Mon Jun 27 16:50:54 2011

Asterisk developer's documentation


func_speex.c

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00001 /*
00002  * Asterisk -- An open source telephony toolkit.
00003  *
00004  * Copyright (C) 2008, Digium, Inc.
00005  *
00006  * Brian Degenhardt <bmd@digium.com>
00007  * Brett Bryant <bbryant@digium.com> 
00008  *
00009  * See http://www.asterisk.org for more information about
00010  * the Asterisk project. Please do not directly contact
00011  * any of the maintainers of this project for assistance;
00012  * the project provides a web site, mailing lists and IRC
00013  * channels for your use.
00014  *
00015  * This program is free software, distributed under the terms of
00016  * the GNU General Public License Version 2. See the LICENSE file
00017  * at the top of the source tree.
00018  */
00019 
00020 /*! \file
00021  *
00022  * \brief Noise reduction and automatic gain control (AGC)
00023  *
00024  * \author Brian Degenhardt <bmd@digium.com> 
00025  * \author Brett Bryant <bbryant@digium.com> 
00026  *
00027  * \ingroup functions
00028  *
00029  * \extref The Speex 1.2 library - http://www.speex.org
00030  * \note Requires the 1.2 version of the Speex library (which might not be what you find in Linux packages)
00031  */
00032 
00033 /*** MODULEINFO
00034    <depend>speex</depend>
00035    <depend>speex_preprocess</depend>
00036    <use>speexdsp</use>
00037  ***/
00038 
00039 #include "asterisk.h"
00040 
00041 ASTERISK_FILE_VERSION(__FILE__, "$Revision: 227237 $")
00042 
00043 #include <speex/speex_preprocess.h>
00044 #include "asterisk/module.h"
00045 #include "asterisk/channel.h"
00046 #include "asterisk/pbx.h"
00047 #include "asterisk/utils.h"
00048 #include "asterisk/audiohook.h"
00049 
00050 #define DEFAULT_AGC_LEVEL 8000.0
00051 
00052 /*** DOCUMENTATION
00053    <function name="AGC" language="en_US">
00054       <synopsis>
00055          Apply automatic gain control to audio on a channel.
00056       </synopsis>
00057       <syntax>
00058          <parameter name="channeldirection" required="true">
00059             <para>This can be either <literal>rx</literal> or <literal>tx</literal></para>
00060          </parameter>
00061       </syntax>
00062       <description>
00063          <para>The AGC function will apply automatic gain control to the audio on the
00064          channel that it is executed on. Using <literal>rx</literal> for audio received
00065          and <literal>tx</literal> for audio transmitted to the channel. When using this
00066          function you set a target audio level. It is primarily intended for use with
00067          analog lines, but could be useful for other channels as well. The target volume 
00068          is set with a number between <literal>1-32768</literal>. The larger the number
00069          the louder (more gain) the channel will receive.</para>
00070          <para>Examples:</para>
00071          <para>exten => 1,1,Set(AGC(rx)=8000)</para>
00072          <para>exten => 1,2,Set(AGC(tx)=off)</para>
00073       </description>
00074    </function>
00075    <function name="DENOISE" language="en_US">
00076       <synopsis>
00077          Apply noise reduction to audio on a channel.
00078       </synopsis>
00079       <syntax>
00080          <parameter name="channeldirection" required="true">
00081             <para>This can be either <literal>rx</literal> or <literal>tx</literal> 
00082             the values that can be set to this are either <literal>on</literal> and
00083             <literal>off</literal></para>
00084          </parameter>
00085       </syntax>
00086       <description>
00087          <para>The DENOISE function will apply noise reduction to audio on the channel
00088          that it is executed on. It is very useful for noisy analog lines, especially
00089          when adjusting gains or using AGC. Use <literal>rx</literal> for audio received from the channel
00090          and <literal>tx</literal> to apply the filter to the audio being sent to the channel.</para>
00091          <para>Examples:</para>
00092          <para>exten => 1,1,Set(DENOISE(rx)=on)</para>
00093          <para>exten => 1,2,Set(DENOISE(tx)=off)</para>
00094       </description>
00095    </function>
00096  ***/
00097 
00098 struct speex_direction_info {
00099    SpeexPreprocessState *state;  /*!< speex preprocess state object */
00100    int agc;                /*!< audio gain control is enabled or not */
00101    int denoise;               /*!< denoise is enabled or not */
00102    int samples;               /*!< n of 8Khz samples in last frame */
00103    float agclevel;               /*!< audio gain control level [1.0 - 32768.0] */
00104 };
00105 
00106 struct speex_info {
00107    struct ast_audiohook audiohook;
00108    struct speex_direction_info *tx, *rx;
00109 };
00110 
00111 static void destroy_callback(void *data) 
00112 {
00113    struct speex_info *si = data;
00114 
00115    ast_audiohook_destroy(&si->audiohook);
00116 
00117    if (si->rx && si->rx->state) {
00118       speex_preprocess_state_destroy(si->rx->state);
00119    }
00120 
00121    if (si->tx && si->tx->state) {
00122       speex_preprocess_state_destroy(si->tx->state);
00123    }
00124 
00125    if (si->rx) {
00126       ast_free(si->rx);
00127    }
00128 
00129    if (si->tx) {
00130       ast_free(si->tx);
00131    }
00132 
00133    ast_free(data);
00134 };
00135 
00136 static const struct ast_datastore_info speex_datastore = {
00137    .type = "speex",
00138    .destroy = destroy_callback
00139 };
00140 
00141 static int speex_callback(struct ast_audiohook *audiohook, struct ast_channel *chan, struct ast_frame *frame, enum ast_audiohook_direction direction)
00142 {
00143    struct ast_datastore *datastore = NULL;
00144    struct speex_direction_info *sdi = NULL;
00145    struct speex_info *si = NULL;
00146    char source[80];
00147 
00148    /* If the audiohook is stopping it means the channel is shutting down.... but we let the datastore destroy take care of it */
00149    if (audiohook->status == AST_AUDIOHOOK_STATUS_DONE || frame->frametype != AST_FRAME_VOICE) {
00150       return -1;
00151    }
00152 
00153    /* We are called with chan already locked */
00154    if (!(datastore = ast_channel_datastore_find(chan, &speex_datastore, NULL))) {
00155       return -1;
00156    }
00157 
00158    si = datastore->data;
00159 
00160    sdi = (direction == AST_AUDIOHOOK_DIRECTION_READ) ? si->rx : si->tx;
00161 
00162    if (!sdi) {
00163       return -1;
00164    }
00165 
00166    if (sdi->samples != frame->samples) {
00167       if (sdi->state) {
00168          speex_preprocess_state_destroy(sdi->state);
00169       }
00170 
00171       if (!(sdi->state = speex_preprocess_state_init((sdi->samples = frame->samples), 8000))) {
00172          return -1;
00173       }
00174 
00175       speex_preprocess_ctl(sdi->state, SPEEX_PREPROCESS_SET_AGC, &sdi->agc);
00176 
00177       if (sdi->agc) {
00178          speex_preprocess_ctl(sdi->state, SPEEX_PREPROCESS_SET_AGC_LEVEL, &sdi->agclevel);
00179       }
00180 
00181       speex_preprocess_ctl(sdi->state, SPEEX_PREPROCESS_SET_DENOISE, &sdi->denoise);
00182    }
00183 
00184    speex_preprocess(sdi->state, frame->data.ptr, NULL);
00185    snprintf(source, sizeof(source), "%s/speex", frame->src);
00186    if (frame->mallocd & AST_MALLOCD_SRC) {
00187       ast_free((char *) frame->src);
00188    }
00189    frame->src = ast_strdup(source);
00190    frame->mallocd |= AST_MALLOCD_SRC;
00191 
00192    return 0;
00193 }
00194 
00195 static int speex_write(struct ast_channel *chan, const char *cmd, char *data, const char *value)
00196 {
00197    struct ast_datastore *datastore = NULL;
00198    struct speex_info *si = NULL;
00199    struct speex_direction_info **sdi = NULL;
00200    int is_new = 0;
00201 
00202    ast_channel_lock(chan);
00203    if (!(datastore = ast_channel_datastore_find(chan, &speex_datastore, NULL))) {
00204       ast_channel_unlock(chan);
00205 
00206       if (!(datastore = ast_datastore_alloc(&speex_datastore, NULL))) {
00207          return 0;
00208       }
00209 
00210       if (!(si = ast_calloc(1, sizeof(*si)))) {
00211          ast_datastore_free(datastore);
00212          return 0;
00213       }
00214 
00215       ast_audiohook_init(&si->audiohook, AST_AUDIOHOOK_TYPE_MANIPULATE, "speex");
00216       si->audiohook.manipulate_callback = speex_callback;
00217 
00218       is_new = 1;
00219    } else {
00220       ast_channel_unlock(chan);
00221       si = datastore->data;
00222    }
00223 
00224    if (!strcasecmp(data, "rx")) {
00225       sdi = &si->rx;
00226    } else if (!strcasecmp(data, "tx")) {
00227       sdi = &si->tx;
00228    } else {
00229       ast_log(LOG_ERROR, "Invalid argument provided to the %s function\n", cmd);
00230 
00231       if (is_new) {
00232          ast_datastore_free(datastore);
00233          return -1;
00234       }
00235    }
00236 
00237    if (!*sdi) {
00238       if (!(*sdi = ast_calloc(1, sizeof(**sdi)))) {
00239          return 0;
00240       }
00241       /* Right now, the audiohooks API will _only_ provide us 8 kHz slinear
00242        * audio.  When it supports 16 kHz (or any other sample rates, we will
00243        * have to take that into account here. */
00244       (*sdi)->samples = -1;
00245    }
00246 
00247    if (!strcasecmp(cmd, "agc")) {
00248       if (!sscanf(value, "%30f", &(*sdi)->agclevel))
00249          (*sdi)->agclevel = ast_true(value) ? DEFAULT_AGC_LEVEL : 0.0;
00250    
00251       if ((*sdi)->agclevel > 32768.0) {
00252          ast_log(LOG_WARNING, "AGC(%s)=%.01f is greater than 32768... setting to 32768 instead\n", 
00253                ((*sdi == si->rx) ? "rx" : "tx"), (*sdi)->agclevel);
00254          (*sdi)->agclevel = 32768.0;
00255       }
00256    
00257       (*sdi)->agc = !!((*sdi)->agclevel);
00258 
00259       if ((*sdi)->state) {
00260          speex_preprocess_ctl((*sdi)->state, SPEEX_PREPROCESS_SET_AGC, &(*sdi)->agc);
00261          if ((*sdi)->agc) {
00262             speex_preprocess_ctl((*sdi)->state, SPEEX_PREPROCESS_SET_AGC_LEVEL, &(*sdi)->agclevel);
00263          }
00264       }
00265    } else if (!strcasecmp(cmd, "denoise")) {
00266       (*sdi)->denoise = (ast_true(value) != 0);
00267 
00268       if ((*sdi)->state) {
00269          speex_preprocess_ctl((*sdi)->state, SPEEX_PREPROCESS_SET_DENOISE, &(*sdi)->denoise);
00270       }
00271    }
00272 
00273    if (!(*sdi)->agc && !(*sdi)->denoise) {
00274       if ((*sdi)->state)
00275          speex_preprocess_state_destroy((*sdi)->state);
00276 
00277       ast_free(*sdi);
00278       *sdi = NULL;
00279    }
00280 
00281    if (!si->rx && !si->tx) {
00282       if (is_new) {
00283          is_new = 0;
00284       } else {
00285          ast_channel_lock(chan);
00286          ast_channel_datastore_remove(chan, datastore);
00287          ast_channel_unlock(chan);
00288          ast_audiohook_remove(chan, &si->audiohook);
00289          ast_audiohook_detach(&si->audiohook);
00290       }
00291       
00292       ast_datastore_free(datastore);
00293    }
00294 
00295    if (is_new) { 
00296       datastore->data = si;
00297       ast_channel_lock(chan);
00298       ast_channel_datastore_add(chan, datastore);
00299       ast_channel_unlock(chan);
00300       ast_audiohook_attach(chan, &si->audiohook);
00301    }
00302 
00303    return 0;
00304 }
00305 
00306 static int speex_read(struct ast_channel *chan, const char *cmd, char *data, char *buf, size_t len)
00307 {
00308    struct ast_datastore *datastore = NULL;
00309    struct speex_info *si = NULL;
00310    struct speex_direction_info *sdi = NULL;
00311 
00312    if (!chan) {
00313       ast_log(LOG_ERROR, "%s cannot be used without a channel!\n", cmd);
00314       return -1;
00315    }
00316 
00317    ast_channel_lock(chan);
00318    if (!(datastore = ast_channel_datastore_find(chan, &speex_datastore, NULL))) {
00319       ast_channel_unlock(chan);
00320       return -1;
00321    }
00322    ast_channel_unlock(chan);
00323 
00324    si = datastore->data;
00325 
00326    if (!strcasecmp(data, "tx"))
00327       sdi = si->tx;
00328    else if (!strcasecmp(data, "rx"))
00329       sdi = si->rx;
00330    else {
00331       ast_log(LOG_ERROR, "%s(%s) must either \"tx\" or \"rx\"\n", cmd, data);
00332       return -1;
00333    }
00334 
00335    if (!strcasecmp(cmd, "agc"))
00336       snprintf(buf, len, "%.01f", sdi ? sdi->agclevel : 0.0);
00337    else
00338       snprintf(buf, len, "%d", sdi ? sdi->denoise : 0);
00339 
00340    return 0;
00341 }
00342 
00343 static struct ast_custom_function agc_function = {
00344    .name = "AGC",
00345    .write = speex_write,
00346    .read = speex_read,
00347    .read_max = 22,
00348 };
00349 
00350 static struct ast_custom_function denoise_function = {
00351    .name = "DENOISE",
00352    .write = speex_write,
00353    .read = speex_read,
00354    .read_max = 22,
00355 };
00356 
00357 static int unload_module(void)
00358 {
00359    ast_custom_function_unregister(&agc_function);
00360    ast_custom_function_unregister(&denoise_function);
00361    return 0;
00362 }
00363 
00364 static int load_module(void)
00365 {
00366    if (ast_custom_function_register(&agc_function)) {
00367       return AST_MODULE_LOAD_DECLINE;
00368    }
00369 
00370    if (ast_custom_function_register(&denoise_function)) {
00371       ast_custom_function_unregister(&agc_function);
00372       return AST_MODULE_LOAD_DECLINE;
00373    }
00374 
00375    return AST_MODULE_LOAD_SUCCESS;
00376 }
00377 
00378 AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Noise reduction and Automatic Gain Control (AGC)");

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