Questions tagged [audio-processing]

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27 views

How can i solve .edf.seizure file error? [closed]

I need to read chb01_03.edf.seizures file from CHB MIT database in MATLAB for seizure detection. i execute this command in matlab ...
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2answers
43 views

Separation of sinusoidal oscillations from the frequency spectrum

It is necessary to extract the harmonic signal from the audio spectrum. That is, from the spectrum of 20 ... 20,000 Hz it is necessary to select only a signal with a frequency of 1,000 Hz, and ...
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20 views

How can i read .edf.seizures format in MATLAB?

I need to read chb01_03.edf.seizures file from CHB MIT database in MATLAB for seizure detection
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2answers
57 views

DSP signal filtering and number formats

I'm working on a dsPIC33, using Audio Codec Board - Proto to read audio samples and for reproduction. In order to implement a relatively simple signal processing algorithm, I'm reading audio samples ...
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0answers
18 views

The best method to convert piano sound to music notes in real time [duplicate]

I was looking for a way to convert piano sounds to piano notes for my research. I discovered the conversion of piano sound to piano notes using FFT and piano note recognition using CQT. I am also a ...
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29 views

Possibilities in Audio Convolution Math

New Python user here! I am a musician working on a program to streamline a process of using convolution math on a folder of .wav samples. The idea is that each audio sample will be multiplied with ...
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1answer
31 views

How to find peak frequency in an audio frequency spectrum?

I have a audio file consisting of multiple frequencies, I need to find all the frequency peaks in the frequency spectrum after doing FFT. But the issue is how can I be able to set the threshold line ...
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1answer
34 views

If video uses color bars to test and align color values among TV sets, what could be the equivalent in Audio?

Are there any sample Audio files, encoded in the simplest format known, that can mathematically explain or illustrate the fundamentals of (audio) signal processing that might also relate to known ...
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2answers
184 views

Can Principal Component Analysis (PCA) Solve the Cocktail Party Problem?

I'm looking into the cocktail party problem and trying to figure out whether something like Principal Component Analysis is enough to separate out all the various voices at the cocktail party into its ...
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1answer
51 views

Do i need to remove mean in STFT when working on music audio?

i need to apply a STFT to some music audio files but i can't understand if i should remove the mean or not. I know that for stochastic signals that is a must and that sometimes also necessary for ...
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2answers
54 views

Comparing two small audio files (e.g., hand clap) Android

Dears, I am new in signal processing and I am dealing with an Android application that should do some audio files (having the same "small" duration) comparison in real time. My main focus is ...
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19 views

Understanding CQT (Constant-Q Transformation) parameters for piano AMR (automatic music recognition) with Librosa Library

I am doing an automatic music recognition project with a deep learning model. For my data preprocessing, I am trying to calculate the Constant Q Transform for Polyphonic 88-key piano audio using ...
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40 views

Manipulating IO from an audio interface

So this is a relatively complex question I’m sure, as I have no idea what I’m getting myself into here. The gist of the project I would like to undertake is that I want to turn an old AIO PC (computer ...
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1answer
64 views

Determining statistically the bandwidth of a signal from a spectrogram

My question is how to determine the parameters of a signal from a spectrogram statistically. The most important parameters that I am looking for are the lower and higher frequencies of the signal. I ...
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22 views

Audio RMS normalization prior to a DNN classification

I want to normalize my data prior to a deep neural network model (DNN). It was recommended to me to use real mean square (RMS) normalization for audio, though I am not sure this is the best for a DNN. ...
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1answer
31 views

Most generally-accepted audio term, for “pre” and “post” lengths of a symmetric filter's impulse response?

Consider a FIR audio filter with symmetric impulse response. In my case, the filter kernel width is 15 frames on each side (so including the center frame, 31 frames total). When examining the audio ...
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2answers
76 views

Scipy Butterworth highpass filter results in distortion and bloated output

Trying to use Scipy Butterworth filter to filter frequencies below 54 Hz from audio files. Audio files are stored as WAVs. Sample rate is 44100. When I try using this filter the result is distorted (...
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0answers
48 views

Reading Waveform from microphone - signal shape problem

I read 2kHz sinusoidal signal that I generate using youtube and mobile phone by shotgun microphone and sound card in my laptop. Results (python): But why are the values zero at the beginning and then ...
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0answers
16 views

Which audio features to use for content based music recommendation

I am attempting to write a content-based music recommendation system using machine learning. Using a python library, I am able to extract the features from raw audio files. For each audio file, I have ...
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5answers
124 views

Effect of One Sample Latency in Audio Application

I am dealing with a customized processor that has a processing latency of one sample (20.8 microseconds) at 48kHz Sampling rate . What I would like to know is ...does this latency of one sample have ...
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36 views

Newbie want to self-study to make guitar effect/amp plugin

I am a senior Game developer want to self-study how to create electric guitar amp/effect plugin. I read around and managed to made a simple distortion plugin using JUCE. But because of lacking in ...
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0answers
26 views

Communicate to user that SNR is too low - Speech procesing

I am working in a project to estimate the signal to noise ratio (SNR) for an automatic speech recognition system in order to generate an alert to the user if the noise is too high in the place where ...
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1answer
22 views

Dimension error using findpeaks function in Matlab on Power Spectral Density to calculate SNR of audio data

I have a set of audio files for which I would like to determine a SNR metric. I tried to implement in Matlab as follows: ...
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0answers
26 views

What is Ad-hoc microphone array?

I have stumbled upon this term in many papers, though no one bothers to elaborate. Is this the case where I do not have prior knowledge of the array structure? or otherwise is it the number of ...
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1answer
50 views

Which codec and format are Audible downloads?

Files downloaded with the Audible Download Manager are stored with the .aax file extension, but the information I can find about this format seem to vary. Regarding this source, the files are stored ...
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3answers
116 views

Simulating room impulse response between rooms

I want to simulate a sound source in one room and a microphone in a second, adjacent room. I am familiar with the image method and Habet's very efficient implementation for room impulse response (RIR)....
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25 views

How to identify the vocal folds in a mel spectrogram image?

I am quite new in the dsp and trying to learn How to identify the vocal folds in a mel spectrogram image? My image data looks like this. How to find the vocal folds in this image? I am using Praat ...
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1answer
60 views

FFT results are repeating?

I am trying to detect notes out of a musical sound input. My program can be run here. The problem is the patterns in the frequency response is repeating: I suspect that echoing in the room is causing ...
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0answers
26 views

Expand and shrink waveform visualization depending on tempo changes

I am not quite sure if this is the right state exchange site or not. Pls let me know. So, I noticed that Logic and Pro Tools, etc. have this waveform shrinking and expanding effect when dragging the ...
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0answers
30 views

Is it possible to open 2 streams from the same microphone on Oboe?

I'm trying to open 2 input streams from the same microphone on Oboe but it's not working. One of the audio inputs I'm opening is like this: ...
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3answers
64 views

Sampling a low-frequency sound doesn't work at some sampling rates

When sampling a low-frequency sound (like a 55Hz sine wave) I hear an additional high frequency sound at certain sampling rates, but not at others. This happens in Audacity and in my own program. For ...
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1answer
35 views

How to use MFCC (or any other tool/method/algorithm) for speech verification

I have an original speech .wav file and a few recorded samples of the same file. Each recorded version has some differences wrt the original file. For example, The ...
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0answers
51 views

How do I extract n-dimensional cepstral coefficients with a sliding Hamming window in PyTorch?

I am trying to replicate the features from this paper, described as follows: MFCCs of speech frames are computed using a mel-scale filterbank. We extract 20-dimensional cepstral coefficients with a ...
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1answer
60 views

Actual Sample Rate slightly off?

When capturing microphone audio, I noticed that my static buffer eventually overflows. So I timed the samples, and low and behold I'm getting an actual sample rate of 44102-ish Hz (my timer is very ...
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0answers
21 views

Pitch analysis Audio Unit [duplicate]

I am developing an audio application for macOS that I would like to use to analyse the notes in a given section of music. I am happy to work from an analysis of the frequencies in a given section then ...
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0answers
23 views

In audio analysis, is the first rahmonic from the cepstrum equivalent to the first partial of a pitched sound?

I was reading this text from Oppenheim and Schafer, briefly explaining the concept of the cepstrum. I'm new to this concept. He explains that the cepstrum is useful to quantize the distance between '...
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1answer
27 views

Order of operations when creating a spectrogram

Signals noob here! I'm dipping my toes into your world by trying to recreate the Shazam algorithm, and I have a quick practical question: When creating a spectrogram, is it better to take in a whole ...
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0answers
62 views

Frame lookup using audio

I'm an animator trying to get a program that will use machine learning to automate the lip syncing process. The program I need made will have two files given to it: a seed, and an audio file. The seed ...
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1answer
82 views

How to do Frequency Scaling on an Audio File

Please excuse me if my terminology is wrong. I'm from a music production background and have no experience in signal processing. I was wondering if it was possible to stretch out the overtones (...
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0answers
219 views

How do I change the sampling rate without changing the speed and pitch in Python?

I am building a voice recognition algorithm, and the use case specifies that it should be able to identify the targeted speaker from various audio sources, which means that it must be able to process ...
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0answers
20 views

MPEG Audio Constant bit rate conversion

I am trying to convert few .wav files to .mp3 format The desired .mp3 format is : I tried with FFmpeg with this code : ...
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1answer
39 views

Continuously changing pitch of stable-pitch audio sample

I have an 44100-rate 4 second sample of a legato violin playing at 440 Hz. I would like to modulate this sample so that it starts at 440 Hz, then transforms its frequency continuously and ...
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5answers
235 views

Algorithms to re-spatialize a stereo recording audio signal?

Let's say we have an audio signal stereo recording, made for example with a XY microphone positioning: Which algorithms can be used to "re-spatialize" this recording, i.e. try to virtually &...
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1answer
55 views

What is the purpose of dividing an audio signal into segments and analysis each segment?

I read bunch of materials for extracting feature from audio signal and they all tell me to break signal into segments, why don't we analyze all the audio signal? I don't know what are the advantages ...
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0answers
63 views

How to correctly pass filter states from one processing block to the next

I'm processing a 4 channel audio signal and due to memory requirements need to process the signal in an arbritrary number of blocks. To ensure consistent filtering between blocks I've been told I need ...
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1answer
96 views

What is the actual format of this audio file?

I am collecting some data from jotform .wav widget. All the audio files are in .wav format. But when I tried to open the file using python ...
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1answer
70 views

finding onset of an impulse signal, basics?

New to signal processing, but making good progress, I think. I have a series of (many) impulses I generated, which will be used as impulse responses to model our church's acoustics, in this time of ...
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1answer
79 views

Popping sound after using bandpass filter

I'm trying to create guitar Wah Wah effect (basically band pass filter with dynamic frequency range). I'm using Steinberg SDK for it. I refactored this code to work in real time and it looks like this:...
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0answers
19 views

Perceptual Based Adversarial Audio Attacks

I am reading this paper Perceptual Based Adversarial Audio Attacks I have a doubt what is meant by weighting in the frequency domain and what does Section 3.4-equation-4 wants to convey .Please refer ...
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1answer
119 views

Comparison of results with Direct form1 and Direct form2 implementation of IIR filters

The results obtained from Direct form1 and Direct form2 implementations (in C language) of IIR (BiQuad) filters are a little different. In the implementation, i have made the b0 coefficient as 1. So, ...