Questions tagged [speech]

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29 questions with no upvoted or accepted answers
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4
votes
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 ...
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 ...
3
votes
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 ...
3
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0answers
13k views

MFCC calculation

I've got a project and need to calculate MFCCs. I tried to read some tutorials and then make a MATLAB function but I seem to have wrong answers. I'd like to feed MFCCs to one of the classification ...
2
votes
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 ...
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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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 ...
2
votes
0answers
111 views

What is actually transmitted following LPC of a speech frame?

For each frame, what's sent over to the receiver for decoding? The coefficients , pitch in some bits, voiced/unvoiced classification in another bit? Another broad question to develop intuition about ...
2
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0answers
5k views

Adaptive Delta Modulation For Speech Signals

I need to implement an adaptive delta modulation encoder and decoder using matlab for my term project. How can I code a good adaptive delta modulation? How can I change ...
2
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0answers
128 views

Summer Projects - Suggestions

I'm looking for some suggestions and wondered if anyone had any ideas. I have just completed a project in Speech Recognition using MFCC, Dynamic Time Warping and Hidden Markov Models etc.. (Some of ...
1
vote
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 ...
1
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0answers
65 views

peak-to-peak test between two signal

Suppose that we have different signals (Pulses). I would like to study which signal has deterministic peak-to-peak distant (lag). For example A=[1 0 0 1 0 0 0 1 0 0 1] and B=[1 0 0 1 0 0 1 0 0 1] and ...
1
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0answers
36 views

Examples of cases where "invariance to transformation" on signal is important

I realize that this might not be the most well-formulated question, but I figured I'd throw it out there as an open-ended discussion that could turn up some interesting input. I'm relatively new to ...
0
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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 ...
0
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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?
0
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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 ...
0
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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 ...
0
votes
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 ...
0
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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 ...
0
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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 ...
0
votes
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 ...
0
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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,...
0
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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 ...
0
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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 ...
0
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0answers
62 views

Unique feature value for each phoneme

I'm working in a speech recognition project for the university. I'm using the TIMIT dataset and pylearn2 as machine learning library. The project need to be simple, and it doesn't have to follow the ...
0
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0answers
27 views

Why the formula below are critically-damped second-order linear systems?

This is part formula of command-response model (Fujisaki Model): ...
0
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0answers
101 views

Speech signal Quality & Speech signal improvement

I'm trying find authoritative references for [1] Which analyzing techniques (time/frequency domain) i should use for determining the quality of a speech signal. [2] Which algorithms i should run ...
-1
votes
1answer
137 views

Parameters for signal analysis

I am extremely new to signal analysis. And before posting I did a lot of reading on signal analysis, FFT and windowing. I am working on my thesis which involves comparison of speech signals lets say ...
-2
votes
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 ...