chan_audiosocket.c

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/*
 * Asterisk -- An open source telephony toolkit.
 *
 * Copyright (C) 2019, CyCore Systems, Inc
 *
 * Seán C McCord <[email protected]>
 *
 * 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 <[email protected]>
 *
 * \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",
);