mirror of http://gerrit.asterisk.org/asterisk
and denoising to a channel, AGC() and DENOISE(). Also included, is a change to the audiohook API to add a new function (ast_audiohook_remove) that can remove an audiohook from a channel before it is detached. This code is based on a contribution from Switchvox. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114926 65c4cc65-6c06-0410-ace0-fbb531ad65f31.6.1
Brett Bryant
17 years ago
parent
c48c37909c
commit
5634048c98
@ -0,0 +1,310 @@
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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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***/
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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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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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speex_preprocess(sdi->state, frame->data, NULL);
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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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if (!(datastore = ast_channel_datastore_alloc(&speex_datastore, NULL))) {
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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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ast_channel_datastore_free(datastore);
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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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ast_channel_datastore_free(datastore);
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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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(*sdi)->denoise = ast_true(value);
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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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ast_channel_datastore_free(datastore);
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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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static struct ast_custom_function agc_function = {
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.name = "AGC",
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.synopsis = "Apply automatic gain control to audio on a channel",
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.desc =
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" The AGC function will apply automatic gain control to audio on the channel\n"
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"that this function is executed on. Use rx for audio received from the channel\n"
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"and tx to apply AGC to the audio being sent to the channel. When using this\n"
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"function, you set a target audio level. It is primarily intended for use with\n"
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"analog lines, but could be useful for other channels, as well. The target volume\n"
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"is set with a number between 1 and 32768. Larger numbers are louder.\n"
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" Example Usage:\n"
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" Set(AGC(rx)=8000)\n"
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" Set(AGC(tx)=8000)\n"
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" Set(AGC(rx)=off)\n"
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" Set(AGC(tx)=off)\n"
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"",
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.write = speex_write,
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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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.synopsis = "Apply noise reduction to audio on a channel",
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.desc =
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" The DENOISE function will apply noise reduction to audio on the channel\n"
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"that this function is executed on. It is especially useful for noisy analog\n"
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"lines, especially when adjusting gains or using AGC. Use rx for audio\n"
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"received from the channel and tx to apply the filter to the audio being sent\n"
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"to the channel.\n"
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" Example Usage:\n"
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" Set(DENOISE(rx)=on)\n"
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" Set(DENOISE(tx)=on)\n"
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" Set(DENOISE(rx)=off)\n"
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" Set(DENOISE(tx)=off)\n"
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"",
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.write = speex_write,
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};
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static int unload_module(void)
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{
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ast_custom_function_unregister(&agc_function);
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ast_custom_function_unregister(&denoise_function);
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return 0;
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}
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static int load_module(void)
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{
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if (ast_custom_function_register(&agc_function)) {
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return AST_MODULE_LOAD_DECLINE;
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}
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if (ast_custom_function_register(&denoise_function)) {
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ast_custom_function_unregister(&denoise_function);
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return AST_MODULE_LOAD_DECLINE;
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}
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return AST_MODULE_LOAD_SUCCESS;
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}
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AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Noise reduction and Automatic Gain Control (AGC)");
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