Trivial application to record a sound file. More...

#include "asterisk.h"
#include "asterisk/file.h"
#include "asterisk/pbx.h"
#include "asterisk/module.h"
#include "asterisk/app.h"
#include "asterisk/channel.h"
#include "asterisk/dsp.h"

Include dependency graph for app_record.c:

Go to the source code of this file.

Enumerations
enum	{ OPTION_APPEND = (1 << 0), OPTION_NOANSWER = (1 << 1), OPTION_QUIET = (1 << 2), OPTION_SKIP = (1 << 3), OPTION_STAR_TERMINATE = (1 << 4), OPTION_IGNORE_TERMINATE = (1 << 5), OPTION_KEEP = (1 << 6), FLAG_HAS_PERCENT = (1 << 7), OPTION_ANY_TERMINATE = (1 << 8) }
Functions
static void	__reg_module (void)
static void	__unreg_module (void)
static int	load_module (void)
static int	record_exec (struct ast_channel chan, const char data)
static int	unload_module (void)
Variables
static struct ast_module_info	__mod_info = { .name = AST_MODULE, .flags = AST_MODFLAG_LOAD_ORDER , .description = "Trivial Record 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, .load = load_module, .unload = unload_module, .load_pri = AST_MODPRI_DEFAULT, }
static char *	app = "Record"
static struct ast_app_option	app_opts [128] = { [ 'a' ] = { .flag = OPTION_APPEND }, [ 'k' ] = { .flag = OPTION_KEEP }, [ 'n' ] = { .flag = OPTION_NOANSWER }, [ 'q' ] = { .flag = OPTION_QUIET }, [ 's' ] = { .flag = OPTION_SKIP }, [ 't' ] = { .flag = OPTION_STAR_TERMINATE }, [ 'y' ] = { .flag = OPTION_ANY_TERMINATE }, [ 'x' ] = { .flag = OPTION_IGNORE_TERMINATE },}
static struct ast_module_info *	ast_module_info = &__mod_info

Detailed Description

Trivial application to record a sound file.

Author:: Matthew Fredrickson <creslin@digium.com>

Definition in file app_record.c.

Enumeration Type Documentation

anonymous enum

Enumerator:

OPTION_APPEND
OPTION_NOANSWER
OPTION_QUIET
OPTION_SKIP
OPTION_STAR_TERMINATE
OPTION_IGNORE_TERMINATE
OPTION_KEEP
FLAG_HAS_PERCENT
OPTION_ANY_TERMINATE

Definition at line 118 of file app_record.c.

     {
   OPTION_APPEND = (1 << 0),
   OPTION_NOANSWER = (1 << 1),
   OPTION_QUIET = (1 << 2),
   OPTION_SKIP = (1 << 3),
   OPTION_STAR_TERMINATE = (1 << 4),
   OPTION_IGNORE_TERMINATE = (1 << 5),
   OPTION_KEEP = (1 << 6),
   FLAG_HAS_PERCENT = (1 << 7),
   OPTION_ANY_TERMINATE = (1 << 8),
};

Function Documentation

static void __reg_module ( void ) [static]

Definition at line 436 of file app_record.c.

static void __unreg_module ( void ) [static]

Definition at line 436 of file app_record.c.

static int load_module ( void ) [static]

Definition at line 431 of file app_record.c.

References ast_register_application_xml, and record_exec().

{
   return ast_register_application_xml(app, record_exec);
}

static int record_exec	(	struct ast_channel *	chan,
		const char *	data
	)		`[static]`

Definition at line 141 of file app_record.c.

References ast_channel::_state, app_opts, args, ast_answer(), AST_APP_ARG, ast_app_parse_options(), ast_channel_start_silence_generator(), ast_channel_stop_silence_generator(), ast_closestream(), AST_CONTROL_VIDUPDATE, ast_copy_string(), ast_debug, AST_DECLARE_APP_ARGS, ast_dsp_free(), ast_dsp_get_threshold_from_settings(), ast_dsp_new(), ast_dsp_set_threshold(), ast_dsp_silence(), AST_FILE_MODE, ast_filedelete(), ast_fileexists(), AST_FORMAT_SLINEAR, AST_FRAME_DTMF, AST_FRAME_VIDEO, AST_FRAME_VOICE, ast_frfree, ast_indicate(), ast_log(), ast_mkdir(), AST_NONSTANDARD_APP_ARGS, ast_opt_transmit_silence, ast_read(), ast_set_flag, ast_set_read_format(), AST_STANDARD_APP_ARGS, AST_STATE_UP, ast_stopstream(), ast_strdupa, ast_stream_rewind(), ast_streamfile(), ast_strlen_zero(), ast_test_flag, ast_truncstream(), ast_waitfor(), ast_waitstream(), ast_writefile(), ast_writestream(), ext, f, FLAG_HAS_PERCENT, ast_frame::frametype, ast_frame_subclass::integer, ast_channel::language, LOG_WARNING, ast_channel::name, OPTION_ANY_TERMINATE, OPTION_APPEND, OPTION_IGNORE_TERMINATE, OPTION_KEEP, OPTION_NOANSWER, OPTION_QUIET, OPTION_SKIP, OPTION_STAR_TERMINATE, parse(), pbx_builtin_setvar_helper(), ast_channel::readformat, ast_frame::subclass, THRESHOLD_SILENCE, and ast_dsp::totalsilence.

Referenced by load_module().

{
   int res = 0;
   int count = 0;
   char *ext = NULL, *opts[0];
   char *parse, *dir, *file;
   int i = 0;
   char tmp[256];

   struct ast_filestream *s = NULL;
   struct ast_frame *f = NULL;
   
   struct ast_dsp *sildet = NULL;      /* silence detector dsp */
   int totalsilence = 0;
   int dspsilence = 0;
   int silence = 0;     /* amount of silence to allow */
   int gotsilence = 0;     /* did we timeout for silence? */
   int maxduration = 0;    /* max duration of recording in milliseconds */
   int gottimeout = 0;     /* did we timeout for maxduration exceeded? */
   int terminator = '#';
   int rfmt = 0;
   int ioflags;
   int waitres;
   struct ast_silence_generator *silgen = NULL;
   struct ast_flags flags = { 0, };
   AST_DECLARE_APP_ARGS(args,
      AST_APP_ARG(filename);
      AST_APP_ARG(silence);
      AST_APP_ARG(maxduration);
      AST_APP_ARG(options);
   );
   
   /* The next few lines of code parse out the filename and header from the input string */
   if (ast_strlen_zero(data)) { /* no data implies no filename or anything is present */
      ast_log(LOG_WARNING, "Record requires an argument (filename)\n");
      pbx_builtin_setvar_helper(chan, "RECORD_STATUS", "ERROR");
      return -1;
   }

   parse = ast_strdupa(data);
   AST_STANDARD_APP_ARGS(args, parse);
   if (args.argc == 4)
      ast_app_parse_options(app_opts, &flags, opts, args.options);

   if (!ast_strlen_zero(args.filename)) {
      if (strstr(args.filename, "%d"))
         ast_set_flag(&flags, FLAG_HAS_PERCENT);
      ext = strrchr(args.filename, '.'); /* to support filename with a . in the filename, not format */
      if (!ext)
         ext = strchr(args.filename, ':');
      if (ext) {
         *ext = '\0';
         ext++;
      }
   }
   if (!ext) {
      ast_log(LOG_WARNING, "No extension specified to filename!\n");
      pbx_builtin_setvar_helper(chan, "RECORD_STATUS", "ERROR");
      return -1;
   }
   if (args.silence) {
      if ((sscanf(args.silence, "%30d", &i) == 1) && (i > -1)) {
         silence = i * 1000;
      } else if (!ast_strlen_zero(args.silence)) {
         ast_log(LOG_WARNING, "'%s' is not a valid silence duration\n", args.silence);
      }
   }
   
   if (args.maxduration) {
      if ((sscanf(args.maxduration, "%30d", &i) == 1) && (i > -1))
         /* Convert duration to milliseconds */
         maxduration = i * 1000;
      else if (!ast_strlen_zero(args.maxduration))
         ast_log(LOG_WARNING, "'%s' is not a valid maximum duration\n", args.maxduration);
   }

   if (ast_test_flag(&flags, OPTION_STAR_TERMINATE))
      terminator = '*';
   if (ast_test_flag(&flags, OPTION_IGNORE_TERMINATE))
      terminator = '\0';

   /* done parsing */

   /* these are to allow the use of the %d in the config file for a wild card of sort to
     create a new file with the inputed name scheme */
   if (ast_test_flag(&flags, FLAG_HAS_PERCENT)) {
      AST_DECLARE_APP_ARGS(fname,
         AST_APP_ARG(piece)[100];
      );
      char *tmp2 = ast_strdupa(args.filename);
      char countstring[15];
      int idx;

      /* Separate each piece out by the format specifier */
      AST_NONSTANDARD_APP_ARGS(fname, tmp2, '%');
      do {
         int tmplen;
         /* First piece has no leading percent, so it's copied verbatim */
         ast_copy_string(tmp, fname.piece[0], sizeof(tmp));
         tmplen = strlen(tmp);
         for (idx = 1; idx < fname.argc; idx++) {
            if (fname.piece[idx][0] == 'd') {
               /* Substitute the count */
               snprintf(countstring, sizeof(countstring), "%d", count);
               ast_copy_string(tmp + tmplen, countstring, sizeof(tmp) - tmplen);
               tmplen += strlen(countstring);
            } else if (tmplen + 2 < sizeof(tmp)) {
               /* Unknown format specifier - just copy it verbatim */
               tmp[tmplen++] = '%';
               tmp[tmplen++] = fname.piece[idx][0];
            }
            /* Copy the remaining portion of the piece */
            ast_copy_string(tmp + tmplen, &(fname.piece[idx][1]), sizeof(tmp) - tmplen);
         }
         count++;
      } while (ast_fileexists(tmp, ext, chan->language) > 0);
      pbx_builtin_setvar_helper(chan, "RECORDED_FILE", tmp);
   } else
      ast_copy_string(tmp, args.filename, sizeof(tmp));
   /* end of routine mentioned */

   if (chan->_state != AST_STATE_UP) {
      if (ast_test_flag(&flags, OPTION_SKIP)) {
         /* At the user's option, skip if the line is not up */
         pbx_builtin_setvar_helper(chan, "RECORD_STATUS", "SKIP");
         return 0;
      } else if (!ast_test_flag(&flags, OPTION_NOANSWER)) {
         /* Otherwise answer unless we're supposed to record while on-hook */
         res = ast_answer(chan);
      }
   }

   if (res) {
      ast_log(LOG_WARNING, "Could not answer channel '%s'\n", chan->name);
      pbx_builtin_setvar_helper(chan, "RECORD_STATUS", "ERROR");
      goto out;
   }

   if (!ast_test_flag(&flags, OPTION_QUIET)) {
      /* Some code to play a nice little beep to signify the start of the record operation */
      res = ast_streamfile(chan, "beep", chan->language);
      if (!res) {
         res = ast_waitstream(chan, "");
      } else {
         ast_log(LOG_WARNING, "ast_streamfile failed on %s\n", chan->name);
      }
      ast_stopstream(chan);
   }

   /* The end of beep code.  Now the recording starts */

   if (silence > 0) {
      rfmt = chan->readformat;
      res = ast_set_read_format(chan, AST_FORMAT_SLINEAR);
      if (res < 0) {
         ast_log(LOG_WARNING, "Unable to set to linear mode, giving up\n");
         pbx_builtin_setvar_helper(chan, "RECORD_STATUS", "ERROR");
         return -1;
      }
      sildet = ast_dsp_new();
      if (!sildet) {
         ast_log(LOG_WARNING, "Unable to create silence detector :(\n");
         pbx_builtin_setvar_helper(chan, "RECORD_STATUS", "ERROR");
         return -1;
      }
      ast_dsp_set_threshold(sildet, ast_dsp_get_threshold_from_settings(THRESHOLD_SILENCE));
   } 

   /* Create the directory if it does not exist. */
   dir = ast_strdupa(tmp);
   if ((file = strrchr(dir, '/')))
      *file++ = '\0';
   ast_mkdir (dir, 0777);

   ioflags = ast_test_flag(&flags, OPTION_APPEND) ? O_CREAT|O_APPEND|O_WRONLY : O_CREAT|O_TRUNC|O_WRONLY;
   s = ast_writefile(tmp, ext, NULL, ioflags, 0, AST_FILE_MODE);

   if (!s) {
      ast_log(LOG_WARNING, "Could not create file %s\n", args.filename);
      pbx_builtin_setvar_helper(chan, "RECORD_STATUS", "ERROR");
      goto out;
   }

   if (ast_opt_transmit_silence)
      silgen = ast_channel_start_silence_generator(chan);

   /* Request a video update */
   ast_indicate(chan, AST_CONTROL_VIDUPDATE);

   if (maxduration <= 0)
      maxduration = -1;

   while ((waitres = ast_waitfor(chan, maxduration)) > -1) {
      if (maxduration > 0) {
         if (waitres == 0) {
            gottimeout = 1;
            pbx_builtin_setvar_helper(chan, "RECORD_STATUS", "TIMEOUT");
            break;
         }
         maxduration = waitres;
      }

      f = ast_read(chan);
      if (!f) {
         res = -1;
         break;
      }
      if (f->frametype == AST_FRAME_VOICE) {
         res = ast_writestream(s, f);

         if (res) {
            ast_log(LOG_WARNING, "Problem writing frame\n");
            ast_frfree(f);
            pbx_builtin_setvar_helper(chan, "RECORD_STATUS", "ERROR");
            break;
         }

         if (silence > 0) {
            dspsilence = 0;
            ast_dsp_silence(sildet, f, &dspsilence);
            if (dspsilence) {
               totalsilence = dspsilence;
            } else {
               totalsilence = 0;
            }
            if (totalsilence > silence) {
               /* Ended happily with silence */
               ast_frfree(f);
               gotsilence = 1;
               pbx_builtin_setvar_helper(chan, "RECORD_STATUS", "SILENCE");
               break;
            }
         }
      } else if (f->frametype == AST_FRAME_VIDEO) {
         res = ast_writestream(s, f);

         if (res) {
            ast_log(LOG_WARNING, "Problem writing frame\n");
            pbx_builtin_setvar_helper(chan, "RECORD_STATUS", "ERROR");
            ast_frfree(f);
            break;
         }
      } else if ((f->frametype == AST_FRAME_DTMF) &&
            ((f->subclass.integer == terminator) ||
             (ast_test_flag(&flags, OPTION_ANY_TERMINATE)))) {
         ast_frfree(f);
         pbx_builtin_setvar_helper(chan, "RECORD_STATUS", "DTMF");
         break;
      }
      ast_frfree(f);
   }
   if (!f) {
      ast_debug(1, "Got hangup\n");
      res = -1;
      pbx_builtin_setvar_helper(chan, "RECORD_STATUS", "HANGUP");
      if (!ast_test_flag(&flags, OPTION_KEEP)) {
         ast_filedelete(args.filename, NULL);
      }
   }

   if (gotsilence) {
      ast_stream_rewind(s, silence - 1000);
      ast_truncstream(s);
   } else if (!gottimeout) {
      /* Strip off the last 1/4 second of it */
      ast_stream_rewind(s, 250);
      ast_truncstream(s);
   }
   ast_closestream(s);

   if (silgen)
      ast_channel_stop_silence_generator(chan, silgen);

out:
   if ((silence > 0) && rfmt) {
      res = ast_set_read_format(chan, rfmt);
      if (res)
         ast_log(LOG_WARNING, "Unable to restore read format on '%s'\n", chan->name);
      if (sildet)
         ast_dsp_free(sildet);
   }

   return res;
}

static int unload_module ( void ) [static]

Definition at line 426 of file app_record.c.

References ast_unregister_application().

{
   return ast_unregister_application(app);
}

Variable Documentation

struct ast_module_info __mod_info = { .name = AST_MODULE, .flags = AST_MODFLAG_LOAD_ORDER , .description = "Trivial Record 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, .load = load_module, .unload = unload_module, .load_pri = AST_MODPRI_DEFAULT, } [static]

Definition at line 436 of file app_record.c.

char* app = "Record" [static]

Definition at line 116 of file app_record.c.

struct ast_app_option app_opts[128] = { [ 'a' ] = { .flag = OPTION_APPEND }, [ 'k' ] = { .flag = OPTION_KEEP }, [ 'n' ] = { .flag = OPTION_NOANSWER }, [ 'q' ] = { .flag = OPTION_QUIET }, [ 's' ] = { .flag = OPTION_SKIP }, [ 't' ] = { .flag = OPTION_STAR_TERMINATE }, [ 'y' ] = { .flag = OPTION_ANY_TERMINATE }, [ 'x' ] = { .flag = OPTION_IGNORE_TERMINATE },} [static]

Definition at line 139 of file app_record.c.

Referenced by record_exec().

struct ast_module_info* ast_module_info = &__mod_info [static]

Definition at line 436 of file app_record.c.

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