123456789101112131415161718192021222324252627282930313233343536373839404142434445464748495051525354555657585960616263646566676869707172737475767778798081828384858687888990919293949596979899100101102103104105106107108109110111112113114115116117118119120121122123124125126127128129130131132133134135136137138139140141142143144145146147148149150151152153154155156157158159160161162163164165166167168169170171172173174175176177178179180181182183184185186187188189190191192193194195196197198199200201202203204205206207208 |
- /*
- * Asterisk -- An open source telephony toolkit.
- *
- * Copyright (C) 1999 - 2005, Digium, Inc.
- *
- * Mark Spencer <markster@digium.com>
- *
- * See http://www.asterisk.org for more information about
- * the Asterisk project. Please do not directly contact
- * any of the maintainers of this project for assistance;
- * the project provides a web site, mailing lists and IRC
- * channels for your use.
- *
- * This program is free software, distributed under the terms of
- * the GNU General Public License Version 2. See the LICENSE file
- * at the top of the source tree.
- */
- /*! \file
- * \brief Convenient Signal Processing routines
- */
- #ifndef _ASTERISK_DSP_H
- #define _ASTERISK_DSP_H
- #define DSP_FEATURE_SILENCE_SUPPRESS (1 << 0)
- #define DSP_FEATURE_BUSY_DETECT (1 << 1)
- #define DSP_FEATURE_DIGIT_DETECT (1 << 3)
- #define DSP_FEATURE_FAX_DETECT (1 << 4)
- #define DSP_DIGITMODE_DTMF 0 /*!< Detect DTMF digits */
- #define DSP_DIGITMODE_MF 1 /*!< Detect MF digits */
- #define DSP_DIGITMODE_NOQUELCH (1 << 8) /*!< Do not quelch DTMF from in-band */
- #define DSP_DIGITMODE_MUTECONF (1 << 9) /*!< Mute conference */
- #define DSP_DIGITMODE_MUTEMAX (1 << 10) /*!< Delay audio by a frame to try to extra quelch */
- #define DSP_DIGITMODE_RELAXDTMF (1 << 11) /*!< "Radio" mode (relaxed DTMF) */
- #define DSP_PROGRESS_TALK (1 << 16) /*!< Enable talk detection */
- #define DSP_PROGRESS_RINGING (1 << 17) /*!< Enable calling tone detection */
- #define DSP_PROGRESS_BUSY (1 << 18) /*!< Enable busy tone detection */
- #define DSP_PROGRESS_CONGESTION (1 << 19) /*!< Enable congestion tone detection */
- #define DSP_FEATURE_CALL_PROGRESS (DSP_PROGRESS_TALK | DSP_PROGRESS_RINGING | DSP_PROGRESS_BUSY | DSP_PROGRESS_CONGESTION)
- #define DSP_FEATURE_WAITDIALTONE (1 << 20) /*!< Enable dial tone detection */
- #define DSP_FAXMODE_DETECT_CNG (1 << 0)
- #define DSP_FAXMODE_DETECT_CED (1 << 1)
- #define DSP_FAXMODE_DETECT_SQUELCH (1 << 2)
- #define DSP_FAXMODE_DETECT_ALL (DSP_FAXMODE_DETECT_CNG | DSP_FAXMODE_DETECT_CED)
- #define DSP_TONE_STATE_SILENCE 0
- #define DSP_TONE_STATE_RINGING 1
- #define DSP_TONE_STATE_DIALTONE 2
- #define DSP_TONE_STATE_TALKING 3
- #define DSP_TONE_STATE_BUSY 4
- #define DSP_TONE_STATE_SPECIAL1 5
- #define DSP_TONE_STATE_SPECIAL2 6
- #define DSP_TONE_STATE_SPECIAL3 7
- #define DSP_TONE_STATE_HUNGUP 8
- struct ast_dsp;
- struct ast_dsp_busy_pattern {
- /*! Number of elements. */
- int length;
- /*! Pattern elements in on/off time durations. */
- int pattern[4];
- };
- enum threshold {
- /* Array offsets */
- THRESHOLD_SILENCE = 0,
- /* Always the last */
- THRESHOLD_MAX = 1,
- };
- /*! \brief Allocates a new dsp with a specific internal sample rate used
- * during processing. */
- struct ast_dsp *ast_dsp_new_with_rate(unsigned int sample_rate);
- /*! \brief Allocates a new dsp, assumes 8khz for internal sample rate */
- struct ast_dsp *ast_dsp_new(void);
- void ast_dsp_free(struct ast_dsp *dsp);
- /*! \brief Retrieve the sample rate this DSP structure was
- * created with */
- unsigned int ast_dsp_get_sample_rate(const struct ast_dsp *dsp);
- /*! \brief Set the minimum average magnitude threshold to determine talking by the DSP. */
- void ast_dsp_set_threshold(struct ast_dsp *dsp, int threshold);
- /*! \brief Set number of required cadences for busy */
- void ast_dsp_set_busy_count(struct ast_dsp *dsp, int cadences);
- /*! \brief Set expected lengths of the busy tone */
- void ast_dsp_set_busy_pattern(struct ast_dsp *dsp, const struct ast_dsp_busy_pattern *cadence);
- /*! \brief Scans for progress indication in audio */
- int ast_dsp_call_progress(struct ast_dsp *dsp, struct ast_frame *inf);
- /*! \brief Set zone for doing progress detection */
- int ast_dsp_set_call_progress_zone(struct ast_dsp *dsp, char *zone);
- /*! \brief Return AST_FRAME_NULL frames when there is silence, AST_FRAME_BUSY on
- busies, and call progress, all dependent upon which features are enabled */
- struct ast_frame *ast_dsp_process(struct ast_channel *chan, struct ast_dsp *dsp, struct ast_frame *inf);
- /*!
- * \brief Process the audio frame for silence.
- *
- * \param dsp DSP processing audio media.
- * \param f Audio frame to process.
- * \param totalsilence Variable to set to the total accumulated silence in ms
- * seen by the DSP since the last noise.
- *
- * \return Non-zero if the frame is silence.
- */
- int ast_dsp_silence(struct ast_dsp *dsp, struct ast_frame *f, int *totalsilence);
- /*!
- * \brief Process the audio frame for silence.
- *
- * \param dsp DSP processing audio media.
- * \param f Audio frame to process.
- * \param totalsilence Variable to set to the total accumulated silence in ms
- * seen by the DSP since the last noise.
- * \param frames_energy Variable to set to the average energy of the samples in the frame.
- *
- * \return Non-zero if the frame is silence.
- */
- int ast_dsp_silence_with_energy(struct ast_dsp *dsp, struct ast_frame *f, int *totalsilence, int *frames_energy);
- /*!
- * \brief Process the audio frame for noise.
- * \since 1.6.1
- *
- * \param dsp DSP processing audio media.
- * \param f Audio frame to process.
- * \param totalnoise Variable to set to the total accumulated noise in ms
- * seen by the DSP since the last silence.
- *
- * \return Non-zero if the frame is silence.
- */
- int ast_dsp_noise(struct ast_dsp *dsp, struct ast_frame *f, int *totalnoise);
- /*! \brief Return non-zero if historically this should be a busy, request that
- ast_dsp_silence has already been called */
- int ast_dsp_busydetect(struct ast_dsp *dsp);
- /*! \brief Return non-zero if DTMF hit was found */
- int ast_dsp_digitdetect(struct ast_dsp *dsp, struct ast_frame *f);
- /*! \brief Reset total silence count */
- void ast_dsp_reset(struct ast_dsp *dsp);
- /*! \brief Reset DTMF detector */
- void ast_dsp_digitreset(struct ast_dsp *dsp);
- /*! \brief Select feature set */
- void ast_dsp_set_features(struct ast_dsp *dsp, int features);
- /*! \brief Get features */
- int ast_dsp_get_features(struct ast_dsp *dsp);
- /*! \brief Get pending DTMF/MF digits */
- int ast_dsp_getdigits(struct ast_dsp *dsp, char *buf, int max);
- /*! \brief Set digit mode
- * \version 1.6.1 renamed from ast_dsp_digitmode to ast_dsp_set_digitmode
- */
- int ast_dsp_set_digitmode(struct ast_dsp *dsp, int digitmode);
- /*! \brief Set fax mode */
- int ast_dsp_set_faxmode(struct ast_dsp *dsp, int faxmode);
- /*!
- * \brief 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
- */
- int ast_dsp_was_muted(struct ast_dsp *dsp);
- /*! \brief Get tstate (Tone State) */
- int ast_dsp_get_tstate(struct ast_dsp *dsp);
- /*! \brief Get tcount (Threshold counter) */
- int ast_dsp_get_tcount(struct ast_dsp *dsp);
- /*!
- * \brief Get silence threshold from dsp.conf
- * \since 1.6.1
- */
- int ast_dsp_get_threshold_from_settings(enum threshold which);
- /*!
- * \brief Reloads dsp settings from dsp.conf
- * \since 1.6.1
- */
- int ast_dsp_reload(void);
- /*!
- * \brief Load dsp settings from dsp.conf
- * \since 1.6.1
- */
- int ast_dsp_init(void);
- #endif /* _ASTERISK_DSP_H */
|