Questions tagged [speech-processing]

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

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Answering Machine vs. Human — Neural Network Features Selection

I've been tasked with creating an artificial neural network that can classify a telephone call as either answered by a human or an answering machine. My knowledge of audio processing is, mildly put, ...
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2k views

how to extract pitch contour of a speech by matlab?

I am totally new in Matlab. Can anyone please explain in details how I can extract pitch contour of a speech? I would like to observe how the speaker's pitch varies during his speech to know whether ...
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3answers
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Speech corpora for VAD

I'm doing a student research project connected with VAD. And I want to find speech corpora with added noise. I found many references to the NIST Speaker Recognition Corpora. But unfortunately, all ...
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1answer
1k views

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 I have other features to look at other than MFCCs, the way I would ...
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1answer
209 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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1answer
142 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
166 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
177 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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1answer
57 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
1k 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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0answers
61 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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1answer
61 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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2answers
1k 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
2k 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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70 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 Laplacian/...
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1answer
772 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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1answer
32 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
143 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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226 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
179 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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1answer
772 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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2answers
4k 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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0answers
145 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 room,...
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2answers
495 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
133 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 (DCT)...
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1answer
2k 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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2answers
4k 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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1answer
469 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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1answer
649 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
135 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
158 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
3k 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
496 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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1answer
274 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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152 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
289 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 of ...
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26 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
448 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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1answer
1k 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
51 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
56 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
1k 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
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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 ...
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1answer
928 views

What is best practice to remove static noise?

What is best practice/algorithms to remove static noise from speech ? static noise aka Stationary Signal/static noise due to 1.humming sound due to of telephonic line/channel, 2.moving Vehicle,or ...
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1answer
2k views

In framing of audio samples', what is need of frame shift while giving frame size??

In framing of audio samples' in audio feature extraction, what is need of frame shift while giving frame size?? i.e. frame size = 20ms, frame shift = 10ms. Rather then shifting/overlapping why ...
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2answers
143 views

Confusion Regarding Bi Linear Transform

I was reading my book where the z-transform of a signal is derived to be ${1-e^{-2bT}z^{-1}}$ . Then it goes on to say that by applying the bilinear transform we can get $$\frac{2(1+bT+(bT-1)z^{-1})}...
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2answers
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Calculation of Reverberation Time (RT60) from the Impulse Response

I have some confusions regarding reverberation time (RT60). I need to calculate reverberation time from a given power envelope. This is what I get. As you might see the line is curved close to zero (...
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1answer
120 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 ...
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2answers
1k views

When should we using moving average in algorithm design?

I'm new to signal processing. And I just read G.720.1. I find there is a lot of moving average used. And I can also recall there is a lot of moving average been used in other audio processing ...
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I'm interested in the state-of-the-art of vocal sound synthesis. Please advise me [closed]

Some of the things I'd like to accomplish: Strip the vocals from a recorded song as best as possible Recreate someone's song voice (like a famous artists') using someone else's voice as input Adjust ...