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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Finding frequencies(and bandwidth) that works best audio in a certain medium

Complete DSP and coding newbie(learning python). I am trying to find frequencies that can pass through well in a musical instrument tube. I am trying to find audio frequencies(audible and inaudible ...
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13 views

Can I see if the audio is recognizable by google speech API using FFT

I want to check if the audio recorded by a user will be recognizable with Google Speech API using the Fast Fourier Transform before using the API. I don't know if it will possible using the FFT, and ...
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130 views

Low-Pass Filtering on Small Chunks of Audio

I've got a script (Python) that chops 44.1 kHz realtime audio into 10 ms chunks, processes each chunk, and stitches the chunks back together with no added artifacts. I'm trying to create a low-pass ...
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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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28 views

What determines peaks in FFT?

I ran FFT on three audio files and found that the results for some have more peaks than the other. Could anyone give me any conceptual explanation as to what determines these peaks? Below are plots of ...
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2answers
59 views

Parallel equalizer, working for boosts but not cuts

I am trying to replicate the behaviour of a parallel equalizer. The approach is to use a parallel sum of bandpass filters like so: $Response = 1 + bandpass1 * (gain1 - 1) + bandpass2 * (gain2 - 1) + ...
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13 views

Hardware that Mixes Two Digital Audio Source into One

Is there any off-the-shelf hardware that extracts two digital audio streaming source from two digital audio stream(like USB+SPDIF), mixes, and transmits in single digital stream real-time(if the ...
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10 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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44 views

How to compute key of a song?

I'm looking for a precise way to compute the key of a song. BPM would be cool too. Analog is ok. I tried to approach it from the DFT angle, but got stuck and can't really put a finger on what's the ...
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32 views

audio processing for audio files quick listening

I need to speed the listening task. I want to listen only 1 second every three seconds. Could you give me some advice? Many thanks! Best regards, Sergio
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1answer
52 views

Biquad high-pass filter not working as expected

I'm trying to implement a high-pass filter using the the coefficients described in RBJ's well-known EQ cookbook (see https://www.w3.org/2011/audio/audio-eq-cookbook.html). However, when I run white ...
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43 views

Preprocess sound for voice recognition in Pocket Sphinx on Raspberry Pi 3

I want to improve my voice recognition system on a Raspberry Pi 3 running Pocket Spinx. It's pretty accurate (limited recognition vocabulary) but is terrible when other noise is around. Is there any ...
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15 views

Sudden loudness changes producing unwanted noises?

What's that effect called? Is that one of the reasons for having attack and release parameters when implementing synths? And is there a generic method to avoid such sound artifacts? Example: I have ...
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53 views

How to reconstruct a sound from magnitude spectrogram?

I have an audio magnitude spectrogram but I don't have the phase, try to randomize the phases of each container and then make a reverse fourier, but only pure noise is heard How can I reconstruct the ...
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2answers
30 views

Can I slow down an audio file by halving the sampling rate?

A question came up, and as a layman I am not sure how to appropriately find an answer; bear with me. Given a 2 minute audio file, say a .wav file sampled at 44,100 Hz, I am curious if it is possible ...
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26 views

Granular synthesis of a periodic waveform - How to derive grain start and end point

I understand to find the length of the grain I need to perform an FFT over the source to find the fundamental harmonic, I can do that and find an accurate grain length, but now I need to find a start ...
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1answer
26 views

Are Total Harmonic Distortion and Noise (THD+N) and THD correlated in practice?

I've read about THD and THD+N due to interest in audio quality. Some audio devices (ej. speakers) report %THD or %THD+N. %THD+N is greater than %THD. I've found that sometimes the difference is huge, ...
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1answer
37 views

Does FFT magnitude of a sound drop to 1/10 of its value if sound level is decreased by 20dB?

I understand decibel = 20 * log (magnitude/reference value). To truly appreciate the meaning of this, if I have a single frequency tone that corresponds to a certain FFT magnitude shown on a PC that's ...
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De-Doppler a signal: Audio Timestretch Factor controlled by function f(x)?

Timestretching usually is constant. Few programs offer bézier handles and such to control the lead-in and lead-out. But that's all. I need better. So the audio I have was in fact recorded in ...
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20 views

Applying binary mask to audio signal

I am trying to implement a simple VAD algorithm (a weighted ES VAD method from http://ceur-ws.org/Vol-2259/aics_17.pdf) and have been able to derive a mask that is either a 1 if speech conditions for ...
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2answers
85 views

Half Window Functions and Formulas

I am looking at implementing different window functions in an application I am making. I have found the formulas for all major symmetrical windows, however I would like to implement half windows. ...
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1answer
46 views

Attenuate the low frequencies in a speech wave

I am designing an FIR filter to attenuate the frequencies below 300Hz. I designed the filter using the ...
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2answers
48 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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926 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
82 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
52 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
58 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
40 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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93 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
50 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
46 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
35 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
34 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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41 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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32 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
58 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
27 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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44 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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7 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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27 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
105 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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71 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
29 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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32 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
62 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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35 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
90 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
45 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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59 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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31 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 ...