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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Programmatically (e.g. in Python) adding clipping to an audio file? [closed]

I am trying to replicate the audio artifact exemplified here to a set of 1000 audio files. In particular, I would like to be able to 'clip' the louder parts of each sound clip so they become distorted ...
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Frequency-specific Delay?

So an equalizer manipulates the gain of certain frequencies by use of a variety of filters, as we all know. But is there a way to manipulate delay in a frequency-specific manner? A parametric ...
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How to make use of a bytes object that represents FLAC format?

I'm streaming audio data from an online server. The audio format itself is FLAC: ...
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77 views

What is difference between frequency and pitch?in nutshell and simple words?

Although both are some how directly related Such that when one increases,the other one also increases I have read different Google results but couldn't understand . I want to know what is exact ...
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Is audio normalisation required for a speaker recognition/speech classification task? How to do it?

Not entirely sure if this Stack or a ML-related stack would be a fit for this question, but I focus on the signal processing here on this question, so I decided this one I'm developing a speech ...
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281 views

Audio signal processing book suggestions

I want to start learning about audio signal processing but I cannot find a good starting point. Please suggest books or lecture series.
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How to convert an audio signal processed by smartphone to PCM format using Android Studio? The ideia is to apply the FFT algorithm

I'm developing an app for Android that records an audio when we press "Record", play it when we press "Play" button, and shows it's spectrum of frequencies for this audio signal recorded, when we ...
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What cause frequency change of the acoustic signal (3 kHz to 10 kHz)?

I am working on Acoustic Resonace testing in which I excite the sample and then record the sound (3 kHz to 10 kHz) by microphones. I do few times repetation of the same measurement in series (one ...
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How to measure quality of an audio/music recording compared to the original

I have 3 wav files: The original recording of a viola note A noisy re-recording of the original clip A different noisy re-recording of the original clip I'm looking for a way to show which of the ...
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Optimization of execution time of audio capture call in c/c++ (ALSA)

I am working on a Audio Deep Learning Project. In this C++ project I need to capture real-time audio and then do pre-processing and the pre-processed output is fed to the Deep Learning Network. I am ...
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pyo: Retrieve amplitudes of audio signal frequenciy ranges

I'm playing around with audio signal processing for the first time and just to start somewhere, I decided to use the Python library pyo. My goal is to analyze an audio input signal from the microphone ...
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Real-time mixing of two audio clips while only modifying the data of one

Problem I want to mix two audio sources in real-time: game audio and microphone audio. The audio is provided at 44 100 Hz in the form of vectors of 1024 numbers between $-1$ and $1$ (PCM data). The ...
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Beamformer implementation methods

I'm currently reading articles about the different type of beamformers. This spatial filtering is gonna be used for acoustic purposes, to focus the beam on the person we need to hear talking in a ...
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Time Frequency bin

I am doing research in audio signal processing for DOA estimation. I came across the term Time-Frequency bin. I have gone through several articles but none of them specify a TF bin rather it specifies ...
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1answer
80 views

How to approximate a moving RMS without iterating over N samples for each required output?

I have a signal with a sampling rate of F Hertz. From it I want to produce an RMS signal, also with F samples per second with a window size of N samples. Using a standard moving RMS algorithm I must ...
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50 views

Reducing one frequency in song

How would I take a song input and output the same song without certain frequency ranges? Based on my research so far, the song should be broken down into chucks, FFT it, reduce the target frequency ...
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How can I calculate time gap in different waves

I want, calculate the time gap between two signals .but that signal has frequently changed the frequency range. I want to combine with these two signal including phase shift I have a circuit for ...
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42 views

Modeling audio effects, reverse engineering

Okay so I'm relatively new to signal processing and don't know the specific terminology to help narrow my search, but I'm looking for advice in the form of books, papers, references, etc that can help ...
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1answer
40 views

Is a “Spectral Tilt Filter” the best way to get an arbitrary slope and cutoff LPF?

I need an arbitrary slope audio LPF for which both the cutoff frequency and slope in dB/oct can be smoothly modulated in real time. Is the "Spectral Tilt Filter" described by Julius Smith the correct ...
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38 views

How do I find the transfer function of a speaker by analysing the audio produced by it?

I have the original audio signal(MATLAB GENERATED) and the recorded audio signal (anechoic conditions). The setup consists of a DAC with a power amp plugged into the speaker. The audio is recorded via ...
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28 views

Audio in Image,Graph or WaveForm Representation

Our task is to feed the audio data into the deep learning tensor flow model in the form of image representation(or graphical,waveform). Question is what is the best way of representing audio in image ...
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30 views

Volume increase of a magnitude spectrogram

Let's say I have an audio waveform and I perform an STFT on the waveform and then take the absolute value to get the magnitude spectrogram. Is there a transformation that I can make to that ...
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1answer
50 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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Phase shifting in frequency domain

I am aware this topic has already been addressed in at least one question here, but despite that I could not find the solution to my problem. Also bear patience with me since I am not a mathematics ...
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Calculating wts in gammatone-gram

I am working on speech segmentation using gammatone filters. I can't understand the variable named GTord to calculate wts(weight) in the ...
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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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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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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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43 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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62 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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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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66 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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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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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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83 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
34 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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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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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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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
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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
129 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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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
41 views

STFT to spectrogram

I would like to know whether I am correct in my understanding of going from STFT to a spectrogram. My goal is to convert a spectrogram back to a wav file. If I have my STFT: ...
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What are the missing steps in the derivation of this equation?

I am trying to understand the model of a piano hammer described here. The relevant excerpt is: As I understand it, they are saying in (12a) that if the hammer spring's uncompressed end's y position ...
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2answers
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How to fix frequency drift when producing an arbitrary length signal by repeatedly looping over a single-period waveform?

I have an audio application that produces an arbitrary length signal by repeatedly looping over a single-period waveform. In pseudo-code: ...
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1answer
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How to measure the “correctness” of a pink noise generator?

I've implemented a pink noise generator for audio applications. How can I measure how close of an ideal pink noise is my output? I do not have access to Matlab or Mathematica. I would prefer a ...
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1answer
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audio classification based only on first few seconds

I record audio notes on my recorder, and each belongs to a certain category ("work", "parenting", "miscellaneous", etc.). At the beginning of each recording, I say the name of the category, then stay ...
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1answer
49 views

How to chose a kernel function for a FIR low pass filter in audio processing?

As the answer of a previous question, I was suggested to use a FIR filter to act as a low-pass filter to reconstruct the envelope of a sound wave (edited, emphasis mine): As I said in a comment, ...
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1answer
32 views

Synchronising audio in realtime without knowledge of original signal

Was doubting whether this is the right SE forum to ask my (elaborate) question, so don’t hesitate to let me know if it’s not. Also, if anyone knows of a better way to title the question, feel free to ...
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ICA School Project

I have been stuck on my school project that consists of independent component analysis. My code will run, but only 50-60% of the time it unmixes the signals. Another problem I have is the amplitudes ...

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