Update: I finished my tool for creating a customized voice model. It encapsulates the best of what I described below. See here:
https://github.com/umhau/vmc.
The idea with a limited vocabulary is that the processor can deal with far less information in order to detect the words needed. You don't have to train it on a complete set of words in the English language, and you don't need a supercomputer. All you have to do is teach it a few words, and how to spell them. The tutorial is
here. I've created a script to automate the voice recording
here, and stashed the needed files there with it.
Preparation
Alright, down to business. You'll find it handy to keep a folder for these sorts of programs.
mkdir ~/tools
cd ~/tools
Install git, if you don't have it already.
sudo apt-get install git
Download the script I made into your new tools folder.
sudo git clone https://github.com/umhau/train-voice-data-pocketsphinx.git
Install SphinxTrain. I included it among the files you just downloaded. Move it up to ~/tools, extract and install it. It's also
here, if you don't want to use the one I provided.
sudo mv ~/tools/train-voice-data-pocketsphinx/extra_files/sphinxtrain-5prealpha.tar.gz ~/tools
sudo tar -xvzf ~/tools/sphinxtrain-5prealpha.tar.gz -C ~/tools
cd sphinxtrain-5prealpha
./configure
make -j 4
make -j 4 install
Record Your Voice
Enter this directory, run the script. It'll have a basic walkthrough built-in. This will help you record the data you need. For experimental purposes, 20 recordings is enough for about 10% relative improvement in accuracy. Use the name neo-en for your training data, assuming you're working in English.
cd ./train-voice-data-pocketsphinx
python train_voice_model.py
You'll find your recordings in a subfolder with the same name as what you specified. Go there.
cd ./neo-en
By the way, if you ever change your mind about what you want your model to be named, there's a fantastic program called pyrenamer that can make it easy to rename all the files you created. Install it with:
sudo apt-get install pyrenamer
Process Your Voice Recordings
Great! Done with that part. Now we're going to copy some other directories into the current working directory to 'work on them'.
cp -a /usr/local/share/pocketsphinx/model/en-us/en-us .
cp -a /usr/local/share/pocketsphinx/model/en-us/cmudict-en-us.dict .
cp -a /usr/local/share/pocketsphinx/model/en-us/en-us.lm.bin .
Based on
this source, it looks like we shouldn't be working with .dmp files. This is a point of deviation from the (outdated) CMU walkthrough. Copy the .bin file instead. Difference is explained below, sourced from the
tutorial.
Language model can be stored and loaded in three different format - text ARPA format, binary format BIN and binary DMP format. ARPA format takes more space but it is possible to edit it. ARPA files have .lm extension. Binary format takes significantly less space and faster to load. Binary files have .lm.bin extension. It is also possible to convert between formats. DMP format is obsolete and not recommended.
Now, while still in this directory, generate some 'acoustic feature files'.
sphinx_fe -argfile en-us/feat.params -samprate 16000 -c neo-en.fileids -di . -do . -ei wav -eo mfc -mswav yes
Get the Full-Sized Language Model
Nice. You have a bunch more files with weird extensions on them. Now it's time to convert them. You need the full version of the language model, which was not shared with your original installation for size reasons. I included it in the github repository, or you can download it from
here (you want the file named cmusphinx-en-us-ptm-5.2.tar.gz). Put the extracted files in your neo-en directory.
Assuming you use the one from the github repo and you're still in the neo-en subdirectory,
tar -xvzf ../extra_files/cmusphinx-en-us-ptm-5.2.tar.gz -C .
There's an folder labeled en-us within the neo-en folder that was created when you made the acoustic feature files. Give it an extension and save it in case of horrible mistakes.
mv ./en-us ./en-us-original
Now move the newly extracted directory to your neo-en folder, and rename it to en-us.
mv ./cmusphinx-en-us-ptm-5.2 ./en-us
This converts the binary mdef file into a text file.
pocketsphinx_mdef_convert -text ./en-us/mdef ./en-us/mdef.txt
Grab Some Tools
Now you need some more tools to work with the data. These are from SphinxTrain, which you installed earlier. You should still be in your working directory, neo-en. Use ls to see what tools are available in the directory.
ls /usr/local/libexec/sphinxtrain
cp /usr/local/libexec/sphinxtrain/bw .
cp /usr/local/libexec/sphinxtrain/map_adapt .
cp /usr/local/libexec/sphinxtrain/mk_s2sendump .
cp /usr/local/libexec/sphinxtrain/mllr_solve .
Run 'bw' Command to Collect Statistics on Your Voice
Now you're going to run a very long command that is designed to collect statistics about your voice. Those backslashes -- the \ things -- tell bash to ignore the following character: in this case, newline characters. That's how this command is stretching over multiple lines.
./bw \
-hmmdir en-us \
-moddeffn en-us/mdef.txt \
-ts2cbfn .ptm. \
-feat 1s_c_d_dd \
-svspec 0-12/13-25/26-38 \
-cmn current \
-agc none \
-dictfn cmudict-en-us.dict \
-ctlfn neo-en.fileids \
-lsnfn neo-en.transcription \
-accumdir .
Future note, for using the
continuous model instead of the PTM model (
from the tutorial):
Make sure the arguments in bw command should match the parameters in feat.params file inside the acoustic model folder. Please note that not all the parameters from feat.param are supported by bw, only a few of them. bw for example doesn't suppport upperf or other feature extraction params. You only need to use parameters which are accepted, other parameters from feat.params should be skipped.
For example, for continuous model you don't need to include the svspec option. Instead, you need to use just -ts2cbfn .cont. For semi-continuous models use -ts2cbfn .semi. If model has `feature_transform` file like en-us continuous model, you need to add -lda feature_transform argument to bw, otherwise it will not work properly.
More Commands
Now it's time to adapt the model. Looks like continuous will be better to use in the long run, but first we're just going to get this working. The tutorial suggests that using MLLR and MAP adaptation methods together is best, but it looks like so far we're just using them sequentially. Here goes:
./mllr_solve \
-meanfn en-us/means \
-varfn en-us/variances \
-outmllrfn mllr_matrix -accumdir .
It appears this adapted model is now completed! Nice work. To use it, add -mllr mllr_matrix to your PocketSphinx command line. I'll put complete commands at the bottom of this note.
Now we're going to do the MAP adaptation method, which is being used on top of the MLLR method. Back up the files you were just working on:
cp -a en-us en-us-adapt
To run the MAP adaptation:
./map_adapt \
-moddeffn en-us/mdef.txt \
-ts2cbfn .ptm. \
-meanfn en-us/means \
-varfn en-us/variances \
-mixwfn en-us/mixture_weights \
-tmatfn en-us/transition_matrices \
-accumdir . \
-mapmeanfn en-us-adapt/means \
-mapvarfn en-us-adapt/variances \
-mapmixwfn en-us-adapt/mixture_weights \
-maptmatfn en-us-adapt/transition_matrices
[Optional; saves some space]
...I think. Apparently it's now important to recreate a sendump file from a newly updated mixture_weights file.
./mk_s2sendump \
-pocketsphinx yes \
-moddeffn en-us-adapt/mdef.txt \
-mixwfn en-us-adapt/mixture_weights \
-sendumpfn en-us-adapt/sendump
Testing the Model
It's also important to test the adaptation quality. This actually gives you a benchmark - a word error rate (WER). See
here.
Create Test Data
Use another script I made to record test data. It's almost the same, but the fileids and transcription file formats are different. The folder with the test data should end up in the neo-en directory. Use the directory name I provide, test-data.
python ../create_test_records.py
Run the decoder on the test files. Go back into the neo-en folder.
pocketsphinx_batch \
-adcin yes \
-cepdir ./test-data \
-cepext .wav \
-ctl ./test-data/test-data.fileids \
-lm en-us.lm.bin \
-dict cmudict-en-us.dict \
-hmm en-us-adapt \
-hyp ./test-data/test-data.hyp
Use this tool to actually test the accuracy of the model. You'll need a working pocketsphinx installation, since it's just a wrapper with a word comparison engine over the transcription engine. Look at the end of the output; it'll give you some percentages indicating accuracy.
../../pocketsphinx-5prealpha/test/word_align.pl \
./test-data/test-data.transcription \
./test-data/test-data.hyp
Live Testing
If you just want to try out your new language model, record a file and try to transcribe it with these commands (assuming you're still in the neo-en working directory):
python ../record_test_voice.py
pocketsphinx_continuous -hmm ./en-us-adapt -infile ../test-file.wav
Or, if you'd rather use a microphone and record live, use this command:
pocketsphinx_continuous -hmm ./en-us-adapt -inmic yes
With 110 voice records and using 20 records as testing, I achieved 60% accuracy. At 400 records and a marginal mic, I achieved 77% accuracy. There's about 1000 records available.
Achieving Optimal Accuracy
You'll want to create your own language model if you're going to be using a specialized language. That's a pain, and you have to know what you're going to use it for ahead of time. If I do that, I'll collect the words from the tools where I specified them and automagically rebuild the language model. For now, I think I can get away with using the default lm.
For actual use of the model, everything you need is in en-us-adapt. That's what you use when you need to refer in a command to your language-model.
Use the following command to transcribe a file, if you've created your own lm and language dictionary:
pocketsphinx_continuous
-hmm <your_new_model_folder> \
-lm <your_lm> \
-dict <your_dict> \
-infile test.wav
Upon testing it appears that a less controlled environment might be useful, as the transcription was almost perfect when I was able to recreate the atmosphere of the original training records and pretty bad otherwise.
Conclusions
I've made a bunch of scripts in the github repo that automate some of this stuff, assuming standard installs. Look for check_accuracy.sh and create_model.sh. Everything should be run inside the neo-en folder, except the original train_voice_model.py script.
TODO next -
- set up the more accurate continuous model
- create a script that generates words in faux-sentences based on my use case scenario.
- find a phonetic dictionary that covers my needs
- figure out what my use case actually is