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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210 views

extracting MPEG-7 low-level audio descriptors using essentia

According to https://www.ntnu.edu/documents/1001201110/1266017954/DAFx-15_submission_43_v2.pdf, the audio processing toolbox Essentia includes the 100% of MPEG-7 low-level audio descriptors. However I ...
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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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2answers
13 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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1answer
135 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 ...
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91 views

does audio feedback consist of harmonic content?

So sound from musical instruments as well as human voice consists of harmonics. just wondering if feedback from speakers also have harmonics component I have heard that feedback does not consist of ...
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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?
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6k views

Feature extraction for sound recognition and classification

I am building an application that would "listen" to the microphone input, analyse it, and compare the analysis to a pre-analysed and pre-classified sound bank (small - maximum 20 sounds). It will then ...
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1answer
89 views

How to keep the same power (sonority) between pure sines?

I have a list of sine waves in 1/3 Octave (63,80,100...1k,1.25k,etc) that I want to reproduce with the same power. Right now I just have a global gain that applies to all this freqs, but of course the ...
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2answers
46 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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1answer
43 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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1answer
112 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 ...
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1answer
70 views

Extract same in phase signal from 3 different sources with shared delayed noise?

If I have 3 signals coming from 3 microphones with noise $n$ delayed like this: $S_1(t) = x(t) + n(t+\phi_1)$ $S_2(t) = x(t) + n(t+\phi_2)$ $S_3(t) = x(t) + n(t+\phi_3)$ Their shared feature $x(t)$...
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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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83 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
237 views

how to create a harmonic mask from fundamental frequency?

Using an algorithm that gives the fundamental frequency across time, example https://www.mathworks.com/help/audio/ref/pitch.html (or some other one?) how would I then use that, to create a harmonic ...
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1answer
43 views

How to recognize for small video sequence with a large sequence?

I'm working on the idea of recognizing a small video sequence (a sequence of 18 seconds) within a long one (a sequence of 15 minutes), the first idea that come to mind stats that I have to ...
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299 views

Noise removal using adaptive noise cancellation algorithms in real time systems

I am newbie in signal processing. I want to remove noise from real time audio signal using adaptive noise cancellation(LMS&RLS) algorithms. I don't know how to choose reference signal(noise signal)...
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1answer
113 views

How can you 'perfectly' reconstruct phase for transients in a modified audio absolute spectrogram?

I am currently working on the absolute spectrogram of an input audio signal. I am using non-negative matrix factorization to separate out the rhythmic components (bass drum, snare drum and hi-hat) of ...
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0answers
14 views

How can I analyse, recognise some part of audio files?

I have 1000+ programmes recorded from a radio channel. There are about 6 kind of programmes among them. I would like to discard one of the six types of programmes: about 1/6 of the programmes, ~...
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93 views

Modeling the human perception of noise

Add together some sinusoids with irrational or "nearly" irrational frequency ratios, and you get something pseudoperiodic. If there aren't too many, you can hear that it has only pitched components, ...
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359 views

When does the laugh track kick in?

One of my weekend projects has brought me into the deep waters of signal processing. As with all my code projects that require some heavy-duty math, I'm more than happy to tinker my way to a solution ...
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0answers
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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2answers
316 views

What is the proper way to implement a real time spectrogram?

I am trying to implement a real time spectrogram and I am not sure that I am implementing it the right way. First I decided to use Matlab, I was taking a short recording of lets say 0.1 seconds and ...
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3answers
87 views

How can I convert audio to a series of notes using pitch detection?

I have a good pitch detection system set up, and I would like to return a series of notes given an array of audio samples. My current approach is as follows: I have a moving window across the audio ...
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1answer
448 views

Bandpass filtering an audio signal with MATLAB's $\tt kaiserord$: unknown new frequency component

I am trying to filter an audio signal with kaiserord in MATLAB, but I'm having a problem. Here is my code: ...
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1answer
957 views

Are there any realtime voice activity detection (VAD) implementations available?

I'm looking for a realtime VAD implementation like Discord/Mumble/Teamspeak have. My goal is to have a system that let's me use the same VAD for all my VoIP applications, as well as something I can ...
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23 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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83 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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292 views

How to make sense of audio file's $\textrm{dBFS}$?

I'm trying to build a song visualizer using the FFT method. When I analyze the audio, I get the $\textrm{dBFS}$ value for each frequency range. From my understanding, $\textrm{dBFS}$ is in the range ...
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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
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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2answers
7k views

Logarithmic fourier transform(LFT) on audio signal

I am trying to analyze the musics as possible as precisely. Of course I tried FFT, but got some problems. I found low frequencies have very low resolution than human's hearing. I tried very long time ...
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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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1answer
1k views

STFT overlap add method - What am i doing wrong?

i am trying to implement a short time fourier transform of an audio signal to do some filtering (subtract another stft signal using spectral subtraction). I am using a hamming window function with 50% ...
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1answer
166 views

Polyphase decimation filter plot in Octave

I'm using de Soras hiir library for decimation and I'm trying to verify the filter design in Octave. The filter is a polyphase 2x downsampling filter with 96dB attenuation and 0.01 transition band, ...
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17 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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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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1answer
301 views

Compute group delay of an audio file from STFT

I'd like to reproduce results from this paper in which the authors used Group Delay-grams as an input to the Machine Learning model. The Machine Learning side is not the one that I have problems with. ...
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1answer
3k views

Calculating the SNR of Audio Signal (Recommended Libraries)

Are there any open source packages or libraries available which can be useful in calculating the SNR(signal to noise ratio) of an audio signal. The input will be just an audio signal and I have to ...
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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
909 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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35 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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1answer
67 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
150 views

What is the best input for de-noising autoencoder for sound data?

I am currently trying to build an autoencoder to de-noise audio data. However I have not found any good articles explaining about the input to the autoencoder, i.e. feature vector. As in speech ...
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1answer
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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2answers
48 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
36 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
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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4answers
1k views

Convolution and windowing using a buffer - how do I do overlap add?

I'm reading chunks of audio signal (1024 samples) using a buffer, applying a Hanning window, doing an FFT on it, then reading an Impulse Response, doing its FFT and then multiplying the two (...
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1answer
44 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*...