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

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Calculating Delta MFCC, correct formula?

I have seen different formulations of how to calculate delta mfccs. In general, I want to be consistent in such a way that if have other features to look at other than mfccs, the way I would ...
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1answer
29 views

Using 8 bit samples for speech comparison

I am making a speech comparison app in javascript. I found this dsp.SE post: Speech comparison algorithm for rating on similarities, where I read it is best to use mfcc and dtw algorithms. I ...
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27 views

Regarding maximizing signal to noise ratio

If I have 3 recordings, each of speech (recorded in a relatively silent room), speech+background noise and background noise then is there a way to utilise the data from all the three to extract clear ...
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1answer
42 views

Computing of speech signals modulation entropy

I got a question regarding to modulation entropy of audio signal. There is some literature I found about this topic, http://www.irit.fr/recherches/SAMOVA/FeaturesExtraction.html#me. I've read sample ...
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1answer
22 views

What are good estimates for phonemes length?

I am working in a phoneme recognition project, the language that I am using is "English". I have a tool that converts each written phrase in phonemes. Also I have an estimation of the time length of ...
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1answer
14 views

Recursively obtaining all-pole coefficients from complex cepstral coefficients?

This is a problem from the book Discrete-Time Speech Signal Processing by Thomas Quatieri, Chapter 6 Problem 13: Suppose $h[n]$ is all-pole (minimum-phase) of order $p$. Show that the all-pole ...
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15 views

LPC analysis without implicit windowing

In MATLAB, the documentation for the LPC function states: lpc uses the autocorrelation method of autoregressive (AR) modeling to find the filter coefficients. ...
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17 views

Recovering spectrum from vowel-sounds

I have this working: https://gist.github.com/p-i-/6e5ab73820b6368a331a The output looks like this: The basic algorithm is as follows: ...
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1answer
47 views

How do I efficiently process 20+ minute audio tracks? [closed]

I have built something that reads in audio files and processes the information for features. Sometimes the audio is rather long, and obviously I cannot keep all of this information in RAM. How can I ...
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2answers
76 views

If we FFT a signal with number of points far less than the length of the signal, the result is unreliable?

For example, there is a very long-term observation and I cut the signal at a sampling rate of 10 into many segments of 1e6-point ...
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20 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 ...
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19 views

Room Impulse Response for CHiME speech database

Where can I find the .wav file or a MATLAB plot for the Bin-aural Room Impulse Response(BRIR) that is used to create the CHiME speech database? http://spandh.dcs.shef.ac.uk/chime_challenge/data.html ...
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45 views

Speech signal, Word Boundary algorithm

I need to implement an algorithm in Matlab that, given a speech signal contaminated with background noise, can detect the boundary of each word, saving only the samples that belong to the actual ...
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34 views

Bandwidth of time-varying harmonics of speech

For speech signals, do the harmonics have a constant bandwidth above some threshold (arbitrarily, -6 dB dropoffs)? Or does this bandwidth decrease with the upper harmonics? If this is not known, how ...
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1answer
132 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
139 views

Phone conversation diarization with LIUM

I am trying split call-center recording by speakers. LIUM seems to fit great. But results look random to me (green is etalon, gray is what I get): So I think I am doing something wrong. Maybe ...
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43 views

Invertible multi-resolution representation of audio/speech signals

What are the invertible (preferably linear) representations of an audio/speech signal which captures the essence of signal in different resolutions and details. Basically, I am looking for ...
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26 views

how to cancel echo with adaptive filter, but without using reference

Ι am trying to cancel echo present in speech. The input is to a single microphone in an auditorium . I am trying with an adaptive filter and as per this method a far end signal provides a reference. ...
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1answer
100 views

Division by zero in Levinson–Durbin recursion?

I am doing a LPC analysis of a speech signal using the autocorrelation method. To calculate the LPCs I am using the Levinson–Durbin recursion. In my literature the error is initialized with the first ...
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39 views

How to detect impulse noises in audio recording?

How do you detect impulse noises in audio recordings of speech? By "impulse noise" I am thinking about noises like: Typing on a keyboard A door slamming Somebody bumping into a table Tapping on a ...
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39 views

How to get rid of speaker dependent information from MFCC

I'm trying to use MFCC vector features in order to compare the same words spoken by different persons. Is it enough to set a starting frequency for MFCC algorithm to 300 Hz, in order to get rid of ...
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1answer
20 views

Data representations for speech data

I'm interesting in knowing more about different kinds of data representations that are used for speech data. I know of FFT, but not any others. More precisely, I have the 3 following questions. What ...
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2answers
56 views

Problem in reading speech waveforms

I have two waveforms, and I'm not sure which of them is for the word "cease" and which one for "shoot". Now I guess based on voiced and unvoiced waveforms(voiced should be periodic), the first one is ...
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82 views

Which algorithms one can use for matrix inversion (for beamforming application MVDR/LCMV) in Fixed Point?

I have implemented beamformer(MVDR/LCMV) for speech enhancement in matlab, now I am trying to convert it to fixed point implementation. during the weight calculation step correlation matrix inversion ...
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1answer
50 views

What is the difference between neural network and deep neural network. an example thank you [closed]

What is the difference between neural network and deep neural network in speaker recognition .Can i have an example of code sur Matlab thank you
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123 views

Invert soundsc function in Matlab

Does anyone know what is the inverse function of soundsc in matlab. I'm playing a sound with a my PC microphone using soundsc, ...
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786 views

Simple VAD (Voice activity Detector) implementation

I am to classify few recorded phone calls as either containing speech or not by sampling lets say first 10 seconds. The files contain either silence, dialtone or ringtones, or real human voice. I ...
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69 views

History of emotion (or affect/prosody) recognition (or detection/analysis)?

I want to know how and why emotion recognition started. What I've come up with so far is an article from 1995: McGilloway, S., Cowie, R., & Douglas-Cowie, E. (1995). Prosodic signs of emotion in ...
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60 views

MATLAB band-stop filter design for room reverberation?

I took acoustic measurements of a room using sine swept method, and I also used Pioneer sc-37 to obtain more accurate data. I used RoomEQ wizard software and measured the impulse response of the ...
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2answers
103 views

How to take acoustic measurements of a clasroom

I'm an electrical engineering major and I have a question regarding a project I'm working on. I'm supposed to design 3 band-stop filters on MATLAB for echo cancellation/suppression. My problem is ...
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1answer
112 views

Markov processes

I am trying to read a paper and I found it difficult to understand. This is a quote: Standard Mel frequency cepstrum coefficient (MFCC) computation technique utilizes discrete cosine transform ...
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40 views

detecting change of speaker in audio file

how can we detect whether the active speaker has changed ( either stopped or someone else started talking or many people started talking together ). I dont need to know who it was ( speaker ...
3
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216 views

Detecting phones in a word audio

I am working on speech recognition for Hindi language. My approach is to map speech to text based on phones. For this, first split the speech into the words and then map each word to text by detecting ...
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457 views

removing vocal from audio file

I want to create a program in matlab that could act like karaoke software. At this moment i have following lines of code: ...
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169 views

autocorrelation A to K conversion for LPC speech analysis?

I am building an LPC analysis tool for speech synthesis and did the following to generate my a0 through a10: ...
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102 views

Speech recognition and comparison algorithms

I am doing several research for a prototype which I will be building. My prototype will consist of single word speech recognition and comparison with a preset recorded words. The prototyped system ...
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1answer
40 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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1answer
61 views

Phase Estimation in speech enhancement

I am reading a recent paper titled "Phase Estimation of Speech Enhancement - unimportant, important, or impossible?" by Grekmann et al. published in the 2012 the IEEE convention. The paper proposes an ...
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2answers
330 views

'A priori' and 'a posteriori' SNR

I am reading this paper about speech enhancement using a minimum mean square error short time spectral amplitude estimator for our speech enhancement project. Basically in the paper they mix a clean ...
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1answer
68 views

What is the distance metric in dynamic time warping?

I am reading the lecture slides from this site http://luthuli.cs.uiuc.edu/~daf/courses/CS-498-DAF-P/Lecture%2018%20-%20Time%20series,%20DTW.pdf In slide 24, the lecture suggest using metric of the ...
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79 views

Filtering noise and identifying phones boundary in speech recongition

I am planning to do a self project on speech recognition for a vocabulary detection system using arduino. I know CMUsphinx or bitvoicer is probably the way to go. But I would like to code the project ...
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87 views

Features for speaker recognition

I'm trying to create my own program for speaker identification (text-independent) and have used MFCC features for that. But result of recognition is very poor. My supervisor told me that using MFCC ...
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1answer
80 views

Variance of white noise and the STFT

I hope someone can help me understand the meaning of variance in this case. I am reading an old research paper by Ephraim and Malah (for my DSP project) and it says: Knowing that $D_k$ is the FFT ...
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19 views

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

This is part formula of command-response model (Fujisaki Model): ...
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6answers
205 views

Call recognition using Digital Signal Processing

We are looking for writing applications which would recognize Bird calls and Frog calls using Sound processing. EDIT: I want to identify the species of bird / frog and do not want to differentiate ...
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241 views

Dividing Audio signals into Barks Bands

I want to apply the following steps in matlab 1) Read the audio data 2) Apply window function on that data 3) Apply fft on the result of window function 4) Make power spectrum of fft 5) divide ...
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2answers
40 views

Re-recording audio vs. convoluting old audio data - differences?

I want to transform a studio audio dataset into a smartphone audio dataset (used for a speech recognition application). Would there be any difference in the resulting audio files using one of the ...
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1answer
39 views

Comparing signals recorded using diffrent methods

Suppose we have two signals( focus to audio) recorded with different methods, or more simply, suppose we have two audio signals recorded with different microphone, and we want to confirm that these ...
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1answer
80 views

The logarithm Energy measure in Speech

I am studying the mel-frequency cepstral coefficient (MFCC) and have downloaded a pdf which is related to MFCC from this link. It is about the "ETSI ES 201 108" ETSI ES 201 108 - Speech ...
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1answer
229 views

How do I construct input to neural network from audio signals?

Input: Microphone recordings of digits from 0 to 9 from different speakers. Output: The digit from 0 to 9. I am doing this for fun. So first I will train my neural network using some samples and ...