audiohook.c

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/*
 * Asterisk -- An open source telephony toolkit.
 *
 * Copyright (C) 1999 - 2007, Digium, Inc.
 *
 * Joshua Colp <[email protected]>
 *
 * 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 <[email protected]>
 */

/*** MODULEINFO
   <support_level>core</support_level>
 ***/

#include "asterisk.h"

#include <signal.h>

#include "asterisk/channel.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"
#include "asterisk/format_cache.h"

#define AST_AUDIOHOOK_SYNC_TOLERANCE 100 /*!< Tolerance in milliseconds for audiohooks synchronization */
#define AST_AUDIOHOOK_SMALL_QUEUE_TOLERANCE 100 /*!< When small queue is enabled, this is the maximum amount of audio that can remain queued at a time. */
#define AST_AUDIOHOOK_LONG_QUEUE_TOLERANCE 500 /*!< Otheriwise we still don't want the queue to grow indefinitely */

#define DEFAULT_INTERNAL_SAMPLE_RATE 8000

struct ast_audiohook_translate {
   struct ast_trans_pvt *trans_pvt;
   struct ast_format *format;
};

struct ast_audiohook_list {
   /* If all the audiohooks in this list are capable
    * of processing slinear at any sample rate, this
    * variable will be set and the sample rate will
    * be preserved during ast_audiohook_write_list()*/
   int native_slin_compatible;
   int list_internal_samp_rate;/*!< Internal sample rate used when writing to the 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;
};

static int audiohook_set_internal_rate(struct ast_audiohook *audiohook, int rate, int reset)
{
   struct ast_format *slin;

   if (audiohook->hook_internal_samp_rate == rate) {
      return 0;
   }

   audiohook->hook_internal_samp_rate = rate;

   slin = ast_format_cache_get_slin_by_rate(rate);

   /* Setup the factories that are needed for this audiohook type */
   switch (audiohook->type) {
   case AST_AUDIOHOOK_TYPE_SPY:
   case AST_AUDIOHOOK_TYPE_WHISPER:
      if (reset) {
         ast_slinfactory_destroy(&audiohook->read_factory);
         ast_slinfactory_destroy(&audiohook->write_factory);
      }
      ast_slinfactory_init_with_format(&audiohook->read_factory, slin);
      ast_slinfactory_init_with_format(&audiohook->write_factory, slin);
      break;
   default:
      break;
   }

   return 0;
}

/*! \brief Initialize an audiohook structure
 *
 * \param audiohook Audiohook structure
 * \param type
 * \param source, init_flags
 *
 * \return Returns 0 on success, -1 on failure
 */
int ast_audiohook_init(struct ast_audiohook *audiohook, enum ast_audiohook_type type, const char *source, enum ast_audiohook_init_flags init_flags)
{
   /* 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);

   audiohook->init_flags = init_flags;

   /* initialize internal rate at 8khz, this will adjust if necessary */
   audiohook_set_internal_rate(audiohook, DEFAULT_INTERNAL_SAMPLE_RATE, 0);

   /* Since we are just starting out... this audiohook is new */
   ast_audiohook_update_status(audiohook, 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:
   case AST_AUDIOHOOK_TYPE_WHISPER:
      ast_slinfactory_destroy(&audiohook->read_factory);
      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);

   ao2_cleanup(audiohook->format);

   /* Lock and trigger be gone! */
   ast_cond_destroy(&audiohook->trigger);
   ast_mutex_destroy(&audiohook->lock);

   return 0;
}

#define SHOULD_MUTE(hook, dir) \
   ((ast_test_flag(hook, AST_AUDIOHOOK_MUTE_READ) && (dir == AST_AUDIOHOOK_DIRECTION_READ)) || \
   (ast_test_flag(hook, AST_AUDIOHOOK_MUTE_WRITE) && (dir == AST_AUDIOHOOK_DIRECTION_WRITE)) || \
   (ast_test_flag(hook, AST_AUDIOHOOK_MUTE_READ | AST_AUDIOHOOK_MUTE_WRITE) == (AST_AUDIOHOOK_MUTE_READ | AST_AUDIOHOOK_MUTE_WRITE)))

/*! \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);
   struct ast_slinfactory *other_factory = (direction == AST_AUDIOHOOK_DIRECTION_READ ? &audiohook->write_factory : &audiohook->read_factory);
   struct timeval *rwtime = (direction == AST_AUDIOHOOK_DIRECTION_READ ? &audiohook->read_time : &audiohook->write_time), previous_time = *rwtime;
   int our_factory_samples;
   int our_factory_ms;
   int other_factory_samples;
   int other_factory_ms;

   /* Update last feeding time to be current */
   *rwtime = ast_tvnow();

   our_factory_samples = ast_slinfactory_available(factory);
   our_factory_ms = ast_tvdiff_ms(*rwtime, previous_time) + (our_factory_samples / (audiohook->hook_internal_samp_rate / 1000));
   other_factory_samples = ast_slinfactory_available(other_factory);
   other_factory_ms = other_factory_samples / (audiohook->hook_internal_samp_rate / 1000);

   if (ast_test_flag(audiohook, AST_AUDIOHOOK_TRIGGER_SYNC) && (our_factory_ms - other_factory_ms > AST_AUDIOHOOK_SYNC_TOLERANCE)) {
      ast_debug(1, "Flushing audiohook %p so it remains in sync\n", audiohook);
      ast_slinfactory_flush(factory);
      ast_slinfactory_flush(other_factory);
   }

   if (ast_test_flag(audiohook, AST_AUDIOHOOK_SMALL_QUEUE) && ((our_factory_ms > AST_AUDIOHOOK_SMALL_QUEUE_TOLERANCE) || (other_factory_ms > AST_AUDIOHOOK_SMALL_QUEUE_TOLERANCE))) {
      ast_debug(1, "Audiohook %p has stale audio in its factories. Flushing them both\n", audiohook);
      ast_slinfactory_flush(factory);
      ast_slinfactory_flush(other_factory);
   } else if ((our_factory_ms > AST_AUDIOHOOK_LONG_QUEUE_TOLERANCE) || (other_factory_ms > AST_AUDIOHOOK_LONG_QUEUE_TOLERANCE)) {
      ast_debug(1, "Audiohook %p has stale audio in its factories. Flushing them both\n", audiohook);
      ast_slinfactory_flush(factory);
      ast_slinfactory_flush(other_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 */
   if ((ast_test_flag(audiohook, AST_AUDIOHOOK_TRIGGER_MODE) == AST_AUDIOHOOK_TRIGGER_READ) && (direction == AST_AUDIOHOOK_DIRECTION_READ)) {
      ast_cond_signal(&audiohook->trigger);
   } else if ((ast_test_flag(audiohook, AST_AUDIOHOOK_TRIGGER_MODE) == AST_AUDIOHOOK_TRIGGER_WRITE) && (direction == AST_AUDIOHOOK_DIRECTION_WRITE)) {
      ast_cond_signal(&audiohook->trigger);
   } else if (ast_test_flag(audiohook, AST_AUDIOHOOK_TRIGGER_SYNC)) {
      ast_cond_signal(&audiohook->trigger);
   }

   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.format = ast_format_cache_get_slin_by_rate(audiohook->hook_internal_samp_rate),
      .data.ptr = 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 (SHOULD_MUTE(audiohook, direction)) {
      /* Swap frame data for zeros if mute is required */
      ast_frame_clear(&frame);
   } else if (vol) {
      /* If a volume adjustment needs to be applied apply it */
      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, struct ast_frame **read_reference, struct ast_frame **write_reference)
{
   int count;
   int usable_read;
   int usable_write;
   short adjust_value;
   short buf1[samples];
   short buf2[samples];
   short *read_buf = NULL;
   short *write_buf = NULL;
   struct ast_frame frame = {
      .frametype = AST_FRAME_VOICE,
      .datalen = sizeof(buf1),
      .samples = samples,
   };

   /* Make sure both factories have the required samples */
   usable_read = (ast_slinfactory_available(&audiohook->read_factory) >= samples ? 1 : 0);
   usable_write = (ast_slinfactory_available(&audiohook->write_factory) >= samples ? 1 : 0);

   if (!usable_read && !usable_write) {
      /* If both factories are unusable bail out */
      ast_debug(1, "Read factory %p and write factory %p both fail to provide %zu samples\n", &audiohook->read_factory, &audiohook->write_factory, samples);
      return NULL;
   }

   /* If we want to provide only a read factory make sure we aren't waiting for other audio */
   if (usable_read && !usable_write && (ast_tvdiff_ms(ast_tvnow(), audiohook->write_time) < (samples/8)*2)) {
      ast_debug(3, "Write factory %p was pretty quick last time, waiting for them.\n", &audiohook->write_factory);
      return NULL;
   }

   /* If we want to provide only a write factory make sure we aren't waiting for other audio */
   if (usable_write && !usable_read && (ast_tvdiff_ms(ast_tvnow(), audiohook->read_time) < (samples/8)*2)) {
      ast_debug(3, "Read factory %p was pretty quick last time, waiting for them.\n", &audiohook->read_factory);
      return NULL;
   }

   /* Start with the read factory... if there are enough samples, read them in */
   if (usable_read) {
      if (ast_slinfactory_read(&audiohook->read_factory, buf1, samples)) {
         read_buf = buf1;

         if ((ast_test_flag(audiohook, AST_AUDIOHOOK_MUTE_READ))) {
            /* Clear the frame data if we are muting */
            memset(buf1, 0, sizeof(buf1));
         } else if (audiohook->options.read_volume) {
            /* Adjust read volume if need be */
            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 {
      ast_debug(1, "Failed to get %d samples from read factory %p\n", (int)samples, &audiohook->read_factory);
   }

   /* Move on to the write factory... if there are enough samples, read them in */
   if (usable_write) {
      if (ast_slinfactory_read(&audiohook->write_factory, buf2, samples)) {
         write_buf = buf2;

         if ((ast_test_flag(audiohook, AST_AUDIOHOOK_MUTE_WRITE))) {
            /* Clear the frame data if we are muting */
            memset(buf2, 0, sizeof(buf2));
         } else if (audiohook->options.write_volume) {
            /* Adjust write volume if need be */
            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 {
      ast_debug(1, "Failed to get %d samples from write factory %p\n", (int)samples, &audiohook->write_factory);
   }

   frame.subclass.format = ast_format_cache_get_slin_by_rate(audiohook->hook_internal_samp_rate);

   /* Should we substitute silence if one side lacks audio? */
   if ((ast_test_flag(audiohook, AST_AUDIOHOOK_SUBSTITUTE_SILENCE))) {
      if (read_reference && !read_buf && write_buf) {
         read_buf = buf1;
         memset(buf1, 0, sizeof(buf1));
      } else if (write_reference && read_buf && !write_buf) {
         write_buf = buf2;
         memset(buf2, 0, sizeof(buf2));
      }
   }

   /* Basically we figure out which buffer to use... and if mixing can be done here */
   if (read_buf && read_reference) {
      frame.data.ptr = read_buf;
      *read_reference = ast_frdup(&frame);
   }
   if (write_buf && write_reference) {
      frame.data.ptr = write_buf;
      *write_reference = ast_frdup(&frame);
   }

   /* Make the correct buffer part of the built frame, so it gets duplicated. */
   if (read_buf) {
      frame.data.ptr = read_buf;
      if (write_buf) {
         for (count = 0; count < samples; count++) {
            ast_slinear_saturated_add(read_buf++, write_buf++);
         }
      }
   } else if (write_buf) {
      frame.data.ptr = write_buf;
   } else {
      return NULL;
   }

   /* Yahoo, a combined copy of the audio! */
   return ast_frdup(&frame);
}

static struct ast_frame *audiohook_read_frame_helper(struct ast_audiohook *audiohook, size_t samples, enum ast_audiohook_direction direction, struct ast_format *format, struct ast_frame **read_reference, struct ast_frame **write_reference)
{
   struct ast_frame *read_frame = NULL, *final_frame = NULL;
   struct ast_format *slin;

   /*
    * Update the rate if compatibility mode is turned off or if it is
    * turned on and the format rate is higher than the current rate.
    *
    * This makes it so any unnecessary rate switching/resetting does
    * not take place and also any associated audiohook_list's internal
    * sample rate maintains the highest sample rate between hooks.
    */
   if (!ast_test_flag(audiohook, AST_AUDIOHOOK_COMPATIBLE) ||
       (ast_test_flag(audiohook, AST_AUDIOHOOK_COMPATIBLE) &&
         ast_format_get_sample_rate(format) > audiohook->hook_internal_samp_rate)) {
      audiohook_set_internal_rate(audiohook, ast_format_get_sample_rate(format), 1);
   }

   /* If the sample rate of the requested format differs from that of the underlying audiohook
    * sample rate determine how many samples we actually need to get from the audiohook. This
    * needs to occur as the signed linear factory stores them at the rate of the audiohook.
    * We do this by determining the duration of audio they've requested and then determining
    * how many samples that would be in the audiohook format.
    */
   if (ast_format_get_sample_rate(format) != audiohook->hook_internal_samp_rate) {
      samples = (audiohook->hook_internal_samp_rate / 1000) * (samples / (ast_format_get_sample_rate(format) / 1000));
   }

   if (!(read_frame = (direction == AST_AUDIOHOOK_DIRECTION_BOTH ?
      audiohook_read_frame_both(audiohook, samples, read_reference, write_reference) :
      audiohook_read_frame_single(audiohook, samples, direction)))) {
      return NULL;
   }

   slin = ast_format_cache_get_slin_by_rate(audiohook->hook_internal_samp_rate);

   /* If they don't want signed linear back out, we'll have to send it through the translation path */
   if (ast_format_cmp(format, slin) != AST_FORMAT_CMP_EQUAL) {
      /* Rebuild translation path if different format then previously */
      if (ast_format_cmp(format, audiohook->format) == AST_FORMAT_CMP_NOT_EQUAL) {
         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, slin))) {
            ast_frfree(read_frame);
            return NULL;
         }
         ao2_replace(audiohook->format, format);
      }
      /* 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 Reads a frame in from the audiohook structure
 * \param audiohook Audiohook structure
 * \param samples Number of samples wanted in requested output format
 * \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, struct ast_format *format)
{
   return audiohook_read_frame_helper(audiohook, samples, direction, format, NULL, NULL);
}

/*! \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
 * \param read_frame frame pointer for copying read frame data
 * \param write_frame frame pointer for copying write frame data
 * \return Returns frame on success, NULL on failure
 */
struct ast_frame *ast_audiohook_read_frame_all(struct ast_audiohook *audiohook, size_t samples, struct ast_format *format, struct ast_frame **read_frame, struct ast_frame **write_frame)
{
   return audiohook_read_frame_helper(audiohook, samples, AST_AUDIOHOOK_DIRECTION_BOTH, format, read_frame, write_frame);
}

static void audiohook_list_set_samplerate_compatibility(struct ast_audiohook_list *audiohook_list)
{
   struct ast_audiohook *ah = NULL;

   /*
    * Anytime the samplerate compatibility is set (attach/remove an audiohook) the
    * list's internal sample rate needs to be reset so that the next time processing
    * through write_list, if needed, it will get updated to the correct rate.
    *
    * A list's internal rate always chooses the higher between its own rate and a
    * given rate. If the current rate is being driven by an audiohook that wanted a
    * higher rate then when this audiohook is removed the list's rate would remain
    * at that level when it should be lower, and with no way to lower it since any
    * rate compared against it would be lower.
    *
    * By setting it back to the lowest rate it can recalulate the new highest rate.
    */
   audiohook_list->list_internal_samp_rate = DEFAULT_INTERNAL_SAMPLE_RATE;

   audiohook_list->native_slin_compatible = 1;
   AST_LIST_TRAVERSE(&audiohook_list->manipulate_list, ah, list) {
      if (!(ah->init_flags & AST_AUDIOHOOK_MANIPULATE_ALL_RATES)) {
         audiohook_list->native_slin_compatible = 0;
         return;
      }
   }
}

/*! \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);

   /* Don't allow an audiohook to be attached to a channel that is already hung up.
    * The hang up process is what actually notifies the audiohook that it should
    * stop.
    */
   if (ast_test_flag(ast_channel_flags(chan), AST_FLAG_ZOMBIE)) {
      ast_channel_unlock(chan);
      return -1;
   }

   if (!ast_channel_audiohooks(chan)) {
      struct ast_audiohook_list *ahlist;
      /* Whoops... allocate a new structure */
      if (!(ahlist = ast_calloc(1, sizeof(*ahlist)))) {
         ast_channel_unlock(chan);
         return -1;
      }
      ast_channel_audiohooks_set(chan, ahlist);
      AST_LIST_HEAD_INIT_NOLOCK(&ast_channel_audiohooks(chan)->spy_list);
      AST_LIST_HEAD_INIT_NOLOCK(&ast_channel_audiohooks(chan)->whisper_list);
      AST_LIST_HEAD_INIT_NOLOCK(&ast_channel_audiohooks(chan)->manipulate_list);
      /* This sample rate will adjust as necessary when writing to the list. */
      ast_channel_audiohooks(chan)->list_internal_samp_rate = DEFAULT_INTERNAL_SAMPLE_RATE;
   }

   /* Drop into respective list */
   if (audiohook->type == AST_AUDIOHOOK_TYPE_SPY) {
      AST_LIST_INSERT_TAIL(&ast_channel_audiohooks(chan)->spy_list, audiohook, list);
   } else if (audiohook->type == AST_AUDIOHOOK_TYPE_WHISPER) {
      AST_LIST_INSERT_TAIL(&ast_channel_audiohooks(chan)->whisper_list, audiohook, list);
   } else if (audiohook->type == AST_AUDIOHOOK_TYPE_MANIPULATE) {
      AST_LIST_INSERT_TAIL(&ast_channel_audiohooks(chan)->manipulate_list, audiohook, list);
   }

   /*
    * Initialize the audiohook's rate to the default. If it needs to be,
    * it will get updated later.
    */
   audiohook_set_internal_rate(audiohook, DEFAULT_INTERNAL_SAMPLE_RATE, 1);
   audiohook_list_set_samplerate_compatibility(ast_channel_audiohooks(chan));

   /* Change status over to running since it is now attached */
   ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_RUNNING);

   if (ast_channel_is_bridged(chan)) {
      ast_channel_set_unbridged_nolock(chan, 1);
   }

   ast_channel_unlock(chan);

   return 0;
}

/*! \brief Update audiohook's status
 * \param audiohook Audiohook structure
 * \param status Audiohook status enum
 *
 * \note once status is updated to DONE, this function can not be used to set the
 * status back to any other setting.  Setting DONE effectively locks the status as such.
 */

void ast_audiohook_update_status(struct ast_audiohook *audiohook, enum ast_audiohook_status status)
{
   ast_audiohook_lock(audiohook);
   if (audiohook->status != AST_AUDIOHOOK_STATUS_DONE) {
      audiohook->status = status;
      ast_cond_signal(&audiohook->trigger);
   }
   ast_audiohook_unlock(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)
{
   if (audiohook->status == AST_AUDIOHOOK_STATUS_NEW || audiohook->status == AST_AUDIOHOOK_STATUS_DONE) {
      return 0;
   }

   ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_SHUTDOWN);

   while (audiohook->status != AST_AUDIOHOOK_STATUS_DONE) {
      ast_audiohook_trigger_wait(audiohook);
   }

   return 0;
}

void ast_audiohook_detach_list(struct ast_audiohook_list *audiohook_list)
{
   int i;
   struct ast_audiohook *audiohook;

   if (!audiohook_list) {
      return;
   }

   /* Drop any spies */
   while ((audiohook = AST_LIST_REMOVE_HEAD(&audiohook_list->spy_list, list))) {
      ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_DONE);
   }

   /* Drop any whispering sources */
   while ((audiohook = AST_LIST_REMOVE_HEAD(&audiohook_list->whisper_list, list))) {
      ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_DONE);
   }

   /* Drop any manipulaters */
   while ((audiohook = AST_LIST_REMOVE_HEAD(&audiohook_list->manipulate_list, list))) {
      ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_DONE);
      audiohook->manipulate_callback(audiohook, NULL, NULL, 0);
   }

   /* 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);
         ao2_cleanup(audiohook_list->in_translate[i].format);
      }
      if (audiohook_list->out_translate[i].trans_pvt) {
         ast_translator_free_path(audiohook_list->out_translate[i].trans_pvt);
         ao2_cleanup(audiohook_list->in_translate[i].format);
      }
   }

   /* Free ourselves */
   ast_free(audiohook_list);
}

/*! \brief find an audiohook based on its source
 * \param audiohook_list The list of audiohooks to search in
 * \param source The source of the audiohook we wish to find
 * \return Return the corresponding audiohook or NULL if it cannot be found.
 */
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;
}

static void audiohook_move(struct ast_channel *old_chan, struct ast_channel *new_chan, struct ast_audiohook *audiohook)
{
   enum ast_audiohook_status oldstatus;

   /* By locking both channels and the audiohook, we can assure that
    * another thread will not have a chance to read the audiohook's status
    * as done, even though ast_audiohook_remove signals the trigger
    * condition.
    */
   ast_audiohook_lock(audiohook);
   oldstatus = audiohook->status;

   ast_audiohook_remove(old_chan, audiohook);
   ast_audiohook_attach(new_chan, audiohook);

   audiohook->status = oldstatus;
   ast_audiohook_unlock(audiohook);
}

void ast_audiohook_move_by_source(struct ast_channel *old_chan, struct ast_channel *new_chan, const char *source)
{
   struct ast_audiohook *audiohook;

   if (!ast_channel_audiohooks(old_chan)) {
      return;
   }

   audiohook = find_audiohook_by_source(ast_channel_audiohooks(old_chan), source);
   if (!audiohook) {
      return;
   }

   audiohook_move(old_chan, new_chan, audiohook);
}

void ast_audiohook_move_all(struct ast_channel *old_chan, struct ast_channel *new_chan)
{
   struct ast_audiohook *audiohook;
   struct ast_audiohook_list *audiohook_list;

   audiohook_list = ast_channel_audiohooks(old_chan);
   if (!audiohook_list) {
      return;
   }

   AST_LIST_TRAVERSE_SAFE_BEGIN(&audiohook_list->spy_list, audiohook, list) {
      audiohook_move(old_chan, new_chan, audiohook);
   }
   AST_LIST_TRAVERSE_SAFE_END;

   AST_LIST_TRAVERSE_SAFE_BEGIN(&audiohook_list->whisper_list, audiohook, list) {
      audiohook_move(old_chan, new_chan, audiohook);
   }
   AST_LIST_TRAVERSE_SAFE_END;

   AST_LIST_TRAVERSE_SAFE_BEGIN(&audiohook_list->manipulate_list, audiohook, list) {
      audiohook_move(old_chan, new_chan, audiohook);
   }
   AST_LIST_TRAVERSE_SAFE_END;
}

/*! \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 (!ast_channel_audiohooks(chan)) {
      ast_channel_unlock(chan);
      return -1;
   }

   audiohook = find_audiohook_by_source(ast_channel_audiohooks(chan), source);

   ast_channel_unlock(chan);

   if (audiohook && audiohook->status != AST_AUDIOHOOK_STATUS_DONE) {
      ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_SHUTDOWN);
   }

   return (audiohook ? 0 : -1);
}

/*!
 * \brief Remove an audiohook from a specified channel
 *
 * \param chan Channel to remove from
 * \param audiohook Audiohook to remove
 *
 * \return Returns 0 on success, -1 on failure
 *
 * \note The channel does not need to be locked before calling this function
 */
int ast_audiohook_remove(struct ast_channel *chan, struct ast_audiohook *audiohook)
{
   ast_channel_lock(chan);

   if (!ast_channel_audiohooks(chan)) {
      ast_channel_unlock(chan);
      return -1;
   }

   if (audiohook->type == AST_AUDIOHOOK_TYPE_SPY) {
      AST_LIST_REMOVE(&ast_channel_audiohooks(chan)->spy_list, audiohook, list);
   } else if (audiohook->type == AST_AUDIOHOOK_TYPE_WHISPER) {
      AST_LIST_REMOVE(&ast_channel_audiohooks(chan)->whisper_list, audiohook, list);
   } else if (audiohook->type == AST_AUDIOHOOK_TYPE_MANIPULATE) {
      AST_LIST_REMOVE(&ast_channel_audiohooks(chan)->manipulate_list, audiohook, list);
   }

   audiohook_list_set_samplerate_compatibility(ast_channel_audiohooks(chan));
   ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_DONE);

   if (ast_channel_is_bridged(chan)) {
      ast_channel_set_unbridged_nolock(chan, 1);
   }

   ast_channel_unlock(chan);

   return 0;
}

/*! \brief Pass a DTMF frame off to be handled by the audiohook core
 * \param chan Channel that the list is coming off of
 * \param audiohook_list List of audiohooks
 * \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;
   int removed = 0;

   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(list);
         removed = 1;
         ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_DONE);
         ast_audiohook_unlock(audiohook);
         audiohook->manipulate_callback(audiohook, NULL, NULL, 0);
         if (ast_channel_is_bridged(chan)) {
            ast_channel_set_unbridged_nolock(chan, 1);
         }
         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;

   /* if an audiohook got removed, reset samplerate compatibility */
   if (removed) {
      audiohook_list_set_samplerate_compatibility(audiohook_list);
   }
   return frame;
}

static struct ast_frame *audiohook_list_translate_to_slin(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_frame *new_frame = frame;
   struct ast_format *slin;

   /*
    * If we are capable of sample rates other that 8khz, update the internal
    * audiohook_list's rate and higher sample rate audio arrives. If native
    * slin compatibility is turned on all audiohooks in the list will be
    * updated as well during read/write processing.
    */
   audiohook_list->list_internal_samp_rate =
      MAX(ast_format_get_sample_rate(frame->subclass.format), audiohook_list->list_internal_samp_rate);

   slin = ast_format_cache_get_slin_by_rate(audiohook_list->list_internal_samp_rate);
   if (ast_format_cmp(frame->subclass.format, slin) == AST_FORMAT_CMP_EQUAL) {
      return new_frame;
   }

   if (!in_translate->format ||
      ast_format_cmp(frame->subclass.format, in_translate->format) != AST_FORMAT_CMP_EQUAL) {
      struct ast_trans_pvt *new_trans;

      new_trans = ast_translator_build_path(slin, frame->subclass.format);
      if (!new_trans) {
         return NULL;
      }

      if (in_translate->trans_pvt) {
         ast_translator_free_path(in_translate->trans_pvt);
      }
      in_translate->trans_pvt = new_trans;

      ao2_replace(in_translate->format, frame->subclass.format);
   }

   if (!(new_frame = ast_translate(in_translate->trans_pvt, frame, 0))) {
      return NULL;
   }

   return new_frame;
}

static struct ast_frame *audiohook_list_translate_to_native(struct ast_audiohook_list *audiohook_list,
   enum ast_audiohook_direction direction, struct ast_frame *slin_frame, struct ast_format *outformat)
{
   struct ast_audiohook_translate *out_translate = (direction == AST_AUDIOHOOK_DIRECTION_READ ? &audiohook_list->out_translate[0] : &audiohook_list->out_translate[1]);
   struct ast_frame *outframe = NULL;
   if (ast_format_cmp(slin_frame->subclass.format, outformat) == AST_FORMAT_CMP_NOT_EQUAL) {
      /* rebuild translators if necessary */
      if (ast_format_cmp(out_translate->format, outformat) == AST_FORMAT_CMP_NOT_EQUAL) {
         if (out_translate->trans_pvt) {
            ast_translator_free_path(out_translate->trans_pvt);
         }
         if (!(out_translate->trans_pvt = ast_translator_build_path(outformat, slin_frame->subclass.format))) {
            return NULL;
         }
         ao2_replace(out_translate->format, outformat);
      }
      /* translate back to the format the frame came in as. */
      if (!(outframe = ast_translate(out_translate->trans_pvt, slin_frame, 0))) {
         return NULL;
      }
   }
   return outframe;
}

/*!
 *\brief Set the audiohook's internal sample rate to the audiohook_list's rate,
 *       but only when native slin compatibility is turned on.
 *
 * \param audiohook_list audiohook_list data object
 * \param audiohook the audiohook to update
 * \param rate the current max internal sample rate
 */
static void audiohook_list_set_hook_rate(struct ast_audiohook_list *audiohook_list,
                struct ast_audiohook *audiohook, int *rate)
{
   /* The rate should always be the max between itself and the hook */
   if (audiohook->hook_internal_samp_rate > *rate) {
      *rate = audiohook->hook_internal_samp_rate;
   }

   /*
    * If native slin compatibility is turned on then update the audiohook
    * with the audiohook_list's current rate. Note, the audiohook's rate is
    * set to the audiohook_list's rate and not the given rate. If there is
    * a change in rate the hook's rate is changed on its next check.
    */
   if (audiohook_list->native_slin_compatible) {
      ast_set_flag(audiohook, AST_AUDIOHOOK_COMPATIBLE);
      audiohook_set_internal_rate(audiohook, audiohook_list->list_internal_samp_rate, 1);
   } else {
      ast_clear_flag(audiohook, AST_AUDIOHOOK_COMPATIBLE);
   }
}

/*!
 * \brief Pass an AUDIO frame off to be handled by the audiohook core
 *
 * \details
 * This function has 3 ast_frames and 3 parts to handle each.  At the beginning of this
 * function all 3 frames, start_frame, middle_frame, and end_frame point to the initial
 * input frame.
 *
 * Part_1: Translate the start_frame into SLINEAR audio if it is not already in that
 *         format.  The result of this part is middle_frame is guaranteed to be in
 *         SLINEAR format for Part_2.
 * Part_2: Send middle_frame off to spies and manipulators.  At this point middle_frame is
 *         either a new frame as result of the translation, or points directly to the start_frame
 *         because no translation to SLINEAR audio was required.
 * Part_3: Translate end_frame's audio back into the format of start frame if necessary.  This
 *         is only necessary if manipulation of middle_frame occurred.
 *
 * \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_frame *start_frame = frame, *middle_frame = frame, *end_frame = frame;
   struct ast_audiohook *audiohook = NULL;
   int samples;
   int middle_frame_manipulated = 0;
   int removed = 0;
   int internal_sample_rate;

   /* ---Part_1. translate start_frame to SLINEAR if necessary. */
   if (!(middle_frame = audiohook_list_translate_to_slin(audiohook_list, direction, start_frame))) {
      return frame;
   }

   /* If the translation resulted in an interpolated frame then immediately return as audiohooks
    * rely on actual media being present to do things.
    */
   if (!middle_frame->data.ptr) {
      if (middle_frame != start_frame) {
         ast_frfree(middle_frame);
      }
      return start_frame;
   }

   samples = middle_frame->samples;

   /*
    * While processing each audiohook check to see if the internal sample rate needs
    * to be adjusted (it should be the highest rate specified between formats and
    * hooks). The given audiohook_list's internal sample rate is then set to the
    * updated value before returning.
    *
    * If slin compatibility mode is turned on then an audiohook's internal sample
    * rate is set to its audiohook_list's rate. If an audiohook_list's rate is
    * adjusted during this pass then the change is picked up by the audiohooks
    * on the next pass.
    */
   internal_sample_rate = audiohook_list->list_internal_samp_rate;

   /* ---Part_2: Send middle_frame to spy and manipulator lists.  middle_frame is guaranteed to be SLINEAR here.*/
   /* 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(list);
         removed = 1;
         ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_DONE);
         ast_audiohook_unlock(audiohook);
         if (ast_channel_is_bridged(chan)) {
            ast_channel_set_unbridged_nolock(chan, 1);
         }
         continue;
      }
      audiohook_list_set_hook_rate(audiohook_list, audiohook, &internal_sample_rate);
      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 (!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) {
         struct ast_slinfactory *factory = (direction == AST_AUDIOHOOK_DIRECTION_READ ? &audiohook->read_factory : &audiohook->write_factory);
         ast_audiohook_lock(audiohook);
         if (audiohook->status != AST_AUDIOHOOK_STATUS_RUNNING) {
            AST_LIST_REMOVE_CURRENT(list);
            removed = 1;
            ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_DONE);
            ast_audiohook_unlock(audiohook);
            if (ast_channel_is_bridged(chan)) {
               ast_channel_set_unbridged_nolock(chan, 1);
            }
            continue;
         }
         audiohook_list_set_hook_rate(audiohook_list, audiohook, &internal_sample_rate);
         if (ast_slinfactory_available(factory) >= samples && ast_slinfactory_read(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.ptr, data2 = combine_buf; i < samples; i++, data1++, data2++) {
         ast_slinear_saturated_add(data1, data2);
      }
      middle_frame_manipulated = 1;
   }

   /* 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(list);
            removed = 1;
            ast_audiohook_update_status(audiohook, 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);
            if (ast_channel_is_bridged(chan)) {
               ast_channel_set_unbridged_nolock(chan, 1);
            }
            continue;
         }
         audiohook_list_set_hook_rate(audiohook_list, audiohook, &internal_sample_rate);
         /*
          * Feed in frame to manipulation.
          */
         if (!audiohook->manipulate_callback(audiohook, chan, middle_frame, direction)) {
            /*
             * XXX FAILURES ARE IGNORED XXX
             * If the manipulation fails then the frame will be returned in its original state.
             * Since there are potentially more manipulator callbacks in the list, no action should
             * be taken here to exit early.
             */
            middle_frame_manipulated = 1;
         }
         ast_audiohook_unlock(audiohook);
      }
      AST_LIST_TRAVERSE_SAFE_END;
   }

   /* ---Part_3: Decide what to do with the end_frame (whether to transcode or not) */
   if (middle_frame_manipulated) {
      if (!(end_frame = audiohook_list_translate_to_native(audiohook_list, direction, middle_frame, start_frame->subclass.format))) {
         /* translation failed, so just pass back the input frame */
         end_frame = start_frame;
      }
   } else {
      end_frame = start_frame;
   }
   /* clean up our middle_frame if required */
   if (middle_frame != end_frame) {
      ast_frfree(middle_frame);
      middle_frame = NULL;
   }

   /* Before returning, if an audiohook got removed, reset samplerate compatibility */
   if (removed) {
      audiohook_list_set_samplerate_compatibility(audiohook_list);
   } else {
      /*
       * Set the audiohook_list's rate to the updated rate. Note that if a hook
       * was removed then the list's internal rate is reset to the default.
       */
      audiohook_list->list_internal_samp_rate = internal_sample_rate;
   }

   return end_frame;
}

int ast_audiohook_write_list_empty(struct ast_audiohook_list *audiohook_list)
{
   return !audiohook_list
      || (AST_LIST_EMPTY(&audiohook_list->spy_list)
         && AST_LIST_EMPTY(&audiohook_list->whisper_list)
         && AST_LIST_EMPTY(&audiohook_list->manipulate_list));
}

/*! \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 wait;
   struct timespec ts;

   wait = ast_tvadd(ast_tvnow(), ast_samp2tv(50000, 1000));
   ts.tv_sec = wait.tv_sec;
   ts.tv_nsec = wait.tv_usec * 1000;

   ast_cond_timedwait(&audiohook->trigger, &audiohook->lock, &ts);

   return;
}

/* Count number of channel audiohooks by type, regardless of type */
int ast_channel_audiohook_count_by_source(struct ast_channel *chan, const char *source, enum ast_audiohook_type type)
{
   int count = 0;
   struct ast_audiohook *ah = NULL;

   if (!ast_channel_audiohooks(chan)) {
      return -1;
   }

   switch (type) {
      case AST_AUDIOHOOK_TYPE_SPY:
         AST_LIST_TRAVERSE(&ast_channel_audiohooks(chan)->spy_list, ah, list) {
            if (!strcmp(ah->source, source)) {
               count++;
            }
         }
         break;
      case AST_AUDIOHOOK_TYPE_WHISPER:
         AST_LIST_TRAVERSE(&ast_channel_audiohooks(chan)->whisper_list, ah, list) {
            if (!strcmp(ah->source, source)) {
               count++;
            }
         }
         break;
      case AST_AUDIOHOOK_TYPE_MANIPULATE:
         AST_LIST_TRAVERSE(&ast_channel_audiohooks(chan)->manipulate_list, ah, list) {
            if (!strcmp(ah->source, source)) {
               count++;
            }
         }
         break;
      default:
         ast_debug(1, "Invalid audiohook type supplied, (%u)\n", type);
         return -1;
   }

   return count;
}

/* Count number of channel audiohooks by type that are running */
int ast_channel_audiohook_count_by_source_running(struct ast_channel *chan, const char *source, enum ast_audiohook_type type)
{
   int count = 0;
   struct ast_audiohook *ah = NULL;
   if (!ast_channel_audiohooks(chan))
      return -1;

   switch (type) {
      case AST_AUDIOHOOK_TYPE_SPY:
         AST_LIST_TRAVERSE(&ast_channel_audiohooks(chan)->spy_list, ah, list) {
            if ((!strcmp(ah->source, source)) && (ah->status == AST_AUDIOHOOK_STATUS_RUNNING))
               count++;
         }
         break;
      case AST_AUDIOHOOK_TYPE_WHISPER:
         AST_LIST_TRAVERSE(&ast_channel_audiohooks(chan)->whisper_list, ah, list) {
            if ((!strcmp(ah->source, source)) && (ah->status == AST_AUDIOHOOK_STATUS_RUNNING))
               count++;
         }
         break;
      case AST_AUDIOHOOK_TYPE_MANIPULATE:
         AST_LIST_TRAVERSE(&ast_channel_audiohooks(chan)->manipulate_list, ah, list) {
            if ((!strcmp(ah->source, source)) && (ah->status == AST_AUDIOHOOK_STATUS_RUNNING))
               count++;
         }
         break;
      default:
         ast_debug(1, "Invalid audiohook type supplied, (%u)\n", type);
         return -1;
   }
   return count;
}

/*! \brief Audiohook volume adjustment structure */
struct audiohook_volume {
   struct ast_audiohook audiohook; /*!< Audiohook attached to the channel */
   int read_adjustment;            /*!< Value to adjust frames read from the channel by */
   int write_adjustment;           /*!< Value to adjust frames written to the channel by */
};

/*! \brief Callback used to destroy the audiohook volume datastore
 * \param data Volume information structure
 * \return Returns nothing
 */
static void audiohook_volume_destroy(void *data)
{
   struct audiohook_volume *audiohook_volume = data;

   /* Destroy the audiohook as it is no longer in use */
   ast_audiohook_destroy(&audiohook_volume->audiohook);

   /* Finally free ourselves, we are of no more use */
   ast_free(audiohook_volume);

   return;
}

/*! \brief Datastore used to store audiohook volume information */
static const struct ast_datastore_info audiohook_volume_datastore = {
   .type = "Volume",
   .destroy = audiohook_volume_destroy,
};

/*! \brief Helper function which actually gets called by audiohooks to perform the adjustment
 * \param audiohook Audiohook attached to the channel
 * \param chan Channel we are attached to
 * \param frame Frame of audio we want to manipulate
 * \param direction Direction the audio came in from
 * \return Returns 0 on success, -1 on failure
 */
static int audiohook_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 audiohook_volume *audiohook_volume = NULL;
   int *gain = NULL;

   /* If the audiohook is shutting down don't even bother */
   if (audiohook->status == AST_AUDIOHOOK_STATUS_DONE) {
      return 0;
   }

   /* Try to find the datastore containg adjustment information, if we can't just bail out */
   if (!(datastore = ast_channel_datastore_find(chan, &audiohook_volume_datastore, NULL))) {
      return 0;
   }

   audiohook_volume = datastore->data;

   /* Based on direction grab the appropriate adjustment value */
   if (direction == AST_AUDIOHOOK_DIRECTION_READ) {
      gain = &audiohook_volume->read_adjustment;
   } else if (direction == AST_AUDIOHOOK_DIRECTION_WRITE) {
      gain = &audiohook_volume->write_adjustment;
   }

   /* If an adjustment value is present modify the frame */
   if (gain && *gain) {
      ast_frame_adjust_volume(frame, *gain);
   }

   return 0;
}

/*! \brief Helper function which finds and optionally creates an audiohook_volume_datastore datastore on a channel
 * \param chan Channel to look on
 * \param create Whether to create the datastore if not found
 * \return Returns audiohook_volume structure on success, NULL on failure
 */
static struct audiohook_volume *audiohook_volume_get(struct ast_channel *chan, int create)
{
   struct ast_datastore *datastore = NULL;
   struct audiohook_volume *audiohook_volume = NULL;

   /* If we are able to find the datastore return the contents (which is actually an audiohook_volume structure) */
   if ((datastore = ast_channel_datastore_find(chan, &audiohook_volume_datastore, NULL))) {
      return datastore->data;
   }

   /* If we are not allowed to create a datastore or if we fail to create a datastore, bail out now as we have nothing for them */
   if (!create || !(datastore = ast_datastore_alloc(&audiohook_volume_datastore, NULL))) {
      return NULL;
   }

   /* Create a new audiohook_volume structure to contain our adjustments and audiohook */
   if (!(audiohook_volume = ast_calloc(1, sizeof(*audiohook_volume)))) {
      ast_datastore_free(datastore);
      return NULL;
   }

   /* Setup our audiohook structure so we can manipulate the audio */
   ast_audiohook_init(&audiohook_volume->audiohook, AST_AUDIOHOOK_TYPE_MANIPULATE, "Volume", AST_AUDIOHOOK_MANIPULATE_ALL_RATES);
   audiohook_volume->audiohook.manipulate_callback = audiohook_volume_callback;

   /* Attach the audiohook_volume blob to the datastore and attach to the channel */
   datastore->data = audiohook_volume;
   ast_channel_datastore_add(chan, datastore);

   /* All is well... put the audiohook into motion */
   ast_audiohook_attach(chan, &audiohook_volume->audiohook);

   return audiohook_volume;
}

/*! \brief Adjust the volume on frames read from or written to a channel
 * \param chan Channel to muck with
 * \param direction Direction to set on
 * \param volume Value to adjust the volume by
 * \return Returns 0 on success, -1 on failure
 */
int ast_audiohook_volume_set(struct ast_channel *chan, enum ast_audiohook_direction direction, int volume)
{
   struct audiohook_volume *audiohook_volume = NULL;

   /* Attempt to find the audiohook volume information, but only create it if we are not setting the adjustment value to zero */
   if (!(audiohook_volume = audiohook_volume_get(chan, (volume ? 1 : 0)))) {
      return -1;
   }

   /* Now based on the direction set the proper value */
   if (direction == AST_AUDIOHOOK_DIRECTION_READ || direction == AST_AUDIOHOOK_DIRECTION_BOTH) {
      audiohook_volume->read_adjustment = volume;
   }
   if (direction == AST_AUDIOHOOK_DIRECTION_WRITE || direction == AST_AUDIOHOOK_DIRECTION_BOTH) {
      audiohook_volume->write_adjustment = volume;
   }

   return 0;
}

/*! \brief Retrieve the volume adjustment value on frames read from or written to a channel
 * \param chan Channel to retrieve volume adjustment from
 * \param direction Direction to retrieve
 * \return Returns adjustment value
 */
int ast_audiohook_volume_get(struct ast_channel *chan, enum ast_audiohook_direction direction)
{
   struct audiohook_volume *audiohook_volume = NULL;
   int adjustment = 0;

   /* Attempt to find the audiohook volume information, but do not create it as we only want to look at the values */
   if (!(audiohook_volume = audiohook_volume_get(chan, 0))) {
      return 0;
   }

   /* Grab the adjustment value based on direction given */
   if (direction == AST_AUDIOHOOK_DIRECTION_READ) {
      adjustment = audiohook_volume->read_adjustment;
   } else if (direction == AST_AUDIOHOOK_DIRECTION_WRITE) {
      adjustment = audiohook_volume->write_adjustment;
   }

   return adjustment;
}

/*! \brief Adjust the volume on frames read from or written to a channel
 * \param chan Channel to muck with
 * \param direction Direction to increase
 * \param volume Value to adjust the adjustment by
 * \return Returns 0 on success, -1 on failure
 */
int ast_audiohook_volume_adjust(struct ast_channel *chan, enum ast_audiohook_direction direction, int volume)
{
   struct audiohook_volume *audiohook_volume = NULL;

   /* Attempt to find the audiohook volume information, and create an audiohook if none exists */
   if (!(audiohook_volume = audiohook_volume_get(chan, 1))) {
      return -1;
   }

   /* Based on the direction change the specific adjustment value */
   if (direction == AST_AUDIOHOOK_DIRECTION_READ || direction == AST_AUDIOHOOK_DIRECTION_BOTH) {
      audiohook_volume->read_adjustment += volume;
   }
   if (direction == AST_AUDIOHOOK_DIRECTION_WRITE || direction == AST_AUDIOHOOK_DIRECTION_BOTH) {
      audiohook_volume->write_adjustment += volume;
   }

   return 0;
}

/*! \brief Mute frames read from or written to a channel
 * \param chan Channel to muck with
 * \param source Type of audiohook
 * \param flag which flag to set / clear
 * \param clear set or clear
 * \return Returns 0 on success, -1 on failure
 */
int ast_audiohook_set_mute(struct ast_channel *chan, const char *source, enum ast_audiohook_flags flag, int clear)
{
   struct ast_audiohook *audiohook = NULL;

   ast_channel_lock(chan);

   /* Ensure the channel has audiohooks on it */
   if (!ast_channel_audiohooks(chan)) {
      ast_channel_unlock(chan);
      return -1;
   }

   audiohook = find_audiohook_by_source(ast_channel_audiohooks(chan), source);

   if (audiohook) {
      if (clear) {
         ast_clear_flag(audiohook, flag);
      } else {
         ast_set_flag(audiohook, flag);
      }
   }

   ast_channel_unlock(chan);

   return (audiohook ? 0 : -1);
}