Sat Nov 1 06:28:23 2008

Asterisk developer's documentation

---

audiohook.c

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/*
 * Asterisk -- An open source telephony toolkit.
 *
 * Copyright (C) 1999 - 2007, Digium, Inc.
 *
 * Joshua Colp <jcolp@digium.com>
 *
 * See http://www.asterisk.org for more information about
 * the Asterisk project. Please do not directly contact
 * any of the maintainers of this project for assistance;
 * the project provides a web site, mailing lists and IRC
 * channels for your use.
 *
 * This program is free software, distributed under the terms of
 * the GNU General Public License Version 2. See the LICENSE file
 * at the top of the source tree.
 */

/*! \file
 *
 * \brief Audiohooks Architecture
 *
 * \author Joshua Colp <jcolp@digium.com>
 */

#include "asterisk.h"

ASTERISK_FILE_VERSION(__FILE__, "$Revision: 149204 $")

#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <signal.h>
#include <errno.h>
#include <unistd.h>

#include "asterisk/logger.h"
#include "asterisk/channel.h"
#include "asterisk/options.h"
#include "asterisk/utils.h"
#include "asterisk/lock.h"
#include "asterisk/linkedlists.h"
#include "asterisk/audiohook.h"
#include "asterisk/slinfactory.h"
#include "asterisk/frame.h"
#include "asterisk/translate.h"

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

struct ast_audiohook_list {
   struct ast_audiohook_translate in_translate[2];
   struct ast_audiohook_translate out_translate[2];
   AST_LIST_HEAD_NOLOCK(, ast_audiohook) spy_list;
   AST_LIST_HEAD_NOLOCK(, ast_audiohook) whisper_list;
   AST_LIST_HEAD_NOLOCK(, ast_audiohook) manipulate_list;
};

/*! \brief Initialize an audiohook structure
 * \param audiohook Audiohook structure
 * \param type
 * \param source
 * \return Returns 0 on success, -1 on failure
 */
int ast_audiohook_init(struct ast_audiohook *audiohook, enum ast_audiohook_type type, const char *source)
{
   /* Need to keep the type and source */
   audiohook->type = type;
   audiohook->source = source;

   /* Initialize lock that protects our audiohook */
   ast_mutex_init(&audiohook->lock);
   ast_cond_init(&audiohook->trigger, NULL);

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

   /* Since we are just starting out... this audiohook is new */
   audiohook->status = AST_AUDIOHOOK_STATUS_NEW;

   return 0;
}

/*! \brief Destroys an audiohook structure
 * \param audiohook Audiohook structure
 * \return Returns 0 on success, -1 on failure
 */
int ast_audiohook_destroy(struct ast_audiohook *audiohook)
{
   /* Drop the factories used by this audiohook type */
   switch (audiohook->type) {
   case AST_AUDIOHOOK_TYPE_SPY:
      ast_slinfactory_destroy(&audiohook->read_factory);
   case AST_AUDIOHOOK_TYPE_WHISPER:
      ast_slinfactory_destroy(&audiohook->write_factory);
      break;
   default:
      break;
   }

   /* Destroy translation path if present */
   if (audiohook->trans_pvt)
      ast_translator_free_path(audiohook->trans_pvt);

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

   return 0;
}

/*! \brief Writes a frame into the audiohook structure
 * \param audiohook Audiohook structure
 * \param direction Direction the audio frame came from
 * \param frame Frame to write in
 * \return Returns 0 on success, -1 on failure
 */
int ast_audiohook_write_frame(struct ast_audiohook *audiohook, enum ast_audiohook_direction direction, struct ast_frame *frame)
{
   struct ast_slinfactory *factory = (direction == AST_AUDIOHOOK_DIRECTION_READ ? &audiohook->read_factory : &audiohook->write_factory);
   struct ast_slinfactory *other_factory = (direction == AST_AUDIOHOOK_DIRECTION_READ ? &audiohook->write_factory : &audiohook->read_factory);
   struct timeval *time = (direction == AST_AUDIOHOOK_DIRECTION_READ ? &audiohook->read_time : &audiohook->write_time), previous_time = *time;
   int our_factory_ms;
   int other_factory_samples;
   int other_factory_ms;

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

   our_factory_ms = ast_tvdiff_ms(*time, previous_time) + (ast_slinfactory_available(factory) / 8);
   other_factory_samples = ast_slinfactory_available(other_factory);
   other_factory_ms = other_factory_samples / 8;

   /* If we are using a sync trigger and this factory suddenly got audio fed in after a lapse, then flush both factories to ensure they remain in sync */
   if (ast_test_flag(audiohook, AST_AUDIOHOOK_TRIGGER_SYNC) && other_factory_samples && (our_factory_ms - other_factory_ms > AST_AUDIOHOOK_SYNC_TOLERANCE)) {
      if (option_debug)
         ast_log(LOG_DEBUG, "Flushing audiohook %p so it remains in sync\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 = AST_FORMAT_SLINEAR,
      .data = buf,
      .datalen = sizeof(buf),
      .samples = samples,
   };

   /* Ensure the factory is able to give us the samples we want */
   if (samples > ast_slinfactory_available(factory))
      return NULL;
   
   /* Read data in from factory */
   if (!ast_slinfactory_read(factory, buf, samples))
      return NULL;

   /* If a volume adjustment needs to be applied apply it */
   if (vol)
      ast_frame_adjust_volume(&frame, vol);

   return ast_frdup(&frame);
}

static struct ast_frame *audiohook_read_frame_both(struct ast_audiohook *audiohook, size_t samples)
{
   int i = 0, usable_read, usable_write;
   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,
   };

   /* 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 */
      if (option_debug)
         ast_log(LOG_DEBUG, "Read factory %p and write factory %p both fail to provide %zd 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)) {
      if (option_debug > 2)
         ast_log(LOG_DEBUG, "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)) {
      if (option_debug > 2)
         ast_log(LOG_DEBUG, "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;
         /* Adjust read volume if need be */
         if (audiohook->options.read_volume) {
            int count = 0;
            short adjust_value = abs(audiohook->options.read_volume);
            for (count = 0; count < samples; count++) {
               if (audiohook->options.read_volume > 0)
                  ast_slinear_saturated_multiply(&buf1[count], &adjust_value);
               else if (audiohook->options.read_volume < 0)
                  ast_slinear_saturated_divide(&buf1[count], &adjust_value);
            }
         }
      }
   } else if (option_debug)
      ast_log(LOG_DEBUG, "Failed to get %zd samples from read factory %p\n", samples, &audiohook->read_factory);

   /* Move on to the write factory... if there are enough samples, read them in */
   if (usable_write) {
      if (ast_slinfactory_read(&audiohook->write_factory, buf2, samples)) {
         write_buf = buf2;
         /* Adjust write volume if need be */
         if (audiohook->options.write_volume) {
            int count = 0;
            short adjust_value = abs(audiohook->options.write_volume);
            for (count = 0; count < samples; count++) {
               if (audiohook->options.write_volume > 0)
                  ast_slinear_saturated_multiply(&buf2[count], &adjust_value);
               else if (audiohook->options.write_volume < 0)
                  ast_slinear_saturated_divide(&buf2[count], &adjust_value);
            }
         }
      }
   } else if (option_debug)
      ast_log(LOG_DEBUG, "Failed to get %zd samples from write factory %p\n", samples, &audiohook->write_factory);

   /* Basically we figure out which buffer to use... and if mixing can be done here */
   if (!read_buf && !write_buf)
      return NULL;
   else if (read_buf && write_buf) {
      for (i = 0, data1 = read_buf, data2 = write_buf; i < samples; i++, data1++, data2++)
         ast_slinear_saturated_add(data1, data2);
      final_buf = buf1;
   } else if (read_buf)
      final_buf = buf1;
   else if (write_buf)
      final_buf = buf2;

   /* Make the final buffer part of the frame, so it gets duplicated fine */
   frame.data = final_buf;

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

/*! \brief Reads a frame in from the audiohook structure
 * \param audiohook Audiohook structure
 * \param samples Number of samples wanted
 * \param direction Direction the audio frame came from
 * \param format Format of frame remote side wants back
 * \return Returns frame on success, NULL on failure
 */
struct ast_frame *ast_audiohook_read_frame(struct ast_audiohook *audiohook, size_t samples, enum ast_audiohook_direction direction, int format)
{
   struct ast_frame *read_frame = NULL, *final_frame = NULL;

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

   /* If they don't want signed linear back out, we'll have to send it through the translation path */
   if (format != AST_FORMAT_SLINEAR) {
      /* Rebuild translation path if different format then previously */
      if (audiohook->format != format) {
         if (audiohook->trans_pvt) {
            ast_translator_free_path(audiohook->trans_pvt);
            audiohook->trans_pvt = NULL;
         }
         /* Setup new translation path for this format... if we fail we can't very well return signed linear so free the frame and return nothing */
         if (!(audiohook->trans_pvt = ast_translator_build_path(format, AST_FORMAT_SLINEAR))) {
            ast_frfree(read_frame);
            return NULL;
         }
      }
      /* Convert to requested format, and allow the read in frame to be freed */
      final_frame = ast_translate(audiohook->trans_pvt, read_frame, 1);
   } else {
      final_frame = read_frame;
   }

   return final_frame;
}

/*! \brief Attach audiohook to channel
 * \param chan Channel
 * \param audiohook Audiohook structure
 * \return Returns 0 on success, -1 on failure
 */
int ast_audiohook_attach(struct ast_channel *chan, struct ast_audiohook *audiohook)
{
   ast_channel_lock(chan);

   if (!chan->audiohooks) {
      /* Whoops... allocate a new structure */
      if (!(chan->audiohooks = ast_calloc(1, sizeof(*chan->audiohooks)))) {
         ast_channel_unlock(chan);
         return -1;
      }
      AST_LIST_HEAD_INIT_NOLOCK(&chan->audiohooks->spy_list);
      AST_LIST_HEAD_INIT_NOLOCK(&chan->audiohooks->whisper_list);
      AST_LIST_HEAD_INIT_NOLOCK(&chan->audiohooks->manipulate_list);
   }

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

   /* Change status over to running since it is now attached */
   audiohook->status = AST_AUDIOHOOK_STATUS_RUNNING;

   ast_channel_unlock(chan);

   return 0;
}

/*! \brief Detach audiohook from channel
 * \param audiohook Audiohook structure
 * \return Returns 0 on success, -1 on failure
 */
int ast_audiohook_detach(struct ast_audiohook *audiohook)
{
   if (audiohook->status == AST_AUDIOHOOK_STATUS_DONE)
      return 0;

   audiohook->status = AST_AUDIOHOOK_STATUS_SHUTDOWN;

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

   return 0;
}

/*! \brief Detach audiohooks from list and destroy said list
 * \param audiohook_list List of audiohooks
 * \return Returns 0 on success, -1 on failure
 */
int ast_audiohook_detach_list(struct ast_audiohook_list *audiohook_list)
{
   int i = 0;
   struct ast_audiohook *audiohook = NULL;

   /* Drop any spies */
   AST_LIST_TRAVERSE_SAFE_BEGIN(&audiohook_list->spy_list, audiohook, list) {
      ast_audiohook_lock(audiohook);
      AST_LIST_REMOVE_CURRENT(&audiohook_list->spy_list, list);
      audiohook->status = AST_AUDIOHOOK_STATUS_DONE;
      ast_cond_signal(&audiohook->trigger);
      ast_audiohook_unlock(audiohook);
   }
   AST_LIST_TRAVERSE_SAFE_END

   /* Drop any whispering sources */
   AST_LIST_TRAVERSE_SAFE_BEGIN(&audiohook_list->whisper_list, audiohook, list) {
      ast_audiohook_lock(audiohook);
      AST_LIST_REMOVE_CURRENT(&audiohook_list->whisper_list, list);
      audiohook->status = AST_AUDIOHOOK_STATUS_DONE;
      ast_cond_signal(&audiohook->trigger);
      ast_audiohook_unlock(audiohook);
   }
   AST_LIST_TRAVERSE_SAFE_END

   /* Drop any manipulaters */
   AST_LIST_TRAVERSE_SAFE_BEGIN(&audiohook_list->manipulate_list, audiohook, list) {
      ast_audiohook_lock(audiohook);
      AST_LIST_REMOVE_CURRENT(&audiohook_list->manipulate_list, list);
      audiohook->status = AST_AUDIOHOOK_STATUS_DONE;
      ast_audiohook_unlock(audiohook);
      audiohook->manipulate_callback(audiohook, NULL, NULL, 0);
   }
   AST_LIST_TRAVERSE_SAFE_END

   /* Drop translation paths if present */
   for (i = 0; i < 2; i++) {
      if (audiohook_list->in_translate[i].trans_pvt)
         ast_translator_free_path(audiohook_list->in_translate[i].trans_pvt);
      if (audiohook_list->out_translate[i].trans_pvt)
         ast_translator_free_path(audiohook_list->out_translate[i].trans_pvt);
   }
   
   /* Free ourselves */
   ast_free(audiohook_list);

   return 0;
}

static struct ast_audiohook *find_audiohook_by_source(struct ast_audiohook_list *audiohook_list, const char *source)
{
   struct ast_audiohook *audiohook = NULL;

   AST_LIST_TRAVERSE(&audiohook_list->spy_list, audiohook, list) {
      if (!strcasecmp(audiohook->source, source))
         return audiohook;
   }

   AST_LIST_TRAVERSE(&audiohook_list->whisper_list, audiohook, list) {
      if (!strcasecmp(audiohook->source, source))
         return audiohook;
   }

   AST_LIST_TRAVERSE(&audiohook_list->manipulate_list, audiohook, list) {
      if (!strcasecmp(audiohook->source, source))
         return audiohook;
   }

   return NULL;
}

/*! \brief Detach specified source audiohook from channel
 * \param chan Channel to detach from
 * \param source Name of source to detach
 * \return Returns 0 on success, -1 on failure
 */
int ast_audiohook_detach_source(struct ast_channel *chan, const char *source)
{
   struct ast_audiohook *audiohook = NULL;

   ast_channel_lock(chan);

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

   audiohook = find_audiohook_by_source(chan->audiohooks, source);

   ast_channel_unlock(chan);

   if (audiohook && audiohook->status != AST_AUDIOHOOK_STATUS_DONE)
      audiohook->status = AST_AUDIOHOOK_STATUS_SHUTDOWN;

   return (audiohook ? 0 : -1);
}

/*! \brief Pass a DTMF frame off to be handled by the audiohook core
 * \param chan Channel that the list is coming off of
 * \param audiohook_list List of audiohooks
 * \param direction Direction frame is coming in from
 * \param frame The frame itself
 * \return Return frame on success, NULL on failure
 */
static struct ast_frame *dtmf_audiohook_write_list(struct ast_channel *chan, struct ast_audiohook_list *audiohook_list, enum ast_audiohook_direction direction, struct ast_frame *frame)
{
   struct ast_audiohook *audiohook = NULL;

   AST_LIST_TRAVERSE_SAFE_BEGIN(&audiohook_list->manipulate_list, audiohook, list) {
      ast_audiohook_lock(audiohook);
      if (audiohook->status != AST_AUDIOHOOK_STATUS_RUNNING) {
         AST_LIST_REMOVE_CURRENT(&audiohook_list->manipulate_list, list);
         audiohook->status = AST_AUDIOHOOK_STATUS_DONE;
         ast_audiohook_unlock(audiohook);
         audiohook->manipulate_callback(audiohook, NULL, NULL, 0);
         continue;
      }
      if (ast_test_flag(audiohook, AST_AUDIOHOOK_WANTS_DTMF))
         audiohook->manipulate_callback(audiohook, chan, frame, direction);
      ast_audiohook_unlock(audiohook);
   }
   AST_LIST_TRAVERSE_SAFE_END

   return frame;
}

/*! \brief Pass an AUDIO frame off to be handled by the audiohook core
 * \param chan Channel that the list is coming off of
 * \param audiohook_list List of audiohooks
 * \param direction Direction frame is coming in from
 * \param frame The frame itself
 * \return Return frame on success, NULL on failure
 */
static struct ast_frame *audio_audiohook_write_list(struct ast_channel *chan, struct ast_audiohook_list *audiohook_list, enum ast_audiohook_direction direction, struct ast_frame *frame)
{
   struct ast_audiohook_translate *in_translate = (direction == AST_AUDIOHOOK_DIRECTION_READ ? &audiohook_list->in_translate[0] : &audiohook_list->in_translate[1]);
   struct ast_audiohook_translate *out_translate = (direction == AST_AUDIOHOOK_DIRECTION_READ ? &audiohook_list->out_translate[0] : &audiohook_list->out_translate[1]);
   struct ast_frame *start_frame = frame, *middle_frame = frame, *end_frame = frame;
   struct ast_audiohook *audiohook = NULL;
   int samples = frame->samples;
   
   /* If the frame coming in is not signed linear we have to send it through the in_translate path */
   if (frame->subclass != AST_FORMAT_SLINEAR) {
      if (in_translate->format != frame->subclass) {
         if (in_translate->trans_pvt)
            ast_translator_free_path(in_translate->trans_pvt);
         if (!(in_translate->trans_pvt = ast_translator_build_path(AST_FORMAT_SLINEAR, frame->subclass)))
            return frame;
         in_translate->format = frame->subclass;
      }
      if (!(middle_frame = ast_translate(in_translate->trans_pvt, frame, 0)))
         return frame;
   }

   /* Queue up signed linear frame to each spy */
   AST_LIST_TRAVERSE_SAFE_BEGIN(&audiohook_list->spy_list, audiohook, list) {
      ast_audiohook_lock(audiohook);
      if (audiohook->status != AST_AUDIOHOOK_STATUS_RUNNING) {
         AST_LIST_REMOVE_CURRENT(&audiohook_list->spy_list, list);
         audiohook->status = AST_AUDIOHOOK_STATUS_DONE;
         ast_cond_signal(&audiohook->trigger);
         ast_audiohook_unlock(audiohook);
         continue;
      }
      ast_audiohook_write_frame(audiohook, direction, middle_frame);
      ast_audiohook_unlock(audiohook);
   }
   AST_LIST_TRAVERSE_SAFE_END

   /* If this frame is being written out to the channel then we need to use whisper sources */
   if (direction == AST_AUDIOHOOK_DIRECTION_WRITE && !AST_LIST_EMPTY(&audiohook_list->whisper_list)) {
      int i = 0;
      short read_buf[samples], combine_buf[samples], *data1 = NULL, *data2 = NULL;
      memset(&combine_buf, 0, sizeof(combine_buf));
      AST_LIST_TRAVERSE_SAFE_BEGIN(&audiohook_list->whisper_list, audiohook, list) {
         ast_audiohook_lock(audiohook);
         if (audiohook->status != AST_AUDIOHOOK_STATUS_RUNNING) {
            AST_LIST_REMOVE_CURRENT(&audiohook_list->whisper_list, list);
            audiohook->status = AST_AUDIOHOOK_STATUS_DONE;
            ast_cond_signal(&audiohook->trigger);
            ast_audiohook_unlock(audiohook);
            continue;
         }
         if (ast_slinfactory_available(&audiohook->write_factory) >= samples && ast_slinfactory_read(&audiohook->write_factory, read_buf, samples)) {
            /* Take audio from this whisper source and combine it into our main buffer */
            for (i = 0, data1 = combine_buf, data2 = read_buf; i < samples; i++, data1++, data2++)
               ast_slinear_saturated_add(data1, data2);
         }
         ast_audiohook_unlock(audiohook);
      }
      AST_LIST_TRAVERSE_SAFE_END
      /* We take all of the combined whisper sources and combine them into the audio being written out */
      for (i = 0, data1 = middle_frame->data, data2 = combine_buf; i < samples; i++, data1++, data2++)
         ast_slinear_saturated_add(data1, data2);
      end_frame = middle_frame;
   }

   /* Pass off frame to manipulate audiohooks */
   if (!AST_LIST_EMPTY(&audiohook_list->manipulate_list)) {
      AST_LIST_TRAVERSE_SAFE_BEGIN(&audiohook_list->manipulate_list, audiohook, list) {
         ast_audiohook_lock(audiohook);
         if (audiohook->status != AST_AUDIOHOOK_STATUS_RUNNING) {
            AST_LIST_REMOVE_CURRENT(&audiohook_list->manipulate_list, list);
            audiohook->status = AST_AUDIOHOOK_STATUS_DONE;
            ast_audiohook_unlock(audiohook);
            /* We basically drop all of our links to the manipulate audiohook and prod it to do it's own destructive things */
            audiohook->manipulate_callback(audiohook, chan, NULL, direction);
            continue;
         }
         /* Feed in frame to manipulation */
         audiohook->manipulate_callback(audiohook, chan, middle_frame, direction);
         ast_audiohook_unlock(audiohook);
      }
      AST_LIST_TRAVERSE_SAFE_END
      end_frame = middle_frame;
   }

   /* Now we figure out what to do with our end frame (whether to transcode or not) */
   if (middle_frame == end_frame) {
      /* Middle frame was modified and became the end frame... let's see if we need to transcode */
      if (end_frame->subclass != start_frame->subclass) {
         if (out_translate->format != start_frame->subclass) {
            if (out_translate->trans_pvt)
               ast_translator_free_path(out_translate->trans_pvt);
            if (!(out_translate->trans_pvt = ast_translator_build_path(start_frame->subclass, AST_FORMAT_SLINEAR))) {
               /* We can't transcode this... drop our middle frame and return the original */
               ast_frfree(middle_frame);
               return start_frame;
            }
            out_translate->format = start_frame->subclass;
         }
         /* Transcode from our middle (signed linear) frame to new format of the frame that came in */
         if (!(end_frame = ast_translate(out_translate->trans_pvt, middle_frame, 0))) {
            /* Failed to transcode the frame... drop it and return the original */
            ast_frfree(middle_frame);
            return start_frame;
         }
         /* Here's the scoop... middle frame is no longer of use to us */
         ast_frfree(middle_frame);
      }
   } else {
      /* No frame was modified, we can just drop our middle frame and pass the frame we got in out */
      ast_frfree(middle_frame);
   }

   return end_frame;
}

/*! \brief Pass a frame off to be handled by the audiohook core
 * \param chan Channel that the list is coming off of
 * \param audiohook_list List of audiohooks
 * \param direction Direction frame is coming in from
 * \param frame The frame itself
 * \return Return frame on success, NULL on failure
 */
struct ast_frame *ast_audiohook_write_list(struct ast_channel *chan, struct ast_audiohook_list *audiohook_list, enum ast_audiohook_direction direction, struct ast_frame *frame)
{
   /* Pass off frame to it's respective list write function */
   if (frame->frametype == AST_FRAME_VOICE)
      return audio_audiohook_write_list(chan, audiohook_list, direction, frame);
   else if (frame->frametype == AST_FRAME_DTMF)
      return dtmf_audiohook_write_list(chan, audiohook_list, direction, frame);
   else
      return frame;
}
         

/*! \brief Wait for audiohook trigger to be triggered
 * \param audiohook Audiohook to wait on
 */
void ast_audiohook_trigger_wait(struct ast_audiohook *audiohook)
{
   struct timeval tv;
   struct timespec ts;

   tv = ast_tvadd(ast_tvnow(), ast_samp2tv(50000, 1000));
   ts.tv_sec = tv.tv_sec;
   ts.tv_nsec = tv.tv_usec * 1000;
   
   ast_cond_timedwait(&audiohook->trigger, &audiohook->lock, &ts);
   
   return;
}

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