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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1answer
387 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
124 views

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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1answer
138 views

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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1answer
43 views

Visualise/compare numpy arrays from Matlab/Octave to matplotlib

I'm new to python and matplotlib, and I'd like to visualise / compare 3 mfcc files stored as numpy arrays in txt format. I have the Octave code below, and I'd like to know how it can be done using ...
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1answer
23 views

How would I approach implementing a windowed sinc filter with changing cutoff in real time?

I moved this out of another question, where I'd put it to give more context, and answered it here for the next reader. ELI5 how to implement a windowed sinc FIR filter in real time for an audio ...
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3answers
3k views

Measuring Audio Signal Similarities

I have recorded samples of lungs sound (person breath) signals from electronic stethoscope. How can I measure (In percentage) the similarity between two audio signals (lung sound), the general idea is ...
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5answers
1k views

How to deal with audio with different length in Audio Classification

Currently, I am trying to work with the Dataset UrbanSound8K to try some Audio classification. And I got stuck in the preprocessing step already. Since the audios are of different lengths, like 4 ...
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1answer
15 views

Microphone Data Comparison for breathing data… best way to compare two files?

I am currently trying to compare the amplitude vs time data between two different microphone readings (.wav files of inhales and exhales) and detect which reading produced a louder sound, on average. ...
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0answers
25 views

Role of oversampling within resampling / pitch modification

I'm working on a pitch modification plugin involving resampling, and want the quality to be as high as possible. I currently use windowed sinc interpolation and lowpass filter where required. I ...
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2answers
36 views

When applying analog and digital effects on an audio signal is it better to start with a clean digital signal or use the analog effects before ADC?

If I am digitally capturing an audio signal from an analog source and I know I will want some analog effects applied, is it better to use them on the analog signal before doing the ADC, or should I ...
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1answer
93 views

How to make ultrasound audible?

Because of interesting spectral phenomena, I would like to shift the >20kHz content of my audio recordings to audible frequencies, so that I can listen to it. I tried the Audacity pitch correction ...
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1answer
41 views

Chirp zero crossings

I am generating a linear chirp in C code with a f0 , f1 and T. What is the most efficient way to solve where the zero crossings appear? Thanks
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1answer
9k views

numpy.correlate and autocorrelation; audio signal

I want to calculate delay between an input and an output audio signal of my audio processing system. The input and output signals are available as signed 16 bit integers. To try out, I tried the ...
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1answer
28 views

Time delay of direct path in RIR measurement

I am trying to measure an RIR using my laptop and a JBL Flip 5 speaker. I have tried, both, the Matlab tool and a python code shared on gihub. My final goal is to compare with the image method and ...
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2answers
68 views

Specify minimum trigger frequency for recording audio in Python

I'm writing a script for sound-activated recording in Python using pyaudio. I want to trigger a 5s recording after a sound that is above a prespecified volume and frequency. I've managed to get the ...
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1answer
55 views

How to extract noise from a signal in order to get both noise power and signal power

I have to implement a function to get an accurate SNR value. For that, I would need a separate noise signal to get the signal power and noise power but I have only one noisy signal (a .wav file.) ...
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0answers
8 views

Sound activated recording in Julia

I'm recording audio with Julia and want to be able to trigger a 5 second recording after the audio signal exceeds a certain volume. This is my record script so far: ...
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2answers
56 views

Is there a way to convert audio to a list of frequencies preserving the temporal position?

I want to convert an audio file to a list of frequencies. We can apply an FFT on the audio to gain analysis of all the frequencies in a given audio sample, however, I want a list of frequencies as a ...
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2answers
106 views

What is a “pitch period”?

The term pitch period appears in the book Speech and Language Processing by Daniel Jurafsy: As we just said, a pitch-synchronous algorithm is one in which we do something at each pitch period or ...
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0answers
72 views

Extracting audio segments based on detected onsets using libROSA

I have a couple of .wav audio files with very similar percussive signals whose onset times I can identify using libROSA's onset detection. I would now like to extract the associated audio segments ...
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2answers
41 views

Why does the volume increase when I reduce the gain of an all-pass filter connected in series?

I am using the colorless allpass-filter proposed by Schroeder Manfred R to create a reverberator. It consists of about 8 colorless allpass-filters connected in series, each with a different delay ...
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1answer
101 views

Popping sound after using bandpass filter

I'm trying to create guitar Wah Wah effect (basically band pass filter with dynamic frequency range). I'm using Steinberg SDK for it. I refactored this code to work in real time and it looks like this:...
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1answer
53 views

Why use both a high pass and a low pass filter in a Butterworth implementation for noise reduction?

I am a new to audio processing . I want to develop an application that reduces audio noise using a Butterworth filter. I found some existing code doing this, but I still do not understand the use of ...
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2answers
98 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
112 views

Sound activated recording and advanced filtering on Raspberry Pi

I'm making a Raspberry Pi bat detector using a USB-powered ultrasonic microphone. I want to be able to record bats while excluding insects and other non-bat noises. Recording needs to be sound-...
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1answer
287 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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1answer
37 views

How common are audio effects that process a signal as a whole, rather than each buffer separately?

I'm writing a VST plugin using the JUCE library, and the way it is set up is that you can write a block of code to process buffers as they are fed to the program, one by one, meaning the program has ...
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1answer
27 views

Differentiate betwee sibilant “sssh” voice sounds and instruments like hi-hat?

How would I differentiate betwee sibilant "sssh" voice sounds in a music track and a similar sounding instrument sounds like hi-hat or cymbals?
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3answers
999 views

How to edit audios so that they have the same length

I have a dataset of audio files, which differ on their lengths by a couple of milliseconds. Some are little shorter than 1s and some are little longer than 1s. They all have the same sampling rate, ...
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0answers
35 views

Invertible overlap-summation of a vector

I'm using the Nonstationary Gabor Transform (https://github.com/grrrr/nsgt) for a DSP application. It has a sliced transform, called the sliCQ. Background: https://www.sciencedirect.com/science/...
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1answer
179 views

Phase difference calculation between two wave files

I am doing comparison between two wave files, How to calculate phase difference between those files? Second wave is generated by making some changes in the first one .I have tried calculating phase ...
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1answer
49 views

Karplus-Strong with Jaffe-Smith sample rate-independent

I tried to implement the Karplus-Strong algorithm with some of the Jaffe-Smith extensions (improve pitch, dynamics, etc). Here is the code. When I run it with a sampleRate of 48000Hz, I get very ...
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1answer
59 views

Comparing spectrogram similarity over time

I have a data set of spectrograms from moving audio recorders. As the recorders get closer, I expect the spectrograms to become more "similar". I expect this "similarity" to be a ...
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2answers
650 views

Realtime sample rate conversion

Is it possible to have a realtime sample rate conversion in a way that a peer A with an audio stream at 44.1 KHz sends it signal over the network to another peer that is on an audio stream at a 48 KHz ...
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4answers
349 views

Signal and Space/Time

I have been reading a couple of definitions on signal such as: Any quantity measurable through time over space or any higher dimension can be taken as a signal. And this, In the physical ...
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3answers
2k views

What's the impact of aliasing in the time domain?

I've been studying digital audio and come across something I can't understand. There appears to be something like a consensus (among those capable of understanding such things) that the impact of ...
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3answers
71 views

Is there any deterministic downsampling algorithm for audio?

I was trying to use the scipy.decimate function provided in the library, but I had a question - I want a pre-processing algorithm to downsample audio in such a way ...
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2answers
244 views

Is it possible to use a Volterra series to generate subharmonics?

Taylor series can create harmonics of the frequencies in an input signal. I'm wondering if it is likewise possible to use Volterra series to create sub-harmonics of the frequencies in that signal. ...
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2answers
211 views

Where do RBJ's analog shelving filters come from?

In RBJ's Audio EQ Cookbook, the analog prototype for a low shelf filter is given as such: $$H(s)=A\cdot\frac{s^2+\left(\frac{\sqrt{A}}{Q}\right)\cdot s + A}{A\cdot s^2 + \left(\frac{\sqrt{A}}{Q}\right)...
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1answer
49 views

far field and near field definition for microphone array

What is the definition of far-field and near-field in acoustic sensor / microphone array? Does it depend on the distance or the power of the sound?
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1answer
221 views

How do audiobook players manage to quickly speedup/slowdown playback speed without using more memory?

Many audiobook player apps on Android have a reading speed slider which lets us choose the reading speed e.g. 0.5x 0.75x 1.00x 1.25x 1.50x. This is time-stretching ...
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1answer
45 views

Balance Fader in Digital Amplifiers

What is the purpose of Balance / Fader in a digital Audio amplifier? In a signal flow what is the most apt position to place balance and fader(for example after before a compressor unit or at the ...
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1answer
37 views

Why does my zoom audio, when there are nearby microphones, seem to usually take an open loop format?

The signal from the speaker on device A gets into the microphone from device B, then the amplifier and then the speaker from device B, back into device A. The result is a sort of runaway signal that ...
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1answer
32 views

Amplify filtered smooth brown noise to range [-1.0f, +1.0f]

I am trying to make a "noise generator" like Pioneer example here. I am using a smoothed brown noise and passing it to a bandpass filter to achieve this noise in C++. For smooth brown noise. ...
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1answer
82 views

Audio Equalization, Is Parallel or Cascade better

I'm not really an audiophile, so I can't determine the difference. I was playing around with Matlab's Graphic EQ and I noticed that there is a difference in the Magnitude response for the same gain ...
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1answer
65 views

Generate Smooth Brown Noise Mathematically

I have a sound engine where I feed float PCM values between -1, +1. I am trying to make a noise generator. I want to be able to generate brown noise Mathematically as float PCM valaues so that I can ...
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1answer
116 views

How to compute key of a song?

I'm looking for a precise way to compute the key of a song. BPM would be cool too. Analog is ok. I tried to approach it from the DFT angle, but got stuck and can't really put a finger on what's the ...
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1answer
93 views

Can lossless audio be wirelessly transferred via Bluetooth?

I have a question that is that possible to wirelessly transfer lossless audio via Bluetooth? Please help me explain technical reasons behind it such as the size of the the lossless data, the maximum ...
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2answers
98 views

How to generate a noise like Pioneers white noise?

I am making a music app and I want to put a noise generator in it. The problem is the usual white, pink noise is just too boring and not useful for the user. Pioneer mixers has an amazing noise ...
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1answer
62 views

Frequency response of a long signal

I am trying to get the frequency response of a system, so, in theory I would need to do: H = fft(y)./fft(x); I have measured the system with a long stepped sine ...

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