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Questions tagged [speech-processing]

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

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How do I extract a part of an audio clip whose start and end times are given into a .wav file?

I have data on which I have performed Voice Activity Detection (VAD) and this returns a file containing columns of data in the following order : Segment Id, Audio file name, Start time, End time. For ...
zero_day's user avatar
1 vote
1 answer
74 views

Deriving the posterior distribution parameters of a normal distribution in the context of dereverberation

Context: $\bar{\Theta}$ is the room regression filter coefficients (RRC); $$X_{t} = \bar{\Theta}^{H}\bar{X}_{t-1} + s_{t}$$ means in words: the filter that defines how the room causes reverberation to ...
user3371266's user avatar
2 votes
0 answers
68 views

A filter to remove f0 and lower harmonics from the signal

I'm studying the perception of vowel formants (resonances of the vocal tract) and need to create stimuli where the signal below the first (lowest) formant is removed. I have some synthesised vowels ...
Renata Koch's user avatar
1 vote
1 answer
43 views

Two level Dynamic Programming

I am going through Fundamentals of Speech Recognition (Rabiner). I stumble upon the concept of Two Level Dynamic Programming . Can you suggest me any online resources to study the same?
Anantha Krishnan's user avatar
1 vote
1 answer
82 views

How does Pre-emphasis mitigate finite precision effects?

"The digitised speech signal $s(n)$ is put through a low order digital system (usually first order FIR filter) to spectrally flatten the signal and make it less susceptible to finite precision ...
Anantha Krishnan's user avatar
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1 answer
23 views

Constraints in Dynamic Time Warping for Speech

The Dynamic time warping is applied for time normalization. As shown in the diagram, two different signals with $Tx$ and $Ty$ time instants, are time-normalized to have $T$ time instants. $\phi$ is ...
Anantha Krishnan's user avatar
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21 views

Speech Endpoints Detection using Explicit approach

The "Fundamentals of Speech Recognition" text by Rabiner, mentions three approaches for Speech (Endpoints) detection - Explicit, Implicit, and Hybrid approaches. I am studying the block ...
Anantha Krishnan's user avatar
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14 views

Why HMM transition probabilities plays no significant role in speech modeling

I'm reading a thesis, it mainly uses a GMM-HMM to tackle the acoustic unit discovery task. It proposes a delicate Bayesian framework on top of the GMM-HMM. In page 15 it says Notice that we have not ...
sicheng mao's user avatar
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42 views

Formants extraction from LPC

I would like to do formants extraction using LPC. I made the necessary pre-emphasis before feeding the speech samples to the LPC function. ...
Anantha Krishnan's user avatar
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0 answers
38 views

Estimate the number of words in audio record

I'm trying to estimate the number of words in a record I tried to extract the Short term energy(STE) feature, denoise it, then count the number of peaks in the STE, it gives a very close result to the ...
ahmad alghadban's user avatar
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37 views

Understanding Baum's auxiliary function used in Hidden Markov Model

I am studying HMM from "Fundamentals of Speech Recognition" by Rabiner. Regarding the problem of how to adjust the parameters of a HMM, the proposed method was Baum Welch method (Expectation-...
Anantha Krishnan's user avatar
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Exact computation cost using Forward Procedure for HMM probability evaluation

I am understanding the HMM concept for Speech Recognition. One of the three problems (HMM) to be solved is Probability evaluation $P(O|\lambda)$. Instead of a straightforward approach which is ...
Anantha Krishnan's user avatar
1 vote
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70 views

Advantage and Disadvantage using Mel Spectrograms over STFT in speech/waveform synthesis

I want to ask you a question about the waveform synthesis or more spesifically speech synthesis. Most of the state-of-the art papers use mel-spectrograms as their inputs, because it mimics the human ...
Yalçın Cenik's user avatar
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0 answers
25 views

Musical Instrument Characterization

I am doing an undergraduate project where I want to first characterize an instrument based on its pitch and timbre and make an instrument recognition system afterward. Pitch characterization is pretty ...
Flash's user avatar
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1 vote
0 answers
91 views

In "Mel spectrogram" or "Mel filterbanks", what does Mel mean and why is it often capitalized? [closed]

In "Mel spectrogram" or "Mel filterbanks", what does Mel mean and why is it capitalized ? It doesn't seem to be the name of a person.
f10w's user avatar
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Formant bandwidths

I have some intuition about bandwidths from these videos (1 and 2) explaining how the frequency response corresponds to the Z-surface. The only thing I could find on the internet about calculating the ...
Tom Huntington's user avatar
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0 answers
63 views

How do I eliminate background noise?

After I use deep learning algorithm to enhance the speech, the speech will still have a weak background noise.The background noise of this audio has little effect on the calculation of SNR, but it ...
Killuaisaack's user avatar
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12 views

Looking for UBM-GMM trained on the TIMIT database

Hello I was wondering if there was any publicly available pretrained UBM-GMM on the TIMIT acoustic-phonetic continuous speech corpus.
Asmodeus's user avatar
2 votes
1 answer
352 views

Is it possible to calculate DRR (Direct Reverberant Ratio) from an Room Impulse Response? If not, how is this parameter calculated?

DRR (Direct-to-Reverberant Ratio) and Reverberation time (T60) are said to be two of the most important characteristics of sound in Reverberation. DRR is said to be ratio of the energies of the direct ...
Murali Kadambi's user avatar
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0 answers
12 views

Close embedding similarity issue in Korean speaker verification using X-vector

I'm working on a speaker verification system using X-vector on Korean speakers' data. On unseen data, the system achieves fairly good EER (from 10% to 20%), but I'm encountering a challenge with close ...
Luka Nguyen's user avatar
1 vote
1 answer
326 views

How to calculate the reverberation time RT60 given dimensions of a room?

I'm trying to estimate RT60 based on reverberant sound. In order to generate reverberant sound, I am required to give dimensions of a room so that RIR is generated and is convolved with clean sound. ...
Murali Kadambi's user avatar
2 votes
2 answers
471 views

What is the explanation that filter EQ changes "Ah" to "O"?

I recorded myself saying "Ah" in the tone of the note C3 which is about 130Hz, this is what it looks like: Then I ...
google dev's user avatar
0 votes
1 answer
88 views

How can sound waves be modeled in a manner that distinguishes individual voices but also recognizes words?

More specifically, I'm curious how we can represent sound waves in a way that would both distinguish between individual voices and also recognize, e.g. that recordings of different people saying the ...
Mikayla Eckel Cifrese's user avatar
9 votes
5 answers
1k views

Identify abrupt changes in an audio waveform

I have an audio file which was created by substituting multiple speech segments into an original utterance. I want to find the timestamps where such substitutions happened, so I guess that one thing I ...
Long's user avatar
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2 votes
0 answers
50 views

Global gain and frame gain in audio and speech processing

I'm wondering if there is a standard definition for global gain and frame gain in audio and speech processing. I found these terms together with an imprecise definition in a text about bandwidth ...
JosefMaier93's user avatar
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220 views

Good models to separate speech and noise?

I have an audio clip containing speech and noise. I want to separate the noise signal from the speech signal. I've looked at some deep-learning based models, but they only remove the noise, without ...
Sophiefy's user avatar
1 vote
1 answer
135 views

Implementing Statistical Pitchmark Correction Method formula in programming code

I am trying to follow along a paper: The IVO Software Blizzard 2007 Entry: Improving Ivona Speech Synthesis System This paper explains how automatically estimated pitchmarks (certain time periods in ...
tmighty's user avatar
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2 votes
1 answer
84 views

What are some speech features that vary with what the speaker says?

I have been working on a text-dependent speaker verification project. I have used mel-frequency cepstrum coefficients(MFCC) for the project. As far as I know, MFCC coefficients depend on the vocal ...
Wanderer's user avatar
2 votes
1 answer
120 views

MFCC: Does the DCT discard the information on pitch?

Given that the pitch harmonics of speech, are higher frequency features than the spectral envelope (of the pitch harmonics), I had an idea that computing the DCT of a spectrogram and discarding the ...
Tom Huntington's user avatar
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0 answers
60 views

How to decided the score threshold in speaker recognition?

I am new to speaker recognition and now starting from the GMM-UBM algorithm. Say I have a pre-trained UBM and speech from target speakers ...
LCS's user avatar
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0 answers
40 views

How to generate PSHC carrier used in vocoders

Vocoders are often used to acoustically simulate the information transmitted by a cochlear implant. The different types of vocoders used in cochlear implants include noise-vocoding, sine-vocoding and ...
Aami's user avatar
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3 votes
3 answers
235 views

Using spectogram to speed up a signal - Time Scaling/Phase Vocoder

Background About half a year ago, while learning about spectograms as part of an Image Processing course I took, I was told you can speed up audio using spectograms as follows: Calculate the ...
snatchysquid's user avatar
2 votes
0 answers
50 views

Looking for audio/speech filtering expert for mobile hearing app

I am an Ear-Nose-Throat doctor (hearing doctor), looking for some help from someone who has built a mobile app with Frequency Filtering and Amplification. I am working on a social impact program for a ...
Deepam 's user avatar
0 votes
0 answers
64 views

How does phase spectrum compensation works?

I'm trying to find some information that explains how phase spectrum compensation of non-stationary signals methods works, but I can't find an exact description of the procedure. I can't understand ...
E. Ginzburg's user avatar
1 vote
1 answer
55 views

Similar voice features for imitated voice

What kind of audio signal features/properties are appropriate for signal similarity measurement invariant to imitation? Basically I would like to do the following: Having a template (e.g. a spoken ...
Micka's user avatar
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0 answers
28 views

How can I speech enhancement with DNN?

I want to do speech enhancement with DNN on Matlab. I downloaded the TIMIT dataset from the internet as a .wav file. I also downloaded the noises from the Noisex-92 database as .m file. Now I need to ...
Zang Li's user avatar
  • 59
1 vote
1 answer
148 views

Should i use window with hop_size in Wavelet Transform or Discrete Wavelet Transform?

I have a signal (audio - voice) with 1 second of duration with sample rate of 50000 Hz. It is big signal and I wish extract some features and apply pattern recognition or classification. My question ...
Rafael Santos's user avatar
4 votes
1 answer
173 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-...
Zang Li's user avatar
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0 votes
0 answers
116 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 ...
El kouk Ilyas's user avatar
0 votes
2 answers
51 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 ...
Muhammad Sohaib Khalid's user avatar
0 votes
0 answers
148 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?
Aniiya0978's user avatar
0 votes
0 answers
380 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 $x$ ...
Aneela Jaffer's user avatar
2 votes
2 answers
1k 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 ...
Khubaib Ahmad's user avatar
0 votes
0 answers
62 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 ...
Muhammad Sohaib Khalid's user avatar
1 vote
1 answer
92 views

What is the "chirped response FIR filter"?

In Pitch synchronous split-band LPC (PS-SBLPC) vocoder the authors describe an algorithm to identify pitch cycle boundaries. In "Pitch cycle detection" section, this sentence describes ...
makkarpov's user avatar
  • 111
3 votes
2 answers
1k 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 ...
benjaminchanming's user avatar
0 votes
0 answers
36 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 ...
Kadir Gunel's user avatar
0 votes
0 answers
96 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 ...
stck8888's user avatar
0 votes
1 answer
41 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 ...
P2000's user avatar
  • 171
1 vote
2 answers
140 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 ...
karlosos's user avatar
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