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asterisk/main/audiohook.c

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/*
* Asterisk -- An open source telephony toolkit.
*
* Copyright (C) 1999 - 2007, Digium, Inc.
*
* Joshua Colp <jcolp@digium.com>
*
* See http://www.asterisk.org for more information about
* the Asterisk project. Please do not directly contact
* any of the maintainers of this project for assistance;
* the project provides a web site, mailing lists and IRC
* channels for your use.
*
* This program is free software, distributed under the terms of
* the GNU General Public License Version 2. See the LICENSE file
* at the top of the source tree.
*/
/*! \file
*
* \brief Audiohooks Architecture
*
* \author Joshua Colp <jcolp@digium.com>
*/
#include "asterisk.h"
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include <signal.h>
#include "asterisk/channel.h"
#include "asterisk/utils.h"
#include "asterisk/lock.h"
#include "asterisk/linkedlists.h"
#include "asterisk/audiohook.h"
#include "asterisk/slinfactory.h"
#include "asterisk/frame.h"
#include "asterisk/translate.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. */
struct ast_audiohook_translate {
struct ast_trans_pvt *trans_pvt;
struct ast_format format;
};
struct ast_audiohook_list {
/* If all the audiohooks in this list are capable
* of processing slinear at any sample rate, this
* 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 */
struct ast_audiohook_translate in_translate[2];
struct ast_audiohook_translate out_translate[2];
AST_LIST_HEAD_NOLOCK(, ast_audiohook) spy_list;
AST_LIST_HEAD_NOLOCK(, ast_audiohook) whisper_list;
AST_LIST_HEAD_NOLOCK(, ast_audiohook) manipulate_list;
};
static int audiohook_set_internal_rate(struct ast_audiohook *audiohook, int rate, int reset)
{
struct ast_format slin;
if (audiohook->hook_internal_samp_rate == rate) {
return 0;
}
audiohook->hook_internal_samp_rate = rate;
ast_format_set(&slin, ast_format_slin_by_rate(rate), 0);
/* 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->write_factory);
}
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
* \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)
{
/* Need to keep the type and source */
audiohook->type = type;
audiohook->source = source;
/* Initialize lock that protects our audiohook */
ast_mutex_init(&audiohook->lock);
ast_cond_init(&audiohook->trigger, NULL);
audiohook->init_flags = init_flags;
/* initialize internal rate at 8khz, this will adjust if necessary */
audiohook_set_internal_rate(audiohook, 8000, 0);
/* Since we are just starting out... this audiohook is new */
ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_NEW);
return 0;
}
/*! \brief Destroys an audiohook structure
* \param audiohook Audiohook structure
* \return Returns 0 on success, -1 on failure
*/
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->write_factory);
break;
default:
break;
}
/* Destroy translation path if present */
if (audiohook->trans_pvt)
ast_translator_free_path(audiohook->trans_pvt);
/* Lock and trigger be gone! */
ast_cond_destroy(&audiohook->trigger);
ast_mutex_destroy(&audiohook->lock);
return 0;
}
/*! \brief Writes a frame into the audiohook structure
* \param audiohook Audiohook structure
* \param direction Direction the audio frame came from
* \param frame Frame to write in
* \return Returns 0 on success, -1 on failure
*/
int ast_audiohook_write_frame(struct ast_audiohook *audiohook, enum ast_audiohook_direction direction, struct ast_frame *frame)
{
struct ast_slinfactory *factory = (direction == AST_AUDIOHOOK_DIRECTION_READ ? &audiohook->read_factory : &audiohook->write_factory);
struct ast_slinfactory *other_factory = (direction == AST_AUDIOHOOK_DIRECTION_READ ? &audiohook->write_factory : &audiohook->read_factory);
struct timeval *rwtime = (direction == AST_AUDIOHOOK_DIRECTION_READ ? &audiohook->read_time : &audiohook->write_time), previous_time = *rwtime;
int our_factory_samples;
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();
our_factory_samples = ast_slinfactory_available(factory);
our_factory_ms = ast_tvdiff_ms(*rwtime, previous_time) + (our_factory_samples / (audiohook->hook_internal_samp_rate / 1000));
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)) {
ast_debug(1, "Flushing audiohook %p so it remains in sync\n", audiohook);
ast_slinfactory_flush(factory);
ast_slinfactory_flush(other_factory);
}
if (ast_test_flag(audiohook, AST_AUDIOHOOK_SMALL_QUEUE) && ((our_factory_ms > AST_AUDIOHOOK_SMALL_QUEUE_TOLERANCE) || (other_factory_ms > AST_AUDIOHOOK_SMALL_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);
}
/* 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);
}
/* Write frame out to respective factory */
ast_slinfactory_feed(factory, frame);
/* If we need to notify the respective handler of this audiohook, do so */
if ((ast_test_flag(audiohook, AST_AUDIOHOOK_TRIGGER_MODE) == AST_AUDIOHOOK_TRIGGER_READ) && (direction == AST_AUDIOHOOK_DIRECTION_READ)) {
ast_cond_signal(&audiohook->trigger);
} else if ((ast_test_flag(audiohook, AST_AUDIOHOOK_TRIGGER_MODE) == AST_AUDIOHOOK_TRIGGER_WRITE) && (direction == AST_AUDIOHOOK_DIRECTION_WRITE)) {
ast_cond_signal(&audiohook->trigger);
} else if (ast_test_flag(audiohook, AST_AUDIOHOOK_TRIGGER_SYNC)) {
ast_cond_signal(&audiohook->trigger);
}
return 0;
}
static struct ast_frame *audiohook_read_frame_single(struct ast_audiohook *audiohook, size_t samples, enum ast_audiohook_direction direction)
{
struct ast_slinfactory *factory = (direction == AST_AUDIOHOOK_DIRECTION_READ ? &audiohook->read_factory : &audiohook->write_factory);
int vol = (direction == AST_AUDIOHOOK_DIRECTION_READ ? audiohook->options.read_volume : audiohook->options.write_volume);
short buf[samples];
struct ast_frame frame = {
.frametype = AST_FRAME_VOICE,
.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))
return NULL;
/* Read data in from factory */
if (!ast_slinfactory_read(factory, buf, samples))
return NULL;
/* If a volume adjustment needs to be applied apply it */
if (vol)
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;
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);
usable_write = (ast_slinfactory_available(&audiohook->write_factory) >= samples ? 1 : 0);
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);
return NULL;
}
/* If we want to provide only a read factory make sure we aren't waiting for other audio */
if (usable_read && !usable_write && (ast_tvdiff_ms(ast_tvnow(), audiohook->write_time) < (samples/8)*2)) {
ast_debug(3, "Write factory %p was pretty quick last time, waiting for them.\n", &audiohook->write_factory);
return NULL;
}
/* If we want to provide only a write factory make sure we aren't waiting for other audio */
if (usable_write && !usable_read && (ast_tvdiff_ms(ast_tvnow(), audiohook->read_time) < (samples/8)*2)) {
ast_debug(3, "Read factory %p was pretty quick last time, waiting for them.\n", &audiohook->read_factory);
return NULL;
}
/* Start with the read factory... if there are enough samples, read them in */
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);
for (count = 0; count < samples; count++) {
if (audiohook->options.read_volume > 0)
ast_slinear_saturated_multiply(&buf1[count], &adjust_value);
else if (audiohook->options.read_volume < 0)
ast_slinear_saturated_divide(&buf1[count], &adjust_value);
}
}
}
}
ast_debug(1, "Failed to get %d samples from read factory %p\n", (int)samples, &audiohook->read_factory);
/* Move on to the write factory... if there are enough samples, read them in */
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);
for (count = 0; count < samples; count++) {
if (audiohook->options.write_volume > 0)
ast_slinear_saturated_multiply(&buf2[count], &adjust_value);
else if (audiohook->options.write_volume < 0)
ast_slinear_saturated_divide(&buf2[count], &adjust_value);
}
}
}
}
ast_debug(1, "Failed to get %d samples from write factory %p\n", (int)samples, &audiohook->write_factory);
/* Basically we figure out which buffer to use... and if mixing can be done here */
if (read_buf && read_reference) {
frame.data.ptr = buf1;
*read_reference = ast_frdup(&frame);
}
if (write_buf && write_reference) {
frame.data.ptr = buf2;
*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);
}
final_buf = buf1;
} else if (read_buf) {
final_buf = buf1;
} else if (write_buf) {
final_buf = buf2;
} 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);
}
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);
}
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)))) {
return NULL;
}
/* 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)) {
/* Rebuild translation path if different format then previously */
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)))) {
ast_frfree(read_frame);
return NULL;
}
ast_format_copy(&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);
} else {
final_frame = read_frame;
}
return final_frame;
}
/*! \brief Reads a frame in from the audiohook structure
* \param audiohook Audiohook structure
* \param samples Number of samples wanted in requested output format
* \param direction Direction the audio frame came from
* \param format Format of frame remote side wants back
* \return Returns frame on success, NULL on failure
*/
struct ast_frame *ast_audiohook_read_frame(struct ast_audiohook *audiohook, size_t samples, enum ast_audiohook_direction direction, struct ast_format *format)
{
return audiohook_read_frame_helper(audiohook, samples, direction, format, NULL, NULL);
}
/*! \brief Reads a frame in from the audiohook structure
* \param audiohook Audiohook structure
* \param samples Number of samples wanted
* \param direction Direction the audio frame came from
* \param format Format of frame remote side wants back
* \param read_frame frame pointer for copying read frame data
* \param write_frame frame pointer for copying write frame data
* \return Returns frame on success, NULL on failure
*/
struct ast_frame *ast_audiohook_read_frame_all(struct ast_audiohook *audiohook, size_t samples, struct ast_format *format, struct ast_frame **read_frame, struct ast_frame **write_frame)
{
return audiohook_read_frame_helper(audiohook, samples, AST_AUDIOHOOK_DIRECTION_BOTH, format, read_frame, write_frame);
}
static void audiohook_list_set_samplerate_compatibility(struct ast_audiohook_list *audiohook_list)
{
struct ast_audiohook *ah = NULL;
audiohook_list->native_slin_compatible = 1;
AST_LIST_TRAVERSE(&audiohook_list->manipulate_list, ah, list) {
if (!(ah->init_flags & AST_AUDIOHOOK_MANIPULATE_ALL_RATES)) {
audiohook_list->native_slin_compatible = 0;
return;
}
}
}
/*! \brief Attach audiohook to channel
* \param chan Channel
* \param audiohook Audiohook structure
* \return Returns 0 on success, -1 on failure
*/
int ast_audiohook_attach(struct ast_channel *chan, struct ast_audiohook *audiohook)
{
ast_channel_lock(chan);
if (!chan->audiohooks) {
/* Whoops... allocate a new structure */
if (!(chan->audiohooks = ast_calloc(1, sizeof(*chan->audiohooks)))) {
ast_channel_unlock(chan);
return -1;
}
AST_LIST_HEAD_INIT_NOLOCK(&chan->audiohooks->spy_list);
AST_LIST_HEAD_INIT_NOLOCK(&chan->audiohooks->whisper_list);
AST_LIST_HEAD_INIT_NOLOCK(&chan->audiohooks->manipulate_list);
/* This sample rate will adjust as necessary when writing to the list. */
chan->audiohooks->list_internal_samp_rate = 8000;
}
/* Drop into respective list */
if (audiohook->type == AST_AUDIOHOOK_TYPE_SPY)
AST_LIST_INSERT_TAIL(&chan->audiohooks->spy_list, audiohook, list);
else if (audiohook->type == AST_AUDIOHOOK_TYPE_WHISPER)
AST_LIST_INSERT_TAIL(&chan->audiohooks->whisper_list, audiohook, list);
else if (audiohook->type == AST_AUDIOHOOK_TYPE_MANIPULATE)
AST_LIST_INSERT_TAIL(&chan->audiohooks->manipulate_list, audiohook, list);
audiohook_set_internal_rate(audiohook, chan->audiohooks->list_internal_samp_rate, 1);
audiohook_list_set_samplerate_compatibility(chan->audiohooks);
/* Change status over to running since it is now attached */
ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_RUNNING);
ast_channel_unlock(chan);
return 0;
}
/*! \brief Update audiohook's status
* \param audiohook Audiohook structure
* \param status Audiohook status enum
*
* \note once status is updated to DONE, this function can not be used to set the
* status back to any other setting. Setting DONE effectively locks the status as such.
*/
void ast_audiohook_update_status(struct ast_audiohook *audiohook, enum ast_audiohook_status status)
{
ast_audiohook_lock(audiohook);
if (audiohook->status != AST_AUDIOHOOK_STATUS_DONE) {
audiohook->status = status;
ast_cond_signal(&audiohook->trigger);
}
ast_audiohook_unlock(audiohook);
}
/*! \brief Detach audiohook from channel
* \param audiohook Audiohook structure
* \return Returns 0 on success, -1 on failure
*/
int ast_audiohook_detach(struct ast_audiohook *audiohook)
{
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)
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)
{
int i = 0;
struct ast_audiohook *audiohook = NULL;
/* Drop any spies */
while ((audiohook = AST_LIST_REMOVE_HEAD(&audiohook_list->spy_list, list))) {
ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_DONE);
}
/* Drop any whispering sources */
while ((audiohook = AST_LIST_REMOVE_HEAD(&audiohook_list->whisper_list, list))) {
ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_DONE);
}
/* Drop any manipulaters */
while ((audiohook = AST_LIST_REMOVE_HEAD(&audiohook_list->manipulate_list, list))) {
ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_DONE);
audiohook->manipulate_callback(audiohook, NULL, NULL, 0);
}
/* Drop translation paths if present */
for (i = 0; i < 2; i++) {
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)
ast_translator_free_path(audiohook_list->out_translate[i].trans_pvt);
}
/* Free ourselves */
ast_free(audiohook_list);
return 0;
}
/*! \brief find an audiohook based on its source
* \param audiohook_list The list of audiohooks to search in
* \param source The source of the audiohook we wish to find
* \return Return the corresponding audiohook or NULL if it cannot be found.
*/
static struct ast_audiohook *find_audiohook_by_source(struct ast_audiohook_list *audiohook_list, const char *source)
{
struct ast_audiohook *audiohook = NULL;
AST_LIST_TRAVERSE(&audiohook_list->spy_list, audiohook, list) {
if (!strcasecmp(audiohook->source, source))
return audiohook;
}
AST_LIST_TRAVERSE(&audiohook_list->whisper_list, audiohook, list) {
if (!strcasecmp(audiohook->source, source))
return audiohook;
}
AST_LIST_TRAVERSE(&audiohook_list->manipulate_list, audiohook, list) {
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)
{
struct ast_audiohook *audiohook;
enum ast_audiohook_status oldstatus;
if (!old_chan->audiohooks || !(audiohook = find_audiohook_by_source(old_chan->audiohooks, 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
* condition.
*/
ast_audiohook_lock(audiohook);
oldstatus = audiohook->status;
ast_audiohook_remove(old_chan, audiohook);
ast_audiohook_attach(new_chan, audiohook);
audiohook->status = oldstatus;
ast_audiohook_unlock(audiohook);
}
/*! \brief Detach specified source audiohook from channel
* \param chan Channel to detach from
* \param source Name of source to detach
* \return Returns 0 on success, -1 on failure
*/
int ast_audiohook_detach_source(struct ast_channel *chan, const char *source)
{
struct ast_audiohook *audiohook = NULL;
ast_channel_lock(chan);
/* Ensure the channel has audiohooks on it */
if (!chan->audiohooks) {
ast_channel_unlock(chan);
return -1;
}
audiohook = find_audiohook_by_source(chan->audiohooks, source);
ast_channel_unlock(chan);
if (audiohook && audiohook->status != AST_AUDIOHOOK_STATUS_DONE)
ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_SHUTDOWN);
return (audiohook ? 0 : -1);
}
/*!
* \brief Remove an audiohook from a specified channel
*
* \param chan Channel to remove from
* \param audiohook Audiohook to remove
*
* \return Returns 0 on success, -1 on failure
*
* \note The channel does not need to be locked before calling this function
*/
int ast_audiohook_remove(struct ast_channel *chan, struct ast_audiohook *audiohook)
{
ast_channel_lock(chan);
if (!chan->audiohooks) {
ast_channel_unlock(chan);
return -1;
}
if (audiohook->type == AST_AUDIOHOOK_TYPE_SPY)
AST_LIST_REMOVE(&chan->audiohooks->spy_list, audiohook, list);
else if (audiohook->type == AST_AUDIOHOOK_TYPE_WHISPER)
AST_LIST_REMOVE(&chan->audiohooks->whisper_list, audiohook, list);
else if (audiohook->type == AST_AUDIOHOOK_TYPE_MANIPULATE)
AST_LIST_REMOVE(&chan->audiohooks->manipulate_list, audiohook, list);
audiohook_list_set_samplerate_compatibility(chan->audiohooks);
ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_DONE);
ast_channel_unlock(chan);
return 0;
}
/*! \brief Pass a DTMF frame off to be handled by the audiohook core
* \param chan Channel that the list is coming off of
* \param audiohook_list List of audiohooks
* \param direction Direction frame is coming in from
* \param frame The frame itself
* \return Return frame on success, NULL on failure
*/
static struct ast_frame *dtmf_audiohook_write_list(struct ast_channel *chan, struct ast_audiohook_list *audiohook_list, enum ast_audiohook_direction direction, struct ast_frame *frame)
{
struct ast_audiohook *audiohook = NULL;
int removed = 0;
AST_LIST_TRAVERSE_SAFE_BEGIN(&audiohook_list->manipulate_list, audiohook, list) {
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);
audiohook->manipulate_callback(audiohook, NULL, NULL, 0);
continue;
}
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;
/* if an audiohook got removed, reset samplerate compatibility */
if (removed) {
audiohook_list_set_samplerate_compatibility(audiohook_list);
}
return frame;
}
static struct ast_frame *audiohook_list_translate_to_slin(struct ast_audiohook_list *audiohook_list,
enum ast_audiohook_direction direction, struct ast_frame *frame)
{
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;
/* 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 (frame->subclass.format.id == slin_id) {
return new_frame;
}
if (ast_format_cmp(&frame->subclass.format, &in_translate->format) == AST_FORMAT_CMP_NOT_EQUAL) {
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);
}
if (!(new_frame = ast_translate(in_translate->trans_pvt, frame, 0))) {
return NULL;
}
return new_frame;
}
static struct ast_frame *audiohook_list_translate_to_native(struct ast_audiohook_list *audiohook_list,
enum ast_audiohook_direction direction, struct ast_frame *slin_frame, struct ast_format *outformat)
{
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) {
/* rebuild translators if necessary */
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))) {
return NULL;
}
ast_format_copy(&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))) {
return NULL;
}
}
return outframe;
}
/*!
* \brief Pass an AUDIO frame off to be handled by the audiohook core
*
* \details
* This function has 3 ast_frames and 3 parts to handle each. At the beginning of this
* function all 3 frames, start_frame, middle_frame, and end_frame point to the initial
* input frame.
*
* Part_1: Translate the start_frame into SLINEAR audio if it is not already in that
* format. The result of this part is middle_frame is guaranteed to be in
* SLINEAR format for Part_2.
* Part_2: Send middle_frame off to spies and manipulators. At this point middle_frame is
* either a new frame as result of the translation, or points directly to the start_frame
* because no translation to SLINEAR audio was required.
* Part_3: Translate end_frame's audio back into the format of start frame if necessary. This
* is only necessary if manipulation of middle_frame occurred.
*
* \param chan Channel that the list is coming off of
* \param audiohook_list List of audiohooks
* \param direction Direction frame is coming in from
* \param frame The frame itself
* \return Return frame on success, NULL on failure
*/
static struct ast_frame *audio_audiohook_write_list(struct ast_channel *chan, struct ast_audiohook_list *audiohook_list, enum ast_audiohook_direction direction, struct ast_frame *frame)
{
struct ast_frame *start_frame = frame, *middle_frame = frame, *end_frame = frame;
struct ast_audiohook *audiohook = NULL;
int samples;
int middle_frame_manipulated = 0;
int removed = 0;
/* ---Part_1. translate start_frame to SLINEAR if necessary. */
if (!(middle_frame = audiohook_list_translate_to_slin(audiohook_list, direction, start_frame))) {
return frame;
}
samples = middle_frame->samples;
/* ---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) {
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);
continue;
}
audiohook_set_internal_rate(audiohook, audiohook_list->list_internal_samp_rate, 1);
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)) {
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) {
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);
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)) {
/* 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);
}
ast_audiohook_unlock(audiohook);
}
AST_LIST_TRAVERSE_SAFE_END;
/* We take all of the combined whisper sources and combine them into the audio being written out */
for (i = 0, data1 = middle_frame->data.ptr, data2 = combine_buf; i < samples; i++, data1++, data2++) {
ast_slinear_saturated_add(data1, data2);
}
middle_frame_manipulated = 1;
}
/* Pass off frame to manipulate audiohooks */
if (!AST_LIST_EMPTY(&audiohook_list->manipulate_list)) {
AST_LIST_TRAVERSE_SAFE_BEGIN(&audiohook_list->manipulate_list, audiohook, list) {
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);
/* 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);
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. */
}
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))) {
/* translation failed, so just pass back the input frame */
end_frame = start_frame;
}
} else {
end_frame = start_frame;
}
/* clean up our middle_frame if required */
if (middle_frame != end_frame) {
ast_frfree(middle_frame);
middle_frame = NULL;
}
/* Before returning, if an audiohook got removed, reset samplerate compatibility */
if (removed) {
audiohook_list_set_samplerate_compatibility(audiohook_list);
}
return end_frame;
}
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;
}
/*! \brief Pass a frame off to be handled by the audiohook core
* \param chan Channel that the list is coming off of
* \param audiohook_list List of audiohooks
* \param direction Direction frame is coming in from
* \param frame The frame itself
* \return Return frame on success, NULL on failure
*/
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)
return audio_audiohook_write_list(chan, audiohook_list, direction, frame);
else if (frame->frametype == AST_FRAME_DTMF)
return dtmf_audiohook_write_list(chan, audiohook_list, direction, frame);
else
return frame;
}
/*! \brief Wait for audiohook trigger to be triggered
* \param audiohook Audiohook to wait on
*/
void ast_audiohook_trigger_wait(struct ast_audiohook *audiohook)
{
struct timeval wait;
struct timespec ts;
wait = ast_tvadd(ast_tvnow(), ast_samp2tv(50000, 1000));
ts.tv_sec = wait.tv_sec;
ts.tv_nsec = wait.tv_usec * 1000;
ast_cond_timedwait(&audiohook->trigger, &audiohook->lock, &ts);
return;
}
/* Count number of channel audiohooks by type, regardless of type */
int ast_channel_audiohook_count_by_source(struct ast_channel *chan, const char *source, enum ast_audiohook_type type)
{
int count = 0;
struct ast_audiohook *ah = NULL;
if (!chan->audiohooks)
return -1;
switch (type) {
case AST_AUDIOHOOK_TYPE_SPY:
AST_LIST_TRAVERSE(&chan->audiohooks->spy_list, ah, list) {
if (!strcmp(ah->source, source)) {
count++;
}
}
break;
case AST_AUDIOHOOK_TYPE_WHISPER:
AST_LIST_TRAVERSE(&chan->audiohooks->whisper_list, ah, list) {
if (!strcmp(ah->source, source)) {
count++;
}
}
break;
case AST_AUDIOHOOK_TYPE_MANIPULATE:
AST_LIST_TRAVERSE(&chan->audiohooks->manipulate_list, ah, list) {
if (!strcmp(ah->source, source)) {
count++;
}
}
break;
default:
ast_debug(1, "Invalid audiohook type supplied, (%d)\n", type);
return -1;
}
return count;
}
/* Count number of channel audiohooks by type that are running */
int ast_channel_audiohook_count_by_source_running(struct ast_channel *chan, const char *source, enum ast_audiohook_type type)
{
int count = 0;
struct ast_audiohook *ah = NULL;
if (!chan->audiohooks)
return -1;
switch (type) {
case AST_AUDIOHOOK_TYPE_SPY:
AST_LIST_TRAVERSE(&chan->audiohooks->spy_list, ah, list) {
if ((!strcmp(ah->source, source)) && (ah->status == AST_AUDIOHOOK_STATUS_RUNNING))
count++;
}
break;
case AST_AUDIOHOOK_TYPE_WHISPER:
AST_LIST_TRAVERSE(&chan->audiohooks->whisper_list, ah, list) {
if ((!strcmp(ah->source, source)) && (ah->status == AST_AUDIOHOOK_STATUS_RUNNING))
count++;
}
break;
case AST_AUDIOHOOK_TYPE_MANIPULATE:
AST_LIST_TRAVERSE(&chan->audiohooks->manipulate_list, ah, list) {
if ((!strcmp(ah->source, source)) && (ah->status == AST_AUDIOHOOK_STATUS_RUNNING))
count++;
}
break;
default:
ast_debug(1, "Invalid audiohook type supplied, (%d)\n", type);
return -1;
}
return count;
}
/*! \brief Audiohook volume adjustment structure */
struct audiohook_volume {
struct ast_audiohook audiohook; /*!< Audiohook attached to the channel */
int read_adjustment; /*!< Value to adjust frames read from the channel by */
int write_adjustment; /*!< Value to adjust frames written to the channel by */
};
/*! \brief Callback used to destroy the audiohook volume datastore
* \param data Volume information structure
* \return Returns nothing
*/
static void audiohook_volume_destroy(void *data)
{
struct audiohook_volume *audiohook_volume = data;
/* Destroy the audiohook as it is no longer in use */
ast_audiohook_destroy(&audiohook_volume->audiohook);
/* Finally free ourselves, we are of no more use */
ast_free(audiohook_volume);
return;
}
/*! \brief Datastore used to store audiohook volume information */
static const struct ast_datastore_info audiohook_volume_datastore = {
.type = "Volume",
.destroy = audiohook_volume_destroy,
};
/*! \brief Helper function which actually gets called by audiohooks to perform the adjustment
* \param audiohook Audiohook attached to the channel
* \param chan Channel we are attached to
* \param frame Frame of audio we want to manipulate
* \param direction Direction the audio came in from
* \return Returns 0 on success, -1 on failure
*/
static int audiohook_volume_callback(struct ast_audiohook *audiohook, struct ast_channel *chan, struct ast_frame *frame, enum ast_audiohook_direction direction)
{
struct ast_datastore *datastore = NULL;
struct audiohook_volume *audiohook_volume = NULL;
int *gain = NULL;
/* If the audiohook is shutting down don't even bother */
if (audiohook->status == AST_AUDIOHOOK_STATUS_DONE) {
return 0;
}
/* Try to find the datastore containg adjustment information, if we can't just bail out */
if (!(datastore = ast_channel_datastore_find(chan, &audiohook_volume_datastore, NULL))) {
return 0;
}
audiohook_volume = datastore->data;
/* Based on direction grab the appropriate adjustment value */
if (direction == AST_AUDIOHOOK_DIRECTION_READ) {
gain = &audiohook_volume->read_adjustment;
} else if (direction == AST_AUDIOHOOK_DIRECTION_WRITE) {
gain = &audiohook_volume->write_adjustment;
}
/* If an adjustment value is present modify the frame */
if (gain && *gain) {
ast_frame_adjust_volume(frame, *gain);
}
return 0;
}
/*! \brief Helper function which finds and optionally creates an audiohook_volume_datastore datastore on a channel
* \param chan Channel to look on
* \param create Whether to create the datastore if not found
* \return Returns audiohook_volume structure on success, NULL on failure
*/
static struct audiohook_volume *audiohook_volume_get(struct ast_channel *chan, int create)
{
struct ast_datastore *datastore = NULL;
struct audiohook_volume *audiohook_volume = NULL;
/* If we are able to find the datastore return the contents (which is actually an audiohook_volume structure) */
if ((datastore = ast_channel_datastore_find(chan, &audiohook_volume_datastore, NULL))) {
return datastore->data;
}
/* If we are not allowed to create a datastore or if we fail to create a datastore, bail out now as we have nothing for them */
if (!create || !(datastore = ast_datastore_alloc(&audiohook_volume_datastore, NULL))) {
return NULL;
}
/* Create a new audiohook_volume structure to contain our adjustments and audiohook */
if (!(audiohook_volume = ast_calloc(1, sizeof(*audiohook_volume)))) {
ast_datastore_free(datastore);
return NULL;
}
/* Setup our audiohook structure so we can manipulate the audio */
ast_audiohook_init(&audiohook_volume->audiohook, AST_AUDIOHOOK_TYPE_MANIPULATE, "Volume", AST_AUDIOHOOK_MANIPULATE_ALL_RATES);
audiohook_volume->audiohook.manipulate_callback = audiohook_volume_callback;
/* Attach the audiohook_volume blob to the datastore and attach to the channel */
datastore->data = audiohook_volume;
ast_channel_datastore_add(chan, datastore);
/* All is well... put the audiohook into motion */
ast_audiohook_attach(chan, &audiohook_volume->audiohook);
return audiohook_volume;
}
/*! \brief Adjust the volume on frames read from or written to a channel
* \param chan Channel to muck with
* \param direction Direction to set on
* \param volume Value to adjust the volume by
* \return Returns 0 on success, -1 on failure
*/
int ast_audiohook_volume_set(struct ast_channel *chan, enum ast_audiohook_direction direction, int volume)
{
struct audiohook_volume *audiohook_volume = NULL;
/* Attempt to find the audiohook volume information, but only create it if we are not setting the adjustment value to zero */
if (!(audiohook_volume = audiohook_volume_get(chan, (volume ? 1 : 0)))) {
return -1;
}
/* Now based on the direction set the proper value */
if (direction == AST_AUDIOHOOK_DIRECTION_READ || direction == AST_AUDIOHOOK_DIRECTION_BOTH) {
audiohook_volume->read_adjustment = volume;
}
if (direction == AST_AUDIOHOOK_DIRECTION_WRITE || direction == AST_AUDIOHOOK_DIRECTION_BOTH) {
audiohook_volume->write_adjustment = volume;
}
return 0;
}
/*! \brief Retrieve the volume adjustment value on frames read from or written to a channel
* \param chan Channel to retrieve volume adjustment from
* \param direction Direction to retrieve
* \return Returns adjustment value
*/
int ast_audiohook_volume_get(struct ast_channel *chan, enum ast_audiohook_direction direction)
{
struct audiohook_volume *audiohook_volume = NULL;
int adjustment = 0;
/* Attempt to find the audiohook volume information, but do not create it as we only want to look at the values */
if (!(audiohook_volume = audiohook_volume_get(chan, 0))) {
return 0;
}
/* Grab the adjustment value based on direction given */
if (direction == AST_AUDIOHOOK_DIRECTION_READ) {
adjustment = audiohook_volume->read_adjustment;
} else if (direction == AST_AUDIOHOOK_DIRECTION_WRITE) {
adjustment = audiohook_volume->write_adjustment;
}
return adjustment;
}
/*! \brief Adjust the volume on frames read from or written to a channel
* \param chan Channel to muck with
* \param direction Direction to increase
* \param volume Value to adjust the adjustment by
* \return Returns 0 on success, -1 on failure
*/
int ast_audiohook_volume_adjust(struct ast_channel *chan, enum ast_audiohook_direction direction, int volume)
{
struct audiohook_volume *audiohook_volume = NULL;
/* Attempt to find the audiohook volume information, and create an audiohook if none exists */
if (!(audiohook_volume = audiohook_volume_get(chan, 1))) {
return -1;
}
/* Based on the direction change the specific adjustment value */
if (direction == AST_AUDIOHOOK_DIRECTION_READ || direction == AST_AUDIOHOOK_DIRECTION_BOTH) {
audiohook_volume->read_adjustment += volume;
}
if (direction == AST_AUDIOHOOK_DIRECTION_WRITE || direction == AST_AUDIOHOOK_DIRECTION_BOTH) {
audiohook_volume->write_adjustment += volume;
}
return 0;
}
/*! \brief Mute frames read from or written to a channel
* \param chan Channel to muck with
* \param source Type of audiohook
* \param flag which flag to set / clear
* \param clear set or clear
* \return Returns 0 on success, -1 on failure
*/
int ast_audiohook_set_mute(struct ast_channel *chan, const char *source, enum ast_audiohook_flags flag, int clear)
{
struct ast_audiohook *audiohook = NULL;
ast_channel_lock(chan);
/* Ensure the channel has audiohooks on it */
if (!chan->audiohooks) {
ast_channel_unlock(chan);
return -1;
}
audiohook = find_audiohook_by_source(chan->audiohooks, source);
if (audiohook) {
if (clear) {
ast_clear_flag(audiohook, flag);
} else {
ast_set_flag(audiohook, flag);
}
}
ast_channel_unlock(chan);
return (audiohook ? 0 : -1);
}
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