/* * Asterisk -- An open source telephony toolkit. * * Copyright (C) 1999 - 2005, Digium, Inc. * * Mark Spencer * * 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_V21 (1 << 2) #define DSP_FAXMODE_DETECT_ALL (DSP_FAXMODE_DETECT_CNG | DSP_FAXMODE_DETECT_CED | DSP_FAXMODE_DETECT_V21) #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 threshold value for silence */ 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 Return non-zero if this is silence. Updates "totalsilence" with the total number of seconds of silence */ int ast_dsp_silence(struct ast_dsp *dsp, struct ast_frame *f, int *totalsilence); /*! \brief 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_silence_with_energy(struct ast_dsp *dsp, struct ast_frame *f, int *totalsilence, int *frames_energy); /*! * \brief Return non-zero if this is noise. Updates "totalnoise" with the total * number of seconds of noise * \since 1.6.1 */ 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 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 */