chan_audiosocket.c File Reference

AudioSocket Channel. More...

#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"

Include dependency graph for chan_audiosocket.c:

Go to the source code of this file.

Data Structures
struct	audiosocket_instance

Macros
#define	FD_OUTPUT   1 /* A fd of -1 means an error, 0 is stdin */

Enumerations
enum	{ OPT_AUDIOSOCKET_CODEC = (1 << 0) }
enum	{ OPT_ARG_AUDIOSOCKET_CODEC = (1 << 0) , OPT_ARG_ARRAY_SIZE }

Functions
static void	__reg_module (void)
static void	__unreg_module (void)
struct ast_module *	AST_MODULE_SELF_SYM (void)
static int	audiosocket_call (struct ast_channel *ast, const char *dest, int timeout)
	Function called when we should actually call the destination. More...
static int	audiosocket_hangup (struct ast_channel *ast)
	Function called when we should hang the channel up. More...
static struct ast_frame *	audiosocket_read (struct ast_channel *ast)
	Function called when we should read a frame from the channel. More...
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)
	Function called when we should prepare to call the unicast destination. More...
static int	audiosocket_send_dtmf (struct ast_channel *ast, char digit, unsigned int duration)
	Function called when we should write a DTMF frame to the channel. More...
static int	audiosocket_write (struct ast_channel *ast, struct ast_frame *f)
	Function called when we should write a frame to the channel. More...
static int	load_module (void)
	Function called when our module is loaded. More...
static int	unload_module (void)
	Function called when our module is unloaded. More...

Variables
static struct ast_module_info	__mod_info = { .name = AST_MODULE, .flags = AST_MODFLAG_LOAD_ORDER , .description = "AudioSocket Channel" , .key = "This paragraph is copyright (c) 2006 by Digium, Inc. \In order for your module to load, it must return this \key via a function called \"key\". Any code which \includes this paragraph must be licensed under the GNU \General Public License version 2 or later (at your \option). In addition to Digium's general reservations \of rights, Digium expressly reserves the right to \allow other parties to license this paragraph under \different terms. Any use of Digium, Inc. trademarks or \logos (including \"Asterisk\" or \"Digium\") without \express written permission of Digium, Inc. is prohibited.\n" , .buildopt_sum = AST_BUILDOPT_SUM, .support_level = AST_MODULE_SUPPORT_EXTENDED, .load = load_module, .unload = unload_module, .load_pri = AST_MODPRI_CHANNEL_DRIVER, .requires = "res_audiosocket", }
static const struct ast_module_info *	ast_module_info = &__mod_info
static struct ast_channel_tech	audiosocket_channel_tech
struct audiosocket_instance	audiosocket_instance
static const struct ast_app_option	audiosocket_options [128] = { [ 'c' ] = { .flag = OPT_AUDIOSOCKET_CODEC , .arg_index = OPT_ARG_AUDIOSOCKET_CODEC + 1 }, }

Detailed Description

AudioSocket Channel.

Author

Seán C McCord scm@c.nosp@m.ycor.nosp@m.esys..nosp@m.com

Definition in file chan_audiosocket.c.

Macro Definition Documentation

FD_OUTPUT

#define FD_OUTPUT   1 /* A fd of -1 means an error, 0 is stdin */

Definition at line 46 of file chan_audiosocket.c.

Enumeration Type Documentation

anonymous enum

anonymous enum

Enumerator
OPT_AUDIOSOCKET_CODEC

Definition at line 168 of file chan_audiosocket.c.

     {
    OPT_AUDIOSOCKET_CODEC = (1 << 0),
};

anonymous enum

anonymous enum

Enumerator
OPT_ARG_AUDIOSOCKET_CODEC
OPT_ARG_ARRAY_SIZE

Definition at line 172 of file chan_audiosocket.c.

     {
    OPT_ARG_AUDIOSOCKET_CODEC = (1 << 0),
    OPT_ARG_ARRAY_SIZE
};

Function Documentation

__reg_module()

static void __reg_module ( void  )

static

Definition at line 346 of file chan_audiosocket.c.

__unreg_module()

static void __unreg_module ( void  )

static

Definition at line 346 of file chan_audiosocket.c.

AST_MODULE_SELF_SYM()

struct ast_module * AST_MODULE_SELF_SYM

(

void

)

Definition at line 346 of file chan_audiosocket.c.

audiosocket_call()

static int audiosocket_call ( struct ast_channel *  ast, const char *  dest, int  timeout  )

static

Function called when we should actually call the destination.

Definition at line 142 of file chan_audiosocket.c.

{
    struct audiosocket_instance *instance = ast_channel_tech_pvt(ast);
 
    ast_queue_control(ast, AST_CONTROL_ANSWER);
 
    return ast_audiosocket_init(instance->svc, instance->id);
}

References ast_audiosocket_init(), ast_channel_tech_pvt(), AST_CONTROL_ANSWER, ast_queue_control(), audiosocket_instance::id, and audiosocket_instance::svc.

audiosocket_hangup()

static int audiosocket_hangup ( struct ast_channel *  ast )

static

Function called when we should hang the channel up.

Definition at line 152 of file chan_audiosocket.c.

{
    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;
}

References ast_channel_tech_pvt(), ast_channel_tech_pvt_set(), ast_free, NULL, and audiosocket_instance::svc.

audiosocket_read()

static struct ast_frame * audiosocket_read ( struct ast_channel *  ast )

static

Function called when we should read a frame from the channel.

Definition at line 77 of file chan_audiosocket.c.

{
    struct audiosocket_instance *instance;
    struct ast_frame *frame;
    int hangup;
 
 
    /* The channel should always be present from the API */
    instance = ast_channel_tech_pvt(ast);
    if (instance == NULL || instance->svc < FD_OUTPUT) {
        return NULL;
    }
 
    frame = ast_audiosocket_receive_frame_with_hangup(instance->svc, &hangup);
    if (!frame && !hangup) {
        ast_log(LOG_ERROR, "Failed to receive frame from AudioSocket server '%s'"
            " connected to channel '%s'\n", instance->server, ast_channel_name(ast));
    }
    return frame;
}

References ast_audiosocket_receive_frame_with_hangup(), ast_channel_name(), ast_channel_tech_pvt(), ast_log, FD_OUTPUT, hangup(), LOG_ERROR, NULL, audiosocket_instance::server, and audiosocket_instance::svc.

audiosocket_request()

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

Function called when we should prepare to call the unicast destination.

Definition at line 182 of file chan_audiosocket.c.

{
    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;
    }
    /* Make a copy of the server address for display in error messages. */
    ast_copy_string(instance->server, args.destination, sizeof(instance->server));
    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;
}

References ao2_cleanup, ao2_ref, args, AST_AF_UNSPEC, AST_APP_ARG, ast_app_parse_options(), ast_audiosocket_connect(), ast_calloc, AST_CAUSE_FAILURE, ast_channel_alloc, ast_channel_nativeformats_set(), ast_channel_set_fd(), ast_channel_set_rawreadformat(), ast_channel_set_rawwriteformat(), ast_channel_set_readformat(), ast_channel_set_writeformat(), ast_channel_tech_pvt_set(), ast_channel_tech_set(), ast_channel_unlock, ast_copy_string(), AST_DECLARE_APP_ARGS, ast_format_cache_get, ast_format_cap_alloc, ast_format_cap_append, AST_FORMAT_CAP_FLAG_DEFAULT, ast_format_cap_get_format(), ast_free, ast_log, AST_NONSTANDARD_APP_ARGS, ast_sockaddr_resolve_first_af(), AST_STATE_DOWN, ast_strdupa, ast_strlen_zero(), ast_test_flag, audiosocket_channel_tech, audiosocket_options, audiosocket_instance::id, LOG_ERROR, NULL, OPT_ARG_ARRAY_SIZE, OPT_ARG_AUDIOSOCKET_CODEC, OPT_AUDIOSOCKET_CODEC, options, PARSE_PORT_REQUIRE, pbx_builtin_setvar_helper(), audiosocket_instance::server, and audiosocket_instance::svc.

audiosocket_send_dtmf()

static int audiosocket_send_dtmf ( struct ast_channel *  ast, char  digit, unsigned int  duration  )

static

Function called when we should write a DTMF frame to the channel.

Definition at line 118 of file chan_audiosocket.c.

{
    struct audiosocket_instance *instance;
    struct ast_frame f;
 
    /* The channel should always be present from the API */
    instance = ast_channel_tech_pvt(ast);
    if (instance == NULL || instance->svc < 1) {
        return -1;
    }
 
    f.frametype = AST_FRAME_DTMF;
    f.subclass.integer = digit;
    f.len = duration;
 
    if (ast_audiosocket_send_frame(instance->svc, &f)) {
        ast_log(LOG_ERROR, "Failed to forward DTMF frame from channel '%s' to AudioSocket server '%s'\n",
                ast_channel_name(ast), instance->server);
        return -1;
    }
    return 0;
}

References ast_audiosocket_send_frame(), ast_channel_name(), ast_channel_tech_pvt(), AST_FRAME_DTMF, ast_log, digit, ast_frame::frametype, ast_frame_subclass::integer, ast_frame::len, LOG_ERROR, NULL, audiosocket_instance::server, ast_frame::subclass, and audiosocket_instance::svc.

audiosocket_write()

static int audiosocket_write ( struct ast_channel *  ast, struct ast_frame *  f  )

static

Function called when we should write a frame to the channel.

Definition at line 99 of file chan_audiosocket.c.

{
    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;
    }
 
    if (ast_audiosocket_send_frame(instance->svc, f)) {
        ast_log(LOG_ERROR, "Failed to forward frame from channel '%s' to AudioSocket server '%s'\n",
                ast_channel_name(ast), instance->server);
        return -1;
    }
    return 0;
}

References ast_audiosocket_send_frame(), ast_channel_name(), ast_channel_tech_pvt(), ast_log, LOG_ERROR, NULL, audiosocket_instance::server, and audiosocket_instance::svc.

load_module()

static int load_module ( void  )

static

Function called when our module is loaded.

Definition at line 324 of file chan_audiosocket.c.

{
    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;
}

References ao2_ref, ast_channel_register(), ast_format_cap_alloc, ast_format_cap_append_by_type(), AST_FORMAT_CAP_FLAG_DEFAULT, ast_log, AST_MEDIA_TYPE_UNKNOWN, AST_MODULE_LOAD_DECLINE, AST_MODULE_LOAD_SUCCESS, audiosocket_channel_tech, ast_channel_tech::capabilities, LOG_ERROR, and NULL.

unload_module()

static int unload_module ( void  )

static

Function called when our module is unloaded.

Definition at line 314 of file chan_audiosocket.c.

{
    ast_channel_unregister(&audiosocket_channel_tech);
    ao2_cleanup(audiosocket_channel_tech.capabilities);
    audiosocket_channel_tech.capabilities = NULL;
 
    return 0;
}

References ao2_cleanup, ast_channel_unregister(), audiosocket_channel_tech, ast_channel_tech::capabilities, and NULL.

Variable Documentation

__mod_info

struct ast_module_info __mod_info = { .name = AST_MODULE, .flags = AST_MODFLAG_LOAD_ORDER , .description = "AudioSocket Channel" , .key = "This paragraph is copyright (c) 2006 by Digium, Inc. \In order for your module to load, it must return this \key via a function called \"key\". Any code which \includes this paragraph must be licensed under the GNU \General Public License version 2 or later (at your \option). In addition to Digium's general reservations \of rights, Digium expressly reserves the right to \allow other parties to license this paragraph under \different terms. Any use of Digium, Inc. trademarks or \logos (including \"Asterisk\" or \"Digium\") without \express written permission of Digium, Inc. is prohibited.\n" , .buildopt_sum = AST_BUILDOPT_SUM, .support_level = AST_MODULE_SUPPORT_EXTENDED, .load = load_module, .unload = unload_module, .load_pri = AST_MODPRI_CHANNEL_DRIVER, .requires = "res_audiosocket", }

static

Definition at line 346 of file chan_audiosocket.c.

ast_module_info

const struct ast_module_info* ast_module_info = &__mod_info

static

Definition at line 346 of file chan_audiosocket.c.

audiosocket_channel_tech

struct ast_channel_tech audiosocket_channel_tech

static

Definition at line 65 of file chan_audiosocket.c.

Referenced by audiosocket_request(), load_module(), and unload_module().

audiosocket_instance

struct audiosocket_instance audiosocket_instance

audiosocket_options

const struct ast_app_option audiosocket_options[128] = { [ 'c' ] = { .flag = OPT_AUDIOSOCKET_CODEC , .arg_index = OPT_ARG_AUDIOSOCKET_CODEC + 1 }, }

static

Definition at line 179 of file chan_audiosocket.c.

Referenced by audiosocket_request().