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  * 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 type="external">speexdsp</use>
00037    <support_level>core</support_level>
00038  ***/
00039 
00040 #include "asterisk.h"
00041 
00042 ASTERISK_FILE_VERSION(__FILE__, "$Revision: 419044 $")
00043 
00044 #include <speex/speex_preprocess.h>
00045 #include "asterisk/module.h"
00046 #include "asterisk/channel.h"
00047 #include "asterisk/pbx.h"
00048 #include "asterisk/utils.h"
00049 #include "asterisk/audiohook.h"
00050 
00051 #define DEFAULT_AGC_LEVEL 8000.0
00052 
00053 /*** DOCUMENTATION
00054    <function name="AGC" language="en_US">
00055       <synopsis>
00056          Apply automatic gain control to audio on a channel.
00057       </synopsis>
00058       <syntax>
00059          <parameter name="channeldirection" required="true">
00060             <para>This can be either <literal>rx</literal> or <literal>tx</literal></para>
00061          </parameter>
00062       </syntax>
00063       <description>
00064          <para>The AGC function will apply automatic gain control to the audio on the
00065          channel that it is executed on. Using <literal>rx</literal> for audio received
00066          and <literal>tx</literal> for audio transmitted to the channel. When using this
00067          function you set a target audio level. It is primarily intended for use with
00068          analog lines, but could be useful for other channels as well. The target volume 
00069          is set with a number between <literal>1-32768</literal>. The larger the number
00070          the louder (more gain) the channel will receive.</para>
00071          <para>Examples:</para>
00072          <para>exten => 1,1,Set(AGC(rx)=8000)</para>
00073          <para>exten => 1,2,Set(AGC(tx)=off)</para>
00074       </description>
00075    </function>
00076    <function name="DENOISE" language="en_US">
00077       <synopsis>
00078          Apply noise reduction to audio on a channel.
00079       </synopsis>
00080       <syntax>
00081          <parameter name="channeldirection" required="true">
00082             <para>This can be either <literal>rx</literal> or <literal>tx</literal> 
00083             the values that can be set to this are either <literal>on</literal> and
00084             <literal>off</literal></para>
00085          </parameter>
00086       </syntax>
00087       <description>
00088          <para>The DENOISE function will apply noise reduction to audio on the channel
00089          that it is executed on. It is very useful for noisy analog lines, especially
00090          when adjusting gains or using AGC. Use <literal>rx</literal> for audio received from the channel
00091          and <literal>tx</literal> to apply the filter to the audio being sent to the channel.</para>
00092          <para>Examples:</para>
00093          <para>exten => 1,1,Set(DENOISE(rx)=on)</para>
00094          <para>exten => 1,2,Set(DENOISE(tx)=off)</para>
00095       </description>
00096    </function>
00097  ***/
00098 
00099 struct speex_direction_info {
00100    SpeexPreprocessState *state;  /*!< speex preprocess state object */
00101    int agc;                /*!< audio gain control is enabled or not */
00102    int denoise;               /*!< denoise is enabled or not */
00103    int samples;               /*!< n of 8Khz samples in last frame */
00104    float agclevel;               /*!< audio gain control level [1.0 - 32768.0] */
00105 };
00106 
00107 struct speex_info {
00108    struct ast_audiohook audiohook;
00109    int lastrate;
00110    struct speex_direction_info *tx, *rx;
00111 };
00112 
00113 static void destroy_callback(void *data) 
00114 {
00115    struct speex_info *si = data;
00116 
00117    ast_audiohook_destroy(&si->audiohook);
00118 
00119    if (si->rx && si->rx->state) {
00120       speex_preprocess_state_destroy(si->rx->state);
00121    }
00122 
00123    if (si->tx && si->tx->state) {
00124       speex_preprocess_state_destroy(si->tx->state);
00125    }
00126 
00127    if (si->rx) {
00128       ast_free(si->rx);
00129    }
00130 
00131    if (si->tx) {
00132       ast_free(si->tx);
00133    }
00134 
00135    ast_free(data);
00136 };
00137 
00138 static const struct ast_datastore_info speex_datastore = {
00139    .type = "speex",
00140    .destroy = destroy_callback
00141 };
00142 
00143 static int speex_callback(struct ast_audiohook *audiohook, struct ast_channel *chan, struct ast_frame *frame, enum ast_audiohook_direction direction)
00144 {
00145    struct ast_datastore *datastore = NULL;
00146    struct speex_direction_info *sdi = NULL;
00147    struct speex_info *si = NULL;
00148    char source[80];
00149 
00150    /* If the audiohook is stopping it means the channel is shutting down.... but we let the datastore destroy take care of it */
00151    if (audiohook->status == AST_AUDIOHOOK_STATUS_DONE || frame->frametype != AST_FRAME_VOICE) {
00152       return -1;
00153    }
00154 
00155    /* We are called with chan already locked */
00156    if (!(datastore = ast_channel_datastore_find(chan, &speex_datastore, NULL))) {
00157       return -1;
00158    }
00159 
00160    si = datastore->data;
00161 
00162    sdi = (direction == AST_AUDIOHOOK_DIRECTION_READ) ? si->rx : si->tx;
00163 
00164    if (!sdi) {
00165       return -1;
00166    }
00167 
00168    if ((sdi->samples != frame->samples) || (ast_format_get_sample_rate(frame->subclass.format) != si->lastrate)) {
00169       si->lastrate = ast_format_get_sample_rate(frame->subclass.format);
00170       if (sdi->state) {
00171          speex_preprocess_state_destroy(sdi->state);
00172       }
00173 
00174       if (!(sdi->state = speex_preprocess_state_init((sdi->samples = frame->samples), si->lastrate))) {
00175          return -1;
00176       }
00177 
00178       speex_preprocess_ctl(sdi->state, SPEEX_PREPROCESS_SET_AGC, &sdi->agc);
00179 
00180       if (sdi->agc) {
00181          speex_preprocess_ctl(sdi->state, SPEEX_PREPROCESS_SET_AGC_LEVEL, &sdi->agclevel);
00182       }
00183 
00184       speex_preprocess_ctl(sdi->state, SPEEX_PREPROCESS_SET_DENOISE, &sdi->denoise);
00185    }
00186 
00187    speex_preprocess(sdi->state, frame->data.ptr, NULL);
00188    snprintf(source, sizeof(source), "%s/speex", frame->src);
00189    if (frame->mallocd & AST_MALLOCD_SRC) {
00190       ast_free((char *) frame->src);
00191    }
00192    frame->src = ast_strdup(source);
00193    frame->mallocd |= AST_MALLOCD_SRC;
00194 
00195    return 0;
00196 }
00197 
00198 static int speex_write(struct ast_channel *chan, const char *cmd, char *data, const char *value)
00199 {
00200    struct ast_datastore *datastore = NULL;
00201    struct speex_info *si = NULL;
00202    struct speex_direction_info **sdi = NULL;
00203    int is_new = 0;
00204 
00205    if (!chan) {
00206       ast_log(LOG_WARNING, "No channel was provided to %s function.\n", cmd);
00207       return -1;
00208    }
00209 
00210    if (strcasecmp(data, "rx") && strcasecmp(data, "tx")) {
00211       ast_log(LOG_ERROR, "Invalid argument provided to the %s function\n", cmd);
00212       return -1;
00213    }
00214 
00215    ast_channel_lock(chan);
00216    if (!(datastore = ast_channel_datastore_find(chan, &speex_datastore, NULL))) {
00217       ast_channel_unlock(chan);
00218 
00219       if (!(datastore = ast_datastore_alloc(&speex_datastore, NULL))) {
00220          return 0;
00221       }
00222 
00223       if (!(si = ast_calloc(1, sizeof(*si)))) {
00224          ast_datastore_free(datastore);
00225          return 0;
00226       }
00227 
00228       ast_audiohook_init(&si->audiohook, AST_AUDIOHOOK_TYPE_MANIPULATE, "speex", AST_AUDIOHOOK_MANIPULATE_ALL_RATES);
00229       si->audiohook.manipulate_callback = speex_callback;
00230       si->lastrate = 8000;
00231       is_new = 1;
00232    } else {
00233       ast_channel_unlock(chan);
00234       si = datastore->data;
00235    }
00236 
00237    if (!strcasecmp(data, "rx")) {
00238       sdi = &si->rx;
00239    } else {
00240       sdi = &si->tx;
00241    }
00242 
00243    if (!*sdi) {
00244       if (!(*sdi = ast_calloc(1, sizeof(**sdi)))) {
00245          return 0;
00246       }
00247       /* Right now, the audiohooks API will _only_ provide us 8 kHz slinear
00248        * audio.  When it supports 16 kHz (or any other sample rates, we will
00249        * have to take that into account here. */
00250       (*sdi)->samples = -1;
00251    }
00252 
00253    if (!strcasecmp(cmd, "agc")) {
00254       if (!sscanf(value, "%30f", &(*sdi)->agclevel))
00255          (*sdi)->agclevel = ast_true(value) ? DEFAULT_AGC_LEVEL : 0.0;
00256    
00257       if ((*sdi)->agclevel > 32768.0) {
00258          ast_log(LOG_WARNING, "AGC(%s)=%.01f is greater than 32768... setting to 32768 instead\n", 
00259                ((*sdi == si->rx) ? "rx" : "tx"), (*sdi)->agclevel);
00260          (*sdi)->agclevel = 32768.0;
00261       }
00262    
00263       (*sdi)->agc = !!((*sdi)->agclevel);
00264 
00265       if ((*sdi)->state) {
00266          speex_preprocess_ctl((*sdi)->state, SPEEX_PREPROCESS_SET_AGC, &(*sdi)->agc);
00267          if ((*sdi)->agc) {
00268             speex_preprocess_ctl((*sdi)->state, SPEEX_PREPROCESS_SET_AGC_LEVEL, &(*sdi)->agclevel);
00269          }
00270       }
00271    } else if (!strcasecmp(cmd, "denoise")) {
00272       (*sdi)->denoise = (ast_true(value) != 0);
00273 
00274       if ((*sdi)->state) {
00275          speex_preprocess_ctl((*sdi)->state, SPEEX_PREPROCESS_SET_DENOISE, &(*sdi)->denoise);
00276       }
00277    }
00278 
00279    if (!(*sdi)->agc && !(*sdi)->denoise) {
00280       if ((*sdi)->state)
00281          speex_preprocess_state_destroy((*sdi)->state);
00282 
00283       ast_free(*sdi);
00284       *sdi = NULL;
00285    }
00286 
00287    if (!si->rx && !si->tx) {
00288       if (is_new) {
00289          is_new = 0;
00290       } else {
00291          ast_channel_lock(chan);
00292          ast_channel_datastore_remove(chan, datastore);
00293          ast_channel_unlock(chan);
00294          ast_audiohook_remove(chan, &si->audiohook);
00295          ast_audiohook_detach(&si->audiohook);
00296       }
00297       
00298       ast_datastore_free(datastore);
00299    }
00300 
00301    if (is_new) { 
00302       datastore->data = si;
00303       ast_channel_lock(chan);
00304       ast_channel_datastore_add(chan, datastore);
00305       ast_channel_unlock(chan);
00306       ast_audiohook_attach(chan, &si->audiohook);
00307    }
00308 
00309    return 0;
00310 }
00311 
00312 static int speex_read(struct ast_channel *chan, const char *cmd, char *data, char *buf, size_t len)
00313 {
00314    struct ast_datastore *datastore = NULL;
00315    struct speex_info *si = NULL;
00316    struct speex_direction_info *sdi = NULL;
00317 
00318    if (!chan) {
00319       ast_log(LOG_ERROR, "%s cannot be used without a channel!\n", cmd);
00320       return -1;
00321    }
00322 
00323    ast_channel_lock(chan);
00324    if (!(datastore = ast_channel_datastore_find(chan, &speex_datastore, NULL))) {
00325       ast_channel_unlock(chan);
00326       return -1;
00327    }
00328    ast_channel_unlock(chan);
00329 
00330    si = datastore->data;
00331 
00332    if (!strcasecmp(data, "tx"))
00333       sdi = si->tx;
00334    else if (!strcasecmp(data, "rx"))
00335       sdi = si->rx;
00336    else {
00337       ast_log(LOG_ERROR, "%s(%s) must either \"tx\" or \"rx\"\n", cmd, data);
00338       return -1;
00339    }
00340 
00341    if (!strcasecmp(cmd, "agc"))
00342       snprintf(buf, len, "%.01f", sdi ? sdi->agclevel : 0.0);
00343    else
00344       snprintf(buf, len, "%d", sdi ? sdi->denoise : 0);
00345 
00346    return 0;
00347 }
00348 
00349 static struct ast_custom_function agc_function = {
00350    .name = "AGC",
00351    .write = speex_write,
00352    .read = speex_read,
00353    .read_max = 22,
00354 };
00355 
00356 static struct ast_custom_function denoise_function = {
00357    .name = "DENOISE",
00358    .write = speex_write,
00359    .read = speex_read,
00360    .read_max = 22,
00361 };
00362 
00363 static int unload_module(void)
00364 {
00365    ast_custom_function_unregister(&agc_function);
00366    ast_custom_function_unregister(&denoise_function);
00367    return 0;
00368 }
00369 
00370 static int load_module(void)
00371 {
00372    if (ast_custom_function_register(&agc_function)) {
00373       return AST_MODULE_LOAD_DECLINE;
00374    }
00375 
00376    if (ast_custom_function_register(&denoise_function)) {
00377       ast_custom_function_unregister(&agc_function);
00378       return AST_MODULE_LOAD_DECLINE;
00379    }
00380 
00381    return AST_MODULE_LOAD_SUCCESS;
00382 }
00383 
00384 AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Noise reduction and Automatic Gain Control (AGC)");

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