/* * Asterisk -- An open source telephony toolkit. * * Copyright (C) 2011, Digium, Inc. * * Joshua Colp * * See http://www.asterisk.org for more information about * the Asterisk project. Please do not directly contact * any of the maintainers of this project for assistance; * the project provides a web site, mailing lists and IRC * channels for your use. * * This program is free software, distributed under the terms of * the GNU General Public License Version 2. See the LICENSE file * at the top of the source tree. */ /*! \file * * \brief Technology independent volume control * * \author Joshua Colp * * \ingroup functions * */ /*** MODULEINFO core ***/ #include "asterisk.h" #include "asterisk/module.h" #include "asterisk/channel.h" #include "asterisk/pbx.h" #include "asterisk/utils.h" #include "asterisk/audiohook.h" #include "asterisk/app.h" /*** DOCUMENTATION Set the TX or RX volume of a channel. Must be TX or RX. The VOLUME function can be used to increase or decrease the tx or rx gain of any channel. For example: Set(VOLUME(TX)=3) Set(VOLUME(RX)=2) Set(VOLUME(TX,p)=3) Set(VOLUME(RX,p)=3) ***/ struct volume_information { struct ast_audiohook audiohook; int tx_gain; int rx_gain; unsigned int flags; }; enum volume_flags { VOLUMEFLAG_CHANGE = (1 << 1), }; AST_APP_OPTIONS(volume_opts, { AST_APP_OPTION('p', VOLUMEFLAG_CHANGE), }); static void destroy_callback(void *data) { struct volume_information *vi = data; /* Destroy the audiohook, and destroy ourselves */ ast_audiohook_lock(&vi->audiohook); ast_audiohook_detach(&vi->audiohook); ast_audiohook_unlock(&vi->audiohook); ast_audiohook_destroy(&vi->audiohook); ast_free(vi); return; } /*! \brief Static structure for datastore information */ static const struct ast_datastore_info volume_datastore = { .type = "volume", .destroy = destroy_callback }; static int 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 volume_information *vi = NULL; int *gain = NULL; /* If the audiohook is stopping it means the channel is shutting down.... but we let the datastore destroy take care of it */ if (audiohook->status == AST_AUDIOHOOK_STATUS_DONE) return 0; /* Grab datastore which contains our gain information */ if (!(datastore = ast_channel_datastore_find(chan, &volume_datastore, NULL))) return 0; vi = datastore->data; /* If this is DTMF then allow them to increase/decrease the gains */ if (ast_test_flag(vi, VOLUMEFLAG_CHANGE)) { if (frame->frametype == AST_FRAME_DTMF) { /* Only use DTMF coming from the source... not going to it */ if (direction != AST_AUDIOHOOK_DIRECTION_READ) return 0; if (frame->subclass.integer == '*') { vi->tx_gain += 1; vi->rx_gain += 1; } else if (frame->subclass.integer == '#') { vi->tx_gain -= 1; vi->rx_gain -= 1; } } } if (frame->frametype == AST_FRAME_VOICE) { /* Based on direction of frame grab the gain, and confirm it is applicable */ if (!(gain = (direction == AST_AUDIOHOOK_DIRECTION_READ) ? &vi->rx_gain : &vi->tx_gain) || !*gain) return 0; /* Apply gain to frame... easy as pi */ ast_frame_adjust_volume(frame, *gain); } return 0; } static int volume_write(struct ast_channel *chan, const char *cmd, char *data, const char *value) { struct ast_datastore *datastore = NULL; struct volume_information *vi = NULL; int is_new = 0; /* Separate options from argument */ AST_DECLARE_APP_ARGS(args, AST_APP_ARG(direction); AST_APP_ARG(options); ); if (!chan) { ast_log(LOG_WARNING, "No channel was provided to %s function.\n", cmd); return -1; } AST_STANDARD_APP_ARGS(args, data); ast_channel_lock(chan); if (!(datastore = ast_channel_datastore_find(chan, &volume_datastore, NULL))) { ast_channel_unlock(chan); /* Allocate a new datastore to hold the reference to this volume and audiohook information */ if (!(datastore = ast_datastore_alloc(&volume_datastore, NULL))) return 0; if (!(vi = ast_calloc(1, sizeof(*vi)))) { ast_datastore_free(datastore); return 0; } ast_audiohook_init(&vi->audiohook, AST_AUDIOHOOK_TYPE_MANIPULATE, "Volume", AST_AUDIOHOOK_MANIPULATE_ALL_RATES); vi->audiohook.manipulate_callback = volume_callback; ast_set_flag(&vi->audiohook, AST_AUDIOHOOK_WANTS_DTMF); is_new = 1; } else { ast_channel_unlock(chan); vi = datastore->data; } /* Adjust gain on volume information structure */ if (ast_strlen_zero(args.direction)) { ast_log(LOG_ERROR, "Direction must be specified for VOLUME function\n"); return -1; } if (!strcasecmp(args.direction, "tx")) { vi->tx_gain = atoi(value); } else if (!strcasecmp(args.direction, "rx")) { vi->rx_gain = atoi(value); } else { ast_log(LOG_ERROR, "Direction must be either RX or TX\n"); } if (is_new) { datastore->data = vi; ast_channel_lock(chan); ast_channel_datastore_add(chan, datastore); ast_channel_unlock(chan); ast_audiohook_attach(chan, &vi->audiohook); } /* Add Option data to struct */ if (!ast_strlen_zero(args.options)) { struct ast_flags flags = { 0 }; ast_app_parse_options(volume_opts, &flags, NULL, args.options); vi->flags = flags.flags; } else { vi->flags = 0; } return 0; } static struct ast_custom_function volume_function = { .name = "VOLUME", .write = volume_write, }; static int unload_module(void) { return ast_custom_function_unregister(&volume_function); } static int load_module(void) { return ast_custom_function_register(&volume_function); } AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Technology independent volume control");