JULIAN(1)                                                            JULIAN(1)



NAME
       Julian - grammar based continuous speech recognition parser

SYNOPSIS
       julian [-C jconffile] [options ...]

DESCRIPTION
       Julian  is  a  high-performance, multi-purpose, free speech recognition
       parser based on finite state grammar.   It  is  capable  of  performing
       real-time  recognition  of  continuous  speech  with  over thousands of
       vocabulary.

       Julian is a derived version of Julius, and almost  all  components  are
       the same except language model related part.

       To execute a recognition, it needs an acoustic model and a finite state
       grammar that describes sentence patterns to be recognized.  The grammar
       format  is  an  original one, and tools to create a recognirion grammar
       are included in the distribution.  For acoustic model, standard  format
       (i.e. HTK) with any word/phone units and sizes are supported.  So users
       can build a recognition system customized for specific tasks using  own
       task  grammar and acoustic models.  For details about models and how to
       write a grammar, please see the documents contained in this package.

       Julian can perform recognition on audio files, live  microphone  input,
       network input and feature parameter files.  The maximum size of vocabu-
       lary is 65,535 words.

RECOGNITION MODELS
       Julian supports the following models.

       Acoustic Models
                 Same as Julius: Sub-word HMM (Hidden  Markov  Model)  in  HTK
                 format  are  supported.   Phoneme models (monophone), context
                 dependent phoneme models (triphone),  tied-mixture  and  pho-
                 netic  tied-mixture  models  of  any  unit can be used.  When
                 using context dependent models,  interword  context  is  also
                 handled.   You can further use a tool mkbinhmm to convert the
                 ascii HMM definition file to binary format, for  speeding  up
                 the startup (this format is incompatible with that of HTK).

       Language model
                 The  grammar format is an original one, and tools to create a
                 recognirion grammar are  included  in  the  distribution.   A
                 grammar  consists  of two files: one is a 'grammar' file that
                 describes sentence structures in  a  BNF  style,  using  word
                 'categories'  as terminate symbols.  Another is a 'voca' file
                 that defines  word  with  its  pronunciations  (i.e.  phoneme
                 sequences)  for  each  category.  They should be converted by
                 mkdfa.pl(1) to a deterministic finite automaton  file  (.dfa)
                 and a dictionary file (.dict), respectively.

SPEECH INPUT
       Same  as  Julius: Both live speech input and recorded speech file input
       are supported.  Live  input  stream  from  microphone  device,  DatLink
       (NetAudio)  device and tcpip network input using adintool is supported.
       Speech waveform files (16bit WAV (no compression), RAW format, and many
       other  format  will be acceptable if compiled with libsndfile library).
       Feature parameter files in HTK format are also supported.

       Note that Julian itself can only  extract  MFCC_E_D_N_Z  features  from
       speech data.  If you use an acoustic HMM trained by other feature type,
       only the HTK parameter file of the same feature type can be used.

SEARCH ALGORITHM OVERVIEW
       Recognition algorithm of Julian is based on a  two-pass  strategy.   In
       the  first  pass,  a  high-speed  approximate search is performed using
       weaker constraints then the given grammar.  Here a LR beam search using
       only  inter-category  constraints  extracted  from  the grammar is per-
       formed. The second pass re-searches the input, using the original gram-
       mar  rules and intermediate results from the first pass, to gain a high
       precision result quickly.  In the second pass the optimal  solution  is
       theoretically guaranteed using the A* search.

       When using context dependent phones (triphones), interword contexts are
       taken into consideration.  For tied-mixture and  phonetic  tied-mixture
       models,  high-speed  acoustic  likelihood calculation is possible using
       gaussian pruning.

       For more details, see the related document or web page below.

OPTIONS
       The options below specify the  models,  system  behaviors  and  various
       search  parameters.  These option can be set all at once at the command
       line, but it is recommended that you write them in a  text  file  as  a
       "jconf file", and specify the file with "-C" option.

       Most are the same as Julius.
       Options  only  in Julian: -gram, -gramlist, -dfa, -penalty1, -penalty2,
       -looktrellis
       Options only in Julius: -nlr, -nrl, -d, -lmp, -lmp2, -transp, -silhead,
       -siltail, -spdur, -sepnum, -separatescore


   Speech Input
       -input {rawfile|mfcfile|mic|adinnet|netaudio|stdin}
              Select  speech  data  input source.  'rawfile' is waveform file,
              and specified after startup from stdin).  'mic' means microphone
              device,  and  'adinnet'  means receiving waveform data via tcpip
              network   from   an   adinnet   client.   'netaudio'   is   from
              DatLink/NetAudio  input, and 'stdin' means data input from stan-
              dard input.

              WAV (no compression) and RAW (noheader,  16bit,  BigEndian)  are
              supported  for  waveform  file  input.  Other format can be sup-
              ported using external library.  To see what format  is  actually
              supported, see the help message using option "-help".  For stdin
              input, only WAV and RAW is supported.
              (default: mfcfile)

       -filelist file
              (With -input rawfile|mfcfile) perform recognition on  all  files
              listed in the file.

       -adport portnum
              (with -input adinnet) adinnet port number (default: 5530)

       -NA server:unit
              (with  -input  netaudio)  set the server name and unit ID of the
              Datlink unit.

       -zmean  -nozmean
              This option enables/disables DC offset removal  of  input  wave-
              form.   For  speech  file input, zero mean will be computed from
              the whole input.  For microphone / network input, zero  mean  of
              the  first  48000  samples (3 seconds in 16kHz sampling) will be
              used at the rest.  (default: disabled (-nozmean))

       -zmeanframe  -nozmeanframe
              With speech input, this option  enables/disables  frame-wise  DC
              offset  removal.  This  is the same as HTK's ZMEANSOURCE option,
              and cannot be set with "-zmean".  (default: disabled  (-nozmean-
              frame))

       -nostrip
              Julian by default removes zero samples in input speech data.  In
              some cases, such invalid data may be recorded at  the  start  or
              end of recording.  This option inhibit this automatic removal.

       -record directory
              Auto-save  input  speech  data successively under the directory.
              Each segmented inputs are recorded to a file each by  one.   The
              file  name  of  the  recorded data is generated from system time
              when the input starts, in  a  style  of  "YYYY.MMDD.HHMMSS.wav".
              File  format  is  16bit monoral WAV.  Invalid for mfcfile input.
              With input rejection by "-rejectshort", the rejected input  will
              also be recorded even if they are rejected.

       -rejectshort msec
              Reject  input  shorter than specified milliseconds.  Search will
              be terminated and no result will be  output.   In  module  mode,
              '<REJECTED REASON="..."/>' message will be sent to client.  With
              "-record", the rejected input will also be recorded even if they
              are rejected.  (default: 0 = off)

   Speech Detection
       Options in this section is invalid for mfcfile input.

       -cutsilence

       -nocutsilence
              Force  silence  cutting  (=speech  segment detection) to ON/OFF.
              (default: ON for mic/adinnet, OFF for files)

       -lv threslevel
              Level threshold (0 - 32767) for  speech  triggering.   If  audio
              input  amplitude  goes  over this threshold for a period, Julius
              begin the 1st pass recognition.  If the level  goes  below  this
              level  after  triggering,  it  is the end of the speech segment.
              (default: 2000)

       -zc zerocrossnum
              Zero crossing threshold per a second (default: 60)

       -headmargin msec
              Margin at the start  of  the  speech  segment  in  milliseconds.
              (default: 300)

       -tailmargin msec
              Margin  at  the  end  of  the  speech  segment  in milliseconds.
              (default: 400)

   Acoustic Analysis
       -smpFreq frequency
              Set sampling frequency of input speech in Hz.  Sampling rate can
              also be specified using "-smpPeriod".  Be careful that this fre-
              quency should be the same as the trained conditions of  acoustic
              model  you  use.   This should be specified for microphone input
              and RAW file input when using other than default rate.  Also see
              "-fsize", "-fshift", "-delwin" and "-accwin".
              (default: 16000 (Hz = 625ns))

       -smpPeriod period
              Set  sampling  frequency  of input speech by its sampling period
              (nanoseconds).  The sampling rate can also  be  specified  using
              "-smpFreq".   Be  careful that the input frequency should be the
              same as the trained conditions of acoustic model you  use.  This
              should be specified for microphone input and RAW file input when
              using other than default rate.  Also  see  "-fsize",  "-fshift",
              "-delwin" and "-accwin".
              (default: 625 (ns = 16000Hz))

       -fsize sample
              Analysis window size in number of samples. (default: 400).

       -fshift sample
              Frame shift in number of samples (default: 160).

       -preemph value
              Pre-emphasis coefficient (default: 0.97)

       -fbank num
              Number of filterbank channels (default: 24)

       -ceplif num
              Cepstral liftering coefficient (default: 22)

       -rawe / -norawe
              Enable/disable  using  raw  energy before pre-emphasis (default:
              disabled)

       -enormal / -nornormal
              Enable/disable  normalizing  log  energy  (default:   disabled).
              Note:  normalising  log  energy  should not be specified on live
              input, at both training and recognition (see  sec.  5.9  "Direct
              Audio Input/Output" in HTKBook).

       -escale value
              Scaling  factor  of  log  energy  when  normalizing  log  energy
              (default: 1.0)

       -silfloor value
              Energy silence floor in dB when normalizing log energy (default:
              50.0)

       -delwin frame
              Delta window size in number of frames (default: 2).

       -accwin frame
              Acceleration window size in number of frames (default: 2).

       -lofreq frequency
              Enable  band-limiting for MFCC filterbank computation: set lower
              frequency cut-off.
              (default: -1 = disabled)

       -hifreq frequency
              Enable band-limiting for MFCC filterbank computation: set  upper
              frequency cut-off.
              (default: -1 = disabled)

       -sscalc
              Perform spectral subtraction using head part of each file.  With
              this option, Julius assume there are certain length  of  silence
              at  each  input  file.   Valid only for rawfile input.  Conflict
              with "-ssload".

       -sscalclen
              With "-sscalc", specify the length of head part silence in  mil-
              liseconds (default: 300)

       -ssload filename
              Perform  spectral  subtraction  for speech input using pre-esti-
              mated noise spectrum from file.  The noise spectrum data  should
              be  computed  beforehand  by  mkss.  Valid for all speech input.
              Conflict with "-sscalc".

       -ssalpha value
              Alpha coefficient of spectral subtraction.  Noise will  be  sub-
              tracted  stronger  as  this value gets larger, but distortion of
              the resulting signal also becomes remarkable.  (default: 2.0)

       -ssfloor value
              Flooring coefficient  of  spectral  subtraction.   The  spectral
              parameters  that go under zero after subtraction will be substi-
              tuted by the source signal  with  this  coefficient  multiplied.
              (default: 0.5)

   GMM-based Input Verification and Rejection
       -gmm filename
              GMM definition file in HTK format. If specified, GMM-based input
              verification will be performed concurrently with the  1st  pass,
              and  you  can reject the input according to the result as speci-
              fied by "-gmmreject".  Note that the GMM should  be  defined  as
              one-state  HMMs, and their training parameter should be the same
              as the acoustic model you want to use with.

       -gmmnum N
              Number of Gaussian components to be computed per  frame  on  GMM
              calculation.   Only  the  N-best  Gaussians will be computed for
              rapid calculation.  The default is  10  and  specifying  smaller
              value  will  speed up GMM calculation, but too small value (1 or
              2) may cause degradation of identification performance.

       -gmmreject string
              Comma-separated list of GMM names  to  be  rejected  as  invalid
              input.   When  recognition,  the log likelihoods of GMMs accumu-
              lated for the entire input will be  computed  concurrently  with
              the  1st  pass.   If the GMM name of the maximum score is within
              this string, the 2nd pass will not be  executed  and  the  input
              will be rejected.

   Language Model (Finite State Grammar)
       The  recognition  grammar  can  be  specified  in  three ways: "-gram",
       "-gramlist" or combination of "-dfa" and "-v".
       Multiple grammars can be specified by using  "-gram"  and  "-gramlist".
       When  you  use these options several times, all of them will be read at
       startup.  Note that this is a different  behavior  from  other  options
       (last  one override previous ones).  You can use "-nogram" to reset the
       already specified grammars at that point.

       -gram gramprefix1[,gramprefix2[,gramprefix3,...]]
              Comma-separated list of  grammars  to  be  used.   the  argument
              should  be  prefix  of a grammar, i.e. if you have "foo.dfa" and
              "foo.dict", you can specify them by single argument "foo".  Mul-
              tiple grammars can be specified as comma-separated list.

       -gramlist listfile
              Specify a grammar list file that contains list of grammars to be
              used.  The list file should contain  the  prefixs  of  grammars,
              each per line.  A relative path in the list file will be treated
              as relative to the list file, not the current path or configura-
              tion file.

       -dfa dfa_filename
              Finite state automaton grammar file.

       -v dictionary_file
              Word dictionary file (required)

       -nogram
              Remove  the  current  list  of grammars already specified by the
              options above.

       -penalty1 float
              Word insertion penalty for the first pass. (default: 0.0)

       -penalty2 float
              Word insertion penalty for the second pass. (default: 0.0)

       -spmodel {WORD|WORD[OUTSYM]|#num}
              Name of short pause model as defined in the hmmdefs.  In Julian,
              a  word  whose  pronunciation  consists of only this short pause
              model is called 'short pause word', and  handled  especially  in
              recognition:  even if its appearance in a sentence is explicitly
              specified in the grammar, it can be skipped while parsing.  This
              behavior  is  for  dealing  with insertion and deletion of short
              pause that often  appear  unintensionally  in  user  utterances.
              They can be specified in a style as shown below (default: "sp").


                                       Example
           Word_name                     <s>
           Word_name[output_symbol]   <s>[silB]
           #Word_ID                      #14

            (Word_ID is the word position in the dictionary
             file starting from 0)

       -forcedict
              Ignore dictionary errors and force running.  Words  with  errors
              will be dropped from dictionary at startup.

   Acoustic Model (HMM)
       -h hmmfilename
              HMM  definition file to use. Format (ascii/binary) will be auto-
              matically detected. (required)

       -hlist HMMlistfilename
              HMMList file to use.  Required when using triphone  based  HMMs.
              This file provides a mapping between the logical triphones names
              genertated from the phonetic representation  in  the  dictionary
              and the HMM definition names.

       -iwcd1 {best N|max|avg}
              When  using a triphone model, select method to handle inter-word
              triphone context on the first and last phone of a  word  in  the
              first pass.

              best N: use average likelihood of N-best scores from the same
                      context triphones
              max: use maximum likelihood of the same
                   context triphones
              avg: use average likelihood of the same
                   context triphones (default)

       -force_ccd / -no_ccd
              Normally  Julius determines whether the specified acoustic model
              is a context-dependent model from the model names, i.e., whether
              the  model names contain character '+' and '-'.  You can explic-
              itly specify by these options  to  avoid  mis-detection.   These
              will override the automatic detection result.

       -notypecheck
              Disable checking of the input parameter type. (default: enabled)

   Acoustic Computation
       Gaussian Pruning will be automatically enabled when using  tied-mixture
       based  acoutic  model.   It is disabled by default for non tied-mixture
       models, but you can activate pruning  to  those  models  by  explicitly
       specifying  "-gprune".  Gaussian Selection needs a monophone model con-
       verted by mkgshmm.

       -gprune {safe|heuristic|beam|none}
              Set the Gaussian pruning technique to use.
              (default: 'safe' (setup=standard), 'beam' (setup=fast) for  tied
              mixture model, 'none' for non tied-mixture model)

       -tmix K
              With  Gaussian  Pruning, specify the number of Gaussians to com-
              pute per mixture codebook. Small value will  speed  up  computa-
              tion, but likelihood error will grow larger. (default: 2)

       -gshmm hmmdefs
              Specify  monophone hmmdefs to use for Gaussian Mixture Selectio.
              Monophone model for GMS is generated from an ordinary  monophone
              HMM  model  using  mkgshmm.  This option is disabled by default.
              (no GMS applied)

       -gsnum N
              When using GMS, specify number of monophone state to select from
              whole monophone states. (default: 24)

   Inter-word Short Pause Handling
       -iwsp  (Multi-path  version  only) Enable inter-word context-free short
              pause handling.  This option appends  a  skippable  short  pause
              model  for  every  word end.  The added model will be skipped on
              inter-word context handling.  The HMM model to be  appended  can
              be specified by "-spmodel" option.

   Search Parameters (First Pass)
       -b beamwidth
              Beam  width (number of HMM nodes) on the first pass.  This value
              defines search width on the 1st pass, and has  great  effect  on
              the  total  processing  time.   Smaller  width will speed up the
              decoding, but too small  value  will  result  in  a  substantial
              increase  of  recognition  errors due to search failure.  Larger
              value will make the search stable and will lead to  failure-free
              search,  but  processing time and memory usage will grow in pro-
              portion to the width.

              default value: acoustic model dependent
                400 (monophone)
                800 (triphone,PTM)
               1000 (triphone,PTM, setup=v2.1)

       -1pass Only perform the first pass search.

       -realtime

       -norealtime
              Explicitly specify whether real-time (pipeline) processing  will
              be  done  in the first pass or not.  For file input, the default
              is OFF  (-norealtime),  for  microphone,  adinnet  and  NetAudio
              input,  the  default  is ON (-realtime).  This option relates to
              the way CMN is performed: when OFF, CMN is calculated  for  each
              input using cepstral mean of the whole input.  When the realtime
              option is ON, MAP-CMN will be performed.  When MAP-CMN, the cep-
              stral  mean  of  last 5 seconds are used as the initial cepstral
              mean at the beginning of each input.  Also refer to  "-progout".

       -cmnsave filename
              Save last CMN parameters computed while recognition to the spec-
              ified file.  The parameters will be saved to the  file  in  each
              time  a input is recognized, so the output file always keeps the
              last CMN parameters.  If output file already exist, it  will  be
              overridden.

       -cmnload filename
              Load initial CMN parameters previously saved in a file by "-cmn-
              save".   Loading  an  initial  CMN  enables  Julius  to   better
              recognize  the  first utterance on a microphone / network input.
              Also see "-cmnnoupdate".

       -cmnmapweight
              Specify weight of initial cepstral mean at the beginning of each
              utterance  for microphone / network input.  Specify larger value
              to retain the initial cepstral mean for  a  longer  period,  and
              smaller  value  to  rely  more  on the current input.  (default:
              100.0)

       -cmnnoupdate
              When microphone / network input, this option makes engine not to
              update  the  cepstral mean at each input and force engine to use
              the initial cepstral mean given by "-cmnload" parmanently.

   Search Parameters (Second Pass)
       -b2 hyponum
              Beam width (number of hypothesis) in second pass.  If the  count
              of word expantion at a certain length of hypothesis reaches this
              limit while search, shorter hypotheses are not expanded further.
              This prevents search to fall in breadth-first-like status stack-
              ing on the same position, and improve search failure.  (default:
              30)

       -n candidatenum
              The  search  continues  till 'candidate_num' sentence hypotheses
              have been found.  The obtained sentence hypotheses are sorted by
              score,  and final result is displayed in the order (see also the
              "-output" option).

              The possibility that the optimum hypothesis is  correctly  found
              increases  as this value gets increased, but the processing time
              also becomes longer.

              Default value depends on the  engine setup on compilation time:
                10  (standard)
                 1  (fast, v2.1)

       -output N
              The top N sentence hypothesis will  be  Output  at  the  end  of
              search.  Use with "-n" option. (default: 1)

       -cmalpha float
              This  parameter decides smoothing effect of word confidence mea-
              sure.  (default: 0.05)

       -sb score
              Score envelope width for enveloped  scoring.   When  calculating
              hypothesis  score  for  each  generated  hypothesis, its trellis
              expansion and viterbi operation will be pruned in the middle  of
              the speech if score on a frame goes under [current maximum score
              of the frame- width].  Giving small value makes the second  pass
              faster, but computation error may occur.  (default: 80.0)

       -s stack_size
              The maximum number of hypothesis that can be stored on the stack
              during the search.  A larger value may give more stable results,
              but increases the amount of memory required. (default: 500)

       -m overflow_pop_times
              Number  of  expanded  hypotheses  required  to  discontinue  the
              search.  If the number of expanded hypotheses  is  greater  then
              this  threshold  then, the search is discontinued at that point.
              The larger this value is, The longer  Julius  gets  to  give  up
              search (default: 2000)

       -lookuprange nframe
              When  performing  word expansion on the second pass, this option
              sets the number of frames before and after to look up next  word
              hypotheses  in  the word trellis.  This prevents the omission of
              short words, but with a large  value,  the  number  of  expanded
              hypotheses increases and system becomes slow. (default: 5)

       -looktrellis
              Expand  only  the  words  survived  on the first pass instead of
              expanding all the words predicted by grammar.  This option makes
              second pass decoding slightly faster especially for large vocab-
              ulary condition, but may increase deletion error of short words.
              (default: disabled)

   Graph Output
       -graphrange nframe
              When  graph  output  is  enabled (--enable-graphout), merge same
              words at neighbor position.  If the position of same words  dif-
              fers  smaller than this value, they will be merged.  The default
              is 0 (allow merging on exactly the same location) and specifying
              larger value will result in smaller graph output.  Setting to -1
              will disable merging, in that case same words on the same  loca-
              tion of different scores will be left as they are. (default: 0)

       -graphcut depth
              Cut  the  resulting  graph  by its word depth at post-processing
              stage.  The depth value is the number of words to be allowed  at
              a frame.  Setting to -1 disables this feature. (default: 80)

       -graphboundloop num
              Limit  the number of boundary adjustment loop at post-processing
              stage. This parameter prevents Julius from blocking by  infinite
              adjustment loop by short word oscillation.  (default: 20)

       -graphsearchdelay

       -nographsearchdelay
              When  "-graphsearchdelay"  option  is  set,  Julius modifies its
              graph generation alogrithm on the  2nd  pass  not  to  terminate
              search  by  graph merging, until the first sentence candidate is
              found.  This option may improve graph accuracy, especially  when
              you  are  going  to  generate a huge word graph by setting broad
              search.  Namely, it may result in better graph accuracy when you
              set  wide  beams  on  both 1st pass "-b" and 2nd pass "-b2", and
              large number for "-n".  (default: disabled)

   Forced Alignment
       -walign
              Do viterbi alignment per word units from the recognition result.
              The  word  boundary  frames  and the average acoustic scores per
              frame are calculated.

       -palign
              Do viterbi alignment per phoneme (model) units from the recogni-
              tion result.  The phoneme boundary frames and the average acous-
              tic scores per frame are calculated.

       -salign
              Do viterbi alignment per HMM state from the recognition  result.
              The  state  boundary  frames and the average acoustic scores per
              frame are calculated.

   Server Module Mode
       -module [port]
              Run Julian on "Server Module Mode".  After startup, Julian waits
              for  tcp/ip  connection  from client.  Once connection is estab-
              lished, Julian start communication with the  client  to  process
              incoming  commands  from  the  client,  or to output recognition
              results, input trigger information and other  system  status  to
              the  client.   The  multi-grammar mode is only supported at this
              Server Module Mode.  The default port number is 10500.  jcontrol
              is sample client contained in this package.

       -outcode [W][L][P][S][C][w][l][p][s]
              (Only for Server Module Mode) Switch which symbols of recognized
              words to be sent to client.  Specify 'W' for output symbol,  'L'
              for  grammar entry, 'P' for phoneme sequence, 'S' for score, and
              'C' for confidence score, respectively.  Capital letters are for
              the  second  pass  (final  result),  and  small  letters are for
              results of the first pass.  For example, if  you  want  to  send
              only  the  output  symbols  and phone sequences as a recognition
              result to a client, specify "-outcode WP".

   Message Output
       -multigramout
              Enable multiple grammar output.  Usually, Julian will search for
              the  best  hypothesis  among  the  grammars.   This options will
              change the search to find the best result one by  one  for  each
              grammar.

       -quiet Omit  phoneme  sequence  and  score,  only  output the best word
              sequence hypothesis.

       -progout
              Enable progressive output of the partial results  on  the  first
              pass.

       -proginterval msec
              set the output time interval of "-progout" in milliseconds.

       -demo  Equivalent to "-progout -quiet"

       -charconv from to
              Enable  output  character  set  conversion. "from" is the source
              character set used in the language model, and "to" is the target
              character set you want to get.
              On  Linux,  the arguments should be a code name.  You can obtain
              the list of available code names by invoking the command  "iconv
              --list".   On  Windows,  the  arguments should be a code name or
              codepage number.  Code name should  be  one  of  "ansi",  "mac",
              "oem",  "utf-7", "utf-8", "sjis", "euc".  Or you can specify any
              codepage number supported at your environment.

   OTHERS
       -debug (For debug) output enoumous internal status and  debug  informa-
              tion.

       -C jconffile
              Load  the  jconf  file.   The  options  written  in the file are
              included and expanded at the point.  This  option  can  also  be
              used within other jconf file.

       -check wchmm
              (For  debug)  turn  on  interactive  check  mode of tree lexicon
              structure at startup.

       -check triphone
              (For debug) turn on interactive  check  mode  of  model  mapping
              between Acoustic model, HMMList and dictionary at startup.

       -setting
              Display compile-time engine configuration and exit.

       -help  Display a brief description of all options.

EXAMPLES
       For  examples  of  system  usage,  refer to the tutorial section in the
       Julian documents.

NOTICE
       Note about jconf files: relative paths in a jconf file are  interpreted
       as relative to the jconf file itself, not to the current directory.

SEE ALSO
       julius(1),  jcontrol(1), adinrec(1), adintool(1), mkdfa(1), mkbinhmm(1)
       mkgsmm(1), wav2mfcc(1), mkss(1)

       http://julius.sourceforge.jp/en/

DIAGNOSTICS
       Julian normally will return the exit status 0.   If  an  error  occurs,
       Julian exits abnormally with exit status 1.  If an input file cannot be
       found or cannot be loaded for some reason then Julian  will  skip  pro-
       cessing for that file.

BUGS
       There  are  some restrictions to the type and size of the models Julian
       can use.  For a detailed explanation refer to the Julius documentation.
       For    bug-reports,    inquires    and    comments    please    contact
       julius@kuis.kyoto-u.ac.jp or julius@is.aist-nara.ac.jp.

COPYRIGHT
       Copyright (c) 1991-2006 Kawahara Lab., Kyoto University
       Copyright (c) 2000-2005 Shikano Lab., Nara  Institute  of  Science  and
       Technology
       Copyright  (c) 2005-2006 Julius project team, Nagoya Institute of Tech-
       nology

AUTHORS
       Rev.1.0 (1998/07/20)
              Designed by Tatsuya KAWAHARA and Akinobu LEE (Kyoto University)

       Rev.2.0 (1999/02/20)

       Rev.2.1 (1999/04/20)

       Rev.2.2 (1999/10/04)

       Rev.3.1 (2000/05/11)
              Development of above versions by Akinobu LEE (Kyoto University)

       Rev.3.2 (2001/08/15)

       Rev.3.3 (2002/09/11)

       Rev.3.4 (2003/10/01)

       Rev.3.4.1 (2004/02/25)

       Rev.3.4.2 (2004/04/30)
              Development of above versions by Akinobu LEE (Nara Institute  of
              Science and Technology)

       Rev.3.5 (2005/11/11)

       Rev.3.5.1 (2006/03/31)

       Rev.3.5.2 (2006/07/31)
              Development  of  above versions by Akinobu LEE (Nagoya Institute
              of Technology)

THANKS TO
       From rev.3.2, Julian is released to the member of the "Information Pro-
       cessing  Society,  Continuous Speech Consortium".  From rev.3.4, Julian
       becomes an open-source products incorporated with Julius.

       The Windows Microsoft Speech API compatible version  was  developed  by
       Takashi SUMIYOSHI (Kyoto University).



4.3 Berkeley Distribution            LOCAL                           JULIAN(1)
