ldd /usr/local/bin/pocketsphinx_continuous

Installation
Clone this repo into ~/MagicMirror/modules directory.

Run command bash dependencies.sh in ~/MagicMirror/modules/MMM-voice/installers directory, to install all dependencies. This will need a couple of minutes.

tsphinx_continuous COMPILED ON: Sep 29 2019, AT: 15:24:41
Error opening audio device plughw:1,0 for capture: Invalid argument
FATAL: “continuous.c”, line 245: Failed to open audio device

I don’t know…

i would open an issue on the MMM-voice github repo, with the log file and ask for help .

INFO: feat.c(715): Initializing feature stream to type: ‘1s_c_d_dd’, ceplen=13, CMN=‘batch’, VARNORM=‘no’, AGC=‘none’
INFO: acmod.c(162): Using subvector specification 0-12/13-25/26-38
INFO: mdef.c(518): Reading model definition: /usr/local/share/pocketsphinx/model/en-us/en-us/mdef
INFO: mdef.c(531): Found byte-order mark BMDF, assuming this is a binary mdef file
INFO: bin_mdef.c(336): Reading binary model definition: /usr/local/share/pocketsphinx/model/en-us/en-us/mdef
INFO: bin_mdef.c(516): 42 CI-phone, 137053 CD-phone, 3 emitstate/phone, 126 CI-sen, 5126 Sen, 29324 Sen-Seq
INFO: tmat.c(149): Reading HMM transition probability matrices: /usr/local/share/pocketsphinx/model/en-us/en-us/transition_matrices
INFO: acmod.c(113): Attempting to use PTM computation module
INFO: ms_gauden.c(127): Reading mixture gaussian parameter: /usr/local/share/pocketsphinx/model/en-us/en-us/means
INFO: ms_gauden.c(242): 42 codebook, 3 feature, size:
INFO: ms_gauden.c(244): 128x13
INFO: ms_gauden.c(244): 128x13
INFO: ms_gauden.c(244): 128x13
INFO: ms_gauden.c(127): Reading mixture gaussian parameter: /usr/local/share/pocketsphinx/model/en-us/en-us/variances
INFO: ms_gauden.c(242): 42 codebook, 3 feature, size:
INFO: ms_gauden.c(244): 128x13
INFO: ms_gauden.c(244): 128x13
INFO: ms_gauden.c(244): 128x13
INFO: ms_gauden.c(304): 222 variance values floored
INFO: ptm_mgau.c(475): Loading senones from dump file /usr/local/share/pocketsphinx/model/en-us/en-us/sendump
INFO: ptm_mgau.c(499): BEGIN FILE FORMAT DESCRIPTION
INFO: ptm_mgau.c(562): Rows: 128, Columns: 5126
INFO: ptm_mgau.c(594): Using memory-mapped I/O for senones
INFO: ptm_mgau.c(837): Maximum top-N: 4
INFO: phone_loop_search.c(114): State beam -225 Phone exit beam -225 Insertion penalty 0
INFO: dict.c(320): Allocating 4130 * 20 bytes (80 KiB) for word entries
INFO: dict.c(333): Reading main dictionary: /home/pi/MagicMirror/modules/MMM-voice/MMM-voice.dic
INFO: dict.c(213): Dictionary size 29, allocated 0 KiB for strings, 0 KiB for phones
INFO: dict.c(336): 29 words read
INFO: dict.c(358): Reading filler dictionary: /usr/local/share/pocketsphinx/model/en-us/en-us/noisedict
INFO: dict.c(213): Dictionary size 34, allocated 0 KiB for strings, 0 KiB for phones
INFO: dict.c(361): 5 words read
INFO: dict2pid.c(396): Building PID tables for dictionary
INFO: dict2pid.c(406): Allocating 42^3 * 2 bytes (144 KiB) for word-initial triphones
INFO: dict2pid.c(132): Allocated 21336 bytes (20 KiB) for word-final triphones
INFO: dict2pid.c(196): Allocated 21336 bytes (20 KiB) for single-phone word triphones
INFO: ngram_model_trie.c(354): Trying to read LM in trie binary format
INFO: ngram_model_trie.c(365): Header doesn’t match
INFO: ngram_model_trie.c(177): Trying to read LM in arpa format
INFO: ngram_model_trie.c(193): LM of order 3
INFO: ngram_model_trie.c(195): #1-grams: 25
INFO: ngram_model_trie.c(195): #2-grams: 46
INFO: ngram_model_trie.c(195): #3-grams: 23
INFO: lm_trie.c(474): Training quantizer
INFO: lm_trie.c(482): Building LM trie
INFO: ngram_search_fwdtree.c(74): Initializing search tree
INFO: ngram_search_fwdtree.c(101): 28 unique initial diphones
INFO: ngram_search_fwdtree.c(186): Creating search channels
INFO: ngram_search_fwdtree.c(323): Max nonroot chan increased to 157
INFO: ngram_search_fwdtree.c(333): Created 28 root, 29 non-root channels, 5 single-phone words
INFO: ngram_search_fwdflat.c(157): fwdflat: min_ef_width = 4, max_sf_win = 25
INFO: continuous.c(307): pocketsphinx_continuous COMPILED ON: Sep 29 2019, AT: 15:24:41

Error opening audio device plughw:1,0 for capture: Invalid argument
FATAL: “continuous.c”, line 245: Failed to open audio device

~/MagicMirror/modules/MMM-voice/node_modules/pocketsphinx_continuous/index.js

same file u just showed me

> this._psc = spawn('pocketsphinx_continuous', [
>       '-adcdev',
>       mic,
>       '-inmic',
>       'yes',
>       '-lm',
>       `modules/${this.setId}/${this.setId}.lm`,
>       '-dict',
>       `modules/${this.setId}/${this.setId}.dic`
>     ]);
> 

so, only the path to the files can be wrong…

so, change them to (assume u are running as user pi)

 this._psc = spawn('pocketsphinx_continuous', [
       '-adcdev',
       mic,
       '-inmic',
       'yes',
       '-lm',
       `/home/pi/MagicMirror/modules/${this.setId}/${this.setId}.lm`,
       '-dict',
       `/home/pi/MagicMirror/modules/${this.setId}/${this.setId}.dic`
     ]); 

const spawn = require('child_process').spawn;
const EventEmitter = require('events').EventEmitter;
const util = require('util');

const reservedEvents = {
  close: true,
  error: true,
  data: true,
  end: true,
  readable: true,
  drain: true,
  finish: true,
  pipe: true,
  unpipe: true
};

function PocketSphinxContinuous(config) {
  this.setId = config.setId;
  this.verbose = config.verbose;
  this.microphone = config.microphone;
  this.autostart = config.autostart;

  if (!config.hasOwnProperty('autostart') || config.autostart === true) {
    this.start();
  }

  EventEmitter.call(this);
}

PocketSphinxContinuous.prototype.start = function() {
  if (!this.isListening()) {
    let mic = this.microphone;
    // if name starts with number
    if (typeof this.microphone === 'number' || this.microphone.match(/^\d/)) {
      mic = `plughw:${this.microphone}`;
    }

    this._psc = spawn('pocketsphinx_continuous', [
      '-adcdev',
      mic,
      '-inmic',
      'yes',
      '-lm',
      `modules/${this.setId}/${this.setId}.lm`,
      '-dict',
      `modules/${this.setId}/${this.setId}.dic`
    ]);

    this._psc.stdout.on('data', data => {
      const output = data.toString().trim();
      if (output) {
        this.emit('data', output);
      }
      // Also try to emit an event for the actual data, unless of course the
      // event is a reserved one.
      const eventName = output.toLowerCase();
      if (!reservedEvents[eventName]) {
        this.emit(eventName, output);
      }
    });

    this._psc.stderr.on('data', data => {
      const output = data.toString().trim();
      if (this.verbose && output) {
        this.emit('debug', data);
      }
    });

    this._psc.on('close', code => {
      this.emit('error', code);
    });

    this._psc.on('error', err => {
      this.emit('error', err);
    });
  }
};

PocketSphinxContinuous.prototype.stop = function() {
  if (this.isListening()) {
    this._psc.kill();
  }
};

PocketSphinxContinuous.prototype.isListening = function() {
  return this._psc && !this._psc.killed;
};

util.inherits(PocketSphinxContinuous, EventEmitter);

module.exports = PocketSphinxContinuous;

also, both of those files are there.

show me this

 var pc = spawn('pocketsphinx_continuous', [
    '-adcdev',
    'plughw:' + this.microphone,
    '-inmic',
    'yes',
    '-lm',
    'modules/MMM-voice/' + this.setId + '.lm',
    '-dict',
    'modules/MMM-voice/' + this.setId + '.dic'

also check for the two files in the mmm-voice folder MMM-voice.lm and MMM-voice.dic

ERROR: "cmd_ln.c", line 682: No arguments given, available options are:
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: continuous.c(295): Specify '-infile <file.wav>' to recognize from file or '-inmic yes' to recognize from microphone.

SO, pocketsphinx_continuous 2>somefile

to get the list of possible parms