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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37 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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How can an audio segment report an integrated loudness (LUFS) greater than 0

According to my very rudimentary understanding of the ITU-R BS.1770 standard, I thought that the LUFS (LKFS) value for any audio signals should not ever go positive. However, there is a 5-second ...
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37 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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3k 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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696 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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329 views

can MFCC coefficients alone be used to do voice detection?

Aim of my project is to differentiate between cases of silence, single active speaker and multiple ( simultaneous ) active speakers using as basic ( but effective ) logic as possible. Is it possible ...
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57 views

HRIR interpolation using VBAP

1) Problem description I am trying to implement a 3D audio simulator in Python. I am using the HUTUBS dataset as HRIR database (more informations here: https://depositonce.tu-berlin.de/handle/11303/...
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72 views

What is this strange pulse in my audio output?

Firstly, if this is the wrong place for this, I'd be happy to know the correct place for this and I'll move it there. I have two PCs which share an audio interface. My laptop is connected via USB and ...
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34 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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44 views

Power Spectral Density Reconstruction

I am from an ML background and I am working on MP3 audio reconstruction and I have managed some amount of spectrogram reconstruction but I have noticed that the higher frequency end of the generated ...
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2answers
210 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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24 views

Best way to compare audible alarms

I'm working on an alarm recognition program in MATLAB in which multiple reference audio samples of different alarms would be recorded. Afterwards, the program would sample every so often and compare ...
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31 views

How to find peak frequency in an audio frequency spectrum?

I have a audio file consisting of multiple frequencies, I need to find all the frequency peaks in the frequency spectrum after doing FFT. But the issue is how can I be able to set the threshold line ...
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32 views

How do I set window length and shift length for computing STFT on Scipy?

Does nperseg=32, noverlap=16 mean the window length is 32mS with a shift of 16mS? Refer here for more details
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Cleaning up voice recordings - where should I start? [closed]

Hi I'm relativity new to cleaning up recordings, as this has just became a new hobby of mine. Is there any good sites that have a good forum/chat etc that a beginner would feel comfortable in? I'm ...
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365 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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Can Principal Component Analysis (PCA) Solve the Cocktail Party Problem?

I'm looking into the cocktail party problem and trying to figure out whether something like Principal Component Analysis is enough to separate out all the various voices at the cocktail party into its ...
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1answer
57 views

What kind of filter is used here?

I'm very new to the DSP world, and honestly I'm not even sure if this is the right place for this question. Anyway, I found this piece of code on github, it's a kind of filter (I think), and I would ...
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Properly stagger an FFT? How to handle output data (Have highs weigh as much as lows?)

I'm new to audio processing so I'm mostly running into problems with not being able to properly google something. Sorry. Basically I want to create something like an equalizer (just visually without ...
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172 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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How to deal with low fundamental when using AMDF for pitch extraction?

I am using the Average Magnitude Difference Function to estimate the fundamental frequency of a quasi-periodic audio signal. The AMDF is defined as $$ D_n = \frac{1}{N-n}\sum_{k=n}^{N-1}|S_k - S_{k-n}...
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1answer
64 views

Designing FIR filter for real-time decimation of audio on embedded system

I need to design a lowpass FIR filter to downsample an audio stream that is oversampled at 250Khz. The goal is to write a real-time audio player of an audio stream on an embedded system which is ...
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1answer
32 views

Ambient noise levels in desert and recording studio

Are there mathematical models which suggest which of the two - desert or recording studio - is likely to be quieter? I would think a recording studio would be quieter than a desert. Is there a ...
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119 views

Can harmonics be created or simulated a la carte

I am trying to learn about adding harmonics (audio) with a goal to make a plugin that can allow adjustment of the amplitude of each harmonic being added. I would not have the ability to write this but ...
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69 views

Calculate periodicity of audio signal

I have to calculate the periodicity of a audio signal like this: You can see by eye, that the volume rises the highest every second "blob". The spectogram (that is visualized ...
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3answers
127 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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37 views

How can I filter a noise with pattern type in audio?

I have a set of audio files rescued from a source of phonograph record like a gramophone. Very old family stuff records from my elders back in year 1946. A relative has digitalized these with a ...
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86 views

Implementing a pitch shifter

I am trying to implement a pitch shifter similar to this one, in MATLAB for quick prototyping. My goal is to do the reverse, but I assume it is similar. The implementation details are missing. So I am ...
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26 views

High frequency noise out of a dummy phase vocoder

I am trying to understand where does the high-frequency noise comes from when I use a phase vocoder with no modification to the spectrum or speed. Here is the MATLAB code I used. I changed it to my ...
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60 views

Questions about MP3 bitrate conversions

I was trying to analyse the lossy compression of MP3 audio files through their Spectrograms and the results were something like this (where the upper is 80Kbps and lower is 128Kbps). It is obvious ...
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56 views

What framesize is usually used for audio effects?

I want to implement an audio effect that reacts to an external parameter that varies over time. The book I have describes an effect in a function that processes a certain framesize of the audio input. ...
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193 views

FFT filter to remove cd scratch

I have a .wav file which came off of a cd with a scratch, so there is a periodic "clicking" of about 6 Hz. I was playing around with trying to squash the clicking by taking FFT and filtering the ...
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1answer
20 views

Derive Time-Domain Biquad Coefficients from Z-Domain

I'm trying to derive the time-domain coefficients of a biquad filter based on its z-domain coefficients. The purpose is to import externally-generated EQ filters (in z-domain) into Max/MSP (in time-...
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How to apply audio superposition in mel spectrogram domain

Very rusty physicist here. I have two audio spectrograms and I want to work out how to calculate the spectrogram of the superposition of the original waveforms (without having access to the original ...
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3answers
941 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 ...
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34 views

If video uses color bars to test and align color values among TV sets, what could be the equivalent in Audio?

Are there any sample Audio files, encoded in the simplest format known, that can mathematically explain or illustrate the fundamentals of (audio) signal processing that might also relate to known ...
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69 views

Removing cracking in real time audio filtering

I am getting back to DSP applications after a while and I've written a real time audio player in Python. Basically I'm reading chunks of data from a .wav file (I am doing this to simulate a real time ...
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328 views

How long does it take for snd_pcm_writei() (ALSA) to send data?

Using a pretty typical call to Linux's ALSA, snd_pcm_writei(pcm_handle, pcm_buffer, pcm_buffer_size) how do you know how long it should take to send this data? What ...
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3answers
216 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
29 views

Error introduced by smoothing filter at the end of a processing block

I'm running analysis on a multichannel audio signal and due to the size have decided to process it in blocks (my computer doesn't have enough memory to process it in one go). Some of the data produced ...
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99 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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207 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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116 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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3answers
51 views

What are typical audio waveform synchronization methods?

I have two roughly similar (not identical) audio files, both being recordings of the same audio source material, but made independently of each other. I need to calculate the time offset between each ...
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400 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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942 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
23 views

Convolution output signal is very large, do I need to scale inputs or outputs?

I am writing some simple matlab code to refresh my convolution knowledge (it has been awhile) in regards to DSP. I have an input signal x and a filter - ...
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374 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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896 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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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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