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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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1answer
59 views

Silence removal from the audio

Suppose i've a mp3 file. I performed FFT of that mp3 file. Then is it good to state the following: 1. Silence occurs by that particular DFT point whose magnitude is lesser than some threshold. 2. ...
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2answers
37 views

Distinguishing Spoken Speech from Played-back Recording of Speech

Do any general-purpose techniques exist to distinguish live spoken speech from played-back speech samples? For example, if I were to deploy a speaker verification system to authenticate users by voice,...
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1answer
216 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 ...
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0answers
12 views

Stack features in wakeword model

I'm not clear how is the input features are stacked in wakeword applications. Taking https://static.googleusercontent.com/media/research.google.com/en//pubs/archive/43969.pdf as an example, the author ...
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1answer
107 views

Speech dereverbaration via maximum kurtosis adaptive filtering

I'm trying to code the algorithm described in Speech dereverbaration via maximum-kurtosis subband adaptive filtering by Gillespie, Malvar and Florencio, and the signal looks cleaner in when I plot it. ...
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0answers
428 views

How to overcome different MFCC coefficients distributions from different speech datasets?

I am using MFCC values as features for a machine learning model of speech, detecting age from a voice recording of a person. I work with voice datasets I found on the web: common-voice and vctk. ...
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1answer
23 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 ...
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1answer
35 views

Cone of Confusion

In sound perception, humans adapt by tilting the head or relying on other location cues such as visual input when the source of the sound lies on the cone of confusion. On this cone, no time and level ...
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0answers
12 views

How to use DTW for keyword spotting in human speech?

I am trying to use Dynamic Time Wrapping (DTW) + KNN to detect word occurrences in human speech audio. I am to implement the same approach suggested in this DSP StackExchange post, but I am seeing ...
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2answers
610 views

Quantify the difference between two audio spectrograms

I have two speech signals of the same length. They contain the exact same speech content but are recorded from two different microphones. I want to compare the differences between them. 1) Assuming ...
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1answer
181 views

How to prevent Octave Jumping in ACF of Speech Signal?

I'm working with speech signals and my aim is to estimate the fundamental frequency $\ F_0$ of this signal often called as "pitch". The main idea is taking small blocks of the speech signal such ...
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24 views

Objective Noise Classification for speech enhancement

I'm working on my thesis and it is about an speech enhancement system to use proper noise reduction method/algorithm for different types of noise. I want to test and compare some filters/algorithms ...
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2answers
335 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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1answer
302 views

Dynamic Time Warp through SPTK tool

I'm an IT student and got an assignment to do about Dynamic Time Warping(DTW) using the Speech Signal Processing Toolkit (SPTK) and comparing some words spoken by 2 speakers and finding the ...
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1answer
126 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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2answers
282 views

Long term spectral flatness measure voice activity detection

This is the Long term spectral flatness measure code that I have written so far. Can someone please help me, I don't know if I'm writing it correctly. It is working fine and gives high peak for ...
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4answers
86 views

Cleaning telephone audio

I have telephone audio at 16kHz and I am trying to use it with a speech transcription engine that converts spoken letters to text. However, the engine has some difficulty transcribing certain letters ...
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1answer
143 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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2answers
3k views

What are some python packages I can use to cut audio files

I have a project in which I have a batch of audio files and I need to remove the audio in it from say time 2secs to 5secs i.e. I have a set duration to be removed. How can I do this in python?
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2answers
62 views

Modelling Unwanted Signal in a LMS Adaptive Filter

I'm having some confusion learning about the LMS Adaptive Filter. I understand that the whole model of adaptive filters relies on the fact that we give it a reference signal to which it keeps ...
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1answer
115 views

audio filter designing

I know the basic filtering techniques available in signal processing.Does audio signal specifically have any different filtering method?.My application is to filter out the background noise in speech ...
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1answer
317 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 ...
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1answer
862 views

Real time background noise reduction or removal

I am trying to build a real time simple audio recognition that can recognize few keywords. Following this tensor flow tutorial i was able to train a simple neural network. Currently the signal flow ...
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1answer
42 views

Speech Processing applications: stereo data for features

I am working on speech databases that have a stereo format. I want to extract spectral information (Mel-filterbanks, MFCC, LPCC) and also some other prosody features like the fundamental Frequency F0. ...
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1answer
383 views

MFCC Classification

I'm working on gender estimation from speech signal and I completed MFCC feature extraction. So now I'm trying to estimate gender from these features. But I have frames for an audio file and I ...
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2answers
262 views

Harmonics to Noise Ratio Estimation

I'm willing to estimate the Harmonics to Noise Ratio (HNR) of a speech signal x[k] and using autocorrelation method. Theoretically, HNR is given as, $\ HNR = \frac{R_{xx}[T_0] }{R_{xx}[0]-R_{xx}[...
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1answer
71 views

Difference between applying a window to a whole signal and frames.

I'm going to do feature extraction using a speech signal. The first step is the windowing of the signal using Hamming window. I'd like to know if I should apply one single window to the whole ...
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1answer
67 views

Why is the Hamming and Hanning window the most widely used for speech spectograms?

When taking the Short-Time Fourier Transform of the speech signal, windowing is applied to taper away the signal frame at both ends and mitigate introduced artifact at expense of the maximum ...
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1answer
33 views

Need some references on sound quality evaluation and syncing

I have to do a project where I'm receiving the same sound signal from different channels (with different offsets), but they may not be consistent all the time. At any given time, I should decide which ...
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1answer
104 views

How to reduce synthesis artifacts produced by phase vocoder?

I implemented the phase vocoder algorithm in Python to time-stretch speech signals by following this paper and referring to this MATLAB tutorial. I can distinguish words in the original signal from ...
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1answer
45 views

Audio Processing Glossary

I'm new to audio processing and I hope here be a good place to ask about the definition of terms in this context. My question is what is difference between sound, audio, voice, speech and tone, from ...
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1answer
509 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: $$\...
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1answer
82 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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3answers
560 views

Signal processing for audio and speech

I have started learning about signals and I am interested in sound signals. There are some questions that I need to resolve. For example in music, difference of notes like 'sol' and 'la' are about ...
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1answer
20 views

Are delta and double delta features needed when classifying with LSTM

I understand that the combination of MFCCs, deltas and double deltas is a good feature to be used with HMMs for keyword detection problems. HMMs are limited by Markov Property and this limitation is ...
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2answers
313 views

How to edit audios so that they have the same length

I have a dataset of audio files, which differ on their lengths by a couple of milliseconds. Some are little shorter than 1s and some are little longer than 1s. They all have the same sampling rate, ...
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1answer
126 views

Mel separated filter banks

I am currently a bit confused. I am currently trying to understand the MFCC extraction process, and don't understand how the Mel filter banks only provide a scalar output from each filter bank. What ...
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1answer
36 views

how does the Frequency spectrum of a signal change in the logarithmic space?

I am trying to implement basic homomorphic filtering but I can't seem to understand what happens in the frequency domain when you take the natural logarithm of the function. In Matlab I'm getting not ...
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1answer
39 views

Is there any way that we can perform speech recognition without using Fourier transforms?

I am trying to research about speech recognition and why everyone uses Fourier transforms in going about the topic. I know that we get information related to the frequency of each sound uttered which ...
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0answers
15 views

How to perform a speaker normalization of features before model training?

I am building a model using a supervised machine learning based on features I extract from speech signals. The features include MFCC, auto correlation and energy derivatives. According to this paper,...
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1answer
33 views

Relationship between Frequency response, magnitude spectrum and gain

I have been reading quite a lot of DSP related materials online and offline (books etc.) lately. I come across the terms frequency response and ...
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1answer
97 views

Split audio file into audio phrases

I have couple of audio recordings of length 5 -7 min. I also have corresponding text to each audio file. I am not subject matter expert on speech to text but I am vaguely aware that much smaller ...
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2answers
38 views

MATLAB - How do I get the Filter Function to get the output Energy Spectrum?

The image was from a question I posted on reddit, so ignore the title. I'm trying to write a program that plots these graphs I know that the first plot is just a plot of the source-signal so the "...
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2answers
137 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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1answer
471 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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0answers
16 views

Analysis by synthesis in speech coding

I am trying to understand the analysis by synthesis in speech coding. In wikipedia it is mentioned for the CELP encoder: The main principle behind CELP is called Analysis-by-Synthesis (AbS) and ...
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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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0answers
149 views

Language detection from audio

I'm trying to do a simple test on language detection on a video input. Basically, given a .mp4 file, I need to process the audio tracks (can have primary and ...
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3answers
80 views

How does real-time signal processing work when the signal we deal with is assumed to be known for all times?

One thing that never sat right with me after my signal processing course is that in order to do signal processing, we must know the entire signal $x(t)$. For instance, I cannot take the Fourier or Z ...
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0answers
45 views

How can I quantify the distortion in a speech signal?

I am working with certain speech enhancement algorithms, for instance to remove stationary noise or impulse noise from a speech signal. Filters such as those can enhance the speech signal, but they ...