Hello, i dont understand why the pocketsphinx demo doesn’t work… (https://github.com/matrix-io/matrix-creator-pocketsphinx)
when i want to test the : pocketsphinx_demo i have this on my terminal :
Arguments list definition: [NAME] [DEFLT] [DESCR] -adcdev Name of audio device to use for input. -agc none Automatic gain control for c0 ('max', 'emax', 'noise', or 'none') -agcthresh 2.0 Initial threshold for automatic gain control -allphone Perform phoneme decoding with phonetic lm -allphone_ci yes Perform phoneme decoding with phonetic lm and context-independent units only -alpha 0.97 Preemphasis parameter -argfile Argument file giving extra arguments. -ascale 20.0 Inverse of acoustic model scale for confidence score calculation -aw 1 Inverse weight applied to acoustic scores. -backtrace no Print results and backtraces to log. -beam 1e-48 Beam width applied to every frame in Viterbi search (smaller values mean wider beam) -bestpath yes Run bestpath (Dijkstra) search over word lattice (3rd pass) -bestpathlw 9.5 Language model probability weight for bestpath search -ceplen 13 Number of components in the input feature vector -cmn live Cepstral mean normalization scheme ('live', 'batch', or 'none') -cmninit 40,3,-1 Initial values (comma-separated) for cepstral mean when 'live' is used -compallsen no Compute all senone scores in every frame (can be faster when there are many senones) -dict Main pronunciation dictionary (lexicon) input file -dictcase no Dictionary is case sensitive (NOTE: case insensitivity applies to ASCII characters only) -dither no Add 1/2-bit noise -doublebw no Use double bandwidth filters (same center freq) -ds 1 Frame GMM computation downsampling ratio -fdict Noise word pronunciation dictionary input file -feat 1s_c_d_dd Feature stream type, depends on the acoustic model -featparams File containing feature extraction parameters. -fillprob 1e-8 Filler word transition probability -frate 100 Frame rate -fsg Sphinx format finite state grammar file -fsgusealtpron yes Add alternate pronunciations to FSG -fsgusefiller yes Insert filler words at each state. -fwdflat yes Run forward flat-lexicon search over word lattice (2nd pass) -fwdflatbeam 1e-64 Beam width applied to every frame in second-pass flat search -fwdflatefwid 4 Minimum number of end frames for a word to be searched in fwdflat search -fwdflatlw 8.5 Language model probability weight for flat lexicon (2nd pass) decoding -fwdflatsfwin 25 Window of frames in lattice to search for successor words in fwdflat search -fwdflatwbeam 7e-29 Beam width applied to word exits in second-pass flat search -fwdtree yes Run forward lexicon-tree search (1st pass) -hmm Directory containing acoustic model files. -infile Audio file to transcribe. -inmic no Transcribe audio from microphone. -input_endian little Endianness of input data, big or little, ignored if NIST or MS Wav -jsgf JSGF grammar file -keyphrase Keyphrase to spot -kws A file with keyphrases to spot, one per line -kws_delay 10 Delay to wait for best detection score -kws_plp 1e-1 Phone loop probability for keyphrase spotting -kws_threshold 1e-30 Threshold for p(hyp)/p(alternatives) ratio -latsize 5000 Initial backpointer table size -lda File containing transformation matrix to be applied to features (single-stream features only) -ldadim 0 Dimensionality of output of feature transformation (0 to use entire matrix) -lifter 0 Length of sin-curve for liftering, or 0 for no liftering. -lm Word trigram language model input file -lmctl Specify a set of language model -lmname Which language model in -lmctl to use by default -logbase 1.0001 Base in which all log-likelihoods calculated -logfn File to write log messages in -logspec no Write out logspectral files instead of cepstra -lowerf 133.33334 Lower edge of filters -lpbeam 1e-40 Beam width applied to last phone in words -lponlybeam 7e-29 Beam width applied to last phone in single-phone words -lw 6.5 Language model probability weight -maxhmmpf 30000 Maximum number of active HMMs to maintain at each frame (or -1 for no pruning) -maxwpf -1 Maximum number of distinct word exits at each frame (or -1 for no pruning) -mdef Model definition input file -mean Mixture gaussian means input file -mfclogdir Directory to log feature files to -min_endfr 0 Nodes ignored in lattice construction if they persist for fewer than N frames -mixw Senone mixture weights input file (uncompressed) -mixwfloor 0.0000001 Senone mixture weights floor (applied to data from -mixw file) -mllr MLLR transformation to apply to means and variances -mmap yes Use memory-mapped I/O (if possible) for model files -ncep 13 Number of cep coefficients -nfft 512 Size of FFT -nfilt 40 Number of filter banks -nwpen 1.0 New word transition penalty -pbeam 1e-48 Beam width applied to phone transitions -pip 1.0 Phone insertion penalty -pl_beam 1e-10 Beam width applied to phone loop search for lookahead -pl_pbeam 1e-10 Beam width applied to phone loop transitions for lookahead -pl_pip 1.0 Phone insertion penalty for phone loop -pl_weight 3.0 Weight for phoneme lookahead penalties -pl_window 5 Phoneme lookahead window size, in frames -rawlogdir Directory to log raw audio files to -remove_dc no Remove DC offset from each frame -remove_noise yes Remove noise with spectral subtraction in mel-energies -remove_silence yes Enables VAD, removes silence frames from processing -round_filters yes Round mel filter frequencies to DFT points -samprate 16000 Sampling rate -seed -1 Seed for random number generator; if less than zero, pick our own -sendump Senone dump (compressed mixture weights) input file -senlogdir Directory to log senone score files to -senmgau Senone to codebook mapping input file (usually not needed) -silprob 0.005 Silence word transition probability -smoothspec no Write out cepstral-smoothed logspectral files -svspec Subvector specification (e.g., 24,0-11/25,12-23/26-38 or 0-12/13-25/26-38) -time no Print word times in file transcription. -tmat HMM state transition matrix input file -tmatfloor 0.0001 HMM state transition probability floor (applied to -tmat file) -topn 4 Maximum number of top Gaussians to use in scoring. -topn_beam 0 Beam width used to determine top-N Gaussians (or a list, per-feature) -toprule Start rule for JSGF (first public rule is default) -transform legacy Which type of transform to use to calculate cepstra (legacy, dct, or htk) -unit_area yes Normalize mel filters to unit area -upperf 6855.4976 Upper edge of filters -uw 1.0 Unigram weight -vad_postspeech 50 Num of silence frames to keep after from speech to silence. -vad_prespeech 20 Num of speech frames to keep before silence to speech. -vad_startspeech 10 Num of speech frames to trigger vad from silence to speech. -vad_threshold 3.0 Threshold for decision between noise and silence frames. Log-ratio between signal level and noise level. -var Mixture gaussian variances input file -varfloor 0.0001 Mixture gaussian variance floor (applied to data from -var file) -varnorm no Variance normalize each utterance (only if CMN == current) -verbose no Show input filenames -warp_params Parameters defining the warping function -warp_type inverse_linear Warping function type (or shape) -wbeam 7e-29 Beam width applied to word exits -wip 0.65 Word insertion penalty -wlen 0.025625 Hamming window length INFO: pocketsphinx_demo.cpp(197): Specify '-infile <file.wav>' to recognize from file or '-inmic yes' to recognize from microphone.