Voice Recognition Implementation
-
Well if all works and says no "error" of any kind,
I would try one of the existing examples
and see if it works.Then I would start think about how to use it in Qt.
@mrjj after running the command pocketsphinx_continuous i get the following result
ubuntu@ub:~/Desktop/sphinx-source/pocketsphinx$ pocketsphinx_continuous
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 no 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)
-debug Verbosity level for debugging messages
-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 1 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 2.0 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 lengthINFO: continuous.c(295): Specify '-infile <file.wav>' to recognize from file or '-inmic yes' to recognize from microphone.
ubuntu@ub:~/Desktop/sphinx-source/pocketsphinx$ -
Does look like the tuts so I think its working :)
\o/ good work -
@mrjj after running the command pocketsphinx_continuous i get the following result
ubuntu@ub:~/Desktop/sphinx-source/pocketsphinx$ pocketsphinx_continuous
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 no 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)
-debug Verbosity level for debugging messages
-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 1 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 2.0 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 lengthINFO: continuous.c(295): Specify '-infile <file.wav>' to recognize from file or '-inmic yes' to recognize from microphone.
ubuntu@ub:~/Desktop/sphinx-source/pocketsphinx$ -
@mrjj u said I would try one of the existing examples
and see if it works. where i can get existing examples ? -
@Naveen_D said in Voice Recognition Implementation:
pocketsphinx
http://cmusphinx.sourceforge.net/wiki/tutorialpocketsphinx
There is a Basic Usage (hello world) sample.
That should do it :) -
@Naveen_D said in Voice Recognition Implementation:
pocketsphinx
http://cmusphinx.sourceforge.net/wiki/tutorialpocketsphinx
There is a Basic Usage (hello world) sample.
That should do it :) -
@mrjj hi,
When run this command
pocketsphinx_continuous
-hmm /usr/share/pocketsphinx/model/hmm/en_US/hub4wsj_sc_8k
-dict /usr/share/pocketsphinx/model/lm/en_US/cmu07a.dic
-lm /usr/share/pocketsphinx/model/lm/en_US/hub4.5000.DMPi get this output can u pls tell me wats the prb here...
INFO: continuous.c(295): Specify '-infile <file.wav>' to recognize from file or '-inmic yes' to recognize from microphone. -
Also i am not able to understand what they are doing in the hello world example in
http://cmusphinx.sourceforge.net/wiki/tutorialpocketsphinx can anyone pls help me out in this concern -
@mrjj hi,
When run this command
pocketsphinx_continuous
-hmm /usr/share/pocketsphinx/model/hmm/en_US/hub4wsj_sc_8k
-dict /usr/share/pocketsphinx/model/lm/en_US/cmu07a.dic
-lm /usr/share/pocketsphinx/model/lm/en_US/hub4.5000.DMPi get this output can u pls tell me wats the prb here...
INFO: continuous.c(295): Specify '-infile <file.wav>' to recognize from file or '-inmic yes' to recognize from microphone.@Naveen_D said in Voice Recognition Implementation:
INFO: continuous.c(295): Specify '-infile <file.wav>' to recognize from file or '-inmic yes' to recognize from microphone.
Sadly I dont know pocketsphinx, i just browsed over the docs to help you,
if I should guess, i think it ask where to get the input from.
You give it the data files ( dic + friends) and then it says
give me -INFILE for a file with input or -inmic to use your mic.- Also i am not able to understand what they are doing in the hello world example
Which part? There are pretty good explaining in between the code.
Not sure we can make it much better unless some user comes by that
actually use pocketsphinx :)
- Also i am not able to understand what they are doing in the hello world example
-
@Naveen_D
Hi, you can try the QtSpeech as SGaist mentions.
It seems it uses pocket also
http://cmusphinx.sourceforge.net/2015/10/qtspeechrecognition-api-for-qt-using-pocketsphinx/
"You can find the sources in review in qtspeech project, branch wip/speech-recognition."So maybe :)
-
@Naveen_D
Hi, you can try the QtSpeech as SGaist mentions.
It seems it uses pocket also
http://cmusphinx.sourceforge.net/2015/10/qtspeechrecognition-api-for-qt-using-pocketsphinx/
"You can find the sources in review in qtspeech project, branch wip/speech-recognition."So maybe :)
-
@mrjj hi yes i will try with that also thanks
QtSpeech is it for text to speech or speech to text ? -
@Naveen_D
HI
As the Champ says:"Hi, Just in case, there's a speech recognition branch in the QtSpeech "
-
@mrjj Hi i am trying voice recognition with julius, I have installed julius on my ubuntu system and configured it, i have created one my.jconfg file from the sample.jconfg file which i got after installing julius. but when i run that, I get the following output
ubuntu@ub:~/Documents/julius-4.2.2/test$ padsp julius -C my.jconf
STAT: include config: my.jconf<<< please speak >>>^C
I am not sure is it running or not, How to confirm that it is running or not ?
Naveen_D
-
@mrjj Hi i am trying voice recognition with julius, I have installed julius on my ubuntu system and configured it, i have created one my.jconfg file from the sample.jconfg file which i got after installing julius. but when i run that, I get the following output
ubuntu@ub:~/Documents/julius-4.2.2/test$ padsp julius -C my.jconf
STAT: include config: my.jconf<<< please speak >>>^C
I am not sure is it running or not, How to confirm that it is running or not ?
-
@mrjj Hi i am trying voice recognition with julius, I have installed julius on my ubuntu system and configured it, i have created one my.jconfg file from the sample.jconfg file which i got after installing julius. but when i run that, I get the following output
ubuntu@ub:~/Documents/julius-4.2.2/test$ padsp julius -C my.jconf
STAT: include config: my.jconf<<< please speak >>>^C
I am not sure is it running or not, How to confirm that it is running or not ?
