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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256 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
679 views

$1/n$ octave complex smoothing

An excellent answer to this post explains how to do $1/n$ octave energy smoothing and mentions complex smoothing can be done as well, but it's tricky business because of phase wrapping. How is phase ...
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2answers
21 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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0answers
8 views

What features capture the semantics of a sentence from a gien audio?

I have been thinking lately and looking up on what could be the possible features that capture the semantics of an audio and came across MFCC: MFCC features represent phonemes (distinct units of sound)...
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1answer
46 views

How to normalize frequency response charts

I want write a simplistic program that can transform a high-quality voice recording in a way that it sounds like it was recorded using a specific low-quality microphone. I have obtained the mic's FR ...
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1answer
27 views

FastICA: what happens when we have more source than channel?

I am reading the original article proposing FastICA and I have a couple of question on point not covered in the article. Both answering me or providing a source will be very appreciated. I want to ...
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1answer
30 views

Is it possible for Matlab to read and decode an mp3 file?

I have run the following Matlab code and I got a signal as shown in the Figure below. ...
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1answer
305 views

What are F0 counters in Speech signals? What is represented by F0?

I can't find any dedicated definition for Frequency F0 counters. Can anyone tell me what does F0 exactly mean, I am confused about that.
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1answer
36 views

Can an audio recording tell whether it is made in a large or small room?

The conversation of one or more people will be analyzed. I think need to look for something similar to reverberation or echo in an audio signal. How can this be implemented? Update: small room is ...
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1answer
77 views

Is there any science to explain the “weight” or “3D depth” of hardware audio vs software that some people claim?

Is there any science to explain the "weight" or "3D depth" of hardware audio vs software that some people claim? There's, I think, strong subjective evidence to suggest that the phenomenon is true. ...
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2answers
121 views

How to graph magnitude plots for basic 4-pole filters?

I have developed magnitude plots for basic one-pole and two-pole filters derived from various Physics PDF's and tutorials I have found online and calculated from: https://www.desmos.com/calculator/...
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19 views

Pitch detection using Cepstrum in matlab

What i am trying to do is to detect the pitch of a given sound signal, so for that purpose, i wanted to find cepstrum of a given signal. This is my code in MATLAB: ...
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1answer
88 views

4-phase FIR LPF in MATLAB

I'm trying to create a low pass FIR filter using the coefficients I found in ITU 1770-3 on page 18. This is the last piece of the puzzle for an audio loudness metering algorithm I'm working on. I'm ...
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2answers
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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3answers
115 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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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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42 views

How convert Given DBFS into DB in iOS?

I am getting DBFS value of an audio ranging from -160 to 1. I am looking for to create a SPL meter in iOS to show actual DB values of audio capture through mic in device. Using recorder class, I'm ...
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1answer
23 views

Digital Audio Mixing Fundamental. What will be the data rate of mixed audio?

I have voice data at 128 kbps coming from three different sources. I'm looking for mixing all three voices together and sending it back to speakers. What methodologies are there for mixing digital ...
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1answer
78 views

Which (Fourier transform?) processing method to use for time variant audio processing? (newbie)

Edit: I've since been able to understand the how the relvant parts of Fourier transform work and how they relate to my problem / what I want to do. And have come to the conclusion that Fourier ...
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1answer
313 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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3answers
3k views

Impulse Response to Frequency Response in Octave

Having an impulse response of an audio system recorded as a wav file, how to calculate the frequency response of the system with octave?
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3answers
711 views

Distribution of energy in frequency bands

On my wav-file (music), I do a short-time Fourier transform (STFT). I have spectrum, which spans from $0\textrm{ Hz}$ to $f_s/2$. I took out the range between $1$ and $1000\textrm{ Hz}$, which I ...
3
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0answers
36 views

Audio sharpening

I'm working on a project where the purpose is to enhance FFT peaks in a signal (musical instruments), by other words, reduce non-harmonic components and enhance the identified FFT peaks. So far I'm ...
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3answers
156 views

Resample audio signal with a low factors

Currently, we need to handle a piece of an audio signal. We want to resample it with different factor rate (p/q) starting with 1.0002 to 0.9998 with a decrement of 0.000005. But Matlab function cannot ...
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1answer
94 views

Introductory resources to mathematical properties of “dynamically recalculable” filters in audio/musical equalizers?

Introductory resources to mathematical properties of "dynamically recalculable" filters in audio/musical equalizers? Particularly, I want to understand What mathematical features (e.g. monotonicity ...
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1answer
98 views

Audio texture analysis for visualization

I'm making a complex music visualizer as a personal project. I've already determined that I will be using a series of modified Goertzel algorithms for frequency detection to place the frequencies on a ...
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2answers
183 views

What Is the Meaning of the Term "Frequency Response'' When We Design a Pitch Detector

This is from Klapuri 2003. What is the "frequency response" and why do we model is as multiplied with the vibrating system we want to get the frequency from?
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1answer
27 views

Hydrophone sensitivity rating

If I use a hydrophone with the sensitivity given as: "-200 dB re 1V/μPa". Question: How should I interpret this? Is it the voltage given at -200 dB SPL or?? All the Best!
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1answer
36 views

Shazam / SoundHound Music Analysis - Recommendation for Reading

I need to explore some area around music frequency analysis. A good example is the way Shazam / SoundHound app works (mapping frequency domain, comparing, creating some vector map, handling noises and ...
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2answers
51 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
44 views

Given 1 recording of my voice, how can I process it to sound like 2 independent recordings of my voice played in unison?

I have one recording of my voice $x[n]$ singing a song. One thing I could do is independently record me singing the song again to obtain $y[n]$. Then, $z[n] = \frac{1}{2}\left(x[n] + y[n]\right)$ ...
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1answer
15 views

script to check zero crossings to check if microphones phases are aligned

I have 'n' microphones placed in square fashion and I want to make sure that all the channels must be aligned in time exactly the same if the signal is equidistant from all the 'n' microphones i.e. in ...
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1answer
32 views

generate a repeated square pulse train in python

Generate a pulse train with -2dBFS pulses of 20 sample duration every 8 seconds. I have referred here to generate a pulse train but not able to repeat it. How can I repeat this audio signal every 8 ...
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1answer
366 views

Machine Learning - any suggestions to solve Python rounding errors?

I am working with Python to isolate voiced segments from music using the Jamendo corpus for singing detection. Training a model, I break my audio into frames, and have a label (0,1) for each frame. ...
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1answer
851 views

Verification of all steps needed to extract MFCCs from raw audio data

I need some verification on steps required to extract MFCCs from raw audio as I'm finding that some sources outline them slightly differently. Get raw audio frame Run window function (eg Blackman) on ...
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1answer
329 views

Dynamic Time Warp through SPTK tool

I'm an IT student and got an assignment to do about Dynamic Time Warping(DTW) using the Speech Signal Processing Toolkit (SPTK) and comparing some words spoken by 2 speakers and finding the ...
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1answer
659 views

Applying DTW on MFCC frames

I want to use MFCC (Mel-frequency cepstral coefficients) and DTW (dynamic time warping) algorithms to make a speech comparison script. I calculated MFCC and got 13 coefficients for each frame. How ...
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1answer
1k views

Are there any signal processing libraries that implement TD-PSOLA?

For such a popular and old algorithm, I've had surprisingly little luck finding implementations. I've got it working in MATLAB, but am curious if any other languages have libraries containing it (...
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1answer
4k views

Demodulating FSK audio in Python

I am trying to write a Python script that can demodulate an FSK modulated audio file and return the data encoded in the audio. The data being transmitted is GPS NMEA strings which are embedded as the ...
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1answer
32 views

Detect playback position in swept sine sweep

I am wondering what the most precise method is to detect the playback sample position of a sine sweep? The real playback sample position is unknown and an DSP with constraint computing capabilities ...
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1answer
279 views

audio for VOIP - filter and resample or viceversa?

I am a noob to signal processing, but I need to create some audio files for IVR for Asterisk VOIP. So my lady friend with a nice voice records her message on a Windows machine and sends me a 44KHz, ...
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1answer
233 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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1answer
154 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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2answers
122 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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2answers
65 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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26 views

Mixing signal with noise in frequency space (STFT)

I am working on a preprocessing function which is supposed to mix two audio signals clean and noise in the context of neural ...
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1answer
182 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
122 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
71 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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2answers
95 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 ...