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

1 files changed, 310 insertions, 0 deletions

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)");