Questions tagged [speech-processing]

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

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35 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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2answers
61 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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310 views

Harmonics to Noise Ratio Estimation

I'm willing to estimate the Harmonics to Noise Ratio (HNR) of a speech signal x[k] and using autocorrelation method. Theoretically, HNR is given as, $$ \ HNR = \frac{R_{xx}[T_0] }{R_{xx}[0]-R_{xx}[...
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24 views

Techniques for removing noise aside from Spectral Subtraction/Statistical Method?

I am studying how to remove noise from speech audio and I figured the most common error in spectral processing comes in the form of musical noise. Now what are the techniques on reducing this ...
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1answer
269 views

How to reduce synthesis artifacts produced by phase vocoder?

I implemented the phase vocoder algorithm in Python to time-stretch speech signals by following this paper and referring to this MATLAB tutorial. I can distinguish words in the original signal from ...
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20 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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7 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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15 views

Is audio normalisation required for a speaker recognition/speech classification task? How to do it?

Not entirely sure if this Stack or a ML-related stack would be a fit for this question, but I focus on the signal processing here on this question, so I decided this one I'm developing a speech ...
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2answers
1k views

Mel Cepstral Distortion

I am working on a speech synthesis model and I am looking to evaluate my synthesized speech. I found that most people use the Mel Cepstral Distortion (MCD) which can be calculated by the formula: $$\...
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2answers
961 views

Initialize a Room Impulse Response using reverberation time(T60)

I am doing Speech dereverberation using Non-negative matrix factorization. I need to initialize an impulse response($H$) such that its reverberation time ($t_{60}$) is $300$ ms. Let the length of $H$ ...
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1answer
89 views

Silence removal from the audio

Suppose i've a mp3 file. I performed FFT of that mp3 file. Then is it good to state the following: 1. Silence occurs by that particular DFT point whose magnitude is lesser than some threshold. 2. ...
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245 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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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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1answer
316 views

What are F0 counters in Speech signals? What is represented by F0?

I can't find any dedicated definition for Frequency F0 counters. Can anyone tell me what does F0 exactly mean, I am confused about that.
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1answer
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MATLAB & Power Spectral Density - Error - Getting the same range of values regardless of sound type?

I wrote code [attached at end of question] to take .wav files of child vocalizations and tried to see if there are valid ranges for the formants F1:F2:F3 for the 5 ...
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26 views

Difference between Cepstrum and Spectral Envelope

I have no background in Signal processing and i don't understand the difference between cepstrum and spectral envelope. IDFT of the log magnitude of the DFT of a signal is cepstrum then what is ...
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2answers
849 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
218 views

How to prevent Octave Jumping in ACF of Speech Signal?

I'm working with speech signals and my aim is to estimate the fundamental frequency $\ F_0$ of this signal often called as "pitch". The main idea is taking small blocks of the speech signal such ...
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2answers
389 views

Matching LPC frequency response to raw sample data frequency response

Background: The intention is feature/formant extraction from speech input. I have used an FFT on raw samples in the past and have done simple peak picking from a weighted average of frequencies over ...
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1answer
44 views

Given 1 recording of my voice, how can I process it to sound like 2 independent recordings of my voice played in unison?

I have one recording of my voice $x[n]$ singing a song. One thing I could do is independently record me singing the song again to obtain $y[n]$. Then, $z[n] = \frac{1}{2}\left(x[n] + y[n]\right)$ ...
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1answer
341 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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1answer
128 views

Pesq numbers and clean speech comparison to processed speech

I am trying to evaluate my processed speech for my DSP project. My project is an implementation of an MMSE algorithm proposed by Y.Ephraim and D. Malah, "Speech Enhancement Using a Minimum Mean Square ...
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2answers
298 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
159 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
122 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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1answer
382 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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1answer
44 views

Speech Processing applications: stereo data for features

I am working on speech databases that have a stereo format. I want to extract spectral information (Mel-filterbanks, MFCC, LPCC) and also some other prosody features like the fundamental Frequency F0. ...
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1answer
505 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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1answer
90 views

python3: speech recognition - calculating features of a WAV file

I am supposed to do this as a school project, but I am kind of lost when it comes to signals and would appreciate your help. Also, English is not my first language, and I will be translating the ...
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23 views

Stereo echo cancellation for stationary source locations: why does adaption stagnate?

In a stereo echo cancellation system, the echo path for the far-end-signal in the near-end room is estimated by an adaptive filter. Since the same signal is transmitted by two similar far-end paths, ...
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1answer
78 views

Difference between applying a window to a whole signal and frames.

I'm going to do feature extraction using a speech signal. The first step is the windowing of the signal using Hamming window. I'd like to know if I should apply one single window to the whole ...
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1answer
26 views

Removing vocal but retaining all background ambient noises

I have one audio file containing human speeches, a lot of ambient noises like audience laughter, birds chirping, sounds of natures etc.. Now I want to separate it into two audio files where one ...
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1answer
53 views

audio classification based only on first few seconds

I record audio notes on my recorder, and each belongs to a certain category ("work", "parenting", "miscellaneous", etc.). At the beginning of each recording, I say the name of the category, then stay ...
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8 views

Detecting disinfluencies in speech

Can someone tell how to find disinfluencies (like ah, umm etc.) in speech using Kaldi or any other open source tool (preferably in Python)
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28 views

Is fundamental frequency information present in speech-shaped white noise?

I am using a so-called speech-shaped stationary white noise, which means so much as a white noise that has been filtered to match the long-term spectrum of a female speaker. The way it was generated ...
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0answers
187 views

Problem using pitch shifting with TD-PSOLA and formant preservation

I tried out this implementation of TD-PSOLA (https://github.com/sannawag/TD-PSOLA) and even though I know how the algorithm works, I can't understand why I get NO audible differences in the output ...
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2answers
40 views

Distinguishing Spoken Speech from Played-back Recording of Speech

Do any general-purpose techniques exist to distinguish live spoken speech from played-back speech samples? For example, if I were to deploy a speaker verification system to authenticate users by voice,...
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20 views

Stack features in wakeword model

I'm not clear how is the input features are stacked in wakeword applications. Taking https://static.googleusercontent.com/media/research.google.com/en//pubs/archive/43969.pdf as an example, the author ...
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1answer
123 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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544 views

How to overcome different MFCC coefficients distributions from different speech datasets?

I am using MFCC values as features for a machine learning model of speech, detecting age from a voice recording of a person. I work with voice datasets I found on the web: common-voice and vctk. ...
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1answer
45 views

Cone of Confusion

In sound perception, humans adapt by tilting the head or relying on other location cues such as visual input when the source of the sound lies on the cone of confusion. On this cone, no time and level ...
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38 views

How to use DTW for keyword spotting in human speech?

I am trying to use Dynamic Time Wrapping (DTW) + KNN to detect word occurrences in human speech audio. I am to implement the same approach suggested in this DSP StackExchange post, but I am seeing ...
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28 views

Objective Noise Classification for speech enhancement

I'm working on my thesis and it is about an speech enhancement system to use proper noise reduction method/algorithm for different types of noise. I want to test and compare some filters/algorithms ...
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4answers
102 views

Cleaning telephone audio

I have telephone audio at 16kHz and I am trying to use it with a speech transcription engine that converts spoken letters to text. However, the engine has some difficulty transcribing certain letters ...
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4k views

What are some python packages I can use to cut audio files

I have a project in which I have a batch of audio files and I need to remove the audio in it from say time 2secs to 5secs i.e. I have a set duration to be removed. How can I do this in python?
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2answers
81 views

Modelling Unwanted Signal in a LMS Adaptive Filter

I'm having some confusion learning about the LMS Adaptive Filter. I understand that the whole model of adaptive filters relies on the fact that we give it a reference signal to which it keeps ...
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1answer
1k views

Real time background noise reduction or removal

I am trying to build a real time simple audio recognition that can recognize few keywords. Following this tensor flow tutorial i was able to train a simple neural network. Currently the signal flow ...
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1answer
367 views

Why is the Hamming and Hanning window the most widely used for speech spectograms?

When taking the Short-Time Fourier Transform of the speech signal, windowing is applied to taper away the signal frame at both ends and mitigate introduced artifact at expense of the maximum ...
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1answer
34 views

Need some references on sound quality evaluation and syncing

I have to do a project where I'm receiving the same sound signal from different channels (with different offsets), but they may not be consistent all the time. At any given time, I should decide which ...
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1answer
49 views

Audio Processing Glossary

I'm new to audio processing and I hope here be a good place to ask about the definition of terms in this context. My question is what is difference between sound, audio, voice, speech and tone, from ...

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