Questions tagged [audio-processing]

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53 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
62 views

Ultrasonic TDoA / ToA Measurement?

I am trying to make an ultrasonic flowmeter. The device has two ultrasonic tranducers (resonance frequency of about 200kHz) that sample one ultrasonic signal in the upstream directon and then ...
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0answers
18 views

Smooth transition between audio without crossfade

Suppose we have two audio signals, and we want to be able to play them back to back, without pops, dips in volume, or other noticeable sounds at the transition point. A similar question on sound ...
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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 ...
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23 views

Audio sample analysis - Comparing two audio samples, the original sample with a reproduction capture sample [duplicate]

Already asked in sound.stackexchange.com but it was suggested by one of the comments that it would be better suited to be asked here. Introduction I'm currently working on a project where I'm ...
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2answers
835 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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1answer
52 views

Composing digital filters

I'm interested in composing filters for realtime audio processing on an microcontroller (MCU). Ideal frequency response is unity as a default, with deviations up and down at specific freq-domain ...
2
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1answer
35 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. ...
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3answers
305 views

What is really the Mel-filterbank?

After applying STFT to a signal, you typically need to convert the frequencies to the Mel scale. Typically the frequency values range from $0$ to $N/2$, where $N$ is the width of each sliding window. ...
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1answer
124 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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0answers
88 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
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1answer
173 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 ...
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1answer
20 views

How to differentiate two different signals from their combined signal

I am having two omnidirectional microphones connected in endfire configuration. I am playing a male speech from 0 degree (front direction) of the endfire microphone configuration and female speech ...
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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 ...
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1answer
43 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 ...
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1answer
44 views

I want to perform aec on a wav file and calculate erle

I am trying to evaluate the performance of an AEC by comparing the i/p and output wav file. I want to calculate echo return loss enhancement to predict how much(db) echo has been attenuated but i am ...
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1answer
32 views

Phase in FIR filters [closed]

Does anyone have any idea what is Clear Phase technology ? How is it implemented ? And which audio channels whether left ,right center , sub woofer should have this technology implemented and why? ...
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1answer
282 views

Frequency domain processing

I'm a little fuzzy on how the Fourier transform works. It's supposed to map time domain to frequency domain, and vice versa, so my intuition says that if I wanted to make an equalizer, I'd take the ...
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1answer
336 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 ...
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1answer
94 views

Digital Audio Amplifiers

Can anyone tell me why a filter that blocks DC signal in an Amplifier would have a cutoff frequency of 4 Hz . Why would someone choose a frequency as low as 4Hz to filter out DC content before ...
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1answer
85 views

Audio Activity Detection and Classification

I am starting a new project. Actually, my real intention is to learn speech recognition but for warming up I want to improve in audio signal processing. In my project, I aim to record sounds except ...
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2answers
45 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 ...
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1answer
42 views

How to remove or smooth the comb filter effect in real time audio signal mixing?

I have been working on a project where I have to mix multiple audio signals of the same source coming from different slave smartphones on one master smartphone in a distributed way. Now I have aligned ...
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1answer
33 views

Identify and remove repeated audio chunks

I have some audio produced by a buggy recorder, at some point it may start two recording streams, each stream captures chunks of audio and write to a file. The buffer queue is processed asynchronously,...
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0answers
20 views

Multi-thread Audio Processing With C and JACK

I'm currently trying to create a NUPOLS (Non-Uniformly Partitioned OverLap Save) convolution reverb processor using C and JACK. According to the literature I've consulted; this is equivalent to ...
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1answer
28 views

Eliminate different microphones textures from their recordings

In my setting, I have different microphones recording a same speech signal. For each microphone recording, I can feel a slight difference in the playing sound. I am calling this property of a ...
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2answers
137 views

Getting a single feedback peak from a basic 4 stages phaser

I try to set a very basic 4 stages phaser (two 2-pole all-pass filters), with a particular feedback path where only one peak is raised at center frequency when the feedback is above 0. The following ...
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1answer
52 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
62 views

How to use MFCC (or any other tool/method/algorithm) for speech verification

I have an original speech .wav file and a few recorded samples of the same file. Each recorded version has some differences wrt the original file. For example, The ...
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2answers
233 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 ...
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2answers
51 views

What signal processing is done in music amplifiers that use non linear power supplies? [closed]

Some digital amplifiers for sound use simple non linear power supplies that lower voltage on higher loads. What is it doing to the input signal from the source compared to the signal sent to the ...
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1answer
40 views

xcorr MATLAB is not giving accurate result in case of noisy signals

I've a signal $x(n)$. I've generated its delayed copy named $x1(n)$ and applied xcorr function of MATLAB to find out the delay. It gives accurate delay. Now I ...
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1answer
52 views

Amplification of a speech signal in the recorded audio

I have recorded an audio. I dont know how it happened that only one sided speech is recorded and the other speech is recorded with a very low sound. Is there any solution to amplify the other side ...
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1answer
21 views

Beep Signal for Latency Calculation

I want to play audios on two mobiles with sync. But mobile phones are not in sync. I am trying to find two Mobile phones playing latency so I can pad latency on each phone so they can play together. ...
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1answer
53 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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1answer
48 views

Timeshifting in streaming based audio processing

I am currently running into a larger problem whilst programming an audio processing software. My target is time-shifting signals in frequency domain, then transform them back to time domain and ...
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2answers
32 views

Real time estimation of room impulse response using the sine sweep method

I am working on real time Room Impulse Response estimation using sine sweep. $x(n)$ is my sine sweep signal. $f(n)$ is its amplitude modulated inverse. $beep$ consists of 3 sine sweeps with a delay ...
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3answers
136 views

Modeling the human perception of noise

Add together some sinusoids with irrational or "nearly" irrational frequency ratios, and you get something pseudoperiodic. If there aren't too many, you can hear that it has only pitched components, ...
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1answer
56 views

How can I filter a noise with pattern type in audio?

I have a set of audio files rescued from a source of phonograph record like a gramophone. Very old family stuff records from my elders back in year 1946. A relative has digitalized these with a ...
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1answer
44 views

Need some references on sound quality evaluation and syncing

I have to do a project where I'm receiving the same sound signal from different channels (with different offsets), but they may not be consistent all the time. At any given time, I should decide which ...
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0answers
54 views

Voice glitch while selecting signal to play from a series incoming recorded signals

In my setting, I have multiple microphones recording the same speech signal in real-time. The speaker is moving while speaking. I am selecting one of the recorded signal to play based on their RMS ...
0
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1answer
78 views

Is the phase difference between recorded sounds from left and right ear the same as the phase difference between their HRTF?

Pardon my ignorance if my question is silly. Lets say I did a binaural recording and found the phase difference between right and left channel. If the recording was done in anechoic chamber, where ...
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1answer
101 views

Determining statistically the bandwidth of a signal from a spectrogram

My question is how to determine the parameters of a signal from a spectrogram statistically. The most important parameters that I am looking for are the lower and higher frequencies of the signal. I ...
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1answer
43 views

Varying shift between microphone data

I had given a sine wave of frequency 2kHz as input to two microphones placed near to each other. The data grabbed by two microphones are then plotted in MATLAB and it was found out that the initial ...
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0answers
17 views

Smooth transistion of music notes

I need advice regarding a small dataset of individual music notes played on a harmonica that I created a while ago. I want to build a system that reads notations in a text file and create realistic ...
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0answers
22 views

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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1answer
67 views

Varying phase shift between two signals

I have two microphones separated by a distance d placed in end-fire configuration. Audio input data was played from the front side (i.e corresponding to 0 degrees.) The data picked by microphone1 and ...
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0answers
43 views

Simulating traffic noise in Python

I am trying to simulate traffic noise for doing some measurements in Acoustic Vehicle Alerting Systems (AVAS). I am trying to figure out whether this can be done in the form of additive or ...
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1answer
135 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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