2008-05-01 16:57:19 +00:00
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/*
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* Asterisk -- An open source telephony toolkit.
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*
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* Copyright (C) 2008, Digium, Inc.
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*
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* Brian Degenhardt <bmd@digium.com>
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* Brett Bryant <bbryant@digium.com>
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*
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* See http://www.asterisk.org for more information about
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* the Asterisk project. Please do not directly contact
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* any of the maintainers of this project for assistance;
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* the project provides a web site, mailing lists and IRC
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* channels for your use.
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*
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* This program is free software, distributed under the terms of
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* the GNU General Public License Version 2. See the LICENSE file
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* at the top of the source tree.
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*/
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/*! \file
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*
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* \brief Noise reduction and automatic gain control (AGC)
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*
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* \author Brian Degenhardt <bmd@digium.com>
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* \author Brett Bryant <bbryant@digium.com>
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*
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* \ingroup functions
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*
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* \extref The Speex library - http://www.speex.org
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*/
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/*** MODULEINFO
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<depend>speex</depend>
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2008-05-05 22:13:57 +00:00
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<depend>speex_preprocess</depend>
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<use>speexdsp</use>
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2008-05-01 16:57:19 +00:00
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***/
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#include "asterisk.h"
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ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
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#include <speex/speex_preprocess.h>
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#include "asterisk/module.h"
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#include "asterisk/channel.h"
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#include "asterisk/pbx.h"
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#include "asterisk/utils.h"
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#include "asterisk/audiohook.h"
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#define DEFAULT_AGC_LEVEL 8000.0
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2008-11-01 21:10:07 +00:00
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/*** DOCUMENTATION
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<function name="AGC" language="en_US">
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<synopsis>
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Apply automatic gain control to audio on a channel.
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</synopsis>
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<syntax>
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<parameter name="channeldirection" required="true">
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<para>This can be either <literal>rx</literal> or <literal>tx</literal></para>
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</parameter>
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</syntax>
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<description>
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<para>The AGC function will apply automatic gain control to the audio on the
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2008-11-02 02:50:33 +00:00
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channel that it is executed on. Using <literal>rx</literal> for audio received
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2008-11-01 21:10:07 +00:00
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and <literal>tx</literal> for audio transmitted to the channel. When using this
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2008-11-02 02:50:33 +00:00
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function you set a target audio level. It is primarily intended for use with
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2008-11-01 21:10:07 +00:00
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analog lines, but could be useful for other channels as well. The target volume
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is set with a number between <literal>1-32768</literal>. The larger the number
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2008-11-02 02:50:33 +00:00
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the louder (more gain) the channel will receive.</para>
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2008-11-01 21:10:07 +00:00
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<para>Examples:</para>
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<para>exten => 1,1,Set(AGC(rx)=8000)</para>
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<para>exten => 1,2,Set(AGC(tx)=off)</para>
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</description>
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</function>
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<function name="DENOISE" language="en_US">
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<synopsis>
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Apply noise reduction to audio on a channel.
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</synopsis>
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<syntax>
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<parameter name="channeldirection" required="true">
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<para>This can be either <literal>rx</literal> or <literal>tx</literal>
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the values that can be set to this are either <literal>on</literal> and
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<literal>off</literal></para>
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</parameter>
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</syntax>
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<description>
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<para>The DENOISE function will apply noise reduction to audio on the channel
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that it is executed on. It is very useful for noisy analog lines, especially
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when adjusting gains or using AGC. Use <literal>rx</literal> for audio received from the channel
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and <literal>tx</literal> to apply the filter to the audio being sent to the channel.</para>
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<para>Examples:</para>
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<para>exten => 1,1,Set(DENOISE(rx)=on)</para>
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<para>exten => 1,2,Set(DENOISE(tx)=off)</para>
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</description>
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</function>
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***/
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2008-05-01 16:57:19 +00:00
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struct speex_direction_info {
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SpeexPreprocessState *state; /*!< speex preprocess state object */
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int agc; /*!< audio gain control is enabled or not */
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int denoise; /*!< denoise is enabled or not */
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int samples; /*!< n of 8Khz samples in last frame */
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float agclevel; /*!< audio gain control level [1.0 - 32768.0] */
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};
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struct speex_info {
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struct ast_audiohook audiohook;
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struct speex_direction_info *tx, *rx;
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};
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static void destroy_callback(void *data)
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{
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struct speex_info *si = data;
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ast_audiohook_destroy(&si->audiohook);
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if (si->rx && si->rx->state) {
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speex_preprocess_state_destroy(si->rx->state);
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}
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if (si->tx && si->tx->state) {
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speex_preprocess_state_destroy(si->tx->state);
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}
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if (si->rx) {
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ast_free(si->rx);
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}
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if (si->tx) {
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ast_free(si->tx);
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}
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ast_free(data);
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};
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static const struct ast_datastore_info speex_datastore = {
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.type = "speex",
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.destroy = destroy_callback
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};
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static int speex_callback(struct ast_audiohook *audiohook, struct ast_channel *chan, struct ast_frame *frame, enum ast_audiohook_direction direction)
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{
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struct ast_datastore *datastore = NULL;
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struct speex_direction_info *sdi = NULL;
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struct speex_info *si = NULL;
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/* If the audiohook is stopping it means the channel is shutting down.... but we let the datastore destroy take care of it */
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if (audiohook->status == AST_AUDIOHOOK_STATUS_DONE || frame->frametype != AST_FRAME_VOICE) {
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return 0;
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}
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ast_channel_lock(chan);
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if (!(datastore = ast_channel_datastore_find(chan, &speex_datastore, NULL))) {
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ast_channel_unlock(chan);
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return 0;
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}
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ast_channel_unlock(chan);
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si = datastore->data;
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sdi = (direction == AST_AUDIOHOOK_DIRECTION_READ) ? si->rx : si->tx;
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if (!sdi) {
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return 0;
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}
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if (sdi->samples != frame->samples) {
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if (sdi->state) {
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speex_preprocess_state_destroy(sdi->state);
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}
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if (!(sdi->state = speex_preprocess_state_init((sdi->samples = frame->samples), 8000))) {
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return -1;
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}
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speex_preprocess_ctl(sdi->state, SPEEX_PREPROCESS_SET_AGC, &sdi->agc);
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if (sdi->agc) {
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speex_preprocess_ctl(sdi->state, SPEEX_PREPROCESS_SET_AGC_LEVEL, &sdi->agclevel);
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}
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speex_preprocess_ctl(sdi->state, SPEEX_PREPROCESS_SET_DENOISE, &sdi->denoise);
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}
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2008-05-22 17:10:53 +00:00
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speex_preprocess(sdi->state, frame->data.ptr, NULL);
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2008-05-01 16:57:19 +00:00
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return 0;
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}
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static int speex_write(struct ast_channel *chan, const char *cmd, char *data, const char *value)
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{
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struct ast_datastore *datastore = NULL;
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struct speex_info *si = NULL;
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struct speex_direction_info **sdi = NULL;
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int is_new = 0;
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ast_channel_lock(chan);
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if (!(datastore = ast_channel_datastore_find(chan, &speex_datastore, NULL))) {
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ast_channel_unlock(chan);
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2008-08-05 16:56:11 +00:00
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if (!(datastore = ast_datastore_alloc(&speex_datastore, NULL))) {
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2008-05-01 16:57:19 +00:00
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return 0;
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}
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if (!(si = ast_calloc(1, sizeof(*si)))) {
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2008-08-05 16:56:11 +00:00
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ast_datastore_free(datastore);
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2008-05-01 16:57:19 +00:00
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return 0;
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}
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ast_audiohook_init(&si->audiohook, AST_AUDIOHOOK_TYPE_MANIPULATE, "speex");
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si->audiohook.manipulate_callback = speex_callback;
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is_new = 1;
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} else {
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ast_channel_unlock(chan);
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si = datastore->data;
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}
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if (!strcasecmp(data, "rx")) {
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sdi = &si->rx;
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} else if (!strcasecmp(data, "tx")) {
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sdi = &si->tx;
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} else {
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ast_log(LOG_ERROR, "Invalid argument provided to the %s function\n", cmd);
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if (is_new) {
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2008-08-05 16:56:11 +00:00
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ast_datastore_free(datastore);
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2008-05-01 16:57:19 +00:00
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return -1;
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}
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}
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if (!*sdi) {
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if (!(*sdi = ast_calloc(1, sizeof(**sdi)))) {
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return 0;
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}
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/* Right now, the audiohooks API will _only_ provide us 8 kHz slinear
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* audio. When it supports 16 kHz (or any other sample rates, we will
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* have to take that into account here. */
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(*sdi)->samples = -1;
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}
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if (!strcasecmp(cmd, "agc")) {
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if (!sscanf(value, "%f", &(*sdi)->agclevel))
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(*sdi)->agclevel = ast_true(value) ? DEFAULT_AGC_LEVEL : 0.0;
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if ((*sdi)->agclevel > 32768.0) {
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ast_log(LOG_WARNING, "AGC(%s)=%.01f is greater than 32768... setting to 32768 instead\n",
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((*sdi == si->rx) ? "rx" : "tx"), (*sdi)->agclevel);
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(*sdi)->agclevel = 32768.0;
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}
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(*sdi)->agc = !!((*sdi)->agclevel);
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if ((*sdi)->state) {
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speex_preprocess_ctl((*sdi)->state, SPEEX_PREPROCESS_SET_AGC, &(*sdi)->agc);
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if ((*sdi)->agc) {
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speex_preprocess_ctl((*sdi)->state, SPEEX_PREPROCESS_SET_AGC_LEVEL, &(*sdi)->agclevel);
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}
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}
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} else if (!strcasecmp(cmd, "denoise")) {
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2008-05-01 18:28:38 +00:00
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(*sdi)->denoise = (ast_true(value) != 0);
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2008-05-01 16:57:19 +00:00
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if ((*sdi)->state) {
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speex_preprocess_ctl((*sdi)->state, SPEEX_PREPROCESS_SET_DENOISE, &(*sdi)->denoise);
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}
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}
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if (!(*sdi)->agc && !(*sdi)->denoise) {
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if ((*sdi)->state)
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speex_preprocess_state_destroy((*sdi)->state);
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ast_free(*sdi);
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*sdi = NULL;
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}
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if (!si->rx && !si->tx) {
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if (is_new) {
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is_new = 0;
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} else {
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ast_channel_lock(chan);
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ast_channel_datastore_remove(chan, datastore);
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ast_channel_unlock(chan);
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ast_audiohook_remove(chan, &si->audiohook);
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ast_audiohook_detach(&si->audiohook);
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}
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2008-08-05 16:56:11 +00:00
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ast_datastore_free(datastore);
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2008-05-01 16:57:19 +00:00
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}
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if (is_new) {
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datastore->data = si;
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ast_channel_lock(chan);
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ast_channel_datastore_add(chan, datastore);
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ast_channel_unlock(chan);
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ast_audiohook_attach(chan, &si->audiohook);
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}
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return 0;
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}
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2008-05-01 18:28:38 +00:00
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static int speex_read(struct ast_channel *chan, const char *cmd, char *data, char *buf, size_t len)
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{
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struct ast_datastore *datastore = NULL;
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struct speex_info *si = NULL;
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struct speex_direction_info *sdi = NULL;
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if (!chan) {
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ast_log(LOG_ERROR, "%s cannot be used without a channel!\n", cmd);
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return -1;
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}
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ast_channel_lock(chan);
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if (!(datastore = ast_channel_datastore_find(chan, &speex_datastore, NULL))) {
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ast_channel_unlock(chan);
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return -1;
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}
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ast_channel_unlock(chan);
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si = datastore->data;
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if (!strcasecmp(data, "tx"))
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sdi = si->tx;
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else if (!strcasecmp(data, "rx"))
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sdi = si->rx;
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else {
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ast_log(LOG_ERROR, "%s(%s) must either \"tx\" or \"rx\"\n", cmd, data);
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return -1;
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}
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if (!strcasecmp(cmd, "agc"))
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snprintf(buf, len, "%.01f", sdi ? sdi->agclevel : 0.0);
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else
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snprintf(buf, len, "%d", sdi ? sdi->denoise : 0);
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return 0;
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}
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2008-05-01 16:57:19 +00:00
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static struct ast_custom_function agc_function = {
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.name = "AGC",
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.write = speex_write,
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2008-05-01 18:28:38 +00:00
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.read = speex_read
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2008-05-01 16:57:19 +00:00
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};
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static struct ast_custom_function denoise_function = {
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.name = "DENOISE",
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.write = speex_write,
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2008-05-01 18:28:38 +00:00
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.read = speex_read
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2008-05-01 16:57:19 +00:00
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};
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static int unload_module(void)
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{
|
|
|
|
ast_custom_function_unregister(&agc_function);
|
|
|
|
ast_custom_function_unregister(&denoise_function);
|
|
|
|
return 0;
|
|
|
|
}
|
|
|
|
|
|
|
|
static int load_module(void)
|
|
|
|
{
|
2008-05-13 17:42:17 +00:00
|
|
|
if (ast_custom_function_register(&agc_function)) {
|
|
|
|
return AST_MODULE_LOAD_DECLINE;
|
|
|
|
}
|
2008-05-01 16:57:19 +00:00
|
|
|
|
2008-05-13 17:42:17 +00:00
|
|
|
if (ast_custom_function_register(&denoise_function)) {
|
|
|
|
ast_custom_function_unregister(&agc_function);
|
|
|
|
return AST_MODULE_LOAD_DECLINE;
|
|
|
|
}
|
2008-05-01 16:57:19 +00:00
|
|
|
|
2008-05-13 17:42:17 +00:00
|
|
|
return AST_MODULE_LOAD_SUCCESS;
|
2008-05-01 16:57:19 +00:00
|
|
|
}
|
|
|
|
|
|
|
|
AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Noise reduction and Automatic Gain Control (AGC)");
|