app_audiosocket.c File Reference

AudioSocket application – transmit and receive audio through a TCP socket. More...

#include "asterisk.h"
#include "errno.h"
#include <uuid/uuid.h>
#include "asterisk/file.h"
#include "asterisk/module.h"
#include "asterisk/channel.h"
#include "asterisk/app.h"
#include "asterisk/pbx.h"
#include "asterisk/res_audiosocket.h"
#include "asterisk/utils.h"
#include "asterisk/format_cache.h"

Include dependency graph for app_audiosocket.c:

Go to the source code of this file.

Macros
#define	AST_MODULE   "app_audiosocket"
#define	AUDIOSOCKET_CONFIG   "audiosocket.conf"
#define	MAX_WAIT_TIMEOUT_MSEC   2000

Functions
static void	__reg_module (void)
static void	__unreg_module (void)
struct ast_module *	AST_MODULE_SELF_SYM (void)
static int	audiosocket_exec (struct ast_channel *chan, const char *data)
static int	audiosocket_run (struct ast_channel *chan, const char *id, const int svc, const char *server)
static int	load_module (void)
static int	unload_module (void)

Variables
static struct ast_module_info	__mod_info = { .name = "app_audiosocket" , .flags = AST_MODFLAG_LOAD_ORDER , .description = "AudioSocket Application" , .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 char	app [] = "AudioSocket"
static const struct ast_module_info *	ast_module_info = &__mod_info

Detailed Description

AudioSocket application – transmit and receive audio through a TCP socket.

Author

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

Definition in file app_audiosocket.c.

Macro Definition Documentation

AST_MODULE

#define AST_MODULE   "app_audiosocket"

Definition at line 46 of file app_audiosocket.c.

AUDIOSOCKET_CONFIG

#define AUDIOSOCKET_CONFIG   "audiosocket.conf"

Definition at line 47 of file app_audiosocket.c.

MAX_WAIT_TIMEOUT_MSEC

#define MAX_WAIT_TIMEOUT_MSEC   2000

Definition at line 48 of file app_audiosocket.c.

Function Documentation

__reg_module()

static void __reg_module ( void  )

static

Definition at line 247 of file app_audiosocket.c.

__unreg_module()

static void __unreg_module ( void  )

static

Definition at line 247 of file app_audiosocket.c.

AST_MODULE_SELF_SYM()

struct ast_module * AST_MODULE_SELF_SYM

(

void

)

Definition at line 247 of file app_audiosocket.c.

audiosocket_exec()

static int audiosocket_exec ( struct ast_channel *  chan, const char *  data  )

static

Definition at line 78 of file app_audiosocket.c.

{
    char *parse;
    struct ast_format *readFormat, *writeFormat;
    const char *chanName;
    int res;
 
    AST_DECLARE_APP_ARGS(args,
        AST_APP_ARG(idStr);
        AST_APP_ARG(server);
    );
 
    int s = 0;
    uuid_t uu;
 
 
    chanName = ast_channel_name(chan);
 
    /* Parse and validate arguments */
    parse = ast_strdupa(data);
    AST_STANDARD_APP_ARGS(args, parse);
    if (ast_strlen_zero(args.idStr)) {
        ast_log(LOG_ERROR, "UUID is required\n");
        return -1;
    }
    if (uuid_parse(args.idStr, uu)) {
        ast_log(LOG_ERROR, "Failed to parse UUID '%s'\n", args.idStr);
        return -1;
    }
    if ((s = ast_audiosocket_connect(args.server, chan)) < 0) {
        /* The res module will already output a log message, so another is not needed */
        return -1;
    }
 
    /* Save current channel audio format and force to linear PCM. */
 
    writeFormat = ao2_bump(ast_channel_writeformat(chan));
    readFormat = ao2_bump(ast_channel_readformat(chan));
 
    if (ast_set_write_format(chan, ast_format_slin)) {
        ast_log(LOG_ERROR, "Failed to set write format to SLINEAR for channel '%s'\n", chanName);
        res = -1;
        goto end;
    }
 
    if (ast_set_read_format(chan, ast_format_slin)) {
        ast_log(LOG_ERROR, "Failed to set read format to SLINEAR for channel '%s'\n", chanName);
        res = -1;
        goto end;
    }
 
    /* Only a requested hangup or socket closure from the remote end will
       return a 0 value (normal exit). All other events that disrupt an
       active connection are treated as errors for now (non-zero). */
 
    res = audiosocket_run(chan, args.idStr, s, args.server);
 
end:
 
    /* Restore previous formats and close the connection */
    if (ast_set_write_format(chan, writeFormat)) {
        ast_log(LOG_ERROR, "Failed to restore write format for channel '%s'\n", chanName);
    }
    if (ast_set_read_format(chan, readFormat)) {
        ast_log(LOG_ERROR, "Failed to restore read format for channel '%s'\n", chanName);
    }
    ao2_ref(writeFormat, -1);
    ao2_ref(readFormat, -1);
    close(s);
 
    return res;
}

References ao2_bump, ao2_ref, args, AST_APP_ARG, ast_audiosocket_connect(), ast_channel_name(), ast_channel_readformat(), ast_channel_writeformat(), AST_DECLARE_APP_ARGS, ast_format_slin, ast_log, ast_set_read_format(), ast_set_write_format(), AST_STANDARD_APP_ARGS, ast_strdupa, ast_strlen_zero(), audiosocket_run(), end, and LOG_ERROR.

Referenced by load_module().

audiosocket_run()

static int audiosocket_run ( struct ast_channel *  chan, const char *  id, const int  svc, const char *  server  )

static

Definition at line 151 of file app_audiosocket.c.

{
    const char *chanName;
    struct ast_channel *targetChan;
    int ms = MAX_WAIT_TIMEOUT_MSEC;
    int outfd = -1;
    struct ast_frame *f;
    int hangup;
 
    if (!chan || ast_channel_state(chan) != AST_STATE_UP) {
        ast_log(LOG_ERROR, "Channel is %s\n", chan ? "not answered" : "missing");
        return -1;
    }
 
    if (ast_audiosocket_init(svc, id)) {
        ast_log(LOG_ERROR, "Failed to initialize AudioSocket\n");
        return -1;
    }
 
    chanName = ast_channel_name(chan);
 
    while (1) {
        /* Timeout is hard-coded currently, could be made into an
           argument if needed, but 2 seconds seems like a realistic
           time range to give. */
        targetChan = ast_waitfor_nandfds(&chan, 1, &svc, 1, NULL, &outfd, &ms);
        ms = MAX_WAIT_TIMEOUT_MSEC;
 
        if (targetChan) {
            /* Receive frame from connected channel. */
            f = ast_read(chan);
            if (!f) {
                ast_log(LOG_WARNING, "Failed to receive frame from channel '%s' connected to AudioSocket server '%s'", chanName, server);
                return -1;
            }
 
            if (f->frametype == AST_FRAME_VOICE || f->frametype == AST_FRAME_DTMF) {
                /* Send audio frame or DTMF frame to audiosocket */
                if (ast_audiosocket_send_frame(svc, f)) {
                    ast_log(LOG_WARNING, "Failed to forward frame from channel '%s' to AudioSocket server '%s'\n",
                        chanName, server);
                    ast_frfree(f);
                    return -1;
                }
            }
            ast_frfree(f);
 
        } else if (outfd >= 0) {
            /* Receive audio frame from audiosocket. */
            f = ast_audiosocket_receive_frame_with_hangup(svc, &hangup);
            if (hangup) {
                /* Graceful termination, no frame to free. */
                return 0;
            }
            if (!f) {
                ast_log(LOG_WARNING, "Failed to receive frame from AudioSocket server '%s'"
                    " connected to channel '%s'\n", server, chanName);
                return -1;
            }
            /* Send audio frame to connected channel. */
            if (ast_write(chan, f)) {
                ast_log(LOG_WARNING, "Failed to forward frame from AudioSocket server '%s' to channel '%s'\n", server, chanName);
                ast_frfree(f);
                return -1;
            }
            ast_frfree(f);
 
        } else {
            /* Neither the channel nor audio socket had activity
               before timeout. Assume connection was lost. */
            ast_log(LOG_ERROR, "Reached timeout after %d ms of no activity on AudioSocket connection between '%s' and '%s'\n", MAX_WAIT_TIMEOUT_MSEC, chanName, server);
            return -1;
        }
    }
    return 0;
}

References ast_audiosocket_init(), ast_audiosocket_receive_frame_with_hangup(), ast_audiosocket_send_frame(), ast_channel_name(), AST_FRAME_DTMF, AST_FRAME_VOICE, ast_frfree, ast_log, ast_read(), AST_STATE_UP, ast_waitfor_nandfds(), ast_write(), ast_frame::frametype, hangup(), LOG_ERROR, LOG_WARNING, MAX_WAIT_TIMEOUT_MSEC, and NULL.

Referenced by audiosocket_exec().

load_module()

static int load_module ( void  )

static

Definition at line 233 of file app_audiosocket.c.

{
    return ast_register_application_xml(app, audiosocket_exec);
}

References app, ast_register_application_xml, and audiosocket_exec().

unload_module()

static int unload_module ( void  )

static

Definition at line 228 of file app_audiosocket.c.

{
    return ast_unregister_application(app);
}

References app, and ast_unregister_application().

Variable Documentation

__mod_info

struct ast_module_info __mod_info = { .name = "app_audiosocket" , .flags = AST_MODFLAG_LOAD_ORDER , .description = "AudioSocket Application" , .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 247 of file app_audiosocket.c.

app

const char app[] = "AudioSocket"

static

Definition at line 74 of file app_audiosocket.c.

Referenced by load_module(), and unload_module().

ast_module_info

const struct ast_module_info* ast_module_info = &__mod_info

static

Definition at line 247 of file app_audiosocket.c.