MM won't run, blank screen with cursor showing after updating

[sdetweil]

@sdetweil try arecord -D plughw:1,0 test.wav

[jmartin1009]

@sdetweil i wish i had a fix for this lol ive had issues with every voice control module ive tried so far, just want one for closing the screen and stuff, don’t need an assistant or anything

[sdetweil]

@jmartin1009 yeh, all I use it for too. Sleep and wake up

[jmartin1009]

@sdetweil the plughw record test worked!! where do we go now?

[sdetweil]

@jmartin1009 edit that file back. But… I don’t know how to fix the root problem

[jmartin1009]

@sdetweil edited the file back, and changed it in config too, but stil showing error when MM runs :/

[sdetweil]

@jmartin1009 yes, it has the invalid parameter

[jmartin1009]

@sdetweil any more ideas?

[sdetweil]

@jmartin1009 do

pocketsphinx_continuous --help

[jmartin1009]

@sdetweil in which folder, like cd what

[sdetweil]

@jmartin1009 it should be in the path, so right where u are

[jmartin1009]

@sdetweil

pi@raspberrypi:~/MagicMirror $ pocketsphinx_continuous --help
ERROR: "cmd_ln.c", line 607: Unknown argument name '--help'
ERROR: "cmd_ln.c", line 704: Failed to parse arguments list
INFO: continuous.c(295): Specify '-infile <file.wav>' to recognize from file or '-inmic yes' to recognize from microphone.

[sdetweil]

@jmartin1009 try -v instead of --help

[jmartin1009]

@sdetweil

ERROR: "cmd_ln.c", line 607: Unknown argument name '-v'
ERROR: "cmd_ln.c", line 704: Failed to parse arguments list
INFO: continuous.c(295): Specify '-infile <file.wav>' to recognize from file or '-inmic yes' to recognize from microphone.

[sdetweil]

@jmartin1009 grrr. Programmers

[jmartin1009]

@sdetweil as a programmer in college right now, i relate lmao any leads or ideas about it?

[sdetweil]

@jmartin1009 if you give it NO parms, it dumps them out…

SO, pocketsphinx_continuous 2>somefile

to get the list of possible parms

[jmartin1009]

@sdetweil

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.

[sdetweil]

@sdetweil also, in the node_modules/pocketsphinx_continuous/index.js

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

[jmartin1009]

@sdetweil this is the entire index file, didn’t see that anywhere

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.