#include "asterisk.h"
-ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
-
#include <signal.h>
#include "asterisk/channel.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;
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);
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
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();
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);
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 */
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);
}
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,
};
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);
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);
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);
}
{
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;
}
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)) {
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 */
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 */
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);
}
}
/*!
+ *\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
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) {
}
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);
}
}
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++) {
}
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);
}
/* 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;
projects / asterisk/asterisk.git / blobdiff