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POCKETSPHINX(1) 	     General Commands Manual		 POCKETSPHINX(1)

NAME
     pocketsphinx - Run speech recognition on audio data

SYNOPSIS
     pocketsphinx [ options... ] [ live | single | help | soxflags ] INPUTS...

DESCRIPTION
     The apocketsphinxa command-line program reads single-channel 16-bit PCM au-
     dio  one  or more input files (or a-a to read from standard input), and at-
     tempts to recognize speech in it using the default  acoustic  and	language
     model.  The input files can be raw audio, WAV, or NIST Sphere files, though
     some of these may not be recognized properly.  It accepts a large number of
     options which you probably don't care about, and a command  which	defaults
     to alivea. The commands are as follows:

     help   Print a long list of those options you don't care about.

     config
	    Dump  configuration  as  JSON to standard output (can be loaded with
	    the a-configa option).

     live   Detect speech segments in input files, run recognition on them  (us-
	    ing  those	options  you don't care about), and write the results to
	    standard output in line-delimited JSON. I  realize	this  isn't  the
	    prettiest  format,	but it sure beats XML. Each line contains a JSON
	    object with these fields, which have short names to make  the  lines
	    more readable:

	    "b": Start time in seconds, from the beginning of the stream

	    "d": Duration in seconds

	    "p":  Estimated probability of the recognition result, i.e. a number
	    between 0 and 1 which may be used as a confidence score

	    "t": Full text of recognition result

	    "w": List of segments (usually words), each of which  in  turn  con-
	    tains  the	aba, ada, apa, and ata fields, for start, end, probabil-
	    ity, and the text of the word. In the future we may also support hi-
	    erarchical results in which case awa could be present.

     single
	    Recognize the input as a single utterance, and write a  JSON  object
	    in the same format described above.

     align

	    Align  a single input file (or '-' for standard input) to a word se-
	    quence, and write a JSON object in the same format described  above.
	    The  first positional argument is the input, and all subsequent ones
	    are concatenated to make the text, to avoid surprises if you  forget
	    to quote it.  You are responsible for normalizing the text to remove
	    punctuation, uppercase, centipedes, etc. For example:

		pocketsphinx align goforward.wav "go forward ten meters"

	    By	default, only word-level alignment is done.  To get phone align-
	    ments, pass `-phone_align yes` in the flags, e.g.:

		pocketsphinx -phone_align yes align audio.wav $text

	    To get state-level alignments, pass `-state_align yes`  (this  auto-
	    matically enables phone alignment as well):

		pocketsphinx -state_align yes align audio.wav $text

	    This  will make not particularly readable output, but you can use jq
	    (https://stedolan.github.io/jq/) to clean it up.  For  example,  you
	    can get just the word names and start times like this:

		pocketsphinx align audio.wav $text | jq '.w[]|[.t,.b]'

	    Or you could get the phone names and durations like this:

		pocketsphinx -phone_align yes align audio.wav $text | jq '.w[]|.w[]|[.t,.d]'

	    There are many, many other possibilities, of course.

     help   Print  a usage and help text with a list of possible arguments.  You
	    can also get command-specific help by  specifying  a  command  after
	    help, for example 'pocketsphinx help align' will show alignment-spe-
	    cific options.

     help-config
	    Print  a  long  list  of all configuration parameters with their de-
	    scriptions.

     soxflags
	    Return arguments to asoxa which will create  the  appropriate  input
	    format.  Note that because the asoxa command-line is slightly quirky
	    these must always come after the filename or a-da (which tells asoxa
	    to read from the microphone). You  can  run  live  recognition  like
	    this:

		sox -d $(pocketsphinx soxflags) | pocketsphinx -

	    or decode from a file named "audio.mp3" like this:

		sox audio.mp3 $(pocketsphinx soxflags) | pocketsphinx -

     By  default only errors are printed to standard error, but if you want more
     information you can pass a-loglevel INFOa. Partial results are not printed,
     maybe they will be in the future, but don't hold your breath.  Force-align-
     ment is likely to be supported soon, however.

OPTIONS
     -agc   Automatic gain control for c0 ('max', 'emax', 'noise', or 'none')

     -agcthresh
	    Initial threshold for automatic gain control

     -allphone
	    phoneme decoding with phonetic lm (given here)

     -allphone_ci
	    Perform  phoneme  decoding	with phonetic lm and context-independent
	    units only

     -alpha
	    Preemphasis parameter

     -ascale
	    Inverse of acoustic model scale for confidence score calculation

     -aw    Inverse weight applied to acoustic scores.

     -backtrace
	    Print results and backtraces to log.

     -beam  Beam width applied to every frame in Viterbi search (smaller  values
	    mean wider beam)

     -bestpath
	    Run bestpath (Dijkstra) search over word lattice (3rd pass)

     -bestpathlw
	    Language model probability weight for bestpath search

     -ceplen
	    Number of components in the input feature vector

     -cmn   Cepstral mean normalization scheme ('live', 'batch', or 'none')

     -cmninit
	    Initial  values  (comma-separated)	for cepstral mean when 'live' is
	    used

     -compallsen
	    Compute all senone scores in every frame (can be faster  when  there
	    are many senones)

     -dict  pronunciation dictionary (lexicon) input file

     -dictcase
	    Dictionary	is  case  sensitive (NOTE: case insensitivity applies to
	    ASCII characters only)

     -dither
	    Add 1/2-bit noise

     -doublebw
	    Use double bandwidth filters (same center freq)

     -ds    Frame GMM computation downsampling ratio

     -fdict
	    word pronunciation dictionary input file

     -feat  Feature stream type, depends on the acoustic model

     -featparams
	    containing feature extraction parameters.

     -fillprob
	    Filler word transition probability

     -frate
	    Frame rate

     -fsg   format finite state grammar file

     -fsgusealtpron
	    Add alternate pronunciations to FSG

     -fsgusefiller
	    Insert filler words at each state.

     -fwdflat
	    Run forward flat-lexicon search over word lattice (2nd pass)

     -fwdflatbeam
	    Beam width applied to every frame in second-pass flat search

     -fwdflatefwid
	    Minimum number of end frames for a word to be  searched  in  fwdflat
	    search

     -fwdflatlw
	    Language model probability weight for flat lexicon (2nd pass) decod-
	    ing

     -fwdflatsfwin
	    Window of frames in lattice to search for successor words in fwdflat
	    search

     -fwdflatwbeam
	    Beam width applied to word exits in second-pass flat search

     -fwdtree
	    Run forward lexicon-tree search (1st pass)

     -hmm   containing acoustic model files.

     -input_endian
	    Endianness of input data, big or little, ignored if NIST or MS Wav

     -jsgf  grammar file

     -keyphrase
	    to spot

     -kws   file with keyphrases to spot, one per line

     -kws_delay
	    Delay to wait for best detection score

     -kws_plp
	    Phone loop probability for keyphrase spotting

     -kws_threshold
	    Threshold for p(hyp)/p(alternatives) ratio

     -latsize
	    Initial backpointer table size

     -lda   containing	transformation matrix to be applied to features (single-
	    stream features only)

     -ldadim
	    Dimensionality of output of feature transformation (0 to use  entire
	    matrix)

     -lifter
	    Length of sin-curve for liftering, or 0 for no liftering.

     -lm    trigram language model input file

     -lmctl
	    a set of language model

     -lmname
	    language model in -lmctl to use by default

     -logbase
	    Base in which all log-likelihoods calculated

     -logfn
	    to write log messages in

     -loglevel
	    Minimum level of log messages (DEBUG, INFO, WARN, ERROR)

     -logspec
	    Write out logspectral files instead of cepstra

     -lowerf
	    Lower edge of filters

     -lpbeam
	    Beam width applied to last phone in words

     -lponlybeam
	    Beam width applied to last phone in single-phone words

     -lw    Language model probability weight

     -maxhmmpf
	    Maximum  number  of active HMMs to maintain at each frame (or -1 for
	    no pruning)

     -maxwpf
	    Maximum number of distinct word exits at each frame (or  -1  for  no
	    pruning)

     -mdef  definition input file

     -mean  gaussian means input file

     -mfclogdir
	    to log feature files to

     -min_endfr
	    Nodes ignored in lattice construction if they persist for fewer than
	    N frames

     -mixw  mixture weights input file (uncompressed)

     -mixwfloor
	    Senone mixture weights floor (applied to data from -mixw file)

     -mllr  transformation to apply to means and variances

     -mmap  Use memory-mapped I/O (if possible) for model files

     -ncep  Number of cep coefficients

     -nfft  Size of FFT, or 0 to set automatically (recommended)

     -nfilt
	    Number of filter banks

     -nwpen
	    New word transition penalty

     -pbeam
	    Beam width applied to phone transitions

     -pip   Phone insertion penalty

     -pl_beam
	    Beam width applied to phone loop search for lookahead

     -pl_pbeam
	    Beam width applied to phone loop transitions for lookahead

     -pl_pip
	    Phone insertion penalty for phone loop

     -pl_weight
	    Weight for phoneme lookahead penalties

     -pl_window
	    Phoneme lookahead window size, in frames

     -rawlogdir
	    to log raw audio files to

     -remove_dc
	    Remove DC offset from each frame

     -remove_noise
	    Remove noise using spectral subtraction

     -round_filters
	    Round mel filter frequencies to DFT points

     -samprate
	    Sampling rate

     -seed  Seed for random number generator; if less than zero, pick our own

     -sendump
	    dump (compressed mixture weights) input file

     -senlogdir
	    to log senone score files to

     -senmgau
	    to codebook mapping input file (usually not needed)

     -silprob
	    Silence word transition probability

     -smoothspec
	    Write out cepstral-smoothed logspectral files

     -svspec
	    specification (e.g., 24,0-11/25,12-23/26-38 or 0-12/13-25/26-38)

     -tmat  state transition matrix input file

     -tmatfloor
	    HMM state transition probability floor (applied to -tmat file)

     -topn  Maximum number of top Gaussians to use in scoring.

     -topn_beam
	    Beam  width  used  to determine top-N Gaussians (or a list, per-fea-
	    ture)

     -toprule
	    rule for JSGF (first public rule is default)

     -transform
	    Which type of transform to use to calculate cepstra (legacy, dct, or
	    htk)

     -unit_area
	    Normalize mel filters to unit area

     -upperf
	    Upper edge of filters

     -uw    Unigram weight

     -var   gaussian variances input file

     -varfloor
	    Mixture gaussian variance floor (applied to data from -var file)

     -varnorm
	    Variance normalize each utterance (only if CMN == current)

     -verbose
	    Show input filenames

     -warp_params
	    defining the warping function

     -warp_type
	    Warping function type (or shape)

     -wbeam
	    Beam width applied to word exits

     -wip   Word insertion penalty

     -wlen  Hamming window length

AUTHOR
     Written by numerous people at CMU from 1994 onwards.  This manual	page  by
     David Huggins-Daines <dhdaines@gmail.com>

COPYRIGHT
     Copyright	(C)  1994-2016 Carnegie Mellon University.  See the file LICENSE
     included with this package for more information.

SEE ALSO
     pocketsphinx_batch(1), sphinx_fe(1).

				   2022-09-27			 POCKETSPHINX(1)

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