 1859 of file dsp.c.

References ast_freq_detect_init(), and ast_dsp::freqmode.

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

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

◆ 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 1775 of file dsp.c.

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().

 {
    dsp->threshold = threshold;
 }

◆ 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 1483 of file dsp.c.

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().

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

◆ 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 1433 of file dsp.c.

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_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, and ast_frame::subclass.

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

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

◆ 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 1478 of file dsp.c.

References ast_dsp_silence_noise_with_energy().

Referenced by softmix_bridge_write_voice().

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

◆ 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 1893 of file dsp.c.

References ast_dsp::mute_fragments.

Referenced by dahdi_read().

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

◆ ast_dtmf_detect_init()

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

static

Definition at line 531 of file dsp.c.

References dtmf_detect_state_t::col_out, dtmf_detect_state_t::current_hit, dtmf_detect_state_t::current_sample, dtmf_col, 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().

 {
    int i;
 
    for (i = 0; i < 4; 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;
 }

◆ ast_fax_detect_init()

static void ast_fax_detect_init ( struct ast_dsp * s )

static

Definition at line 511 of file dsp.c.

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_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;
    }
 
 }

◆ 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 522 of file dsp.c.

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

Referenced by ast_dsp_set_freqmode().

 {
    /* 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;
    }
 }

◆ ast_mf_detect_init()

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

static

Definition at line 547 of file dsp.c.

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().

 {
    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;
 }

◆ AST_MODULE_SELF_SYM()

struct ast_module* AST_MODULE_SELF_SYM ( void )

Definition at line 2466 of file dsp.c.

◆ AST_TEST_DEFINE() [1/2]

AST_TEST_DEFINE ( test_dsp_fax_detect )

Definition at line 2336 of file dsp.c.

References ast_dsp_free(), ast_dsp_new(), AST_TEST_FAIL, AST_TEST_NOT_RUN, AST_TEST_PASS, ast_dsp::ced_tone_state, ast_dsp::cng_tone_state, sip_to_pjsip::info(), result, TEST_EXECUTE, TEST_INIT, test_tone_amplitude_sweep(), test_tone_freq_sweep(), TONE_AMPLITUDE_MAX, and TONE_AMPLITUDE_MIN.

 {
    struct ast_dsp *dsp;
    enum ast_test_result_state result;
 
    switch (cmd) {
    case TEST_INIT:
       info->name = "fax";
       info->category = "/main/dsp/";
       info->summary = "DSP fax tone detect unit test";
       info->description =
          "Tests fax tone detection code.";
       return AST_TEST_NOT_RUN;
    case TEST_EXECUTE:
       break;
    }
 
    dsp = ast_dsp_new();
    if (!dsp) {
       return AST_TEST_FAIL;
    }
 
    result = AST_TEST_PASS;
 
    /* Test CNG tone amplitude detection */
    if (test_tone_amplitude_sweep(test, dsp, &dsp->cng_tone_state)) {
       result = AST_TEST_FAIL;
    }
 
    /* Test CED tone amplitude detection */
    if (test_tone_amplitude_sweep(test, dsp, &dsp->ced_tone_state)) {
       result = AST_TEST_FAIL;
    }
 
    /* Test CNG tone frequency detection */
    if (test_tone_freq_sweep(test, dsp, &dsp->cng_tone_state, TONE_AMPLITUDE_MAX)) {
       result = AST_TEST_FAIL;
    }
    if (test_tone_freq_sweep(test, dsp, &dsp->cng_tone_state, TONE_AMPLITUDE_MIN)) {
       result = AST_TEST_FAIL;
    }
 
    /* Test CED tone frequency detection */
    if (test_tone_freq_sweep(test, dsp, &dsp->ced_tone_state, TONE_AMPLITUDE_MAX)) {
       result = AST_TEST_FAIL;
    }
    if (test_tone_freq_sweep(test, dsp, &dsp->ced_tone_state, TONE_AMPLITUDE_MIN)) {
       result = AST_TEST_FAIL;
    }
 
    ast_dsp_free(dsp);
    return result;
 }

◆ AST_TEST_DEFINE() [2/2]

AST_TEST_DEFINE ( test_dsp_dtmf_detect )

Definition at line 2392 of file dsp.c.

References ast_dsp_free(), ast_dsp_new(), AST_TEST_FAIL, AST_TEST_NOT_RUN, AST_TEST_PASS, dtmf_positions, sip_to_pjsip::info(), result, test_dtmf_amplitude_sweep(), test_dtmf_twist_sweep(), TEST_EXECUTE, and TEST_INIT.

 {
    int idx;
    struct ast_dsp *dsp;
    enum ast_test_result_state result;
 
    switch (cmd) {
    case TEST_INIT:
       info->name = "dtmf";
       info->category = "/main/dsp/";
       info->summary = "DSP DTMF detect unit test";
       info->description =
          "Tests DTMF detection code.";
       return AST_TEST_NOT_RUN;
    case TEST_EXECUTE:
       break;
    }
 
    dsp = ast_dsp_new();
    if (!dsp) {
       return AST_TEST_FAIL;
    }
 
    result = AST_TEST_PASS;
 
    for (idx = 0; dtmf_positions[idx]; ++idx) {
       if (test_dtmf_amplitude_sweep(test, dsp, idx)) {
          result = AST_TEST_FAIL;
       }
    }
 
    for (idx = 0; dtmf_positions[idx]; ++idx) {
       if (test_dtmf_twist_sweep(test, dsp, idx)) {
          result = AST_TEST_FAIL;
       }
    }
 
    ast_dsp_free(dsp);
    return result;
 }

◆ 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 450 of file dsp.c.

References ast_debug, tone_detect_state_t::block_size, tone_detect_state_t::energy, tone_detect_state_t::freq, goertzel_init(), tone_detect_state_t::hit_count, tone_detect_state_t::hits_required, tone_detect_state_t::last_hit, 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().

 {
    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 analisys.
       This may make block larger than expected packet and will make squelching impossible
       but at least we will be detecting the tone */
    if (periods_in_block < 5) {
       periods_in_block = 5;
    }
 
    /* Now calculate final block size. It will contain integer number of periods */
    s->block_size = periods_in_block * sample_rate / freq;
 
    /* tone_detect is 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);
 }

◆ 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 693 of file dsp.c.

Referenced by ast_dsp_process(), test_dtmf_amplitude_sweep(), and test_dtmf_twist_sweep().

 {
    float row_energy[4];
    float col_energy[4];
    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);
       }
       s->td.dtmf.current_sample += (limit - sample);
       if (s->td.dtmf.current_sample < DTMF_GSIZE) {
          continue;
       }
       /* We are at the end of a DTMF detection block */
       /* Find the peak row and the peak column */
       row_energy[0] = goertzel_result(&s->td.dtmf.row_out[0]);
       col_energy[0] = goertzel_result(&s->td.dtmf.col_out[0]);
 
       for (best_row = best_col = 0, i = 1; i < 4; i++) {
          row_energy[i] = goertzel_result(&s->td.dtmf.row_out[i]);
          if (row_energy[i] > row_energy[best_row]) {
             best_row = i;
          }
          col_energy[i] = goertzel_result(&s->td.dtmf.col_out[i]);
          if (col_energy[i] > col_energy[best_col]) {
             best_col = i;
          }
       }
       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 < 4; i++) {
             if ((i != best_col &&
                 col_energy[i] * DTMF_RELATIVE_PEAK_COL > col_energy[best_col]) ||
                 (i != best_row
                  && row_energy[i] * DTMF_RELATIVE_PEAK_ROW > row_energy[best_row])) {
                break;
             }
          }
          /* ... and fraction of total energy test */
          if (i >= 4 &&
              (row_energy[best_row] + col_energy[best_col]) > DTMF_TO_TOTAL_ENERGY * s->td.dtmf.energy) {
             /* Got a hit */
             hit = dtmf_positions[(best_row << 2) + best_col];
             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 < 4; i++) {
          goertzel_reset(&s->td.dtmf.row_out[i]);
          goertzel_reset(&s->td.dtmf.col_out[i]);
       }
       s->td.dtmf.energy = 0.0;
       s->td.dtmf.current_sample = 0;
    }
 
    if (squelch && mute.end) {
       if (mute.end > samples) {
          s->td.dtmf.mute_samples = mute.end - samples;
          mute.end = samples;
       }
       mute_fragment(dsp, &mute);
    }
 
    return (s->td.dtmf.current_hit); /* return the debounced hit */
 }

◆ goertzel_init()

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

inlinestatic

Definition at line 376 of file dsp.c.

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().

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

◆ goertzel_reset()

static void goertzel_reset ( goertzel_state_t * s )

inlinestatic

Definition at line 382 of file dsp.c.

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().

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

◆ goertzel_result()

static float goertzel_result ( goertzel_state_t * s )

inlinestatic

Definition at line 362 of file dsp.c.

References goertzel_state_t::chunky, goertzel_state_t::fac, goertzel_result_t::power, goertzel_state_t::v2, goertzel_state_t::v3, and goertzel_result_t::value.

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

 {
    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);
 }

◆ goertzel_sample()

static void goertzel_sample	(	goertzel_state_t *	s,
		short	sample
	)

inlinestatic

Definition at line 336 of file dsp.c.

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().

 {
    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;
    }
 }

◆ load_module()

static int load_module ( void )

static

Definition at line 2442 of file dsp.c.

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

Referenced by reload_module().

 {
    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;
 }

◆ 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 900 of file dsp.c.

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, 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().

 {
    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 */
 }

◆ mute_fragment()

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

static

Definition at line 440 of file dsp.c.

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().

 {
    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;
 }

◆ pair_there()

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

inlinestatic

Definition at line 1051 of file dsp.c.

References TONE_MIN_THRESH, and TONE_THRESH.

Referenced by __ast_dsp_call_progress().

 {
    /* 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;
 }

◆ reload_module()

static int reload_module ( void )

static

Definition at line 2454 of file dsp.c.

References _dsp_init(), AST_MODFLAG_GLOBAL_SYMBOLS, AST_MODFLAG_LOAD_ORDER, AST_MODPRI_CORE, AST_MODULE_INFO(), AST_MODULE_SUPPORT_CORE, ASTERISK_GPL_KEY, load_module(), reload(), and unload_module().

 {
    return _dsp_init(1);
 }

◆ store_digit()

static void store_digit	(	digit_detect_state_t *	s,
		char	digit
	)

static

Definition at line 680 of file dsp.c.

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().

 {
    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++;
    }
 }

◆ test_dtmf_amplitude_sweep()

static int test_dtmf_amplitude_sweep	(	struct ast_test *	test,
		struct ast_dsp *	dsp,
		int	digit_index
	)

static

Definition at line 2096 of file dsp.c.

References ARRAY_LEN, ast_debug, ast_dsp_digitreset(), ast_test_status_update, DEFAULT_SAMPLE_RATE, digit, ast_dsp::digit_state, dtmf_col, dtmf_detect(), DTMF_GSIZE, dtmf_positions, dtmf_row, result, test_dual_sample_gen(), TONE_AMPLITUDE_MAX, and TONE_AMPLITUDE_MIN.

Referenced by AST_TEST_DEFINE().

 {
    short slin_buf[DTMF_GSIZE];
    int result;
    int row;
    int column;
    int idx;
    struct {
       short amp_val;
       int digit;
    } amp_tests[] = {
       /*
        * XXX Since there is no current DTMF level detection issue.  This test
        * just checks the current detection levels.
        */
       { .amp_val = TONE_AMPLITUDE_MAX/2,  .digit = dtmf_positions[digit_index], },
       { .amp_val = 10000,              .digit = dtmf_positions[digit_index], },
       { .amp_val = 1000,               .digit = dtmf_positions[digit_index], },
       { .amp_val = 500,             .digit = dtmf_positions[digit_index], },
       { .amp_val = 250,             .digit = dtmf_positions[digit_index], },
       { .amp_val = 200,             .digit = dtmf_positions[digit_index], },
       { .amp_val = 180,             .digit = dtmf_positions[digit_index], },
       /* Various digits detect and not detect in this range */
       { .amp_val = 170,             .digit = 0, },
       { .amp_val = 100,             .digit = 0, },
       /*
        * Amplitudes below TONE_AMPLITUDE_MIN start having questionable detection
        * over quantization and background noise.
        */
       { .amp_val = TONE_AMPLITUDE_MIN, .digit = 0, },
       { .amp_val = 75,              .digit = 0, },
       { .amp_val = 10,              .digit = 0, },
       { .amp_val = 1,                  .digit = 0, },
    };
 
    row = (digit_index >> 2) & 0x03;
    column = digit_index & 0x03;
 
    result = 0;
 
    for (idx = 0; idx < ARRAY_LEN(amp_tests); ++idx) {
       int digit;
       int duration;
 
       ast_debug(1, "Test '%c' at amplitude %d\n",
          dtmf_positions[digit_index], amp_tests[idx].amp_val);
       test_dual_sample_gen(slin_buf, ARRAY_LEN(slin_buf), DEFAULT_SAMPLE_RATE,
          (int) dtmf_row[row], amp_tests[idx].amp_val,
          (int) dtmf_col[column], amp_tests[idx].amp_val);
 
       digit = 0;
       for (duration = 0; !digit && duration < 3; ++duration) {
          digit = dtmf_detect(dsp, &dsp->digit_state, slin_buf, ARRAY_LEN(slin_buf),
             0, 0);
       }
       if (amp_tests[idx].digit != digit) {
          /*
           * Both messages are needed.  ast_debug for when figuring out
           * what went wrong and the test update for normal output before
           * you start debugging.  The different logging methods are not
           * synchronized.
           */
          ast_debug(1,
             "Test '%c' at amplitude %d failed.  Detected Digit: '%c'\n",
             dtmf_positions[digit_index], amp_tests[idx].amp_val,
             digit ?: ' ');
          ast_test_status_update(test,
             "Test '%c' at amplitude %d failed.  Detected Digit: '%c'\n",
             dtmf_positions[digit_index], amp_tests[idx].amp_val,
             digit ?: ' ');
          result = -1;
       }
       ast_dsp_digitreset(dsp);
    }
 
    return result;
 }

◆ test_dtmf_twist_sweep()

static int test_dtmf_twist_sweep	(	struct ast_test *	test,
		struct ast_dsp *	dsp,
		int	digit_index
	)

static

Definition at line 2176 of file dsp.c.

References ARRAY_LEN, ast_debug, ast_dsp_digitreset(), ast_test_status_update, DEF_DTMF_NORMAL_TWIST, DEF_DTMF_REVERSE_TWIST, DEFAULT_SAMPLE_RATE, digit, ast_dsp::digit_state, dtmf_col, dtmf_detect(), DTMF_GSIZE, dtmf_normal_twist, dtmf_positions, dtmf_reverse_twist, dtmf_row, result, and test_dual_sample_gen().

Referenced by AST_TEST_DEFINE().

 {
    short slin_buf[DTMF_GSIZE];
    int result;
    int row;
    int column;
    int idx;
    struct {
       short amp_row;
       short amp_col;
       int digit;
    } twist_tests[] = {
       /*
        * XXX Since there is no current DTMF twist detection issue.  This test
        * just checks the current detection levels.
        *
        * Normal twist has the column higher than the row amplitude.
        * Reverse twist is the other way.
        */
       { .amp_row = 1000 + 1800, .amp_col = 1000 +    0, .digit = 0, },
       { .amp_row = 1000 + 1700, .amp_col = 1000 +    0, .digit = 0, },
       /* Various digits detect and not detect in this range */
       { .amp_row = 1000 + 1400, .amp_col = 1000 +    0, .digit = dtmf_positions[digit_index], },
       { .amp_row = 1000 + 1300, .amp_col = 1000 +    0, .digit = dtmf_positions[digit_index], },
       { .amp_row = 1000 + 1200, .amp_col = 1000 +    0, .digit = dtmf_positions[digit_index], },
       { .amp_row = 1000 + 1100, .amp_col = 1000 +    0, .digit = dtmf_positions[digit_index], },
       { .amp_row = 1000 + 1000, .amp_col = 1000 +    0, .digit = dtmf_positions[digit_index], },
       { .amp_row = 1000 +  100, .amp_col = 1000 +    0, .digit = dtmf_positions[digit_index], },
       { .amp_row = 1000 +    0, .amp_col = 1000 +  100, .digit = dtmf_positions[digit_index], },
       { .amp_row = 1000 +    0, .amp_col = 1000 +  200, .digit = dtmf_positions[digit_index], },
       { .amp_row = 1000 +    0, .amp_col = 1000 +  300, .digit = dtmf_positions[digit_index], },
       { .amp_row = 1000 +    0, .amp_col = 1000 +  400, .digit = dtmf_positions[digit_index], },
       { .amp_row = 1000 +    0, .amp_col = 1000 +  500, .digit = dtmf_positions[digit_index], },
       { .amp_row = 1000 +    0, .amp_col = 1000 +  550, .digit = dtmf_positions[digit_index], },
       /* Various digits detect and not detect in this range */
       { .amp_row = 1000 +    0, .amp_col = 1000 +  650, .digit = 0, },
       { .amp_row = 1000 +    0, .amp_col = 1000 +  700, .digit = 0, },
       { .amp_row = 1000 +    0, .amp_col = 1000 +  800, .digit = 0, },
    };
    float save_normal_twist;
    float save_reverse_twist;
 
    save_normal_twist = dtmf_normal_twist;
    save_reverse_twist = dtmf_reverse_twist;
    dtmf_normal_twist = DEF_DTMF_NORMAL_TWIST;
    dtmf_reverse_twist = DEF_DTMF_REVERSE_TWIST;
 
    row = (digit_index >> 2) & 0x03;
    column = digit_index & 0x03;
 
    result = 0;
 
    for (idx = 0; idx < ARRAY_LEN(twist_tests); ++idx) {
       int digit;
       int duration;
 
       ast_debug(1, "Test '%c' twist row %d col %d amplitudes\n",
          dtmf_positions[digit_index],
          twist_tests[idx].amp_row, twist_tests[idx].amp_col);
       test_dual_sample_gen(slin_buf, ARRAY_LEN(slin_buf), DEFAULT_SAMPLE_RATE,
          (int) dtmf_row[row], twist_tests[idx].amp_row,
          (int) dtmf_col[column], twist_tests[idx].amp_col);
 
       digit = 0;
       for (duration = 0; !digit && duration < 3; ++duration) {
          digit = dtmf_detect(dsp, &dsp->digit_state, slin_buf, ARRAY_LEN(slin_buf),
             0, 0);
       }
       if (twist_tests[idx].digit != digit) {
          /*
           * Both messages are needed.  ast_debug for when figuring out
           * what went wrong and the test update for normal output before
           * you start debugging.  The different logging methods are not
           * synchronized.
           */
          ast_debug(1,
             "Test '%c' twist row %d col %d amplitudes failed.  Detected Digit: '%c'\n",
             dtmf_positions[digit_index],
             twist_tests[idx].amp_row, twist_tests[idx].amp_col,
             digit ?: ' ');
          ast_test_status_update(test,
             "Test '%c' twist row %d col %d amplitudes failed.  Detected Digit: '%c'\n",
             dtmf_positions[digit_index],
             twist_tests[idx].amp_row, twist_tests[idx].amp_col,
             digit ?: ' ');
          result = -1;
       }
       ast_dsp_digitreset(dsp);
    }
 
    dtmf_normal_twist = save_normal_twist;
    dtmf_reverse_twist = save_reverse_twist;
 
    return result;
 }

◆ test_dual_sample_gen()

static void test_dual_sample_gen	(	short *	slin_buf,
		int	samples,
		int	rate,
		int	f1,
		short	a1,
		int	f2,
		short	a2
	)

static

Definition at line 2026 of file dsp.c.

References test_tone_sample_gen(), and test_tone_sample_gen_add().

Referenced by test_dtmf_amplitude_sweep(), and test_dtmf_twist_sweep().

 {
    test_tone_sample_gen(slin_buf, samples, rate, f1, a1);
    test_tone_sample_gen_add(slin_buf, samples, rate, f2, a2);
 }

◆ test_tone_amplitude_sweep()

static int test_tone_amplitude_sweep	(	struct ast_test *	test,
		struct ast_dsp *	dsp,
		tone_detect_state_t *	tone_state
	)

static

Definition at line 2037 of file dsp.c.

References ARRAY_LEN, ast_debug, ast_test_status_update, tone_detect_state_t::block_size, DEFAULT_SAMPLE_RATE, tone_detect_state_t::freq, tone_detect_state_t::hit_count, tone_detect_state_t::hits_required, result, test_tone_sample_gen(), TONE_AMPLITUDE_MAX, TONE_AMPLITUDE_MIN, and tone_detect().

Referenced by AST_TEST_DEFINE().

 {
    short slin_buf[tone_state->block_size];
    int result;
    int idx;
    struct {
       short amp_val;
       int detect;
    } amp_tests[] = {
       { .amp_val = TONE_AMPLITUDE_MAX, .detect = 1, },
       { .amp_val = 10000,              .detect = 1, },
       { .amp_val = 1000,               .detect = 1, },
       { .amp_val = 100,             .detect = 1, },
       { .amp_val = TONE_AMPLITUDE_MIN, .detect = 1, },
       { .amp_val = 75,              .detect = 0, },
       { .amp_val = 10,              .detect = 0, },
       { .amp_val = 1,                  .detect = 0, },
    };
 
    result = 0;
 
    for (idx = 0; idx < ARRAY_LEN(amp_tests); ++idx) {
       int detected;
       int duration;
 
       ast_debug(1, "Test %d Hz at amplitude %d\n",
          tone_state->freq, amp_tests[idx].amp_val);
       test_tone_sample_gen(slin_buf, tone_state->block_size, DEFAULT_SAMPLE_RATE,
          tone_state->freq, amp_tests[idx].amp_val);
 
       detected = 0;
       for (duration = 0; !detected && duration < tone_state->hits_required + 3; ++duration) {
          detected = tone_detect(dsp, tone_state, slin_buf, tone_state->block_size) ? 1 : 0;
       }
       if (amp_tests[idx].detect != detected) {
          /*
           * Both messages are needed.  ast_debug for when figuring out
           * what went wrong and the test update for normal output before
           * you start debugging.  The different logging methods are not
           * synchronized.
           */
          ast_debug(1,
             "Test %d Hz at amplitude %d failed.  Detected: %s\n",
             tone_state->freq, amp_tests[idx].amp_val,
             detected ? "yes" : "no");
          ast_test_status_update(test,
             "Test %d Hz at amplitude %d failed.  Detected: %s\n",
             tone_state->freq, amp_tests[idx].amp_val,
             detected ? "yes" : "no");
          result = -1;
       }
       tone_state->hit_count = 0;
    }
 
    return result;
 }

◆ test_tone_freq_sweep()

static int test_tone_freq_sweep	(	struct ast_test *	test,
		struct ast_dsp *	dsp,
		tone_detect_state_t *	tone_state,
		short	amplitude
	)

static

Definition at line 2274 of file dsp.c.

References ast_debug, ast_test_status_update, tone_detect_state_t::block_size, DEFAULT_SAMPLE_RATE, tone_detect_state_t::freq, tone_detect_state_t::hit_count, tone_detect_state_t::hits_required, result, test_tone_sample_gen(), and tone_detect().

Referenced by AST_TEST_DEFINE().

 {
    short slin_buf[tone_state->block_size];
    int result;
    int freq;
    int lower_freq;
    int upper_freq;
 
    /* Calculate detection frequency range */
    lower_freq = tone_state->freq - 4;
    upper_freq = tone_state->freq + 4;
 
    result = 0;
 
    /* Sweep frequencies loop. */
    for (freq = 100; freq <= 3500; freq += 1) {
       int detected;
       int duration;
       int expect_detection;
 
       if (freq == tone_state->freq) {
          /* This case is done by the amplitude sweep. */
          continue;
       }
 
       expect_detection = (lower_freq <= freq && freq <= upper_freq) ? 1 : 0;
 
       ast_debug(1, "Test %d Hz detection given %d Hz tone at amplitude %d.  Range:%d-%d Expect detect: %s\n",
          tone_state->freq, freq, amplitude, lower_freq, upper_freq,
          expect_detection ? "yes" : "no");
       test_tone_sample_gen(slin_buf, tone_state->block_size, DEFAULT_SAMPLE_RATE, freq,
          amplitude);
 
       detected = 0;
       for (duration = 0; !detected && duration < tone_state->hits_required + 3; ++duration) {
          detected = tone_detect(dsp, tone_state, slin_buf, tone_state->block_size) ? 1 : 0;
       }
       if (expect_detection != detected) {
          /*
           * Both messages are needed.  ast_debug for when figuring out
           * what went wrong and the test update for normal output before
           * you start debugging.  The different logging methods are not
           * synchronized.
           */
          ast_debug(1,
             "Test %d Hz detection given %d Hz tone at amplitude %d failed.  Range:%d-%d Detected: %s\n",
             tone_state->freq, freq, amplitude, lower_freq, upper_freq,
             detected ? "yes" : "no");
          ast_test_status_update(test,
             "Test %d Hz detection given %d Hz tone at amplitude %d failed.  Range:%d-%d Detected: %s\n",
             tone_state->freq, freq, amplitude, lower_freq, upper_freq,
             detected ? "yes" : "no");
          result = -1;
       }
       tone_state->hit_count = 0;
    }
 
    return result;
 }

◆ test_tone_sample_gen()

static void test_tone_sample_gen	(	short *	slin_buf,
		int	samples,
		int	rate,
		int	freq,
		short	amplitude
	)

static

Definition at line 2002 of file dsp.c.

References M_PI.

Referenced by test_dual_sample_gen(), test_tone_amplitude_sweep(), and test_tone_freq_sweep().

 {
    int idx;
    double sample_step = 2.0 * M_PI * freq / rate;/* radians per step */
 
    for (idx = 0; idx < samples; ++idx) {
       slin_buf[idx] = amplitude * sin(sample_step * idx);
    }
 }

◆ test_tone_sample_gen_add()

static void test_tone_sample_gen_add	(	short *	slin_buf,
		int	samples,
		int	rate,
		int	freq,
		short	amplitude
	)

static

Definition at line 2014 of file dsp.c.

References M_PI.

Referenced by test_dual_sample_gen().

 {
    int idx;
    double sample_step = 2.0 * M_PI * freq / rate;/* radians per step */
 
    for (idx = 0; idx < samples; ++idx) {
       slin_buf[idx] += amplitude * sin(sample_step * idx);
    }
 }

◆ tone_detect()

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

static

Definition at line 573 of file dsp.c.

Referenced by ast_dsp_process(), test_tone_amplitude_sweep(), and test_tone_freq_sweep().

 {
    float tone_energy;
    int i;
    int hit = 0;
    int limit;
    int res = 0;
    int16_t *ptr;
    short samp;
    int start, end;
    fragment_t mute = {0, 0};
 
    if (s->squelch && s->mute_samples > 0) {
       mute.end = (s->mute_samples < samples) ? s->mute_samples : samples;
       s->mute_samples -= mute.end;
    }
 
    for (start = 0; start < samples; start = end) {
       /* Process in blocks. */
       limit = samples - start;
       if (limit > s->samples_pending) {
          limit = s->samples_pending;
       }
       end = start + limit;
 
       for (i = limit, ptr = amp ; i > 0; i--, ptr++) {
          samp = *ptr;
          /* signed 32 bit int should be enough to square any possible signed 16 bit value */
          s->energy += (int32_t) samp * (int32_t) samp;
 
          goertzel_sample(&s->tone, samp);
       }
 
       s->samples_pending -= limit;
 
       if (s->samples_pending) {
          /* Finished incomplete (last) block */
          break;
       }
 
       tone_energy = goertzel_result(&s->tone);
 
       /* Scale to make comparable */
       tone_energy *= 2.0;
       s->energy *= s->block_size;
 
       ast_debug(10, "%d Hz tone %2d Ew=%.4E, Et=%.4E, s/n=%10.2f\n", s->freq, s->hit_count, tone_energy, s->energy, tone_energy / (s->energy - tone_energy));
       hit = 0;
       if (TONE_THRESHOLD <= tone_energy
          && tone_energy > s->energy * s->threshold) {
          ast_debug(10, "%d Hz tone Hit! %2d Ew=%.4E, Et=%.4E, s/n=%10.2f\n", s->freq, s->hit_count, tone_energy, s->energy, tone_energy / (s->energy - tone_energy));
          hit = 1;
       }
 
       if (s->hit_count) {
          s->hit_count++;
       }
 
       if (hit == s->last_hit) {
          if (!hit) {
             /* Two successive misses. Tone ended */
             s->hit_count = 0;
          } else if (!s->hit_count) {
             s->hit_count++;
          }
 
       }
 
       if (s->hit_count == s->hits_required) {
          ast_debug(1, "%d Hz tone detected\n", s->freq);
          res = 1;
       }
 
       s->last_hit = hit;
 
       /* If we had a hit in this block, include it into mute fragment */
       if (s->squelch && hit) {
          if (mute.end < start - s->block_size) {
             /* There is a gap between fragments */
             mute_fragment(dsp, &mute);
             mute.start = (start > s->block_size) ? (start - s->block_size) : 0;
          }
          mute.end = end + s->block_size;
       }
 
       /* Reinitialise the detector for the next block */
       /* Reset for the next block */
       goertzel_reset(&s->tone);
 
       /* Advance to the next block */
       s->energy = 0.0;
       s->samples_pending = s->block_size;
 
       amp += limit;
    }
 
    if (s->squelch && mute.end) {
       if (mute.end > samples) {
          s->mute_samples = mute.end - samples;
          mute.end = samples;
       }
       mute_fragment(dsp, &mute);
    }
 
    return res;
 }

◆ unload_module()

static int unload_module ( void )

static

Definition at line 2434 of file dsp.c.

References AST_TEST_UNREGISTER.

Referenced by reload_module().

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

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 = "This paragraph is copyright (c) 2006 by Digium, Inc. \In order for your module to load, it must return this \key via a function called \"key\". Any code which \includes this paragraph must be licensed under the GNU \General Public License version 2 or later (at your \option). In addition to Digium's general reservations \of rights, Digium expressly reserves the right to \allow other parties to license this paragraph under \different terms. Any use of Digium, Inc. trademarks or \logos (including \"Asterisk\" or \"Digium\") without \express written permission of Digium, Inc. is prohibited.\n" , .buildopt_sum = "30ef0c93b36035ec78c9cfd712d36d9b" , .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 2466 of file dsp.c.

◆ aliases

struct progalias aliases[]

static

Referenced by ast_dsp_set_call_progress_zone().

◆ ast_module_info

const struct ast_module_info* ast_module_info = &__mod_info

static

Definition at line 2466 of file dsp.c.

◆ bell_mf_positions

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

static

Definition at line 327 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 243 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 320 of file dsp.c.

Referenced by ast_dtmf_detect_init(), test_dtmf_amplitude_sweep(), and test_dtmf_twist_sweep().

◆ dtmf_hits_to_begin

int dtmf_hits_to_begin

static

Definition at line 333 of file dsp.c.

Referenced by _dsp_init(), and dtmf_detect().

◆ dtmf_misses_to_end

int dtmf_misses_to_end

static

Definition at line 334 of file dsp.c.

Referenced by _dsp_init(), and dtmf_detect().

◆ dtmf_normal_twist

float dtmf_normal_twist

static

Definition at line 329 of file dsp.c.

Referenced by _dsp_init(), dtmf_detect(), and test_dtmf_twist_sweep().

◆ dtmf_positions

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

static

Definition at line 326 of file dsp.c.

Referenced by AST_TEST_DEFINE(), dtmf_detect(), test_dtmf_amplitude_sweep(), and test_dtmf_twist_sweep().

◆ dtmf_reverse_twist

float dtmf_reverse_twist

static

Definition at line 330 of file dsp.c.

Referenced by _dsp_init(), dtmf_detect(), and test_dtmf_twist_sweep().

◆ dtmf_row

const float dtmf_row[]

static

Initial value:

= {
   697.0,  770.0,  852.0,  941.0
}

Definition at line 317 of file dsp.c.

Referenced by ast_dtmf_detect_init(), test_dtmf_amplitude_sweep(), and test_dtmf_twist_sweep().

◆ mf_tones

const float mf_tones[]

static

Initial value:

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

Definition at line 323 of file dsp.c.

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 331 of file dsp.c.

Referenced by _dsp_init(), and dtmf_detect().

◆ relax_dtmf_reverse_twist

float relax_dtmf_reverse_twist

static

Definition at line 332 of file dsp.c.

Referenced by _dsp_init(), and dtmf_detect().

◆ thresholds

int thresholds[THRESHOLD_MAX]

static

Definition at line 328 of file dsp.c.

Referenced by _dsp_init(), and ast_dsp_get_threshold_from_settings().

Data Structures

Macros

Enumerations

Functions

Variables

Detailed Description

Macro Definition Documentation

◆ BELL_MF_RELATIVE_PEAK

◆ BELL_MF_THRESHOLD

◆ BELL_MF_TWIST

◆ CONFIG_FILE_NAME

◆ 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_SAMPLE_RATE

◆ DEFAULT_THRESHOLD

◆ DSP_HISTORY

◆ DTMF_GSIZE

◆ DTMF_RELATIVE_PEAK_COL

◆ DTMF_RELATIVE_PEAK_ROW

◆ DTMF_THRESHOLD

◆ DTMF_TO_TOTAL_ENERGY

◆ FAX_TONE_CED_DB

◆ FAX_TONE_CED_DURATION

◆ FAX_TONE_CED_FREQ

◆ FAX_TONE_CNG_DB

◆ FAX_TONE_CNG_DURATION

◆ FAX_TONE_CNG_FREQ

◆ FREQ_ARRAY_SIZE

◆ MAX_DTMF_DIGITS

◆ MF_GSIZE

◆ TONE_AMPLITUDE_MAX

◆ TONE_AMPLITUDE_MIN

◆ TONE_MIN_THRESH

◆ TONE_THRESH

◆ TONE_THRESHOLD

Enumeration Type Documentation

◆ busy_detect

◆ freq_index

◆ gsamp_size

◆ gsamp_thresh

◆ prog_mode

Function Documentation

◆ __ast_dsp_call_progress()

◆ __ast_dsp_new()

◆ __ast_dsp_silence_noise()

◆ __reg_module()

◆ __unreg_module()

◆ _dsp_init()

◆ ast_digit_detect_init()

◆ ast_dsp_busydetect()

◆ ast_dsp_call_progress()

◆ ast_dsp_digitreset()

◆ ast_dsp_free()

◆ ast_dsp_get_features()

◆ ast_dsp_get_sample_rate()

◆ ast_dsp_get_tcount()

◆ ast_dsp_get_threshold_from_settings()

◆ ast_dsp_get_tstate()

◆ ast_dsp_new()

◆ ast_dsp_new_with_rate()

◆ ast_dsp_noise()

◆ ast_dsp_process()

◆ ast_dsp_prog_reset()

◆ ast_dsp_reset()

◆ ast_dsp_set_busy_count()

◆ ast_dsp_set_busy_pattern()

◆ ast_dsp_set_call_progress_zone()

◆ ast_dsp_set_digitmode()

◆ ast_dsp_set_faxmode()

◆ ast_dsp_set_features()

◆ ast_dsp_set_freqmode()

◆ ast_dsp_set_threshold()

◆ ast_dsp_silence()

◆ ast_dsp_silence_noise_with_energy()

◆ ast_dsp_silence_with_energy()

◆ ast_dsp_was_muted()