Questions tagged [speech]

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Identifying incorrectly sampled recordings and finding their actual sampling rate

I have a dataset that contains a lot of recordings sampled incorrectly. This incorrect resampling could be explained in two ways: A recording S, which was recorded with a sampling rate of 16kHz, was ...
thanatoz's user avatar
  • 131
0 votes
1 answer
53 views

Simple speech recognition with hmm

I'm new to the field of Machine Learning and currently watching some lectures on speech recognition. I'm attempting to write a simple class implementation for a speech recognition model using Hidden ...
Owen K's user avatar
  • 1
1 vote
0 answers
33 views

How to calculate phase estimation error

I am working on speech enhancement and I want to compare the phase and magnitude between two signals. I compute the STFT for both, and comparing the magnitudes is easy via mean-squared error or some ...
Mirao Da Silva's user avatar
4 votes
1 answer
5k views

As of 2023, is it possible to extract two human voice from single audio track?

Isolation of different human voices from audio Separate two voices from a speech signal Several years ago it was hard to extract voice from music and almost impossible to separate two human voices ...
Schezuk's user avatar
  • 143
1 vote
1 answer
91 views

Use multichannel noise spatially diffused for multichannel diarization

I'm trying to implement speaker diarization (who speaks when) for mono recording audio (1 channel), but now I'm starting to work with multichannel audio (stereo - 2 channels), so I need for data ...
chaimae hadrouch's user avatar
1 vote
0 answers
65 views

How to detect prematurely cut-off speech recording?

I'm searching for simple method to detect whether the audio recording is prematurely ended in the middle of the pronounciation of a word. I haven't found any literature on this topic, my idea is to ...
bully44's user avatar
  • 11
3 votes
3 answers
234 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
0 votes
0 answers
35 views

Why when ploting the ICLabel in EEGLAB some of the IC topographs look exaclty the same?

I am attaching an image with the output of ICLabel. So you can see that IC5 and IC6 are the same. What does this mean? Thank you in advance:)
Eva's user avatar
  • 1
0 votes
2 answers
85 views

How to detect acoustic and/or network degradations in speech as early as possible before automatic speech recognition?

I'd like to learn about techniques and approaches to identify speech degradations before they enter automatic speech recognition system and cause false recognitions. For example, I'd like to detect ...
bully44's user avatar
  • 11
0 votes
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
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
133 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
60 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
0 votes
0 answers
34 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
-1 votes
1 answer
97 views

How do I detect by ear if a quarter second audio clip is that of someone speaking or not?

I have a dataset of quarter second 16 bit mono PCM audio clips, each clip labelled as speech or nonspeech. I am trying to estimate the quality of the labelling by randomly sampling a few of them and ...
Anand's user avatar
  • 99
1 vote
2 answers
130 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
  • 111
0 votes
0 answers
41 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 ...
Blue482's user avatar
  • 103
3 votes
2 answers
1k views

Implementing a pitch shifter

I am trying to implement a pitch shifter similar to this one, in MATLAB for quick prototyping. My goal is to do the reverse, but I assume it is similar. The implementation details are missing. So I am ...
Andrew Au's user avatar
  • 141
8 votes
5 answers
6k views

How can I upsample 22 kHz speech audio recording to 44 kHz, maybe using AI?

I have a 22 kHz mono audio recording, which is mainly speech, a reading. I would like to upsample somehow to 44 kHz, to improve the audible quality. I have read about there are AI methods for ...
Konstantin's user avatar
3 votes
1 answer
452 views

Convolution : Anechoic Speech signal and Impulse response

I am trying to perform convolution in MATLAB of an anechoic speech signal (2.1 s) and a impulse response (20 s) using the following code: ...
Tanmayee Pathre's user avatar
1 vote
1 answer
2k 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 ...
Harry Stuart's user avatar
2 votes
2 answers
204 views

How many bps would be required to reproduce the words in speech recognizably?

In Robert Gallager's lecture notes for 6.450 Principles of Digital Communications I, Exercise 2.1 asks an interesting conceptual question: Voice waveforms could be converted to binary data by ...
gtak's user avatar
  • 83
7 votes
1 answer
18k views

What are i-vectors and x-vectors in the context of Speech Recognition?

I have read that i-vectors and x-vectors are widely used in speaker recognition tasks but I don't get the difference between them and how exactly they work. Can someone explain it starting from the ...
mausamsion's user avatar
1 vote
1 answer
74 views

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 ...
ThermoRestart's user avatar
4 votes
2 answers
5k 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: $$\...
MrHat's user avatar
  • 81
0 votes
1 answer
1k views

Audio/speech detection, verification

I would like to write a program to compare two '.wav' files, one of which is the original played file with human speech and the other is a recorded '.wav' file. The program I wish to write in python ...
skrowten_hermit's user avatar
2 votes
2 answers
841 views

Converting speech audio to telephone audio

I am trying to train a machine learning algorithm on telephony speech audio. However, there isn't really enough data for this anywhere that I can find. My solution is to just use speech audio from ...
Harry Stuart's user avatar
0 votes
1 answer
34 views

Characterization of the amount of modulation in a signal

We have used the ICRA babble noise to generate an 8-talker babble. As a background - babble noise is used as a substitute for competing talkers as babble noise lacks lexical content, but contains the ...
AliceD's user avatar
  • 298
0 votes
1 answer
254 views

Frequency detection in audio and what parameter or set of parameters help/limit frequency detection

I'm trying to learn audio and voice/speech processing techniques out of interest. I intend to learn, experiment with algorithms and design, implementation of such systems, frameworks using programming ...
skrowten_hermit's user avatar
3 votes
2 answers
3k views

MFCC window size at different sampling rates

The general recommendation for window size when calculating MFCC seems to be 20-40 msec. This is most often recommended in a context of 16000 samples per second, so leading to a window containing 320-...
EddieM's user avatar
  • 43
1 vote
1 answer
328 views

About speech preprocessing in mobile phones

I understand that cell phones transmit speech signals in digital form through 2G/3G or other standards. But I could not find anything about the signal processing method lying in between the analog ...
h_r_kr_shn_n's user avatar
0 votes
0 answers
195 views

Passband filter before Hilbert transform

I have a .wav file of human speech sampled at 48kHz. Before making the analytic signal with: hilbert(s_real) lets say in range 200 - 3,500 Hz, I probably need to do pass-band filtering, say 300 - 3,...
K-man's user avatar
  • 27
2 votes
3 answers
346 views

Phase change rate of human speech

I have a .wav file of human speech, sampled at 48kHz, with useful signal up to 5kHz, and almost no noise. If I do Hilbert transform in MatLab, I get the Analytic signal: $$ s_\mathrm{a}[n] = s_\...
K-man's user avatar
  • 27
1 vote
1 answer
3k views

How to convert NIST to RIFF wav file

I have a dataset of audios(namely TIDIGITS, collected at Texas Instruments, Inc.) and they are in .wav format. However, when I try to read them from python, I get the following error: ...
UrmLmn's user avatar
  • 151
-2 votes
1 answer
268 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 ...
Anold Atha's user avatar
-1 votes
2 answers
401 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 ...
cdancette's user avatar
1 vote
1 answer
3k views

Pitch estimation using the autocorrelation method

I have the following code: ...
user33584's user avatar
2 votes
1 answer
100 views

How exactly does sound get encoded by a cochlear implant processor?

I've been wanting to know more about how sound is processed by a cochlear implant processor, how they are programmed, and how they stimulate the magnets and electrodes inside. How do you take pure ...
HeavenlyHarmony's user avatar
2 votes
0 answers
135 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 ...
Karol's user avatar
  • 83
2 votes
2 answers
501 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 ...
Youngjae's user avatar
  • 185
-1 votes
1 answer
450 views

symmetrical wav audio in time domain and Phase spectrum significance

I have plotted many wav files in MALTAB and tried to understand their time domain plot , Magnitude spectrum and Phase spectrum. But I don't get why time domain plots of speech wav files are ...
jonny789's user avatar
  • 101
2 votes
2 answers
418 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 ...
Ernani's user avatar
  • 131
4 votes
1 answer
1k 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 ...
Tendero's user avatar
  • 5,010
2 votes
1 answer
616 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 ...
Richard's user avatar
  • 31
0 votes
1 answer
146 views

Trying to compare two speech samples that have been recorded by smartphones in pockets

I'm trying to compare two audio samples of speech from two different Android smartphones to determine if the people are co-located. I'm doing this using the frequency bins of an FFT between 300 Hz - ...
Jonty800's user avatar
  • 101
1 vote
1 answer
109 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 ...
anony's user avatar
  • 15
1 vote
1 answer
2k 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 ...
Yu-Chih's user avatar
  • 113
2 votes
0 answers
2k views

Speex echo cancellation takes a long time (~20s) to converge

I have been experimenting with Speex's echo canceller during the past few days. I got the echo canceller to work, however, it takes a long time until it actually begins filtering out the echo. It ...
nevos's user avatar
  • 161
0 votes
1 answer
574 views

Raw speech data into spectrogram

I am using Delphi trying to make a very simple speech recognition system. The system collects raw data well, but I have a hard time turning the data into a spectrogram: The system collects an array ...
Thomas Riedel's user avatar
-1 votes
1 answer
3k views

Compare two audio tracks

I need to compare two audio tracks and compute the difference. The difference is how close they are in terms of pronunciation, rhythm etc. ignoring speaker voice characteristics. The first track is ...
Andriy Tylychko's user avatar