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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14 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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82 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
231 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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1answer
290 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
112 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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13 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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2answers
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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2answers
355 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
311 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
85 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
446 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
954 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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1answer
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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82 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
291 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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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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16 views

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
165 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
296 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
899 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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2answers
34 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
62 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
145 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
42 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
47 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
39 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
35 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*...
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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
177 views

Teager's Energy Operator

Does anyone know how to find information on the derivation (rather than the definition and application) of Teager's Energy Operator, both for the continuous and sampled case. I have Kaiser's early ...
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3answers
538 views

Signal processing for audio and speech

I have started learning about signals and I am interested in sound signals. There are some questions that I need to resolve. For example in music, difference of notes like 'sol' and 'la' are about ...
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1answer
60 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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1answer
32 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
911 views

What does Fast and Slow Time Weighting signify for real-time audio signal processing?

I have a streaming audio signal that I am recording in real time. To make the explanation easy, say the signal that I receive is calibrated and has a sampling frequency, $F_s = 44100\textrm{ Hz}$. I ...
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1answer
41 views

how to lower frequency of a soubd file in matlab?

I am new to programming and sound processing, I have been using matlab to process sound files, a few of which are ultrasonic. my sampling rate is 250kHz. I Would like to be able to play the sounds at ...
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1answer
40 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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24 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
77 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
44 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 ...