[asterisk-commits] bbryant: trunk r114926 - in /trunk: ./ funcs/ include/asterisk/ main/
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Thu May 1 11:57:19 CDT 2008
Author: bbryant
Date: Thu May 1 11:57:19 2008
New Revision: 114926
URL: http://svn.digium.com/view/asterisk?view=rev&rev=114926
Log:
Add two new dialplan functions from libspeex for applying audio gain control
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.
Added:
trunk/funcs/func_speex.c (with props)
Modified:
trunk/CHANGES
trunk/include/asterisk/audiohook.h
trunk/main/audiohook.c
Modified: trunk/CHANGES
URL: http://svn.digium.com/view/asterisk/trunk/CHANGES?view=diff&rev=114926&r1=114925&r2=114926
==============================================================================
--- trunk/CHANGES (original)
+++ trunk/CHANGES Thu May 1 11:57:19 2008
@@ -7,6 +7,9 @@
* Added a new dialplan function, AST_CONFIG(), which allows you to access
variables from an Asterisk configuration file.
* The JACK_HOOK function now has a c() option to supply a custom client name.
+ * Added two new dialplan functions from libspeex for audio gain control and
+ denoise, AGC() and DENOISE(). Both functions can be applied to the tx and
+ rx directions of a channel from the dialplan.
Zaptel channel driver (chan_zap) Changes
----------------------------------------
Added: trunk/funcs/func_speex.c
URL: http://svn.digium.com/view/asterisk/trunk/funcs/func_speex.c?view=auto&rev=114926
==============================================================================
--- trunk/funcs/func_speex.c (added)
+++ trunk/funcs/func_speex.c Thu May 1 11:57:19 2008
@@ -1,0 +1,310 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 2008, Digium, Inc.
+ *
+ * Brian Degenhardt <bmd at digium.com>
+ * Brett Bryant <bbryant at digium.com>
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*! \file
+ *
+ * \brief Noise reduction and automatic gain control (AGC)
+ *
+ * \author Brian Degenhardt <bmd at digium.com>
+ * \author Brett Bryant <bbryant at digium.com>
+ *
+ * \ingroup functions
+ *
+ * \extref The Speex library - http://www.speex.org
+ */
+
+/*** MODULEINFO
+ <depend>speex</depend>
+ ***/
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include <speex/speex_preprocess.h>
+#include "asterisk/module.h"
+#include "asterisk/channel.h"
+#include "asterisk/pbx.h"
+#include "asterisk/utils.h"
+#include "asterisk/audiohook.h"
+
+#define DEFAULT_AGC_LEVEL 8000.0
+
+struct speex_direction_info {
+ SpeexPreprocessState *state; /*!< speex preprocess state object */
+ int agc; /*!< audio gain control is enabled or not */
+ int denoise; /*!< denoise is enabled or not */
+ int samples; /*!< n of 8Khz samples in last frame */
+ float agclevel; /*!< audio gain control level [1.0 - 32768.0] */
+};
+
+struct speex_info {
+ struct ast_audiohook audiohook;
+ struct speex_direction_info *tx, *rx;
+};
+
+static void destroy_callback(void *data)
+{
+ struct speex_info *si = data;
+
+ ast_audiohook_destroy(&si->audiohook);
+
+ if (si->rx && si->rx->state) {
+ speex_preprocess_state_destroy(si->rx->state);
+ }
+
+ if (si->tx && si->tx->state) {
+ speex_preprocess_state_destroy(si->tx->state);
+ }
+
+ if (si->rx) {
+ ast_free(si->rx);
+ }
+
+ if (si->tx) {
+ ast_free(si->tx);
+ }
+
+ ast_free(data);
+};
+
+static const struct ast_datastore_info speex_datastore = {
+ .type = "speex",
+ .destroy = destroy_callback
+};
+
+static int speex_callback(struct ast_audiohook *audiohook, struct ast_channel *chan, struct ast_frame *frame, enum ast_audiohook_direction direction)
+{
+ struct ast_datastore *datastore = NULL;
+ struct speex_direction_info *sdi = NULL;
+ struct speex_info *si = NULL;
+
+ /* If the audiohook is stopping it means the channel is shutting down.... but we let the datastore destroy take care of it */
+ if (audiohook->status == AST_AUDIOHOOK_STATUS_DONE || frame->frametype != AST_FRAME_VOICE) {
+ return 0;
+ }
+
+ ast_channel_lock(chan);
+ if (!(datastore = ast_channel_datastore_find(chan, &speex_datastore, NULL))) {
+ ast_channel_unlock(chan);
+ return 0;
+ }
+ ast_channel_unlock(chan);
+
+ si = datastore->data;
+
+ sdi = (direction == AST_AUDIOHOOK_DIRECTION_READ) ? si->rx : si->tx;
+
+ if (!sdi) {
+ return 0;
+ }
+
+ if (sdi->samples != frame->samples) {
+ if (sdi->state) {
+ speex_preprocess_state_destroy(sdi->state);
+ }
+
+ if (!(sdi->state = speex_preprocess_state_init((sdi->samples = frame->samples), 8000))) {
+ return -1;
+ }
+
+ speex_preprocess_ctl(sdi->state, SPEEX_PREPROCESS_SET_AGC, &sdi->agc);
+
+ if (sdi->agc) {
+ speex_preprocess_ctl(sdi->state, SPEEX_PREPROCESS_SET_AGC_LEVEL, &sdi->agclevel);
+ }
+
+ speex_preprocess_ctl(sdi->state, SPEEX_PREPROCESS_SET_DENOISE, &sdi->denoise);
+ }
+
+ speex_preprocess(sdi->state, frame->data, NULL);
+
+ return 0;
+}
+
+static int speex_write(struct ast_channel *chan, const char *cmd, char *data, const char *value)
+{
+ struct ast_datastore *datastore = NULL;
+ struct speex_info *si = NULL;
+ struct speex_direction_info **sdi = NULL;
+ int is_new = 0;
+
+ ast_channel_lock(chan);
+ if (!(datastore = ast_channel_datastore_find(chan, &speex_datastore, NULL))) {
+ ast_channel_unlock(chan);
+
+ if (!(datastore = ast_channel_datastore_alloc(&speex_datastore, NULL))) {
+ return 0;
+ }
+
+ if (!(si = ast_calloc(1, sizeof(*si)))) {
+ ast_channel_datastore_free(datastore);
+ return 0;
+ }
+
+ ast_audiohook_init(&si->audiohook, AST_AUDIOHOOK_TYPE_MANIPULATE, "speex");
+ si->audiohook.manipulate_callback = speex_callback;
+
+ is_new = 1;
+ } else {
+ ast_channel_unlock(chan);
+ si = datastore->data;
+ }
+
+ if (!strcasecmp(data, "rx")) {
+ sdi = &si->rx;
+ } else if (!strcasecmp(data, "tx")) {
+ sdi = &si->tx;
+ } else {
+ ast_log(LOG_ERROR, "Invalid argument provided to the %s function\n", cmd);
+
+ if (is_new) {
+ ast_channel_datastore_free(datastore);
+ return -1;
+ }
+ }
+
+ if (!*sdi) {
+ if (!(*sdi = ast_calloc(1, sizeof(**sdi)))) {
+ return 0;
+ }
+ /* Right now, the audiohooks API will _only_ provide us 8 kHz slinear
+ * audio. When it supports 16 kHz (or any other sample rates, we will
+ * have to take that into account here. */
+ (*sdi)->samples = -1;
+ }
+
+ if (!strcasecmp(cmd, "agc")) {
+ if (!sscanf(value, "%f", &(*sdi)->agclevel))
+ (*sdi)->agclevel = ast_true(value) ? DEFAULT_AGC_LEVEL : 0.0;
+
+ if ((*sdi)->agclevel > 32768.0) {
+ ast_log(LOG_WARNING, "AGC(%s)=%.01f is greater than 32768... setting to 32768 instead\n",
+ ((*sdi == si->rx) ? "rx" : "tx"), (*sdi)->agclevel);
+ (*sdi)->agclevel = 32768.0;
+ }
+
+ (*sdi)->agc = !!((*sdi)->agclevel);
+
+ if ((*sdi)->state) {
+ speex_preprocess_ctl((*sdi)->state, SPEEX_PREPROCESS_SET_AGC, &(*sdi)->agc);
+ if ((*sdi)->agc) {
+ speex_preprocess_ctl((*sdi)->state, SPEEX_PREPROCESS_SET_AGC_LEVEL, &(*sdi)->agclevel);
+ }
+ }
+ } else if (!strcasecmp(cmd, "denoise")) {
+ (*sdi)->denoise = ast_true(value);
+
+ if ((*sdi)->state) {
+ speex_preprocess_ctl((*sdi)->state, SPEEX_PREPROCESS_SET_DENOISE, &(*sdi)->denoise);
+ }
+ }
+
+ if (!(*sdi)->agc && !(*sdi)->denoise) {
+ if ((*sdi)->state)
+ speex_preprocess_state_destroy((*sdi)->state);
+
+ ast_free(*sdi);
+ *sdi = NULL;
+ }
+
+ if (!si->rx && !si->tx) {
+ if (is_new) {
+ is_new = 0;
+ } else {
+ ast_channel_lock(chan);
+ ast_channel_datastore_remove(chan, datastore);
+ ast_channel_unlock(chan);
+ ast_audiohook_remove(chan, &si->audiohook);
+ ast_audiohook_detach(&si->audiohook);
+ }
+
+ ast_channel_datastore_free(datastore);
+ }
+
+ if (is_new) {
+ datastore->data = si;
+ ast_channel_lock(chan);
+ ast_channel_datastore_add(chan, datastore);
+ ast_channel_unlock(chan);
+ ast_audiohook_attach(chan, &si->audiohook);
+ }
+
+ return 0;
+}
+
+static struct ast_custom_function agc_function = {
+ .name = "AGC",
+ .synopsis = "Apply automatic gain control to audio on a channel",
+ .desc =
+ " The AGC function will apply automatic gain control to audio on the channel\n"
+ "that this function is executed on. Use rx for audio received from the channel\n"
+ "and tx to apply AGC to the audio being sent to the channel. When using this\n"
+ "function, you set a target audio level. It is primarily intended for use with\n"
+ "analog lines, but could be useful for other channels, as well. The target volume\n"
+ "is set with a number between 1 and 32768. Larger numbers are louder.\n"
+ " Example Usage:\n"
+ " Set(AGC(rx)=8000)\n"
+ " Set(AGC(tx)=8000)\n"
+ " Set(AGC(rx)=off)\n"
+ " Set(AGC(tx)=off)\n"
+ "",
+ .write = speex_write,
+};
+
+static struct ast_custom_function denoise_function = {
+ .name = "DENOISE",
+ .synopsis = "Apply noise reduction to audio on a channel",
+ .desc =
+ " The DENOISE function will apply noise reduction to audio on the channel\n"
+ "that this function is executed on. It is especially useful for noisy analog\n"
+ "lines, especially when adjusting gains or using AGC. Use rx for audio\n"
+ "received from the channel and tx to apply the filter to the audio being sent\n"
+ "to the channel.\n"
+ " Example Usage:\n"
+ " Set(DENOISE(rx)=on)\n"
+ " Set(DENOISE(tx)=on)\n"
+ " Set(DENOISE(rx)=off)\n"
+ " Set(DENOISE(tx)=off)\n"
+ "",
+ .write = speex_write,
+};
+
+static int unload_module(void)
+{
+ ast_custom_function_unregister(&agc_function);
+ ast_custom_function_unregister(&denoise_function);
+ return 0;
+}
+
+static int load_module(void)
+{
+ if (ast_custom_function_register(&agc_function)) {
+ return AST_MODULE_LOAD_DECLINE;
+ }
+
+ if (ast_custom_function_register(&denoise_function)) {
+ ast_custom_function_unregister(&denoise_function);
+ return AST_MODULE_LOAD_DECLINE;
+ }
+
+ return AST_MODULE_LOAD_SUCCESS;
+}
+
+AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Noise reduction and Automatic Gain Control (AGC)");
Propchange: trunk/funcs/func_speex.c
------------------------------------------------------------------------------
svn:eol-style = native
Propchange: trunk/funcs/func_speex.c
------------------------------------------------------------------------------
svn:keywords = Author Date Id Revision
Propchange: trunk/funcs/func_speex.c
------------------------------------------------------------------------------
svn:mime-type = text/plain
Modified: trunk/include/asterisk/audiohook.h
URL: http://svn.digium.com/view/asterisk/trunk/include/asterisk/audiohook.h?view=diff&rev=114926&r1=114925&r2=114926
==============================================================================
--- trunk/include/asterisk/audiohook.h (original)
+++ trunk/include/asterisk/audiohook.h Thu May 1 11:57:19 2008
@@ -160,6 +160,18 @@
*/
int ast_audiohook_detach_source(struct ast_channel *chan, const char *source);
+/*!
+ * \brief Remove an audiohook from a specified channel
+ *
+ * \param chan Channel to remove from
+ * \param audiohook Audiohook to remove
+ *
+ * \return Returns 0 on success, -1 on failure
+ *
+ * \note The channel does not need to be locked before calling this function
+ */
+int ast_audiohook_remove(struct ast_channel *chan, struct ast_audiohook *audiohook);
+
/*! \brief Pass a frame off to be handled by the audiohook core
* \param chan Channel that the list is coming off of
* \param audiohook_list List of audiohooks
Modified: trunk/main/audiohook.c
URL: http://svn.digium.com/view/asterisk/trunk/main/audiohook.c?view=diff&rev=114926&r1=114925&r2=114926
==============================================================================
--- trunk/main/audiohook.c (original)
+++ trunk/main/audiohook.c Thu May 1 11:57:19 2008
@@ -453,6 +453,42 @@
audiohook->status = AST_AUDIOHOOK_STATUS_SHUTDOWN;
return (audiohook ? 0 : -1);
+}
+
+/*!
+ * \brief Remove an audiohook from a specified channel
+ *
+ * \param chan Channel to remove from
+ * \param audiohook Audiohook to remove
+ *
+ * \return Returns 0 on success, -1 on failure
+ *
+ * \note The channel does not need to be locked before calling this function
+ */
+int ast_audiohook_remove(struct ast_channel *chan, struct ast_audiohook *audiohook)
+{
+ ast_channel_lock(chan);
+
+ if (!chan->audiohooks) {
+ ast_channel_unlock(chan);
+ return -1;
+ }
+
+ if (audiohook->type == AST_AUDIOHOOK_TYPE_SPY)
+ AST_LIST_REMOVE(&chan->audiohooks->spy_list, audiohook, list);
+ else if (audiohook->type == AST_AUDIOHOOK_TYPE_WHISPER)
+ AST_LIST_REMOVE(&chan->audiohooks->whisper_list, audiohook, list);
+ else if (audiohook->type == AST_AUDIOHOOK_TYPE_MANIPULATE)
+ AST_LIST_REMOVE(&chan->audiohooks->manipulate_list, audiohook, list);
+
+ ast_audiohook_lock(audiohook);
+ audiohook->status = AST_AUDIOHOOK_STATUS_DONE;
+ ast_cond_signal(&audiohook->trigger);
+ ast_audiohook_unlock(audiohook);
+
+ ast_channel_unlock(chan);
+
+ return 0;
}
/*! \brief Pass a DTMF frame off to be handled by the audiohook core
More information about the asterisk-commits
mailing list