Questions tagged [speech-processing]

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

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2answers
313 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 ...
1answer
144 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 ...
0answers
910 views

Pre-emphasizing in speech recognition

Usually, in speech recognition, the techniques that are used are based on the linear prediction model (Fant, 1960). Parallel to this, the human speech production mechanism causes energy to drop ...
0answers
9k 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 ...
1answer
67 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 ...
0answers
220 views

What are F0 counters in Speech signals? What is represented by F0?

I can't find any dedicated definition for Frequency F0 counters. Can anyone tell me what does F0 exactly mean, I am confused about that.
1answer
330 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 ...
0answers
115 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 ...
0answers
94 views

2answers
65 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 ...
0answers
335 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. ...
1answer
609 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 ...
0answers
624 views

Python scipy.signal.wiener filter for speech processing

As far as I know Wiener filter can be used in Image processing.I wonder that if i can use Python scipy.signal.wiener() for speech processing as well. Thanks in advance.
2answers
238 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}[...
2answers
470 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 ...
1answer
282 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 ...
0answers
154 views

What is advantage of Logarithmic Sampling?

Logarithmic Fourier Transformation This paper starts off in the abstract saying that the given audio signal is sampled logarithmically. We also know that the human ear has a sort of a filter bank ...
0answers
144 views

What is logarithmic mapping in a filter bank such as the one in the human ear?

"The spacing of the filters in a filter bank used to make an ordinary spectrogram is linear, whereas an auditory filter bank has a spacing that corresponds to the way in which sinusoidal frequency ...
0answers
104 views

difference between logged power spectrum and power spectrum?

What is the difference between a logged power spectrum and power spectrum... Well the log part. So can a normal power spectrum be converted to a logged power spectrum just by taking the log of the ...
0answers
51 views

Finding the closest match of a recorded word to a 'corpus' of singly-recorded words.

I am interested in learning more about processing audio and thought that it will be helpful to learn while doing a project. The project involves a pseudo-speec recognition where I have a corpus of ...
0answers
146 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 ...
0answers
425 views

audio features extraction using restricted boltzmann machine

I want to extract Audio Features using RBM (Restricted Boltzmann Machine). For this, I am giving the spectrogram (PCA whitened) as an input to the RBM. For each audio file, The spectrogram is a matrix ...
0answers
184 views

Reducing frequency of a signal

I have a signal that I want to make it sound like a whisper. This is probably better than using a voice conversion system, but I was just want to do some experimenting before. I was thinking maybe ...
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 ...
1answer
13 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 ...
1answer
41 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. ...
0answers
12 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,...
1answer
16 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 ...
0answers
13 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 ...
0answers
47 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 ...
0answers
33 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 ...
2answers
2k 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?
1answer
39 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. ...
1answer
64 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 ...
1answer
105 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 ...
1answer
127 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 ...
1answer
194 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 ...
1answer
279 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 ...
0answers
409 views

The essential pre-processing steps in speech signal analysis

My question is that for simulating a new scheme for speech emotion recognition (feature extraction and classification), what are the necessary pre-processing steps (I mean any kind of normalization ...
0answers
149 views

How to deal with variations in amplitudes when doing voiced/silence detection?

I have a collection of wave files that has a few well separated words spoken. My goal is to split them into individual files. I wanted the program to work on all files, not just one of them or require ...
2answers
267 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 ...
0answers
59 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 ...
0answers
66 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/...
0answers
194 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 ...
0answers
142 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,...
1answer
124 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 ...
0answers
26 views

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

This is part formula of command-response model (Fujisaki Model)： ...