[svn-commits] file: branch file/ah r58875 - in /team/file/ah: apps/ include/asterisk/ main/

svn-commits at lists.digium.com svn-commits at lists.digium.com
Tue Mar 13 16:44:53 MST 2007


Author: file
Date: Tue Mar 13 18:44:52 2007
New Revision: 58875

URL: http://svn.digium.com/view/asterisk?view=rev&rev=58875
Log:
Add audiohook architecture. This is basically chanspy cleaned up with a few things done differently. It also implements a new type, manipulate, alongside the old spy and whisper. Currently the spy type is working fine.

Added:
    team/file/ah/include/asterisk/audiohook.h   (with props)
    team/file/ah/main/audiohook.c   (with props)
Modified:
    team/file/ah/apps/app_chanspy.c
    team/file/ah/apps/app_mixmonitor.c
    team/file/ah/include/asterisk/channel.h
    team/file/ah/main/Makefile
    team/file/ah/main/channel.c

Modified: team/file/ah/apps/app_chanspy.c
URL: http://svn.digium.com/view/asterisk/team/file/ah/apps/app_chanspy.c?view=diff&rev=58875&r1=58874&r2=58875
==============================================================================
--- team/file/ah/apps/app_chanspy.c (original)
+++ team/file/ah/apps/app_chanspy.c Tue Mar 13 18:44:52 2007
@@ -40,7 +40,7 @@
 #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 +166,7 @@
 
 struct chanspy_translation_helper {
 	/* spy data */
-	struct ast_channel_spy spy;
+	struct ast_audiohook audiohook;
 	int fd;
 	int volfactor;
 };
@@ -185,15 +185,18 @@
 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_mutex_lock(&csth->audiohook.lock);
+	if (csth->audiohook.status != AST_AUDIOHOOK_STATUS_RUNNING) {
 		/* Channel is already gone more than likely */
+		ast_mutex_unlock(&csth->audiohook.lock);
 		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->audiohook, samples, AST_AUDIOHOOK_DIRECTION_BOTH, AST_FORMAT_SLINEAR);
+
+	ast_mutex_unlock(&csth->audiohook.lock);
 		
 	if (!f)
 		return 0;
@@ -217,50 +220,19 @@
 	.generate = spy_generate, 
 };
 
-static int start_spying(struct ast_channel *chan, struct ast_channel *spychan, struct ast_channel_spy *spy) 
-{
-	int res;
-	struct ast_channel *peer;
+static int start_spying(struct ast_channel *chan, struct ast_channel *spychan, struct ast_audiohook *audiohook) 
+{
+	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);	
 
 	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,
@@ -281,49 +253,12 @@
 		ast_verbose(VERBOSE_PREFIX_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.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.audiohook)) {
+		ast_audiohook_destroy(&csth.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);
 	}
 
 	if (ast_test_flag(flags, OPTION_PRIVATE))
@@ -345,21 +280,12 @@
 	   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.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);
-			ast_frfree(f);
-			continue;
-		}
-
+		
 		res = (f->frametype == AST_FRAME_DTMF) ? f->subclass : 0;
 		ast_frfree(f);
 		if (!res)
@@ -403,36 +329,20 @@
 				*volfactor = -4;
 			if (option_verbose > 2)
 				ast_verbose(VERBOSE_PREFIX_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);
 
+	ast_mutex_lock(&csth.audiohook.lock);
+	ast_audiohook_detach(&csth.audiohook);
+	ast_mutex_unlock(&csth.audiohook.lock);
+
 	/* If a channel still exists on our spy structure then we need to remove ourselves */
-	if (csth.spy.chan) {
-		csth.spy.status = CHANSPY_DONE;
-		ast_channel_lock(csth.spy.chan);
-		ast_channel_spy_remove(csth.spy.chan, &csth.spy);
-		ast_channel_unlock(csth.spy.chan);
-	}
-	ast_channel_spy_free(&csth.spy);
+	ast_audiohook_destroy(&csth.audiohook);
 	
 	if (option_verbose >= 2)
 		ast_verbose(VERBOSE_PREFIX_2 "Done Spying on channel %s\n", name);

Modified: team/file/ah/apps/app_mixmonitor.c
URL: http://svn.digium.com/view/asterisk/team/file/ah/apps/app_mixmonitor.c?view=diff&rev=58875&r1=58874&r2=58875
==============================================================================
--- team/file/ah/apps/app_mixmonitor.c (original)
+++ team/file/ah/apps/app_mixmonitor.c Tue Mar 13 18:44:52 2007
@@ -45,7 +45,7 @@
 #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 @@
 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_OPTION_ARG('W', MUXFLAG_VOLUME, OPT_ARG_VOLUME),
 });
 
-static int startmon(struct ast_channel *chan, struct ast_channel_spy *spy) 
-{
-	struct ast_channel *peer;
-	int res;
+static int startmon(struct ast_channel *chan, struct ast_audiohook *audiohook) 
+{
+	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 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 +152,50 @@
 	if (option_verbose > 1)
 		ast_verbose(VERBOSE_PREFIX_2 "Begin MixMonitor Recording %s\n", mixmonitor->name);
 	
-	ast_mutex_lock(&mixmonitor->spy.lock);
-
-	while (mixmonitor->spy.chan) {
-		struct ast_frame *next;
-		int write;
-
-		ast_channel_spy_trigger_wait(&mixmonitor->spy);
-		
-		if (!mixmonitor->spy.chan || mixmonitor->spy.status != CHANSPY_RUNNING)
+	ast_mutex_lock(&mixmonitor->audiohook.lock);
+
+	while (mixmonitor->audiohook.status == AST_AUDIOHOOK_STATUS_RUNNING) {
+		struct ast_frame *fr = NULL;
+
+		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;
 			}
 		}
-	}
-
-	ast_mutex_unlock(&mixmonitor->spy.lock);
-
-	ast_channel_spy_free(&mixmonitor->spy);
-	
+
+		/* Write out frame */
+		if (fs)
+			ast_writestream(fs, fr);
+
+		/* All done! free it. */
+		ast_frame_free(fr, 0);
+
+	}
+
+	ast_audiohook_detach(&mixmonitor->audiohook);
+
+	ast_mutex_unlock(&mixmonitor->audiohook.lock);
+
+	ast_audiohook_destroy(&mixmonitor->audiohook);
+
 	if (option_verbose > 1)
 		ast_verbose(VERBOSE_PREFIX_2 "End MixMonitor Recording %s\n", mixmonitor->name);
 
@@ -271,27 +259,17 @@
 	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;
-	}
-	ast_mutex_init(&mixmonitor->spy.lock);
-
-	if (startmon(chan, &mixmonitor->spy)) {
+	if (ast_audiohook_init(&mixmonitor->audiohook, AST_AUDIOHOOK_TYPE_SPY, mixmonitor_spy_type)) {
+		free(mixmonitor);
+		return;
+	}
+
+	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);
 		free(mixmonitor);
 		return;
 	}
@@ -390,11 +368,11 @@
 	struct ast_module_user *u;
 
 	u = ast_module_user_add(chan);
-
+/*
 	ast_channel_lock(chan);
 	ast_channel_spy_stop_by_type(chan, mixmonitor_spy_type);
 	ast_channel_unlock(chan);
-
+*/
 	ast_module_user_remove(u);
 
 	return 0;
@@ -411,12 +389,12 @@
 		ast_cli(fd, "No channel matching '%s' found.\n", argv[2]);
 		return RESULT_SUCCESS;
 	}
-
+/*
 	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);
 
 	return RESULT_SUCCESS;

Added: team/file/ah/include/asterisk/audiohook.h
URL: http://svn.digium.com/view/asterisk/team/file/ah/include/asterisk/audiohook.h?view=auto&rev=58875
==============================================================================
--- team/file/ah/include/asterisk/audiohook.h (added)
+++ team/file/ah/include/asterisk/audiohook.h Tue Mar 13 18:44:52 2007
@@ -1,0 +1,154 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 1999 - 2007, Digium, Inc.
+ *
+ * Joshua Colp <jcolp at digium.com>
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*! \file
+ * \brief 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,
+	AST_AUDIOHOOK_TYPE_WHISPER,
+	AST_AUDIOHOOK_TYPE_MANIPULATE,
+};
+
+enum ast_audiohook_status {
+	AST_AUDIOHOOK_STATUS_NEW = 0,
+	AST_AUDIOHOOK_STATUS_RUNNING,
+	AST_AUDIOHOOK_STATUS_SHUTDOWN,
+	AST_AUDIOHOOK_STATUS_DONE,
+};
+
+enum ast_audiohook_direction {
+	AST_AUDIOHOOK_DIRECTION_READ = 0,
+	AST_AUDIOHOOK_DIRECTION_WRITE,
+	AST_AUDIOHOOK_DIRECTION_BOTH,
+};
+
+enum ast_audiohook_flags {
+	AST_AUDIOHOOK_TRIGGER_MODE = (3 << 0),
+	AST_AUDIOHOOK_TRIGGER_READ = (1 << 0),
+	AST_AUDIOHOOK_TRIGGER_WRITE = (2 << 0),
+};
+
+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_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 Writes a frame out to spies
+ * \param list List of audiohooks
+ * \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_spy_write_frame(struct ast_audiohook_list *audiohook_list, enum ast_audiohook_direction direction, struct ast_frame *frame);
+
+/*! \brief Merges frame with data from whisper sources
+ * \param list List of audiohooks
+ * \param frame Frame to merge data into
+ * \return Returns 0 on success, -1 on failure
+ */
+int ast_audiohook_whisper_read_frame(struct ast_audiohook_list *audiohook_list, struct ast_frame *frame);
+
+/*! \brief Passes frame to manipulate audiohook to be modified
+ * \param list List of audiohooks
+ * \param frame Frame to be manipulated
+ * \return Returns 0 on success, -1 on failure
+ */
+int ast_audiohook_manipulate_frame(struct ast_audiohook_list *audiohook_list, 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);
+
+#if defined(__cplusplus) || defined(c_plusplus)
+}
+#endif
+
+#endif /* _ASTERISK_AUDIOHOOK_H */

Propchange: team/file/ah/include/asterisk/audiohook.h
------------------------------------------------------------------------------
    svn:eol-style = native

Propchange: team/file/ah/include/asterisk/audiohook.h
------------------------------------------------------------------------------
    svn:keywords = Author Date Id Revision

Propchange: team/file/ah/include/asterisk/audiohook.h
------------------------------------------------------------------------------
    svn:mime-type = text/plain

Modified: team/file/ah/include/asterisk/channel.h
URL: http://svn.digium.com/view/asterisk/team/file/ah/include/asterisk/channel.h?view=diff&rev=58875&r1=58874&r2=58875
==============================================================================
--- team/file/ah/include/asterisk/channel.h (original)
+++ team/file/ah/include/asterisk/channel.h Tue Mar 13 18:44:52 2007
@@ -314,9 +314,6 @@
 	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.
@@ -479,8 +476,8 @@
 	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  */

Modified: team/file/ah/main/Makefile
URL: http://svn.digium.com/view/asterisk/team/file/ah/main/Makefile?view=diff&rev=58875&r1=58874&r2=58875
==============================================================================
--- team/file/ah/main/Makefile (original)
+++ team/file/ah/main/Makefile Tue Mar 13 18:44:52 2007
@@ -26,7 +26,7 @@
 	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
+	strcompat.o threadstorage.o dial.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

Added: team/file/ah/main/audiohook.c
URL: http://svn.digium.com/view/asterisk/team/file/ah/main/audiohook.c?view=auto&rev=58875
==============================================================================
--- team/file/ah/main/audiohook.c (added)
+++ team/file/ah/main/audiohook.c Tue Mar 13 18:44:52 2007
@@ -1,0 +1,425 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 1999 - 2007, Digium, Inc.
+ *
+ * Joshua Colp <jcolp at digium.com>
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*! \file
+ *
+ * \brief Audiohooks Architecture
+ *
+ * \author Joshua Colp <jcolp at 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_list {
+	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 */
+	audiohook->type = type;
+
+	/* Initialize lock that protects our audiohook */
+	ast_mutex_init(&audiohook->lock);
+	ast_cond_init(&audiohook->trigger, NULL);
+
+	/* Initialize our factories */
+	ast_slinfactory_init(&audiohook->read_factory);
+	/* Manipulate type only needs one factory */
+	if (type != AST_AUDIOHOOK_TYPE_MANIPULATE)
+		ast_slinfactory_init(&audiohook->write_factory);
+
+	/* 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 */
+	ast_slinfactory_destroy(&audiohook->read_factory);
+	if (audiohook->type != AST_AUDIOHOOK_TYPE_MANIPULATE)
+		ast_slinfactory_destroy(&audiohook->write_factory);
+
+	/* 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);
+	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;
+
+	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;
+	} 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;
+	} 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(AST_FORMAT_SLINEAR, format))) {
+				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);
+	}
+
+	/* 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);
+
+	/* 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)
+{
+	struct ast_audiohook *audiohook = NULL;
+
+	/* Drop any spies */
+	AST_LIST_TRAVERSE_SAFE_BEGIN(&audiohook_list->spy_list, audiohook, list) {
+		ast_mutex_lock(&audiohook->lock);
+		AST_LIST_REMOVE_CURRENT(&audiohook_list->spy_list, list);
+		audiohook->status = AST_AUDIOHOOK_STATUS_DONE;
+		ast_cond_signal(&audiohook->trigger);
+		ast_mutex_unlock(&audiohook->lock);
+	}
+	AST_LIST_TRAVERSE_SAFE_END
+
+	/* Drop any whispering sources */
+	AST_LIST_TRAVERSE_SAFE_BEGIN(&audiohook_list->whisper_list, audiohook, list) {
+		ast_mutex_lock(&audiohook->lock);
+		AST_LIST_REMOVE_CURRENT(&audiohook_list->whisper_list, list);
+		audiohook->status = AST_AUDIOHOOK_STATUS_DONE;
+		ast_cond_signal(&audiohook->trigger);
+		ast_mutex_unlock(&audiohook->lock);
+	}
+	AST_LIST_TRAVERSE_SAFE_END
+
+	/* Drop any manipulaters */
+	AST_LIST_TRAVERSE_SAFE_BEGIN(&audiohook_list->manipulate_list, audiohook, list) {
+		ast_mutex_lock(&audiohook->lock);
+		AST_LIST_REMOVE_CURRENT(&audiohook_list->manipulate_list, list);
+		audiohook->status = AST_AUDIOHOOK_STATUS_DONE;
+		ast_cond_signal(&audiohook->trigger);
+		ast_mutex_unlock(&audiohook->lock);
+	}
+	AST_LIST_TRAVERSE_SAFE_END
+
+	/* Free ourselves */
+	free(audiohook_list);
+
+	return 0;
+}
+
+/*! \brief Writes a frame out to spies
+ * \param list List of audiohooks
+ * \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_spy_write_frame(struct ast_audiohook_list *audiohook_list, enum ast_audiohook_direction direction, struct ast_frame *frame)
+{
+	struct ast_audiohook *audiohook = NULL;
+
+	/* Write out frame to each spy */
+	AST_LIST_TRAVERSE_SAFE_BEGIN(&audiohook_list->spy_list, audiohook, list) {
+		ast_mutex_lock(&audiohook->lock);
+		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_mutex_unlock(&audiohook->lock);
+			continue;
+		}
+		ast_audiohook_write_frame(audiohook, direction, frame);
+		ast_mutex_unlock(&audiohook->lock);
+	}
+	AST_LIST_TRAVERSE_SAFE_END
+
+	return 0;
+}
+
+/*! \brief Merges frame with data from whisper sources
+ * \param list List of audiohooks
+ * \param frame Frame to merge data into
+ * \return Returns 0 on success, -1 on failure
+ */
+int ast_audiohook_whisper_read_frame(struct ast_audiohook_list *audiohook_list, struct ast_frame *frame)
+{
+	struct ast_audiohook *audiohook = NULL;
+
+	AST_LIST_TRAVERSE_SAFE_BEGIN(&audiohook_list->whisper_list, audiohook, list) {
+		ast_mutex_lock(&audiohook->lock);
+		/* If the audiohook wants to detach, let it go */
+		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_mutex_unlock(&audiohook->lock);
+			continue;
+		}
+		ast_mutex_unlock(&audiohook->lock);
+	}
+	AST_LIST_TRAVERSE_SAFE_END
+
+	return 0;
+}
+
+/*! \brief Passes frame to manipulate audiohook to be modified
+ * \param list List of audiohooks
+ * \param frame Frame to be manipulated
+ * \return Returns 0 on success, -1 on failure
+ */
+int ast_audiohook_manipulate_frame(struct ast_audiohook_list *audiohook_list, struct ast_frame *frame)
+{
+	struct ast_audiohook *audiohook = NULL;
+
+	AST_LIST_TRAVERSE_SAFE_BEGIN(&audiohook_list->manipulate_list, audiohook, list) {
+		ast_mutex_lock(&audiohook->lock);
+		/* If the audiohook wants to detach, let it go */
+		if (audiohook->status != AST_AUDIOHOOK_STATUS_RUNNING) {
+			AST_LIST_REMOVE_CURRENT(&audiohook_list->manipulate_list, list);
+			audiohook->status = AST_AUDIOHOOK_STATUS_DONE;
+			ast_cond_signal(&audiohook->trigger);
+			ast_mutex_unlock(&audiohook->lock);
+			continue;
+		}
+		ast_mutex_unlock(&audiohook->lock);
+	}
+	AST_LIST_TRAVERSE_SAFE_END
+
+	return 0;
+}
+
+/*! \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;
+}

Propchange: team/file/ah/main/audiohook.c
------------------------------------------------------------------------------
    svn:eol-style = native

Propchange: team/file/ah/main/audiohook.c
------------------------------------------------------------------------------
    svn:keywords = Author Date Id Revision

Propchange: team/file/ah/main/audiohook.c
------------------------------------------------------------------------------
    svn:mime-type = text/plain

Modified: team/file/ah/main/channel.c
URL: http://svn.digium.com/view/asterisk/team/file/ah/main/channel.c?view=diff&rev=58875&r1=58874&r2=58875
==============================================================================
--- team/file/ah/main/channel.c (original)
+++ team/file/ah/main/channel.c Tue Mar 13 18:44:52 2007
@@ -65,6 +65,7 @@
 #include "asterisk/sha1.h"
 #include "asterisk/threadstorage.h"
 #include "asterisk/slinfactory.h"
+#include "asterisk/audiohook.h"
 
 /* uncomment if you have problems with 'monitoring' synchronized files */
 #if 0
@@ -1242,6 +1243,11 @@
 	/* Don't actually hang up a channel that will masquerade as someone else, or
 	   if someone is going to masquerade as us */
 	ast_channel_lock(chan);
+
+	if (chan->audiohooks) {
+		ast_audiohook_detach_list(chan->audiohooks);
+		chan->audiohooks = NULL;
+	}
 
 	if (chan->masq) {
 		if (ast_do_masquerade(chan))
@@ -1908,6 +1914,9 @@
 				ast_frfree(f);
 				f = &ast_null_frame;
 			} else {
+				/* Send frame to audiohooks if present */
+				if (chan->audiohooks)
+					ast_audiohook_spy_write_frame(chan->audiohooks, AST_AUDIOHOOK_DIRECTION_READ, f);
 				if (chan->monitor && chan->monitor->read_stream ) {
 					/* XXX what does this do ? */
 #ifndef MONITOR_CONSTANT_DELAY
@@ -2296,6 +2305,10 @@
 	case AST_FRAME_VOICE:
 		if (chan->tech->write == NULL)
 			break;	/*! \todo XXX should return 0 maybe ? */
+
+		/* If audiohooks are present, write the frame out */
+		if (chan->audiohooks)
+			ast_audiohook_spy_write_frame(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)
@@ -3576,7 +3589,8 @@
 		    (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) &&

[... 4 lines stripped ...]


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