Questions tagged [audio-processing]

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A python-based implementation of time-frequency block thresholding for audio signals?

I'm looking for a python implementation of the block-thresholding algorithm introduced in this classic paper. I've been searching a while but I couldn't find it :(. There is of course a Matlab ...
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45 views

Excessive low frequency component of electroaccoustic system impulse response

I'm using GNU Octave to achive speaker to my laptop microphone path impulse response. For this I use NLMS system identification algorithm. I've analyzed impulse response of system with fft to being ...
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3 votes
2 answers
87 views

SNR from 2 audio clips

I'm a beginner in audio signal processing, and I'm trying to figure out how to compute the signal-to-noise (SNR) ratio from 2 audio clips. I have two single-channel audio clips, where one is a ...
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9 votes
5 answers
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Why does the derivative of an audio file act like a high-pass filter?

Code: https://github.com/echometerain/sound-derivative/blob/master/sd.py I wrote a program which would subtract the current audio frame by the former one. Why does this act like a high-pass filter?
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Confusion regarding frequency spectrum in MATLAB

I am trying to observe frequency spectrum of an audio (mp3) file on MATLAB. The file'abc.mp3' is a 14 second music clip My MATLAB code is below: ...
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Accessing first few elements of a circular buffer in real time Python

I've made a circular buffer of size 10,000 in which I'm getting real time input from the mic. Now I want to access first 1920 samples every time and do the post processing but I'm only able to access ...
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Seeking expert on Android Dynamics Processing Effect for designing simple hearing amplifier [duplicate]

I am looking for someone who understand or has prior experience with Android Dynamic Processing Effect, to design a low-latency, frequency gain amplifier. Some public links for this Android feature ...
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4 votes
1 answer
293 views

Analysis of two audio signal by autocorrelation

I have recorded my audio by two hardware, PC and Microcontroller, I know the PC result are correct but my signal have some deficit. I have recorded by my PC and MCU my voice 6s 7812Hz and ...
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1 vote
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How to detect prematurely cut-off speech recording?

I'm searching for simple method to detect whether the audio recording is prematurely ended in the middle of the pronounciation of a word. I haven't found any literature on this topic, my idea is to ...
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2 answers
51 views

Which of an array's contiguous subarrays is an algorithm best applied to?

A function $f$ accepts two equally-long arrays $A$ and $B$ as input, and returns a real number $s$ such that the root mean square of $A-sB$ is minimal. I'm hoping to come up with a better-than-brute-...
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How to decided the score threshold in speaker recognition?

I am new to speaker recognition and now starting from the GMM-UBM algorithm. Say I have a pre-trained UBM and speech from target speakers ...
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2 answers
112 views

How do audio readers interpolate?

$124$ Hz sampled at $256$ Hz has "beats", but $128$ doesn't - a phenomenon of this question: Whether these are "legitimate" beats is up to interpretation, but suppose the goal is ...
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5 votes
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113 views

What is the relationship between beamforming and Independent Component Analysis (ICA)?

My first inclination when thinking about the Cocktail Party Problem would be to use adaptive beamforming to isolate different signals, but this does not seem to be how the problem is commonly thought ...
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How to generate PSHC carrier used in vocoders

Vocoders are often used to acoustically simulate the information transmitted by a cochlear implant. The different types of vocoders used in cochlear implants include noise-vocoding, sine-vocoding and ...
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1 answer
52 views

Is there a way to get the "mode" of several recordings? (to remove 1-off noise)

I've got several (4) not-great audio recordings of the same short (a few minutes long) performance, each taken from different microphones. I'd like one high-quality clip of the event. This is... ...
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1 answer
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Need for rectification in cochlear implants

In cochlear implants, the input signal is fed to a set of bandpass filters and then half wave rectified and low pass filtered to estimate the envelope signal. What is the need for doing rectification? ...
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2 votes
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Looking for audio/speech filtering expert for mobile hearing app

I am an Ear-Nose-Throat doctor (hearing doctor), looking for some help from someone who has built a mobile app with Frequency Filtering and Amplification. I am working on a social impact program for a ...
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1 vote
1 answer
70 views

Correct way to draw zoomable audio waveform

I am trying to implement smooth zoomable audio waveform but am puzzled with the correct approach to implement zoom. I searched internet but there is very little or no information. So here is what I ...
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1 answer
43 views

Is it correct to say that increasing the sampling frequency is same as compressing the duration of an audio signal in MATLAB?

I am trying to read and play with an audio signal in MATLAB. I have coded below clear [y,fs]=audioread('guitartune.wav'); sound(y,fs) This audio clip'guitartune....
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3 answers
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Why mixing two sines sounds like a triangle?

I'm not sure where to post this question that is related to audio synthesis through programmation. More specifically, I'm using a library called synthplayer that implement basic synthesizer in Python. ...
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Detecting whether the signal is increasing intensity or decreasing intensity

I am trying to determine whether the signal is increasing intensity, decreasing intensity, fade-in, fade-out or peak using an audio file. The graph has an x-axis with time and a y-axis with the data. ...
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51 views

How is the 'difference' of two audio tracks best computed, in order to maximise instrument isolation?

I'm hoping to improve the results of a simple application I've been working on, which aims to isolate as fully as possible the electric guitar heard (along with drums/bass/vocals) on a single track of ...
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1 vote
1 answer
43 views

Detection of Loopable Audio sample

I have a bunch of samples some of which are loopable [1]. I'm searching for a methodology to programmatically detect them. Any paper or approach is welcomed. Thanks. [1] https://www.wikiwand.com/en/...
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3 votes
1 answer
136 views

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 ...
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0 votes
4 answers
143 views

do audio compression algorithms encode channels in parralel?

I'm looking to compress multiple mono sound tracks together, as one track (with loads of channels), however, I was wondering if this is stupid - because I may lose out on real compression, as these ...
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What's the state of the art way to detect / change audio pitch? [duplicate]

Detecting pitch: It seems that selecting the modal frequency in the frequency spectrum is too simple an algorithm for reliable results. For detecting, and changing, the pitch of a note, what are the ...
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Separating Instruments Within a Music File - python code

Is there a way to distinguish between different instruments? Let's say I have a song that contains vocals, piano, guitar, drums (etc). I would like that song to be separated into different files, each ...
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1 vote
1 answer
29 views

Effective bit depth of mixed signals

I have three signals of 16bit signed integers. Each signal represents a sine wave of unknown (and variable) frequency / phase in the audible range (20hz-20khz) and the amplitude is anywhere between <...
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1 answer
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Fingerprint Recognition of Short Audio Samples

I'm working on a project related to fingerprinting of audios with different sizes. The library that I'm using is called Dejavu. Generally, it is the implementation of this paper that hashes the ...
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1 vote
1 answer
46 views

Similar voice features for imitated voice

What kind of audio signal features/properties are appropriate for signal similarity measurement invariant to imitation? Basically I would like to do the following: Having a template (e.g. a spoken ...
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1 vote
1 answer
166 views

Which Audio Codec/Bitrate Relationship delivers best Audio Quality: 128 kbit/s MP3 vs. 192 kbit/s AAC vs. 64 kbit/s OPUS?

Assumed we have those three audio streams available for listening to: MP3 with a bitrate of 128 kbit/s ...
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What is the best approach to cancel realtime noise from a participant audio for conference purpose?

I am new to audio signal processing and I wanna do real-time noise cancelation for a participant in a conference. I am capturing audio at 16bit, 16khz, little-endian config. I tried to use the webRTC ...
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2 votes
1 answer
54 views

How to implement Auto Gain Control (AGC) before and after mixng realtime audio for a conference?

I mix audio(16bit,16khz, little-endian, mono,frame-duration:60ms) packets after dividing the short values by participant count so it never gets clipped. but before that, I need to apply a certain gain ...
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0 votes
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44 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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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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0 votes
1 answer
58 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 ...
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3 votes
0 answers
136 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 ...
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1 vote
1 answer
25 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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4 votes
1 answer
105 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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1 vote
1 answer
343 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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2 votes
2 answers
149 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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1 vote
1 answer
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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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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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2 answers
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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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0 votes
1 answer
309 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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1 answer
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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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0 votes
1 answer
57 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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0 votes
2 answers
47 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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0 votes
1 answer
51 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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0 votes
2 answers
58 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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