Signal Processing Stack Exchange
Signal Processing Stack Exchange

Questions tagged [audio-processing]

222 questions
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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 constituent sound sources. If its not enough,…
hotmeatballsoup
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Multi-channel audio upsampling interpolation

I have a four-channel audio signal from a microphone tetrahedral array. I wish to upsample it from 48 kHz to 240 kHz. Is there a preferred interpolation method for audio? Does cubic interpolation (or any other) have any advantages over linear for…
havakok
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How do impulse response guitar amp simulators work?

I am wondering how impulse response guitar amp simulators/modelers work. I thought it was a matter of convolving a signal of recorded impulse response in time-space with a guitar sample. I tried to do this with Matlab using conv function. I loaded…
Klemen
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Finding Reference Audio Signal in Test Audio Signal and Cropping Accordingly

As seen in the diagram, below I have a reference audio and a test audio. I want to find at what part of the test clip the reference audio be heard. Once, that is found I want to crop the test file from the point where the match begins and the point…
listener
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Audio Processing - How to create a spectral pitch display?

I want to create an application that analyzes human voice pitch, but spectrograms are very noisy. However, in Adobe Audition, there is a feature called the spectral pitch display, and it successfully filtered the spectrogram so that only relevant…
Hykilpikonna
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Actual Sample Rate slightly off?

When capturing microphone audio, I noticed that my static buffer eventually overflows. So I timed the samples, and low and behold I'm getting an actual sample rate of 44102-ish Hz (my timer is very accurate). Is this the actual sample rate recorded…
Tom Huntington
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How to do Frequency Scaling on an Audio File

Please excuse me if my terminology is wrong. I'm from a music production background and have no experience in signal processing. I was wondering if it was possible to stretch out the overtones (partial tones) of sound through digital signal…
ab97
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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 crowded environment. I think that there is a…
zou
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Noise reduction using multiple recordings of the same signal

I have two (audio) recordings of one analog signal, I used two separate devices (smart-phones) recording from two different locations. Each recording device is pretty crappy, lot's of electrical hum and interference etc... My assumption is that each…
Reuben Crimp
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What is the difference between undersampling and oversampling in analog to digital conversion ?

When converting a signal from analog to digital, we observe undersampling and oversampling. Does oversampling means that the sampling frequency is greater that the signal's frequency and does undersampling means that the sampling frequency is less…
Nzui Manto
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Detecting background noise from audio time-freq domain analysis

I have a requirement to detect/reduce sidetalk/background noise in real-time audio. I am stuck in how can I detect this from audio time-frequency domain analysis. I am already getting the time-freq data from stft (I am using java for an easier way…
Nafiul Alam Fuji
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How timbre shifting is done?

I've recently came across two programs - Morphvox, VCSdiamond that are able to preform pitch shift, but also timbre shift. As far as I know the timbre is nothing but the amplitude of the harmonics in the signal. I'm looking for any algorithm or…
Dannynis
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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-triggered to avoid filling the SD card too quickly and…
Thomas
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Eliminate Signal A from Signal B

i have 2 audio signals, both available in time and frequency domain: A: Noise B: Noise(A with minor variations) + Additive sounds (C) i want to eliminate Signal A from B to leave only C. C is for testing almost zero, so after removing A from B…
Maximilian Körner
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Beyond "mid-side": decompose an audio stereo recording (coincident microphones) into Mid + 45° + 90°

Context: I have done in the past stereo recordings in XY position (coincident microphones): from a source far from at least 20 meters (example: piano in a big reverberant building). Since the microphones are coincident, there are IID (interaural…
g6kxjv1ozn
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