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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Plotting the difference in magnitude and phase between two signals using specgram

I have two synthesised audio signals that I've convolved with impulse responses from a multi-channel microphone and I'm wanting to plot the difference in phase and magnitude using matplotlib's ...
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26 views

Paper replication: Validating the proper way to pass .wav audio breathing data through a bandpass filter

I am working on trying to apply a low and high pass filter to an audio file that contains a set of exhalations over a microphone. The inhalations have been cut out of the file, and the exhalations are ...
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2answers
41 views

What is the advantage of overlapping windows while evaluating Spectrogram?

I've seen many documents that refer to calculate spectrogram through overlapping windows but haven't found clear documents on why this is done and how to calculate the number of points that need to be ...
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1answer
25 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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1answer
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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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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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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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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
29 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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1answer
57 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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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
70 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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76 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
108 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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58 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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42 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
48 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
116 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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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
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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3answers
72 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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1answer
50 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
83 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
33 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
95 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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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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1answer
68 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
35 views

How is voice sound on TV or phone distorted through a face mask?

After a year of listening to people talk through face masks, it appeared to me that the sound is not merely muffled, but it has a very specific and recognizable tinge and texture to it. It reminds me ...
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2answers
37 views

Waveform independent frequency analysis

If a musician (or a vocal coach) hears a human voice at a certain note, he/she can easily find the same note on piano. Not only for human voice, if the waveform is square, or created with a synth, ...
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35 views

Finding “Max Change” in a Audio Signal

I'm working on studying an endangered language and have the following problem: while I already have labeled exactly what the speakers I worked with have said at given timesteps on the word level and I ...
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4answers
2k views

How to tell if a high-res flac file has been upsampled from a CD-quality file?

Because I might encode my music in lossy format for other devices or edit them, I prefer to keep a high-res copy of my music. However, I know that it is possible to upsample a file using ffmpeg and it ...
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47 views

Use Fourier Transformation to compare 2 short audio files [duplicate]

I want to use Fourier Transformation to compute the similarity of 2 audios. I am dealing with small audios having the same size: 1 second length, mono channel. My question is how can I use the Fourier ...
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1answer
99 views

Audio signal arrival timing, impulse response, and reverberation

I'm building an application that requires determining the arrival time of an audio signal with a very high degree of accuracy (within 1ms would be ideal) using commodity smartphone hardware. I found ...
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17 views

Speaker Walkaround

Does anyone know what Speaker Walkaround means in context of a digital amplifier? Also how is this test performed? Lastly, how is it different from Load Diagnostics in Amplifiers?
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3answers
546 views

Seamless looping of a signal without pops

I'm trying to seamlessly loop an audio signal. When I loop the current audio signal I have, I hear audible clicks or pops right as it loops back to the beginning. I want to be able to loop the audio ...
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30 views

Emulating tape distortion of low-level audio signals

I'm working on a tape echo emulation in Reaktor, and I've based the frequency response, tape speed warble, high-amplitude saturation and tape hiss on a paper I found about the Roland Space Echo. ...
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1answer
78 views

generating log mel spectrogram using librosa

I have come across 2 different ways of generating log-mel spectrograms for audio files using librosa and I don't know why they differ in the final output, which one is "correct" or how ...
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1answer
77 views

what is silence in audio .wav file and how to compute decibels?

I'm a beginner of audio signal processing, and trying to understand some basic things about audio signals. I primary work with .wav files, so please allow me to use wav as my data to describe my ...
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1answer
984 views

CD quality bitrate

Question 1: Here are two facts that I know about CD quality audio: Audio is sampled to 16 bits at a rate of 44.1 kHz The bitrate is 1411 kbps. I can't quite work out the math. $ \frac{16\ \text{bits}...
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29 views

Equalization in CAR- Automotive Audio

Can someone tell me what is the purpose of CAR Equalization in audio and why before CAR Equalization ,gain adjustments are necessary ? Also why would someone use FIR filters during CAR Equalization ...
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1answer
46 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
29 views

What happens to the audio of a reconstructed signal if we remove too much frequency?

Assuming we have an audio with noise embedded in it, I was wondering what happens theoretically if we remove (filter out) too much frequency when we pass LPF with a cutoff frequency lower than the ...
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1answer
30 views

What phase rotation occurs when you take the derivative of an audio signal?

If you take the derivative of an audio signal, it provides a 6 dB/oct upward sloping filter (increasing high frequencies / cutting low frequencies) all the way across the spectrum. What is the result ...
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50 views

Short Time Fourier Transform has different frequencies than Fourier Transform?

The reason we do the STFT is so that we can analyse for short segments of time how much of the components in the frequencies of the FT are present. However, it may be possible that completely ...
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3answers
155 views

There is an implant in me, is there a way to identify the frequency it is receiving signals on?

There is an implant in me, is there a way to identify the frequency it is receiving signals on? I don't want to share the evidence and my reasoning for believing there is an implant in me, you can ...
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1answer
63 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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1answer
48 views

How to compute dBFS from rms-values if the samples exceed 32bits

For a first approach on audio-sampling I'm currently working with the raspberry pi's pcm pins and MEMS-microphones, using the ALSA library. A problem arises when I want to calculate dBFS from a buffer'...
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2answers
36 views

Centre channel derivation using correlation

I would like to know the algorithm for how centre channel is derived from rejecting left and right channel in audio upmixing process. Can someone explain intuitively rather than hard hitting ...
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2answers
99 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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