This library provides common speech features for ASR including MFCCs and filterbank energies. If you are not sure what MFCCs are, and would like to know more have a look at this MFCC tutorial
This project is on pypi
To install from pypi:
pip install python_speech_features
From this repository:
git clone https://github.com/jameslyons/python_speech_features python setup.py develop
Supported features:
- Mel Frequency Cepstral Coefficients
- Filterbank Energies
- Log Filterbank Energies
- Spectral Subband Centroids
From here you can write the features to a file etc.
The default parameters should work fairly well for most cases, if you want to change the MFCC parameters, the following parameters are supported:
python
def mfcc(signal,samplerate=16000,winlen=0.025,winstep=0.01,numcep=13,
nfilt=26,nfft=512,lowfreq=0,highfreq=None,preemph=0.97,
ceplifter=22,appendEnergy=True)
| Parameter | Description |
|---|---|
| signal | the audio signal from which to compute features. Should be an N*1 array |
| samplerate | the samplerate of the signal we are working with. |
| winlen | the length of the analysis window in seconds. Default is 0.025s (25 milliseconds) |
| winstep | the step between successive windows in seconds. Default is 0.01s (10 milliseconds) |
| numcep | the number of cepstrum to return, default 13 |
| nfilt | the number of filters in the filterbank, default 26. |
| nfft | the FFT size. Default is 512 |
| lowfreq | lowest band edge of mel filters. In Hz, default is 0 |
| highfreq | highest band edge of mel filters. In Hz, default is samplerate/2 |
| preemph | apply preemphasis filter with preemph as coefficient. 0 is no filter. Default is 0.97 |
| ceplifter | apply a lifter to final cepstral coefficients. 0 is no lifter. Default is 22 |
| appendEnergy | if this is true, the zeroth cepstral coefficient is replaced with the log of the total frame energy. |
| returns | A numpy array of size (NUMFRAMES by numcep) containing features. Each row holds 1 feature vector. |
These filters are raw filterbank energies. For most applications you will want the logarithm of these features. The default parameters should work fairly well for most cases. If you want to change the fbank parameters, the following parameters are supported:
python
def fbank(signal,samplerate=16000,winlen=0.025,winstep=0.01,
nfilt=26,nfft=512,lowfreq=0,highfreq=None,preemph=0.97)
| Parameter | Description |
|---|---|
| signal | the audio signal from which to compute features. Should be an N*1 array |
| samplerate | the samplerate of the signal we are working with |
| winlen | the length of the analysis window in seconds. Default is 0.025s (25 milliseconds) |
| winstep | the step between seccessive windows in seconds. Default is 0.01s (10 milliseconds) |
| nfilt | the number of filters in the filterbank, default 26. |
| nfft | the FFT size. Default is 512. |
| lowfreq | lowest band edge of mel filters. In Hz, default is 0 |
| highfreq | highest band edge of mel filters. In Hz, default is samplerate/2 |
| preemph | apply preemphasis filter with preemph as coefficient. 0 is no filter. Default is 0.97 |
| returns | A numpy array of size (NUMFRAMES by nfilt) containing features. Each row holds 1 feature vector. The second return value is the energy in each frame (total energy, unwindowed) |
sample english.wav obtained from:
wget http://voyager.jpl.nasa.gov/spacecraft/audio/english.au sox english.au -e signed-integer english.wav