Questions tagged [speech-processing]

Speech processing is the study of speech signals and the processing methods of these signals.

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59 views

Which Programming Language Should Be Used for Deep Learning (Deep Neural Network [DNN])?

I will do voice activity detection and speech enhancement based deep neural network. However, I don't know whether to do this via matlab or pyhton. In which programming language can I find more ready-...
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31 views

Realtime Independent Vector Analysis

I have been working on an implementation of Real-time (Online) IVA (independent vector analysis). The paper I am referring to is: Real-Time Independent Vector Analysis for Convolutive Blind Source ...
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1answer
28 views

Eliminate different microphones textures from their recordings

In my setting, I have different microphones recording a same speech signal. For each microphone recording, I can feel a slight difference in the playing sound. I am calling this property of a ...
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18 views

Calculation of ITD (Interaural time delay) using linear prediction - python

How is it possible to calculate the interaural time delay between two signals recorded underwater (ship sound signals)? Will linear prediction weight method be used for this?
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27 views

How can we find Room Impulse Response in real-time using Sine Sweep Method?

I want to calculate the room impulse response in real-time using sine sweep method. For that, I generated a sine sweep $x$ & its amplitude modulated inverse signal $f$. Then I played that $x$ ...
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2answers
78 views

How to use deconvolution technique to find out impulse response?

I have been working to find out room for impulse response. I am using Logarithmic sweep sine wave as input say $x(n)$ and my recorded signal is $y(n)$. I know the room impulse response is ...
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53 views

Voice glitch while selecting signal to play from a series incoming recorded signals

In my setting, I have multiple microphones recording the same speech signal in real-time. The speaker is moving while speaking. I am selecting one of the recorded signal to play based on their RMS ...
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1answer
49 views

What is the "chirped response FIR filter"?

In this paper authors describe an algorithm to identify pitch cycle boundaries. In "Pitch cycle detection" section, this sentence describes something that authors termed a "chirped ...
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2answers
158 views

What is a "pitch period"?

The term pitch period appears in the book Speech and Language Processing by Daniel Jurafsy: As we just said, a pitch-synchronous algorithm is one in which we do something at each pitch period or ...
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25 views

with PCM-16 no speech sound

I am trying to create a synthethic speech dataset. A quick overview of the my process is: read wav apply stft do some stuff (ex. add noise, etc.) apply istft write wav Now, the .wav file that I am ...
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42 views

Is there a technical term for smoothing or flattening the floor of an amplitude envelope?

I am working with the speech envelope, and I have been trying to make it act a little more like a bilevel signal by exaggerating the peaks and flattening out the floor. I detrended and smoothed my ...
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19 views

Stack audio frames for classification

I have audio files for which I extracted the mfcc features, and I ended up with train dataset of size 3082092 frames by 30 mfcc: torch.Size([3082092, 31]). What is ...
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1answer
36 views

How is voice sound on TV or phone distorted through a face mask?

After a year of listening to people talk through face masks, it appeared to me that the sound is not merely muffled, but it has a very specific and recognizable tinge and texture to it. It reminds me ...
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55 views

What are the fundamental differences between speech and music in frequency domain?

What are the fundamental differences between speech and music in the frequency domain? I can think mainly of features in the time domain that can differentiate speech from music signals, e.g. short ...
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35 views

Finding "Max Change" in a Audio Signal

I'm working on studying an endangered language and have the following problem: while I already have labeled exactly what the speakers I worked with have said at given timesteps on the word level and I ...
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1answer
52 views

Amplification of a speech signal in the recorded audio

I have recorded an audio. I dont know how it happened that only one sided speech is recorded and the other speech is recorded with a very low sound. Is there any solution to amplify the other side ...
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40 views

Python - preparing wav file so speech recognition (google) can return accurate results

I am trying to convert my wav files from playing tennis into text. I intend to use Google Cloud so ideally this would work with their Speech Recognition tool. TLDR: Tried denoising Tried slowing ...
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27 views

Automatically identifying speech disfluencies from audio recordings

I want to automatically identify speech disfluencies (e.g. um(s) and uh(s) ) from my speech recordings. I am a bit surprised I couldn't find any disfluency detection models using speech only. I know ...
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49 views

Splitting Arabic speech by letter

I have the audio stream below, which is a recording of Arabic. I am trying to split the stream into individual letters. I have manually marked the ideal locations with black lines in the image below. ...
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12 views

Viterbi algorithm in HMM (Speech processing-oriented)

Currently, I'm following a tutorial about Viterbi which uses 12 time steps, and as an example, it gives the last 3 MFCC coeffs per time step: https://arxiv.org/pdf/1908.03143.pdf In the page 3 (E. ...
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1answer
48 views

Sound recording anonymization

I want to study stress based on sound recordings. However, there are many concerns about keeping the identity of the recorded persons anonymous. I want to keep intensity and spectral contents intact ...
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42 views

Logarithmic Amplitude Spectral Subtraction

I'm diving into audio processing and I'm trying to wrap my head around spectral subtraction. I learned there are different approaches do it based on power, magnitude, oversubtraction. My task is to ...
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1answer
33 views

How to select the order of LPC to model formants?

I am having some difficulties understanding LPC and speech processing in general. The professor asked us to provide an heuristic way to select the order of LPC? For example if I wanted to model 3 ...
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119 views

What is the difference between MFCC features and Mel Filter Banks (FBANKs)?

I haven't been able to find any source explaining what the FBANK features are. As a reference, I have been reading this paper here which mentions that they have used 20 MFCC coefficients from 40 mel ...
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1answer
92 views

Male and female voice spectrum

I have a set of data which consists of male and female voices. They are pronunciation of a same sentence. What's the appropriate method for getting an average spectrum for the male and female voices ...
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1answer
80 views

How can you perform echo cancellation on features of audio signals?

I am working on a project that takes some features of audio signals and classify them with a neural network. However, the audio signals are added with their own echo with the environment. What would ...
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1answer
120 views

How to compare 2 MFCC (compare them if they where created from same speaker)

There are multiple speakers. Each speaker generates multiple MFCC. If I'm getting two different MFCC, can we know if those MFCC came from 2 different speakers or same speaker ? How can we compare ...
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2answers
52 views

Why are males' pitch lower than females?

Hi :) I'm not sure this is the right place to ask this question, but I'll try anyway: Here are two well-known facts, and I don't understand how they work together: Males usually have deeper voices ...
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1answer
90 views

Formant analysis LPC (Matlab)

I wanted to do formant analysis using MATLAB and did everything according to the following tutorial: https://de.mathworks.com/help/signal/ug/formant-estimation-with-lpc-coefficients.html I have speech ...
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40 views

Average spectral shape

In this article about pre-emphasis on here, its use is defined as "to compensate for the average spectral shape". My question is, how is the average spectral shape calculated? This is the ...
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2answers
162 views

Non-Uniform Quantization

I was reading a research paper on companding schemes for non uniform quantisation. In paper one of the motivations for non-uniform quantisation is that distortion at larger amplitude values is less ...
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1answer
96 views

Determining statistically the bandwidth of a signal from a spectrogram

My question is how to determine the parameters of a signal from a spectrogram statistically. The most important parameters that I am looking for are the lower and higher frequencies of the signal. I ...
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15 views

Source Filter Modelling of Speech Production

I am reading about mathematically modeling Speech using Source- Filter model. I want to know how can we prove that that the source filter model becomes an LTI system provided the configuration of ...
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1answer
32 views

Does speech volume affect frequencies?

I have created a little classifier that listens for certain keywords. Sometimes it will not react to a keyword, but if I shout the word, then the classifier will be able to pick it up. The classifier ...
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45 views

Sinusoidal tone in Signal-to-Noise measurements

Mengapa kita menggunakan nada uji sinusoidal saat mengukur signal-to-noise pada saluran bicara daripada hanya sinyal ucapan itu sendiri ? EDIT: Translation from Indonesian according to Google: Why do ...
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1answer
60 views

Is this statement correct from DSP aspect?

So, In my report I am comparing the related works, In most of the previous work, researchers have used spectrogram as input to vanilla CNN and classify. Where I am using different handcrafted features ...
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74 views

How to use cross-correlation to verify speech data and determine delay

I have been experimenting with cross-correlation function to verify the presence of speech in a recorded file wrt a source file. I tried the following in Matlab: <...
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1answer
61 views

LPC Formant Estimation: Why resample?

The Burg algorithm implemented in PRAAT for formant estimation also works via LPC. First the sound is resampled to a sampling frequency of twice the value of maximum formant. Then a pre-emphasis is ...
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1answer
99 views

What do different Cepstral or MFCC coefficients represent intuitively?

I understand the explanation for separating slow and rapid changing log spectral components but i need to understand: Why lower coefficients have higher and mostly positive magnitudes? Why Higher ...
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1answer
56 views

How to use MFCC (or any other tool/method/algorithm) for speech verification

I have an original speech .wav file and a few recorded samples of the same file. Each recorded version has some differences wrt the original file. For example, The ...
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0answers
63 views

Frame lookup using audio

I'm an animator trying to get a program that will use machine learning to automate the lip syncing process. The program I need made will have two files given to it: a seed, and an audio file. The seed ...
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0answers
84 views

Understanding MFCC

I am not from electrical or electronic background so my knowledge will be lacking. MFCC is represented by 39 values for each window frame. 12 values are the mel filter-bank and we get 13th value by ...
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2answers
118 views

Conceptually confused by LPC for speech: Do we synthesize by the inverse filter (FIR)?

I've been following the process of using LPC to analyse the speech and then synthesize the speech by swapping out coefficients over time. I'm purely concerned with pitched vowels at this point. ...
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2answers
104 views

Speech SNR estimation

I'm trying to estimate SNR level in db to use it as a switch for an speech enhancement method that i'm working on. The noise level is not known and can be estimated using a posterior SNR algorithm. ...
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1answer
480 views

Removing background noise algorithm

What is the algorithm for removing background noise from an audio clip of human speech? Say I have a 10 second audio clip of somebody talking, and there is background noise of them tapping on their ...
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1answer
113 views

Is zero-padding necessary for speech processing to satisfy linear convolution property?

In the FFT domain, for example, if I multiply a noise reduction gain to the noisy signal, $ Y(w)H(w) $, this translates to a circular convolution in the time domain of the same length $ y(t) \...
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31 views

Capture the ambient sound

I sit in a cafe and start recording my voice, reading a poem. Then I sit at my room, at complete isolation, and I read the same exact poem once again and record it. Now, Considering I have these two ...
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1answer
31 views

Using MFCCs for acoustic machine failure prediction

MFCCs are ubiquitously extracted for speech processing tasks, but I would like to know how suitable they are for non-speech processing tasks. Intuitively, it is my understanding that MFCCs are ...
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2answers
190 views

log base in log-power spectral features

Is the log in log-power spectral features natural logarithm, or common log of base 10? does it even matter? I found sources for how to calculate it, but they all call it 'log'. no one mentions the ...
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54 views

Why do we use the term "cepstral" in MFCCs, and how does it related to the cepstral domain?

As I know, we can get the cepstrum by taking the Inverse Discrete Fourier Transform (IDFT) of the log of the power spectrum. And we can get the Mel-frequency cepstral coefficients (MFCCs) by taking ...

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