Convenience Signal Processing routines. More...

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

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
Defines
#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
	This value is the minimum threshold, calculated by averaging all of the samples within a frame, for which a frame is determined to either be silence (below the threshold) or noise (above the threshold). Please note that while the default threshold is an even exponent of 2, there is no requirement that it be so. The threshold will accept any value between 0 and 32767.
#define	DSP_HISTORY 15
#define	DTMF_2ND_HARMONIC_COL 63.1 /* 18dB */
#define	DTMF_2ND_HARMONIC_ROW (relax ? 1.7 : 2.5) /* 4dB normal */
#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_2ND_HARMONIC 2.0 /* 4dB */
#define	FAX_THRESHOLD 8.0e7
#define	FAX_TONE_CED_DB 16
#define	FAX_TONE_CED_DURATION 2600
#define	FAX_TONE_CED_FREQ 2100
#define	FAX_TONE_CNG_DB 16
#define	FAX_TONE_CNG_DURATION 500
#define	FAX_TONE_CNG_FREQ 1100
#define	MAX_DTMF_DIGITS 128
#define	MF_GSIZE 120
#define	TONE_MIN_THRESH 1e8
#define	TONE_THRESH 10.0
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 int	_dsp_init (int reload)
static void	ast_digit_detect_init (digit_detect_state_t *s, int mf, unsigned int sample_rate)
int	ast_dsp_busydetect (struct ast_dsp *dsp)
	Return non-zero if historically this should be a busy, request that ast_dsp_silence has already been called.
int	ast_dsp_call_progress (struct ast_dsp dsp, struct ast_frame inf)
	Scans for progress indication in audio.
void	ast_dsp_digitreset (struct ast_dsp *dsp)
	Reset DTMF detector.
void	ast_dsp_free (struct ast_dsp *dsp)
unsigned int	ast_dsp_get_sample_rate (const struct ast_dsp *dsp)
	Retrieve the sample rate this DSP structure was created with.
int	ast_dsp_get_tcount (struct ast_dsp *dsp)
	Get tcount (Threshold counter)
int	ast_dsp_get_threshold_from_settings (enum threshold which)
	Get silence threshold from dsp.conf.
int	ast_dsp_get_tstate (struct ast_dsp *dsp)
	Get tstate (Tone State)
int	ast_dsp_init (void)
	Load dsp settings from dsp.conf.
struct ast_dsp *	ast_dsp_new (void)
	Allocates a new dsp, assumes 8khz for internal sample rate.
struct ast_dsp *	ast_dsp_new_with_rate (unsigned int sample_rate)
	Allocates a new dsp with a specific internal sample rate used during processing.
int	ast_dsp_noise (struct ast_dsp dsp, struct ast_frame f, int *totalnoise)
	Return non-zero if this is noise. Updates "totalnoise" with the total number of seconds of noise.
struct ast_frame *	ast_dsp_process (struct ast_channel chan, struct ast_dsp dsp, struct ast_frame *af)
	Return AST_FRAME_NULL frames when there is silence, AST_FRAME_BUSY on busies, and call progress, all dependent upon which features are enabled.
static void	ast_dsp_prog_reset (struct ast_dsp *dsp)
int	ast_dsp_reload (void)
	Reloads dsp settings from dsp.conf.
void	ast_dsp_reset (struct ast_dsp *dsp)
	Reset total silence count.
void	ast_dsp_set_busy_count (struct ast_dsp *dsp, int cadences)
	Set number of required cadences for busy.
void	ast_dsp_set_busy_pattern (struct ast_dsp dsp, const struct ast_dsp_busy_pattern cadence)
	Set expected lengths of the busy tone.
int	ast_dsp_set_call_progress_zone (struct ast_dsp dsp, char zone)
	Set zone for doing progress detection.
int	ast_dsp_set_digitmode (struct ast_dsp *dsp, int digitmode)
	Set digit mode.
int	ast_dsp_set_faxmode (struct ast_dsp *dsp, int faxmode)
	Set fax mode.
void	ast_dsp_set_features (struct ast_dsp *dsp, int features)
	Select feature set.
void	ast_dsp_set_threshold (struct ast_dsp *dsp, int threshold)
	Set threshold value for silence.
int	ast_dsp_silence (struct ast_dsp dsp, struct ast_frame f, int *totalsilence)
	Return non-zero if this is silence. Updates "totalsilence" with the total number of seconds of silence.
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)
	Return non-zero if this is silence. Updates "totalsilence" with the total number of seconds of silence. Returns the average energy of the samples in the frame in frames_energy variable.
int	ast_dsp_was_muted (struct ast_dsp *dsp)
	Returns true if DSP code was muting any fragment of the last processed frame. Muting (squelching) happens when DSP code removes DTMF/MF/generic tones from the audio.
static void	ast_dtmf_detect_init (dtmf_detect_state_t *s, unsigned int sample_rate)
static void	ast_fax_detect_init (struct ast_dsp *s)
static void	ast_mf_detect_init (mf_detect_state_t *s, unsigned int sample_rate)
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, int samples, 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 void	goertzel_update (goertzel_state_t s, short samps, int count)
static int	mf_detect (struct ast_dsp dsp, digit_detect_state_t s, int16_t amp[], int samples, int squelch, int relax)
static void	mute_fragment (struct ast_dsp dsp, fragment_t fragment)
static int	pair_there (float p1, float p2, float i1, float i2, float e)
static void	store_digit (digit_detect_state_t *s, char digit)
static int	tone_detect (struct ast_dsp dsp, tone_detect_state_t s, int16_t *amp, int samples)
Variables
static struct progalias	aliases []
static const char	bell_mf_positions [] = "1247C-358A--69*---0B----#"
static const int	DEFAULT_SILENCE_THRESHOLD = 256
	The default silence threshold we will use if an alternate configured value is not present or is invalid.
static const float	dtmf_col []
static int	dtmf_hits_to_begin
static int	dtmf_misses_to_end
static float	dtmf_normal_twist
static const char	dtmf_positions [] = "*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 <markster@digium.com>; Steve Underwood <steveu@coppice.org>

Definition in file dsp.c.

Define Documentation

#define BELL_MF_RELATIVE_PEAK 12.6 /* 11dB */

Definition at line 184 of file dsp.c.

Referenced by mf_detect().

#define BELL_MF_THRESHOLD 1.6e9

Definition at line 182 of file dsp.c.

Referenced by mf_detect().

#define BELL_MF_TWIST 4.0 /* 6dB */

Definition at line 183 of file dsp.c.

Referenced by mf_detect().

#define CONFIG_FILE_NAME "dsp.conf"

Definition at line 229 of file dsp.c.

Referenced by _dsp_init().

#define DEF_DTMF_HITS_TO_BEGIN 2

Definition at line 216 of file dsp.c.

Referenced by _dsp_init().

#define DEF_DTMF_MISSES_TO_END 3

Definition at line 221 of file dsp.c.

Referenced by _dsp_init().

#define DEF_DTMF_NORMAL_TWIST 6.31 /* 8.0dB */

Definition at line 165 of file dsp.c.

Referenced by _dsp_init().

#define DEF_DTMF_REVERSE_TWIST 2.51 /* 4.01dB */

Definition at line 172 of file dsp.c.

Referenced by _dsp_init().

#define DEF_RELAX_DTMF_NORMAL_TWIST 6.31 /* 8.0dB */

Definition at line 166 of file dsp.c.

Referenced by _dsp_init().

#define DEF_RELAX_DTMF_REVERSE_TWIST 3.98 /* 6.0dB */

Definition at line 173 of file dsp.c.

Referenced by _dsp_init().

#define DEFAULT_SAMPLE_RATE 8000

Definition at line 205 of file dsp.c.

Referenced by ast_dsp_new().

#define DEFAULT_THRESHOLD 512

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

Definition at line 121 of file dsp.c.

Referenced by __ast_dsp_new().

#define DSP_HISTORY 15

Remember last 15 units

Definition at line 132 of file dsp.c.

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

#define DTMF_2ND_HARMONIC_COL 63.1 /* 18dB */

Definition at line 179 of file dsp.c.

#define DTMF_2ND_HARMONIC_ROW (relax ? 1.7 : 2.5) /* 4dB normal */

Definition at line 178 of file dsp.c.

#define DTMF_GSIZE 102

Definition at line 211 of file dsp.c.

Referenced by ast_dtmf_detect_init(), and dtmf_detect().

#define DTMF_RELATIVE_PEAK_COL 6.3 /* 8dB */

Definition at line 177 of file dsp.c.

Referenced by dtmf_detect().

#define DTMF_RELATIVE_PEAK_ROW 6.3 /* 8dB */

Definition at line 176 of file dsp.c.

Referenced by dtmf_detect().

#define DTMF_THRESHOLD 8.0e7

Definition at line 161 of file dsp.c.

Referenced by dtmf_detect().

#define DTMF_TO_TOTAL_ENERGY 42.0

Definition at line 180 of file dsp.c.

Referenced by dtmf_detect().

#define FAX_2ND_HARMONIC 2.0 /* 4dB */

Definition at line 163 of file dsp.c.

#define FAX_THRESHOLD 8.0e7

Definition at line 162 of file dsp.c.

#define FAX_TONE_CED_DB 16

Definition at line 203 of file dsp.c.

Referenced by ast_fax_detect_init().

#define FAX_TONE_CED_DURATION 2600

Definition at line 202 of file dsp.c.

Referenced by ast_fax_detect_init().

#define FAX_TONE_CED_FREQ 2100

Definition at line 201 of file dsp.c.

Referenced by ast_fax_detect_init().

#define FAX_TONE_CNG_DB 16

Definition at line 195 of file dsp.c.

Referenced by ast_fax_detect_init().

#define FAX_TONE_CNG_DURATION 500

Definition at line 194 of file dsp.c.

Referenced by ast_fax_detect_init().

#define FAX_TONE_CNG_FREQ 1100

Definition at line 193 of file dsp.c.

Referenced by ast_fax_detect_init().

#define MAX_DTMF_DIGITS 128

Definition at line 147 of file dsp.c.

Referenced by store_digit().

#define MF_GSIZE 120

Definition at line 208 of file dsp.c.

Referenced by ast_mf_detect_init(), and mf_detect().

#define TONE_MIN_THRESH 1e8

How much tone there should be at least to attempt

Definition at line 135 of file dsp.c.

Referenced by __ast_dsp_call_progress(), and pair_there().

#define TONE_THRESH 10.0

How much louder the tone should be than channel energy

Definition at line 134 of file dsp.c.

Referenced by __ast_dsp_call_progress(), and pair_there().

Enumeration Type Documentation

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 123 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 */
};

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 75 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
};

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 63 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 */
};

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 138 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) */
};

enum prog_mode

Enumerator:

PROG_MODE_NA
PROG_MODE_CR
PROG_MODE_UK

Definition at line 69 of file dsp.c.

               {
   PROG_MODE_NA = 0,
   PROG_MODE_CR,
   PROG_MODE_UK
};

Function Documentation

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

Definition at line 1032 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, 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().

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

         /* Reset goertzel */
         for (x = 0; x < 7; x++) {
            dsp->freqs[x].v2 = dsp->freqs[x].v3 = 0.0;
         }
         dsp->gsamps = 0;
         dsp->genergy = 0.0;
      }
   }

   return res;
}

static struct ast_dsp* __ast_dsp_new ( unsigned int sample_rate ) [static, read]

Definition at line 1670 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::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;
      /* 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;
}

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 1183 of file dsp.c.

References 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;
}

static int _dsp_init ( int reload ) [static]

Definition at line 1835 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, LOG_WARNING, ast_variable::name, ast_variable::next, THRESHOLD_SILENCE, and ast_variable::value.

Referenced by ast_dsp_init(), and ast_dsp_reload().

{
   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;
}

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

Definition at line 523 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);
   }
}

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 1249 of file dsp.c.

References 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;
}

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

Scans for progress indication in audio.

Definition at line 1170 of file dsp.c.

References __ast_dsp_call_progress(), ast_format_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_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);
}

void ast_dsp_digitreset ( struct ast_dsp * dsp )

Reset DTMF detector.

Definition at line 1737 of file dsp.c.

Referenced by analog_ss_thread(), and my_dsp_reset_and_flush_digits().

{
   int i;

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

   dsp->digit_state.digits[0] = '\0';
   dsp->digit_state.current_digits = 0;
}

void ast_dsp_free ( struct ast_dsp * dsp )

Definition at line 1710 of file dsp.c.

References ast_free.

Referenced by __ast_play_and_record(), __oh323_destroy(), analog_ss_thread(), background_detect_exec(), chan_list_destructor(), cleanup_connection(), conf_run(), dahdi_hangup(), destroy_endpoint(), destroy_faxdetect(), destroy_session(), disable_dsp_detect(), do_waiting(), handle_recordfile(), isAnsweringMachine(), mgcp_hangup(), my_all_subchannels_hungup(), my_dsp_set_digitmode(), record_exec(), set_softmix_bridge_data(), sip_rtp_read(), and softmix_bridge_leave().

{
   ast_free(dsp);
}

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 1665 of file dsp.c.

References ast_dsp::sample_rate.

{
   return dsp->sample_rate;
}

int ast_dsp_get_tcount ( struct ast_dsp * dsp )

Get tcount (Threshold counter)

Definition at line 1830 of file dsp.c.

References ast_dsp::tcount.

Referenced by dahdi_read().

{
   return dsp->tcount;
}

int ast_dsp_get_threshold_from_settings ( enum threshold which )

Get silence threshold from dsp.conf.

Since:: 1.6.1

Definition at line 1923 of file dsp.c.

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

{
   return thresholds[which];
}

int ast_dsp_get_tstate ( struct ast_dsp * dsp )

Get tstate (Tone State)

Definition at line 1825 of file dsp.c.

References ast_dsp::tstate.

Referenced by dahdi_read().

{
   return dsp->tstate;
}

int ast_dsp_init ( void )

Load dsp settings from dsp.conf.

Since:: 1.6.1

Definition at line 1928 of file dsp.c.

References _dsp_init().

Referenced by main().

{
   return _dsp_init(0);
}

struct ast_dsp* ast_dsp_new ( void ) [read]

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

Definition at line 1692 of file dsp.c.

References __ast_dsp_new(), and DEFAULT_SAMPLE_RATE.

Referenced by __ast_play_and_record(), __oh323_new(), background_detect_exec(), conf_run(), dahdi_new(), do_waiting(), enable_dsp_detect(), fax_detect_new(), fax_session_new(), handle_recordfile(), isAnsweringMachine(), mgcp_new(), misdn_set_opt_exec(), my_dsp_set_digitmode(), read_config(), and record_exec().

{
   return __ast_dsp_new(DEFAULT_SAMPLE_RATE);
}

struct ast_dsp* ast_dsp_new_with_rate ( unsigned int sample_rate ) [read]

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

Definition at line 1697 of file dsp.c.

References __ast_dsp_new().

Referenced by set_softmix_bridge_data().

{
   return __ast_dsp_new(sample_rate);
}

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

Return non-zero if this is noise. Updates "totalnoise" with the total number of seconds of noise.

Since:: 1.6.1

Definition at line 1442 of file dsp.c.

References ast_dsp_silence_noise_with_energy().

Referenced by do_waiting().

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

struct ast_frame* ast_dsp_process	(	struct ast_channel *	chan,
		struct ast_dsp *	dsp,
		struct ast_frame *	af
	)		`[read]`

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 1448 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_is_slinear(), AST_FORMAT_TESTLAW, 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_getformatname(), 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, ast_dsp::digit_state, digit_detect_state_t::digitlen, ast_dsp::digitmode, digit_detect_state_t::digits, ast_dsp::display_inband_dtmf_warning, 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_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::historicnoise, ast_dsp::historicsilence, ast_format::id, ast_frame_subclass::integer, ast_frame::len, len(), LOG_WARNING, mf_detect(), ast_dsp::mute_data, ast_dsp::mute_fragments, ast_frame::ptr, ast_dsp::sample_rate, ast_frame::src, fragment_t::start, ast_frame::subclass, and tone_detect().

Referenced by dahdi_read(), fax_detect_framehook(), mgcp_rtp_read(), oh323_rtp_read(), process_ast_dsp(), and sip_rtp_read().

{
   int silence;
   int res;
   int digit = 0, fax_digit = 0;
   int x;
   short *shortdata;
   unsigned char *odata;
   int len;
   struct ast_frame *outf = NULL;

   if (!af) {
      return NULL;
   }
   if (af->frametype != AST_FRAME_VOICE) {
      return af;
   }

   odata = af->data.ptr;
   len = af->datalen;
   /* Make sure we have short data */
   if (ast_format_is_slinear(&af->subclass.format)) {
      shortdata = af->data.ptr;
      len = af->datalen / 2;
   } else {
      switch (af->subclass.format.id) {
      case AST_FORMAT_ULAW:
      case AST_FORMAT_TESTLAW:
         shortdata = ast_alloca(af->datalen * 2);
         for (x = 0;x < len; x++) {
            shortdata[x] = AST_MULAW(odata[x]);
         }
         break;
      case AST_FORMAT_ALAW:
         shortdata = ast_alloca(af->datalen * 2);
         for (x = 0; x < len; x++) {
            shortdata[x] = AST_ALAW(odata[x]);
         }
         break;
      default:
         /*Display warning only once. Otherwise you would get hundreds of warnings every second */
         if (dsp->display_inband_dtmf_warning)
            ast_log(LOG_WARNING, "Inband DTMF is not supported on codec %s. Use RFC2833\n", ast_getformatname(&af->subclass.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_DIGIT_DETECT | DSP_FEATURE_BUSY_DETECT)) {
      if (dsp->digitmode & DSP_DIGITMODE_MF) {
         digit = mf_detect(dsp, &dsp->digit_state, shortdata, len, (dsp->digitmode & DSP_DIGITMODE_NOQUELCH) == 0, (dsp->digitmode & DSP_DIGITMODE_RELAXDTMF));
      } else {
         digit = dtmf_detect(dsp, &dsp->digit_state, shortdata, len, (dsp->digitmode & DSP_DIGITMODE_NOQUELCH) == 0, (dsp->digitmode & DSP_DIGITMODE_RELAXDTMF));
      }

      if (dsp->digit_state.current_digits) {
         int event = 0, event_len = 0;
         char event_digit = 0;

         if (!dsp->dtmf_began) {
            /* We have not reported DTMF_BEGIN for anything yet */

            if (dsp->features & DSP_FEATURE_DIGIT_DETECT) {
               event = AST_FRAME_DTMF_BEGIN;
               event_digit = dsp->digit_state.digits[0];
            }
            dsp->dtmf_began = 1;

         } else if (dsp->digit_state.current_digits > 1 || digit != dsp->digit_state.digits[0]) {
            /* Digit changed. This means digit we have reported with DTMF_BEGIN ended */
            if (dsp->features & DSP_FEATURE_DIGIT_DETECT) {
               event = AST_FRAME_DTMF_END;
               event_digit = dsp->digit_state.digits[0];
               event_len = dsp->digit_state.digitlen[0] * 1000 / dsp->sample_rate;
            }
            memmove(&dsp->digit_state.digits[0], &dsp->digit_state.digits[1], dsp->digit_state.current_digits);
            memmove(&dsp->digit_state.digitlen[0], &dsp->digit_state.digitlen[1], dsp->digit_state.current_digits * sizeof(dsp->digit_state.digitlen[0]));
            dsp->digit_state.current_digits--;
            dsp->dtmf_began = 0;

            if (dsp->features & DSP_FEATURE_BUSY_DETECT) {
               /* Reset Busy Detector as we have some confirmed activity */
               memset(dsp->historicsilence, 0, sizeof(dsp->historicsilence));
               memset(dsp->historicnoise, 0, sizeof(dsp->historicnoise));
               ast_debug(1, "DTMF Detected - Reset busydetector\n");
            }
         }

         if (event) {
            memset(&dsp->f, 0, sizeof(dsp->f));
            dsp->f.frametype = event;
            dsp->f.subclass.integer = event_digit;
            dsp->f.len = event_len;
            outf = &dsp->f;
            goto done;
         }
      }
   }

   if (fax_digit) {
      /* Fax was detected - digit is either 'f' or 'e' */

      memset(&dsp->f, 0, sizeof(dsp->f));
      dsp->f.frametype = AST_FRAME_DTMF;
      dsp->f.subclass.integer = fax_digit;
      outf = &dsp->f;
      goto done;
   }

   if ((dsp->features & DSP_FEATURE_CALL_PROGRESS)) {
      res = __ast_dsp_call_progress(dsp, shortdata, len);
      if (res) {
         switch (res) {
         case AST_CONTROL_ANSWER:
         case AST_CONTROL_BUSY:
         case AST_CONTROL_RINGING:
         case AST_CONTROL_CONGESTION:
         case AST_CONTROL_HANGUP:
            memset(&dsp->f, 0, sizeof(dsp->f));
            dsp->f.frametype = AST_FRAME_CONTROL;
            dsp->f.subclass.integer = res;
            dsp->f.src = "dsp_progress";
            if (chan) {
               ast_queue_frame(chan, &dsp->f);
            }
            break;
         default:
            ast_log(LOG_WARNING, "Don't know how to represent call progress message %d\n", res);
         }
      }
   } else if ((dsp->features & DSP_FEATURE_WAITDIALTONE)) {
      res = __ast_dsp_call_progress(dsp, shortdata, len);
   }

done:
   /* Mute fragment of the frame */
   for (x = 0; x < dsp->mute_fragments; x++) {
      memset(shortdata + dsp->mute_data[x].start, 0, sizeof(int16_t) * (dsp->mute_data[x].end - dsp->mute_data[x].start));
   }

   switch (af->subclass.format.id) {
   case AST_FORMAT_ULAW:
      for (x = 0; x < len; x++) {
         odata[x] = AST_LIN2MU((unsigned short) shortdata[x]);
      }
      break;
   case AST_FORMAT_ALAW:
      for (x = 0; x < len; x++) {
         odata[x] = AST_LIN2A((unsigned short) shortdata[x]);
      }
      /* fall through */
   default:
      break;
   }

   if (outf) {
      if (chan) {
         ast_queue_frame(chan, af);
      }
      ast_frfree(af);
      return ast_frisolate(outf);
   } else {
      return af;
   }
}

static void ast_dsp_prog_reset ( struct ast_dsp * dsp ) [static]

Definition at line 1648 of file dsp.c.

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

int ast_dsp_reload ( void )

Reloads dsp settings from dsp.conf.

Since:: 1.6.1

Definition at line 1933 of file dsp.c.

References _dsp_init().

{
   return _dsp_init(1);
}

void ast_dsp_reset ( struct ast_dsp * dsp )

Reset total silence count.

Definition at line 1768 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;
}

void ast_dsp_set_busy_count	(	struct ast_dsp *	dsp,
		int	cadences
	)

Set number of required cadences for busy.

Definition at line 1720 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;
}

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 1731 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);
}

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

Set zone for doing progress detection.

Definition at line 1806 of file dsp.c.

References aliases, ARRAY_LEN, ast_dsp_prog_reset(), progalias::mode, 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;
}

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 1782 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(), mgcp_new(), mkintf(), and my_dsp_set_digitmode().

{
   int new;
   int old;

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

int ast_dsp_set_faxmode	(	struct ast_dsp *	dsp,
		int	faxmode
	)

Set fax mode.

Definition at line 1797 of file dsp.c.

References ast_fax_detect_init(), and ast_dsp::faxmode.

Referenced by fax_detect_new().

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

void ast_dsp_set_features	(	struct ast_dsp *	dsp,
		int	features
	)

Select feature set.

Definition at line 1702 of file dsp.c.

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

Referenced by __oh323_new(), dahdi_handle_dtmf(), dahdi_new(), dahdi_read(), dahdi_setoption(), disable_dtmf_detect(), enable_dsp_detect(), enable_dtmf_detect(), fax_detect_new(), mgcp_new(), misdn_set_opt_exec(), my_handle_dtmf(), my_pri_ss7_open_media(), read_config(), and sip_rtp_read().

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

void ast_dsp_set_threshold	(	struct ast_dsp *	dsp,
		int	threshold
	)

Set threshold value for silence.

Definition at line 1715 of file dsp.c.

References ast_dsp::threshold.

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

{
   dsp->threshold = threshold;
}

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

Return non-zero if this is silence. Updates "totalsilence" with the total number of seconds of silence.

Definition at line 1437 of file dsp.c.

References ast_dsp_silence_noise_with_energy().

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

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

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 1386 of file dsp.c.

References __ast_dsp_silence_noise(), AST_ALAW, ast_alloca, AST_FORMAT_ALAW, ast_format_is_slinear(), AST_FORMAT_ULAW, AST_FRAME_VOICE, ast_log(), AST_MULAW, ast_frame::data, ast_frame::datalen, ast_frame_subclass::format, ast_frame::frametype, ast_format::id, len(), LOG_WARNING, 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_is_slinear(&f->subclass.format)) {
      odata = f->data.ptr;
      len = f->datalen;
      switch (f->subclass.format.id) {
         case AST_FORMAT_ULAW:
            s = ast_alloca(len * 2);
            for (x = 0;x < len; x++) {
               s[x] = AST_MULAW(odata[x]);
            }
            break;
         case AST_FORMAT_ALAW:
            s = ast_alloca(len * 2);
            for (x = 0;x < len; x++) {
               s[x] = AST_ALAW(odata[x]);
            }
            break;
         default:
            ast_log(LOG_WARNING, "Can only calculate silence on signed-linear, alaw or ulaw frames :(\n");
         return 0;
      }
   } else {
      s = f->data.ptr;
      len = f->datalen/2;
   }
   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);
   }
}

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

Return non-zero if this is silence. Updates "totalsilence" with the total number of seconds of silence. Returns the average energy of the samples in the frame in frames_energy variable.

Definition at line 1432 of file dsp.c.

References ast_dsp_silence_noise_with_energy().

Referenced by softmix_bridge_write().

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

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 1820 of file dsp.c.

References ast_dsp::mute_fragments.

Referenced by dahdi_read().

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

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

Definition at line 496 of file dsp.c.

References dtmf_detect_state_t::col_out, dtmf_detect_state_t::current_hit, dtmf_detect_state_t::current_sample, DTMF_GSIZE, dtmf_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;

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

static void ast_fax_detect_init ( struct ast_dsp * s ) [static]

Definition at line 485 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;
   }

}

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

Definition at line 512 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_GSIZE, and mf_detect_state_t::tone_out.

Referenced by ast_digit_detect_init().

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

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 424 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().

{
   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 currently 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, s->block_size, 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);
}

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 656 of file dsp.c.

Referenced by ast_dsp_process().

{
   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;
         }
      }
      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];
         }
      }

/*
 * 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 */
}

static void goertzel_init	(	goertzel_state_t *	s,
		float	freq,
		int	samples,
		unsigned int	sample_rate
	)		`[inline, static]`

Definition at line 352 of file dsp.c.

References goertzel_state_t::chunky, cos, goertzel_state_t::fac, M_PI, goertzel_state_t::samples, goertzel_state_t::v2, and goertzel_state_t::v3.

Referenced by ast_dsp_prog_reset(), ast_dtmf_detect_init(), ast_mf_detect_init(), and ast_tone_detect_init().

{
   s->v2 = s->v3 = s->chunky = 0.0;
   s->fac = (int)(32768.0 * 2.0 * cos(2.0 * M_PI * freq / sample_rate));
   s->samples = samples;
}

static void goertzel_reset ( goertzel_state_t * s ) [inline, static]

Definition at line 359 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 = 0.0;
}

static float goertzel_result ( goertzel_state_t * s ) [inline, static]

Definition at line 343 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;
   r.power = s->chunky * 2;
   return (float)r.value * (float)(1 << r.power);
}

static void goertzel_sample	(	goertzel_state_t *	s,
		short	sample
	)		`[inline, static]`

Definition at line 317 of file dsp.c.

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

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

{
   int v1;

   v1 = s->v2;
   s->v2 = s->v3;

   s->v3 = (s->fac * s->v2) >> 15;
   s->v3 = s->v3 - v1 + (sample >> s->chunky);
   if (abs(s->v3) > 32768) {
      s->chunky++;
      s->v3 = s->v3 >> 1;
      s->v2 = s->v2 >> 1;
   }
}

static void goertzel_update	(	goertzel_state_t *	s,
		short *	samps,
		int	count
	)		`[inline, static]`

Definition at line 333 of file dsp.c.

References goertzel_sample().

{
   int i;

   for (i = 0; i < count; i++) {
      goertzel_sample(s, samps[i]);
   }
}

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 858 of file dsp.c.

References 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 */
}

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

Definition at line 414 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;
}

static int pair_there	(	float	p1,
		float	p2,
		float	i1,
		float	i2,
		float	e
	)		`[inline, static]`

Definition at line 1009 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;
}

static void store_digit	(	digit_detect_state_t *	s,
		char	digit
	)		`[static]`

Definition at line 643 of file dsp.c.

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

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++;
   }
}

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

Definition at line 537 of file dsp.c.

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

Referenced by ast_dsp_process().

{
   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, "tone %d, Ew=%.2E, Et=%.2E, s/n=%10.2f\n", s->freq, tone_energy, s->energy, tone_energy / (s->energy - tone_energy));
      hit = 0;
      if (tone_energy > s->energy * s->threshold) {
         ast_debug(10, "Hit! count=%d\n", s->hit_count);
         hit = 1;
      }

      if (s->hit_count) {
         s->hit_count++;
      }

      if (hit == s->last_hit) {
         if (!hit) {
            /* Two successive misses. Tone ended */
            s->hit_count = 0;
         } else if (!s->hit_count) {
            s->hit_count++;
         }

      }

      if (s->hit_count == s->hits_required) {
         ast_debug(1, "%d Hz done detected\n", s->freq);
         res = 1;
      }

      s->last_hit = hit;

      /* If we had a hit in this block, include it into mute fragment */
      if (s->squelch && hit) {
         if (mute.end < start - s->block_size) {
            /* There is a gap between fragments */
            mute_fragment(dsp, &mute);
            mute.start = (start > s->block_size) ? (start - s->block_size) : 0;
         }
         mute.end = end + s->block_size;
      }

      /* Reinitialise the detector for the next block */
      /* Reset for the next block */
      goertzel_reset(&s->tone);

      /* Advance to the next block */
      s->energy = 0.0;
      s->samples_pending = s->block_size;

      amp += limit;
   }

   if (s->squelch && mute.end) {
      if (mute.end > samples) {
         s->mute_samples = mute.end - samples;
         mute.end = samples;
      }
      mute_fragment(dsp, &mute);
   }

   return res;
}

Variable Documentation

struct progalias aliases[] [static]

Referenced by ast_dsp_set_call_progress_zone().

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

Definition at line 308 of file dsp.c.

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 227 of file dsp.c.

Referenced by _dsp_init(), and confbridge_exec().

const float dtmf_col[] [static]

Initial value:

 {
   1209.0, 1336.0, 1477.0, 1633.0
}

Definition at line 301 of file dsp.c.

int dtmf_hits_to_begin [static]

Definition at line 314 of file dsp.c.

int dtmf_misses_to_end [static]

Definition at line 315 of file dsp.c.

float dtmf_normal_twist [static]

Definition at line 310 of file dsp.c.

const char dtmf_positions[] = "*0#D" [static]

Definition at line 307 of file dsp.c.

float dtmf_reverse_twist [static]

Definition at line 311 of file dsp.c.

const float dtmf_row[] [static]

Initial value:

 {
   697.0,  770.0,  852.0,  941.0
}

Definition at line 298 of file dsp.c.

const float mf_tones[] [static]

Initial value:

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

Definition at line 304 of file dsp.c.

struct progress modes[] [static]

Referenced by ast_dsp_prog_reset().

float relax_dtmf_normal_twist [static]

Definition at line 312 of file dsp.c.

float relax_dtmf_reverse_twist [static]

Definition at line 313 of file dsp.c.

int thresholds[THRESHOLD_MAX] [static]

Definition at line 309 of file dsp.c.

Asterisk developer's documentation

Data Structures

Defines

Enumerations

Functions

Variables

Detailed Description

Define Documentation

Enumeration Type Documentation

Function Documentation

Variable Documentation