Interpolated frames are frames which contain a number of samples but have no actual data. Audiohooks did not handle this case when translating an incoming frame into signed linear. It assumed that a frame would always contain media when it may not. If this occurs audiohooks will now immediately return and not act on the frame. As well for users of ast_trans_frameout the function has been changed to be a bit more sane and ensure that the data pointer on a frame is set to NULL if no data is actually on the frame. This allows the various spots in Asterisk that check for an interpolated frame based on the presence of a data pointer to work as expected. ASTERISK-26926 Change-Id: I7fa22f631fa28d540722ed789ce28e84c7f8662b
1428 lines
50 KiB
C
1428 lines
50 KiB
C
/*
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* Asterisk -- An open source telephony toolkit.
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*
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* Copyright (C) 1999 - 2007, Digium, Inc.
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*
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* Joshua Colp <jcolp@digium.com>
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*
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* See http://www.asterisk.org for more information about
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* the Asterisk project. Please do not directly contact
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* any of the maintainers of this project for assistance;
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* the project provides a web site, mailing lists and IRC
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* channels for your use.
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*
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* This program is free software, distributed under the terms of
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* the GNU General Public License Version 2. See the LICENSE file
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* at the top of the source tree.
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*/
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/*! \file
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*
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* \brief Audiohooks Architecture
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*
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* \author Joshua Colp <jcolp@digium.com>
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*/
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/*** MODULEINFO
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<support_level>core</support_level>
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***/
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#include "asterisk.h"
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#include <signal.h>
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#include "asterisk/channel.h"
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#include "asterisk/utils.h"
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#include "asterisk/lock.h"
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#include "asterisk/linkedlists.h"
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#include "asterisk/audiohook.h"
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#include "asterisk/slinfactory.h"
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#include "asterisk/frame.h"
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#include "asterisk/translate.h"
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#include "asterisk/format_cache.h"
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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 AST_AUDIOHOOK_LONG_QUEUE_TOLERANCE 500 /*!< Otheriwise we still don't want the queue to grow indefinitely */
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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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};
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struct ast_audiohook_list {
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/* If all the audiohooks in this list are capable
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* of processing slinear at any sample rate, this
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* variable will be set and the sample rate will
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* be preserved during ast_audiohook_write_list()*/
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int native_slin_compatible;
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int list_internal_samp_rate;/*!< Internal sample rate used when writing to the audiohook list */
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struct ast_audiohook_translate in_translate[2];
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struct ast_audiohook_translate out_translate[2];
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AST_LIST_HEAD_NOLOCK(, ast_audiohook) spy_list;
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AST_LIST_HEAD_NOLOCK(, ast_audiohook) whisper_list;
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AST_LIST_HEAD_NOLOCK(, ast_audiohook) manipulate_list;
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};
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static int audiohook_set_internal_rate(struct ast_audiohook *audiohook, int rate, int reset)
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{
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struct ast_format *slin;
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if (audiohook->hook_internal_samp_rate == rate) {
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return 0;
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}
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audiohook->hook_internal_samp_rate = rate;
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slin = ast_format_cache_get_slin_by_rate(rate);
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/* Setup the factories that are needed for this audiohook type */
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switch (audiohook->type) {
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case AST_AUDIOHOOK_TYPE_SPY:
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case AST_AUDIOHOOK_TYPE_WHISPER:
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if (reset) {
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ast_slinfactory_destroy(&audiohook->read_factory);
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ast_slinfactory_destroy(&audiohook->write_factory);
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}
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ast_slinfactory_init_with_format(&audiohook->read_factory, slin);
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ast_slinfactory_init_with_format(&audiohook->write_factory, slin);
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break;
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default:
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break;
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}
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return 0;
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}
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/*! \brief Initialize an audiohook structure
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*
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* \param audiohook Audiohook structure
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* \param type
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* \param source, init_flags
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*
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* \return Returns 0 on success, -1 on failure
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*/
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int ast_audiohook_init(struct ast_audiohook *audiohook, enum ast_audiohook_type type, const char *source, enum ast_audiohook_init_flags init_flags)
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{
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/* Need to keep the type and source */
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audiohook->type = type;
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audiohook->source = source;
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/* Initialize lock that protects our audiohook */
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ast_mutex_init(&audiohook->lock);
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ast_cond_init(&audiohook->trigger, NULL);
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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, 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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return 0;
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}
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/*! \brief Destroys an audiohook structure
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* \param audiohook Audiohook structure
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* \return Returns 0 on success, -1 on failure
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*/
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int ast_audiohook_destroy(struct ast_audiohook *audiohook)
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{
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/* Drop the factories used by this audiohook type */
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switch (audiohook->type) {
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case AST_AUDIOHOOK_TYPE_SPY:
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case AST_AUDIOHOOK_TYPE_WHISPER:
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ast_slinfactory_destroy(&audiohook->read_factory);
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ast_slinfactory_destroy(&audiohook->write_factory);
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break;
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default:
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break;
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}
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/* Destroy translation path if present */
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if (audiohook->trans_pvt)
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ast_translator_free_path(audiohook->trans_pvt);
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ao2_cleanup(audiohook->format);
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/* Lock and trigger be gone! */
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ast_cond_destroy(&audiohook->trigger);
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ast_mutex_destroy(&audiohook->lock);
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return 0;
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}
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#define SHOULD_MUTE(hook, dir) \
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((ast_test_flag(hook, AST_AUDIOHOOK_MUTE_READ) && (dir == AST_AUDIOHOOK_DIRECTION_READ)) || \
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(ast_test_flag(hook, AST_AUDIOHOOK_MUTE_WRITE) && (dir == AST_AUDIOHOOK_DIRECTION_WRITE)) || \
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(ast_test_flag(hook, AST_AUDIOHOOK_MUTE_READ | AST_AUDIOHOOK_MUTE_WRITE) == (AST_AUDIOHOOK_MUTE_READ | AST_AUDIOHOOK_MUTE_WRITE)))
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/*! \brief Writes a frame into the audiohook structure
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* \param audiohook Audiohook structure
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* \param direction Direction the audio frame came from
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* \param frame Frame to write in
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* \return Returns 0 on success, -1 on failure
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*/
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int ast_audiohook_write_frame(struct ast_audiohook *audiohook, enum ast_audiohook_direction direction, struct ast_frame *frame)
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{
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struct ast_slinfactory *factory = (direction == AST_AUDIOHOOK_DIRECTION_READ ? &audiohook->read_factory : &audiohook->write_factory);
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struct ast_slinfactory *other_factory = (direction == AST_AUDIOHOOK_DIRECTION_READ ? &audiohook->write_factory : &audiohook->read_factory);
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struct timeval *rwtime = (direction == AST_AUDIOHOOK_DIRECTION_READ ? &audiohook->read_time : &audiohook->write_time), previous_time = *rwtime;
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int our_factory_samples;
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int our_factory_ms;
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int other_factory_samples;
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int other_factory_ms;
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/* Update last feeding time to be current */
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*rwtime = ast_tvnow();
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our_factory_samples = ast_slinfactory_available(factory);
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our_factory_ms = ast_tvdiff_ms(*rwtime, previous_time) + (our_factory_samples / (audiohook->hook_internal_samp_rate / 1000));
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other_factory_samples = ast_slinfactory_available(other_factory);
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other_factory_ms = other_factory_samples / (audiohook->hook_internal_samp_rate / 1000);
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if (ast_test_flag(audiohook, AST_AUDIOHOOK_TRIGGER_SYNC) && (our_factory_ms - other_factory_ms > AST_AUDIOHOOK_SYNC_TOLERANCE)) {
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ast_debug(1, "Flushing audiohook %p so it remains in sync\n", audiohook);
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ast_slinfactory_flush(factory);
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ast_slinfactory_flush(other_factory);
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}
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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))) {
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ast_debug(1, "Audiohook %p has stale audio in its factories. Flushing them both\n", audiohook);
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ast_slinfactory_flush(factory);
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ast_slinfactory_flush(other_factory);
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} else if ((our_factory_ms > AST_AUDIOHOOK_LONG_QUEUE_TOLERANCE) || (other_factory_ms > AST_AUDIOHOOK_LONG_QUEUE_TOLERANCE)) {
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ast_debug(1, "Audiohook %p has stale audio in its factories. Flushing them both\n", audiohook);
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ast_slinfactory_flush(factory);
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ast_slinfactory_flush(other_factory);
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}
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/* Write frame out to respective factory */
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ast_slinfactory_feed(factory, frame);
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/* If we need to notify the respective handler of this audiohook, do so */
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if ((ast_test_flag(audiohook, AST_AUDIOHOOK_TRIGGER_MODE) == AST_AUDIOHOOK_TRIGGER_READ) && (direction == AST_AUDIOHOOK_DIRECTION_READ)) {
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ast_cond_signal(&audiohook->trigger);
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} else if ((ast_test_flag(audiohook, AST_AUDIOHOOK_TRIGGER_MODE) == AST_AUDIOHOOK_TRIGGER_WRITE) && (direction == AST_AUDIOHOOK_DIRECTION_WRITE)) {
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ast_cond_signal(&audiohook->trigger);
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} else if (ast_test_flag(audiohook, AST_AUDIOHOOK_TRIGGER_SYNC)) {
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ast_cond_signal(&audiohook->trigger);
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}
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return 0;
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}
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static struct ast_frame *audiohook_read_frame_single(struct ast_audiohook *audiohook, size_t samples, enum ast_audiohook_direction direction)
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{
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struct ast_slinfactory *factory = (direction == AST_AUDIOHOOK_DIRECTION_READ ? &audiohook->read_factory : &audiohook->write_factory);
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int vol = (direction == AST_AUDIOHOOK_DIRECTION_READ ? audiohook->options.read_volume : audiohook->options.write_volume);
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short buf[samples];
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struct ast_frame frame = {
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.frametype = AST_FRAME_VOICE,
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.subclass.format = ast_format_cache_get_slin_by_rate(audiohook->hook_internal_samp_rate),
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.data.ptr = buf,
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.datalen = sizeof(buf),
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.samples = samples,
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};
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/* Ensure the factory is able to give us the samples we want */
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if (samples > ast_slinfactory_available(factory)) {
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return NULL;
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}
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/* Read data in from factory */
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if (!ast_slinfactory_read(factory, buf, samples)) {
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return NULL;
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}
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if (SHOULD_MUTE(audiohook, direction)) {
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/* Swap frame data for zeros if mute is required */
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ast_frame_clear(&frame);
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} else if (vol) {
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/* If a volume adjustment needs to be applied apply it */
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ast_frame_adjust_volume(&frame, vol);
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}
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return ast_frdup(&frame);
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}
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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)
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{
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int count;
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int usable_read;
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int usable_write;
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short adjust_value;
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short buf1[samples];
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short buf2[samples];
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short *read_buf = NULL;
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short *write_buf = NULL;
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struct ast_frame frame = {
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.frametype = AST_FRAME_VOICE,
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.datalen = sizeof(buf1),
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.samples = samples,
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};
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/* Make sure both factories have the required samples */
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usable_read = (ast_slinfactory_available(&audiohook->read_factory) >= samples ? 1 : 0);
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usable_write = (ast_slinfactory_available(&audiohook->write_factory) >= samples ? 1 : 0);
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if (!usable_read && !usable_write) {
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/* If both factories are unusable bail out */
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ast_debug(1, "Read factory %p and write factory %p both fail to provide %zu samples\n", &audiohook->read_factory, &audiohook->write_factory, samples);
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return NULL;
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}
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/* If we want to provide only a read factory make sure we aren't waiting for other audio */
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if (usable_read && !usable_write && (ast_tvdiff_ms(ast_tvnow(), audiohook->write_time) < (samples/8)*2)) {
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ast_debug(3, "Write factory %p was pretty quick last time, waiting for them.\n", &audiohook->write_factory);
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return NULL;
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}
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/* If we want to provide only a write factory make sure we aren't waiting for other audio */
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if (usable_write && !usable_read && (ast_tvdiff_ms(ast_tvnow(), audiohook->read_time) < (samples/8)*2)) {
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ast_debug(3, "Read factory %p was pretty quick last time, waiting for them.\n", &audiohook->read_factory);
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return NULL;
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}
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/* Start with the read factory... if there are enough samples, read them in */
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if (usable_read) {
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if (ast_slinfactory_read(&audiohook->read_factory, buf1, samples)) {
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read_buf = buf1;
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if ((ast_test_flag(audiohook, AST_AUDIOHOOK_MUTE_READ))) {
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/* Clear the frame data if we are muting */
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memset(buf1, 0, sizeof(buf1));
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} else if (audiohook->options.read_volume) {
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/* Adjust read volume if need be */
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adjust_value = abs(audiohook->options.read_volume);
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for (count = 0; count < samples; count++) {
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if (audiohook->options.read_volume > 0) {
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ast_slinear_saturated_multiply(&buf1[count], &adjust_value);
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} else if (audiohook->options.read_volume < 0) {
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ast_slinear_saturated_divide(&buf1[count], &adjust_value);
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}
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}
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}
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}
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} else {
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ast_debug(1, "Failed to get %d samples from read factory %p\n", (int)samples, &audiohook->read_factory);
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}
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/* Move on to the write factory... if there are enough samples, read them in */
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if (usable_write) {
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if (ast_slinfactory_read(&audiohook->write_factory, buf2, samples)) {
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write_buf = buf2;
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if ((ast_test_flag(audiohook, AST_AUDIOHOOK_MUTE_WRITE))) {
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/* Clear the frame data if we are muting */
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memset(buf2, 0, sizeof(buf2));
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} else if (audiohook->options.write_volume) {
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/* Adjust write volume if need be */
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adjust_value = abs(audiohook->options.write_volume);
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for (count = 0; count < samples; count++) {
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if (audiohook->options.write_volume > 0) {
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ast_slinear_saturated_multiply(&buf2[count], &adjust_value);
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} else if (audiohook->options.write_volume < 0) {
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ast_slinear_saturated_divide(&buf2[count], &adjust_value);
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}
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}
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}
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}
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} else {
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ast_debug(1, "Failed to get %d samples from write factory %p\n", (int)samples, &audiohook->write_factory);
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}
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frame.subclass.format = ast_format_cache_get_slin_by_rate(audiohook->hook_internal_samp_rate);
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/* Basically we figure out which buffer to use... and if mixing can be done here */
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if (read_buf && read_reference) {
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frame.data.ptr = read_buf;
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*read_reference = ast_frdup(&frame);
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}
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if (write_buf && write_reference) {
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frame.data.ptr = write_buf;
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*write_reference = ast_frdup(&frame);
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}
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/* Make the correct buffer part of the built frame, so it gets duplicated. */
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if (read_buf) {
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frame.data.ptr = read_buf;
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if (write_buf) {
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for (count = 0; count < samples; count++) {
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ast_slinear_saturated_add(read_buf++, write_buf++);
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}
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}
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} else if (write_buf) {
|
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frame.data.ptr = write_buf;
|
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} else {
|
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return NULL;
|
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}
|
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|
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/* Yahoo, a combined copy of the audio! */
|
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return ast_frdup(&frame);
|
|
}
|
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|
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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)
|
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{
|
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struct ast_frame *read_frame = NULL, *final_frame = NULL;
|
|
struct ast_format *slin;
|
|
|
|
/*
|
|
* 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) ||
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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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}
|
|
|
|
/* 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, read_reference, write_reference) :
|
|
audiohook_read_frame_single(audiohook, samples, direction)))) {
|
|
return NULL;
|
|
}
|
|
|
|
slin = ast_format_cache_get_slin_by_rate(audiohook->hook_internal_samp_rate);
|
|
|
|
/* If they don't want signed linear back out, we'll have to send it through the translation path */
|
|
if (ast_format_cmp(format, slin) != AST_FORMAT_CMP_EQUAL) {
|
|
/* 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, slin))) {
|
|
ast_frfree(read_frame);
|
|
return NULL;
|
|
}
|
|
ao2_replace(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;
|
|
|
|
/*
|
|
* 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)) {
|
|
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 (!ast_channel_audiohooks(chan)) {
|
|
struct ast_audiohook_list *ahlist;
|
|
/* Whoops... allocate a new structure */
|
|
if (!(ahlist = ast_calloc(1, sizeof(*ahlist)))) {
|
|
ast_channel_unlock(chan);
|
|
return -1;
|
|
}
|
|
ast_channel_audiohooks_set(chan, ahlist);
|
|
AST_LIST_HEAD_INIT_NOLOCK(&ast_channel_audiohooks(chan)->spy_list);
|
|
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 = DEFAULT_INTERNAL_SAMPLE_RATE;
|
|
}
|
|
|
|
/* Drop into respective list */
|
|
if (audiohook->type == AST_AUDIOHOOK_TYPE_SPY) {
|
|
AST_LIST_INSERT_TAIL(&ast_channel_audiohooks(chan)->spy_list, audiohook, list);
|
|
} else if (audiohook->type == AST_AUDIOHOOK_TYPE_WHISPER) {
|
|
AST_LIST_INSERT_TAIL(&ast_channel_audiohooks(chan)->whisper_list, audiohook, list);
|
|
} else if (audiohook->type == AST_AUDIOHOOK_TYPE_MANIPULATE) {
|
|
AST_LIST_INSERT_TAIL(&ast_channel_audiohooks(chan)->manipulate_list, audiohook, list);
|
|
}
|
|
|
|
/*
|
|
* 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 */
|
|
ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_RUNNING);
|
|
|
|
if (ast_channel_is_bridged(chan)) {
|
|
ast_channel_set_unbridged_nolock(chan, 1);
|
|
}
|
|
|
|
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;
|
|
}
|
|
|
|
void ast_audiohook_detach_list(struct ast_audiohook_list *audiohook_list)
|
|
{
|
|
int i;
|
|
struct ast_audiohook *audiohook;
|
|
|
|
if (!audiohook_list) {
|
|
return;
|
|
}
|
|
|
|
/* 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);
|
|
ao2_cleanup(audiohook_list->in_translate[i].format);
|
|
}
|
|
if (audiohook_list->out_translate[i].trans_pvt) {
|
|
ast_translator_free_path(audiohook_list->out_translate[i].trans_pvt);
|
|
ao2_cleanup(audiohook_list->in_translate[i].format);
|
|
}
|
|
}
|
|
|
|
/* Free ourselves */
|
|
ast_free(audiohook_list);
|
|
}
|
|
|
|
/*! \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;
|
|
}
|
|
|
|
static void audiohook_move(struct ast_channel *old_chan, struct ast_channel *new_chan, struct ast_audiohook *audiohook)
|
|
{
|
|
enum ast_audiohook_status oldstatus;
|
|
|
|
/* 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);
|
|
}
|
|
|
|
void ast_audiohook_move_by_source(struct ast_channel *old_chan, struct ast_channel *new_chan, const char *source)
|
|
{
|
|
struct ast_audiohook *audiohook;
|
|
|
|
if (!ast_channel_audiohooks(old_chan)) {
|
|
return;
|
|
}
|
|
|
|
audiohook = find_audiohook_by_source(ast_channel_audiohooks(old_chan), source);
|
|
if (!audiohook) {
|
|
return;
|
|
}
|
|
|
|
audiohook_move(old_chan, new_chan, audiohook);
|
|
}
|
|
|
|
void ast_audiohook_move_all(struct ast_channel *old_chan, struct ast_channel *new_chan)
|
|
{
|
|
struct ast_audiohook *audiohook;
|
|
struct ast_audiohook_list *audiohook_list;
|
|
|
|
audiohook_list = ast_channel_audiohooks(old_chan);
|
|
if (!audiohook_list) {
|
|
return;
|
|
}
|
|
|
|
AST_LIST_TRAVERSE_SAFE_BEGIN(&audiohook_list->spy_list, audiohook, list) {
|
|
audiohook_move(old_chan, new_chan, audiohook);
|
|
}
|
|
AST_LIST_TRAVERSE_SAFE_END;
|
|
|
|
AST_LIST_TRAVERSE_SAFE_BEGIN(&audiohook_list->whisper_list, audiohook, list) {
|
|
audiohook_move(old_chan, new_chan, audiohook);
|
|
}
|
|
AST_LIST_TRAVERSE_SAFE_END;
|
|
|
|
AST_LIST_TRAVERSE_SAFE_BEGIN(&audiohook_list->manipulate_list, audiohook, list) {
|
|
audiohook_move(old_chan, new_chan, audiohook);
|
|
}
|
|
AST_LIST_TRAVERSE_SAFE_END;
|
|
}
|
|
|
|
/*! \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 (!ast_channel_audiohooks(chan)) {
|
|
ast_channel_unlock(chan);
|
|
return -1;
|
|
}
|
|
|
|
audiohook = find_audiohook_by_source(ast_channel_audiohooks(chan), 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 (!ast_channel_audiohooks(chan)) {
|
|
ast_channel_unlock(chan);
|
|
return -1;
|
|
}
|
|
|
|
if (audiohook->type == AST_AUDIOHOOK_TYPE_SPY) {
|
|
AST_LIST_REMOVE(&ast_channel_audiohooks(chan)->spy_list, audiohook, list);
|
|
} else if (audiohook->type == AST_AUDIOHOOK_TYPE_WHISPER) {
|
|
AST_LIST_REMOVE(&ast_channel_audiohooks(chan)->whisper_list, audiohook, list);
|
|
} else if (audiohook->type == AST_AUDIOHOOK_TYPE_MANIPULATE) {
|
|
AST_LIST_REMOVE(&ast_channel_audiohooks(chan)->manipulate_list, audiohook, list);
|
|
}
|
|
|
|
audiohook_list_set_samplerate_compatibility(ast_channel_audiohooks(chan));
|
|
ast_audiohook_update_status(audiohook, AST_AUDIOHOOK_STATUS_DONE);
|
|
|
|
if (ast_channel_is_bridged(chan)) {
|
|
ast_channel_set_unbridged_nolock(chan, 1);
|
|
}
|
|
|
|
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);
|
|
if (ast_channel_is_bridged(chan)) {
|
|
ast_channel_set_unbridged_nolock(chan, 1);
|
|
}
|
|
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 *slin;
|
|
|
|
/*
|
|
* 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 (!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);
|
|
}
|
|
in_translate->trans_pvt = new_trans;
|
|
|
|
ao2_replace(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;
|
|
}
|
|
ao2_replace(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 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
|
|
* 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;
|
|
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) {
|
|
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) {
|
|
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);
|
|
if (ast_channel_is_bridged(chan)) {
|
|
ast_channel_set_unbridged_nolock(chan, 1);
|
|
}
|
|
continue;
|
|
}
|
|
audiohook_list_set_hook_rate(audiohook_list, audiohook, &internal_sample_rate);
|
|
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 (!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) {
|
|
struct ast_slinfactory *factory = (direction == AST_AUDIOHOOK_DIRECTION_READ ? &audiohook->read_factory : &audiohook->write_factory);
|
|
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);
|
|
if (ast_channel_is_bridged(chan)) {
|
|
ast_channel_set_unbridged_nolock(chan, 1);
|
|
}
|
|
continue;
|
|
}
|
|
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++) {
|
|
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);
|
|
if (ast_channel_is_bridged(chan)) {
|
|
ast_channel_set_unbridged_nolock(chan, 1);
|
|
}
|
|
continue;
|
|
}
|
|
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)) {
|
|
/*
|
|
* 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;
|
|
}
|
|
ast_audiohook_unlock(audiohook);
|
|
}
|
|
AST_LIST_TRAVERSE_SAFE_END;
|
|
}
|
|
|
|
/* ---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);
|
|
} 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;
|
|
}
|
|
|
|
int ast_audiohook_write_list_empty(struct ast_audiohook_list *audiohook_list)
|
|
{
|
|
return !audiohook_list
|
|
|| (AST_LIST_EMPTY(&audiohook_list->spy_list)
|
|
&& AST_LIST_EMPTY(&audiohook_list->whisper_list)
|
|
&& AST_LIST_EMPTY(&audiohook_list->manipulate_list));
|
|
}
|
|
|
|
/*! \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 (!ast_channel_audiohooks(chan)) {
|
|
return -1;
|
|
}
|
|
|
|
switch (type) {
|
|
case AST_AUDIOHOOK_TYPE_SPY:
|
|
AST_LIST_TRAVERSE(&ast_channel_audiohooks(chan)->spy_list, ah, list) {
|
|
if (!strcmp(ah->source, source)) {
|
|
count++;
|
|
}
|
|
}
|
|
break;
|
|
case AST_AUDIOHOOK_TYPE_WHISPER:
|
|
AST_LIST_TRAVERSE(&ast_channel_audiohooks(chan)->whisper_list, ah, list) {
|
|
if (!strcmp(ah->source, source)) {
|
|
count++;
|
|
}
|
|
}
|
|
break;
|
|
case AST_AUDIOHOOK_TYPE_MANIPULATE:
|
|
AST_LIST_TRAVERSE(&ast_channel_audiohooks(chan)->manipulate_list, ah, list) {
|
|
if (!strcmp(ah->source, source)) {
|
|
count++;
|
|
}
|
|
}
|
|
break;
|
|
default:
|
|
ast_debug(1, "Invalid audiohook type supplied, (%u)\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 (!ast_channel_audiohooks(chan))
|
|
return -1;
|
|
|
|
switch (type) {
|
|
case AST_AUDIOHOOK_TYPE_SPY:
|
|
AST_LIST_TRAVERSE(&ast_channel_audiohooks(chan)->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(&ast_channel_audiohooks(chan)->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(&ast_channel_audiohooks(chan)->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, (%u)\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 (!ast_channel_audiohooks(chan)) {
|
|
ast_channel_unlock(chan);
|
|
return -1;
|
|
}
|
|
|
|
audiohook = find_audiohook_by_source(ast_channel_audiohooks(chan), source);
|
|
|
|
if (audiohook) {
|
|
if (clear) {
|
|
ast_clear_flag(audiohook, flag);
|
|
} else {
|
|
ast_set_flag(audiohook, flag);
|
|
}
|
|
}
|
|
|
|
ast_channel_unlock(chan);
|
|
|
|
return (audiohook ? 0 : -1);
|
|
}
|