Questions tagged [audio-processing]

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15 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 ...
1
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1answer
23 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
13 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
126 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
37 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
35 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 ...
0
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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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1answer
46 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
30 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
52 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
20 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
42 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
16 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
21 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
65 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
42 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 ...
0
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1answer
49 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
128 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-...
0
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1answer
27 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 ...
2
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1answer
30 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
38 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, ...
2
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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
32 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
164 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
22 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
34 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
16 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
68 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 ...
1
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1answer
50 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 ...
1
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1answer
40 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
51 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 ...
0
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1answer
24 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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1answer
46 views

How to filter sudden spikes and baseline changes in microphone signal?

The orange is a signal I recorded, the blue - my first restoration attempt, after which I decided to ask. The signal is audible, however there are 2 types of defects 1) baseline changes 2) sudden ...
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2answers
48 views

Peaking filter (loudness) cutoff frequency in an equalizer

An equalizer we have consists of Loudness ,Bass ,Mid and Treble .The loudness is nothing but a Peaking filter with cutoff frequency 20Hz .I read this in a book that outlines the EQ design. I was ...
0
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1answer
68 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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0answers
35 views

Algorithm to remove vocal from audio file

I have tried some tutorials with audacity but was highly disappointed with the result (to be fair the resulting output is not that bad, but not good enough for prod). What reputable algorithm could I ...
2
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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: ...
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0answers
31 views

Audio-Processing for DeepSpeech model

I try to understand audio signal processing from Deep Speech: Scaling up end-to-end speech recognition paper. Quote: Since the Switchboard and Fisher corpora are distributed at a sample rate of 8kHz, ...
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0answers
54 views

A mp3 file is reeoncoded in a video software (shotcut), which bitrate (Audio Bitrate vs Lame Bitrate) do I need to use?

I have downloaded a .mp3 file from the internet. "Right click" and "Properties" on Linux does not show up any bitrate (N/A) in the "Audio" tab. Then, I executed following ...
2
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1answer
45 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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1answer
60 views

Audio classification without deep learning

I am looking for code or papers, for methods to classify audio signals, without the use of deep neural networks (DNN). A current paper which reviews methods will also be most welcome. I have searched ...
0
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1answer
81 views

Why there is static noise and voice distortion after convolving it with impulse response?

I have been working on the decorrelation of audio signals while following this paper : The Decorrelation of Audio Signals and Its Impact on Spatial Imagery So far I have generated impulse responses in ...
1
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1answer
56 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 ...
0
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0answers
104 views

Low-pass filter + HRTF cascade audio artifacts

I am implementing a 3D audio application. I have correctly implemented a set of HRTF to perform binaural simulation and interpolation. To achieve this, I converted the HRTF into minimum-phase ones, ...
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0answers
323 views

Python: How to generate log-frequency spectrogram from an audio?

I want to make a log-frequency spectrogram out of this audio. Later, I need this spectrogram for pitch sequence analysis. This is a sample sequence I want to achieve: ...
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0answers
51 views

Comparing two audio files of the same sentence and determining if the speaker is the same person

I have a library of audio files where different speakers say the exact same sentence multiple times and I am trying to compare the audio files with each other and determine if the speaker is the same ...
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0answers
19 views

Seperate music instruments and estimate note density

Here is my problem: I have a .wav file of a musical performance with three instruments: a clarinet, a bass and drums. My goal is to detect the note speed played by the clarinet player (so I want the ...

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