Original Source Here
Let us walk through all these steps and processes one by one, in detail, with the corresponding pseudo-code. Also, before we start, below is a link for the complete code repository that will be handy to go through alongside this tutorial.
Step 1: Reading a File for Audio Signals
File I/O in Python (scipy.io): SciPy has numerous methods of performing file operations in Python. The I/O module that includes methods read(filename[, mmap]) and write(filename, rate, data) is used to read from a .wav file and write a NumPy array in the form of a .wav file. We will be using these methods to read from and write to sound (audio) file formats.
The first step in starting a speech recognition algorithm is to create a system that can read files that contain audio (.wav, .mp3, etc.) and understanding the information present in these files. Python has libraries that we can use to read from these files and interpret them for analysis. The purpose of this step is to visualize audio signals as structured data points.
- Recording: A recording is a file we give to the algorithm as its input. The algorithm then works on this input to analyze its contents and build a speech recognition model. This could be a saved file or a live recording, Python allows for both.
- Sampling: All signals of a recording are stored in a digitized manner. These digital signatures are hard for software to work upon since machines only understand numeric input. Sampling is the technique used to convert these digital signals into a discrete numeric form. Sampling is done at a certain frequency and it generates numeric signals. Choosing the frequency levels depends on the human perception of sound. For instance, choosing a high frequency implies that the human perception of that audio signal is continuous.
# Using IO module to read Audio Files
from scipy.io import wavfile
freq_sample, sig_audio = wavfile.read("/content/Welcome.wav")# Output the parameters: Signal Data Type, Sampling Frequency and Duration
print('\nShape of Signal:', sig_audio.shape)
print('Signal Datatype:', sig_audio.dtype)
print('Signal duration:', round(sig_audio.shape / float(freq_sample), 2), 'seconds')
>>> Shape of Signal: (645632,)
>>> Signal Datatype: int16
>>> Signal duration: 40.35 seconds# Normalize the Signal Value and Plot it on a graph
pow_audio_signal = sig_audio / np.power(2, 15)
pow_audio_signal = pow_audio_signal [:100]
time_axis = 1000 * np.arange(0, len(pow_audio_signal), 1) / float(freq_sample)plt.plot(time_axis, pow_audio_signal, color='blue')
This is the representation of the sound amplitude of the input file against its duration of play. We have successfully extracted numerical data from an audio (.wav) file.
Step 2: Transforming Audio Frequencies
The representation of the audio signal we did in the first section represents a time-domain audio signal. It shows the intensity (loudness or amplitude) of the sound wave with respect to time. Portions with amplitude = 0, represent silence.
In terms of sound engineering, amplitude = 0 is the sound of static or moving air particles when no other sound is present in the environment.
Frequency-Domain Representation: To better understand an audio signal, it is necessary to look at it through a frequency domain. This representation of an audio signal will give us details about the presence of different frequencies in the signal. Fourier Transform is a mathematical concept that can be used in the conversion of a continuous signal from its original time-domain state to a frequency-domain state. We will be using Fourier Transforms (FT) in Python to convert audio signals to a frequency-centric representation.
Fourier Transforms in Python: Fourier Transforms is a mathematical concept that can decompose this signal and bring out the individual frequencies. This is vital for understanding all the frequencies that are combined together to form the sound we hear. Fourier Transform (FT) gives all the frequencies present in the signal and also shows the magnitude of each frequency.
All audio signals are composed of a collection of many single-frequency sound waves that travel together and create a disturbance in the medium of movement, for instance, a room. Capturing sound is essentially the capturing of the amplitudes that these waves generated in space.
NumPy (np.fft.fft): This NumPy function allows us to compute a 1-D discrete Fourier Transform. The function uses Fast Fourier Transform (FFT) algorithm to convert a given sequence to a Discrete Fourier Transform (DFT). In the file we are processing, we have a sequence of amplitudes drawn from an audio file, that were originally sampled from a continuous signal. We will use this function to covert this time-domain to a discrete frequency-domain signal.
Let us now walk through some code to implement Fourier Transform to Audio signals with the aim of representing sound to its intensity (decibels (dB))
# Working on the same input file
# Extracting the length and the half-length of the signal to input to the foruier transform
sig_length = len(sig_audio)
half_length = np.ceil((sig_length + 1) / 2.0).astype(np.int)# We will now be using the Fourier Transform to form the frequency domain of the signal
signal_freq = np.fft.fft(sig_audio)# Normalize the frequency domain and square it
signal_freq = abs(signal_freq[0:half_length]) / sig_length
signal_freq **= 2
transform_len = len(signal_freq)# The Fourier transformed signal now needs to be adjusted for both even and odd cases
if sig_length % 2:
signal_freq[1:transform_len] *= 2
signal_freq[1:transform_len-1] *= 2# Extract the signal's strength in decibels (dB)
exp_signal = 10 * np.log10(signal_freq)
x_axis = np.arange(0, half_length, 1) * (freq_sample / sig_length) / 1000.0plt.plot(x_axis, exp_signal, color='green', linewidth=1)
With this, we were able to apply Fourier Transforms to the Audio input file and subsequently see a frequency domain (frequency against signal strength) representation of the audio.
Step 3: Extracting Features from Speech
Once the speech is moved from a time-domain signal to a frequency domain signal, the next step is to convert this frequency domain data into a usable feature vector. Before starting this, we have to know about a new concept called MFCC.
Mel Frequency Cepstral Coefficients (MFCCs)
MFCC is a technique designed to extract features from an audio signal. It uses the MEL scale to divide the audio signal’s frequency bands and then extracts coefficients from each individual frequency band, thus, creating a separation between frequencies. MFCC uses the Discrete Cosine Transform (DCT) to perform this operation. The MEL scale is established on the human perception of sound, i.e., how the human brain process audio signals and differentiates between the varied frequencies. Let us look at the formation of the MEL scale below.
- Human voice sound perception: An adult human, has a fundamental hearing capacity that ranges from 85 Hz to 255 Hz, and this can further be distinguished between genders (85Hz to 180 Hz for Male and 165 Hz to 255 Hz for females). Above these fundamental frequencies, there also are harmonics that the human ear processes. Harmonics are multiplications of the fundamental frequency. These are simple multipliers, for instance, a 100 Hz frequency’s second harmonic will be 200 Hz, third would be 300 Hz, and so on.
The rough hearing range for humans is 20Hz to 20KHz and this sound perception is also non-linear. We can distinguish low-frequency sounds better in comparison to high-frequency sounds. For example, we can clearly state the difference between signals of 100Hz and 200Hz but cannot distinguish between 15000 Hz and 15100 Hz. To generate tones of varied frequencies we can use the program above or use this tool.
- MEL Scale: Stevens, Volkmann, and Newmann proposed a pitch in 1937 that introduced the MEL scale to the world. It is a pitch scale (scale of audio signals with varying pitch levels) that is judged by humans on the basis of equality in their distances. It is basically a scale that is derived from human perception. For example, if you were exposed to two sound sources distant from each other, the brain will perceive a distance between these sources without actually seeing them. This scale is based on how we humans measure audio signal distances with the sense of hearing. Because our perception is non-linear, the distances on this scale increase with frequency.
- MEL-spaced Filterbank: To compute the power (strength) of every frequency band, the first step is to distinguish the different feature bands available (done by MFCC). Once these segregations are made, we use filter banks to create partitions in the frequencies and separate them. Filter banks can be created using any specified frequency for partitions. The spacing between filters within a filter bank grows exponentially as the frequency grows. In the code section, we will see how to separate frequency bands.
Mathematics of MFCCs and Filter Banks
MFCC and the creation of filter banks are all motivated by the nature of audio signals and impacted by the way in which humans perceive sound. But this processing also requires a lot of mathematical computation that goes behind the scenes in its implementation. Python directly gives us methods to build filters and perform the MFCC functionality on sound but let us glance at the maths behind these functions.
Three discrete mathematical models that go into this processing are the Discrete Cosine Transform (DCT), which is used for decorrelation of filter bank coefficients, also termed as whitening of sound, and Gaussian Mixture Models — Hidden Markov Models (GMMs-HMMs) that are a standard for Automatic Speech Recognition (ASR) algorithms.
Although, in the present day, when computation costs have gone down (thanks to Cloud Computing), deep learning speech systems that are less susceptible to noise, are used over these techniques.
DCT is a linear transformation algorithm, and it will therefore rule out a lot of useful signals, given sound is highly non-linear.
# Installing and importing necessary libraries
pip install python_speech_features
from python_speech_features import mfcc, logfbank
sampling_freq, sig_audio = wavfile.read("Welcome.wav")# We will now be taking the first 15000 samples from the signal for analysis
sig_audio = sig_audio[:15000]# Using MFCC to extract features from the signal
mfcc_feat = mfcc(sig_audio, sampling_freq)
print('\nMFCC Parameters\nWindow Count =', mfcc_feat.shape)
print('Individual Feature Length =', mfcc_feat.shape)
>>> MFCC Parameters Window Count = 93
>>> Individual Feature Length = 13mfcc_feat = mfcc_feat.T
# Generating filter bank features
fb_feat = logfbank(sig_audio, sampling_freq)
print('\nFilter bank\nWindow Count =', fb_feat.shape)
print('Individual Feature Length =', fb_feat.shape)
>>> Filter bank Window Count = 93
>>> Individual Feature Length = 26fb_feat = fb_feat.T
If we see the two distributions, it is evident that the low frequency and high-frequency sound distributions are separated in the second image.
The MFCC, along with application of Filter Banks is a good algorithm to separate the high and low frequency signals. This expedites the analysis process as we can trim sound signals into two or more separate segments and individually analyze them based on their frequencies.
Step 4: Recognizing Spoken Words
Speech Recognition is the process of understanding the human voice and transcribing it to text in the machine. There are several libraries available to process speech to text, namely, Bing Speech, Google Speech, Houndify, IBM Speech to Text, etc. We will be using the Google Speech library to convert Speech to Text.
Google Speech API
More about the Google Speech API can be read from the Google Cloud Page and the Speech Recognition PyPi page. A few key features that the Google Speech API is capable of are the adaptation of speech. This means that the API understands the domain of the speech. For instance, currencies, addresses, years are all prescribed into the speech-to-text conversion. There are domain-specific classes defined in the algorithm that recognize these occurrences in the input speech. The API works with both on-prem, pre-recorded files as well as live recordings on the microphone in the present working environment. We will analyze live speech through microphonic input in the next section.
Trending AI/ML Article Identified & Digested via Granola by Ramsey Elbasheer; a Machine-Driven RSS Bot