Questions tagged [audio-processing]

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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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21 views

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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1answer
26 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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1answer
28 views

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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1answer
44 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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1answer
54 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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28 views

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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1answer
31 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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25 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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24 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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1answer
54 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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0answers
96 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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1answer
21 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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1answer
52 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
342 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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2answers
67 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
36 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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27 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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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
308 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
48 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
43 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
31 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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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
35 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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0answers
55 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 ...
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1answer
31 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
44 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
18 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
25 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
73 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
44 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
55 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 ...
3
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1answer
147 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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1answer
32 views

Static noise which changes based on which usb port I've audio interface connected to?

I've audio interface umc22. I thought I could hear some static noise Even with 0 gain. I recorded audio sample without any gain or even microphone attached. Yet there was still static white noise. I ...
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1answer
37 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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2answers
40 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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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 ...
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1answer
50 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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0answers
48 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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3answers
655 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
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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0answers
39 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
17 views

Training a NN classifier on a single\double channel out of a surround sound dataset

I want to train a neural network for a classification task on the VSD2014 dataset. I have downloaded the movies, but they have a 6 channel audio format (surround sound). 6 channels will cost a very ...
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0answers
90 views

Python library for audio filtering, gating, and sound/silence detection

Is there a Python library that can do all of the following?: Band-pass filter; Noise gate, with different parameters set for different segments of audio file; Detect and count the number of discrete ...
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1answer
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
62 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
63 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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1answer
57 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
25 views

Indian Space Research Organization - Extent of spectral spread 1/3rd Octave Analysis [closed]

I am a scientist at Indian Space Research Organization (ISRO) and will be representing my team. We work on Telemetry and Data Handling units for our Launch Vehicles. We are preparing for our upcoming ...

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