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.
static int	audiosocket_hangup (struct ast_channel *ast)
	Function called when we should hang the channel up.
static struct ast_frame *	audiosocket_read (struct ast_channel *ast)
	Function called when we should read a frame from the channel.
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.
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.
static int	audiosocket_write (struct ast_channel *ast, struct ast_frame *f)
	Function called when we should write a frame to the channel.
static int	load_module (void)
	Function called when our module is loaded.
static int	unload_module (void)
	Function called when our module is unloaded.

Variables
static struct ast_module_info	__mod_info = { .name = AST_MODULE, .flags = AST_MODFLAG_LOAD_ORDER , .description = "AudioSocket Channel" , .key = ASTERISK_GPL_KEY , .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 = ASTERISK_GPL_KEY , .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.

                                                          {
    .type = "AudioSocket",
    .description = "AudioSocket Channel Driver",
    .requester = audiosocket_request,
    .call = audiosocket_call,
    .hangup = audiosocket_hangup,
    .read = audiosocket_read,
    .write = audiosocket_write,
    .send_digit_end = audiosocket_send_dtmf,
};

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().