loader: Correct overly strict startup checks.
[asterisk/asterisk.git] / main / audiohook.c
index e01b1ce..04a379f 100644 (file)
@@ -29,8 +29,6 @@
 
 #include "asterisk.h"
 
-ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
-
 #include <signal.h>
 
 #include "asterisk/channel.h"
@@ -45,6 +43,9 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 
 #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;
@@ -117,7 +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);
@@ -155,6 +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
@@ -170,7 +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();
@@ -180,7 +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);
@@ -190,17 +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 */
@@ -241,8 +239,11 @@ static struct ast_frame *audiohook_read_frame_single(struct ast_audiohook *audio
                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);
        }
 
@@ -251,11 +252,16 @@ static struct ast_frame *audiohook_read_frame_single(struct ast_audiohook *audio
 
 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,
        };
@@ -286,10 +292,13 @@ 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) {
                                                ast_slinear_saturated_multiply(&buf1[count], &adjust_value);
@@ -307,10 +316,13 @@ 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) {
                                                ast_slinear_saturated_multiply(&buf2[count], &adjust_value);
@@ -324,34 +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;
-
-       frame.subclass.format = ast_format_cache_get_slin_by_rate(audiohook->hook_internal_samp_rate);
-
        /* Yahoo, a combined copy of the audio! */
        return ast_frdup(&frame);
 }
@@ -360,21 +370,34 @@ static struct ast_frame *audiohook_read_frame_helper(struct ast_audiohook *audio
 {
        struct ast_frame *read_frame = NULL, *final_frame = NULL;
        struct ast_format *slin;
-       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_get_sample_rate(format)) {
-               samples_converted = samples;
-       } else if (audiohook->hook_internal_samp_rate > ast_format_get_sample_rate(format)) {
-               samples_converted = samples * (audiohook->hook_internal_samp_rate / (float) ast_format_get_sample_rate(format));
-       } else {
-               samples_converted = samples * (ast_format_get_sample_rate(format) / (float) audiohook->hook_internal_samp_rate);
+
+       /*
+        * 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;
        }
 
@@ -434,6 +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)) {
@@ -464,7 +503,7 @@ 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 */
@@ -476,8 +515,11 @@ int ast_audiohook_attach(struct ast_channel *chan, struct ast_audiohook *audioho
                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 */
@@ -775,27 +817,34 @@ static struct ast_frame *audiohook_list_translate_to_slin(struct ast_audiohook_l
        struct ast_frame *new_frame = frame;
        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_get_sample_rate(frame->subclass.format), 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);
 
        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(slin, frame->subclass.format))) {
-                       return NULL;
-               }
+               in_translate->trans_pvt = new_trans;
+
                ao2_replace(in_translate->format, frame->subclass.format);
        }
 
@@ -831,6 +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
@@ -860,13 +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) {
@@ -881,7 +985,7 @@ static struct ast_frame *audio_audiohook_write_list(struct ast_channel *chan, st
                        }
                        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);
        }
@@ -905,7 +1009,7 @@ static struct ast_frame *audio_audiohook_write_list(struct ast_channel *chan, st
                                }
                                continue;
                        }
-                       audiohook_set_internal_rate(audiohook, audiohook_list->list_internal_samp_rate, 1);
+                       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++) {
@@ -938,13 +1042,18 @@ static struct ast_frame *audio_audiohook_write_list(struct ast_channel *chan, st
                                }
                                continue;
                        }
-                       audiohook_set_internal_rate(audiohook, audiohook_list->list_internal_samp_rate, 1);
-                       /* Feed in frame to manipulation. */
+                       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)) {
-                               /* If the manipulation fails then the frame will be returned in its original state.
+                               /*
+                                * 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;
+                                * be taken here to exit early.
+                                */
+                               middle_frame_manipulated = 1;
                        }
                        ast_audiohook_unlock(audiohook);
                }
@@ -969,6 +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;