git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@78649 65c4cc65-6c06-0410-ace0-fbb531ad65f3
626 lines
22 KiB
C
626 lines
22 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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#include "asterisk.h"
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ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
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#include <stdio.h>
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#include <stdlib.h>
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#include <string.h>
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#include <signal.h>
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#include <errno.h>
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#include <unistd.h>
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#include "asterisk/logger.h"
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#include "asterisk/channel.h"
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#include "asterisk/options.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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struct ast_audiohook_translate {
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struct ast_trans_pvt *trans_pvt;
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int format;
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};
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struct ast_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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/*! \brief Initialize 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_init(struct ast_audiohook *audiohook, enum ast_audiohook_type type, const char *source)
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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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/* Setup the factories that are needed for this audiohook type */
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switch (type) {
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case AST_AUDIOHOOK_TYPE_SPY:
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ast_slinfactory_init(&audiohook->read_factory);
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case AST_AUDIOHOOK_TYPE_WHISPER:
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ast_slinfactory_init(&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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/* Since we are just starting out... this audiohook is new */
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audiohook->status = 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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ast_slinfactory_destroy(&audiohook->read_factory);
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case AST_AUDIOHOOK_TYPE_WHISPER:
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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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/* 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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/*! \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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/* 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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switch (ast_test_flag(audiohook, AST_AUDIOHOOK_TRIGGER_MODE)) {
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case AST_AUDIOHOOK_TRIGGER_READ:
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if (direction == AST_AUDIOHOOK_DIRECTION_READ)
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ast_cond_signal(&audiohook->trigger);
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break;
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case AST_AUDIOHOOK_TRIGGER_WRITE:
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if (direction == AST_AUDIOHOOK_DIRECTION_WRITE)
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ast_cond_signal(&audiohook->trigger);
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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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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 = AST_FORMAT_SLINEAR,
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.data = 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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/* 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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/* If a volume adjustment needs to be applied apply it */
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if (vol)
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ast_frame_adjust_volume(&frame, vol);
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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)
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{
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int i = 0;
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short buf1[samples], buf2[samples], *read_buf = NULL, *write_buf = NULL, *final_buf = NULL, *data1 = NULL, *data2 = NULL;
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struct ast_frame frame = {
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.frametype = AST_FRAME_VOICE,
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.subclass = AST_FORMAT_SLINEAR,
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.data = NULL,
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.datalen = sizeof(buf1),
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.samples = samples,
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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 (ast_slinfactory_available(&audiohook->read_factory) >= samples) {
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if (ast_slinfactory_read(&audiohook->read_factory, buf1, samples))
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read_buf = buf1;
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/* Adjust read volume if need be */
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if (audiohook->options.read_volume) {
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int count = 0;
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short 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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} else if (option_debug)
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ast_log(LOG_DEBUG, "Failed to get %zd samples from read factory %p\n", samples, &audiohook->read_factory);
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/* Move on to the write factory... if there are enough samples, read them in */
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if (ast_slinfactory_available(&audiohook->write_factory) >= samples) {
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if (ast_slinfactory_read(&audiohook->write_factory, buf2, samples))
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write_buf = buf2;
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/* Adjust write volume if need be */
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if (audiohook->options.write_volume) {
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int count = 0;
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short 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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} else if (option_debug)
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ast_log(LOG_DEBUG, "Failed to get %zd samples from write factory %p\n", samples, &audiohook->write_factory);
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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 && !write_buf)
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return NULL;
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else if (read_buf && write_buf) {
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for (i = 0, data1 = read_buf, data2 = write_buf; i < samples; i++, data1++, data2++)
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ast_slinear_saturated_add(data1, data2);
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final_buf = buf1;
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} else if (read_buf)
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final_buf = buf1;
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else if (write_buf)
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final_buf = buf2;
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/* Make the final buffer part of the frame, so it gets duplicated fine */
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frame.data = final_buf;
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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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/*! \brief Reads a frame in from the audiohook structure
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* \param audiohook Audiohook structure
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* \param samples Number of samples wanted
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* \param direction Direction the audio frame came from
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* \param format Format of frame remote side wants back
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* \return Returns frame on success, NULL on failure
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*/
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struct ast_frame *ast_audiohook_read_frame(struct ast_audiohook *audiohook, size_t samples, enum ast_audiohook_direction direction, int format)
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{
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struct ast_frame *read_frame = NULL, *final_frame = NULL;
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if (!(read_frame = (direction == AST_AUDIOHOOK_DIRECTION_BOTH ? audiohook_read_frame_both(audiohook, samples) : audiohook_read_frame_single(audiohook, samples, direction))))
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return NULL;
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/* If they don't want signed linear back out, we'll have to send it through the translation path */
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if (format != AST_FORMAT_SLINEAR) {
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/* Rebuild translation path if different format then previously */
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if (audiohook->format != format) {
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if (audiohook->trans_pvt) {
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ast_translator_free_path(audiohook->trans_pvt);
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audiohook->trans_pvt = NULL;
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}
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/* 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 */
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if (!(audiohook->trans_pvt = ast_translator_build_path(format, AST_FORMAT_SLINEAR))) {
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ast_frfree(read_frame);
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return NULL;
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}
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}
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/* Convert to requested format, and allow the read in frame to be freed */
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final_frame = ast_translate(audiohook->trans_pvt, read_frame, 1);
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} else {
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final_frame = read_frame;
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}
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return final_frame;
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}
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/*! \brief Attach audiohook to channel
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* \param chan Channel
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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_attach(struct ast_channel *chan, struct ast_audiohook *audiohook)
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{
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ast_channel_lock(chan);
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if (!chan->audiohooks) {
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/* Whoops... allocate a new structure */
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if (!(chan->audiohooks = ast_calloc(1, sizeof(*chan->audiohooks)))) {
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ast_channel_unlock(chan);
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return -1;
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}
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AST_LIST_HEAD_INIT_NOLOCK(&chan->audiohooks->spy_list);
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AST_LIST_HEAD_INIT_NOLOCK(&chan->audiohooks->whisper_list);
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AST_LIST_HEAD_INIT_NOLOCK(&chan->audiohooks->manipulate_list);
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}
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/* Drop into respective list */
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if (audiohook->type == AST_AUDIOHOOK_TYPE_SPY)
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AST_LIST_INSERT_TAIL(&chan->audiohooks->spy_list, audiohook, list);
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else if (audiohook->type == AST_AUDIOHOOK_TYPE_WHISPER)
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AST_LIST_INSERT_TAIL(&chan->audiohooks->whisper_list, audiohook, list);
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else if (audiohook->type == AST_AUDIOHOOK_TYPE_MANIPULATE)
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AST_LIST_INSERT_TAIL(&chan->audiohooks->manipulate_list, audiohook, list);
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/* Change status over to running since it is now attached */
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audiohook->status = AST_AUDIOHOOK_STATUS_RUNNING;
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ast_channel_unlock(chan);
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return 0;
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}
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/*! \brief Detach audiohook from channel
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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_detach(struct ast_audiohook *audiohook)
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{
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if (audiohook->status == AST_AUDIOHOOK_STATUS_DONE)
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return 0;
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audiohook->status = AST_AUDIOHOOK_STATUS_SHUTDOWN;
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while (audiohook->status != AST_AUDIOHOOK_STATUS_DONE)
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ast_audiohook_trigger_wait(audiohook);
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return 0;
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}
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/*! \brief Detach audiohooks from list and destroy said list
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* \param audiohook_list List of audiohooks
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* \return Returns 0 on success, -1 on failure
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*/
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int ast_audiohook_detach_list(struct ast_audiohook_list *audiohook_list)
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{
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int i = 0;
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struct ast_audiohook *audiohook = NULL;
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/* Drop any spies */
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AST_LIST_TRAVERSE_SAFE_BEGIN(&audiohook_list->spy_list, audiohook, list) {
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ast_audiohook_lock(audiohook);
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AST_LIST_REMOVE_CURRENT(&audiohook_list->spy_list, list);
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audiohook->status = AST_AUDIOHOOK_STATUS_DONE;
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ast_cond_signal(&audiohook->trigger);
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ast_audiohook_unlock(audiohook);
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}
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AST_LIST_TRAVERSE_SAFE_END
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/* Drop any whispering sources */
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AST_LIST_TRAVERSE_SAFE_BEGIN(&audiohook_list->whisper_list, audiohook, list) {
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ast_audiohook_lock(audiohook);
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AST_LIST_REMOVE_CURRENT(&audiohook_list->whisper_list, list);
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audiohook->status = AST_AUDIOHOOK_STATUS_DONE;
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ast_cond_signal(&audiohook->trigger);
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ast_audiohook_unlock(audiohook);
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}
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AST_LIST_TRAVERSE_SAFE_END
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/* Drop any manipulaters */
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AST_LIST_TRAVERSE_SAFE_BEGIN(&audiohook_list->manipulate_list, audiohook, list) {
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ast_audiohook_lock(audiohook);
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ast_mutex_lock(&audiohook->lock);
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AST_LIST_REMOVE_CURRENT(&audiohook_list->manipulate_list, list);
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audiohook->status = AST_AUDIOHOOK_STATUS_DONE;
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ast_audiohook_unlock(audiohook);
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audiohook->manipulate_callback(audiohook, NULL, NULL, 0);
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}
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AST_LIST_TRAVERSE_SAFE_END
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/* Drop translation paths if present */
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for (i = 0; i < 2; i++) {
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if (audiohook_list->in_translate[i].trans_pvt)
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ast_translator_free_path(audiohook_list->in_translate[i].trans_pvt);
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if (audiohook_list->out_translate[i].trans_pvt)
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ast_translator_free_path(audiohook_list->out_translate[i].trans_pvt);
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}
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/* Free ourselves */
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ast_free(audiohook_list);
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return 0;
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}
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static struct ast_audiohook *find_audiohook_by_source(struct ast_audiohook_list *audiohook_list, const char *source)
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{
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struct ast_audiohook *audiohook = NULL;
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AST_LIST_TRAVERSE(&audiohook_list->spy_list, audiohook, list) {
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if (!strcasecmp(audiohook->source, source))
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return audiohook;
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}
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AST_LIST_TRAVERSE(&audiohook_list->whisper_list, audiohook, list) {
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if (!strcasecmp(audiohook->source, source))
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return audiohook;
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}
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AST_LIST_TRAVERSE(&audiohook_list->manipulate_list, audiohook, list) {
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if (!strcasecmp(audiohook->source, source))
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return audiohook;
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}
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return NULL;
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}
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/*! \brief Detach specified source audiohook from channel
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* \param chan Channel to detach from
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* \param source Name of source to detach
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* \return Returns 0 on success, -1 on failure
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*/
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int ast_audiohook_detach_source(struct ast_channel *chan, const char *source)
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{
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struct ast_audiohook *audiohook = NULL;
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ast_channel_lock(chan);
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/* Ensure the channel has audiohooks on it */
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if (!chan->audiohooks) {
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ast_channel_unlock(chan);
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return -1;
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}
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audiohook = find_audiohook_by_source(chan->audiohooks, source);
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ast_channel_unlock(chan);
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if (audiohook && audiohook->status != AST_AUDIOHOOK_STATUS_DONE)
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audiohook->status = AST_AUDIOHOOK_STATUS_SHUTDOWN;
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return (audiohook ? 0 : -1);
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}
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/*! \brief Pass a DTMF frame off to be handled by the audiohook core
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* \param chan Channel that the list is coming off of
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* \param audiohook_list List of audiohooks
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* \param direction Direction frame is coming in from
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* \param frame The frame itself
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* \return Return frame on success, NULL on failure
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*/
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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)
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{
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struct ast_audiohook *audiohook = NULL;
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AST_LIST_TRAVERSE_SAFE_BEGIN(&audiohook_list->manipulate_list, audiohook, list) {
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ast_audiohook_lock(audiohook);
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if (audiohook->status != AST_AUDIOHOOK_STATUS_RUNNING) {
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AST_LIST_REMOVE_CURRENT(&audiohook_list->manipulate_list, list);
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audiohook->status = AST_AUDIOHOOK_STATUS_DONE;
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ast_audiohook_unlock(audiohook);
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audiohook->manipulate_callback(audiohook, NULL, NULL, 0);
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continue;
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}
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if (ast_test_flag(audiohook, AST_AUDIOHOOK_WANTS_DTMF))
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audiohook->manipulate_callback(audiohook, chan, frame, direction);
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ast_audiohook_unlock(audiohook);
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}
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AST_LIST_TRAVERSE_SAFE_END
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return frame;
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}
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/*! \brief Pass an AUDIO frame off to be handled by the audiohook core
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* \param chan Channel that the list is coming off of
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* \param audiohook_list List of audiohooks
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* \param direction Direction frame is coming in from
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* \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_audiohook_translate *in_translate = (direction == AST_AUDIOHOOK_DIRECTION_READ ? &audiohook_list->in_translate[0] : &audiohook_list->in_translate[1]);
|
|
struct ast_audiohook_translate *out_translate = (direction == AST_AUDIOHOOK_DIRECTION_READ ? &audiohook_list->out_translate[0] : &audiohook_list->out_translate[1]);
|
|
struct ast_frame *start_frame = frame, *middle_frame = frame, *end_frame = frame;
|
|
struct ast_audiohook *audiohook = NULL;
|
|
int samples = frame->samples;
|
|
|
|
/* If the frame coming in is not signed linear we have to send it through the in_translate path */
|
|
if (frame->subclass != AST_FORMAT_SLINEAR) {
|
|
if (in_translate->format != frame->subclass) {
|
|
if (in_translate->trans_pvt)
|
|
ast_translator_free_path(in_translate->trans_pvt);
|
|
if (!(in_translate->trans_pvt = ast_translator_build_path(AST_FORMAT_SLINEAR, frame->subclass)))
|
|
return frame;
|
|
in_translate->format = frame->subclass;
|
|
}
|
|
if (!(middle_frame = ast_translate(in_translate->trans_pvt, frame, 0)))
|
|
return frame;
|
|
}
|
|
|
|
/* 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(&audiohook_list->spy_list, list);
|
|
audiohook->status = AST_AUDIOHOOK_STATUS_DONE;
|
|
ast_cond_signal(&audiohook->trigger);
|
|
ast_audiohook_unlock(audiohook);
|
|
continue;
|
|
}
|
|
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(&audiohook_list->whisper_list, list);
|
|
audiohook->status = AST_AUDIOHOOK_STATUS_DONE;
|
|
ast_cond_signal(&audiohook->trigger);
|
|
ast_audiohook_unlock(audiohook);
|
|
continue;
|
|
}
|
|
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, data2 = combine_buf; i < samples; i++, data1++, data2++)
|
|
ast_slinear_saturated_add(data1, data2);
|
|
end_frame = middle_frame;
|
|
}
|
|
|
|
/* 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(&audiohook_list->manipulate_list, list);
|
|
audiohook->status = 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;
|
|
}
|
|
/* Feed in frame to manipulation */
|
|
audiohook->manipulate_callback(audiohook, chan, middle_frame, direction);
|
|
ast_audiohook_unlock(audiohook);
|
|
}
|
|
AST_LIST_TRAVERSE_SAFE_END
|
|
end_frame = middle_frame;
|
|
}
|
|
|
|
/* Now we figure out what to do with our end frame (whether to transcode or not) */
|
|
if (middle_frame == end_frame) {
|
|
/* Middle frame was modified and became the end frame... let's see if we need to transcode */
|
|
if (end_frame->subclass != start_frame->subclass) {
|
|
if (out_translate->format != start_frame->subclass) {
|
|
if (out_translate->trans_pvt)
|
|
ast_translator_free_path(out_translate->trans_pvt);
|
|
if (!(out_translate->trans_pvt = ast_translator_build_path(start_frame->subclass, AST_FORMAT_SLINEAR))) {
|
|
/* We can't transcode this... drop our middle frame and return the original */
|
|
ast_frfree(middle_frame);
|
|
return start_frame;
|
|
}
|
|
out_translate->format = start_frame->subclass;
|
|
}
|
|
/* Transcode from our middle (signed linear) frame to new format of the frame that came in */
|
|
if (!(end_frame = ast_translate(out_translate->trans_pvt, middle_frame, 0))) {
|
|
/* Failed to transcode the frame... drop it and return the original */
|
|
ast_frfree(middle_frame);
|
|
return start_frame;
|
|
}
|
|
/* Here's the scoop... middle frame is no longer of use to us */
|
|
ast_frfree(middle_frame);
|
|
}
|
|
/* Yay let's rid ourselves of the start frame */
|
|
ast_frfree(start_frame);
|
|
} else {
|
|
/* No frame was modified, we can just drop our middle frame and pass the frame we got in out */
|
|
ast_frfree(middle_frame);
|
|
}
|
|
|
|
return end_frame;
|
|
}
|
|
|
|
/*! \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 tv;
|
|
struct timespec ts;
|
|
|
|
tv = ast_tvadd(ast_tvnow(), ast_samp2tv(50000, 1000));
|
|
ts.tv_sec = tv.tv_sec;
|
|
ts.tv_nsec = tv.tv_usec * 1000;
|
|
|
|
ast_cond_timedwait(&audiohook->trigger, &audiohook->lock, &ts);
|
|
|
|
return;
|
|
}
|