Questions tagged [audio]

Audio, or in terms of signal processing, an audio signal is an analog or digital representation of sound, typically as an electrical voltage.

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2answers
98 views

Phase information of a signal when creating a spectrogram

Sorry for the uninformed question but I need some help understanding this. I'm trying to understand what phase information is and it would be very helpful if someone could correct my understanding of ...
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2answers
988 views

How do I explain a complex exponential intuitively?

What is a complex exponential, explained intuitively? How do I explain to an adolescent a complex exponential function?
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1answer
180 views

Removing breaths using python

My wife is trying her hand at making audio books and spends a lot of time cleaning up some voice actors breaths in their "finished" audio. We have tried some filters in Adobe Audition and it helps but ...
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1answer
136 views

High resolution spectrograms

I have a recorded audio signal that I would like to convert into spectrogram with the highest possible yield of information. The audio clips I am working with is about 30 ms long and contains ...
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2answers
85 views

Advantages of an orthogonal filterbank

What are the advantages orthogonal filterbanks such as the DFT or DCT have over non-orthogonal filterbanks? I have heard quite often that they have attributes that are desirable when modifying the ...
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1answer
46 views

Loudness detection for natural conversation

I'm working on a project where I want to be able to analyze natural conversation as an audio file and determine the points in the conversation where the speech is louder than normal i.e. be able to ...
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0answers
95 views

Audio experiment with intermodulation by multiplying two analytic signals

1) First, I describe intended processing with a basic case: Let's say on left channel I have a sin wave with frequency F1, and constant amplitude A1. Then on right channel a sin wave with frequency ...
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1answer
53 views

Low-Pass FIR that keeps all values positive

Let's say, that I want to low-pass the norm (hence positive values) of an analytic signal (related to Hilbert transform). This allows me to perform experiments with the envelope of audio. For low-...
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1answer
53 views

Is the sum of all FFT bins from Audacity equal to RMS amplitude?

Say I open an audio segment with Audacity. Then I perform a size 4096+ FFT with a rectangular window. The program outputs bin magnitude values in dB. If I add them all together like this: sum = 10*...
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1answer
44 views

DFT of an audio signal stored as a multi-dimensional array

I'm using the Python scipy library to get the data of a particular .wav file into array format. Now, I'd like to find the Discrete Fourier Transform of that signal. The formula for the DFT is, $X_k = ...
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1answer
44 views

Re-mixing high and low pass filtered audio has interference in the shared transition band?

Disclaimer: this feels like something that should be obvious from an intro class, but for some reason I can't find anything around this. I'm working on an algorithm that takes a signal, let's call it ...
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1answer
46 views

What do these values in the freeverb diagram mean?

I am implementing a reverb based on the Freeverb implementation detailed here. There is also a more in-depth description of the LBCF in question here, but I am not able to find the information I want ...
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0answers
39 views

Dual FFT - difficulties extracting phase

I am performing dual FFT analysis on a system. 1) I generate a Log Sine Sweep signal and save it as my reference array 2) I then produce an inverse filter to correct for amplitude and save this as ...
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1answer
59 views

Sampling Audio Signal [closed]

I generate a 19kHz audio signal with sampling frequency of 44100 using Matlab. Due to the sampling, some other frequencies are generated in addition to the 19 kHz signal. When I play this audio, I ...
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1answer
28 views

How to find out that a wav file is edited or original in Python? [closed]

I want to verify that if a audio (.wav or any format) file is original or edited. How can it possible in python? Here, "edited" means modification of audio file using software. For example, for an ...
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2answers
58 views

Subtract audio to another audio

suppose I have a sample of a snare and that snare is used in a song,¿ there is an algorithm or tool that detects the sound of the snare and subtracts it, that is to say that cancels its phase or ...
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9 views

Why can ARSS not recreate audio from a spectrogram without sounding muffled?

Why do ARSS and all audio to spectrogram then back to audio program's create audio files that sound muffled? I know their is some physics and math with the short-time Fourier transform and etc. I ...
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0answers
28 views

What does KRT mean in this diagram?

I am working on building a reverb. This article explains how to build a reverb pretty well, but I do not understand what the KRT in the main diagram stands for. The article says ...
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4answers
176 views

What are some alternatives to Spectrogram charts? [closed]

Spectrogram renders of audio are limited in their representation of audio. For example, one could not use a Spectrogram to re-render the original audio. What are some alternatives to Spectrogram ...
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0answers
256 views

How to measure SINAD of an audio signal?

I tried to measure the SINAD and SNR of a recorded audio signal using MATLAB. The following is the process I did to measure the 2 parameters: Firstly, I generated an FM modulated signal(1 kHz audio ...
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1answer
32 views

From pressure to SPL : microphone calibration using sound level meter calibrator and sound level meter

I have a microphone + signal conditioner with adjustable gain and filter. I capture amplified signal using oscilloscope. I'm using this signal to compute equivalent continuous sound level With signal ...
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40 views

Generating a colored waveform of an audio file?

I would like to ask on what approaches could one attempt to generate a colored waveform out of an audio file. Below are two examples of waveform coloring by commercial products along an audio sample ...
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1answer
100 views

How to interpret these MP3 spectrums?

Recently, I found I had this song duplicated in my music library, but when listening to them I noticed one version was of noticeably higher quality than the other despite both claiming to have a ...
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0answers
36 views

How to crop a song and mantain the same values in the sample vector?

Does anyone know how can I crop an audio signal and maintain the same values in the sample vector? I am working on a audio signal processing project for song recognition in Python and I wanted to ...
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1answer
168 views

MFCC classification model

I have audio samples which MFCCs i want to train, but there is a problem. I can't find a classification model, because the samples have different length and consequently the MFCC matrices will also ...
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1answer
69 views

Distort voice to hide talker identity as much as possible

I need to develop an app which makes speech recognition. Because I need to be able to constantly improve the recognition capability, I would like to train the engine with real recordings and match the ...
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0answers
65 views

Additive Synthesis - clean playback of a spectrum frame

First of all - I have only a limited of Fourier transforms and the mathematics involved, so please pardon any potentially bad terminology or huge gaps in conceptual knowledge. I'm experimenting with ...
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42 views

Time gating of FFT for audio Measurements or Multi Time Windowing

I am learning more about DSP specifically for audio measurement purposes. I can an app, where I can create a log sine sweep from start frequency to end frequency over a certain length. I can perform ...
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1answer
40 views

How to separate silence from frequencies in Matlab

For a project, I was asked to create a DTMF detector on Matlab. I know that the audio sample that will be provided will have 4 tones maximum, possibly less, and that the audio begins with a 0.1 second ...
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0answers
61 views

Designing Matched Filters on Arduino

I'm trying to develop a prototype at the moment that includes a sensor to detect how fast someone is typing on a keyboard. Essentially this sensor is a microphone that will be able to detect keyboard ...
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2answers
261 views

audio spectrogram normalization

I am performing language classification from audio signals using mel-spectrograms as my inputs for a ResNet. It works well as long as all of my audio data from different languages is from the same ...
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0answers
29 views

What is the difference in the algorithm for noise reduction in SoX and Audacity?

I've looked at this question, and I understand that SoX works way FFT of the noise, then subtracting that from the FFT audio, and then resynthesizing using FFT, but I don't quite understand how this ...
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1answer
22 views

What are the ranges of values for each audio feature/descriptor?

I need to normalize some audio feature measurements I've done with the software Wolfram Mathematica. Coming from a non signal processing background, I have some difficulties understanding what ...
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0answers
12 views

Wavelet Analysis Magnitude Scale

I am trying to plot magnitude against frequency averaged over a time window. To do this I make a morlet wavelet Filterbank. And multiply the dft of each kernel by the dft of my incoming data set. ...
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1answer
27 views

Deriving difference equation

I have this block-diagram of Karplus-Strong basic algorithm. Where $$H_{d}(z) = \frac{1+z^{-1}}{2}$$ The problem occurs when I try to get the difference equation. As I can see from the diagram, that ...
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0answers
22 views

Is it possible to create a causal filter with a positively-sloped phase response?

I am currently prototyping a set of noise-cancelling headphones that utilize active EQ to modify the noise-cancelling signal, allowing the user to modify the frequencies that they hear or don't hear. ...
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23 views

Methodology for log some sweep audio measurement impulse response

I am trying to implement a method for measuring the response of a loudspeaker in a room using a log sine sweep stimulus. To check my method: 1. I create the stimulus using: Make recording length ...
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0answers
133 views

Implementing real-time variable fractional delay line in Matlab

I am attempting to implement a fractional delay line in Matlab which operates in real time, uses an object-oriented approach, and whose length can be modulated each time a sample is processed. I have ...
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0answers
28 views

Log Sine Sweep Acoustic Measurements

I am currently trying to understand the Sine Sweep Method for taking acoustic measurements. I am following this paper: https://www.thinksrs.com/downloads/pdfs/applicationnotes/SR1_SweptSine.pdf I ...
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2answers
118 views

Significance of signal energy in audio?

Probably a dumb question but I can't find the answer anywhere. It seems to me that the energy carried by a sound wave should be proportional to frequency squared. For example, the waves $x(t)=\sin(t)$,...
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1answer
286 views

Change durration/tempo of a .Wav file without introducing pops or pitch shift?

I am attempting to develop a similar function as Audacity's Effects>Change Tempo effect. Or more specifically: I am in need of a way to create a funciton in my ...
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1answer
109 views

Reading a Wav File [closed]

I need to read two .wav files. One of them is a song, and the other one is impulse response. Then, I have to convolve them. In fact, I have read them by using ...
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1answer
1k views

Bandpass filter for audio wav file

How do I apply a bandpass filter on an audio file which is in wav format? And write back the filtered data in a new wav file for further analysis.
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0answers
18 views

Interpreting results of multiplying signal and wavelets in the frequency domain

I have followed a lot of stuff on mike cohens website and have managed to make a filterbank of Morlet wavelets and convolve with an incoming block of samples. This of course is a computationally heave ...
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42 views

Speech processing using matlab

My teacher gave me a project speech processing using matlab. I've completed up to point 6 but I do not know anything about points 7 and 8. Is there anyone who can help me please
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34 views

Help in identifying low speed image transmission protocol over RF

In the seventies I worked for Muirhead on press fax machines that used to transmit pictures over radio and landline as I recall two FM systems were used a carrier 800 Hz modulated +- 150 Hz and 1900Hz ...
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1answer
42 views

Similarity in signal plots but dissimilarity in sensed content

I have two audio signals that look incredibly similar when plotted. However, when I play them, they sound quite different. The only difference I can even see in their plots is that one signal looks ...
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1answer
36 views

Why do AAC files seem to have an inexact bitrate?

I'm using ffmpeg to encode audio files as AAC at 256kbps. However, the actual bitrate of the resulting file is rarely 256. It's usually a number close to 256 like 241, 255, 257, 254. In one case it ...
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0answers
56 views

Detecting similar behaviour in audio files

Can anyone help, I'm trying to find a solution that would allow me the ability to look at a batch of .WAV files and detect ones which share a similar behaviour in the recording like the image attached....
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70 views

How enhance amplitude of audio signal without distortion?

After FFT I've set to zero some frequencies and I'd like to enhance the amplitude in order to recover the leak. I'd like to do a normalization in order to avoid distortion. I've done in this way but ...