Questions tagged [audio-processing]

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5answers
107 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 ...
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0answers
10 views

short frame subband coding

I am implementing a subband ADPCM codec, Fs=16 kHz, frame length 3ms, band1=50Hz-2kHz, band2= 2kHz-8kHz. I used the FIR filter for subband coding and %50 overlap. My reconstructed signal does not ...
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0answers
30 views

Newbie want to self-study to make guitar effect/amp plugin

I am a senior Game developer want to self-study how to create electric guitar amp/effect plugin. I read around and managed to made a simple distortion plugin using JUCE. But because of lacking in ...
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0answers
24 views

Communicate to user that SNR is too low - Speech procesing

I am working in a project to estimate the signal to noise ratio (SNR) for an automatic speech recognition system in order to generate an alert to the user if the noise is too high in the place where ...
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1answer
19 views

Dimension error using findpeaks function in Matlab on Power Spectral Density to calculate SNR of audio data

I have a set of audio files for which I would like to determine a SNR metric. I tried to implement in Matlab as follows: ...
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0answers
43 views

Monophonic audio processing [closed]

I need to find a tool written in python that given a monophonic audio file as input, returns all played notes' starting time. Let's say we have the "Twinkle Twinkle Little Star" performed on ...
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0answers
21 views

What is Ad-hoc microphone array?

I have stumbled upon this term in many papers, though no one bothers to elaborate. Is this the case where I do not have prior knowledge of the array structure? or otherwise is it the number of ...
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1answer
35 views

Which codec and format are Audible downloads?

Files downloaded with the Audible Download Manager are stored with the .aax file extension, but the information I can find about this format seem to vary. Regarding this source, the files are stored ...
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3answers
114 views

Simulating room impulse response between rooms

I want to simulate a sound source in one room and a microphone in a second, adjacent room. I am familiar with the image method and Habet's very efficient implementation for room impulse response (RIR)....
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0answers
22 views

How to identify the vocal folds in a mel spectrogram image?

I am quite new in the dsp and trying to learn How to identify the vocal folds in a mel spectrogram image? My image data looks like this. How to find the vocal folds in this image? I am using Praat ...
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1answer
46 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 ...
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0answers
25 views

Expand and shrink waveform visualization depending on tempo changes

I am not quite sure if this is the right state exchange site or not. Pls let me know. So, I noticed that Logic and Pro Tools, etc. have this waveform shrinking and expanding effect when dragging the ...
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0answers
30 views

Is it possible to open 2 streams from the same microphone on Oboe?

I'm trying to open 2 input streams from the same microphone on Oboe but it's not working. One of the audio inputs I'm opening is like this: ...
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3answers
60 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 ...
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1answer
30 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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0answers
15 views

Correct normalization of audio samples in time domain for classification

I am training an audio classification neural network. I wish to normalize the audio samples in the time domain before extracting features (MFCCs). I can: Normalize all samples to $\pm1$. Normalize ...
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0answers
19 views

How do I extract n-dimensional cepstral coefficients with a sliding Hamming window in PyTorch?

I am trying to replicate the features from this paper, described as follows: MFCCs of speech frames are computed using a mel-scale filterbank. We extract 20-dimensional cepstral coefficients with a ...
4
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1answer
58 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 ...
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0answers
18 views

Pitch analysis Audio Unit [duplicate]

I am developing an audio application for macOS that I would like to use to analyse the notes in a given section of music. I am happy to work from an analysis of the frequencies in a given section then ...
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0answers
22 views

In audio analysis, is the first rahmonic from the cepstrum equivalent to the first partial of a pitched sound?

I was reading this text from Oppenheim and Schafer, briefly explaining the concept of the cepstrum. I'm new to this concept. He explains that the cepstrum is useful to quantize the distance between '...
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1answer
27 views

Order of operations when creating a spectrogram

Signals noob here! I'm dipping my toes into your world by trying to recreate the Shazam algorithm, and I have a quick practical question: When creating a spectrogram, is it better to take in a whole ...
3
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0answers
62 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 ...
4
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1answer
68 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 (...
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0answers
26 views

How do I change the sampling rate without changing the speed and pitch in Python?

I am building a voice recognition algorithm, and the use case specifies that it should be able to identify the targeted speaker from various audio sources, which means that it must be able to process ...
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0answers
19 views

MPEG Audio Constant bit rate conversion

I am trying to convert few .wav files to .mp3 format The desired .mp3 format is : I tried with FFmpeg with this code : ...
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1answer
37 views

Continuously changing pitch of stable-pitch audio sample

I have an 44100-rate 4 second sample of a legato violin playing at 440 Hz. I would like to modulate this sample so that it starts at 440 Hz, then transforms its frequency continuously and ...
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5answers
215 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 &...
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1answer
49 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 ...
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0answers
37 views

How to correctly pass filter states from one processing block to the next

I'm processing a 4 channel audio signal and due to memory requirements need to process the signal in an arbritrary number of blocks. To ensure consistent filtering between blocks I've been told I need ...
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1answer
57 views

What is the actual format of this audio file?

I am collecting some data from jotform .wav widget. All the audio files are in .wav format. But when I tried to open the file using python ...
2
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1answer
65 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 ...
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1answer
67 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
14 views

Perceptual Based Adversarial Audio Attacks

I am reading this paper Perceptual Based Adversarial Audio Attacks I have a doubt what is meant by weighting in the frequency domain and what does Section 3.4-equation-4 wants to convey .Please refer ...
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1answer
39 views

Comparison of results with Direct form1 and Direct form2 implementation of IIR filters

The results obtained from Direct form1 and Direct form2 implementations (in C language) of IIR (BiQuad) filters are a little different. In the implementation, i have made the b0 coefficient as 1. So, ...
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2answers
51 views

Conceptually confused by LPC for speech: Do we synthesize by the inverse filter (FIR)?

I've been following the process of using LPC to analyse the speech and then synthesize the speech by swapping out coefficients over time. I'm purely concerned with pitched vowels at this point. ...
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0answers
23 views

Masking Threshold in psychoacoustics model

I am trying to calculate masking threshold using power spectral density but is is given in the attached document that global masking threshold is the sum of individual masking threshold and threshold ...
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1answer
52 views

Phase difference between two wave files

I am doing comparison between two wave files, How to calculate phase difference between those files? Second wave is generated by making some changes in the first one .I have tried calculating phase ...
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0answers
30 views

How can I detect the presence of echo in an audio?

The presence of echo during a phone call can be extremely irritating. Echo consists typically of the speakers sound being picked up by the microphone with a certain delay (10-200ms). One of the causes ...
0
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1answer
37 views

Calculate Collision force using Audio Signal

I am having an audio sample hitting an aluminium object (a small piece of hard ball 200g) at varying speed. I am trying to calculate the amount of force the object is hitting with using the audio ...
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2answers
103 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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2answers
178 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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0answers
27 views

PCM_SIGNED 8000.0 HZ, 16 bit, mono, 2 bytes/frame, 48000.0 frames/second, big-endian from PCM-SIGNED 48000.0 HZ, 16 bit, mono, 2 bytes/frame, big end

How can I resolve the following issue: Unsupported conversion: PCM_SIGNED 8000.0 HZ, 16 bit, mono, 2 bytes/frame, 48000.0 frames/second, big-endian from PCM-SIGNED 48000.0 HZ, 16 bit, mono, 2 bytes/...
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1answer
80 views

audio reprocessing for machine learning

I have a bunch of phone interviews where there is large sound volume difference between two voices in some of the interview audios, because they were recorded only on one side of the phone. May I ask ...
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1answer
40 views

Time stretching allowing for pitch change

I have implemented the WSOLA time stretching algorithm for audio data, but there are still many artifacts from the stretching. I don't actually care about maintaining pitch and am totally ok with ...
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0answers
25 views

Signal is not audible after frequency re-sampling

I'm trying to change the frequency of an audio signal from 22050hz to 500hz. I'm using librosa to read the wav audio and pass the custom sampling rate as 500. Once read, I'm saving it back to audio ...
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0answers
15 views

Classifying audio files by their instruments

I have a small set of sound files where each sound file contains a note or chord played on some instrument. Some of those sound files may be generated by the same instrument and I want to detect that ...
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0answers
42 views

Remove aliasing caused by scanning a wave file at arbitrary playback rates

I am writing a sound generator that needs to scan through audio samples at an arbitrary rate. When the sample is played at it's original rate, it sounds fine. But when I change the playback speed, it ...
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0answers
25 views

Remove known sound from signal

I'm currently getting FFT values from the music playing from a user's PC. However, my application also generates noise which interferes with the FFT data as my app's sound is also coming out of the ...
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0answers
84 views

How to write the program to compare two audio files

I want to make the program which would take one file and check the similarity with other. It would be music/melody file, and the program would ignore small deviations between two of them similarly to ...
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0answers
35 views

Finding the noise only part in a signal [duplicate]

I have signal which contains some noise only regions (glitch with the recording it seems). The question is: I would like to identify the discrete time samples where this noise part starts and ends. ...