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https://github.com/asterisk/asterisk.git
synced 2025-10-12 15:45:18 +00:00
Merge "audiohook.c: Difference in read/write rates caused continuous buffer resets" into 13
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@@ -63,6 +63,7 @@ enum ast_audiohook_flags {
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AST_AUDIOHOOK_SMALL_QUEUE = (1 << 4),
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AST_AUDIOHOOK_MUTE_READ = (1 << 5), /*!< audiohook should be mute frames read */
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AST_AUDIOHOOK_MUTE_WRITE = (1 << 6), /*!< audiohook should be mute frames written */
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AST_AUDIOHOOK_COMPATIBLE = (1 << 7), /*!< is the audiohook native slin compatible */
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};
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enum ast_audiohook_init_flags {
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131
main/audiohook.c
131
main/audiohook.c
@@ -46,6 +46,8 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
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#define AST_AUDIOHOOK_SYNC_TOLERANCE 100 /*!< Tolerance in milliseconds for audiohooks synchronization */
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#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. */
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#define DEFAULT_INTERNAL_SAMPLE_RATE 8000
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struct ast_audiohook_translate {
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struct ast_trans_pvt *trans_pvt;
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struct ast_format *format;
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@@ -117,7 +119,7 @@ int ast_audiohook_init(struct ast_audiohook *audiohook, enum ast_audiohook_type
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audiohook->init_flags = init_flags;
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/* initialize internal rate at 8khz, this will adjust if necessary */
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audiohook_set_internal_rate(audiohook, 8000, 0);
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audiohook_set_internal_rate(audiohook, DEFAULT_INTERNAL_SAMPLE_RATE, 0);
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/* Since we are just starting out... this audiohook is new */
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ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_NEW);
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@@ -361,7 +363,19 @@ static struct ast_frame *audiohook_read_frame_helper(struct ast_audiohook *audio
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struct ast_frame *read_frame = NULL, *final_frame = NULL;
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struct ast_format *slin;
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audiohook_set_internal_rate(audiohook, ast_format_get_sample_rate(format), 1);
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/*
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* Update the rate if compatibility mode is turned off or if it is
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* turned on and the format rate is higher than the current rate.
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*
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* This makes it so any unnecessary rate switching/resetting does
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* not take place and also any associated audiohook_list's internal
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* sample rate maintains the highest sample rate between hooks.
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*/
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if (!ast_test_flag(audiohook, AST_AUDIOHOOK_COMPATIBLE) ||
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(ast_test_flag(audiohook, AST_AUDIOHOOK_COMPATIBLE) &&
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ast_format_get_sample_rate(format) > audiohook->hook_internal_samp_rate)) {
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audiohook_set_internal_rate(audiohook, ast_format_get_sample_rate(format), 1);
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}
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if (!(read_frame = (direction == AST_AUDIOHOOK_DIRECTION_BOTH ?
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audiohook_read_frame_both(audiohook, samples, read_reference, write_reference) :
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@@ -425,6 +439,22 @@ struct ast_frame *ast_audiohook_read_frame_all(struct ast_audiohook *audiohook,
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static void audiohook_list_set_samplerate_compatibility(struct ast_audiohook_list *audiohook_list)
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{
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struct ast_audiohook *ah = NULL;
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/*
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* Anytime the samplerate compatibility is set (attach/remove an audiohook) the
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* list's internal sample rate needs to be reset so that the next time processing
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* through write_list, if needed, it will get updated to the correct rate.
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*
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* A list's internal rate always chooses the higher between its own rate and a
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* given rate. If the current rate is being driven by an audiohook that wanted a
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* higher rate then when this audiohook is removed the list's rate would remain
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* at that level when it should be lower, and with no way to lower it since any
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* rate compared against it would be lower.
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*
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* By setting it back to the lowest rate it can recalulate the new highest rate.
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*/
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audiohook_list->list_internal_samp_rate = DEFAULT_INTERNAL_SAMPLE_RATE;
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audiohook_list->native_slin_compatible = 1;
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AST_LIST_TRAVERSE(&audiohook_list->manipulate_list, ah, list) {
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if (!(ah->init_flags & AST_AUDIOHOOK_MANIPULATE_ALL_RATES)) {
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@@ -455,7 +485,7 @@ int ast_audiohook_attach(struct ast_channel *chan, struct ast_audiohook *audioho
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AST_LIST_HEAD_INIT_NOLOCK(&ast_channel_audiohooks(chan)->whisper_list);
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AST_LIST_HEAD_INIT_NOLOCK(&ast_channel_audiohooks(chan)->manipulate_list);
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/* This sample rate will adjust as necessary when writing to the list. */
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ast_channel_audiohooks(chan)->list_internal_samp_rate = 8000;
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ast_channel_audiohooks(chan)->list_internal_samp_rate = DEFAULT_INTERNAL_SAMPLE_RATE;
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}
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/* Drop into respective list */
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@@ -467,8 +497,11 @@ int ast_audiohook_attach(struct ast_channel *chan, struct ast_audiohook *audioho
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AST_LIST_INSERT_TAIL(&ast_channel_audiohooks(chan)->manipulate_list, audiohook, list);
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}
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audiohook_set_internal_rate(audiohook, ast_channel_audiohooks(chan)->list_internal_samp_rate, 1);
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/*
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* Initialize the audiohook's rate to the default. If it needs to be,
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* it will get updated later.
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*/
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audiohook_set_internal_rate(audiohook, DEFAULT_INTERNAL_SAMPLE_RATE, 1);
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audiohook_list_set_samplerate_compatibility(ast_channel_audiohooks(chan));
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/* Change status over to running since it is now attached */
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@@ -766,14 +799,14 @@ static struct ast_frame *audiohook_list_translate_to_slin(struct ast_audiohook_l
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struct ast_frame *new_frame = frame;
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struct ast_format *slin;
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/* If we are capable of maintaining doing samplerates other that 8khz, update
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* the internal audiohook_list's rate and higher samplerate audio arrives. By
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* updating the list's rate, all the audiohooks in the list will be updated as well
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* as the are written and read from. */
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if (audiohook_list->native_slin_compatible) {
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audiohook_list->list_internal_samp_rate =
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MAX(ast_format_get_sample_rate(frame->subclass.format), audiohook_list->list_internal_samp_rate);
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}
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/*
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* If we are capable of sample rates other that 8khz, update the internal
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* audiohook_list's rate and higher sample rate audio arrives. If native
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* slin compatibility is turned on all audiohooks in the list will be
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* updated as well during read/write processing.
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*/
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audiohook_list->list_internal_samp_rate =
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MAX(ast_format_get_sample_rate(frame->subclass.format), audiohook_list->list_internal_samp_rate);
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slin = ast_format_cache_get_slin_by_rate(audiohook_list->list_internal_samp_rate);
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if (ast_format_cmp(frame->subclass.format, slin) == AST_FORMAT_CMP_EQUAL) {
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@@ -821,6 +854,36 @@ static struct ast_frame *audiohook_list_translate_to_native(struct ast_audiohook
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return outframe;
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}
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/*!
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*\brief Set the audiohook's internal sample rate to the audiohook_list's rate,
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* but only when native slin compatibility is turned on.
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*
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* \param audiohook_list audiohook_list data object
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* \param audiohook the audiohook to update
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* \param rate the current max internal sample rate
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*/
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static void audiohook_list_set_hook_rate(struct ast_audiohook_list *audiohook_list,
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struct ast_audiohook *audiohook, int *rate)
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{
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/* The rate should always be the max between itself and the hook */
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if (audiohook->hook_internal_samp_rate > *rate) {
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*rate = audiohook->hook_internal_samp_rate;
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}
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/*
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* If native slin compatibility is turned on then update the audiohook
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* with the audiohook_list's current rate. Note, the audiohook's rate is
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* set to the audiohook_list's rate and not the given rate. If there is
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* a change in rate the hook's rate is changed on its next check.
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*/
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if (audiohook_list->native_slin_compatible) {
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ast_set_flag(audiohook, AST_AUDIOHOOK_COMPATIBLE);
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audiohook_set_internal_rate(audiohook, audiohook_list->list_internal_samp_rate, 1);
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} else {
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ast_clear_flag(audiohook, AST_AUDIOHOOK_COMPATIBLE);
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}
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}
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/*!
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* \brief Pass an AUDIO frame off to be handled by the audiohook core
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*
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@@ -851,6 +914,7 @@ static struct ast_frame *audio_audiohook_write_list(struct ast_channel *chan, st
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int samples;
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int middle_frame_manipulated = 0;
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int removed = 0;
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int internal_sample_rate;
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/* ---Part_1. translate start_frame to SLINEAR if necessary. */
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if (!(middle_frame = audiohook_list_translate_to_slin(audiohook_list, direction, start_frame))) {
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@@ -858,6 +922,19 @@ static struct ast_frame *audio_audiohook_write_list(struct ast_channel *chan, st
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}
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samples = middle_frame->samples;
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/*
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* While processing each audiohook check to see if the internal sample rate needs
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* to be adjusted (it should be the highest rate specified between formats and
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* hooks). The given audiohook_list's internal sample rate is then set to the
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* updated value before returning.
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*
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* If slin compatibility mode is turned on then an audiohook's internal sample
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* rate is set to its audiohook_list's rate. If an audiohook_list's rate is
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* adjusted during this pass then the change is picked up by the audiohooks
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* on the next pass.
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*/
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internal_sample_rate = audiohook_list->list_internal_samp_rate;
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/* ---Part_2: Send middle_frame to spy and manipulator lists. middle_frame is guaranteed to be SLINEAR here.*/
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/* Queue up signed linear frame to each spy */
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AST_LIST_TRAVERSE_SAFE_BEGIN(&audiohook_list->spy_list, audiohook, list) {
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@@ -872,7 +949,7 @@ static struct ast_frame *audio_audiohook_write_list(struct ast_channel *chan, st
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}
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continue;
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}
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audiohook_set_internal_rate(audiohook, audiohook_list->list_internal_samp_rate, 1);
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audiohook_list_set_hook_rate(audiohook_list, audiohook, &internal_sample_rate);
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ast_audiohook_write_frame(audiohook, direction, middle_frame);
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ast_audiohook_unlock(audiohook);
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}
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@@ -896,7 +973,7 @@ static struct ast_frame *audio_audiohook_write_list(struct ast_channel *chan, st
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}
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continue;
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}
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audiohook_set_internal_rate(audiohook, audiohook_list->list_internal_samp_rate, 1);
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audiohook_list_set_hook_rate(audiohook_list, audiohook, &internal_sample_rate);
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if (ast_slinfactory_available(factory) >= samples && ast_slinfactory_read(factory, read_buf, samples)) {
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/* Take audio from this whisper source and combine it into our main buffer */
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for (i = 0, data1 = combine_buf, data2 = read_buf; i < samples; i++, data1++, data2++) {
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@@ -929,14 +1006,16 @@ static struct ast_frame *audio_audiohook_write_list(struct ast_channel *chan, st
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}
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continue;
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}
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audiohook_set_internal_rate(audiohook, audiohook_list->list_internal_samp_rate, 1);
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/* Feed in frame to manipulation. */
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if (!audiohook->manipulate_callback(audiohook, chan, middle_frame, direction)) {
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/* If the manipulation fails then the frame will be returned in its original state.
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* Since there are potentially more manipulator callbacks in the list, no action should
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* be taken here to exit early. */
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middle_frame_manipulated = 1;
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}
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audiohook_list_set_hook_rate(audiohook_list, audiohook, &internal_sample_rate);
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/*
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* Feed in frame to manipulation.
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*
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* XXX FAILURES ARE IGNORED XXX
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* If the manipulation fails then the frame will be returned in its original state.
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* Since there are potentially more manipulator callbacks in the list, no action should
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* be taken here to exit early.
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*/
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audiohook->manipulate_callback(audiohook, chan, middle_frame, direction);
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ast_audiohook_unlock(audiohook);
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}
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AST_LIST_TRAVERSE_SAFE_END;
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@@ -960,6 +1039,12 @@ static struct ast_frame *audio_audiohook_write_list(struct ast_channel *chan, st
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/* Before returning, if an audiohook got removed, reset samplerate compatibility */
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if (removed) {
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audiohook_list_set_samplerate_compatibility(audiohook_list);
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} else {
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/*
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* Set the audiohook_list's rate to the updated rate. Note that if a hook
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* was removed then the list's internal rate is reset to the default.
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*/
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audiohook_list->list_internal_samp_rate = internal_sample_rate;
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}
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return end_frame;
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