Merge audiohooks branch into trunk. This is a new API for developers to listen and manipulate the audio going through a channel.

git-svn-id: http://svn.digium.com/svn/asterisk/trunk@78649 f38db490-d61c-443f-a65b-d21fe96a405b

11 files changed, 1117 insertions, 917 deletions

diff --git a/apps/app_chanspy.c b/apps/app_chanspy.c
index edfd5290d..bb1510bd2 100644
--- a/apps/app_chanspy.c
+++ b/apps/app_chanspy.c
@@ -40,7 +40,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 #include "asterisk/file.h"
 #include "asterisk/logger.h"
 #include "asterisk/channel.h"
-#include "asterisk/chanspy.h"
+#include "asterisk/audiohook.h"
 #include "asterisk/features.h"
 #include "asterisk/options.h"
 #include "asterisk/app.h"
@@ -166,7 +166,8 @@ AST_APP_OPTIONS(spy_opts, {
 
 struct chanspy_translation_helper {
 	/* spy data */
-	struct ast_channel_spy spy;
+	struct ast_audiohook spy_audiohook;
+	struct ast_audiohook whisper_audiohook;
 	int fd;
 	int volfactor;
 };
@@ -185,15 +186,18 @@ static void spy_release(struct ast_channel *chan, void *data)
 static int spy_generate(struct ast_channel *chan, void *data, int len, int samples) 
 {
 	struct chanspy_translation_helper *csth = data;
-	struct ast_frame *f;
+	struct ast_frame *f = NULL;
 		
-	if (csth->spy.status != CHANSPY_RUNNING)
+	ast_audiohook_lock(&csth->spy_audiohook);
+	if (csth->spy_audiohook.status != AST_AUDIOHOOK_STATUS_RUNNING) {
 		/* Channel is already gone more than likely */
+		ast_audiohook_unlock(&csth->spy_audiohook);
 		return -1;
+	}
 
-	ast_mutex_lock(&csth->spy.lock);
-	f = ast_channel_spy_read_frame(&csth->spy, samples);
-	ast_mutex_unlock(&csth->spy.lock);
+	f = ast_audiohook_read_frame(&csth->spy_audiohook, samples, AST_AUDIOHOOK_DIRECTION_BOTH, AST_FORMAT_SLINEAR);
+
+	ast_audiohook_unlock(&csth->spy_audiohook);
 		
 	if (!f)
 		return 0;
@@ -217,16 +221,14 @@ static struct ast_generator spygen = {
 	.generate = spy_generate, 
 };
 
-static int start_spying(struct ast_channel *chan, struct ast_channel *spychan, struct ast_channel_spy *spy) 
+static int start_spying(struct ast_channel *chan, struct ast_channel *spychan, struct ast_audiohook *audiohook) 
 {
-	int res;
-	struct ast_channel *peer;
+	int res = 0;
+	struct ast_channel *peer = NULL;
 
 	ast_log(LOG_NOTICE, "Attaching %s to %s\n", spychan->name, chan->name);
 
-	ast_channel_lock(chan);
-	res = ast_channel_spy_add(chan, spy);
-	ast_channel_unlock(chan);
+	res = ast_audiohook_attach(chan, audiohook);
 
 	if (!res && ast_test_flag(chan, AST_FLAG_NBRIDGE) && (peer = ast_bridged_channel(chan)))
 		ast_softhangup(peer, AST_SOFTHANGUP_UNBRIDGE);	
@@ -234,35 +236,6 @@ static int start_spying(struct ast_channel *chan, struct ast_channel *spychan, s
 	return res;
 }
 
-/* Map 'volume' levels from -4 through +4 into
-   decibel (dB) settings for channel drivers
-*/
-static signed char volfactor_map[] = {
-	-24,
-	-18,
-	-12,
-	-6,
-	0,
-	6,
-	12,
-	18,
-	24,
-};
-
-/* attempt to set the desired gain adjustment via the channel driver;
-   if successful, clear it out of the csth structure so the
-   generator will not attempt to do the adjustment itself
-*/
-static void set_volume(struct ast_channel *chan, struct chanspy_translation_helper *csth)
-{
-	signed char volume_adjust = volfactor_map[csth->volfactor + 4];
-
-	if (!ast_channel_setoption(chan, AST_OPTION_TXGAIN, &volume_adjust, sizeof(volume_adjust), 0))
-		csth->volfactor = 0;
-	csth->spy.read_vol_adjustment = csth->volfactor;
-	csth->spy.write_vol_adjustment = csth->volfactor;
-}
-
 static int channel_spy(struct ast_channel *chan, struct ast_channel *spyee, int *volfactor, int fd,
 		       const struct ast_flags *flags, char *exitcontext) 
 {
@@ -280,49 +253,17 @@ static int channel_spy(struct ast_channel *chan, struct ast_channel *spyee, int
 	ast_verb(2, "Spying on channel %s\n", name);
 
 	memset(&csth, 0, sizeof(csth));
-	ast_set_flag(&csth.spy, CHANSPY_FORMAT_AUDIO);
-	ast_set_flag(&csth.spy, CHANSPY_TRIGGER_NONE);
-	if (!ast_test_flag(flags, OPTION_READONLY))
-		ast_set_flag(&csth.spy, CHANSPY_MIXAUDIO);
-	csth.spy.type = "ChanSpy";
-	csth.spy.status = CHANSPY_RUNNING;
-	csth.spy.read_queue.format = AST_FORMAT_SLINEAR;
-	csth.spy.write_queue.format = AST_FORMAT_SLINEAR;
-	ast_mutex_init(&csth.spy.lock);
-	csth.volfactor = *volfactor;
-	set_volume(chan, &csth);
-	if (csth.volfactor) {
-		ast_set_flag(&csth.spy, CHANSPY_READ_VOLADJUST);
-		csth.spy.read_vol_adjustment = csth.volfactor;
-		ast_set_flag(&csth.spy, CHANSPY_WRITE_VOLADJUST);
-		csth.spy.write_vol_adjustment = csth.volfactor;
-	}
-	csth.fd = fd;
+
+	ast_audiohook_init(&csth.spy_audiohook, AST_AUDIOHOOK_TYPE_SPY, "ChanSpy");
 	
-	if (start_spying(spyee, chan, &csth.spy)) {
-		ast_mutex_destroy(&csth.spy.lock);
+	if (start_spying(spyee, chan, &csth.spy_audiohook)) {
+		ast_audiohook_destroy(&csth.spy_audiohook);
 		return 0;
 	}
 
 	if (ast_test_flag(flags, OPTION_WHISPER)) {
-		struct ast_filestream *beepstream;
-		int old_write_format = 0;
-
-		ast_channel_whisper_start(csth.spy.chan);
-		old_write_format = chan->writeformat;
-		if ((beepstream = ast_openstream_full(chan, "beep", chan->language, 1))) {
-			struct ast_frame *f;
-
-			while ((f = ast_readframe(beepstream))) {
-				ast_channel_whisper_feed(csth.spy.chan, f);
-				ast_frfree(f);
-			}
-
-			ast_closestream(beepstream);
-			chan->stream = NULL;
-		}
-		if (old_write_format)
-			ast_set_write_format(chan, old_write_format);
+		ast_audiohook_init(&csth.whisper_audiohook, AST_AUDIOHOOK_TYPE_WHISPER, "ChanSpy");
+		start_spying(spyee, chan, &csth.whisper_audiohook);
 	}
 
 	if (ast_test_flag(flags, OPTION_PRIVATE))
@@ -344,21 +285,20 @@ static int channel_spy(struct ast_channel *chan, struct ast_channel *spyee, int
 	   has arrived, since the spied-on channel could have gone away while
 	   we were waiting
 	*/
-	while ((res = ast_waitfor(chan, -1) > -1) &&
-	       csth.spy.status == CHANSPY_RUNNING &&
-	       csth.spy.chan) {
+	while ((res = ast_waitfor(chan, -1) > -1) && csth.spy_audiohook.status == AST_AUDIOHOOK_STATUS_RUNNING) {
 		if (!(f = ast_read(chan)) || ast_check_hangup(chan)) {
 			running = -1;
 			break;
 		}
 
-		if (ast_test_flag(flags, OPTION_WHISPER) &&
-		    (f->frametype == AST_FRAME_VOICE)) {
-			ast_channel_whisper_feed(csth.spy.chan, f);
+		if (ast_test_flag(flags, OPTION_WHISPER) && f->frametype == AST_FRAME_VOICE) {
+			ast_audiohook_lock(&csth.whisper_audiohook);
+			ast_audiohook_write_frame(&csth.whisper_audiohook, AST_AUDIOHOOK_DIRECTION_WRITE, f);
+			ast_audiohook_unlock(&csth.whisper_audiohook);
 			ast_frfree(f);
 			continue;
 		}
-
+		
 		res = (f->frametype == AST_FRAME_DTMF) ? f->subclass : 0;
 		ast_frfree(f);
 		if (!res)
@@ -401,37 +341,25 @@ static int channel_spy(struct ast_channel *chan, struct ast_channel *spyee, int
 			if (*volfactor > 4)
 				*volfactor = -4;
 			ast_verb(3, "Setting spy volume on %s to %d\n", chan->name, *volfactor);
-			csth.volfactor = *volfactor;
-			set_volume(chan, &csth);
-			if (csth.volfactor) {
-				ast_set_flag(&csth.spy, CHANSPY_READ_VOLADJUST);
-				csth.spy.read_vol_adjustment = csth.volfactor;
-				ast_set_flag(&csth.spy, CHANSPY_WRITE_VOLADJUST);
-				csth.spy.write_vol_adjustment = csth.volfactor;
-			} else {
-				ast_clear_flag(&csth.spy, CHANSPY_READ_VOLADJUST);
-				ast_clear_flag(&csth.spy, CHANSPY_WRITE_VOLADJUST);
-			}
 		}
 	}
 
-	if (ast_test_flag(flags, OPTION_WHISPER) && csth.spy.chan)
-		ast_channel_whisper_stop(csth.spy.chan);
-
 	if (ast_test_flag(flags, OPTION_PRIVATE))
 		ast_channel_stop_silence_generator(chan, silgen);
 	else
 		ast_deactivate_generator(chan);
 
-	csth.spy.status = CHANSPY_DONE;
-
-	/* If a channel still exists on our spy structure then we need to remove ourselves */
-	if (csth.spy.chan) {
-		ast_channel_lock(csth.spy.chan);
-		ast_channel_spy_remove(csth.spy.chan, &csth.spy);
-		ast_channel_unlock(csth.spy.chan);
+	if (ast_test_flag(flags, OPTION_WHISPER)) {
+		ast_audiohook_lock(&csth.whisper_audiohook);
+		ast_audiohook_detach(&csth.whisper_audiohook);
+		ast_audiohook_unlock(&csth.whisper_audiohook);
+		ast_audiohook_destroy(&csth.whisper_audiohook);
 	}
-	ast_channel_spy_free(&csth.spy);
+
+	ast_audiohook_lock(&csth.spy_audiohook);
+	ast_audiohook_detach(&csth.spy_audiohook);
+	ast_audiohook_unlock(&csth.spy_audiohook);
+	ast_audiohook_destroy(&csth.spy_audiohook);
 	
 	if (option_verbose >= 2)
 		ast_verbose(VERBOSE_PREFIX_2 "Done Spying on channel %s\n", name);
diff --git a/apps/app_mixmonitor.c b/apps/app_mixmonitor.c
index cd532b321..614adcba1 100644
--- a/apps/app_mixmonitor.c
+++ b/apps/app_mixmonitor.c
@@ -45,7 +45,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 #include "asterisk/file.h"
 #include "asterisk/logger.h"
 #include "asterisk/channel.h"
-#include "asterisk/chanspy.h"
+#include "asterisk/audiohook.h"
 #include "asterisk/pbx.h"
 #include "asterisk/module.h"
 #include "asterisk/lock.h"
@@ -93,7 +93,7 @@ struct module_symbols *me;
 static const char *mixmonitor_spy_type = "MixMonitor";
 
 struct mixmonitor {
-	struct ast_channel_spy spy;
+	struct ast_audiohook audiohook;
 	char *filename;
 	char *post_process;
 	char *name;
@@ -123,17 +123,15 @@ AST_APP_OPTIONS(mixmonitor_opts, {
 	AST_APP_OPTION_ARG('W', MUXFLAG_VOLUME, OPT_ARG_VOLUME),
 });
 
-static int startmon(struct ast_channel *chan, struct ast_channel_spy *spy) 
+static int startmon(struct ast_channel *chan, struct ast_audiohook *audiohook) 
 {
-	struct ast_channel *peer;
-	int res;
+	struct ast_channel *peer = NULL;
+	int res = 0;
 
 	if (!chan)
 		return -1;
 
-	ast_channel_lock(chan);
-	res = ast_channel_spy_add(chan, spy);
-	ast_channel_unlock(chan);
+	ast_audiohook_attach(chan, audiohook);
 
 	if (!res && ast_test_flag(chan, AST_FLAG_NBRIDGE) && (peer = ast_bridged_channel(chan)))
 		ast_softhangup(peer, AST_SOFTHANGUP_UNBRIDGE);	
@@ -146,7 +144,6 @@ static int startmon(struct ast_channel *chan, struct ast_channel_spy *spy)
 static void *mixmonitor_thread(void *obj) 
 {
 	struct mixmonitor *mixmonitor = obj;
-	struct ast_frame *f = NULL;
 	struct ast_filestream *fs = NULL;
 	unsigned int oflags;
 	char *ext;
@@ -155,59 +152,48 @@ static void *mixmonitor_thread(void *obj)
 	if (option_verbose > 1)
 		ast_verbose(VERBOSE_PREFIX_2 "Begin MixMonitor Recording %s\n", mixmonitor->name);
 	
-	ast_mutex_lock(&mixmonitor->spy.lock);
+	ast_audiohook_lock(&mixmonitor->audiohook);
 
-	while (mixmonitor->spy.chan) {
-		struct ast_frame *next;
-		int write;
+	while (mixmonitor->audiohook.status == AST_AUDIOHOOK_STATUS_RUNNING) {
+		struct ast_frame *fr = NULL;
 
-		ast_channel_spy_trigger_wait(&mixmonitor->spy);
-		
-		if (!mixmonitor->spy.chan || mixmonitor->spy.status != CHANSPY_RUNNING)
+		ast_audiohook_trigger_wait(&mixmonitor->audiohook);
+
+		if (mixmonitor->audiohook.status != AST_AUDIOHOOK_STATUS_RUNNING)
 			break;
-		
-		while (1) {
-			if (!(f = ast_channel_spy_read_frame(&mixmonitor->spy, SAMPLES_PER_FRAME)))
-				break;
-
-			write = (!ast_test_flag(mixmonitor, MUXFLAG_BRIDGED) ||
-				 ast_bridged_channel(mixmonitor->spy.chan));
-
-			/* it is possible for ast_channel_spy_read_frame() to return a chain
-			   of frames if a queue flush was necessary, so process them
-			*/
-			for (; f; f = next) {
-				next = AST_LIST_NEXT(f, frame_list);
-				if (write && errflag == 0) {
-					if (!fs) {
-						/* Determine creation flags and filename plus extension for filestream */
-						oflags = O_CREAT | O_WRONLY;
-						oflags |= ast_test_flag(mixmonitor, MUXFLAG_APPEND) ? O_APPEND : O_TRUNC;
-
-						if ((ext = strrchr(mixmonitor->filename, '.')))
-							*(ext++) = '\0';
-						else
-							ext = "raw";
-
-						/* Move onto actually creating the filestream */
-						if (!(fs = ast_writefile(mixmonitor->filename, ext, NULL, oflags, 0, 0644))) {
-							ast_log(LOG_ERROR, "Cannot open %s.%s\n", mixmonitor->filename, ext);
-							errflag = 1;
-						}
-
-					}
-					if (fs)
-						ast_writestream(fs, f);
-				}
-				ast_frame_free(f, 0);
+
+		if (!(fr = ast_audiohook_read_frame(&mixmonitor->audiohook, SAMPLES_PER_FRAME, AST_AUDIOHOOK_DIRECTION_BOTH, AST_FORMAT_SLINEAR)))
+			continue;
+
+		/* Initialize the file if not already done so */
+		if (!fs && !errflag) {
+			oflags = O_CREAT | O_WRONLY;
+			oflags |= ast_test_flag(mixmonitor, MUXFLAG_APPEND) ? O_APPEND : O_TRUNC;
+
+			if ((ext = strrchr(mixmonitor->filename, '.')))
+				*(ext++) = '\0';
+			else
+				ext = "raw";
+
+			if (!(fs = ast_writefile(mixmonitor->filename, ext, NULL, oflags, 0, 0644))) {
+				ast_log(LOG_ERROR, "Cannot open %s.%s\n", mixmonitor->filename, ext);
+				errflag = 1;
 			}
 		}
+
+		/* Write out frame */
+		if (fs)
+			ast_writestream(fs, fr);
+
+		/* All done! free it. */
+		ast_frame_free(fr, 0);
+
 	}
 
-	ast_mutex_unlock(&mixmonitor->spy.lock);
+	ast_audiohook_detach(&mixmonitor->audiohook);
+	ast_audiohook_unlock(&mixmonitor->audiohook);
+	ast_audiohook_destroy(&mixmonitor->audiohook);
 
-	ast_channel_spy_free(&mixmonitor->spy);
-	
 	if (option_verbose > 1)
 		ast_verbose(VERBOSE_PREFIX_2 "End MixMonitor Recording %s\n", mixmonitor->name);
 
@@ -270,27 +256,17 @@ static void launch_monitor_thread(struct ast_channel *chan, const char *filename
 	strcpy(mixmonitor->filename, filename);
 
 	/* Setup the actual spy before creating our thread */
-	ast_set_flag(&mixmonitor->spy, CHANSPY_FORMAT_AUDIO);
-	ast_set_flag(&mixmonitor->spy, CHANSPY_MIXAUDIO);
-	mixmonitor->spy.type = mixmonitor_spy_type;
-	mixmonitor->spy.status = CHANSPY_RUNNING;
-	mixmonitor->spy.read_queue.format = AST_FORMAT_SLINEAR;
-	mixmonitor->spy.write_queue.format = AST_FORMAT_SLINEAR;
-	if (readvol) {
-		ast_set_flag(&mixmonitor->spy, CHANSPY_READ_VOLADJUST);
-		mixmonitor->spy.read_vol_adjustment = readvol;
-	}
-	if (writevol) {
-		ast_set_flag(&mixmonitor->spy, CHANSPY_WRITE_VOLADJUST);
-		mixmonitor->spy.write_vol_adjustment = writevol;
+	if (ast_audiohook_init(&mixmonitor->audiohook, AST_AUDIOHOOK_TYPE_SPY, mixmonitor_spy_type)) {
+		free(mixmonitor);
+		return;
 	}
-	ast_mutex_init(&mixmonitor->spy.lock);
 
-	if (startmon(chan, &mixmonitor->spy)) {
+	ast_set_flag(&mixmonitor->audiohook, AST_AUDIOHOOK_TRIGGER_WRITE);
+
+	if (startmon(chan, &mixmonitor->audiohook)) {
 		ast_log(LOG_WARNING, "Unable to add '%s' spy to channel '%s'\n",
-			mixmonitor->spy.type, chan->name);
-		/* Since we couldn't add ourselves - bail out! */
-		ast_mutex_destroy(&mixmonitor->spy.lock);
+			mixmonitor_spy_type, chan->name);
+		ast_audiohook_destroy(&mixmonitor->audiohook);
 		ast_free(mixmonitor);
 		return;
 	}
@@ -382,9 +358,7 @@ static int mixmonitor_exec(struct ast_channel *chan, void *data)
 
 static int stop_mixmonitor_exec(struct ast_channel *chan, void *data)
 {
-	ast_channel_lock(chan);
-	ast_channel_spy_stop_by_type(chan, mixmonitor_spy_type);
-	ast_channel_unlock(chan);
+	ast_audiohook_detach_source(chan, mixmonitor_spy_type);
 	return 0;
 }
 
@@ -400,12 +374,13 @@ static int mixmonitor_cli(int fd, int argc, char **argv)
 		return RESULT_SUCCESS;
 	}
 
-	if (!strcasecmp(argv[1], "start"))
+	if (!strcasecmp(argv[1], "start")) {
 		mixmonitor_exec(chan, argv[3]);
-	else if (!strcasecmp(argv[1], "stop"))
-		ast_channel_spy_stop_by_type(chan, mixmonitor_spy_type);
-
-	ast_channel_unlock(chan);
+		ast_channel_unlock(chan);
+	} else {
+		ast_channel_unlock(chan);
+		ast_audiohook_detach_source(chan, mixmonitor_spy_type);
+	}
 
 	return RESULT_SUCCESS;
 }
diff --git a/funcs/func_volume.c b/funcs/func_volume.c
new file mode 100644
index 000000000..79fc13f30
--- /dev/null
+++ b/funcs/func_volume.c
@@ -0,0 +1,163 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 2007, Digium, Inc.
+ *
+ * Joshua Colp <jcolp@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 Technology independent volume control
+ *
+ * \author Joshua Colp <jcolp@digium.com> 
+ *
+ * \ingroup functions
+ *
+ */
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include <stdlib.h>
+
+#include "asterisk/module.h"
+#include "asterisk/channel.h"
+#include "asterisk/pbx.h"
+#include "asterisk/utils.h"
+#include "asterisk/linkedlists.h"
+#include "asterisk/audiohook.h"
+
+struct volume_information {
+	struct ast_audiohook audiohook;
+	int tx_gain;
+	int rx_gain;
+};
+
+static void destroy_callback(void *data)
+{
+	struct volume_information *vi = data;
+
+	/* Destroy the audiohook, and destroy ourselves */
+	ast_audiohook_destroy(&vi->audiohook);
+	free(vi);
+
+	return;
+}
+
+/*! \brief Static structure for datastore information */
+static const struct ast_datastore_info volume_datastore = {
+        .type = "volume",
+        .destroy = destroy_callback
+};
+
+static int volume_callback(struct ast_audiohook *audiohook, struct ast_channel *chan, struct ast_frame *frame, enum ast_audiohook_direction direction)
+{
+	struct ast_datastore *datastore = NULL;
+	struct volume_information *vi = NULL;
+	int *gain = 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)
+		return 0;
+
+	/* Grab datastore which contains our gain information */
+	if (!(datastore = ast_channel_datastore_find(chan, &volume_datastore, NULL)))
+		return 0;
+
+	vi = datastore->data;
+
+	/* If this is DTMF then allow them to increase/decrease the gains */
+	if (frame->frametype == AST_FRAME_DTMF) {
+		/* Only use DTMF coming from the source... not going to it */
+		if (direction != AST_AUDIOHOOK_DIRECTION_READ)
+			return 0;
+		if (frame->subclass == '*') {
+			vi->tx_gain += 1;
+			vi->rx_gain += 1;
+		} else if (frame->subclass == '#') {
+			vi->tx_gain -= 1;
+			vi->rx_gain -= 1;
+		}
+	} else if (frame->frametype == AST_FRAME_VOICE) {
+		/* Based on direction of frame grab the gain, and confirm it is applicable */
+		if (!(gain = (direction == AST_AUDIOHOOK_DIRECTION_READ) ? &vi->rx_gain : &vi->tx_gain) || !*gain)
+			return 0;
+		/* Apply gain to frame... easy as pi */
+		ast_frame_adjust_volume(frame, *gain);
+	}
+
+	return 0;
+}
+
+static int volume_write(struct ast_channel *chan, const char *cmd, char *data, const char *value)
+{
+	struct ast_datastore *datastore = NULL;
+	struct volume_information *vi = NULL;
+	int is_new = 0;
+
+	if (!(datastore = ast_channel_datastore_find(chan, &volume_datastore, NULL))) {
+		/* Allocate a new datastore to hold the reference to this volume and audiohook information */
+		if (!(datastore = ast_channel_datastore_alloc(&volume_datastore, NULL)))
+			return 0;
+		if (!(vi = ast_calloc(1, sizeof(*vi)))) {
+			ast_channel_datastore_free(datastore);
+			return 0;
+		}
+		ast_audiohook_init(&vi->audiohook, AST_AUDIOHOOK_TYPE_MANIPULATE, "Volume");
+		vi->audiohook.manipulate_callback = volume_callback;
+		ast_set_flag(&vi->audiohook, AST_AUDIOHOOK_WANTS_DTMF);
+		is_new = 1;
+	} else {
+		vi = datastore->data;
+	}
+
+	/* Adjust gain on volume information structure */
+	if (!strcasecmp(data, "tx"))
+		vi->tx_gain = atoi(value);
+	else if (!strcasecmp(data, "rx"))
+		vi->rx_gain = atoi(value);
+
+	if (is_new) {
+		datastore->data = vi;
+		ast_channel_datastore_add(chan, datastore);
+		ast_audiohook_attach(chan, &vi->audiohook);
+	}
+
+	return 0;
+}
+
+static struct ast_custom_function volume_function = {
+	.name = "VOLUME",
+	.synopsis = "Set the TX or RX volume of a channel",
+	.syntax = "VOLUME(TX|RX)",
+	.desc =
+	"  The VOLUME function can be used to increase or decrease the tx or\n"
+	"rx gain of any channel.  For example:\n"
+	"  Set(VOLUME(TX)=3)\n"
+	"  Set(VOLUME(RX)=2)\n",
+	.write = volume_write,
+};
+
+static int unload_module(void)
+{
+	return ast_custom_function_unregister(&volume_function);
+}
+
+static int load_module(void)
+{
+	return ast_custom_function_register(&volume_function);
+}
+
+AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Technology independent volume control");
diff --git a/include/asterisk/audiohook.h b/include/asterisk/audiohook.h
new file mode 100644
index 000000000..a374a630a
--- /dev/null
+++ b/include/asterisk/audiohook.h
@@ -0,0 +1,185 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 1999 - 2007, Digium, Inc.
+ *
+ * Joshua Colp <jcolp@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 Audiohooks Architecture
+ */
+
+#ifndef _ASTERISK_AUDIOHOOK_H
+#define _ASTERISK_AUDIOHOOK_H
+
+#if defined(__cplusplus) || defined(c_plusplus)
+extern "C" {
+#endif
+
+#include "asterisk/slinfactory.h"
+
+enum ast_audiohook_type {
+	AST_AUDIOHOOK_TYPE_SPY = 0,    /*!< Audiohook wants to receive audio */
+	AST_AUDIOHOOK_TYPE_WHISPER,    /*!< Audiohook wants to provide audio to be mixed with existing audio */
+	AST_AUDIOHOOK_TYPE_MANIPULATE, /*!< Audiohook wants to manipulate the audio */
+};
+
+enum ast_audiohook_status {
+	AST_AUDIOHOOK_STATUS_NEW = 0,  /*!< Audiohook was just created, not in use yet */
+	AST_AUDIOHOOK_STATUS_RUNNING,  /*!< Audiohook is running on a channel */
+	AST_AUDIOHOOK_STATUS_SHUTDOWN, /*!< Audiohook is being shutdown */
+	AST_AUDIOHOOK_STATUS_DONE,     /*!< Audiohook has shutdown and is not running on a channel any longer */
+};
+
+enum ast_audiohook_direction {
+	AST_AUDIOHOOK_DIRECTION_READ = 0, /*!< Reading audio in */
+	AST_AUDIOHOOK_DIRECTION_WRITE,    /*!< Writing audio out */
+	AST_AUDIOHOOK_DIRECTION_BOTH,     /*!< Both reading audio in and writing audio out */
+};
+
+enum ast_audiohook_flags {
+	AST_AUDIOHOOK_TRIGGER_MODE = (3 << 0),  /*!< When audiohook should be triggered to do something */
+	AST_AUDIOHOOK_TRIGGER_READ = (1 << 0),  /*!< Audiohook wants to be triggered when reading audio in */
+	AST_AUDIOHOOK_TRIGGER_WRITE = (2 << 0), /*!< Audiohook wants to be triggered when writing audio out */
+	AST_AUDIOHOOK_WANTS_DTMF = (1 << 1),    /*!< Audiohook also wants to receive DTMF frames */
+};
+
+struct ast_audiohook;
+
+/*! \brief Callback function for manipulate audiohook type
+ * \param audiohook Audiohook structure
+ * \param chan Channel
+ * \param frame Frame of audio to manipulate
+ * \param direction Direction frame came from
+ * \return Returns 0 on success, -1 on failure
+ * \note An audiohook does not have any reference to a private data structure for manipulate types. It is up to the manipulate callback to store this data
+ *       via it's own method. An example would be datastores.
+ */
+typedef int (*ast_audiohook_manipulate_callback)(struct ast_audiohook *audiohook, struct ast_channel *chan, struct ast_frame *frame, enum ast_audiohook_direction direction);
+
+struct ast_audiohook_options {
+	int read_volume;  /*!< Volume adjustment on frames read from the channel the hook is on */
+	int write_volume; /*!< Volume adjustment on frames written to the channel the hook is on */
+};
+
+struct ast_audiohook {
+	ast_mutex_t lock;                                      /*!< Lock that protects the audiohook structure */
+	ast_cond_t trigger;                                    /*!< Trigger condition (if enabled) */
+	enum ast_audiohook_type type;                          /*!< Type of audiohook */
+	enum ast_audiohook_status status;                      /*!< Status of the audiohook */
+	const char *source;                                    /*!< Who this audiohook ultimately belongs to */
+	unsigned int flags;                                    /*!< Flags on the audiohook */
+	struct ast_slinfactory read_factory;                   /*!< Factory where frames read from the channel, or read from the whisper source will go through */
+	struct ast_slinfactory write_factory;                  /*!< Factory where frames written to the channel will go through */
+	int format;                                            /*!< Format translation path is setup as */
+	struct ast_trans_pvt *trans_pvt;                       /*!< Translation path for reading frames */
+	ast_audiohook_manipulate_callback manipulate_callback; /*!< Manipulation callback */
+	struct ast_audiohook_options options;                  /*!< Applicable options */
+	AST_LIST_ENTRY(ast_audiohook) list;                    /*!< Linked list information */
+};
+
+struct ast_audiohook_list;
+
+/*! \brief Initialize an audiohook structure
+ * \param audiohook Audiohook structure
+ * \param type Type of audiohook to initialize this as
+ * \param source Who is initializing this audiohook
+ * \return Returns 0 on success, -1 on failure
+ */
+int ast_audiohook_init(struct ast_audiohook *audiohook, enum ast_audiohook_type type, const char *source);
+
+/*! \brief Destroys an audiohook structure
+ * \param audiohook Audiohook structure
+ * \return Returns 0 on success, -1 on failure
+ */
+int ast_audiohook_destroy(struct ast_audiohook *audiohook);
+
+/*! \brief Writes a frame into the audiohook structure
+ * \param audiohook Audiohook structure
+ * \param direction Direction the audio frame came from
+ * \param frame Frame to write in
+ * \return Returns 0 on success, -1 on failure
+ */
+int ast_audiohook_write_frame(struct ast_audiohook *audiohook, enum ast_audiohook_direction direction, struct ast_frame *frame);
+
+/*! \brief Reads a frame in from the audiohook structure
+ * \param audiohook Audiohook structure
+ * \param samples Number of samples wanted
+ * \param direction Direction the audio frame came from
+ * \param format Format of frame remote side wants back
+ * \return Returns frame on success, NULL on failure
+ */
+struct ast_frame *ast_audiohook_read_frame(struct ast_audiohook *audiohook, size_t samples, enum ast_audiohook_direction direction, int format);
+
+/*! \brief Attach audiohook to channel
+ * \param chan Channel
+ * \param audiohook Audiohook structure
+ * \return Returns 0 on success, -1 on failure
+ */
+int ast_audiohook_attach(struct ast_channel *chan, struct ast_audiohook *audiohook);
+
+/*! \brief Detach audiohook from channel
+ * \param audiohook Audiohook structure
+ * \return Returns 0 on success, -1 on failure
+ */
+int ast_audiohook_detach(struct ast_audiohook *audiohook);
+
+/*! \brief Detach audiohooks from list and destroy said list
+ * \param audiohook_list List of audiohooks
+ * \return Returns 0 on success, -1 on failure
+ */
+int ast_audiohook_detach_list(struct ast_audiohook_list *audiohook_list);
+
+/*! \brief Detach specified source audiohook from channel
+ * \param chan Channel to detach from
+ * \param source Name of source to detach
+ * \return Returns 0 on success, -1 on failure
+ */
+int ast_audiohook_detach_source(struct ast_channel *chan, const char *source);
+
+/*! \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
+ * \param direction Direction frame is coming in from
+ * \param frame The frame itself
+ * \return Return frame on success, NULL on failure
+ */
+struct ast_frame *ast_audiohook_write_list(struct ast_channel *chan, struct ast_audiohook_list *audiohook_list, enum ast_audiohook_direction direction, struct ast_frame *frame);
+
+/*! \brief Wait for audiohook trigger to be triggered
+ * \param audiohook Audiohook to wait on
+ */
+void ast_audiohook_trigger_wait(struct ast_audiohook *audiohook);
+
+/*! \brief Lock an audiohook
+ * \param audiohook Audiohook structure
+ */
+static inline int ast_audiohook_lock(struct ast_audiohook *audiohook)
+{
+	return ast_mutex_lock(&audiohook->lock);
+}
+
+/*! \brief Unlock an audiohook
+ * \param audiohook Audiohook structure
+ */
+static inline int ast_audiohook_unlock(struct ast_audiohook *audiohook)
+{
+	return ast_mutex_unlock(&audiohook->lock);
+}
+
+#if defined(__cplusplus) || defined(c_plusplus)
+}
+#endif
+
+#endif /* _ASTERISK_AUDIOHOOK_H */
diff --git a/include/asterisk/channel.h b/include/asterisk/channel.h
index 3e63259aa..39f2c636d 100644
--- a/include/asterisk/channel.h
+++ b/include/asterisk/channel.h
@@ -316,9 +316,6 @@ struct ast_channel_tech {
 	int (* func_channel_write)(struct ast_channel *chan, const char *function, char *data, const char *value);
 };
 
-struct ast_channel_spy_list;	/*!< \todo Add explanation here */
-struct ast_channel_whisper_buffer;	/*!< \todo Add explanation here */
-
 /*!
  * The high bit of the frame count is used as a debug marker, so
  * increments of the counters must be done with care.
@@ -481,8 +478,8 @@ struct ast_channel {
 	int rawreadformat;				/*!< Raw read format */
 	int rawwriteformat;				/*!< Raw write format */
 
-	struct ast_channel_spy_list *spies;		/*!< Chan Spy stuff */
-	struct ast_channel_whisper_buffer *whisper;	/*!< Whisper Paging buffer */
+	struct ast_audiohook_list *audiohooks;
+
 	AST_LIST_ENTRY(ast_channel) chan_list;		/*!< For easy linking */
 	
 	struct ast_jb jb;				/*!< The jitterbuffer state  */
diff --git a/include/asterisk/chanspy.h b/include/asterisk/chanspy.h
deleted file mode 100644
index 8550210d0..000000000
--- a/include/asterisk/chanspy.h
+++ /dev/null
@@ -1,150 +0,0 @@
-/*
- * Asterisk -- An open source telephony toolkit.
- *
- * Copyright (C) 1999 - 2006, Digium, Inc.
- *
- * Mark Spencer <markster@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 Asterisk PBX channel spy definitions
- */
-
-#ifndef _ASTERISK_CHANSPY_H
-#define _ASTERISK_CHANSPY_H
-
-#if defined(__cplusplus) || defined(c_plusplus)
-extern "C" {
-#endif
-
-#include "asterisk/linkedlists.h"
-
-enum chanspy_states {
-	CHANSPY_NEW = 0,		/*!< spy not yet operating */
-	CHANSPY_RUNNING = 1,		/*!< normal operation, spy is still operating */
-	CHANSPY_DONE = 2,		/*!< spy is stopped and already removed from channel */
-	CHANSPY_STOP = 3,		/*!< spy requested to stop, still attached to channel */
-};
-
-enum chanspy_flags {
-	CHANSPY_MIXAUDIO = (1 << 0),
-	CHANSPY_READ_VOLADJUST = (1 << 1),
-	CHANSPY_WRITE_VOLADJUST = (1 << 2),
-	CHANSPY_FORMAT_AUDIO = (1 << 3),
-	CHANSPY_TRIGGER_MODE = (3 << 4),
-	CHANSPY_TRIGGER_READ = (1 << 4),
-	CHANSPY_TRIGGER_WRITE = (2 << 4),
-	CHANSPY_TRIGGER_NONE = (3 << 4),
-	CHANSPY_TRIGGER_FLUSH = (1 << 6),
-};
-
-struct ast_channel_spy_queue {
-	AST_LIST_HEAD_NOLOCK(, ast_frame) list;
-	unsigned int samples;
-	unsigned int format;
-};
-
-struct ast_channel_spy {
-	AST_LIST_ENTRY(ast_channel_spy) list;
-	ast_mutex_t lock;
-	ast_cond_t trigger;
-	struct ast_channel *chan;
-	struct ast_channel_spy_queue read_queue;
-	struct ast_channel_spy_queue write_queue;
-	unsigned int flags;
-	enum chanspy_states status;
-	const char *type;
-	/* The volume adjustment values are very straightforward:
-	   positive values cause the samples to be multiplied by that amount
-	   negative values cause the samples to be divided by the absolute value of that amount
-	*/
-	int read_vol_adjustment;
-	int write_vol_adjustment;
-};
-
-/*!
-  \brief Adds a spy to a channel, to begin receiving copies of the channel's audio frames.
-  \param chan The channel to add the spy to.
-  \param spy A pointer to ast_channel_spy structure describing how the spy is to be used.
-  \return 0 for success, non-zero for failure
-
-  Note: This function performs no locking; you must hold the channel's lock before
-  calling this function.
- */
-int ast_channel_spy_add(struct ast_channel *chan, struct ast_channel_spy *spy);
-
-/*!
-  \brief Remove a spy from a channel.
-  \param chan The channel to remove the spy from
-  \param spy The spy to be removed
-  \return nothing
-
-  Note: This function performs no locking; you must hold the channel's lock before
-  calling this function.
- */
-void ast_channel_spy_remove(struct ast_channel *chan, struct ast_channel_spy *spy);
-
-/*!
-  \brief Free a spy.
-  \param spy The spy to free
-  \return nothing
-
-  Note: This function MUST NOT be called with the spy locked.
-*/
-void ast_channel_spy_free(struct ast_channel_spy *spy);
-
-/*!
-  \brief Find all spies of a particular type on a channel and stop them.
-  \param chan The channel to operate on
-  \param type A character string identifying the type of spies to be stopped
-  \return nothing
-
-  Note: This function performs no locking; you must hold the channel's lock before
-  calling this function.
- */
-void ast_channel_spy_stop_by_type(struct ast_channel *chan, const char *type);
-
-/*!
-  \brief Read one (or more) frames of audio from a channel being spied upon.
-  \param spy The spy to operate on
-  \param samples The number of audio samples to read
-  \return NULL for failure, one ast_frame pointer, or a chain of ast_frame pointers
-
-  This function can return multiple frames if the spy structure needs to be 'flushed'
-  due to mismatched queue lengths, or if the spy structure is configured to return
-  unmixed audio (in which case each call to this function will return a frame of audio
-  from each side of channel).
-
-  Note: This function performs no locking; you must hold the spy's lock before calling
-  this function. You must <b>not</b> hold the channel's lock at the same time.
- */
-struct ast_frame *ast_channel_spy_read_frame(struct ast_channel_spy *spy, unsigned int samples);
-
-/*!
-  \brief Efficiently wait until audio is available for a spy, or an exception occurs.
-  \param spy The spy to wait on
-  \return nothing
-
-  Note: The locking rules for this function are non-obvious... first, you must <b>not</b>
-  hold the channel's lock when calling this function. Second, you must hold the spy's lock
-  before making the function call; while the function runs the lock will be released, and
-  when the trigger event occurs, the lock will be re-obtained. This means that when control
-  returns to your code, you will again hold the spy's lock.
- */
-void ast_channel_spy_trigger_wait(struct ast_channel_spy *spy);
-
-#if defined(__cplusplus) || defined(c_plusplus)
-}
-#endif
-
-#endif /* _ASTERISK_CHANSPY_H */
diff --git a/include/asterisk/slinfactory.h b/include/asterisk/slinfactory.h
index b81817d6b..3ab42d283 100644
--- a/include/asterisk/slinfactory.h
+++ b/include/asterisk/slinfactory.h
@@ -46,6 +46,7 @@ void ast_slinfactory_destroy(struct ast_slinfactory *sf);
 int ast_slinfactory_feed(struct ast_slinfactory *sf, struct ast_frame *f);
 int ast_slinfactory_read(struct ast_slinfactory *sf, short *buf, size_t samples);
 unsigned int ast_slinfactory_available(const struct ast_slinfactory *sf);
+void ast_slinfactory_flush(struct ast_slinfactory *sf);
 
 #if defined(__cplusplus) || defined(c_plusplus)
 }
diff --git a/main/Makefile b/main/Makefile
index f3c1b99f9..ba780b281 100644
--- a/main/Makefile
+++ b/main/Makefile
@@ -26,7 +26,7 @@ OBJS=	io.o sched.o logger.o frame.o loader.o config.o channel.o \
 	utils.o plc.o jitterbuf.o dnsmgr.o devicestate.o \
 	netsock.o slinfactory.o ast_expr2.o ast_expr2f.o \
 	cryptostub.o sha1.o http.o fixedjitterbuf.o abstract_jb.o \
-	strcompat.o threadstorage.o dial.o event.o adsistub.o
+	strcompat.o threadstorage.o dial.o event.o adsistub.o audiohook.o
 
 # we need to link in the objects statically, not as a library, because
 # otherwise modules will not have them available if none of the static
diff --git a/main/audiohook.c b/main/audiohook.c
new file mode 100644
index 000000000..a7600a356
--- /dev/null
+++ b/main/audiohook.c
@@ -0,0 +1,625 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 1999 - 2007, Digium, Inc.
+ *
+ * Joshua Colp <jcolp@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 Audiohooks Architecture
+ *
+ * \author Joshua Colp <jcolp@digium.com>
+ */
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+#include <signal.h>
+#include <errno.h>
+#include <unistd.h>
+
+#include "asterisk/logger.h"
+#include "asterisk/channel.h"
+#include "asterisk/options.h"
+#include "asterisk/utils.h"
+#include "asterisk/lock.h"
+#include "asterisk/linkedlists.h"
+#include "asterisk/audiohook.h"
+#include "asterisk/slinfactory.h"
+#include "asterisk/frame.h"
+#include "asterisk/translate.h"
+
+struct ast_audiohook_translate {
+	struct ast_trans_pvt *trans_pvt;
+	int format;
+};
+
+struct ast_audiohook_list {
+	struct ast_audiohook_translate in_translate[2];
+	struct ast_audiohook_translate out_translate[2];
+	AST_LIST_HEAD_NOLOCK(, ast_audiohook) spy_list;
+	AST_LIST_HEAD_NOLOCK(, ast_audiohook) whisper_list;
+	AST_LIST_HEAD_NOLOCK(, ast_audiohook) manipulate_list;
+};
+
+/*! \brief Initialize an audiohook structure
+ * \param audiohook Audiohook structure
+ * \return Returns 0 on success, -1 on failure
+ */
+int ast_audiohook_init(struct ast_audiohook *audiohook, enum ast_audiohook_type type, const char *source)
+{
+	/* Need to keep the type and source */
+	audiohook->type = type;
+	audiohook->source = source;
+
+	/* Initialize lock that protects our audiohook */
+	ast_mutex_init(&audiohook->lock);
+	ast_cond_init(&audiohook->trigger, NULL);
+
+	/* Setup the factories that are needed for this audiohook type */
+	switch (type) {
+	case AST_AUDIOHOOK_TYPE_SPY:
+		ast_slinfactory_init(&audiohook->read_factory);
+	case AST_AUDIOHOOK_TYPE_WHISPER:
+		ast_slinfactory_init(&audiohook->write_factory);
+		break;
+	default:
+		break;
+	}
+
+	/* Since we are just starting out... this audiohook is new */
+	audiohook->status = AST_AUDIOHOOK_STATUS_NEW;
+
+	return 0;
+}
+
+/*! \brief Destroys an audiohook structure
+ * \param audiohook Audiohook structure
+ * \return Returns 0 on success, -1 on failure
+ */
+int ast_audiohook_destroy(struct ast_audiohook *audiohook)
+{
+	/* Drop the factories used by this audiohook type */
+	switch (audiohook->type) {
+	case AST_AUDIOHOOK_TYPE_SPY:
+		ast_slinfactory_destroy(&audiohook->read_factory);
+	case AST_AUDIOHOOK_TYPE_WHISPER:
+		ast_slinfactory_destroy(&audiohook->write_factory);
+		break;
+	default:
+		break;
+	}
+
+	/* Destroy translation path if present */
+	if (audiohook->trans_pvt)
+		ast_translator_free_path(audiohook->trans_pvt);
+
+	/* Lock and trigger be gone! */
+	ast_cond_destroy(&audiohook->trigger);
+	ast_mutex_destroy(&audiohook->lock);
+
+	return 0;
+}
+
+/*! \brief Writes a frame into the audiohook structure
+ * \param audiohook Audiohook structure
+ * \param direction Direction the audio frame came from
+ * \param frame Frame to write in
+ * \return Returns 0 on success, -1 on failure
+ */
+int ast_audiohook_write_frame(struct ast_audiohook *audiohook, enum ast_audiohook_direction direction, struct ast_frame *frame)
+{
+	struct ast_slinfactory *factory = (direction == AST_AUDIOHOOK_DIRECTION_READ ? &audiohook->read_factory : &audiohook->write_factory);
+
+	/* Write frame out to respective factory */
+	ast_slinfactory_feed(factory, frame);
+
+	/* If we need to notify the respective handler of this audiohook, do so */
+	switch (ast_test_flag(audiohook, AST_AUDIOHOOK_TRIGGER_MODE)) {
+	case AST_AUDIOHOOK_TRIGGER_READ:
+		if (direction == AST_AUDIOHOOK_DIRECTION_READ)
+			ast_cond_signal(&audiohook->trigger);
+		break;
+	case AST_AUDIOHOOK_TRIGGER_WRITE:
+		if (direction == AST_AUDIOHOOK_DIRECTION_WRITE)
+			ast_cond_signal(&audiohook->trigger);
+		break;
+	default:
+		break;
+	}
+
+	return 0;
+}
+
+static struct ast_frame *audiohook_read_frame_single(struct ast_audiohook *audiohook, size_t samples, enum ast_audiohook_direction direction)
+{
+	struct ast_slinfactory *factory = (direction == AST_AUDIOHOOK_DIRECTION_READ ? &audiohook->read_factory : &audiohook->write_factory);
+	int vol = (direction == AST_AUDIOHOOK_DIRECTION_READ ? audiohook->options.read_volume : audiohook->options.write_volume);
+	short buf[samples];
+	struct ast_frame frame = {
+		.frametype = AST_FRAME_VOICE,
+		.subclass = AST_FORMAT_SLINEAR,
+		.data = buf,
+		.datalen = sizeof(buf),
+		.samples = samples,
+	};
+
+	/* Ensure the factory is able to give us the samples we want */
+	if (samples > ast_slinfactory_available(factory))
+		return NULL;
+	
+	/* Read data in from factory */
+	if (!ast_slinfactory_read(factory, buf, samples))
+		return NULL;
+
+	/* If a volume adjustment needs to be applied apply it */
+	if (vol)
+		ast_frame_adjust_volume(&frame, vol);
+
+	return ast_frdup(&frame);
+}
+
+static struct ast_frame *audiohook_read_frame_both(struct ast_audiohook *audiohook, size_t samples)
+{
+	int i = 0;
+	short buf1[samples], buf2[samples], *read_buf = NULL, *write_buf = NULL, *final_buf = NULL, *data1 = NULL, *data2 = NULL;
+	struct ast_frame frame = {
+		.frametype = AST_FRAME_VOICE,
+		.subclass = AST_FORMAT_SLINEAR,
+		.data = NULL,
+		.datalen = sizeof(buf1),
+		.samples = samples,
+	};
+
+	/* Start with the read factory... if there are enough samples, read them in */
+	if (ast_slinfactory_available(&audiohook->read_factory) >= samples) {
+		if (ast_slinfactory_read(&audiohook->read_factory, buf1, samples))
+			read_buf = buf1;
+		/* Adjust read volume if need be */
+		if (audiohook->options.read_volume) {
+			int count = 0;
+			short adjust_value = abs(audiohook->options.read_volume);
+			for (count = 0; count < samples; count++) {
+				if (audiohook->options.read_volume > 0)
+					ast_slinear_saturated_multiply(&buf1[count], &adjust_value);
+				else if (audiohook->options.read_volume < 0)
+					ast_slinear_saturated_divide(&buf1[count], &adjust_value);
+			}
+		}
+	} else if (option_debug)
+		ast_log(LOG_DEBUG, "Failed to get %zd samples from read factory %p\n", samples, &audiohook->read_factory);
+
+	/* Move on to the write factory... if there are enough samples, read them in */
+	if (ast_slinfactory_available(&audiohook->write_factory) >= samples) {
+		if (ast_slinfactory_read(&audiohook->write_factory, buf2, samples))
+			write_buf = buf2;
+		/* Adjust write volume if need be */
+		if (audiohook->options.write_volume) {
+			int count = 0;
+			short adjust_value = abs(audiohook->options.write_volume);
+			for (count = 0; count < samples; count++) {
+				if (audiohook->options.write_volume > 0)
+					ast_slinear_saturated_multiply(&buf2[count], &adjust_value);
+				else if (audiohook->options.write_volume < 0)
+					ast_slinear_saturated_divide(&buf2[count], &adjust_value);
+			}
+		}
+	} else if (option_debug)
+		ast_log(LOG_DEBUG, "Failed to get %zd samples from write factory %p\n", samples, &audiohook->write_factory);
+
+	/* Basically we figure out which buffer to use... and if mixing can be done here */
+	if (!read_buf && !write_buf)
+		return NULL;
+	else if (read_buf && write_buf) {
+		for (i = 0, data1 = read_buf, data2 = write_buf; i < samples; i++, data1++, data2++)
+			ast_slinear_saturated_add(data1, data2);
+		final_buf = buf1;
+	} else if (read_buf)
+		final_buf = buf1;
+	else if (write_buf)
+		final_buf = buf2;
+
+	/* Make the final buffer part of the frame, so it gets duplicated fine */
+	frame.data = final_buf;
+
+	/* Yahoo, a combined copy of the audio! */
+	return ast_frdup(&frame);
+}
+
+/*! \brief Reads a frame in from the audiohook structure
+ * \param audiohook Audiohook structure
+ * \param samples Number of samples wanted
+ * \param direction Direction the audio frame came from
+ * \param format Format of frame remote side wants back
+ * \return Returns frame on success, NULL on failure
+ */
+struct ast_frame *ast_audiohook_read_frame(struct ast_audiohook *audiohook, size_t samples, enum ast_audiohook_direction direction, int format)
+{
+	struct ast_frame *read_frame = NULL, *final_frame = NULL;
+
+	if (!(read_frame = (direction == AST_AUDIOHOOK_DIRECTION_BOTH ? audiohook_read_frame_both(audiohook, samples) : audiohook_read_frame_single(audiohook, samples, direction))))
+		return NULL;
+
+	/* If they don't want signed linear back out, we'll have to send it through the translation path */
+	if (format != AST_FORMAT_SLINEAR) {
+		/* Rebuild translation path if different format then previously */
+		if (audiohook->format != format) {
+			if (audiohook->trans_pvt) {
+				ast_translator_free_path(audiohook->trans_pvt);
+				audiohook->trans_pvt = NULL;
+			}
+			/* Setup new translation path for this format... if we fail we can't very well return signed linear so free the frame and return nothing */
+			if (!(audiohook->trans_pvt = ast_translator_build_path(format, AST_FORMAT_SLINEAR))) {
+				ast_frfree(read_frame);
+				return NULL;
+			}
+		}
+		/* Convert to requested format, and allow the read in frame to be freed */
+		final_frame = ast_translate(audiohook->trans_pvt, read_frame, 1);
+	} else {
+		final_frame = read_frame;
+	}
+
+	return final_frame;
+}
+
+/*! \brief Attach audiohook to channel
+ * \param chan Channel
+ * \param audiohook Audiohook structure
+ * \return Returns 0 on success, -1 on failure
+ */
+int ast_audiohook_attach(struct ast_channel *chan, struct ast_audiohook *audiohook)
+{
+	ast_channel_lock(chan);
+
+	if (!chan->audiohooks) {
+		/* Whoops... allocate a new structure */
+		if (!(chan->audiohooks = ast_calloc(1, sizeof(*chan->audiohooks)))) {
+			ast_channel_unlock(chan);
+			return -1;
+		}
+		AST_LIST_HEAD_INIT_NOLOCK(&chan->audiohooks->spy_list);
+		AST_LIST_HEAD_INIT_NOLOCK(&chan->audiohooks->whisper_list);
+		AST_LIST_HEAD_INIT_NOLOCK(&chan->audiohooks->manipulate_list);
+	}
+
+	/* Drop into respective list */
+	if (audiohook->type == AST_AUDIOHOOK_TYPE_SPY)
+		AST_LIST_INSERT_TAIL(&chan->audiohooks->spy_list, audiohook, list);
+	else if (audiohook->type == AST_AUDIOHOOK_TYPE_WHISPER)
+		AST_LIST_INSERT_TAIL(&chan->audiohooks->whisper_list, audiohook, list);
+	else if (audiohook->type == AST_AUDIOHOOK_TYPE_MANIPULATE)
+		AST_LIST_INSERT_TAIL(&chan->audiohooks->manipulate_list, audiohook, list);
+
+	/* Change status over to running since it is now attached */
+	audiohook->status = AST_AUDIOHOOK_STATUS_RUNNING;
+
+	ast_channel_unlock(chan);
+
+	return 0;
+}
+
+/*! \brief Detach audiohook from channel
+ * \param audiohook Audiohook structure
+ * \return Returns 0 on success, -1 on failure
+ */
+int ast_audiohook_detach(struct ast_audiohook *audiohook)
+{
+	if (audiohook->status == AST_AUDIOHOOK_STATUS_DONE)
+		return 0;
+
+	audiohook->status = AST_AUDIOHOOK_STATUS_SHUTDOWN;
+
+	while (audiohook->status != AST_AUDIOHOOK_STATUS_DONE)
+		ast_audiohook_trigger_wait(audiohook);
+
+	return 0;
+}
+
+/*! \brief Detach audiohooks from list and destroy said list
+ * \param audiohook_list List of audiohooks
+ * \return Returns 0 on success, -1 on failure
+ */
+int ast_audiohook_detach_list(struct ast_audiohook_list *audiohook_list)
+{
+	int i = 0;
+	struct ast_audiohook *audiohook = NULL;
+
+	/* Drop any spies */
+	AST_LIST_TRAVERSE_SAFE_BEGIN(&audiohook_list->spy_list, audiohook, list) {
+		ast_audiohook_lock(audiohook);
+		AST_LIST_REMOVE_CURRENT(&audiohook_list->spy_list, list);
+		audiohook->status = AST_AUDIOHOOK_STATUS_DONE;
+		ast_cond_signal(&audiohook->trigger);
+		ast_audiohook_unlock(audiohook);
+	}
+	AST_LIST_TRAVERSE_SAFE_END
+
+	/* Drop any whispering sources */
+	AST_LIST_TRAVERSE_SAFE_BEGIN(&audiohook_list->whisper_list, audiohook, list) {
+		ast_audiohook_lock(audiohook);
+		AST_LIST_REMOVE_CURRENT(&audiohook_list->whisper_list, list);
+		audiohook->status = AST_AUDIOHOOK_STATUS_DONE;
+		ast_cond_signal(&audiohook->trigger);
+		ast_audiohook_unlock(audiohook);
+	}
+	AST_LIST_TRAVERSE_SAFE_END
+
+	/* Drop any manipulaters */
+	AST_LIST_TRAVERSE_SAFE_BEGIN(&audiohook_list->manipulate_list, audiohook, list) {
+		ast_audiohook_lock(audiohook);
+		ast_mutex_lock(&audiohook->lock);
+		AST_LIST_REMOVE_CURRENT(&audiohook_list->manipulate_list, list);
+		audiohook->status = AST_AUDIOHOOK_STATUS_DONE;
+		ast_audiohook_unlock(audiohook);
+		audiohook->manipulate_callback(audiohook, NULL, NULL, 0);
+	}
+	AST_LIST_TRAVERSE_SAFE_END
+
+	/* Drop translation paths if present */
+	for (i = 0; i < 2; i++) {
+		if (audiohook_list->in_translate[i].trans_pvt)
+			ast_translator_free_path(audiohook_list->in_translate[i].trans_pvt);
+		if (audiohook_list->out_translate[i].trans_pvt)
+			ast_translator_free_path(audiohook_list->out_translate[i].trans_pvt);
+	}
+	
+	/* Free ourselves */
+	ast_free(audiohook_list);
+
+	return 0;
+}
+
+static struct ast_audiohook *find_audiohook_by_source(struct ast_audiohook_list *audiohook_list, const char *source)
+{
+	struct ast_audiohook *audiohook = NULL;
+
+	AST_LIST_TRAVERSE(&audiohook_list->spy_list, audiohook, list) {
+		if (!strcasecmp(audiohook->source, source))
+			return audiohook;
+	}
+
+	AST_LIST_TRAVERSE(&audiohook_list->whisper_list, audiohook, list) {
+		if (!strcasecmp(audiohook->source, source))
+			return audiohook;
+	}
+
+	AST_LIST_TRAVERSE(&audiohook_list->manipulate_list, audiohook, list) {
+		if (!strcasecmp(audiohook->source, source))
+			return audiohook;
+	}
+
+	return NULL;
+}
+
+/*! \brief Detach specified source audiohook from channel
+ * \param chan Channel to detach from
+ * \param source Name of source to detach
+ * \return Returns 0 on success, -1 on failure
+ */
+int ast_audiohook_detach_source(struct ast_channel *chan, const char *source)
+{
+	struct ast_audiohook *audiohook = NULL;
+
+	ast_channel_lock(chan);
+
+	/* Ensure the channel has audiohooks on it */
+	if (!chan->audiohooks) {
+		ast_channel_unlock(chan);
+		return -1;
+	}
+
+	audiohook = find_audiohook_by_source(chan->audiohooks, source);
+
+	ast_channel_unlock(chan);
+
+	if (audiohook && audiohook->status != AST_AUDIOHOOK_STATUS_DONE)
+		audiohook->status = AST_AUDIOHOOK_STATUS_SHUTDOWN;
+
+	return (audiohook ? 0 : -1);
+}
+
+/*! \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
+ * \param direction Direction frame is coming in from
+ * \param frame The frame itself
+ * \return Return frame on success, NULL on failure
+ */
+static struct ast_frame *dtmf_audiohook_write_list(struct ast_channel *chan, struct ast_audiohook_list *audiohook_list, enum ast_audiohook_direction direction, struct ast_frame *frame)
+{
+	struct ast_audiohook *audiohook = NULL;
+
+	AST_LIST_TRAVERSE_SAFE_BEGIN(&audiohook_list->manipulate_list, audiohook, list) {
+		ast_audiohook_lock(audiohook);
+		if (audiohook->status != AST_AUDIOHOOK_STATUS_RUNNING) {
+			AST_LIST_REMOVE_CURRENT(&audiohook_list->manipulate_list, list);
+			audiohook->status = AST_AUDIOHOOK_STATUS_DONE;
+			ast_audiohook_unlock(audiohook);
+			audiohook->manipulate_callback(audiohook, NULL, NULL, 0);
+			continue;
+		}
+		if (ast_test_flag(audiohook, AST_AUDIOHOOK_WANTS_DTMF))
+			audiohook->manipulate_callback(audiohook, chan, frame, direction);
+		ast_audiohook_unlock(audiohook);
+	}
+	AST_LIST_TRAVERSE_SAFE_END
+
+	return frame;
+}
+
+/*! \brief Pass an AUDIO 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
+ * \param direction Direction frame is coming in from
+ * \param frame The frame itself
+ * \return Return frame on success, NULL on failure
+ */
+static struct ast_frame *audio_audiohook_write_list(struct ast_channel *chan, struct ast_audiohook_list *audiohook_list, enum ast_audiohook_direction direction, struct ast_frame *frame)
+{
+	struct ast_audiohook_translate *in_translate = (direction == AST_AUDIOHOOK_DIRECTION_READ ? &audiohook_list->in_translate[0] : &audiohook_list->in_translate[1]);
+	struct ast_audiohook_translate *out_translate = (direction == AST_AUDIOHOOK_DIRECTION_READ ? &audiohook_list->out_translate[0] : &audiohook_list->out_translate[1]);
+	struct ast_frame *start_frame = frame, *middle_frame = frame, *end_frame = frame;
+	struct ast_audiohook *audiohook = NULL;
+	int samples = frame->samples;
+	
+	/* If the frame coming in is not signed linear we have to send it through the in_translate path */
+	if (frame->subclass != AST_FORMAT_SLINEAR) {
+		if (in_translate->format != frame->subclass) {
+			if (in_translate->trans_pvt)
+				ast_translator_free_path(in_translate->trans_pvt);
+			if (!(in_translate->trans_pvt = ast_translator_build_path(AST_FORMAT_SLINEAR, frame->subclass)))
+				return frame;
+			in_translate->format = frame->subclass;
+		}
+		if (!(middle_frame = ast_translate(in_translate->trans_pvt, frame, 0)))
+			return frame;
+	}
+
+	/* Queue up signed linear frame to each spy */
+	AST_LIST_TRAVERSE_SAFE_BEGIN(&audiohook_list->spy_list, audiohook, list) {
+		ast_audiohook_lock(audiohook);
+		if (audiohook->status != AST_AUDIOHOOK_STATUS_RUNNING) {
+			AST_LIST_REMOVE_CURRENT(&audiohook_list->spy_list, list);
+			audiohook->status = AST_AUDIOHOOK_STATUS_DONE;
+			ast_cond_signal(&audiohook->trigger);
+			ast_audiohook_unlock(audiohook);
+			continue;
+		}
+		ast_audiohook_write_frame(audiohook, direction, middle_frame);
+		ast_audiohook_unlock(audiohook);
+	}
+	AST_LIST_TRAVERSE_SAFE_END
+
+	/* If this frame is being written out to the channel then we need to use whisper sources */
+	if (direction == AST_AUDIOHOOK_DIRECTION_WRITE && !AST_LIST_EMPTY(&audiohook_list->whisper_list)) {
+		int i = 0;
+		short read_buf[samples], combine_buf[samples], *data1 = NULL, *data2 = NULL;
+		memset(&combine_buf, 0, sizeof(combine_buf));
+		AST_LIST_TRAVERSE_SAFE_BEGIN(&audiohook_list->whisper_list, audiohook, list) {
+			ast_audiohook_lock(audiohook);
+			if (audiohook->status != AST_AUDIOHOOK_STATUS_RUNNING) {
+				AST_LIST_REMOVE_CURRENT(&audiohook_list->whisper_list, list);
+				audiohook->status = AST_AUDIOHOOK_STATUS_DONE;
+				ast_cond_signal(&audiohook->trigger);
+				ast_audiohook_unlock(audiohook);
+				continue;
+			}
+			if (ast_slinfactory_available(&audiohook->write_factory) >= samples && ast_slinfactory_read(&audiohook->write_factory, read_buf, samples)) {
+				/* Take audio from this whisper source and combine it into our main buffer */
+				for (i = 0, data1 = combine_buf, data2 = read_buf; i < samples; i++, data1++, data2++)
+					ast_slinear_saturated_add(data1, data2);
+			}
+			ast_audiohook_unlock(audiohook);
+		}
+		AST_LIST_TRAVERSE_SAFE_END
+		/* We take all of the combined whisper sources and combine them into the audio being written out */
+		for (i = 0, data1 = middle_frame->data, data2 = combine_buf; i < samples; i++, data1++, data2++)
+			ast_slinear_saturated_add(data1, data2);
+		end_frame = middle_frame;
+	}
+
+	/* Pass off frame to manipulate audiohooks */
+	if (!AST_LIST_EMPTY(&audiohook_list->manipulate_list)) {
+		AST_LIST_TRAVERSE_SAFE_BEGIN(&audiohook_list->manipulate_list, audiohook, list) {
+			ast_audiohook_lock(audiohook);
+			if (audiohook->status != AST_AUDIOHOOK_STATUS_RUNNING) {
+				AST_LIST_REMOVE_CURRENT(&audiohook_list->manipulate_list, list);
+				audiohook->status = AST_AUDIOHOOK_STATUS_DONE;
+				ast_audiohook_unlock(audiohook);
+				/* We basically drop all of our links to the manipulate audiohook and prod it to do it's own destructive things */
+				audiohook->manipulate_callback(audiohook, chan, NULL, direction);
+				continue;
+			}
+			/* Feed in frame to manipulation */
+			audiohook->manipulate_callback(audiohook, chan, middle_frame, direction);
+			ast_audiohook_unlock(audiohook);
+		}
+		AST_LIST_TRAVERSE_SAFE_END
+		end_frame = middle_frame;
+	}
+
+	/* Now we figure out what to do with our end frame (whether to transcode or not) */
+	if (middle_frame == end_frame) {
+		/* Middle frame was modified and became the end frame... let's see if we need to transcode */
+		if (end_frame->subclass != start_frame->subclass) {
+			if (out_translate->format != start_frame->subclass) {
+				if (out_translate->trans_pvt)
+					ast_translator_free_path(out_translate->trans_pvt);
+				if (!(out_translate->trans_pvt = ast_translator_build_path(start_frame->subclass, AST_FORMAT_SLINEAR))) {
+					/* We can't transcode this... drop our middle frame and return the original */
+					ast_frfree(middle_frame);
+					return start_frame;
+				}
+				out_translate->format = start_frame->subclass;
+			}
+			/* Transcode from our middle (signed linear) frame to new format of the frame that came in */
+			if (!(end_frame = ast_translate(out_translate->trans_pvt, middle_frame, 0))) {
+				/* Failed to transcode the frame... drop it and return the original */
+				ast_frfree(middle_frame);
+				return start_frame;
+			}
+			/* Here's the scoop... middle frame is no longer of use to us */
+			ast_frfree(middle_frame);
+		}
+		/* Yay let's rid ourselves of the start frame */
+		ast_frfree(start_frame);
+	} else {
+		/* No frame was modified, we can just drop our middle frame and pass the frame we got in out */
+		ast_frfree(middle_frame);
+	}
+
+	return end_frame;
+}
+
+/*! \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
+ * \param direction Direction frame is coming in from
+ * \param frame The frame itself
+ * \return Return frame on success, NULL on failure
+ */
+struct ast_frame *ast_audiohook_write_list(struct ast_channel *chan, struct ast_audiohook_list *audiohook_list, enum ast_audiohook_direction direction, struct ast_frame *frame)
+{
+	/* Pass off frame to it's respective list write function */
+	if (frame->frametype == AST_FRAME_VOICE)
+		return audio_audiohook_write_list(chan, audiohook_list, direction, frame);
+	else if (frame->frametype == AST_FRAME_DTMF)
+		return dtmf_audiohook_write_list(chan, audiohook_list, direction, frame);
+	else
+		return frame;
+}
+			
+
+/*! \brief Wait for audiohook trigger to be triggered
+ * \param audiohook Audiohook to wait on
+ */
+void ast_audiohook_trigger_wait(struct ast_audiohook *audiohook)
+{
+	struct timeval tv;
+	struct timespec ts;
+
+	tv = ast_tvadd(ast_tvnow(), ast_samp2tv(50000, 1000));
+	ts.tv_sec = tv.tv_sec;
+	ts.tv_nsec = tv.tv_usec * 1000;
+	
+	ast_cond_timedwait(&audiohook->trigger, &audiohook->lock, &ts);
+	
+	return;
+}
diff --git a/main/channel.c b/main/channel.c
index 5c8de5694..3f3ce9466 100644
--- a/main/channel.c
+++ b/main/channel.c
@@ -44,7 +44,6 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 #include "asterisk/sched.h"
 #include "asterisk/options.h"
 #include "asterisk/channel.h"
-#include "asterisk/chanspy.h"
 #include "asterisk/musiconhold.h"
 #include "asterisk/logger.h"
 #include "asterisk/say.h"
@@ -66,27 +65,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 #include "asterisk/sha1.h"
 #include "asterisk/threadstorage.h"
 #include "asterisk/slinfactory.h"
-
-struct channel_spy_trans {
-	int last_format;
-	struct ast_trans_pvt *path;
-};
-
-/*! \brief List of SPY structures 
-*/
-struct ast_channel_spy_list {
-	struct channel_spy_trans read_translator;
-	struct channel_spy_trans write_translator;
-	AST_LIST_HEAD_NOLOCK(, ast_channel_spy) list;
-};
-
-/*! \brief Definition of the Whisper buffer */
-struct ast_channel_whisper_buffer {
-	ast_mutex_t lock;
-	struct ast_slinfactory sf;
-	unsigned int original_format;
-	struct ast_trans_pvt *path;
-};
+#include "asterisk/audiohook.h"
 
 /* uncomment if you have problems with 'monitoring' synchronized files */
 #if 0
@@ -1121,10 +1100,6 @@ void ast_channel_free(struct ast_channel *chan)
 	if (chan->music_state)
 		ast_moh_cleanup(chan);
 
-	/* if someone is whispering on the channel, stop them */
-	if (chan->whisper)
-		ast_channel_whisper_stop(chan);
-
 	/* Free translators */
 	if (chan->readtrans)
 		ast_translator_free_path(chan->readtrans);
@@ -1281,176 +1256,6 @@ struct ast_datastore *ast_channel_datastore_find(struct ast_channel *chan, const
 	return datastore;
 }
 
-int ast_channel_spy_add(struct ast_channel *chan, struct ast_channel_spy *spy)
-{
-	/* Link the owner channel to the spy */
-	spy->chan = chan;
-
-	if (!ast_test_flag(spy, CHANSPY_FORMAT_AUDIO)) {
-		ast_log(LOG_WARNING, "Could not add channel spy '%s' to channel '%s', only audio format spies are supported.\n",
-			spy->type, chan->name);
-		return -1;
-	}
-
-	if (ast_test_flag(spy, CHANSPY_READ_VOLADJUST) && (spy->read_queue.format != AST_FORMAT_SLINEAR)) {
-		ast_log(LOG_WARNING, "Cannot provide volume adjustment on '%s' format spies\n",
-			ast_getformatname(spy->read_queue.format));
-		return -1;
-	}
-
-	if (ast_test_flag(spy, CHANSPY_WRITE_VOLADJUST) && (spy->write_queue.format != AST_FORMAT_SLINEAR)) {
-		ast_log(LOG_WARNING, "Cannot provide volume adjustment on '%s' format spies\n",
-			ast_getformatname(spy->write_queue.format));
-		return -1;
-	}
-
-	if (ast_test_flag(spy, CHANSPY_MIXAUDIO) &&
-	    ((spy->read_queue.format != AST_FORMAT_SLINEAR) ||
-	     (spy->write_queue.format != AST_FORMAT_SLINEAR))) {
-		ast_log(LOG_WARNING, "Cannot provide audio mixing on '%s'-'%s' format spies\n",
-			ast_getformatname(spy->read_queue.format), ast_getformatname(spy->write_queue.format));
-		return -1;
-	}
-
-	if (!chan->spies) {
-		if (!(chan->spies = ast_calloc(1, sizeof(*chan->spies)))) {
-			return -1;
-		}
-
-		AST_LIST_HEAD_INIT_NOLOCK(&chan->spies->list);
-		AST_LIST_INSERT_HEAD(&chan->spies->list, spy, list);
-	} else {
-		AST_LIST_INSERT_TAIL(&chan->spies->list, spy, list);
-	}
-
-	if (ast_test_flag(spy, CHANSPY_TRIGGER_MODE) != CHANSPY_TRIGGER_NONE) {
-		ast_cond_init(&spy->trigger, NULL);
-		ast_set_flag(spy, CHANSPY_TRIGGER_READ);
-		ast_clear_flag(spy, CHANSPY_TRIGGER_WRITE);
-	}
-
-	ast_debug(1, "Spy %s added to channel %s\n",
-		spy->type, chan->name);
-
-	return 0;
-}
-
-/* Clean up a channel's spy information */
-static void spy_cleanup(struct ast_channel *chan)
-{
-	if (!AST_LIST_EMPTY(&chan->spies->list))
-		return;
-	if (chan->spies->read_translator.path)
-		ast_translator_free_path(chan->spies->read_translator.path);
-	if (chan->spies->write_translator.path)
-		ast_translator_free_path(chan->spies->write_translator.path);
-	ast_free(chan->spies);
-	chan->spies = NULL;
-	return;
-}
-
-/* Detach a spy from it's channel */
-static void spy_detach(struct ast_channel_spy *spy, struct ast_channel *chan)
-{
-	/* We only need to poke them if they aren't already done */
-	if (spy->status != CHANSPY_DONE) {
-		ast_mutex_lock(&spy->lock);
-		/* Indicate to the spy to stop */
-		spy->status = CHANSPY_STOP;
-		spy->chan = NULL;
-		/* Poke the spy if needed */
-		if (ast_test_flag(spy, CHANSPY_TRIGGER_MODE) != CHANSPY_TRIGGER_NONE)
-			ast_cond_signal(&spy->trigger);
-		ast_mutex_unlock(&spy->lock);
-	}
-
-	/* Print it out while we still have a lock so the structure can't go away (if signalled above) */
-	ast_debug(1, "Spy %s removed from channel %s\n", spy->type, chan->name);
-
-	return;
-}
-
-void ast_channel_spy_stop_by_type(struct ast_channel *chan, const char *type)
-{
-	struct ast_channel_spy *spy = NULL;
-	
-	if (!chan->spies)
-		return;
-
-	AST_LIST_TRAVERSE_SAFE_BEGIN(&chan->spies->list, spy, list) {
-		if ((spy->type == type) && (spy->status == CHANSPY_RUNNING)) {
-			AST_LIST_REMOVE_CURRENT(&chan->spies->list, list);
-			spy_detach(spy, chan);
-		}
-	}
-	AST_LIST_TRAVERSE_SAFE_END
-	spy_cleanup(chan);
-}
-
-void ast_channel_spy_trigger_wait(struct ast_channel_spy *spy)
-{
-	struct timeval tv;
-	struct timespec ts;
-
-	tv = ast_tvadd(ast_tvnow(), ast_samp2tv(50000, 1000));
-	ts.tv_sec = tv.tv_sec;
-	ts.tv_nsec = tv.tv_usec * 1000;
-
-	ast_cond_timedwait(&spy->trigger, &spy->lock, &ts);
-}
-
-void ast_channel_spy_remove(struct ast_channel *chan, struct ast_channel_spy *spy)
-{
-	if (!chan->spies)
-		return;
-
-	AST_LIST_REMOVE(&chan->spies->list, spy, list);
-	spy_detach(spy, chan);
-	spy_cleanup(chan);
-}
-
-void ast_channel_spy_free(struct ast_channel_spy *spy)
-{
-	struct ast_frame *f = NULL;
-
-	if (spy->status == CHANSPY_DONE)
-		return;
-
-	/* Switch status to done in case we get called twice */
-	spy->status = CHANSPY_DONE;
-
-	/* Drop any frames in the queue */
-	while ((f = AST_LIST_REMOVE_HEAD(&spy->write_queue.list, frame_list)))
-		ast_frfree(f);
-	while ((f = AST_LIST_REMOVE_HEAD(&spy->read_queue.list, frame_list)))
-		ast_frfree(f);
-
-	/* Destroy the condition if in use */
-	if (ast_test_flag(spy, CHANSPY_TRIGGER_MODE) != CHANSPY_TRIGGER_NONE)
-		ast_cond_destroy(&spy->trigger);
-
-	/* Destroy our mutex since it is no longer in use */
-	ast_mutex_destroy(&spy->lock);
-
-	return;
-}
-
-static void detach_spies(struct ast_channel *chan)
-{
-	struct ast_channel_spy *spy = NULL;
-
-	if (!chan->spies)
-		return;
-
-	AST_LIST_TRAVERSE_SAFE_BEGIN(&chan->spies->list, spy, list) {
-		AST_LIST_REMOVE_CURRENT(&chan->spies->list, list);
-		spy_detach(spy, chan);
-	}
-	AST_LIST_TRAVERSE_SAFE_END
-
-	spy_cleanup(chan);
-}
-
 /*! \brief Softly hangup a channel, don't lock */
 int ast_softhangup_nolock(struct ast_channel *chan, int cause)
 {
@@ -1474,124 +1279,6 @@ int ast_softhangup(struct ast_channel *chan, int cause)
 	return res;
 }
 
-enum spy_direction {
-	SPY_READ,
-	SPY_WRITE,
-};
-
-#define SPY_QUEUE_SAMPLE_LIMIT 4000			/* half of one second */
-
-static void queue_frame_to_spies(struct ast_channel *chan, struct ast_frame *f, enum spy_direction dir)
-{
-	struct ast_frame *translated_frame = NULL;
-	struct ast_channel_spy *spy;
-	struct channel_spy_trans *trans;
-
-	trans = (dir == SPY_READ) ? &chan->spies->read_translator : &chan->spies->write_translator;
-
-	AST_LIST_TRAVERSE(&chan->spies->list, spy, list) {
-		struct ast_channel_spy_queue *queue;
-		struct ast_frame *duped_fr;
-
-		if (spy->status != CHANSPY_RUNNING)
-			continue;
-
-		ast_mutex_lock(&spy->lock);
-
-		queue = (dir == SPY_READ) ? &spy->read_queue : &spy->write_queue;
-
-		if ((queue->format == AST_FORMAT_SLINEAR) && (f->subclass != AST_FORMAT_SLINEAR)) {
-			if (!translated_frame) {
-				if (trans->path && (trans->last_format != f->subclass)) {
-					ast_translator_free_path(trans->path);
-					trans->path = NULL;
-				}
-				if (!trans->path) {
-					ast_debug(1, "Building translator from %s to SLINEAR for spies on channel %s\n",
-							ast_getformatname(f->subclass), chan->name);
-					if ((trans->path = ast_translator_build_path(AST_FORMAT_SLINEAR, f->subclass)) == NULL) {
-						ast_log(LOG_WARNING, "Cannot build a path from %s to %s\n",
-							ast_getformatname(f->subclass), ast_getformatname(AST_FORMAT_SLINEAR));
-						ast_mutex_unlock(&spy->lock);
-						continue;
-					} else {
-						trans->last_format = f->subclass;
-					}
-				}
-				if (!(translated_frame = ast_translate(trans->path, f, 0))) {
-					ast_log(LOG_ERROR, "Translation to %s failed, dropping frame for spies\n",
-						ast_getformatname(AST_FORMAT_SLINEAR));
-					ast_mutex_unlock(&spy->lock);
-					break;
-				}
-			}
-			duped_fr = ast_frdup(translated_frame);
-		} else if (f->subclass != queue->format) {
-			ast_log(LOG_WARNING, "Spy '%s' on channel '%s' wants format '%s', but frame is '%s', dropping\n",
-				spy->type, chan->name,
-				ast_getformatname(queue->format), ast_getformatname(f->subclass));
-			ast_mutex_unlock(&spy->lock);
-			continue;
-		} else
-			duped_fr = ast_frdup(f);
-
-		AST_LIST_INSERT_TAIL(&queue->list, duped_fr, frame_list);
-
-		queue->samples += f->samples;
-
-		if (queue->samples > SPY_QUEUE_SAMPLE_LIMIT) {
-			if (ast_test_flag(spy, CHANSPY_TRIGGER_MODE) != CHANSPY_TRIGGER_NONE) {
-				switch (ast_test_flag(spy, CHANSPY_TRIGGER_MODE)) {
-				case CHANSPY_TRIGGER_READ:
-					if (dir == SPY_WRITE) {
-						ast_set_flag(spy, CHANSPY_TRIGGER_WRITE);
-						ast_clear_flag(spy, CHANSPY_TRIGGER_READ);
-						ast_debug(1, "Switching spy '%s' on '%s' to write-trigger mode\n",
-							spy->type, chan->name);
-					}
-					break;
-				case CHANSPY_TRIGGER_WRITE:
-					if (dir == SPY_READ) {
-						ast_set_flag(spy, CHANSPY_TRIGGER_READ);
-						ast_clear_flag(spy, CHANSPY_TRIGGER_WRITE);
-						ast_debug(1, "Switching spy '%s' on '%s' to read-trigger mode\n",
-							spy->type, chan->name);
-					}
-					break;
-				}
-				ast_debug(1, "Triggering queue flush for spy '%s' on '%s'\n",
-					spy->type, chan->name);
-				ast_set_flag(spy, CHANSPY_TRIGGER_FLUSH);
-				ast_cond_signal(&spy->trigger);
-			} else {
-				ast_debug(1, "Spy '%s' on channel '%s' %s queue too long, dropping frames\n",
-					spy->type, chan->name, (dir == SPY_READ) ? "read" : "write");
-				while (queue->samples > SPY_QUEUE_SAMPLE_LIMIT) {
-					struct ast_frame *drop = AST_LIST_REMOVE_HEAD(&queue->list, frame_list);
-					queue->samples -= drop->samples;
-					ast_frfree(drop);
-				}
-			}
-		} else {
-			switch (ast_test_flag(spy, CHANSPY_TRIGGER_MODE)) {
-			case CHANSPY_TRIGGER_READ:
-				if (dir == SPY_READ)
-					ast_cond_signal(&spy->trigger);
-				break;
-			case CHANSPY_TRIGGER_WRITE:
-				if (dir == SPY_WRITE)
-					ast_cond_signal(&spy->trigger);
-				break;
-			}
-		}
-
-		ast_mutex_unlock(&spy->lock);
-	}
-
-	if (translated_frame)
-		ast_frfree(translated_frame);
-}
-
 static void free_translation(struct ast_channel *clone)
 {
 	if (clone->writetrans)
@@ -1614,7 +1301,10 @@ int ast_hangup(struct ast_channel *chan)
 	   if someone is going to masquerade as us */
 	ast_channel_lock(chan);
 
-	detach_spies(chan);		/* get rid of spies */
+	if (chan->audiohooks) {
+		ast_audiohook_detach_list(chan->audiohooks);
+		chan->audiohooks = NULL;
+	}
 
 	if (chan->masq) {
 		if (ast_do_masquerade(chan))
@@ -2310,6 +2000,8 @@ static struct ast_frame *__ast_read(struct ast_channel *chan, int dropaudio)
 						chan->emulate_dtmf_duration = AST_DEFAULT_EMULATE_DTMF_DURATION;
 					ast_log(LOG_DTMF, "DTMF begin emulation of '%c' with duration %u queued on %s\n", f->subclass, chan->emulate_dtmf_duration, chan->name);
 				}
+				if (chan->audiohooks)
+					f = ast_audiohook_write_list(chan, chan->audiohooks, AST_AUDIOHOOK_DIRECTION_READ, f);
 			} else {
 				struct timeval now = ast_tvnow();
 				if (ast_test_flag(chan, AST_FLAG_IN_DTMF)) {
@@ -2331,6 +2023,8 @@ static struct ast_frame *__ast_read(struct ast_channel *chan, int dropaudio)
 					ast_log(LOG_DTMF, "DTMF end passthrough '%c' on %s\n", f->subclass, chan->name);
 					chan->dtmf_tv = now;
 				}
+				if (chan->audiohooks)
+					f = ast_audiohook_write_list(chan, chan->audiohooks, AST_AUDIOHOOK_DIRECTION_READ, f);
 			}
 			break;
 		case AST_FRAME_DTMF_BEGIN:
@@ -2392,6 +2086,8 @@ static struct ast_frame *__ast_read(struct ast_channel *chan, int dropaudio)
 					f->subclass = chan->emulate_dtmf_digit;
 					f->len = ast_tvdiff_ms(now, chan->dtmf_tv);
 					chan->dtmf_tv = now;
+					if (chan->audiohooks)
+						f = ast_audiohook_write_list(chan, chan->audiohooks, AST_AUDIOHOOK_DIRECTION_READ, f);
 					ast_log(LOG_DTMF, "DTMF end emulation of '%c' queued on %s\n", f->subclass, chan->name);
 				} else {
 					/* Drop voice frames while we're still in the middle of the digit */
@@ -2406,9 +2102,9 @@ static struct ast_frame *__ast_read(struct ast_channel *chan, int dropaudio)
 				ast_frfree(f);
 				f = &ast_null_frame;
 			} else if ((f->frametype == AST_FRAME_VOICE)) {
-				if (chan->spies)
-					queue_frame_to_spies(chan, f, SPY_READ);
-				
+				/* Send frame to audiohooks if present */
+				if (chan->audiohooks)
+					f = ast_audiohook_write_list(chan, chan->audiohooks, AST_AUDIOHOOK_DIRECTION_READ, f);
 				if (chan->monitor && chan->monitor->read_stream ) {
 					/* XXX what does this do ? */
 #ifndef MONITOR_CONSTANT_DELAY
@@ -2746,6 +2442,8 @@ int ast_write(struct ast_channel *chan, struct ast_frame *fr)
 			chan->tech->indicate(chan, fr->subclass, fr->data, fr->datalen);
 		break;
 	case AST_FRAME_DTMF_BEGIN:
+		if (chan->audiohooks)
+			fr = ast_audiohook_write_list(chan, chan->audiohooks, AST_AUDIOHOOK_DIRECTION_WRITE, fr);
 		send_dtmf_event(chan, "Sent", fr->subclass, "Yes", "No");
 		ast_clear_flag(chan, AST_FLAG_BLOCKING);
 		ast_channel_unlock(chan);
@@ -2754,6 +2452,8 @@ int ast_write(struct ast_channel *chan, struct ast_frame *fr)
 		CHECK_BLOCKING(chan);
 		break;
 	case AST_FRAME_DTMF_END:
+		if (chan->audiohooks)
+			fr = ast_audiohook_write_list(chan, chan->audiohooks, AST_AUDIOHOOK_DIRECTION_WRITE, fr);
 		send_dtmf_event(chan, "Sent", fr->subclass, "No", "Yes");
 		ast_clear_flag(chan, AST_FLAG_BLOCKING);
 		ast_channel_unlock(chan);
@@ -2787,42 +2487,21 @@ int ast_write(struct ast_channel *chan, struct ast_frame *fr)
 		if (chan->tech->write == NULL)
 			break;	/*! \todo XXX should return 0 maybe ? */
 
-		/* If someone is whispering on this channel then we must ensure that we are always getting signed linear frames */
-		if (ast_test_flag(chan, AST_FLAG_WHISPER)) {
-			if (fr->subclass == AST_FORMAT_SLINEAR)
-				f = fr;
-			else {
-				ast_mutex_lock(&chan->whisper->lock);
-				if (chan->writeformat != AST_FORMAT_SLINEAR) {
-					/* Rebuild the translation path and set our write format back to signed linear */
-					chan->whisper->original_format = chan->writeformat;
-					ast_set_write_format(chan, AST_FORMAT_SLINEAR);
-					if (chan->whisper->path)
-						ast_translator_free_path(chan->whisper->path);
-					chan->whisper->path = ast_translator_build_path(AST_FORMAT_SLINEAR, chan->whisper->original_format);
-				}
-				/* Translate frame using the above translation path */
-				f = (chan->whisper->path) ? ast_translate(chan->whisper->path, fr, 0) : fr;
-				ast_mutex_unlock(&chan->whisper->lock);
-			}
-		} else {
-			/* If the frame is in the raw write format, then it's easy... just use the frame - otherwise we will have to translate */
-			if (fr->subclass == chan->rawwriteformat)
-				f = fr;
-			else
-				f = (chan->writetrans) ? ast_translate(chan->writetrans, fr, 0) : fr;
-		}
+		/* If audiohooks are present, write the frame out */
+		if (chan->audiohooks)
+			fr = ast_audiohook_write_list(chan, chan->audiohooks, AST_AUDIOHOOK_DIRECTION_WRITE, fr);
+		
+		/* If the frame is in the raw write format, then it's easy... just use the frame - otherwise we will have to translate */
+		if (fr->subclass == chan->rawwriteformat)
+			f = fr;
+		else
+			f = (chan->writetrans) ? ast_translate(chan->writetrans, fr, 0) : fr;
 
-		/* If we have no frame of audio, then we have to bail out */
-		if (f == NULL) {
+		if (!f) {
 			res = 0;
 			break;
 		}
 
-		/* If spies are on the channel then queue the frame out to them */
-		if (chan->spies)
-			queue_frame_to_spies(chan, f, SPY_WRITE);
-
 		/* If Monitor is running on this channel, then we have to write frames out there too */
 		if (chan->monitor && chan->monitor->write_stream) {
 			/* XXX must explain this code */
@@ -2849,30 +2528,6 @@ int ast_write(struct ast_channel *chan, struct ast_frame *fr)
 					ast_log(LOG_WARNING, "Failed to write data to channel monitor write stream\n");
 			}
 		}
-
-		/* Finally the good part! Write this out to the channel */
-		if (ast_test_flag(chan, AST_FLAG_WHISPER)) {
-			/* frame is assumed to be in SLINEAR, since that is
-			   required for whisper mode */
-			ast_frame_adjust_volume(f, -2);
-			if (ast_slinfactory_available(&chan->whisper->sf) >= f->samples) {
-				short buf[f->samples];
-				struct ast_frame whisper = {
-					.frametype = AST_FRAME_VOICE,
-					.subclass = AST_FORMAT_SLINEAR,
-					.data = buf,
-					.datalen = sizeof(buf),
-					.samples = f->samples,
-				};
-				
-				ast_mutex_lock(&chan->whisper->lock);
-				if (ast_slinfactory_read(&chan->whisper->sf, buf, f->samples))
-					ast_frame_slinear_sum(f, &whisper);
-				ast_mutex_unlock(&chan->whisper->lock);
-			}
-			/* and now put it through the regular translator */
-			f = (chan->writetrans) ? ast_translate(chan->writetrans, f, 0) : f;
-		}
 		res = f ? chan->tech->write(chan, f) : 0;
 		break;
 	case AST_FRAME_NULL:
@@ -3460,8 +3115,6 @@ int ast_do_masquerade(struct ast_channel *original)
 	void *t_pvt;
 	struct ast_callerid tmpcid;
 	struct ast_channel *clone = original->masq;
-	struct ast_channel_spy_list *spy_list = NULL;
-	struct ast_channel_spy *spy = NULL;
 	struct ast_cdr *cdr;
 	int rformat = original->readformat;
 	int wformat = original->writeformat;
@@ -3547,27 +3200,6 @@ int ast_do_masquerade(struct ast_channel *original)
 	original->rawwriteformat = clone->rawwriteformat;
 	clone->rawwriteformat = x;
 
-	/* Swap the spies */
-	spy_list = original->spies;
-	original->spies = clone->spies;
-	clone->spies = spy_list;
-
-	/* Update channel on respective spy lists if present */
-	if (original->spies) {
-		AST_LIST_TRAVERSE(&original->spies->list, spy, list) {
-			ast_mutex_lock(&spy->lock);
-			spy->chan = original;
-			ast_mutex_unlock(&spy->lock);
-		}
-	}
-	if (clone->spies) {
-		AST_LIST_TRAVERSE(&clone->spies->list, spy, list) {
-			ast_mutex_lock(&spy->lock);
-			spy->chan = clone;
-			ast_mutex_unlock(&spy->lock);
-		}
-	}
-
 	/* Save any pending frames on both sides.  Start by counting
 	 * how many we're going to need... */
 	x = 0;
@@ -3632,15 +3264,6 @@ int ast_do_masquerade(struct ast_channel *original)
 
 	ast_app_group_update(clone, original);
 
-	/* move any whisperer over */
-	ast_channel_whisper_stop(original);
-	if (ast_test_flag(clone, AST_FLAG_WHISPER)) {
-		original->whisper = clone->whisper;
-		ast_set_flag(original, AST_FLAG_WHISPER);
-		clone->whisper = NULL;
-		ast_clear_flag(clone, AST_FLAG_WHISPER);
-	}
-
 	/* Move data stores over */
 	if (AST_LIST_FIRST(&clone->datastores))
                 AST_LIST_INSERT_TAIL(&original->datastores, AST_LIST_FIRST(&clone->datastores), entry);
@@ -4152,7 +3775,8 @@ enum ast_bridge_result ast_channel_bridge(struct ast_channel *c0, struct ast_cha
 		    (config->timelimit == 0) &&
 		    (c0->tech->bridge == c1->tech->bridge) &&
 		    !nativefailed && !c0->monitor && !c1->monitor &&
-		    !c0->spies && !c1->spies && !ast_test_flag(&(config->features_callee),AST_FEATURE_REDIRECT) &&
+		    !c0->audiohooks && !c1->audiohooks && 
+		    !ast_test_flag(&(config->features_callee),AST_FEATURE_REDIRECT) &&
 		    !ast_test_flag(&(config->features_caller),AST_FEATURE_REDIRECT) ) {
 			/* Looks like they share a bridge method and nothing else is in the way */
 			ast_set_flag(c0, AST_FLAG_NBRIDGE);
@@ -4525,129 +4149,6 @@ void ast_set_variables(struct ast_channel *chan, struct ast_variable *vars)
 		pbx_builtin_setvar_helper(chan, cur->name, cur->value);	
 }
 
-static void copy_data_from_queue(struct ast_channel_spy_queue *queue, short *buf, unsigned int samples)
-{
-	struct ast_frame *f;
-	int tocopy;
-	int bytestocopy;
-
-	while (samples) {
-		if (!(f = AST_LIST_FIRST(&queue->list))) {
-			ast_log(LOG_ERROR, "Ran out of frames before buffer filled!\n");
-			break;
-		}
-
-		tocopy = (f->samples > samples) ? samples : f->samples;
-		bytestocopy = ast_codec_get_len(queue->format, tocopy);
-		memcpy(buf, f->data, bytestocopy);
-		samples -= tocopy;
-		buf += tocopy;
-		f->samples -= tocopy;
-		f->data += bytestocopy;
-		f->datalen -= bytestocopy;
-		f->offset += bytestocopy;
-		queue->samples -= tocopy;
-
-		if (!f->samples)
-			ast_frfree(AST_LIST_REMOVE_HEAD(&queue->list, frame_list));
-	}
-}
-
-struct ast_frame *ast_channel_spy_read_frame(struct ast_channel_spy *spy, unsigned int samples)
-{
-	struct ast_frame *result;
-	/* buffers are allocated to hold SLINEAR, which is the largest format */
-        short read_buf[samples];
-        short write_buf[samples];
-	struct ast_frame *read_frame;
-	struct ast_frame *write_frame;
-	int need_dup;
-	struct ast_frame stack_read_frame = { .frametype = AST_FRAME_VOICE,
-					      .subclass = spy->read_queue.format,
-					      .data = read_buf,
-					      .samples = samples,
-					      .datalen = ast_codec_get_len(spy->read_queue.format, samples),
-	};
-	struct ast_frame stack_write_frame = { .frametype = AST_FRAME_VOICE,
-					       .subclass = spy->write_queue.format,
-					       .data = write_buf,
-					       .samples = samples,
-					       .datalen = ast_codec_get_len(spy->write_queue.format, samples),
-	};
-
-	/* if a flush has been requested, dump everything in whichever queue is larger */
-	if (ast_test_flag(spy, CHANSPY_TRIGGER_FLUSH)) {
-		if (spy->read_queue.samples > spy->write_queue.samples) {
-			if (ast_test_flag(spy, CHANSPY_READ_VOLADJUST)) {
-				AST_LIST_TRAVERSE(&spy->read_queue.list, result, frame_list)
-					ast_frame_adjust_volume(result, spy->read_vol_adjustment);
-			}
-			result = AST_LIST_FIRST(&spy->read_queue.list);
-			AST_LIST_HEAD_SET_NOLOCK(&spy->read_queue.list, NULL);
-			spy->read_queue.samples = 0;
-		} else {
-			if (ast_test_flag(spy, CHANSPY_WRITE_VOLADJUST)) {
-				AST_LIST_TRAVERSE(&spy->write_queue.list, result, frame_list)
-					ast_frame_adjust_volume(result, spy->write_vol_adjustment);
-			}
-			result = AST_LIST_FIRST(&spy->write_queue.list);
-			AST_LIST_HEAD_SET_NOLOCK(&spy->write_queue.list, NULL);
-			spy->write_queue.samples = 0;
-		}
-		ast_clear_flag(spy, CHANSPY_TRIGGER_FLUSH);
-		return result;
-	}
-
-	if ((spy->read_queue.samples < samples) || (spy->write_queue.samples < samples))
-		return NULL;
-
-	/* short-circuit if both head frames have exactly what we want */
-	if ((AST_LIST_FIRST(&spy->read_queue.list)->samples == samples) &&
-	    (AST_LIST_FIRST(&spy->write_queue.list)->samples == samples)) {
-		read_frame = AST_LIST_REMOVE_HEAD(&spy->read_queue.list, frame_list);
-		write_frame = AST_LIST_REMOVE_HEAD(&spy->write_queue.list, frame_list);
-
-		spy->read_queue.samples -= samples;
-		spy->write_queue.samples -= samples;
-
-		need_dup = 0;
-	} else {
-		copy_data_from_queue(&spy->read_queue, read_buf, samples);
-		copy_data_from_queue(&spy->write_queue, write_buf, samples);
-
-		read_frame = &stack_read_frame;
-		write_frame = &stack_write_frame;
-		need_dup = 1;
-	}
-	
-	if (ast_test_flag(spy, CHANSPY_READ_VOLADJUST))
-		ast_frame_adjust_volume(read_frame, spy->read_vol_adjustment);
-
-	if (ast_test_flag(spy, CHANSPY_WRITE_VOLADJUST))
-		ast_frame_adjust_volume(write_frame, spy->write_vol_adjustment);
-
-	if (ast_test_flag(spy, CHANSPY_MIXAUDIO)) {
-		ast_frame_slinear_sum(read_frame, write_frame);
-
-		if (need_dup)
-			result = ast_frdup(read_frame);
-		else {
-			result = read_frame;
-			ast_frfree(write_frame);
-		}
-	} else {
-		if (need_dup) {
-			result = ast_frdup(read_frame);
-			AST_LIST_NEXT(result, frame_list) = ast_frdup(write_frame);
-		} else {
-			result = read_frame;
-			AST_LIST_NEXT(result, frame_list) = write_frame;
-		}
-	}
-
-	return result;
-}
-
 static void *silence_generator_alloc(struct ast_channel *chan, void *data)
 {
 	/* just store the data pointer in the channel structure */
@@ -4894,46 +4395,3 @@ int ast_say_digits_full(struct ast_channel *chan, int num,
         snprintf(buf, sizeof(buf), "%d", num);
         return ast_say_digit_str_full(chan, buf, ints, lang, audiofd, ctrlfd);
 }
-
-int ast_channel_whisper_start(struct ast_channel *chan)
-{
-	if (chan->whisper)
-		return -1;
-
-	if (!(chan->whisper = ast_calloc(1, sizeof(*chan->whisper))))
-		return -1;
-
-	ast_mutex_init(&chan->whisper->lock);
-	ast_slinfactory_init(&chan->whisper->sf);
-	ast_set_flag(chan, AST_FLAG_WHISPER);
-
-	return 0;
-}
-
-int ast_channel_whisper_feed(struct ast_channel *chan, struct ast_frame *f)
-{
-	if (!chan->whisper)
-		return -1;
-
-	ast_mutex_lock(&chan->whisper->lock);
-	ast_slinfactory_feed(&chan->whisper->sf, f);
-	ast_mutex_unlock(&chan->whisper->lock);
-
-	return 0;
-}
-
-void ast_channel_whisper_stop(struct ast_channel *chan)
-{
-	if (!chan->whisper)
-		return;
-
-	ast_clear_flag(chan, AST_FLAG_WHISPER);
-	if (chan->whisper->path)
-		ast_translator_free_path(chan->whisper->path);
-	if (chan->whisper->original_format && chan->writeformat == AST_FORMAT_SLINEAR)
-		ast_set_write_format(chan, chan->whisper->original_format);
-	ast_slinfactory_destroy(&chan->whisper->sf);
-	ast_mutex_destroy(&chan->whisper->lock);
-	ast_free(chan->whisper);
-	chan->whisper = NULL;
-}
diff --git a/main/slinfactory.c b/main/slinfactory.c
index a42b2b213..038fa0d7b 100644
--- a/main/slinfactory.c
+++ b/main/slinfactory.c
@@ -148,3 +148,21 @@ unsigned int ast_slinfactory_available(const struct ast_slinfactory *sf)
 {
 	return sf->size;
 }
+
+void ast_slinfactory_flush(struct ast_slinfactory *sf)
+{
+	struct ast_frame *fr = NULL;
+
+	if (sf->trans) {
+		ast_translator_free_path(sf->trans);
+		sf->trans = NULL;
+	}
+
+	while ((fr = AST_LIST_REMOVE_HEAD(&sf->queue, frame_list)))
+		ast_frfree(fr);
+
+	sf->size = sf->holdlen = 0;
+	sf->offset = sf->hold;
+
+	return;
+}