loader: Correct overly strict startup checks.
[asterisk/asterisk.git] / main / audiohook.c
index 679e27c..04a379f 100644 (file)
@@ -29,8 +29,6 @@
 
 #include "asterisk.h"
 
-ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
-
 #include <signal.h>
 
 #include "asterisk/channel.h"
@@ -41,13 +39,17 @@ 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;
-       struct ast_format format;
+       struct ast_format *format;
 };
 
 struct ast_audiohook_list {
@@ -67,7 +69,7 @@ struct ast_audiohook_list {
 
 static int audiohook_set_internal_rate(struct ast_audiohook *audiohook, int rate, int reset)
 {
-       struct ast_format slin;
+       struct ast_format *slin;
 
        if (audiohook->hook_internal_samp_rate == rate) {
                return 0;
@@ -75,31 +77,32 @@ static int audiohook_set_internal_rate(struct ast_audiohook *audiohook, int rate
 
        audiohook->hook_internal_samp_rate = rate;
 
-       ast_format_set(&slin, ast_format_slin_by_rate(rate), 0);
+       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:
-               if (reset) {
-                       ast_slinfactory_destroy(&audiohook->read_factory);
-               }
-               ast_slinfactory_init_with_format(&audiohook->read_factory, &slin);
-               /* fall through */
        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->write_factory, &slin);
+               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, enum ast_audiohook_init_flags init_flags)
@@ -115,7 +118,7 @@ int ast_audiohook_init(struct ast_audiohook *audiohook, enum ast_audiohook_type
        audiohook->init_flags = init_flags;
 
        /* initialize internal rate at 8khz, this will adjust if necessary */
-       audiohook_set_internal_rate(audiohook, 8000, 0);
+       audiohook_set_internal_rate(audiohook, DEFAULT_INTERNAL_SAMPLE_RATE, 0);
 
        /* Since we are just starting out... this audiohook is new */
        ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_NEW);
@@ -132,8 +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:
@@ -144,6 +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);
@@ -151,6 +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
@@ -166,7 +176,6 @@ int ast_audiohook_write_frame(struct ast_audiohook *audiohook, enum ast_audiohoo
        int our_factory_ms;
        int other_factory_samples;
        int other_factory_ms;
-       int muteme = 0;
 
        /* Update last feeding time to be current */
        *rwtime = ast_tvnow();
@@ -176,7 +185,7 @@ int ast_audiohook_write_frame(struct ast_audiohook *audiohook, enum ast_audiohoo
        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) && other_factory_samples && (our_factory_ms - other_factory_ms > AST_AUDIOHOOK_SYNC_TOLERANCE)) {
+       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);
@@ -186,17 +195,10 @@ int ast_audiohook_write_frame(struct ast_audiohook *audiohook, enum ast_audiohoo
                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);
-       }
-
-       /* swap frame data for zeros if mute is required */
-       if ((ast_test_flag(audiohook, AST_AUDIOHOOK_MUTE_READ) && (direction == AST_AUDIOHOOK_DIRECTION_READ)) ||
-               (ast_test_flag(audiohook, AST_AUDIOHOOK_MUTE_WRITE) && (direction == AST_AUDIOHOOK_DIRECTION_WRITE)) ||
-               (ast_test_flag(audiohook, AST_AUDIOHOOK_MUTE_READ | AST_AUDIOHOOK_MUTE_WRITE) == (AST_AUDIOHOOK_MUTE_READ | AST_AUDIOHOOK_MUTE_WRITE))) {
-                       muteme = 1;
-       }
-
-       if (muteme && frame->datalen > 0) {
-               ast_frame_clear(frame);
+       } 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 */
@@ -221,38 +223,48 @@ static struct ast_frame *audiohook_read_frame_single(struct ast_audiohook *audio
        short buf[samples];
        struct ast_frame frame = {
                .frametype = AST_FRAME_VOICE,
+               .subclass.format = ast_format_cache_get_slin_by_rate(audiohook->hook_internal_samp_rate),
                .data.ptr = buf,
                .datalen = sizeof(buf),
                .samples = samples,
        };
-       ast_format_set(&frame.subclass.format, ast_format_slin_by_rate(audiohook->hook_internal_samp_rate), 0);
 
        /* 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, struct ast_frame **read_reference, struct ast_frame **write_reference)
 {
-       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;
+       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,
-               .data.ptr = NULL,
                .datalen = sizeof(buf1),
                .samples = samples,
        };
-       ast_format_set(&frame.subclass.format, ast_format_slin_by_rate(audiohook->hook_internal_samp_rate), 0);
 
        /* Make sure both factories have the required samples */
        usable_read = (ast_slinfactory_available(&audiohook->read_factory) >= samples ? 1 : 0);
@@ -260,7 +272,7 @@ static struct ast_frame *audiohook_read_frame_both(struct ast_audiohook *audioho
 
        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 %zd samples\n", &audiohook->read_factory, &audiohook->write_factory, samples);
+               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;
        }
 
@@ -280,15 +292,19 @@ static struct ast_frame *audiohook_read_frame_both(struct ast_audiohook *audioho
        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);
+                                       }
                                }
                        }
                }
@@ -300,15 +316,19 @@ static struct ast_frame *audiohook_read_frame_both(struct ast_audiohook *audioho
        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);
+                                       }
                                }
                        }
                }
@@ -316,32 +336,32 @@ static struct ast_frame *audiohook_read_frame_both(struct ast_audiohook *audioho
                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 && read_reference) {
-               frame.data.ptr = buf1;
+               frame.data.ptr = read_buf;
                *read_reference = ast_frdup(&frame);
        }
        if (write_buf && write_reference) {
-               frame.data.ptr = buf2;
+               frame.data.ptr = write_buf;
                *write_reference = ast_frdup(&frame);
        }
 
-       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);
+       /* 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++);
+                       }
                }
-               final_buf = buf1;
-       } else if (read_buf) {
-               final_buf = buf1;
        } else if (write_buf) {
-               final_buf = buf2;
+               frame.data.ptr = write_buf;
        } else {
                return NULL;
        }
 
-       /* Make the final buffer part of the frame, so it gets duplicated fine */
-       frame.data.ptr = final_buf;
-
        /* Yahoo, a combined copy of the audio! */
        return ast_frdup(&frame);
 }
@@ -349,40 +369,55 @@ static struct ast_frame *audiohook_read_frame_both(struct ast_audiohook *audioho
 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 tmp_fmt;
-       int samples_converted;
-
-       /* the number of samples requested is based on the format they are requesting.  Inorder
-        * to process this correctly samples must be converted to our internal sample rate */
-       if (audiohook->hook_internal_samp_rate == ast_format_rate(format)) {
-               samples_converted = samples;
-       } else if (audiohook->hook_internal_samp_rate > ast_format_rate(format)) {
-               samples_converted = samples * (audiohook->hook_internal_samp_rate / (float) ast_format_rate(format));
-       } else {
-               samples_converted = samples * (ast_format_rate(format) / (float) audiohook->hook_internal_samp_rate);
+       struct ast_format *slin;
+
+       /*
+        * Update the rate if compatibility mode is turned off or if it is
+        * turned on and the format rate is higher than the current rate.
+        *
+        * This makes it so any unnecessary rate switching/resetting does
+        * not take place and also any associated audiohook_list's internal
+        * sample rate maintains the highest sample rate between hooks.
+        */
+       if (!ast_test_flag(audiohook, AST_AUDIOHOOK_COMPATIBLE) ||
+           (ast_test_flag(audiohook, AST_AUDIOHOOK_COMPATIBLE) &&
+             ast_format_get_sample_rate(format) > audiohook->hook_internal_samp_rate)) {
+               audiohook_set_internal_rate(audiohook, ast_format_get_sample_rate(format), 1);
+       }
+
+       /* If the sample rate of the requested format differs from that of the underlying audiohook
+        * sample rate determine how many samples we actually need to get from the audiohook. This
+        * needs to occur as the signed linear factory stores them at the rate of the audiohook.
+        * We do this by determining the duration of audio they've requested and then determining
+        * how many samples that would be in the audiohook format.
+        */
+       if (ast_format_get_sample_rate(format) != audiohook->hook_internal_samp_rate) {
+               samples = (audiohook->hook_internal_samp_rate / 1000) * (samples / (ast_format_get_sample_rate(format) / 1000));
        }
 
        if (!(read_frame = (direction == AST_AUDIOHOOK_DIRECTION_BOTH ?
-               audiohook_read_frame_both(audiohook, samples_converted, read_reference, write_reference) :
-               audiohook_read_frame_single(audiohook, samples_converted, direction)))) {
+               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->id != ast_format_slin_by_rate(audiohook->hook_internal_samp_rate)) {
+       if (ast_format_cmp(format, slin) != AST_FORMAT_CMP_EQUAL) {
                /* Rebuild translation path if different format then previously */
-               if (ast_format_cmp(format, &audiohook->format) == AST_FORMAT_CMP_NOT_EQUAL) {
+               if (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_set(&tmp_fmt, ast_format_slin_by_rate(audiohook->hook_internal_samp_rate), 0)))) {
+                       if (!(audiohook->trans_pvt = ast_translator_build_path(format, slin))) {
                                ast_frfree(read_frame);
                                return NULL;
                        }
-                       ast_format_copy(&audiohook->format, format);
+                       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);
@@ -422,6 +457,22 @@ struct ast_frame *ast_audiohook_read_frame_all(struct ast_audiohook *audiohook,
 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)) {
@@ -452,24 +503,32 @@ int ast_audiohook_attach(struct ast_channel *chan, struct ast_audiohook *audioho
                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 = 8000;
+               ast_channel_audiohooks(chan)->list_internal_samp_rate = DEFAULT_INTERNAL_SAMPLE_RATE;
        }
 
        /* Drop into respective list */
-       if (audiohook->type == AST_AUDIOHOOK_TYPE_SPY)
+       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)
+       } 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)
+       } else if (audiohook->type == AST_AUDIOHOOK_TYPE_MANIPULATE) {
                AST_LIST_INSERT_TAIL(&ast_channel_audiohooks(chan)->manipulate_list, audiohook, list);
+       }
 
-
-       audiohook_set_internal_rate(audiohook, ast_channel_audiohooks(chan)->list_internal_samp_rate, 1);
+       /*
+        * Initialize the audiohook's rate to the default. If it needs to be,
+        * it will get updated later.
+        */
+       audiohook_set_internal_rate(audiohook, DEFAULT_INTERNAL_SAMPLE_RATE, 1);
        audiohook_list_set_samplerate_compatibility(ast_channel_audiohooks(chan));
 
        /* Change status over to running since it is now attached */
        ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_RUNNING);
 
+       if (ast_channel_is_bridged(chan)) {
+               ast_channel_set_unbridged_nolock(chan, 1);
+       }
+
        ast_channel_unlock(chan);
 
        return 0;
@@ -499,25 +558,27 @@ void ast_audiohook_update_status(struct ast_audiohook *audiohook, enum ast_audio
  */
 int ast_audiohook_detach(struct ast_audiohook *audiohook)
 {
-       if (audiohook->status == AST_AUDIOHOOK_STATUS_NEW || audiohook->status == AST_AUDIOHOOK_STATUS_DONE)
+       if (audiohook->status == AST_AUDIOHOOK_STATUS_NEW || audiohook->status == AST_AUDIOHOOK_STATUS_DONE) {
                return 0;
+       }
 
        ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_SHUTDOWN);
 
-       while (audiohook->status != AST_AUDIOHOOK_STATUS_DONE)
+       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 */
        while ((audiohook = AST_LIST_REMOVE_HEAD(&audiohook_list->spy_list, list))) {
@@ -537,16 +598,18 @@ int ast_audiohook_detach_list(struct ast_audiohook_list *audiohook_list)
 
        /* 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
@@ -559,32 +622,30 @@ static struct ast_audiohook *find_audiohook_by_source(struct ast_audiohook_list
        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;
 }
 
-void ast_audiohook_move_by_source(struct ast_channel *old_chan, struct ast_channel *new_chan, const char *source)
+static void audiohook_move(struct ast_channel *old_chan, struct ast_channel *new_chan, struct ast_audiohook *audiohook)
 {
-       struct ast_audiohook *audiohook;
        enum ast_audiohook_status oldstatus;
 
-       if (!ast_channel_audiohooks(old_chan) || !(audiohook = find_audiohook_by_source(ast_channel_audiohooks(old_chan), source))) {
-               return;
-       }
-
        /* 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
@@ -600,6 +661,48 @@ void ast_audiohook_move_by_source(struct ast_channel *old_chan, struct ast_chann
        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
@@ -621,8 +724,9 @@ int ast_audiohook_detach_source(struct ast_channel *chan, const char *source)
 
        ast_channel_unlock(chan);
 
-       if (audiohook && audiohook->status != AST_AUDIOHOOK_STATUS_DONE)
+       if (audiohook && audiohook->status != AST_AUDIOHOOK_STATUS_DONE) {
                ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_SHUTDOWN);
+       }
 
        return (audiohook ? 0 : -1);
 }
@@ -646,16 +750,21 @@ int ast_audiohook_remove(struct ast_channel *chan, struct ast_audiohook *audioho
                return -1;
        }
 
-       if (audiohook->type == AST_AUDIOHOOK_TYPE_SPY)
+       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)
+       } 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)
+       } 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;
@@ -681,10 +790,14 @@ static struct ast_frame *dtmf_audiohook_write_list(struct ast_channel *chan, str
                        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;
@@ -702,33 +815,39 @@ static struct ast_frame *audiohook_list_translate_to_slin(struct ast_audiohook_l
        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 tmp_fmt;
-       enum ast_format_id slin_id;
+       struct ast_format *slin;
 
-       /* If we are capable of maintaining doing samplerates other that 8khz, update
-        * the internal audiohook_list's rate and higher samplerate audio arrives. By
-        * updating the list's rate, all the audiohooks in the list will be updated as well
-        * as the are written and read from. */
-       if (audiohook_list->native_slin_compatible) {
-               audiohook_list->list_internal_samp_rate =
-                       MAX(ast_format_rate(&frame->subclass.format), audiohook_list->list_internal_samp_rate);
-       }
-
-       slin_id = ast_format_slin_by_rate(audiohook_list->list_internal_samp_rate);
+       /*
+        * 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);
 
-       if (frame->subclass.format.id == slin_id) {
+       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 (ast_format_cmp(&frame->subclass.format, &in_translate->format) == AST_FORMAT_CMP_NOT_EQUAL) {
+       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);
                }
-               if (!(in_translate->trans_pvt = ast_translator_build_path(ast_format_set(&tmp_fmt, slin_id, 0), &frame->subclass.format))) {
-                       return NULL;
-               }
-               ast_format_copy(&in_translate->format, &frame->subclass.format);
+               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;
        }
@@ -741,16 +860,16 @@ static struct ast_frame *audiohook_list_translate_to_native(struct ast_audiohook
 {
        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) {
+       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 (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))) {
+                       if (!(out_translate->trans_pvt = ast_translator_build_path(outformat, slin_frame->subclass.format))) {
                                return NULL;
                        }
-                       ast_format_copy(&out_translate->format, outformat);
+                       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))) {
@@ -761,6 +880,36 @@ static struct ast_frame *audiohook_list_translate_to_native(struct ast_audiohook
 }
 
 /*!
+ *\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
@@ -790,13 +939,38 @@ static struct ast_frame *audio_audiohook_write_list(struct ast_channel *chan, st
        int samples;
        int middle_frame_manipulated = 0;
        int removed = 0;
+       int internal_sample_rate;
 
        /* ---Part_1. translate start_frame to SLINEAR if necessary. */
        if (!(middle_frame = audiohook_list_translate_to_slin(audiohook_list, direction, start_frame))) {
                return frame;
        }
+
+       /* If the translation resulted in an interpolated frame then immediately return as audiohooks
+        * rely on actual media being present to do things.
+        */
+       if (!middle_frame->data.ptr) {
+               if (middle_frame != start_frame) {
+                       ast_frfree(middle_frame);
+               }
+               return start_frame;
+       }
+
        samples = middle_frame->samples;
 
+       /*
+        * While processing each audiohook check to see if the internal sample rate needs
+        * to be adjusted (it should be the highest rate specified between formats and
+        * hooks). The given audiohook_list's internal sample rate is then set to the
+        * updated value before returning.
+        *
+        * If slin compatibility mode is turned on then an audiohook's internal sample
+        * rate is set to its audiohook_list's rate. If an audiohook_list's rate is
+        * adjusted during this pass then the change is picked up by the audiohooks
+        * on the next pass.
+        */
+       internal_sample_rate = audiohook_list->list_internal_samp_rate;
+
        /* ---Part_2: Send middle_frame to spy and manipulator lists.  middle_frame is guaranteed to be SLINEAR here.*/
        /* Queue up signed linear frame to each spy */
        AST_LIST_TRAVERSE_SAFE_BEGIN(&audiohook_list->spy_list, audiohook, list) {
@@ -806,33 +980,41 @@ static struct ast_frame *audio_audiohook_write_list(struct ast_channel *chan, st
                        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_set_internal_rate(audiohook, audiohook_list->list_internal_samp_rate, 1);
+               audiohook_list_set_hook_rate(audiohook_list, audiohook, &internal_sample_rate);
                ast_audiohook_write_frame(audiohook, direction, middle_frame);
                ast_audiohook_unlock(audiohook);
        }
        AST_LIST_TRAVERSE_SAFE_END;
 
        /* If this frame is being written out to the channel then we need to use whisper sources */
-       if (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(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_set_internal_rate(audiohook, audiohook_list->list_internal_samp_rate, 1);
-                       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);
                }
@@ -855,26 +1037,32 @@ static struct ast_frame *audio_audiohook_write_list(struct ast_channel *chan, st
                                ast_audiohook_unlock(audiohook);
                                /* We basically drop all of our links to the manipulate audiohook and prod it to do it's own destructive things */
                                audiohook->manipulate_callback(audiohook, chan, NULL, direction);
+                               if (ast_channel_is_bridged(chan)) {
+                                       ast_channel_set_unbridged_nolock(chan, 1);
+                               }
                                continue;
                        }
-                       audiohook_set_internal_rate(audiohook, audiohook_list->list_internal_samp_rate, 1);
-                       /* Feed in frame to manipulation. */
-                       if (audiohook->manipulate_callback(audiohook, chan, middle_frame, direction)) {
-                               /* XXX IGNORE FAILURE */
-
-                               /* If the manipulation fails then the frame will be returned in its original state.
+                       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. */
+                                * be taken here to exit early.
+                                */
+                               middle_frame_manipulated = 1;
                        }
                        ast_audiohook_unlock(audiohook);
                }
                AST_LIST_TRAVERSE_SAFE_END;
-               middle_frame_manipulated = 1;
        }
 
        /* ---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))) {
+               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;
                }
@@ -890,6 +1078,12 @@ static struct ast_frame *audio_audiohook_write_list(struct ast_channel *chan, st
        /* 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;
@@ -897,13 +1091,10 @@ static struct ast_frame *audio_audiohook_write_list(struct ast_channel *chan, st
 
 int ast_audiohook_write_list_empty(struct ast_audiohook_list *audiohook_list)
 {
-       if (AST_LIST_EMPTY(&audiohook_list->spy_list) &&
-               AST_LIST_EMPTY(&audiohook_list->whisper_list) &&
-               AST_LIST_EMPTY(&audiohook_list->manipulate_list)) {
-
-               return 1;
-       }
-       return 0;
+       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
@@ -916,12 +1107,13 @@ int ast_audiohook_write_list_empty(struct ast_audiohook_list *audiohook_list)
 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
@@ -947,8 +1139,9 @@ int ast_channel_audiohook_count_by_source(struct ast_channel *chan, const char *
        int count = 0;
        struct ast_audiohook *ah = NULL;
 
-       if (!ast_channel_audiohooks(chan))
+       if (!ast_channel_audiohooks(chan)) {
                return -1;
+       }
 
        switch (type) {
                case AST_AUDIOHOOK_TYPE_SPY:
@@ -973,7 +1166,7 @@ int ast_channel_audiohook_count_by_source(struct ast_channel *chan, const char *
                        }
                        break;
                default:
-                       ast_debug(1, "Invalid audiohook type supplied, (%d)\n", type);
+                       ast_debug(1, "Invalid audiohook type supplied, (%u)\n", type);
                        return -1;
        }
 
@@ -1008,7 +1201,7 @@ int ast_channel_audiohook_count_by_source_running(struct ast_channel *chan, cons
                        }
                        break;
                default:
-                       ast_debug(1, "Invalid audiohook type supplied, (%d)\n", type);
+                       ast_debug(1, "Invalid audiohook type supplied, (%u)\n", type);
                        return -1;
        }
        return count;