loader: Correct overly strict startup checks.
[asterisk/asterisk.git] / main / audiohook.c
index cddfd10..04a379f 100644 (file)
  * \author Joshua Colp <jcolp@digium.com>
  */
 
-#include "asterisk.h"
+/*** MODULEINFO
+       <support_level>core</support_level>
+ ***/
 
-ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+#include "asterisk.h"
 
-#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"
@@ -44,13 +39,27 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 #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;
-       int format;
+       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;
@@ -58,13 +67,45 @@ struct ast_audiohook_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
+ * \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)
+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;
@@ -74,19 +115,13 @@ int ast_audiohook_init(struct ast_audiohook *audiohook, enum ast_audiohook_type
        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;
-       }
+       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 */
-       audiohook->status = AST_AUDIOHOOK_STATUS_NEW;
+       ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_NEW);
 
        return 0;
 }
@@ -100,8 +135,8 @@ 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->read_factory);
                ast_slinfactory_destroy(&audiohook->write_factory);
                break;
        default:
@@ -112,6 +147,8 @@ int ast_audiohook_destroy(struct ast_audiohook *audiohook)
        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);
@@ -119,6 +156,11 @@ int ast_audiohook_destroy(struct ast_audiohook *audiohook)
        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
@@ -128,22 +170,47 @@ int ast_audiohook_destroy(struct ast_audiohook *audiohook)
 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 */
-       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;
+       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;
@@ -156,123 +223,201 @@ static struct ast_frame *audiohook_read_frame_single(struct ast_audiohook *audio
        short buf[samples];
        struct ast_frame frame = {
                .frametype = AST_FRAME_VOICE,
-               .subclass = AST_FORMAT_SLINEAR,
-               .data = buf,
+               .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))
+       if (samples > ast_slinfactory_available(factory)) {
                return NULL;
-       
+       }
+
        /* Read data in from factory */
-       if (!ast_slinfactory_read(factory, buf, samples))
+       if (!ast_slinfactory_read(factory, buf, samples)) {
                return NULL;
+       }
 
-       /* If a volume adjustment needs to be applied apply it */
-       if (vol)
+       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)
+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 i = 0;
-       short buf1[samples], buf2[samples], *read_buf = NULL, *write_buf = NULL, *final_buf = NULL, *data1 = NULL, *data2 = NULL;
+       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,
-               .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 */
+               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 (ast_slinfactory_available(&audiohook->read_factory) >= samples) {
+       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);
+
+                       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)
+                                       if (audiohook->options.read_volume > 0) {
                                                ast_slinear_saturated_multiply(&buf1[count], &adjust_value);
-                                       else if (audiohook->options.read_volume < 0)
+                                       } 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);
+       } 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 (ast_slinfactory_available(&audiohook->write_factory) >= samples) {
+       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);
+
+                       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)
+                                       if (audiohook->options.write_volume > 0) {
                                                ast_slinear_saturated_multiply(&buf2[count], &adjust_value);
-                                       else if (audiohook->options.write_volume < 0)
+                                       } 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);
+       } 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);
 
        /* 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;
+       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 final buffer part of the frame, so it gets duplicated fine */
-       frame.data = final_buf;
+       /* 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);
 }
 
-/*! \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)
+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 (!(read_frame = (direction == AST_AUDIOHOOK_DIRECTION_BOTH ? audiohook_read_frame_both(audiohook, samples) : audiohook_read_frame_single(audiohook, samples, direction))))
+       /* 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 (format != AST_FORMAT_SLINEAR) {
+       if (ast_format_cmp(format, slin) != AST_FORMAT_CMP_EQUAL) {
                /* Rebuild translation path if different format then previously */
-               if (audiohook->format != format) {
+               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, AST_FORMAT_SLINEAR))) {
+                       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);
@@ -283,6 +428,60 @@ struct ast_frame *ast_audiohook_read_frame(struct ast_audiohook *audiohook, size
        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
@@ -292,126 +491,218 @@ int ast_audiohook_attach(struct ast_channel *chan, struct ast_audiohook *audioho
 {
        ast_channel_lock(chan);
 
-       if (!chan->audiohooks) {
+       if (!ast_channel_audiohooks(chan)) {
+               struct ast_audiohook_list *ahlist;
                /* Whoops... allocate a new structure */
-               if (!(chan->audiohooks = ast_calloc(1, sizeof(*chan->audiohooks)))) {
+               if (!(ahlist = ast_calloc(1, sizeof(*ahlist)))) {
                        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);
+               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(&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);
+       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 */
-       audiohook->status = AST_AUDIOHOOK_STATUS_RUNNING;
+       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_DONE)
+       if (audiohook->status == AST_AUDIOHOOK_STATUS_NEW || audiohook->status == AST_AUDIOHOOK_STATUS_DONE) {
                return 0;
+       }
 
-       audiohook->status = AST_AUDIOHOOK_STATUS_SHUTDOWN;
+       ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_SHUTDOWN);
 
-       while (audiohook->status != AST_AUDIOHOOK_STATUS_DONE)
+       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)
+void ast_audiohook_detach_list(struct ast_audiohook_list *audiohook_list)
 {
-       int i = 0;
-       struct ast_audiohook *audiohook = NULL;
+       int i;
+       struct ast_audiohook *audiohook;
+
+       if (!audiohook_list) {
+               return;
+       }
 
        /* 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);
+       while ((audiohook = AST_LIST_REMOVE_HEAD(&audiohook_list->spy_list, list))) {
+               ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_DONE);
        }
-       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);
+       while ((audiohook = AST_LIST_REMOVE_HEAD(&audiohook_list->whisper_list, list))) {
+               ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_DONE);
        }
-       AST_LIST_TRAVERSE_SAFE_END
 
        /* Drop any manipulaters */
-       AST_LIST_TRAVERSE_SAFE_BEGIN(&audiohook_list->manipulate_list, audiohook, list) {
-               ast_audiohook_lock(audiohook);
-               ast_mutex_lock(&audiohook->lock);
-               AST_LIST_REMOVE_CURRENT(&audiohook_list->manipulate_list, list);
-               audiohook->status = AST_AUDIOHOOK_STATUS_DONE;
-               ast_audiohook_unlock(audiohook);
+       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);
        }
-       AST_LIST_TRAVERSE_SAFE_END
 
        /* Drop translation paths if present */
        for (i = 0; i < 2; i++) {
-               if (audiohook_list->in_translate[i].trans_pvt)
+               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)
+                       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);
-
-       return 0;
 }
 
+/*! \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))
+               if (!strcasecmp(audiohook->source, source)) {
                        return audiohook;
+               }
        }
 
        AST_LIST_TRAVERSE(&audiohook_list->whisper_list, audiohook, list) {
-               if (!strcasecmp(audiohook->source, source))
+               if (!strcasecmp(audiohook->source, source)) {
                        return audiohook;
+               }
        }
 
        AST_LIST_TRAVERSE(&audiohook_list->manipulate_list, audiohook, list) {
-               if (!strcasecmp(audiohook->source, source))
+               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
@@ -424,21 +715,61 @@ int ast_audiohook_detach_source(struct ast_channel *chan, const char *source)
        ast_channel_lock(chan);
 
        /* Ensure the channel has audiohooks on it */
-       if (!chan->audiohooks) {
+       if (!ast_channel_audiohooks(chan)) {
                ast_channel_unlock(chan);
                return -1;
        }
 
-       audiohook = find_audiohook_by_source(chan->audiohooks, source);
+       audiohook = find_audiohook_by_source(ast_channel_audiohooks(chan), source);
 
        ast_channel_unlock(chan);
 
-       if (audiohook && audiohook->status != AST_AUDIOHOOK_STATUS_DONE)
-               audiohook->status = AST_AUDIOHOOK_STATUS_SHUTDOWN;
+       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
@@ -449,26 +780,152 @@ int ast_audiohook_detach_source(struct ast_channel *chan, const char *source)
 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(&audiohook_list->manipulate_list, list);
-                       audiohook->status = AST_AUDIOHOOK_STATUS_DONE;
+                       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))
+               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
+       AST_LIST_TRAVERSE_SAFE_END;
 
+       /* if an audiohook got removed, reset samplerate compatibility */
+       if (removed) {
+               audiohook_list_set_samplerate_compatibility(audiohook_list);
+       }
        return frame;
 }
 
-/*! \brief Pass an AUDIO frame off to be handled by the audiohook core
+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
@@ -477,66 +934,96 @@ static struct ast_frame *dtmf_audiohook_write_list(struct ast_channel *chan, str
  */
 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;
+       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);
                }
-               if (!(middle_frame = ast_translate(in_translate->trans_pvt, frame, 0)))
-                       return 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(&audiohook_list->spy_list, list);
-                       audiohook->status = AST_AUDIOHOOK_STATUS_DONE;
-                       ast_cond_signal(&audiohook->trigger);
+                       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
+       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)) {
+       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(&audiohook_list->whisper_list, list);
-                               audiohook->status = AST_AUDIOHOOK_STATUS_DONE;
-                               ast_cond_signal(&audiohook->trigger);
+                               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;
                        }
-                       if (ast_slinfactory_available(&audiohook->write_factory) >= samples && ast_slinfactory_read(&audiohook->write_factory, read_buf, samples)) {
+                       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++)
+                               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
+               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++)
+               for (i = 0, data1 = middle_frame->data.ptr, data2 = combine_buf; i < samples; i++, data1++, data2++) {
                        ast_slinear_saturated_add(data1, data2);
-               end_frame = middle_frame;
+               }
+               middle_frame_manipulated = 1;
        }
 
        /* Pass off frame to manipulate audiohooks */
@@ -544,54 +1031,72 @@ static struct ast_frame *audio_audiohook_write_list(struct ast_channel *chan, st
                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_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;
                        }
-                       /* Feed in frame to manipulation */
-                       audiohook->manipulate_callback(audiohook, chan, middle_frame, direction);
+                       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
-               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);
+               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;
                }
-               /* Yay let's rid ourselves of the start frame */
-               ast_frfree(start_frame);
        } else {
-               /* No frame was modified, we can just drop our middle frame and pass the frame we got in out */
+               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
@@ -602,28 +1107,323 @@ static struct ast_frame *audio_audiohook_write_list(struct ast_channel *chan, st
 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)
+       if (frame->frametype == AST_FRAME_VOICE) {
                return audio_audiohook_write_list(chan, audiohook_list, direction, frame);
-       else if (frame->frametype == AST_FRAME_DTMF)
+       } else if (frame->frametype == AST_FRAME_DTMF) {
                return dtmf_audiohook_write_list(chan, audiohook_list, direction, frame);
-       else
+       } 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 timeval wait;
        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;
-       
+       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);
+}