JULIUS(1)                                               JULIUS(1)



NAME
       Julius - open source multi-purpose LVCSR engine

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

DESCRIPTION
       Julius  is  a high-performance, multi-purpose, free speech
       recognition engine for researchers and developers.  It  is
       capable of performing almost real-time recognition of con-
       tinuous speech with over 60k-word vocabulary on most  cur-
       rent PCs.

       Julius needs an N-gram language model, word dictionary and
       an acoustic model  to  execute  a  recognition.   Standard
       model  formats  (i.e.  ARPA  and  HTK) with any word/phone
       units and sizes are supported, so users can build a recog-
       nition  system for various target using their own language
       model and acoustic models.  For details about basic models
       and their availability, please see the documents contained
       in this package.

       Julius can perform recognition on audio files, live micro-
       phone  input,  network  input and feature parameter files.
       The maximum size of vocabulary is 65,535 words.

RECOGNITION MODELS
       Julius supports the following models.

       Acoustic Models
                 Sub-word HMM (Hidden Markov Model) in HTK  ascii
                 format  are  supported.   Phoneme  models (mono-
                 phone), context dependent phoneme  models  (tri-
                 phone),  tied-mixture  and phonetic tied-mixture
                 models of any unit can be used.  When using con-
                 text 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
                 2-gram  and  reverse  3-gram language models are
                 used.  The Standard ARPA  format  is  supported.
                 In addition, a binary format N-gram is also sup-
                 ported for efficiency.  The tool mkbingram.  can
                 convert  binary  N-gram  from  the ARPA language
                 models.

SPEECH INPUT
       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 formats  will
       be  acceptable if compiled with libsndfile library).  Fea-
       ture parameter files in HTK format are also supported.

       Note that Julius itself can only extract MFCC_E_D_N_Z fea-
       tures  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  Julius  is based on a two-pass
       strategy.  Word 2-gram and reverse word 3-gram is used  on
       the  respective  passes.  The entire input is processed on
       the first pass, and again the final searching  process  is
       performed  again  for  the  input, using the result of the
       first pass to narrow the search space.  Specifically,  the
       recognition algorithm is based on a tree-trellis heuristic
       search combined with left-to-right frame-synchronous  beam
       search and right-to-left stack decoding 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.

   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  'adin-
              net'  means  receiving waveform data via tcpip net-
              work from an adinnet  client.  'netaudio'  is  from
              DatLink/NetAudio  input,  and  'stdin'  means  data
              input from standard input.

              WAV (no  compression)  and  RAW  (noheader,  16bit,
              BigEndian)  are  supported for waveform file input.
              Other  format  can  be  supported  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  waveform.   For speech file input, zero mean
              will be computed from the whole input.  For  micro-
              phone / 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 (-nozmeanframe))

       -nostrip
              Julius 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  recogni-
              tion.   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 speech segment in millisec-
              onds. (default: 300)

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

   Acoustic Analysis
       -smpFreq frequency
              Set sampling frequency of input speech in Hz.  Sam-
              pling rate can also  be  specified  using  "-smpPe-
              riod".   Be  careful  that this frequency should be
              the same as  the  trained  conditions  of  acoustic
              model you use.  This should be specified for micro-
              phone 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  sam-
              pling  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/Out-
              put" 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 computa-
              tion:  set  lower  frequency  cut-off.   Also   see
              "-hifreq".
              (default: -1 = disabled)

       -hifreq frequency
              Enable  band-limiting  for MFCC filterbank computa-
              tion:  set  upper  frequency  cut-off.   Also   see
              "-lofreq".
              (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 milliseconds (default: 300)

       -ssload filename
              Perform spectral subtraction for speech input using
              pre-estimated  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   for
              "-sscals"  and "-ssload".  Noise will be subtracted
              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 sub-
              traction will be substituted 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 con-
              currently with the 1st pass, and you can reject the
              input according  to  the  result  as  specified  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 Gaus-
              sians 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 perfor-
              mance.

       -gmmreject string
              Comma-separated list of GMM names to be rejected as
              invalid  input.   When recognition, the log likeli-
              hoods of GMMs accumulated 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 (word N-gram)
       -nlr 2gram_filename
              2-gram language model file in standard ARPA format.

       -nrl rev_3gram_filename
              Reverse   3-gram  language  model  file.   This  is
              required for the second search pass.   If  this  is
              not  defined  then  only  the  first pass will take
              place.

       -d bingram_filename
              Use binary format language model  instead  of  ARPA
              formats.   The  2-gram and 3-gram model can be com-
              bined and converted to  this  binary  format  using
              mkbingram.  Julius can read this format much faster
              than ARPA format.

       -lmp lm_weight lm_penalty

       -lmp2 lm_weight2 lm_penalty2
              Language model score  weights  and  word  insertion
              penalties  for  the first and second passes respec-
              tively.

              The hypothesis language scores are scaled as  shown
              below:

              lm_score1  =  lm_weight * 2-gram_score + lm_penalty
              lm_score2 = lm_weight2 * 3-gram_score + lm_penalty2

              The defaults are dependent on acoustic model:

                First-Pass | Second-Pass
               --------------------------
                5.0 -1.0   |  6.0  0.0 (monophone)
                8.0 -2.0   |  8.0 -2.0 (triphone,PTM)
                9.0  8.0   | 11.0 -2.0 (triphone,PTM, setup=v2.1)

       -transp float
              Additional insertion penalty for transparent words.
              (default: 0.0)

   Word Dictionary
       -v dictionary_file
              Word dictionary file (required).

       -silhead {WORD|WORD[OUTSYM]|#num}

       -siltail {WORD|WORD[OUTSYM]|#num}
              Sentence  start  and end silence word as defined in
              the dictionary.  (default: "<s>" / "</s>")

              Julius deal these words  as  fixed  start-word  and
              end-word  of  recognition.   They can be defined in
              several formats as shown below.

                                       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 automatically 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 phone
              sequence in the dictionary and the  HMM  definition
              names.

       -iwcd1 {best N|max|avg}
              When  using a triphone model, select method to han-
              dle 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 (default, N=3)
              max: use maximum likelihood of the same
                   context triphones
              avg: use average likelihood of the same
                   context triphones

       -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 con-
              tain  character  '+'  and  '-'.  You can explicitly
              specify by these options  to  avoid  mis-detection.
              These will override the automatic detection result.

       -notypecheck
              Disable checking of 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
       converted 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  Gaus-
              sians  to compute per mixture codebook. Small value
              will speed up  computation,  but  likelihood  error
              will grow larger. (default: 2)

       -gshmm hmmdefs
              Specify  monophone hmmdefs to use for Gaussian Mix-
              ture Selectio.  Monophone model for GMS  is  gener-
              ated  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
       -iwspword
              Add a word entry to the dictionary that should cor-
              respond to inter-word short pauses that  may  occur
              in  input  speech.   This  may  improve recognition
              accuracy in some language model that has no  inter-
              word  pause modeling.  The word entry can be speci-
              fied by "-iwspentry".

       -iwspentry
              Specify the  word  entry  that  will  be  added  by
              "-iwspword".  (default: "<UNK> [sp] sp sp")

       -iwsp  (Multi-path  version  only)  Enable inter-word con-
              text-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.

       -spmodel
              Specify short-pause model name that will be used in
              "-iwsp". (default: "sp")

   Short-pause Segmentation
       The  short  pause  segmentation  can be used for sucessive
       decoding of a long utterance.  Enabled when compiled  with
       '--enable-sp-segment'.

       -spdur Set the short-pause duration threshold in number of
              frames.  If a  short-pause  word  has  the  maximum
              likelihood  in  successive  frames longer than this
              value, then interrupt the first pass and start  the
              second pass. (default: 10)

   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 proportion to the
              width.

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

       -sepnum N
              Number of high frequency words to be separated from
              the lexicon tree. (default: 150)

       -1pass Only  perform  the first pass search.  This mode is
              automatically set when no 3-gram language model has
              been specified (-nlr).

       -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  calcu-
              lated  for each input independently, when the real-
              time option is ON the previous 5 second of input is
              always used.  Also refer to "-progout".

       -cmnsave filename
              Save last CMN parameters computed while recognition
              to the specified  file.   The  parameters  will  be
              saved  to  the  file in each time a input is recog-
              nized, 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 "-cmnsave".  Loading an initial CMN enables
              Julius to better recognize the first utterance on a
              microphone / network input.   Also  see  "-cmnnoup-
              date".

       -cmnmapweight
              Specify  weight  of  initial  cepstral  mean at the
              beginning of each utterance for microphone  /  net-
              work  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 sta-
              tus stacking 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 cor-
              rectly  found  increases   as   this   value   gets
              increased,  but  the  processing  time also becomes
              longer.

              Default value depends on the  engine setup on  com-
              pilation 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 measure.  (default: 0.05)

       -sb score
              Score  envelope  width for enveloped scoring.  When
              calculating hypothesis  score  for  each  generated
              hypothesis, its trellis expansion and viterbi oper-
              ation 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 discon-
              tinue  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)

       -graphrange nframe
              When graph output is  enabled  (--enable-graphout),
              merge  same  words  at  neighbor  position.  If the
              position of same words differs  smaller  than  this
              value,  they  will be merged.  The default is 0 (no
              merging) and specifying larger value will result in
              smaller graph output.

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

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

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

   Server Module Mode
       -module [port]
              Run Julius on "Server Module Mode".  After startup,
              Julius  waits  for  tcp/ip  connection from client.
              Once connection is established, Julius start commu-
              nication  with  the client to process incoming com-
              mands 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 N-gram 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  sym-
              bols and phone sequences as a recognition result to
              a client, specify "-outcode WP".

   Message Output
       -separatescore
              Output the language and acoustic scores separately.

       -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 mil-
              liseconds.

       -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 envi-
              ronment.

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

       -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 for
              recursive expansion.

       -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 dictio-
              nary 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  sec-
       tion in the Julius 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
       julian(1),  jcontrol(1),  adinrec(1),  adintool(1), mkbin-
       gram(1), mkbinhmm(1), mkgsmm(1), wav2mfcc(1), mkss(1)

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

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

BUGS
       There are some restrictions to the type and  size  of  the
       models  Julius  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) 1997-2000 Information-technology  Promotion
       Agency, Japan
       Copyright (c) 2000-2005 Shikano Lab., Nara Institute of
       Science and Technology
       Copyright (c) 2005-2006 Julius project team, Nagoya Institute
       of Technology

AUTHORS
       Rev.1.0 (1998/02/20)
              Designed by Tatsuya KAWAHARA and Akinobu LEE (Kyoto
              University)

              Development by Akinobu LEE (Kyoto University)

       Rev.1.1 (1998/04/14)

       Rev.1.2 (1998/10/31)

       Rev.2.0 (1999/02/20)

       Rev.2.1 (1999/04/20)

       Rev.2.2 (1999/10/04)

       Rev.3.0 (2000/02/14)

       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)
              Development  of  above  versions  by  Akinobu   LEE
              (Nagoya Institute of Technology)

THANKS TO
       From  rev.3.2, Julius is released by the "Information Pro-
       cessing Society, Continuous Speech Consortium".

       The Windows DLL version  was  developed  and  released  by
       Hideki BANNO (Nagoya University).

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



                              LOCAL                     JULIUS(1)
