Signal Processing

Questions tagged [speech]

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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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0answers
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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0answers
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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0answers
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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1answer
43 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 ...
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2answers
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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0answers
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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0answers
34 views

High frequency noise out of a dummy phase vocoder

I am trying to understand where does the high-frequency noise comes from when I use a phase vocoder with no modification to the spectrum or speed. Here is the MATLAB code I used. I changed it to my ...
2
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2answers
219 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 ...
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4answers
3k 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 ...
3
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1answer
142 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: ...
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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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2answers
65 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 ...
5
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1answer
10k 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 ...
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1answer
52 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 ...
4
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2answers
3k 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: $$\...
0
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1answer
886 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 ...
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2answers
427 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 ...
0
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1answer
28 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 ...
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1answer
228 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 ...
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1answer
2k 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-...
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1answer
190 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 ...
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0answers
129 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,...
2
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3answers
308 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_\...
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1answer
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: ...
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1answer
195 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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2answers
294 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
2k views

Pitch estimation using the autocorrelation method

I have the following code: ...
2
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1answer
78 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 ...
2
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0answers
128 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 ...
2
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2answers
272 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 ...
-1
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1answer
243 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 ...
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2answers
329 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 ...
4
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1answer
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 ...
1
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1answer
554 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 ...
0
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1answer
141 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 - ...
1
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1answer
91 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 ...
1
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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 ...
2
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0answers
1k 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 ...
0
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1answer
444 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 ...
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1answer
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 ...
2
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3answers
1k views

Understanding LPC for Formant Estimation

I went through the Matlab tutorial on Formant Estimation using LPC Coefficients. Though I vaguely understand the details, it's not entirely clear why we need to do this. From http://person2.sol.lu.se/...
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1answer
912 views

Is Cross correlation between stationary random signal and noise zero for all lags?

If $x(n)$ is a stationary random process (let say small 20ms window of speech which can be assumed stationary) and $v(n)$ is said to be uncorrelated noise, can we say that cross correlation of $x(n)$ ...
3
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0answers
5k views

What's spectral "tilt"?

I am looking at article Speech-in-noise intelligibility improvement based on spectral shaping and dynamic range compression. In paragraph 2.2 the article mentions "tilt" of the spectral envelope. The ...
0
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1answer
71 views

Fine-grained evaluation of speech recognition results

I know that usually ASR systems are evaluated using WER (word error rate), and I was wondering if it makes sense to use a more fine grained approach. E.g. transforming both the ASR hypothesis and the ...
2
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1answer
465 views

audio quality evaluation

I am creating a dataset, where in many speakers are recording their speech signals using different microphones and in different environment. I would like to evaluate the quality of audio and discard ...
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2answers
739 views

Why discrete cosine transform may not maintain locality

In paper Convolutional neural network for speech recognition, they say I didn't understand the highlighted sentence.
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2answers
3k views

FFT spectrogram in log frequency space. How?

I was reading a paper and the authors say that they extract FFT spectrograms and log-scale in frequency domain. I understand how the general spectrogram is computed, but how to make it log-scale in ...
3
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0answers
470 views

Implementing blind deconvolution in MATLAB

I want to implement blind deconvolution for the signal $ r(n) = h(n) \star s(n) + a(n)$ in MATLAB where $r(n)$ is the recorded speech $h(n)$ is impulse response of room acoustics $s(n)$ is desired ...
4
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2answers
458 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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