diff --git a/include/asterisk/audiohook.h b/include/asterisk/audiohook.h
index 5e8fc1812047d35d5cfb1f4043104f42b210ba60..38bfd724ee0e025270b6124236401196a5c70a40 100644
--- a/include/asterisk/audiohook.h
+++ b/include/asterisk/audiohook.h
@@ -63,6 +63,7 @@ enum ast_audiohook_flags {
AST_AUDIOHOOK_SMALL_QUEUE = (1 << 4),
AST_AUDIOHOOK_MUTE_READ = (1 << 5), /*!< audiohook should be mute frames read */
AST_AUDIOHOOK_MUTE_WRITE = (1 << 6), /*!< audiohook should be mute frames written */
+ AST_AUDIOHOOK_COMPATIBLE = (1 << 7), /*!< is the audiohook native slin compatible */
};
enum ast_audiohook_init_flags {
diff --git a/main/audiohook.c b/main/audiohook.c
index d81df85e8c91f39fff1a1fe5c0759296d86e26b1..f01a6404539c83c35fd1fffa29a1e7dfc52ed3e8 100644
--- a/main/audiohook.c
+++ b/main/audiohook.c
@@ -45,6 +45,8 @@ 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 DEFAULT_INTERNAL_SAMPLE_RATE 8000
+
struct ast_audiohook_translate {
struct ast_trans_pvt *trans_pvt;
struct ast_format format;
@@ -56,7 +58,17 @@ struct ast_audiohook_list {
* variable will be set and the sample rate will
* be preserved during ast_audiohook_write_list()*/
int native_slin_compatible;
- int list_internal_samp_rate;/*!< Internal sample rate used when writing to the audiohook list */
+
+ /*
+ * The internal sample rate is used when writing to the audiohook lists and
+ * should always be set to the highest value between formats and audiohooks.
+ *
+ * During audiohook invocation (see write_list) if native slin is turned off
+ * the highest sample rate is determined as each audiohook is processed.
+ * If it is turned on then the audiohooks are also updated with the current
+ * highest sample rate.
+ */
+ int list_internal_samp_rate;
struct ast_audiohook_translate in_translate[2];
struct ast_audiohook_translate out_translate[2];
@@ -115,7 +127,7 @@ 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);
@@ -351,7 +363,19 @@ static struct ast_frame *audiohook_read_frame_helper(struct ast_audiohook *audio
struct ast_frame *read_frame = NULL, *final_frame = NULL;
struct ast_format tmp_fmt;
- audiohook_set_internal_rate(audiohook, ast_format_rate(format), 1);
+ /*
+ * 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_rate(format) > audiohook->hook_internal_samp_rate)) {
+ audiohook_set_internal_rate(audiohook, ast_format_rate(format), 1);
+ }
if (!(read_frame = (direction == AST_AUDIOHOOK_DIRECTION_BOTH ?
audiohook_read_frame_both(audiohook, samples, read_reference, write_reference) :
@@ -413,6 +437,22 @@ 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)) {
@@ -443,7 +483,7 @@ 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 */
@@ -455,7 +495,11 @@ 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 */
@@ -646,7 +690,6 @@ int ast_audiohook_remove(struct ast_channel *chan, struct ast_audiohook *audioho
audiohook_list_set_samplerate_compatibility(ast_channel_audiohooks(chan));
ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_DONE);
-
ast_channel_unlock(chan);
return 0;
@@ -696,14 +739,14 @@ static struct ast_frame *audiohook_list_translate_to_slin(struct ast_audiohook_l
struct ast_format tmp_fmt;
enum ast_format_id slin_id;
- /* 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);
- }
+ /*
+ * 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_rate(&frame->subclass.format), audiohook_list->list_internal_samp_rate);
slin_id = ast_format_slin_by_rate(audiohook_list->list_internal_samp_rate);
@@ -751,6 +794,36 @@ static struct ast_frame *audiohook_list_translate_to_native(struct ast_audiohook
return outframe;
}
+/*!
+ *\brief Set the audiohook's internal sample rate to the audiohook_list's rate,
+ * but only when native slin compatibility is turned on.
+ *
+ * \param audiohook_list audiohook_list data object
+ * \param audiohook the audiohook to update
+ * \param rate the current max internal sample rate
+ */
+static void audiohook_list_set_hook_rate(struct ast_audiohook_list *audiohook_list,
+ struct ast_audiohook *audiohook, int *rate)
+{
+ /* The rate should always be the max between itself and the hook */
+ if (audiohook->hook_internal_samp_rate > *rate) {
+ *rate = audiohook->hook_internal_samp_rate;
+ }
+
+ /*
+ * If native slin compatibility is turned on then update the audiohook
+ * with the audiohook_list's current rate. Note, the audiohook's rate is
+ * set to the audiohook_list's rate and not the given rate. If there is
+ * a change in rate the hook's rate is changed on its next check.
+ */
+ if (audiohook_list->native_slin_compatible) {
+ ast_set_flag(audiohook, AST_AUDIOHOOK_COMPATIBLE);
+ audiohook_set_internal_rate(audiohook, audiohook_list->list_internal_samp_rate, 1);
+ } else {
+ ast_clear_flag(audiohook, AST_AUDIOHOOK_COMPATIBLE);
+ }
+}
+
/*!
* \brief Pass an AUDIO frame off to be handled by the audiohook core
*
@@ -781,6 +854,7 @@ 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))) {
@@ -788,6 +862,19 @@ static struct ast_frame *audio_audiohook_write_list(struct ast_channel *chan, st
}
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) {
@@ -799,7 +886,7 @@ static struct ast_frame *audio_audiohook_write_list(struct ast_channel *chan, st
ast_audiohook_unlock(audiohook);
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);
}
@@ -819,8 +906,9 @@ static struct ast_frame *audio_audiohook_write_list(struct ast_channel *chan, st
ast_audiohook_unlock(audiohook);
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(&audiohook->write_factory) >= samples &&
+ ast_slinfactory_read(&audiohook->write_factory, read_buf, samples)) {
/* Take audio from this whisper source and combine it into our main buffer */
for (i = 0, data1 = combine_buf, data2 = read_buf; i < samples; i++, data1++, data2++)
ast_slinear_saturated_add(data1, data2);
@@ -848,15 +936,16 @@ static struct ast_frame *audio_audiohook_write_list(struct ast_channel *chan, st
audiohook->manipulate_callback(audiohook, chan, NULL, direction);
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.
- * Since there are potentially more manipulator callbacks in the list, no action should
- * be taken here to exit early. */
- }
+ audiohook_list_set_hook_rate(audiohook_list, audiohook, &internal_sample_rate);
+ /*
+ * Feed in frame to manipulation.
+ *
+ * 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.
+ */
+ audiohook->manipulate_callback(audiohook, chan, middle_frame, direction);
ast_audiohook_unlock(audiohook);
}
AST_LIST_TRAVERSE_SAFE_END;
@@ -881,6 +970,12 @@ 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;