chan_audiosocket.c

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/*
 * Asterisk -- An open source telephony toolkit.
 *
 * Copyright (C) 2019, CyCore Systems, Inc
 *
 * Seán C McCord <scm@cycoresys.com>
 *
 * See http://www.asterisk.org for more information about
 * the Asterisk project. Please do not directly contact
 * any of the maintainers of this project for assistance;
 * the project provides a web site, mailing lists and IRC
 * channels for your use.
 *
 * This program is free software, distributed under the terms of
 * the GNU General Public License Version 2. See the LICENSE file
 * at the top of the source tree.
 */
 
/*! \file
 *
 * \author Seán C McCord <scm@cycoresys.com>
 *
 * \brief AudioSocket Channel
 *
 * \ingroup channel_drivers
 */
 
/*** MODULEINFO
    <depend>res_audiosocket</depend>
    <support_level>extended</support_level>
 ***/
 
#include "asterisk.h"
#include <uuid/uuid.h>
 
#include "asterisk/channel.h"
#include "asterisk/module.h"
#include "asterisk/res_audiosocket.h"
#include "asterisk/pbx.h"
#include "asterisk/acl.h"
#include "asterisk/app.h"
#include "asterisk/causes.h"
#include "asterisk/format_cache.h"
 
#define FD_OUTPUT 1 /* A fd of -1 means an error, 0 is stdin */
 
struct audiosocket_instance {
    int svc;    /* The file descriptor for the AudioSocket instance */
    char id[38];    /* The UUID identifying this AudioSocket instance */
} audiosocket_instance;
 
/* Forward declarations */
static struct ast_channel *audiosocket_request(const char *type,
    struct ast_format_cap *cap, const struct ast_assigned_ids *assignedids,
    const struct ast_channel *requestor, const char *data, int *cause);
static int audiosocket_call(struct ast_channel *ast, const char *dest, int timeout);
static int audiosocket_hangup(struct ast_channel *ast);
static struct ast_frame *audiosocket_read(struct ast_channel *ast);
static int audiosocket_write(struct ast_channel *ast, struct ast_frame *f);
 
/* AudioSocket channel driver declaration */
static struct ast_channel_tech audiosocket_channel_tech = {
    .type = "AudioSocket",
    .description = "AudioSocket Channel Driver",
    .requester = audiosocket_request,
    .call = audiosocket_call,
    .hangup = audiosocket_hangup,
    .read = audiosocket_read,
    .write = audiosocket_write,
};
 
/*! \brief Function called when we should read a frame from the channel */
static struct ast_frame *audiosocket_read(struct ast_channel *ast)
{
    struct audiosocket_instance *instance;
 
    /* The channel should always be present from the API */
    instance = ast_channel_tech_pvt(ast);
    if (instance == NULL || instance->svc < FD_OUTPUT) {
        return NULL;
    }
    return ast_audiosocket_receive_frame(instance->svc);
}
 
/*! \brief Function called when we should write a frame to the channel */
static int audiosocket_write(struct ast_channel *ast, struct ast_frame *f)
{
    struct audiosocket_instance *instance;
 
    /* The channel should always be present from the API */
    instance = ast_channel_tech_pvt(ast);
    if (instance == NULL || instance->svc < 1) {
        return -1;
    }
    return ast_audiosocket_send_frame(instance->svc, f);
}
 
/*! \brief Function called when we should actually call the destination */
static int audiosocket_call(struct ast_channel *ast, const char *dest, int timeout)
{
    struct audiosocket_instance *instance = ast_channel_tech_pvt(ast);
 
    ast_queue_control(ast, AST_CONTROL_ANSWER);
 
    return ast_audiosocket_init(instance->svc, instance->id);
}
 
/*! \brief Function called when we should hang the channel up */
static int audiosocket_hangup(struct ast_channel *ast)
{
    struct audiosocket_instance *instance;
 
    /* The channel should always be present from the API */
    instance = ast_channel_tech_pvt(ast);
    if (instance != NULL && instance->svc > 0) {
        close(instance->svc);
    }
 
    ast_channel_tech_pvt_set(ast, NULL);
    ast_free(instance);
 
    return 0;
}
 
enum {
    OPT_AUDIOSOCKET_CODEC = (1 << 0),
};
 
enum {
    OPT_ARG_AUDIOSOCKET_CODEC = (1 << 0),
    OPT_ARG_ARRAY_SIZE
};
 
AST_APP_OPTIONS(audiosocket_options, BEGIN_OPTIONS
    AST_APP_OPTION_ARG('c', OPT_AUDIOSOCKET_CODEC, OPT_ARG_AUDIOSOCKET_CODEC),
END_OPTIONS );
 
/*! \brief Function called when we should prepare to call the unicast destination */
static struct ast_channel *audiosocket_request(const char *type,
    struct ast_format_cap *cap, const struct ast_assigned_ids *assignedids,
    const struct ast_channel *requestor, const char *data, int *cause)
{
    char *parse;
    struct audiosocket_instance *instance = NULL;
    struct ast_sockaddr address;
    struct ast_channel *chan;
    struct ast_format_cap *caps = NULL;
    struct ast_format *fmt = NULL;
    uuid_t uu;
    int fd = -1;
    AST_DECLARE_APP_ARGS(args,
        AST_APP_ARG(destination);
        AST_APP_ARG(idStr);
        AST_APP_ARG(options);
    );
    struct ast_flags opts = { 0, };
    char *opt_args[OPT_ARG_ARRAY_SIZE];
 
    if (ast_strlen_zero(data)) {
        ast_log(LOG_ERROR, "Destination is required for the 'AudioSocket' channel\n");
        goto failure;
    }
    parse = ast_strdupa(data);
    AST_NONSTANDARD_APP_ARGS(args, parse, '/');
 
    if (ast_strlen_zero(args.destination)) {
        ast_log(LOG_ERROR, "Destination is required for the 'AudioSocket' channel\n");
        goto failure;
    }
    if (ast_sockaddr_resolve_first_af
        (&address, args.destination, PARSE_PORT_REQUIRE, AST_AF_UNSPEC)) {
        ast_log(LOG_ERROR, "Destination '%s' could not be parsed\n", args.destination);
        goto failure;
    }
 
    if (ast_strlen_zero(args.idStr)) {
        ast_log(LOG_ERROR, "UUID is required for the 'AudioSocket' channel\n");
        goto failure;
    }
    if (uuid_parse(args.idStr, uu)) {
        ast_log(LOG_ERROR, "Failed to parse UUID '%s'\n", args.idStr);
        goto failure;
    }
 
    if (!ast_strlen_zero(args.options)
        && ast_app_parse_options(audiosocket_options, &opts, opt_args,
            ast_strdupa(args.options))) {
        ast_log(LOG_ERROR, "'AudioSocket' channel options '%s' parse error\n",
            args.options);
        goto failure;
    }
 
    if (ast_test_flag(&opts, OPT_AUDIOSOCKET_CODEC)
        && !ast_strlen_zero(opt_args[OPT_ARG_AUDIOSOCKET_CODEC])) {
        fmt = ast_format_cache_get(opt_args[OPT_ARG_AUDIOSOCKET_CODEC]);
        if (!fmt) {
            ast_log(LOG_ERROR, "Codec '%s' not found for AudioSocket connection to '%s'\n",
                opt_args[OPT_ARG_AUDIOSOCKET_CODEC], args.destination);
            goto failure;
        }
    } else {
        fmt = ast_format_cap_get_format(cap, 0);
        if (!fmt) {
            ast_log(LOG_ERROR, "No codec available for AudioSocket connection to '%s'\n",
                args.destination);
            goto failure;
        }
    }
 
    caps = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT);
    if (!caps) {
        goto failure;
    }
 
    instance = ast_calloc(1, sizeof(*instance));
    if (!instance) {
        ast_log(LOG_ERROR, "Failed to allocate AudioSocket channel pvt\n");
        goto failure;
    }
    ast_copy_string(instance->id, args.idStr, sizeof(instance->id));
 
    if ((fd = ast_audiosocket_connect(args.destination, NULL)) < 0) {
        goto failure;
    }
    instance->svc = fd;
 
    chan = ast_channel_alloc(1, AST_STATE_DOWN, "", "", "", "", "", assignedids,
        requestor, 0, "AudioSocket/%s-%s", args.destination, args.idStr);
    if (!chan) {
        goto failure;
    }
    ast_channel_set_fd(chan, 0, fd);
 
    ast_channel_tech_set(chan, &audiosocket_channel_tech);
 
    ast_format_cap_append(caps, fmt, 0);
    ast_channel_nativeformats_set(chan, caps);
    ast_channel_set_writeformat(chan, fmt);
    ast_channel_set_rawwriteformat(chan, fmt);
    ast_channel_set_readformat(chan, fmt);
    ast_channel_set_rawreadformat(chan, fmt);
 
    ast_channel_tech_pvt_set(chan, instance);
 
    pbx_builtin_setvar_helper(chan, "AUDIOSOCKET_UUID", args.idStr);
    pbx_builtin_setvar_helper(chan, "AUDIOSOCKET_SERVICE", args.destination);
 
    ast_channel_unlock(chan);
 
    ao2_ref(fmt, -1);
    ao2_ref(caps, -1);
    return chan;
 
failure:
    *cause = AST_CAUSE_FAILURE;
    ao2_cleanup(fmt);
    ao2_cleanup(caps);
    if (instance != NULL) {
        ast_free(instance);
        if (fd >= 0) {
            close(fd);
        }
    }
 
    return NULL;
}
 
/*! \brief Function called when our module is unloaded */
static int unload_module(void)
{
    ast_channel_unregister(&audiosocket_channel_tech);
    ao2_cleanup(audiosocket_channel_tech.capabilities);
    audiosocket_channel_tech.capabilities = NULL;
 
    return 0;
}
 
/*! \brief Function called when our module is loaded */
static int load_module(void)
{
    if (!(audiosocket_channel_tech.capabilities = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT))) {
        return AST_MODULE_LOAD_DECLINE;
    }
    ast_format_cap_append_by_type(audiosocket_channel_tech.capabilities, AST_MEDIA_TYPE_UNKNOWN);
    if (ast_channel_register(&audiosocket_channel_tech)) {
        ast_log(LOG_ERROR, "Unable to register channel class AudioSocket");
        ao2_ref(audiosocket_channel_tech.capabilities, -1);
        audiosocket_channel_tech.capabilities = NULL;
        return AST_MODULE_LOAD_DECLINE;
    }
 
    return AST_MODULE_LOAD_SUCCESS;
}
 
AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "AudioSocket Channel",
    .support_level = AST_MODULE_SUPPORT_EXTENDED,
    .load = load_module,
    .unload = unload_module,
    .load_pri = AST_MODPRI_CHANNEL_DRIVER,
    .requires = "res_audiosocket",
);