diff options
author | bbryant <bbryant@f38db490-d61c-443f-a65b-d21fe96a405b> | 2008-05-01 16:57:19 +0000 |
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committer | bbryant <bbryant@f38db490-d61c-443f-a65b-d21fe96a405b> | 2008-05-01 16:57:19 +0000 |
commit | 26a549ebfbb5c4014e3d1c8f421eb8f05b5989b0 (patch) | |
tree | 27c35a3350c9aae81be9a3d569d1551d4c0e4a59 | |
parent | 46a00af5ab6a1c7ae5b96d708df7076b51c9c2ba (diff) |
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.
This code is based on a contribution from Switchvox.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@114926 f38db490-d61c-443f-a65b-d21fe96a405b
-rw-r--r-- | CHANGES | 3 | ||||
-rw-r--r-- | funcs/func_speex.c | 310 | ||||
-rw-r--r-- | include/asterisk/audiohook.h | 12 | ||||
-rw-r--r-- | main/audiohook.c | 36 |
4 files changed, 361 insertions, 0 deletions
@@ -7,6 +7,9 @@ Dialplan Functions * 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 ---------------------------------------- diff --git a/funcs/func_speex.c b/funcs/func_speex.c new file mode 100644 index 000000000..33282a1f5 --- /dev/null +++ b/funcs/func_speex.c @@ -0,0 +1,310 @@ +/* + * Asterisk -- An open source telephony toolkit. + * + * Copyright (C) 2008, Digium, Inc. + * + * Brian Degenhardt <bmd@digium.com> + * Brett Bryant <bbryant@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@digium.com> + * \author Brett Bryant <bbryant@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)"); diff --git a/include/asterisk/audiohook.h b/include/asterisk/audiohook.h index 4ebd19e5d..3345c5db7 100644 --- a/include/asterisk/audiohook.h +++ b/include/asterisk/audiohook.h @@ -160,6 +160,18 @@ int ast_audiohook_detach_list(struct ast_audiohook_list *audiohook_list); */ 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 diff --git a/main/audiohook.c b/main/audiohook.c index 1f5bcff4f..37970174e 100644 --- a/main/audiohook.c +++ b/main/audiohook.c @@ -455,6 +455,42 @@ int ast_audiohook_detach_source(struct ast_channel *chan, const char *source) 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 * \param chan Channel that the list is coming off of * \param audiohook_list List of audiohooks |