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Questions tagged [speech-processing]

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

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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 ...
wsdzbm's user avatar
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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 ...
giogix's user avatar
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1 answer
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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 ...
panthyon's user avatar
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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 $H$ ...
shubham sharma's user avatar
2 votes
1 answer
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 ...
alexanderkuk's user avatar
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101 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/...
Soroush's user avatar
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1 answer
953 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 ...
Jumi's user avatar
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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 ...
pir's user avatar
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2 answers
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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 "...
Hanna's user avatar
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295 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 ...
Arpit Jain's user avatar
2 votes
1 answer
193 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
Makrem's user avatar
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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, ...
The Answer's user avatar
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2 answers
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 ...
user1064504's user avatar
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0 answers
151 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,...
engalex's user avatar
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2 answers
523 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 ...
engalex's user avatar
1 vote
1 answer
140 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)...
Celdor's user avatar
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1 answer
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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 ...
Gaurav Deshmukh's user avatar
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2 answers
5k 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: ...
all0star's user avatar
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1 answer
515 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: ...
patrick's user avatar
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1 vote
1 answer
700 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 ...
gurubondin's user avatar
0 votes
1 answer
147 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 ...
Mona's user avatar
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1 answer
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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 ...
Mona's user avatar
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2 votes
2 answers
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'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 ...
D.Zou's user avatar
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1 answer
511 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 ...
user4984's user avatar
1 vote
1 answer
288 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 ...
Kevin's user avatar
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1 vote
0 answers
165 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 ...
kobra's user avatar
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1 answer
404 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 ...
Mona's user avatar
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0 answers
28 views

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

This is part formula of command-response model (Fujisaki Model): ...
Ping Tang's user avatar
  • 101
1 vote
6 answers
468 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 ...
Raj Pawan Gumdal's user avatar
2 votes
1 answer
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 ...
Hassan Anwer's user avatar
0 votes
2 answers
53 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 ...
user2212461's user avatar
1 vote
1 answer
57 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 ...
jaseem's user avatar
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1 answer
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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 ...
user2874612's user avatar
7 votes
1 answer
2k 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 ...
Pratik Deoghare's user avatar
1 vote
1 answer
1k 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 ...
Surendra's user avatar
1 vote
1 answer
3k 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 ...
Surendra's user avatar
3 votes
2 answers
184 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})}...
varunkr's user avatar
  • 233
8 votes
2 answers
20k views

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 (...
varunkr's user avatar
  • 233
-1 votes
1 answer
182 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 ...
Sulayemaan's user avatar
1 vote
2 answers
2k 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 ...
ZijingWu's user avatar
  • 257
1 vote
0 answers
54 views

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 ...
HighAsAKiteOnMath's user avatar
2 votes
2 answers
166 views

What do cloud-based speech-to-text services upload to the cloud?

Services like Siri and Google Now perform speech recognition (speech-to-text) on mobile phones by doing part of the computation on a server in the cloud. What do they upload to the cloud? Do they ...
D.W.'s user avatar
  • 175
2 votes
0 answers
200 views

Speech features that can be inverted?

I am trying to figure out speech features that can be inverted to recover the audio back with no loss or as less loss as possible. I am currently using the power spectrum which is completely ...
nishGladiator's user avatar
0 votes
1 answer
706 views

Is it necessary to taper two ends before fft a time series?

I saw in some codes people always taper the ends of a signal using a cosine taper. Is that necessary and why?
wsdzbm's user avatar
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1 vote
1 answer
2k views

Log energy calculation

I have some code here that I don't quite understand. It calculates the energy for a given frame the following way: En=log(1+energy(32767*frame)) The energy ...
Peter James's user avatar
0 votes
1 answer
217 views

Good methods for hardware based 1-2 word speaker independent speech recognition?

I want to detect, in real-time, a 1-2 word spoken phrase within continuous speech, speaker independently, in hardware. The smaller the number of gates the better. What is the best way to do this? Are ...
user58109's user avatar
0 votes
2 answers
128 views

is there a mathematical relation between percent overlap and window length or amplitude

I am trying to plot short time energy of an utterance with percent overlap as an input variable. As I referred to the short time energy equation it can be represented as square summation of $x(n)*w(n-...
Engtyma's user avatar
0 votes
0 answers
52 views

Time of sound-signal

Let's say I have a sound file, I then compute the Spectrogram and segment each part of it. Assume that it is a recording of someone saying the word "Hello" and therefore, each piece is given like so: ...
Phorce's user avatar
  • 520
6 votes
1 answer
4k views

Voice Audio Detection algorithm

I have to detect speech intervals in large pre-recorded files in my project. I think, there won't be so much noise in the background (audio will be recordered in the room or even in the studio), but ...
emptysamurai's user avatar
-1 votes
2 answers
249 views

How to check the quality of an audio through an algorithm or software program

Can you please help me with this problem?? I have been working on AM335x sitara evm and my project needs self diagnosis of the captured audio from the mic through the speaker on the same board (we ...
user3292664's user avatar