Questions tagged [speech-processing]

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

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114 views

audio filter designing

I know the basic filtering techniques available in signal processing.Does audio signal specifically have any different filtering method?.My application is to filter out the background noise in speech ...
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377 views

MFCC Classification

I'm working on gender estimation from speech signal and I completed MFCC feature extraction. So now I'm trying to estimate gender from these features. But I have frames for an audio file and I ...
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106 views

How do i remove the DC component from an audio signal using Matlab2016?

I have been doing speech processing in matlab2016, and am stack on how to remove DC component. I tried using the the function y=x-x(mean) but it seems not to ...
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80 views

Is there a known correlation between human voice (360Hz to 5kHz) and the frequencies above/below that?

Hello SignalProcessing! Given the important frequencies of a human voice (360Hz to 5kHz) to communicate and the frequencies above and below that make the voice sound way better. Imagine we are ...
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2answers
602 views

Quantify the difference between two audio spectrograms

I have two speech signals of the same length. They contain the exact same speech content but are recorded from two different microphones. I want to compare the differences between them. 1) Assuming ...
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1answer
143 views

Normalize speech features per dimensions or across dimensions

I am working on pipelines for audio processing (audio features that will go into a neural net to create some embeddings). We are using two features : Filter banks, or MFCC. I was wondering what was ...
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1answer
418 views

Cepstrum: IFFT or FFT to enter the cepstral domain?

I'm trying to implement source-filter separation of some pre-recorded speech in Max/MSP. I'm unsure of whether one uses an FFT to enter the cepstral domain, or an IFFT. Some of the textbooks and ...
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1answer
368 views

Octave: Finding the mean fundamental frequency given an audio signal

I have an audio .wav file of someone's speech of which I want to find the mean fundamental frequency. Till now, all I have managed to do is load the data into a vector and (roughly) filter the silence ...
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2answers
167 views

How to make the impulse response sparse? How does one know that the channel is sparse?

I am new to sparse channel estimation algorithms and reading research articles. One such paper is blind sparse channel estimation using a modification of the BOMP technique titled, "Blind Acoustic ...
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1answer
246 views

taking FFT from DCT

There is a speech signal, $y[n]$, in the time-domain. To take it to the frequency-domain, DCT has been got from $y[n]$ (w = dct(y) in MATLAB). After it, I applied ...
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1answer
226 views

The origin of constants in mel-scale formula

To convert frequency to mel, we usually use formula: $$\mathrm{mel}(f) = 1127 \ln \left(1 + \frac{f}{700}\right)$$ I wonder where are $1127$ and $700$ came from? I've read the paper which ...
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1answer
56 views

How to correctly classify voice from other sound events?

I have a huge labeled dataset of several thousand sound events, including human voice, dish washing, things falling to the ground, among others. I need to report when a human voice event takes place....
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115 views

Pre-emphasis: What is connection between parameter a and 6db/oct?

I know that the standard pre-emphasis function is $y[n] = x[n]-\alpha\cdot x[n-1]$ and $\alpha$ is often set as $0.97$. However, when I was learning speech processing using Praat, the pre-emphasis ...
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1answer
121 views

Spectral density interpolation: Applications?

Let $\Phi_1$ and $\Phi_2$ be two matrix-valued power spectral densities. I wonder whether the problem of interpolating $\Phi_1$ and $\Phi_2$ has been studied in the literature and/or applied in ...
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0answers
122 views

Speaker diarization tool for acoustic analysis

I'm researching certain acoustic prosodic features during live clinical interactions. Because of the setting, the recordings always have cross channel bleed, and in some cases everything is recorded ...
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1answer
250 views

File size adjustment using FFT

I have been given the task of doing audio compression using FFT. The assignment is below. Below is my MATLAB code: ...
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1answer
107 views

Speech dereverbaration via maximum kurtosis adaptive filtering

I'm trying to code the algorithm described in Speech dereverbaration via maximum-kurtosis subband adaptive filtering by Gillespie, Malvar and Florencio, and the signal looks cleaner in when I plot it. ...
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1answer
185 views

Speech signal formant spectrogram. How can we determine F1?

The formant spectrogram in this page points F1 and F2 as below; In bottom-right F1 movement, I thought F1 could be below blue one. But how can we determine F1 is upper one not below one? Is it ...
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1answer
262 views

Parseval's Theorem for discrete series

I need to use Parseval's Theorem to find \begin{equation} S\:=\:\sum _{n=-\infty }^{\infty }\:\left[\left(\frac{\sin\left(\frac{\pi }{4}n\right)}{2\pi n}\right)\left(\frac{\sin\left(\frac{\pi \:}{6}n\...
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1answer
203 views

Synchronizing 2 seperate speech audio signals

I have two speech signals: one with a man reading a passage, and one with a woman reading the same passage. I want to synchronize these signals so that, for each word, their readings line up. The ...
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1answer
68 views

why does/ doesn't phase count when you analyze a voice signal?

I am new to digital signal processing, I want to know why phase should not affect the analysis of a voice signal.
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1answer
216 views

MFCC in Speech Recognition and ANN

I have a dataset with over 2000 examples of digits (0 to 9) recorded on wav files. I'm trying to process these files with MFCC so that later I could train an Artificial Neural Network (ANN). My ...
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1answer
1k views

Why is LPCC and LPC used in speech recognition

I am currently trying to understand why LPCC and LPC is being used in speech recognition and why? LPC as I understand is model similar to source filter model, modelling speech production, and LPC ...
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1answer
126 views

Mel separated filter banks

I am currently a bit confused. I am currently trying to understand the MFCC extraction process, and don't understand how the Mel filter banks only provide a scalar output from each filter bank. What ...
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34 views

can you suggest me courses to start DSP? [duplicate]

I'm a cs student who's totally 100 % new to DSP can you recommend me math , physics and all the required courses so i can understand DSP better? thank you and also can you recommend some courses / ...
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1answer
128 views

Cepstrum calculus disambiguation

Learning the cepstrum analysis for speech recognition, I have met two different definitions of cepstrum (for discrete signals): $F^{-1}(ln|F(x[n])|)$ . That is, the cepstrum is the inverse Fourier ...
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1answer
223 views

Autoregressive modeling (linear prediction) of electrical transmission lines?

I've read that the reflection coefficients in speech processing (as computed by the Levinson-Durbin algorithm for solving the Yule-Walker equations) "represent the fraction of energy reflected back" ...
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1k views

Pre-emphasizing in speech recognition

Usually, in speech recognition, the techniques that are used are based on the linear prediction model (Fant, 1960). Parallel to this, the human speech production mechanism causes energy to drop ...
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1answer
301 views

Dynamic Time Warp through SPTK tool

I'm an IT student and got an assignment to do about Dynamic Time Warping(DTW) using the Speech Signal Processing Toolkit (SPTK) and comparing some words spoken by 2 speakers and finding the ...
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424 views

The essential pre-processing steps in speech signal analysis

My question is that for simulating a new scheme for speech emotion recognition (feature extraction and classification), what are the necessary pre-processing steps (I mean any kind of normalization ...
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1answer
853 views

Speech recognition: words vs. phonemes

One could train a speech recognizer with words or phonemes. I guess that they both have their pros and cons: with words there would be a lower error rate as phonemes can be put together building a ...
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1k views

How does noise reduction for speech recognition differ from noise reduction that is supposed to make speech more “intelligible” for humans?

this is a question that has interested me for some time now, mainly because I'm working on noise reduction for an existing speech recognition system myself. Most papers on noise reduction techniques ...
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1answer
77 views

Obtaining databases for speech emotion recognition analysis

For a research work on speech emotion recognition, I need the database of CASIA and EESDB for simulations. I could not find those; how can I get them? Thanks.
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316 views

autoregressive moving average code implementation

I am new to DSP and i am trying to take a wav (human speech) file and apply ARMA filter and plot its PSD graph in python. I see that there are a lot of AR implementations but almost none ARMA. I ...
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179 views

What is advantage of Logarithmic Sampling?

Logarithmic Fourier Transformation This paper starts off in the abstract saying that the given audio signal is sampled logarithmically. We also know that the human ear has a sort of a filter bank ...
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1answer
170 views

Logarithmic Spacing

What is the meaning of sampling logarithmically? How do you space the samples logarithmically? Or to rephrase the second question, How to sample an audio signal logarithmically? It would be ...
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0answers
156 views

What is logarithmic mapping in a filter bank such as the one in the human ear?

"The spacing of the filters in a filter bank used to make an ordinary spectrogram is linear, whereas an auditory filter bank has a spacing that corresponds to the way in which sinusoidal frequency ...
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0answers
97 views

Cepstrum of finite impulse train

In Acero's Spoken language processing (p. 309) it says that the cepstrum of a finite length impulse train $$x[n] = \sum^{M-1}_{k=0} \delta[n-kN]$$ is given by $$\hat{x}[n] = \sum^{ \infty}_{r=1} \frac{...
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1answer
49 views

Suggestions on how to use FIR model in sentence retrieval task

Estimating an impulse response from input-output measurements is called system identification. When the impulse response of the linear system $h(n)$, is of finite duration (M samples) or can be ...
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154 views

How to deal with variations in amplitudes when doing voiced/silence detection?

I have a collection of wave files that has a few well separated words spoken. My goal is to split them into individual files. I wanted the program to work on all files, not just one of them or require ...
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1answer
88 views

What do we mean by Total Factor Space in Audio Processing?

I am referring to the article for understanding Ivectors which says: Over recent years, i-vector-based framework has been proven to provide state-of-the-art performance in speaker verification. ...
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117 views

difference between logged power spectrum and power spectrum?

What is the difference between a logged power spectrum and power spectrum... Well the log part. So can a normal power spectrum be converted to a logged power spectrum just by taking the log of the ...
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1answer
58 views

Is there a method to combine speech signals recorded by different microphones?

I have 3 speech files of the same speech recorded by 3 different microphones at the same time. I'm only concerned to improve the speech quality (in terms of SNR and intelligibility) by combining the ...
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1answer
1k views

The minimal signal to noise ratio (SNR) for people to understand a speech in the noisy background

This question seems easy to find an answer but I got lost in the end. I am trying to find the threshold for people to understand speech in a noisy environment. For example, assume the background noise ...
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1answer
38 views

Pole-Zero Configuration of Glottal Signal

I am currently studying for an exam in my Speech Processing course this Friday and I stumbled upon this question in one of the sample exams: The answer to this question was given as: Can someone ...
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54 views

Finding the closest match of a recorded word to a 'corpus' of singly-recorded words.

I am interested in learning more about processing audio and thought that it will be helpful to learn while doing a project. The project involves a pseudo-speec recognition where I have a corpus of ...
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1answer
216 views

TTS - Diphone database

I am working on text to speech conversion. I have done speech synthesis by concatenation of phonemes.The transition between phonemes is not smooth.I want to shift to diphone synthesis. Where can ...
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0answers
225 views

How to interpret the sign of a cepstral coefficient?

I feel like I've got a decent intuition for what the magnitude of a cepstral coefficient represents (i.e. how much of the signal's energy is contained in harmonics of a given wavelength), but I don't ...
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2answers
281 views

Long term spectral flatness measure voice activity detection

This is the Long term spectral flatness measure code that I have written so far. Can someone please help me, I don't know if I'm writing it correctly. It is working fine and gives high peak for ...
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1answer
777 views

Bandpass filter using wavelet transform

I'm working on a speech recognition project. The first step of this project is to find phoneme in the speech signal. To do that, I found this paper that discusses about it. In the paper, wavelets are ...