Questions tagged [audio-processing]

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7
votes
5answers
2k views

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 ...
7
votes
2answers
380 views

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 ...
5
votes
3answers
3k views

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 ...
5
votes
2answers
2k views

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 ...
4
votes
4answers
1k views

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 ...
4
votes
1answer
61 views

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 ...
4
votes
1answer
7k views

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 ...
4
votes
1answer
45 views

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 ...
4
votes
1answer
171 views

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 (...
4
votes
0answers
79 views

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 ...
3
votes
2answers
8k views

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 ...
3
votes
5answers
311 views

Algorithms to re-spatialize a stereo recording audio signal?

Let's say we have an audio signal stereo recording, made for example with a XY microphone positioning: Which algorithms can be used to "re-spatialize" this recording, i.e. try to virtually &...
3
votes
3answers
140 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 ...
3
votes
2answers
139 views

Modelling Unwanted Signal in a LMS Adaptive Filter

I'm having some confusion learning about the LMS Adaptive Filter. I understand that the whole model of adaptive filters relies on the fact that we give it a reference signal to which it keeps ...
3
votes
2answers
61 views

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 ...
3
votes
1answer
138 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-...
3
votes
1answer
174 views

Count heart beat from audio capture using smartphone

I am working on a project where I am required to perform heart rate BPM calculation from an audio signal captured using smartphone microphone / hands-free. Constraint is not to use ML. So I am left to ...
3
votes
0answers
89 views

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 ...
3
votes
0answers
64 views

Frame lookup using audio

I'm an animator trying to get a program that will use machine learning to automate the lip syncing process. The program I need made will have two files given to it: a seed, and an audio file. The seed ...
2
votes
2answers
842 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 ...
2
votes
1answer
785 views

Compute group delay of an audio file from STFT

I'd like to reproduce results from this paper in which the authors used Group Delay-grams as an input to the Machine Learning model. The Machine Learning side is not the one that I have problems with. ...
2
votes
1answer
1k views

What is the best input for de-noising autoencoder for sound data?

I am currently trying to build an autoencoder to de-noise audio data. However I have not found any good articles explaining about the input to the autoencoder, i.e. feature vector. As in speech ...
2
votes
1answer
66 views

Software Mixing of Multiple Audio Streams From Network

TLDR: How do I mix network streams when streams might be unreliable? I am working on a project which will allow musicians to practice/perform together over the internet with low latency. (Think ...
2
votes
1answer
32 views

Most generally-accepted audio term, for "pre" and "post" lengths of a symmetric filter's impulse response?

Consider a FIR audio filter with symmetric impulse response. In my case, the filter kernel width is 15 frames on each side (so including the center frame, 31 frames total). When examining the audio ...
2
votes
1answer
157 views

finding onset of an impulse signal, basics?

New to signal processing, but making good progress, I think. I have a series of (many) impulses I generated, which will be used as impulse responses to model our church's acoustics, in this time of ...
2
votes
1answer
544 views

Problem using pitch shifting with TD-PSOLA and formant preservation

I tried out this implementation of TD-PSOLA (https://github.com/sannawag/TD-PSOLA) and even though I know how the algorithm works, I can't understand why I get NO audible differences in the output ...
2
votes
1answer
1k views

Removing breaths using python

My wife is trying her hand at making audio books and spends a lot of time cleaning up some voice actors breaths in their "finished" audio. We have tried some filters in Adobe Audition and it helps but ...
2
votes
2answers
1k views

audio spectrogram normalization

I am performing language classification from audio signals using mel-spectrograms as my inputs for a ResNet. It works well as long as all of my audio data from different languages is from the same ...
2
votes
4answers
128 views

Baby Cry recognition robust against the distance [closed]

I am building a baby cry detection system using deep learning algorithm, Raspberry Pi and a microphone as the sound sensor. When the baby starts crying then the system should detect the cry. It is ...
2
votes
2answers
2k views

How to Subtract One Signal from Another?

I am capturing some body internal signals. I am using 2 identical mics. One captures the in body signal and one open air. I want to remove/delete/subtract the open air signal (noise for me) from ...
2
votes
2answers
46 views

Non-Uniformly Partitioned Convolution Implementation

I've succeded in implementing the uniformly partitioned convolution algorithm and now I'm looking to implement the non-uniformly partitioned version. I've had no luck with running parallel threads on ...
2
votes
1answer
510 views

Overlap Add - which length to use?

I am doing realtime audio. Let's say I have a signal $x(n)$ which is passed in blocks of 512 (siobuffer) to the program filter/FFT method. Within this block I want to convolve the signal with the ...
2
votes
2answers
508 views

How to interpret these MP3 spectrums?

Recently, I found I had this song duplicated in my music library, but when listening to them I noticed one version was of noticeably higher quality than the other despite both claiming to have a ...
2
votes
1answer
167 views

Distort voice to hide talker identity as much as possible

I need to develop an app which makes speech recognition. Because I need to be able to constantly improve the recognition capability, I would like to train the engine with real recordings and match the ...
2
votes
1answer
36 views

Getting error while performing upsampling of an audio signal processing using low pass filter in python

I want to perform upsampling of an audio signal introducing zeros between samples to create a longer signal with factor of N=2. The sampling rate is 44100Hz, time = 5sec. ...
2
votes
0answers
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 ...
2
votes
0answers
41 views

Tell a recording of Martian wind from a recording of Earth's wind

Perseverance Rover's SuperCam Records Wind on Mars, sol 11 (Credit: NASA/JPL-Caltech/LANL/CNES/CNRS/ISAE-Supaero) How do I interpret an image of a spectrogram of an audio wind signal? (Credit: ...
2
votes
2answers
235 views

Comparing two small audio files (e.g., hand clap) Android

Dears, I am new in signal processing and I am dealing with an Android application that should do some audio files (having the same "small" duration) comparison in real time. My main focus is ...
2
votes
0answers
270 views

How to work with portaudio?

I am working in a project to record audio and process in real-time in C.For starting, I wanted to check how it works. I downloaded the pa_stable_v190600_20161030.tgz from here. I ran ,/configure and ...
1
vote
4answers
160 views

What is the principle behind my voice echoing perpetually when 2 separate devices are used for playback and recording on video calls?

I had to use 2 separate devices for a video call, desktop for video/audio output and smartphone for recording, since the desktop microphone was not working properly. Unfortunately for me, when I ...
1
vote
1answer
60 views

Audio Processing Glossary

I'm new to audio processing and I hope here be a good place to ask about the definition of terms in this context. My question is what is difference between sound, audio, voice, speech and tone, from ...
1
vote
2answers
8k views

Understanding the frequency domain [duplicate]

I'm trying to understand what frequency domain is. I found general explanations on the Internet, for example: frequency-domain graph shows how much of the signal lies within each given frequency band ...
1
vote
1answer
338 views

Analog-Digital Dilemma on Audio Signal Processing

I am planning to build a hi-fi speaker along with a tweeter and a woofer. Most probably, the predominant stage of the speaker is the audio signal processing stage. According to the need of high ...
1
vote
5answers
178 views

Effect of One Sample Latency in Audio Application

I am dealing with a customized processor that has a processing latency of one sample (20.8 microseconds) at 48kHz Sampling rate . What I would like to know is ...does this latency of one sample have ...
1
vote
3answers
126 views

Sampling a low-frequency sound doesn't work at some sampling rates

When sampling a low-frequency sound (like a 55Hz sine wave) I hear an additional high frequency sound at certain sampling rates, but not at others. This happens in Audacity and in my own program. For ...
1
vote
1answer
156 views

FFT results are repeating?

I am trying to detect notes out of a musical sound input. My program can be run here. The problem is the patterns in the frequency response is repeating: I suspect that echoing in the room is causing ...
1
vote
1answer
126 views

What is the purpose of dividing an audio signal into segments and analysis each segment?

I read bunch of materials for extracting feature from audio signal and they all tell me to break signal into segments, why don't we analyze all the audio signal? I don't know what are the advantages ...
1
vote
4answers
73 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 ...
1
vote
2answers
132 views

Number of microphone limitation for Spherical Harmonics based DOA estimation

I am following this book "Theory and Applications of Spherical Microphone Array Processing", as well as this paper on DOA estimation "Eigenbeam-ESPRIT for DOA-Vector Estimation". My dataset is ...
1
vote
1answer
162 views

How many psychoacoustical critical bands are there?

I understand that the human ear works as a band-pass filter and I have doubt regarding the number of such bandpass filters in the filter bank. I came across these two possible explanations of how it ...

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