cont_ad.h

来自「CMU大名鼎鼎的SPHINX-3大词汇量连续语音识别系统」· C头文件 代码 · 共 313 行

H
313
字号
/* ==================================================================== * Copyright (c) 1999-2001 Carnegie Mellon University.  All rights * reserved. * * Redistribution and use in source and binary forms, with or without * modification, are permitted provided that the following conditions * are met: * * 1. Redistributions of source code must retain the above copyright *    notice, this list of conditions and the following disclaimer.  * * 2. Redistributions in binary form must reproduce the above copyright *    notice, this list of conditions and the following disclaimer in *    the documentation and/or other materials provided with the *    distribution. * * This work was supported in part by funding from the Defense Advanced  * Research Projects Agency and the National Science Foundation of the  * United States of America, and the CMU Sphinx Speech Consortium. * * THIS SOFTWARE IS PROVIDED BY CARNEGIE MELLON UNIVERSITY ``AS IS'' AND  * ANY EXPRESSED OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO,  * THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR * PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL CARNEGIE MELLON UNIVERSITY * NOR ITS EMPLOYEES BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT  * LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,  * DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY  * THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT  * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE  * OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. * * ==================================================================== * *//* * cont_ad.h -- Continuous A/D listening and silence filtering module. *  * ********************************************** * CMU ARPA Speech Project * * Copyright (c) 1996 Carnegie Mellon University. * ALL RIGHTS RESERVED. * ********************************************** *  * HISTORY *  * 13-Jul-98	M K Ravishankar (rkm@cs.cmu.edu) at Carnegie Mellon University * 		Added spf and adbufsize to cont_ad_t in order to support variable * 		frame sizes depending on audio sampling rate. *  * 30-Jun-98	M K Ravishankar (rkm@cs.cmu.edu) at Carnegie Mellon University * 		Added FILE* argument to cont_ad_powhist_dump(). *  * 16-Jan-98	Paul Placeway (pwp@cs.cmu.edu) at Carnegie Mellon University * 		Changed to use dB instead of the weird power measure. * 		Added most system parameters to cont_ad_t instead of hardwiring * 		them in cont_ad.c. * 		Added cont_ad_set_params() and cont_ad_get_params(). *  * 28-Jul-96	M K Ravishankar (rkm@cs.cmu.edu) at Carnegie Mellon University * 		Added cont_ad_t.siglvl. *  * 27-Jun-96	M K Ravishankar (rkm@cs.cmu.edu) at Carnegie Mellon University * 		Added the option for cont_ad_read to return -1 on EOF. *  * 21-Jun-96	M K Ravishankar (rkm@cs.cmu.edu) at Carnegie Mellon University * 		Added cont_ad_set_thresh(). *  * 20-Jun-96	M K Ravishankar (rkm@cs.cmu.edu) at Carnegie Mellon University * 		Separated thresholds for speech and silence. *  * 17-Jun-96	M K Ravishankar (rkm@cs.cmu.edu) at Carnegie Mellon University * 		Created, based loosely on Steve Reed's original implementation. */#ifndef _CONT_AD_H_#define _CONT_AD_H_#include <s3types.h>/** * This module is intended to be interposed as a filter between any raw A/D source and the * application to remove silence regions.  Its main purpose is to remove regions of * silence from the raw input speech.  It is initialized with a raw A/D source function * (during the cont_ad_init call).  The application is responsible for setting up the A/D * source, turning recording on and off as it desires.  Filtered A/D data can be read by * the application using the cont_ad_read function. *  * In other words, the application calls cont_ad_read instead of the raw A/D source function * (e.g., ad_read in libad) to obtain filtered A/D data with silence regions removed.  This * module itself does not enforce any other structural changes to the application. *  * The cont_ad_read function also updates an "absolute" timestamp (see cont_ad_t.read_ts) at * the end of each invocation.  The timestamp indicates the total #samples of A/D data read * until this point, including data discarded as silence frames.  The application is * responsible for using this timestamp to make any policy decisions regarding utterance * boundaries or whatever. */#include <stdio.h>/** * Data structure for maintaining speech (non-silence) segments not yet consumed by the * application.  FOR THE INTERNAL USE OF THIS MODULE. */typedef struct spseg_s {    int32 startfrm;	/* Frame-id in adbuf (see below) of start of this segment */     int32 nfrm;		/* #frames in segment (may wrap around adbuf) */    struct spseg_s *next;	/* Next speech segment (with some intervening silence) */} spseg_t;/** * Continuous listening module or object.  An application can open and maintain several * such objects, if necessary. * FYI: Module always in one of two states: SILENCE or SPEECH.  Transitions between the * two detected by sliding a window spanning several frames and looking for some minimum * number of frames of the other type. */typedef struct {    /* Function to be called for obtaining A/D data (see prototype for ad_read in ad.h) */    int32 (*adfunc)(ad_rec_t *ad, int16 *buf, int32 max);    ad_rec_t *ad;	/* A/D device argument for adfunc.  Also, ad->sps used to			   determine frame size (spf, see below) */    int16 *adbuf;	/* Circular buffer for maintaining A/D data read until consumed */    int32 read_ts;	/* Timestamp (total no. of raw A/D samples read, silence+speech)			   at the end of the most recent cont_ad_read call */    int32 siglvl;	/* Max signal level for most recently read data (0-16; #bits) */    int32 sps;		/* Samples/sec; moved from ad->sps to break dependence on			   ad by N. Roy.*/    int32 spf;		/* Samples/frame; audio level is analyzed within frames */    int32 adbufsize;	/* Buffer size (#samples) */    int32 prev_sample;	/* For pre-emphasis filter */    int32 headfrm;	/* Frame # in adbuf with unconsumed A/D data */    int32 n_frm;	/* #Complete frames of unconsumed A/D data in adbuf */    int32 n_sample;	/* #Samples of unconsumed data in adbuf */    int32 tot_frm;	/* Total #frames of A/D data read, including consumed ones */    int32 noise_level;	/* PWP: what we claim as the "current" noise level */        int32 *pow_hist;	/* Histogram of frame power, moving window, decayed */    char *frm_pow;	/* Frame power */    int32 auto_thresh;  /* Do automatic threshold adjustment or not */    int32 delta_sil;	/* Max silence power/frame ABOVE noise level */    int32 delta_speech;	/* Min speech power/frame ABOVE noise level */    int32 min_noise;	/* noise lower than this we ignore */    int32 max_noise;	/* noise higher than this signals an error */    int32 winsize;	/* how many frames to look at for speech det */    int32 speech_onset;	/* start speech on >= these many frames out of winsize, of >= delta_speech */    int32 sil_onset;	/* end speech on >= these many frames out of winsize, of <= delta_sil */    int32 leader;	/* pad beggining of speech with this many extra frms */    int32 trailer;	/* pad end of speech with this many extra frms */    int32 thresh_speech;/* Frame considered to be speech if power >= thresh_speech			   (for transitioning from SILENCE to SPEECH state) */    int32 thresh_sil;	/* Frame considered to be silence if power <= thresh_sil			   (for transitioning from SPEECH to SILENCE state) */    int32 thresh_update;/* #Frames before next update to pow_hist/thresholds */    float32 adapt_rate;	/* Linear interpolation constant for rate at which noise level adapted			   to each estimate;			   range: 0-1; 0=> no adaptation, 1=> instant adaptation */        int32 state;	/* Current state, SILENCE or SPEECH */    int32 win_startfrm;	/* Where next analysis window begins */    int32 win_validfrm;	/* #Frames currently available from win_startfrm for analysis */    int32 n_other;	/* If in SILENCE state, #frames in analysis window considered to			   be speech; otherwise #frames considered to be silence */    int32 n_in_a_row;	/* number of frames sequentially other side of thresh */    spseg_t *spseg_head;/* First of unconsumed speech segments */    spseg_t *spseg_tail;/* Last of unconsumed speech segments */} cont_ad_t;/* * One time initialization of a continuous listening/silence filtering object/module. * Return value: pointer to a READ-ONLY structure used in other calls to the object. * If any error occurs, the return value is NULL. */cont_ad_t *cont_ad_init (ad_rec_t *ad,	/* In: The A/D source object to be filtered */			 int32 (*adfunc)(ad_rec_t *ad, int16 *buf, int32 max));					/* In: adfunc = source function to be invoked					   to obtain raw A/D data.  See ad.h for the					   required prototype definition. *//** * Calibration to determine an initial silence threshold.  This function can be called * any number of times.  It should be called at least once immediately after cont_ad_init. * The silence threshold is also updated internally once in a while, so this function * only needs to be called in the middle if there is a definite change in the recording * environment. * The application is responsible for making sure that the raw audio source is turned on * before the calibration. * Return value: 0 if successful, <0 otherwise. */int32 cont_ad_calib (cont_ad_t *cont);	/* In: object pointer returned by cont_ad_init *//** * If the application has not passed an audio device into the silence filter * at initialisation,  this routine can be used to calibrate the filter. The * buf (of length max samples) should contain audio data for calibration. This * data is assumed to be completely consumed. More than one call may be * necessary to fully calibrate.  * Return value: 0 if successful, <0 on failure, >0 if calibration not * complete. */int32 cont_ad_calib_loop (cont_ad_t *r, int16 *buf, int32 max); /** * Read A/D data pre-filtered to remove silence segments. * Return value: #samples actually read, possibly 0; <0 if EOF on A/D source. * The function also updates r->read_ts and r->siglvl (see above). */int32 cont_ad_read (cont_ad_t *r,	/* In: Object pointer returned by cont_ad_init */		    int16 *buf,		/* Out: On return, buf contains A/D data returned					   by this function, if any.					   NOTE: buf must be at least 256 samples long */		    int32 max);		/* In: Max #samples to be filled into buf *//** * Set silence and speech threshold parameters.  The silence threshold is the max power * level, RELATIVE to the peak background noise level, in any silence frame.  Similarly, * the speech threshold is the min power level, RELATIVE to the peak background noise * level, in any speech frame.  In general, silence threshold <= speech threshold. * Increasing the thresholds (say, from the default value of 2 to 3 or 4) reduces the * sensitivity to background noise, but may also increase the chances of clipping actual * speech. * Return value: 0 if successful, <0 otherwise. */int32 cont_ad_set_thresh (cont_ad_t *cont,	/* In: Object ptr from cont_ad_init */			  int32 sil,	/* In: silence threshold (default 2) */			  int32 sp);	/* In: speech threshold (default 2) *//** * PWP 1/14/98 -- set the changable params. * *   delta_sil, delta_speech, min_noise, and max_noise are in dB, *   winsize, speech_onset, sil_onset, leader and trailer are in frames of *   16 ms length (256 samples @ 16kHz sampling). */int32 cont_ad_set_params (cont_ad_t *r, int32 delta_sil, int32 delta_speech,			  int32 min_noise, int32 max_noise,			  int32 winsize, int32 speech_onset, int32 sil_onset,			  int32 leader, int32 trailer,			  float32 adapt_rate);/** * PWP 1/14/98 -- get the changable params. * *   delta_sil, delta_speech, min_noise, and max_noise are in dB, *   winsize, speech_onset, sil_onset, leader and trailer are in frames of *   16 ms length (256 samples @ 16kHz sampling). */int32 cont_ad_get_params (cont_ad_t *r, int32 *delta_sil, int32 *delta_speech,			  int32 *min_noise, int32 *max_noise,			  int32 *winsize, int32 *speech_onset, int32 *sil_onset,			  int32 *leader, int32 *trailer,			  float32 *adapt_rate);/** * Reset, discarding any accumulated speech segments. * Return value: 0 if successful, <0 otherwise. */int32 cont_ad_reset (cont_ad_t *cont);	/* In: Object pointer from cont_ad_init *//** * Close the continuous listening object. */int32 cont_ad_close (cont_ad_t *cont);	/* In: Object pointer from cont_ad_init *//** * Dump the power histogram.  For debugging... */void cont_ad_powhist_dump (FILE *fp, cont_ad_t *cont);/** * Detach the given continuous listening module from the associated audio device. * Return 0 if successful, -1 otherwise. */int32 cont_ad_detach (cont_ad_t *c);/** * Attach the continuous listening module to the given audio device/function. * (Like cont_ad_init, but without the calibration.) * Return 0 if successful, -1 otherwise. */int32 cont_ad_attach (cont_ad_t *c, ad_rec_t *a, int32 (*func)(ad_rec_t *, int16 *, int32));void cont_ad_set_logfp (FILE *fp);	/* File containing detailed logs (if non-NULL) *//* * Set the silence and speech thresholds. For this to have any effect, the * auto_thresh field of the continuous listening module should be set to * FALSE. */int32 cont_set_thresh(cont_ad_t *r, int32 silence, int32 speech);#endif

⌨️ 快捷键说明

复制代码Ctrl + C
搜索代码Ctrl + F
全屏模式F11
增大字号Ctrl + =
减小字号Ctrl + -
显示快捷键?