예제 #1
0
파일: sprecog.cpp 프로젝트: nishimotz/dmcpp
bool SpRecog::start(Application *app)
{
  if (j_adin_init(this->recog) == FALSE) {    
    app->tell("error in j_adin_init");
    return false;
  }
  j_recog_info(this->recog); /* output system information to log */
  switch(j_open_stream(this->recog, NULL)) {
  case 0:			
    app->tell("ok j_open_stream");
    break;
  case -1:      		
    app->tell("error in input stream");
    return false;
  case -2:
    app->tell("error in beginning input");
    return false;
  }
  fflush(this->srm_log_fp);
  int ret = j_recognize_stream(this->recog);
  if (ret == -1) {
    app->tell("error j_recognize_stream");
    return false;
  }
  return true;
}
예제 #2
0
void SpeechSystem::setup()
{
	// ewwww
	jconf = j_config_load_file_new(const_cast<char*>(jconf_filename.c_str()));

	/* 2. create recognition instance according to the jconf */
	/* it loads models, setup final parameters, build lexicon
     and set up work area for recognition */
	recog = j_create_instance_from_jconf(jconf);
	if (recog == NULL)
	{
		fprintf(stderr, "Error in startup\n");
		return;
	}

	/*********************/
	/* Register callback */
	/*********************/
	/* register result callback functions */
	callback_add(recog, CALLBACK_EVENT_SPEECH_READY, recready, this);
	callback_add(recog, CALLBACK_EVENT_SPEECH_START, recstart, this);
	callback_add(recog, CALLBACK_RESULT, recdone, this);

	/**************************/
	/* Initialize audio input */
	/**************************/
	/* initialize audio input device */
	/* ad-in thread starts at this time for microphone */
	if (j_adin_init(recog) == FALSE) {    /* error */
		return;
	}

//#ifdef JULIUS_DEBUG
	/* output system information to log */
	j_recog_info(recog);
//#endif

	/***********************************/
	/* Open input stream and recognize */
	/***********************************/
	/* raw speech input (microphone etc.) */

	switch(j_open_stream(recog, NULL)) {
		case 0:			/* succeeded */
			break;
		case -1:      		/* error */
			fprintf(stderr, "error in input stream\n");
			return;
		case -2:			/* end of recognition process */
			fprintf(stderr, "failed to begin input stream\n");
			return;
	}

	startThread(true, false); // blocking, verbose
}
예제 #3
0
파일: adintool.c 프로젝트: xawirq/julius
/** 
 * <JA>
 * メイン関数
 * 
 * @param argc [in] 引数列の長さ
 * @param argv [in] 引数列
 * 
 * @return 
 * </JA>エラー時 1,通常終了時 0 を返す.
 * <EN>
 * Main function.
 * 
 * @param argc [in] number of argument.
 * @param argv [in] array of arguments.
 * 
 * @return 1 on error, 0 on success.
 * </EN>
 */
int
main(int argc, char *argv[])
{
  Recog *recog;
  Jconf *jconf;
  int ret;
  int i;
  boolean is_continues;

  /* create instance */
  recog = j_recog_new();
  jconf = j_jconf_new();
  recog->jconf = jconf;

  /********************/
  /* setup parameters */
  /********************/
  /* register additional options */
  j_add_option("-in", 1, 1, "input from", opt_in);
  j_add_option("-out", 1, 1, "output to", opt_out);
  j_add_option("-server", 1, 1, "hostname (-out adinnet)", opt_server);
  j_add_option("-NA", 1, 1, "NetAudio server host:unit (-in netaudio)", opt_NA);
  j_add_option("-port", 1, 1, "port number (-out adinnet)", opt_port);
  j_add_option("-inport", 1, 1, "port number (-in adinnet)", opt_inport);
  j_add_option("-filename", 1, 1, "(base) filename to record (-out file)", opt_filename);
  j_add_option("-startid", 1, 1, "recording start id (-out file)", opt_startid);
  j_add_option("-freq", 1, 1, "sampling frequency in Hz", opt_freq);
  j_add_option("-nosegment", 0, 0, "not segment input speech, record all", opt_nosegment);
  j_add_option("-segment", 0, 0, "force segment input speech", opt_segment);
  j_add_option("-oneshot", 0, 0, "exit after the first input", opt_oneshot);
  j_add_option("-raw", 0, 0, "save in raw (BE) format", opt_raw);
  j_add_option("-autopause", 0, 0, "automatically pause at each input end", opt_autopause);
  j_add_option("-loosesync", 0, 0, "loose sync of resume among servers", opt_loosesync);
  j_add_option("-rewind", 1, 1, "rewind to the msec", opt_rewind);
  j_add_option("-h", 0, 0, "display this help", opt_help);
  j_add_option("-help", 0, 0, "display this help", opt_help);
  j_add_option("--help", 0, 0, "display this help", opt_help);

  /* when no argument, output help and exit */
  if (argc <= 1) {
    opt_help(jconf, NULL, 0);
    return 0;
  }

  /* read arguments and set parameters */
  if (j_config_load_args(jconf, argc, argv) == -1) {
    fprintf(stderr, "Error reading arguments\n");
    return -1;
  }

  /* check needed arguments */
  if (speech_output == SPOUT_FILE && filename == NULL) {
    fprintf(stderr, "Error: output filename not specified\n");
    return(-1);
  }
  if (speech_output == SPOUT_ADINNET && adinnet_servnum < 1) {
    fprintf(stderr, "Error: adinnet server name for output not specified\n");
    return(-1);
  }
  if (jconf->input.speech_input == SP_ADINNET &&
      speech_output != SPOUT_ADINNET &&
      adinnet_servnum >= 1) {
    fprintf(stderr, "Warning: you specified port num by -port, but it's for output\n");
    fprintf(stderr, "Warning: you may specify input port by -inport instead.\n");
    fprintf(stderr, "Warning: now the default value (%d) will be used\n", ADINNET_PORT);
  }
#ifdef USE_NETAUDIO
  if (jconf->input.speech_input == SP_NETAUDIO && jconf->input.netaudio_devname == NULL) {
    fprintf(stderr, "Error: NetAudio server name not specified\n");
    return(-1);
  }
#endif
  if (adinnet_portnum != adinnet_servnum) {
    /* if only one server, use default */
    if (adinnet_servnum == 1) {
      adinnet_port[0] = ADINNET_PORT;
      adinnet_portnum = 1;
    } else {
      fprintf(stderr, "Error: you should specify both server names and different port for each!\n");
      fprintf(stderr, "\tserver:");
      for(i=0;i<adinnet_servnum;i++) fprintf(stderr, " %s", adinnet_serv[i]);
      fprintf(stderr, "\n\tport  :");
      for(i=0;i<adinnet_portnum;i++) fprintf(stderr, " %d", adinnet_port[i]);
      fprintf(stderr, "\n");
      return(-1);
    }
  }

  /* set Julius default parameters for unspecified acoustic parameters */
  apply_para(&(jconf->am_root->analysis.para), &(jconf->am_root->analysis.para_default));

  /* set some values */
  jconf->input.sfreq = jconf->am_root->analysis.para.smp_freq;
  jconf->input.period = jconf->am_root->analysis.para.smp_period;
  jconf->input.frameshift = jconf->am_root->analysis.para.frameshift;
  jconf->input.framesize = jconf->am_root->analysis.para.framesize;

  /* disable successive segmentation when no segmentation available */
  if (!jconf->detect.silence_cut) continuous_segment = FALSE;
  /* store sampling rate locally */
  sfreq = jconf->am_root->analysis.para.smp_freq;

  /********************/
  /* setup for output */
  /********************/
  if (speech_output == SPOUT_FILE) {
    /* allocate work area for output file name */
    if (continuous_segment) {
      outpath = (char *)mymalloc(strlen(filename) + 10);
    } else {
      if (use_raw) {
	outpath = filename;
      } else {
	outpath = new_output_filename(filename, ".wav");
      }
    }
  } else if (speech_output == SPOUT_ADINNET) {
    /* connect to adinnet server(s) */
    for(i=0;i<adinnet_servnum;i++) {
      fprintf(stderr, "connecting to #%d (%s:%d)...", i+1, adinnet_serv[i], adinnet_port[i]);
      sd[i] = make_connection(adinnet_serv[i], adinnet_port[i]);
      if (sd[i] < 0) return 1;	/* on error */
      fprintf(stderr, "connected\n");
    }
  } else if (speech_output == SPOUT_STDOUT) {
    /* output to stdout */
    fd = 1;
    fprintf(stderr,"[STDOUT]");
  }

  /**********************/
  /* interrupt handling */
  /**********************/
  if (signal(SIGINT, interrupt_record) == SIG_ERR) {
    fprintf(stderr, "Warning: signal interruption may collapse output\n");
  }
  if (signal(SIGTERM, interrupt_record) == SIG_ERR) {
    fprintf(stderr, "Warning: signal interruption may collapse output\n");
  }
#ifdef SIGPIPE
  if (signal(SIGPIPE, interrupt_record) == SIG_ERR) {
    fprintf(stderr, "Warning: signal interruption may collapse output\n");
  }
#endif
#ifdef SIGQUIT
  if (signal(SIGQUIT, interrupt_record) == SIG_ERR) {
    fprintf(stderr, "Warning: signal interruption may collapse output\n");
  }
#endif

  /***************************/
  /* initialize input device */
  /***************************/
  if (jconf->input.speech_input == SP_ADINNET) {
    jconf->input.adinnet_port = adinnet_port_in;
  }
  if (j_adin_init(recog) == FALSE) {
    fprintf(stderr, "Error in initializing adin device\n");
    return -1;
  }
  if (rewind_msec > 0) {
    /* allow adin module to keep triggered speech while pausing */
#ifdef HAVE_PTHREAD
    if (recog->adin->enable_thread) {
      recog->adin->ignore_speech_while_recog = FALSE;
    }
#endif
  }

  /*********************/
  /* add some callback */
  /*********************/
  callback_add(recog, CALLBACK_EVENT_SPEECH_START, record_trigger_time, NULL);


  /**************************************/
  /* display input/output configuration */
  /**************************************/
  put_status(recog);

  /*******************/
  /* begin recording */
  /*******************/
  if (continuous_segment) {	/* reset parameter for successive output */
    total_speechlen = 0;
    sid = startid;
  }
  fprintf(stderr,"[start recording]\n");
  if (jconf->input.speech_input == SP_RAWFILE) file_counter = 0;

  /*********************/
  /* input stream loop */
  /*********************/
  while(1) {

    /* begin A/D input of a stream */
    ret = j_open_stream(recog, NULL);
    switch(ret) {
    case 0:			/* succeeded */
      break;
    case -1:      		/* error */
      /* go on to next input */
      continue;
    case -2:			/* end of recognition process */
      switch(jconf->input.speech_input) {
      case SP_RAWFILE:
	fprintf(stderr, "%d files processed\n", file_counter);
	break;
      case SP_STDIN:
	fprintf(stderr, "reached end of input on stdin\n");
	break;
      default:
	fprintf(stderr, "failed to begin input stream\n");
      }
      /* exit recording */
      goto record_end;
    }

    /*********************************/
    /* do VAD and recording */
    /*********************************/
    do {
      /* process one segment with segmentation */
      /* for incoming speech input, speech detection and segmentation are
	 performed and, adin_callback_* is called for speech output for each segment block.
      */
      /* adin_go() return when input segmented by long silence, or input stream reached to the end */
      speechlen = 0;
      stop_at_next = FALSE;
      if (jconf->input.speech_input == SP_MIC) {
	fprintf(stderr, "<<< please speak >>>");
      }
      if (speech_output == SPOUT_ADINNET) {
	ret = adin_go(adin_callback_adinnet, adinnet_check_command, recog);
      } else {
	ret = adin_go(adin_callback_file, NULL, recog);
      }
      /* return value of adin_go:
	 -2: input terminated by pause command from adinnet server
	 -1: input device read error or callback process error
	 0:  paused by input stream (end of file, etc..)
	 >0: detected end of speech segment:
             by adin-cut, or by callback process
	 (or return value of ad_check (<0) (== not used in this program))
      */
      /* if PAUSE or TERMINATE command has been received while input,
	 stop_at_next is TRUE here  */
      switch(ret) {
      case -2:	     /* terminated by terminate command from server */
	fprintf(stderr, "[terminated by server]\n");
	break;
      case -1:		     /* device read error or callback error */
	fprintf(stderr, "[error]\n");
	break;
      case 0:			/* reached to end of input */
	fprintf(stderr, "[eof]\n");
	break;
      default:	  /* input segmented by silence or callback process */
	fprintf(stderr, "[segmented]\n");
	break;
      }
      
      if (ret == -1) {
	/* error in input device or callback function, so terminate program here */
	return 1;
      }

      /*************************/
      /* one segment processed */
      /*************************/
      if (speech_output == SPOUT_FILE) {
	/* close output files */
	if (close_files() == FALSE) return 1;
      } else if (speech_output == SPOUT_ADINNET) {
	if (speechlen > 0) {
	  if (ret >= 0 || stop_at_next) { /* segmented by adin-cut or end of stream or server-side command */
	    /* send end-of-segment ack to client */
	    adin_send_end_of_segment();
	  }
	  /* output info */
	  printf("sent: %d samples (%.2f sec.) [%6d (%5.2fs) - %6d (%5.2fs)]\n", 
		 speechlen, (float)speechlen / (float)sfreq,
		 trigger_sample, (float)trigger_sample / (float)sfreq, 
		 trigger_sample + speechlen, (float)(trigger_sample + speechlen) / (float)sfreq);
	}
      }

      /*************************************/
      /* increment ID and total sample len */
      /*************************************/
      if (continuous_segment) {
	total_speechlen += speechlen;
	sid++;
      }

      /***************************************************/
      /* with adinnet server, if terminated by           */
      /* server-side PAUSE command, wait for RESUME here */
      /***************************************************/
      if (pause_each) {
	/* pause at each end */
	//if (speech_output == SPOUT_ADINNET && speechlen > 0) {
	if (speech_output == SPOUT_ADINNET) {
	  if (adinnet_wait_command() < 0) {
	    /* command error: terminate program here */
	    return 1;
	  }
	}
      } else {
	if (speech_output == SPOUT_ADINNET && stop_at_next) {
	  if (adinnet_wait_command() < 0) {
	    /* command error: terminate program here */
	    return 1;
	  }
	}
      }

      /* loop condition check */
      is_continues = FALSE;
      if (continuous_segment && (ret > 0 || ret == -2)) {
	is_continues = TRUE;
      }
      
    } while (is_continues); /* to the next segment in this input stream */

    /***********************/
    /* end of input stream */
    /***********************/
    adin_end(recog->adin);

  } /* to the next input stream (i.e. next input file in SP_RAWFILE) */

 record_end:

  if (speech_output == SPOUT_FILE) {
    if (continuous_segment) {
      printf("recorded total %d samples (%.2f sec.) segmented to %s.%04d - %s.%04d files\n", total_speechlen, (float)total_speechlen / (float)sfreq, filename, 0, filename, sid-1);
    }
  }

  return 0;
}
예제 #4
0
int
main(int argc, char *argv[])
{
  Recog *recog;
  Jconf *jconf;
  float *ss;
  MFCCWork *wrk;

  /* create instance */
  recog = j_recog_new();
  jconf = j_jconf_new();
  recog->jconf = jconf;

  /* set application-specific additional options */
  j_add_option("-freq", 1, 1, "sampling freq in Hz", opt_freq);
  j_add_option("-len", 1, 1, "record length in msec", opt_len);
  j_add_option("-h", 0, 0, "display this help", opt_help);
  j_add_option("-help", 0, 0, "display this help", opt_help);
  j_add_option("--help", 0, 0, "display this help", opt_help);

  /* when no argument, output help and exit */
  if (argc <= 1) {
    opt_help(jconf, NULL, 0);
    return 0;
  }

  /* regard last arg as filename */
  if (strmatch(argv[argc-1], "-")) {
    stout = TRUE;
  } else {
    filename = argv[argc-1];
  }

  /* set default as same as "-input mic" */
  jconf->input.type = INPUT_WAVEFORM;
  jconf->input.speech_input = SP_MIC;
  jconf->input.device = SP_INPUT_DEFAULT;
  /* process config and load them */
  if (j_config_load_args(jconf, argc-1, argv) == -1) {
    fprintf(stderr, "Error reading arguments\n");
    return -1;
  }
  /* force some default values */
  jconf->detect.silence_cut  = 0; /* disable silence cut */
  jconf->preprocess.strip_zero_sample = TRUE; /* strip zero samples */
  jconf->detect.level_thres = 0;	/* no VAD, record all */
  /* set Julius default parameters for unspecified acoustic parameters */
  apply_para(&(jconf->am_root->analysis.para), &(jconf->am_root->analysis.para_default));
  /* set some values */
  jconf->input.sfreq = jconf->am_root->analysis.para.smp_freq;
  jconf->input.period = jconf->am_root->analysis.para.smp_period;
  jconf->input.frameshift = jconf->am_root->analysis.para.frameshift;
  jconf->input.framesize = jconf->am_root->analysis.para.framesize;

  sfreq = jconf->am_root->analysis.para.smp_freq;

  /* output file check */
  if (!stout) {
    if (access(filename, F_OK) == 0) {
      if (access(filename, W_OK) == 0) {
	fprintf(stderr, "Warning: overwriting file \"%s\"\n", filename);
      } else {
	perror("mkss");
	return(1);
      }
    }
  }

  /* allocate speech store buffer */
  samples = sfreq * slen / 1000;
  speech = (SP16 *)mymalloc(sizeof(SP16) * samples);

  /* allocate work area to compute spectrum */
  wrk = WMP_work_new(&(jconf->am_root->analysis.para));
  if (wrk == NULL) {
    jlog("ERROR: m_fusion: failed to initialize MFCC computation for SS\n");
    return -1;
  }

  /* initialize input device */
  if (j_adin_init(recog) == FALSE) {
    fprintf(stderr, "Error in initializing adin device\n");
    return -1;
  }

  /* open device */
  if (j_open_stream(recog, NULL) < 0) {
    fprintf(stderr, "Error in opening adin device\n");
  }

  /* record mic input */
  fprintf(stderr, "%dHz recording for %.2f seconds of noise\n", sfreq, (float)slen /(float)1000);
  speechnum = 0;
  adin_go(adin_callback, NULL, recog);

  /* close device */
  adin_end(recog->adin);
  fprintf(stderr, "\n%d samples (%d bytes, %.1f sec) recorded\n", samples, samples * sizeof(SP16), (float)samples / (float)sfreq);

  /* compute SS */
  fprintf(stderr, "compute SS:\n");
  fprintf(stderr, "  fsize : %4d samples (%.1f msec)\n", jconf->input.framesize, (float)jconf->input.framesize * 1000.0/ (float)sfreq);
  fprintf(stderr, "  fshift: %4d samples (%.1f msec)\n", jconf->input.frameshift, (float)jconf->input.frameshift * 1000.0/ (float)sfreq);

  ss = new_SS_calculate(speech, samples, &sslen, wrk, &(jconf->am_root->analysis.para));

  fprintf(stderr, "  points: %4d\n", sslen);
  fprintf(stderr, "noise spectrum was measured\n");
  
  /* open file for recording */
  fprintf(stderr, "writing average noise spectrum to [%s]...", filename);
  if (stout) {
    fd = 1;
  } else {
    if ((fd = open(filename, O_CREAT | O_RDWR
#ifdef O_BINARY
		   | O_BINARY
#endif
		   , 0644)) == -1) {
      perror("mkss");
      return(1);
    }
  }
  x = sslen;
#ifndef WORDS_BIGENDIAN
  swap_bytes((char *)&x, sizeof(int), 1);
#endif
  if (write(fd, &x, sizeof(int)) < sizeof(int)) {
    perror("mkss");
    return(1);
  }
#ifndef WORDS_BIGENDIAN
  swap_bytes((char *)ss, sizeof(float), sslen);
#endif
  if (write(fd, ss, sslen * sizeof(float)) < sslen * sizeof(float)) {
    perror("mkss");
    return(1);
  }
  if (!stout) {
    if (close(fd) < 0) {
      perror("mkss");
      return(1);
    }
  }
  fprintf(stderr, "done\n");

  WMP_free(wrk);

  return 0;
}
예제 #5
0
/** 
 * <JA>
 * メイン関数
 * 
 * @param argc [in] 引数列の長さ
 * @param argv [in] 引数列
 * 
 * @return 
 * </JA>エラー時 1,通常終了時 0 を返す.
 * <EN>
 * Main function.
 * 
 * @param argc [in] number of argument.
 * @param argv [in] array of arguments.
 * 
 * @return 1 on error, 0 on success.
 * </EN>
 */
int
main(int argc, char *argv[])
{
  Recog *recog;
  Jconf *jconf;

  /* create instance */
  jconf = j_jconf_new();

  /* register application options */
  j_add_option("-freq", 1, 1, "sampling frequency in Hz", opt_freq);
  j_add_option("-raw", 0, 0, "save in raw (BE) format", opt_raw);
  j_add_option("-h", 0, 0, "display this help", opt_help);
  j_add_option("-help", 0, 0, "display this help", opt_help);
  j_add_option("--help", 0, 0, "display this help", opt_help);

  /* when no argument, output help and exit */
  if (argc <= 1) {
    opt_help(jconf, NULL, 0);
    return 0;
  }

  /* regard last arg as filename */
  if (strmatch(argv[argc-1], "-")) {
    stout = TRUE;
    use_raw = TRUE;
  } else {
    filename = argv[argc-1];
  }

  /* set default as same as "-input mic" */
  jconf->input.type = INPUT_WAVEFORM;
  jconf->input.speech_input = SP_MIC;
  jconf->input.device = SP_INPUT_DEFAULT;

  /* read arguments and set parameters */
  if (j_config_load_args(jconf, argc-1, argv) == -1) {
    fprintf(stderr, "Error reading arguments\n");
    return -1;
  }

  /* exit if no file name specified */
  if (filename == NULL && stout == FALSE) {
    opt_help(jconf, NULL, 0);
    return -1;
  }

  /* finalize config */
  //if (j_jconf_finalize(jconf) == FALSE) return -1;

  /* set Julius default parameters for unspecified acoustic parameters */
  apply_para(&(jconf->am_root->analysis.para), &(jconf->am_root->analysis.para_default));
  
  /* set some values */
  jconf->input.sfreq = jconf->am_root->analysis.para.smp_freq;
  jconf->input.period = jconf->am_root->analysis.para.smp_period;
  jconf->input.frameshift = jconf->am_root->analysis.para.frameshift;
  jconf->input.framesize = jconf->am_root->analysis.para.framesize;

  /* preliminary check of output file */
  /* (output file will be opened later when input is triggered) */
  if (!stout) {
    if (access(filename, F_OK) == 0) {
      if (access(filename, W_OK) == 0) {
	fprintf(stderr, "Warning: overwriting file \"%s\"\n", filename);
      } else {
	perror("adinrec");
	return(1);
      }
    }
  }
  /* set signal handlers to properly close output file */
  if (signal(SIGINT, interrupt_record) == SIG_ERR) {
    fprintf(stderr, "Warning: signal intterupt may collapse output\n");
  }
  if (signal(SIGTERM, interrupt_record) == SIG_ERR) {
    fprintf(stderr, "Warning: signal intterupt may collapse output\n");
  }
#ifdef SIGPIPE
  if (signal(SIGPIPE, interrupt_record) == SIG_ERR) {
    fprintf(stderr, "Warning: signal intterupt may collapse output\n");
  }
#endif
#ifdef SIGQUIT
  if (signal(SIGQUIT, interrupt_record) == SIG_ERR) {
    fprintf(stderr, "Warning: signal intterupt may collapse output\n");
  }
#endif

  recog = j_recog_new();
  recog->jconf = jconf;

  /* initialize input device */
  if (j_adin_init(recog) == FALSE) {
    fprintf(stderr, "Error in initializing adin device\n");
    return -1;
  }
  /* open device */
  if (j_open_stream(recog, NULL) < 0) {
    fprintf(stderr, "Error in opening adin device\n");
  }
  /* do recoding */
  speechlen = 0;
  sfreq = recog->jconf->input.sfreq;
  fprintf(stderr, "<<< please speak >>>"); /* moved from adin-cut.c */
  adin_go(adin_callback_file, NULL, recog);
  /* close device */
  adin_end(recog->adin);
  /* close output file */
  close_file();

  return 0;
}
예제 #6
0
int
main(int argc, char *argv[])
{
  FILE *fp;
  Recog *recog;
  Jconf *jconf;

  /* inihibit system log output (default: stdout) */
  //jlog_set_output(NULL);
  /* output system log to a file */
  // FILE *fp = fopen(logfile, "w"); jlog_set_output(fp);

  /* if no option argument, output julius usage and exit */
  if (argc == 1) {
    fprintf(stderr, "Julius rev.%s - based on ", JULIUS_VERSION);
    j_put_version(stderr);
    fprintf(stderr, "Try '-setting' for built-in engine configuration.\n");
    fprintf(stderr, "Try '-help' for run time options.\n");
    return -1;
  }

  /* add application options */
  record_add_option();
  module_add_option();
  charconv_add_option();
  j_add_option("-separatescore", 0, 0, "output AM and LM scores separately", opt_separatescore);
  j_add_option("-logfile", 1, 1, "output log to file", opt_logfile);
  j_add_option("-nolog", 0, 0, "not output any log", opt_nolog);
  j_add_option("-outfile", 0, 0, "save result in separate .out file", opt_outfile);
  j_add_option("-help", 0, 0, "display this help", opt_help);
  j_add_option("--help", 0, 0, "display this help", opt_help);

  /* create a configuration variables container */
  jconf = j_jconf_new();
  // j_config_load_file(jconf, jconffile);
  if (j_config_load_args(jconf, argc, argv) == -1) {
    fprintf(stderr, "Try `-help' for more information.\n");
    return -1;
  }

  /* output system log to a file */
  if (nolog) {
    jlog_set_output(NULL);
  } else if (logfile) {
    fp = fopen(logfile, "w");
    jlog_set_output(fp);
  }

  /* here you can set/modify any parameter in the jconf before setup */
  // jconf->input.input_speech = SP_MIC;

  /* Fixate jconf parameters: it checks whether the jconf parameters
     are suitable for recognition or not, and set some internal
     parameters according to the values for recognition.  Modifying
     a value in jconf after this function may be errorous.
  */
  if (j_jconf_finalize(jconf) == FALSE) {
    if (logfile) fclose(fp);
    return -1;
  }

  /* create a recognition instance */
  recog = j_recog_new();
  /* assign configuration to the instance */
  recog->jconf = jconf;
  /* load all files according to the configurations */
  if (j_load_all(recog, jconf) == FALSE) {
    fprintf(stderr, "ERROR: Error in loading model\n");
    if (logfile) fclose(fp);
    return -1;
  }
  
#ifdef USER_LM_TEST
  {
    PROCESS_LM *lm;
    for(lm=recog->lmlist;lm;lm=lm->next) {
      if (lm->lmtype == LM_PROB) {
	j_regist_user_lm_func(lm, my_uni, my_bi, my_lm);
      }
    }
#endif

  /* checkout for recognition: build lexicon tree, allocate cache */
  if (j_final_fusion(recog) == FALSE) {
    fprintf(stderr, "ERROR: Error while setup work area for recognition\n");
    j_recog_free(recog);
    if (logfile) fclose(fp);
    return -1;
  }
  
  /* Set up some application functions */
  /* set character conversion mode */
  if (charconv_setup() == FALSE) {
    if (logfile) fclose(fp);
    return -1;
  }
  if (is_module_mode()) {
    /* set up for module mode */
    /* register result output callback functions to network module */
    module_setup(recog, NULL);
  } else {
    /* register result output callback functions to stdout */
    setup_output_tty(recog, NULL);
  }
  /* if -outfile option specified, callbacks for file output will be
     regitered */
  if (outfile_enabled) {
    if (jconf->input.speech_input == SP_MFCFILE || jconf->input.speech_input == SP_RAWFILE) {
      setup_output_file(recog, NULL);
    } else {
      fprintf(stderr, "Warning: -outfile works only for file input, disabled now\n");
      outfile_enabled = FALSE;
    }
  }

  /* setup recording if option was specified */
  record_setup(recog, NULL);

  /* on module connect with client */
  if (is_module_mode()) module_server();

  /* initialize and standby the specified audio input source */
  /* for microphone or other threaded input, ad-in thread starts here */
  if (j_adin_init(recog) == FALSE) return;

  /* output system information to log */
  j_recog_info(recog);

#ifdef VISUALIZE
  /* Visualize: initialize GTK */
  visual_init(recog);
  callback_add(recog, CALLBACK_EVENT_RECOGNITION_END, visual_show, NULL);
  callback_add(recog, CALLBACK_EVENT_PASS2_BEGIN, visual2_init, NULL);
  callback_add(recog, CALLBACK_DEBUG_PASS2_POP, visual2_popped, NULL);
  callback_add(recog, CALLBACK_DEBUG_PASS2_PUSH, visual2_next_word, NULL);
  /* below should be called at result */
  visual2_best(now, winfo);
  /* 音声取り込みはコールバックで新規作成 */
  /* 第2パスで認識結果出力時に以下を実行 */
  visual2_best(now, recog->model->winfo);
#endif
  
  /* if no grammar specified on startup, start with pause status */
  {
    RecogProcess *r;
    boolean ok_p;
    ok_p = TRUE;
    for(r=recog->process_list;r;r=r->next) {
      if (r->lmtype == LM_DFA) {
	if (r->lm->winfo == NULL) { /* stop when no grammar found */
	  j_request_pause(recog);
	}
      }
    }
  }

  /* enter recongnition loop */
  main_recognition_stream_loop(recog);

  /* end proc */
  if (is_module_mode()) module_disconnect();

  /* release all */
  j_recog_free(recog);

  if (logfile) fclose(fp);
  return(0);
}
예제 #7
0
int main(int argc, char* argv[])
{
	// Jconf: configuration parameters
	// load configurations from command arguments
	Jconf *jconf = j_config_load_args_new(argc, argv);
	if (jconf == NULL) {
		std::cout << "Error @ j_config_load_args_new" << std::endl;
		return -1;
	}

	// Recog: Top level instance for the whole recognition process
	// create recognition instance according to the jconf
	Recog *recog = j_create_instance_from_jconf(jconf);
	if (recog == NULL) {
		std::cout << "Error @ j_create_instance_from_jconf" << std::endl;
		return -1;
	}

	// Regster callback
	callback_add(recog, CALLBACK_EVENT_SPEECH_READY, [](Recog *recog, void*) {
		std::cout << "<<< PLEASE SPEAK! >>>" << std::endl;
	}, NULL);

	callback_add(recog, CALLBACK_EVENT_SPEECH_START, [](Recog *recog, void*) {
		std::cout << "...SPEECH START..." << std::endl;
	}, NULL);

	callback_add(recog, CALLBACK_RESULT, [](Recog *recog, void*) {
		for (const RecogProcess *r = recog->process_list; r; r = r->next) {
			WORD_INFO *winfo = r->lm->winfo;
			for (int n = 0; n < r->result.sentnum; ++n) {
				Sentence *s   = &(r->result.sent[n]);
				WORD_ID *seq = s->word;
				int seqnum   = s->word_num;
				for (int i = 0; i < seqnum; ++i) {
					std::cout << winfo->woutput[seq[i]];
				}
			}
		}
	}, NULL);

	// Initialize audio input
	if (j_adin_init(recog) == FALSE) {
		return -1;
	}

	// output system information to log
	j_recog_info(recog);

	// Open input stream and recognize
	switch (j_open_stream(recog, NULL)) {
		case  0: break; // success
		case -1: std::cout << "Error in input stream" << std::endl; return -1;
		case -2: std::cout << "Failed to begin input stream" << std::endl; return -1;
	}

	// Recognition loop
	int ret = j_recognize_stream(recog);
	if (ret == -1) return -1;

	// exit
	j_close_stream(recog);
	j_recog_free(recog);

	return 0;
}
예제 #8
0
//========================
// Create engine instance
//========================
bool cJulius::createEngine( void )
{
#ifdef APP_ADIN
	ADIn *a;
#endif

	if (!m_jconf) return false;
	if (m_recog) return false;

#ifdef APP_ADIN
	if (m_appsource != 0) {
		switch(m_appsource) {
			case 1: // buffer input, batch
				m_recog->jconf->input.type = INPUT_WAVEFORM;
				m_recog->jconf->input.speech_input = SP_RAWFILE;
				m_recog->jconf->decodeopt.realtime_flag = FALSE;
				break;
			case 2: // buffer input, incremental
				m_recog->jconf->input.type = INPUT_WAVEFORM;
				m_recog->jconf->input.speech_input = SP_RAWFILE;
				m_recog->jconf->decodeopt.realtime_flag = TRUE;
				break;
		}
	}
#endif

	// Create engine instance
	m_recog = j_create_instance_from_jconf(m_jconf);
	if (m_recog == NULL) {
		return false;
	}

	// Register callbacks
	callback_add(m_recog, CALLBACK_EVENT_PROCESS_ONLINE,		::callback_engine_active, this);
	callback_add(m_recog, CALLBACK_EVENT_PROCESS_OFFLINE,		::callback_engine_inactive, this);
	callback_add(m_recog, CALLBACK_EVENT_SPEECH_READY,			::callback_audio_ready, this);
	callback_add(m_recog, CALLBACK_EVENT_SPEECH_START,			::callback_audio_begin, this);
	callback_add(m_recog, CALLBACK_EVENT_SPEECH_STOP,			::callback_audio_end, this);
	callback_add(m_recog, CALLBACK_EVENT_RECOGNITION_BEGIN,		::callback_recog_begin, this);
	callback_add(m_recog, CALLBACK_EVENT_RECOGNITION_END,		::callback_recog_end, this);
	callback_add(m_recog, CALLBACK_EVENT_PASS1_FRAME,			::callback_recog_frame, this);
	callback_add(m_recog, CALLBACK_EVENT_PAUSE,					::callback_engine_pause, this);
	callback_add(m_recog, CALLBACK_EVENT_RESUME,				::callback_engine_resume, this);
	callback_add(m_recog, CALLBACK_RESULT,						::callback_result_final, this);
	callback_add(m_recog, CALLBACK_PAUSE_FUNCTION,				::callback_wait_for_resume, this);

#ifdef APP_ADIN
	// Initialize application side audio input
	if (m_appsource != 0) {
		a = m_recog->adin;
		switch(m_appsource) {
			case 1: // buffer input, batch
				a->ad_standby			= NULL;
				a->ad_begin				= NULL;
				a->ad_end				= NULL;
				a->ad_resume			= NULL;
				a->ad_pause				= NULL;
				a->ad_terminate			= NULL;
				a->ad_read				= callback_adin_fetch_input;
				a->ad_input_name		= NULL;
				a->silence_cut_default	= FALSE;
				a->enable_thread		= FALSE;
				break;
			case 2: // buffer input, incremental
				a->ad_standby			= NULL;
				a->ad_begin				= NULL;
				a->ad_end				= NULL;
				a->ad_resume			= NULL;
				a->ad_pause				= NULL;
				a->ad_terminate			= NULL;
				a->ad_read				= callback_adin_fetch_input;
				a->ad_input_name		= NULL;
				a->silence_cut_default	= FALSE;
				a->enable_thread		= FALSE;
				break;
		}
	    a->ds = NULL;
		a->down_sample = FALSE;
		if (adin_standby(a, m_recog->jconf->input.sfreq, NULL) == FALSE) return false;
		if (adin_setup_param(a, m_recog->jconf) == FALSE) return false;
		a->input_side_segment = FALSE;
	} else {
		// Let JuliusLib get audio input
		if (! j_adin_init( m_recog ) ) return false;
	}
#else
	if (! j_adin_init( m_recog ) ) return false;
#endif

	return true;
}
예제 #9
0
xreturn::r<bool> JuliusPlus::JuliusFileStart()
{
	assert(this->recogFile == NULL);
	assert(this->jconfFile == NULL);
	const char* argv[]={
		"juliusplus"
		,"-C"
		,"testfile.jconf"
	};
	int argc = sizeof(argv)/sizeof(argv[0]);

	//julius は C関数だから、const外して char** にするしかない。
	this->jconfFile = j_config_load_args_new(argc, (char**)argv);
	/* else, you can load configurations from a jconf file */
	//jconf = j_config_load_file_new(jconf_filename);
	if (this->jconfFile == NULL) 
	{
		return xreturn::error("Try `-help' for more information.\n");
	}

	/* 2. create recognition instance according to the jconf */
	/* it loads models, setup final parameters, build lexicon
	and set up work area for recognition */
	this->recogFile = j_create_instance_from_jconf(this->jconfFile);
	if (this->recogFile == NULL)
	{
		return xreturn::error("Error in startup(j_create_instance_from_jconf)\n");
	}


	struct _ref{
		static void output_result(Recog *recog, void *_this)
		{
			((JuliusPlus*)_this)->OnOutputResultFile(recog);
		}
	};
	callback_add(this->recogFile, CALLBACK_RESULT, _ref::output_result, this);

	// Initialize audio input
	if (j_adin_init(this->recogFile) == FALSE) 
	{
		return xreturn::error("Error in startup(j_adin_init)\n");
	}
	//以上、準備だけしておいて、
	//認識ルーチンは、後から呼びます。
/*
	int ret = j_open_stream(recogFile, "nano.wav");
	if(ret < 0)
	{
		return xreturn::error("Error in startup(j_open_stream)\n");
	}

	j_recognize_stream(recogFile);

	ret = j_open_stream(recogFile, "nano.wav");
	if(ret < 0)
	{
		return xreturn::error("Error in startup(j_open_stream)\n");
	}

	j_recognize_stream(recogFile);
*/
	return true;
}
예제 #10
0
xreturn::r<bool> JuliusPlus::JuliusStart()
{
	assert(this->recog == NULL);
	assert(this->jconf == NULL);
	assert(this->Thread == NULL);
	const char* argv[]={
		"juliusplus"
		,"-C"
		,"testmic.jconf"
	};
	int argc = sizeof(argv)/sizeof(argv[0]);
	int ret;

	//julusはC関数なので、const外して char** にするしかない・・・
	this->jconf = j_config_load_args_new(argc, (char**)argv);
	/* else, you can load configurations from a jconf file */
	//jconf = j_config_load_file_new(jconf_filename);
	if (this->jconf == NULL) 
	{
		return xreturn::error("Try `-help' for more information.\n");
	}

	/* 2. create recognition instance according to the jconf */
	/* it loads models, setup final parameters, build lexicon
	and set up work area for recognition */
	this->recog = j_create_instance_from_jconf(this->jconf);
	if (this->recog == NULL)
	{
		return xreturn::error("Error in startup(j_create_instance_from_jconf)\n");
	}
	struct _ref{
		static void status_recready(Recog *recog, void *_this)
		{
			((JuliusPlus*)_this)->OnStatusRecready(recog);
		}
		static void status_recstart(Recog *recog, void *_this)
		{
			((JuliusPlus*)_this)->OnStatusRecstart(recog);
		}
		static void output_result(Recog *recog, void *_this)
		{
			((JuliusPlus*)_this)->OnOutputResult(recog);
		}
		static void record_adin_trigger(Recog *recog, SP16 *speech, int samplenum, void *_this)
		{
			((JuliusPlus*)_this)->OnRecordAdinTrigger(recog,speech,samplenum);
		}
	};
	callback_add(this->recog, CALLBACK_EVENT_SPEECH_READY, _ref::status_recready, this);
	callback_add(this->recog, CALLBACK_EVENT_SPEECH_START, _ref::status_recstart, this);
	callback_add(this->recog, CALLBACK_RESULT, _ref::output_result, this);
	callback_add_adin(this->recog, CALLBACK_ADIN_TRIGGERED, _ref::record_adin_trigger, this);

	// Initialize audio input
	if (j_adin_init(this->recog) == FALSE) 
	{
		return xreturn::error("Error in startup(j_adin_init)\n");
	}

	//output system information to log
	//j_recog_info(this->recog);
	ret = j_open_stream(recog, NULL);
	if(ret < 0)
	{
		return xreturn::error("Error in startup(j_open_stream)\n");
	}

	this->Thread = new boost::thread( [&]()
	{
		j_recognize_stream(recog);
	} );
	return true;
}
/* Julius_Thread::run: main loop */
void Julius_Thread::run()
{
   char *tmp;
   char buff[MMDAGENT_MAXBUFLEN];
   FILE *fp;

   if(m_jconf != NULL || m_recog != NULL || m_mmdagent == NULL || m_thread < 0 || m_languageModel == 0 || m_dictionary == 0 || m_acousticModel == 0 || m_triphoneList == 0 || m_configFile == 0)
      return;

   /* set latency */
   sprintf(buff, "PA_MIN_LATENCY_MSEC=%d", JULIUSTHREAD_LATENCY);
   putenv(buff);
   sprintf(buff, "LATENCY_MSEC=%d", JULIUSTHREAD_LATENCY);
   putenv(buff);

   /* turn off log */
   jlog_set_output(NULL);

   /* load models */
   tmp = MMDAgent_pathdup(m_languageModel);
   sprintf(buff, "-d \"%s\"", tmp);
   free(tmp);
   m_jconf = j_config_load_string_new(buff);
   if (m_jconf == NULL) {
      return;
   }

   tmp = MMDAgent_pathdup(m_dictionary);
   sprintf(buff, "-v \"%s\"", tmp);
   free(tmp);
   if(j_config_load_string(m_jconf, buff) < 0) {
      return;
   }

   tmp = MMDAgent_pathdup(m_acousticModel);
   sprintf(buff, "-h \"%s\"", tmp);
   free(tmp);
   if(j_config_load_string(m_jconf, buff) < 0) {
      return;
   }

   tmp = MMDAgent_pathdup(m_triphoneList);
   sprintf(buff, "-hlist \"%s\"", tmp);
   free(tmp);
   if(j_config_load_string(m_jconf, buff) < 0) {
      return;
   }

   /* load config file */
   tmp = MMDAgent_pathdup(m_configFile);
   if(j_config_load_file(m_jconf, tmp) < 0) {
      free(tmp);
      return;
   }
   free(tmp);

   /* load user dictionary */
   fp = MMDAgent_fopen(m_userDictionary, "r");
   if(fp != NULL) {
      fclose(fp);
      tmp = MMDAgent_pathdup(m_userDictionary);
      j_add_dict(m_jconf->lm_root, tmp);
      free(tmp);
   }

   /* create instance */
   m_recog = j_create_instance_from_jconf(m_jconf);
   if (m_recog == NULL) {
      return;
   }

   /* register callback functions */
   callback_add(m_recog, CALLBACK_EVENT_RECOGNITION_BEGIN, callbackRecogBegin, this);
   callback_add(m_recog, CALLBACK_RESULT, callbackRecogResult, this);
   if (!j_adin_init(m_recog)) {
      return;
   }

   if (j_open_stream(m_recog, NULL) != 0) {
      return;
   }

   /* setup logger */
   m_logger.setup(m_recog);

   /* start logger */
   m_logger.setActiveFlag(true);

   /* start recognize */
   j_recognize_stream(m_recog);
}