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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30 views

Generating intermodulation between two channels, using analytic signals

1) First, I describe intended audio effect with a basic case: Let's say on left channel I have a sin wave with frequency F1, and amplitude A1. Then on right channel a sin wave with frequency F2 and ...
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40 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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35 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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30 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
29 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
38 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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19 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
53 views

Sampling Audio Signal

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
23 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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1answer
24 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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4 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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24 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
50 views

Alternatives to Spectrogram [closed]

What is a more accurate alternative to a spectrogram that can be represented in an image? Also, what is a python script/exe that can be used to convert audio into the image format then back from image ...
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44 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
22 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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24 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
49 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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34 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
71 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
42 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
49 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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22 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
31 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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40 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
70 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
21 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
16 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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8 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
26 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
19 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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15 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
47 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
20 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
82 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
86 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
43 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
319 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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13 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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38 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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32 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
31 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
32 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
40 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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63 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 ...
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1answer
35 views

Generating audio clip by superposing two files

I'm currently trying to generate a set of audio files where some audio events (e.g. dog bark) are immersed in a background audio scene (e.g. crowd noise). I understand that if the event would have ...
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1answer
72 views

Formant Analysis Frequency - What makes a peak value?

from spectrograms, how can you tell which peak is really a formant. Like why is the pink arrow the first formant. And in the spectrogram below, which has fewer samples, what makes a value the peak ...
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43 views

Finding the input from the impulse response and output

I have $y,h,x$ which are all vectors. From $y[n]=x[n]*h[n]$ which is basically how I got $y[n]$. I also know $h[n]$. I put this through a Fourier transform. Let's assume that the capitalized ...
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2answers
65 views

Multiple different length Windows for high resolution realtime FFT

I am looking at trying to achieve a 'realtime' (as quick as possible to acquire, process and present data) FFT application to analyse audio. I have setup my app to acquire a set of samples, apply a ...
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1answer
72 views

Hilbert Transformer processing block by block

I am processing audio and trying to obtain a 90º phase shift when processing block by block. I have implemented a Hilbert Transformer (from Lyon's 3rd. Ed. & MATLAB). MATLAB code below (I believe ...
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1answer
31 views

audio analog input + usb data cable = sound noises [closed]

I connect my android smartphone from audio jack output to my pc audio input and get a good analog sound signal. But when I connect this smartphone to the pc with usb data cable the analog sound ...