VoxForge
VoxForge
1. First you need to create your Julius configuration file. Copy
this sample configuration file sample.jconf
to you 'voxforge/howto' folder. For details on the parameters
contained in your sample.jconf file, see
the Juliusbook. The main parameters are shown below:
# VoxForge configurations: -dfa sample.dfa # finite state automaton grammar file -v sample.dict # pronunciation dictionary -h acoustic_model/hmmdefs # acoustic HMM (ascii or Julius binary) -hlist acoustic_model/tiedlist # HMMList to map logical phone to physical -spmodel "sp" # name of a short-pause silence model -multipath # force enable MULTI-PATH model handling -gprune safe # Gaussian pruning method -iwcd1 max # Inter-word triphone approximation method -iwsppenalty -70.0 # transition penalty for the appended sp models -smpFreq 16000 # sampling rate (Hz) -iwsp # append a skippable sp model at all word ends -penalty1 5.0 # word insertion penalty for grammar (pass1) -penalty2 20.0 # word insertion penalty for grammar (pass2) -b2 200 # beam width on 2nd pass (#words) -sb 200.0 # score beam envelope threshold -n 1 # num of sentences to find # you may need to adjust your "-lv" value to prevent the recognizer inadvertently # recognizing non-speech sounds: -lv 4000 # level threshold (0-32767) # comment these out for debugging: -logfile julius.log -quiet |
2. Make sure your Microphone volume is at a setting that is similar
to when you created your audio files.
4. The Julius will display its startup information on your screen. You can begin speaking when you see:
| ----------------------- System Information end ----------------------- Notice for feature extraction (01), ************************************************************* * Cepstral mean normalization for real-time decoding: * * NOTICE: The first input may not be recognized, since * * no initial mean is available on startup. * ************************************************************* ------ ### read waveform input Stat: capture audio at 16000Hz Stat: adin_alsa: latency set to 32 msec (chunk = 1536 bytes) Error: adin_alsa: unable to get pcm info from card control Warning: adin_alsa: skip output of detailed audio device info STAT: AD-in thread created <<< please speak >>> |
The first 2-3 seconds of your speech will not be recognized as Julius
adjusts its recognition levels. You may need to adjust your volume to get better recognition results.
You should get fair recognition results. To improve recognition, your Acoustic Model needs more audio training data. You need to create new prompts, and record more speech audio files based on these prompts in order to create better Acoustic Models. However, with an automated script, it is much easier to record additional audio data.
To add more speech audio to your Acoustic Model, keep the audio and prompts you have already created, and follow the steps in this how-to:
As you add more audio data, you will begin to notice improvements in your recognition accuracy, and it will reach a point where you no longer need to add any additional training data.
The steps in this how-to are for the creation of a single user Acoustic Model. To create a multi-user Acoustic Models, simply get other people to contribute audio as shown in the previous steps, and re-run your script.