dsp.h File Reference

Convenient Signal Processing routines. More...

This graph shows which files directly or indirectly include this file:

Go to the source code of this file.

Data Structures
struct	ast_dsp_busy_pattern

Macros
#define	DSP_DIGITMODE_DTMF   0
#define	DSP_DIGITMODE_MF   1
#define	DSP_DIGITMODE_MUTECONF   (1 << 9)
#define	DSP_DIGITMODE_MUTEMAX   (1 << 10)
#define	DSP_DIGITMODE_NOQUELCH   (1 << 8)
#define	DSP_DIGITMODE_RELAXDTMF   (1 << 11)
#define	DSP_FAXMODE_DETECT_ALL   (DSP_FAXMODE_DETECT_CNG | DSP_FAXMODE_DETECT_CED)
#define	DSP_FAXMODE_DETECT_CED   (1 << 1)
#define	DSP_FAXMODE_DETECT_CNG   (1 << 0)
#define	DSP_FAXMODE_DETECT_SQUELCH   (1 << 2)
#define	DSP_FEATURE_BUSY_DETECT   (1 << 1)
#define	DSP_FEATURE_CALL_PROGRESS   (DSP_PROGRESS_TALK | DSP_PROGRESS_RINGING | DSP_PROGRESS_BUSY | DSP_PROGRESS_CONGESTION)
#define	DSP_FEATURE_DIGIT_DETECT   (1 << 3)
#define	DSP_FEATURE_FAX_DETECT   (1 << 4)
#define	DSP_FEATURE_FREQ_DETECT   (1 << 21)
#define	DSP_FEATURE_SILENCE_SUPPRESS   (1 << 0)
#define	DSP_FEATURE_WAITDIALTONE   (1 << 20)
#define	DSP_PROGRESS_BUSY   (1 << 18)
#define	DSP_PROGRESS_CONGESTION   (1 << 19)
#define	DSP_PROGRESS_RINGING   (1 << 17)
#define	DSP_PROGRESS_TALK   (1 << 16)
#define	DSP_TONE_STATE_BUSY   4
#define	DSP_TONE_STATE_DIALTONE   2
#define	DSP_TONE_STATE_HUNGUP   8
#define	DSP_TONE_STATE_RINGING   1
#define	DSP_TONE_STATE_SILENCE   0
#define	DSP_TONE_STATE_SPECIAL1   5
#define	DSP_TONE_STATE_SPECIAL2   6
#define	DSP_TONE_STATE_SPECIAL3   7
#define	DSP_TONE_STATE_TALKING   3

Enumerations
enum	threshold { THRESHOLD_SILENCE = 0, THRESHOLD_MAX = 1 }

Functions
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...
int	ast_dsp_digitdetect (struct ast_dsp *dsp, struct ast_frame *f)
	Return non-zero if DTMF hit was found. 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...
int	ast_dsp_getdigits (struct ast_dsp *dsp, char *buf, int max)
	Get pending DTMF/MF digits. 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 *inf)
	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...
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 freq1, 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...
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...

Detailed Description

Convenient Signal Processing routines.

Definition in file dsp.h.

Macro Definition Documentation

DSP_DIGITMODE_DTMF

#define DSP_DIGITMODE_DTMF   0

Detect DTMF digits

Definition at line 31 of file dsp.h.

Referenced by __ast_dsp_new(), analog_ss_thread(), ast_dsp_set_digitmode(), dahdi_hangup(), dahdi_new(), dahdi_setoption(), enable_dsp_detect(), mbl_load_device(), mkintf(), my_dsp_set_digitmode(), and ooh323_new().

DSP_DIGITMODE_MF

#define DSP_DIGITMODE_MF   1

Detect MF digits

Definition at line 32 of file dsp.h.

Referenced by __ast_dsp_new(), analog_ss_thread(), ast_dsp_digitreset(), ast_dsp_process(), ast_dsp_set_digitmode(), my_dsp_set_digitmode(), and read_mf_digits().

DSP_DIGITMODE_MUTECONF

#define DSP_DIGITMODE_MUTECONF   (1 << 9)

Mute conference

Definition at line 35 of file dsp.h.

Referenced by ast_dsp_set_digitmode(), dahdi_setoption(), read_mf_exec(), and read_sf_exec().

DSP_DIGITMODE_MUTEMAX

#define DSP_DIGITMODE_MUTEMAX   (1 << 10)

Delay audio by a frame to try to extra quelch

Definition at line 36 of file dsp.h.

Referenced by ast_dsp_set_digitmode(), dahdi_setoption(), read_mf_exec(), and read_sf_exec().

DSP_DIGITMODE_NOQUELCH

#define DSP_DIGITMODE_NOQUELCH   (1 << 8)

Do not quelch DTMF from in-band

Definition at line 34 of file dsp.h.

Referenced by ast_dsp_process(), mgcp_new(), read_mf_exec(), and read_sf_exec().

DSP_DIGITMODE_RELAXDTMF

#define DSP_DIGITMODE_RELAXDTMF   (1 << 11)

"Radio" mode (relaxed DTMF)

Definition at line 37 of file dsp.h.

Referenced by ast_dsp_process(), dahdi_setoption(), enable_dsp_detect(), mbl_load_device(), ooh323_new(), process_dahdi(), read_mf_exec(), and read_sf_exec().

DSP_FAXMODE_DETECT_ALL

#define DSP_FAXMODE_DETECT_ALL   (DSP_FAXMODE_DETECT_CNG | DSP_FAXMODE_DETECT_CED)

Definition at line 50 of file dsp.h.

DSP_FAXMODE_DETECT_CED

#define DSP_FAXMODE_DETECT_CED   (1 << 1)

Definition at line 48 of file dsp.h.

Referenced by ast_dsp_process(), and ooh323_new().

DSP_FAXMODE_DETECT_CNG

#define DSP_FAXMODE_DETECT_CNG   (1 << 0)

Definition at line 47 of file dsp.h.

Referenced by __ast_dsp_new(), ast_dsp_process(), fax_detect_new(), and ooh323_new().

DSP_FAXMODE_DETECT_SQUELCH

#define DSP_FAXMODE_DETECT_SQUELCH   (1 << 2)

Definition at line 49 of file dsp.h.

Referenced by ast_fax_detect_init(), and fax_detect_new().

DSP_FEATURE_BUSY_DETECT

#define DSP_FEATURE_BUSY_DETECT   (1 << 1)

Definition at line 27 of file dsp.h.

Referenced by ast_dsp_process(), and dahdi_new().

DSP_FEATURE_CALL_PROGRESS

#define DSP_FEATURE_CALL_PROGRESS   (DSP_PROGRESS_TALK | DSP_PROGRESS_RINGING | DSP_PROGRESS_BUSY | DSP_PROGRESS_CONGESTION)

Definition at line 43 of file dsp.h.

Referenced by __ast_dsp_call_progress(), ast_dsp_process(), and dahdi_new().

DSP_FEATURE_DIGIT_DETECT

#define DSP_FEATURE_DIGIT_DETECT   (1 << 3)

Definition at line 28 of file dsp.h.

Referenced by ast_dsp_process(), ast_dsp_set_features(), ast_sip_session_alloc(), dahdi_dtmf_detect_disable(), dahdi_dtmf_detect_enable(), dahdi_new(), enable_dsp_detect(), mbl_load_device(), mgcp_new(), misdn_set_opt_exec(), ooh323_new(), pjsip_acf_dtmf_mode_write(), read_config(), read_mf_digits(), set_caps(), and sip_rtp_read().

DSP_FEATURE_FAX_DETECT

#define DSP_FEATURE_FAX_DETECT   (1 << 4)

Definition at line 29 of file dsp.h.

Referenced by ast_dsp_process(), ast_sip_session_alloc(), chan_pjsip_cng_tone_detected(), chan_pjsip_read_stream(), dahdi_handle_dtmf(), dahdi_new(), dahdi_queryoption(), dahdi_read(), dahdi_setoption(), enable_dsp_detect(), fax_detect_new(), misdn_set_opt_exec(), my_handle_dtmf(), ooh323_new(), and read_config().

DSP_FEATURE_FREQ_DETECT

#define DSP_FEATURE_FREQ_DETECT   (1 << 21)

Enable arbitrary tone detection

Definition at line 45 of file dsp.h.

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

DSP_FEATURE_SILENCE_SUPPRESS

#define DSP_FEATURE_SILENCE_SUPPRESS   (1 << 0)

Definition at line 26 of file dsp.h.

Referenced by __ast_dsp_new(), and ast_dsp_process().

DSP_FEATURE_WAITDIALTONE

#define DSP_FEATURE_WAITDIALTONE   (1 << 20)

Enable dial tone detection

Definition at line 44 of file dsp.h.

Referenced by ast_dsp_process(), dahdi_new(), and dahdi_read().

DSP_PROGRESS_BUSY

#define DSP_PROGRESS_BUSY   (1 << 18)

Enable busy tone detection

Definition at line 41 of file dsp.h.

Referenced by __ast_dsp_call_progress().

DSP_PROGRESS_CONGESTION

#define DSP_PROGRESS_CONGESTION   (1 << 19)

Enable congestion tone detection

Definition at line 42 of file dsp.h.

Referenced by __ast_dsp_call_progress().

DSP_PROGRESS_RINGING

#define DSP_PROGRESS_RINGING   (1 << 17)

Enable calling tone detection

Definition at line 40 of file dsp.h.

Referenced by __ast_dsp_call_progress().

DSP_PROGRESS_TALK

#define DSP_PROGRESS_TALK   (1 << 16)

Enable talk detection

Definition at line 39 of file dsp.h.

Referenced by __ast_dsp_call_progress(), and dahdi_new().

DSP_TONE_STATE_BUSY

#define DSP_TONE_STATE_BUSY   4

Definition at line 56 of file dsp.h.

Referenced by __ast_dsp_call_progress().

DSP_TONE_STATE_DIALTONE

#define DSP_TONE_STATE_DIALTONE   2

Definition at line 54 of file dsp.h.

Referenced by __ast_dsp_call_progress(), and dahdi_read().

DSP_TONE_STATE_HUNGUP

#define DSP_TONE_STATE_HUNGUP   8

Definition at line 60 of file dsp.h.

Referenced by __ast_dsp_call_progress().

DSP_TONE_STATE_RINGING

#define DSP_TONE_STATE_RINGING   1

Definition at line 53 of file dsp.h.

Referenced by __ast_dsp_call_progress(), and dahdi_read().

DSP_TONE_STATE_SILENCE

#define DSP_TONE_STATE_SILENCE   0

Definition at line 52 of file dsp.h.

Referenced by __ast_dsp_call_progress().

DSP_TONE_STATE_SPECIAL1

#define DSP_TONE_STATE_SPECIAL1   5

Definition at line 57 of file dsp.h.

Referenced by __ast_dsp_call_progress().

DSP_TONE_STATE_SPECIAL2

#define DSP_TONE_STATE_SPECIAL2   6

Definition at line 58 of file dsp.h.

Referenced by __ast_dsp_call_progress().

DSP_TONE_STATE_SPECIAL3

#define DSP_TONE_STATE_SPECIAL3   7

Definition at line 59 of file dsp.h.

Referenced by __ast_dsp_call_progress().

DSP_TONE_STATE_TALKING

#define DSP_TONE_STATE_TALKING   3

Definition at line 55 of file dsp.h.

Referenced by __ast_dsp_call_progress().

Enumeration Type Documentation

threshold

enum threshold

Enumerator
THRESHOLD_SILENCE
THRESHOLD_MAX

Definition at line 71 of file dsp.h.

               {
   /* Array offsets */
   THRESHOLD_SILENCE = 0,
   /* Always the last */
   THRESHOLD_MAX = 1,
};

Function Documentation

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_digitdetect()

int ast_dsp_digitdetect	(	struct ast_dsp *	dsp,
		struct ast_frame *	f
	)

Return non-zero if DTMF hit was found.

ast_dsp_digitreset()

void ast_dsp_digitreset

(

struct ast_dsp *

dsp

)

Reset DTMF detector.

Definition at line 1797 of file dsp.c.

References dtmf_detect_state_t::col_out, digit_detect_state_t::current_digits, dtmf_detect_state_t::current_hit, mf_detect_state_t::current_hit, dtmf_detect_state_t::current_sample, mf_detect_state_t::current_sample, ast_dsp::digit_state, ast_dsp::digitmode, digit_detect_state_t::digits, DSP_DIGITMODE_MF, digit_detect_state_t::dtmf, ast_dsp::dtmf_began, dtmf_detect_state_t::energy, goertzel_reset(), dtmf_detect_state_t::hits, mf_detect_state_t::hits, dtmf_detect_state_t::lasthit, digit_detect_state_t::mf, dtmf_detect_state_t::misses, dtmf_detect_state_t::row_out, digit_detect_state_t::td, and mf_detect_state_t::tone_out.

Referenced by analog_ss_thread(), mbl_new(), 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

)

Examples:

/usr/src/asterisk-18.5.0/main/app.c.

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

Examples:

/usr/src/asterisk-18.5.0/main/app.c.

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_getdigits()

int ast_dsp_getdigits	(	struct ast_dsp *	dsp,
		char *	buf,
		int	max
	)

Get pending DTMF/MF digits.

ast_dsp_new()

struct ast_dsp* ast_dsp_new

(

void

)

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

Examples:

/usr/src/asterisk-18.5.0/main/app.c.

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 *	inf
	)

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_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	freq1,
		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.

Examples:

/usr/src/asterisk-18.5.0/main/app.c.

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.

Examples:

/usr/src/asterisk-18.5.0/main/app.c.

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_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);
}