Questions tagged [audio-processing]

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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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16 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 ...
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54 views
+100

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 ...
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1answer
65 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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22 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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16 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
36 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
203 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
43 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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32 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
37 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 ...
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1answer
59 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
64 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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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
33 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
50 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
42 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
27 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 ...
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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
95 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
123 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
46 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
29 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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40 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
22 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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36 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
34 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. ...
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1answer
29 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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35 views

Simulating packet loss in audio

I have loaded a wav file into a Python Numpy array. I would like to "perturb the signal" such that the audio sounds akin to audio streamed over a poor internet connection. How can I mutate the ...
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1answer
20 views

Frequency-specific Delay?

So an equalizer manipulates the gain of certain frequencies by use of a variety of filters, as we all know. But is there a way to manipulate delay in a frequency-specific manner? A parametric ...
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3answers
91 views

What is difference between frequency and pitch?in nutshell and simple words?

Although both are some how directly related Such that when one increases,the other one also increases I have read different Google results but couldn't understand . I want to know what is exact ...
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30 views

Block Processing to Plot Phase in Real Time

I am trying to modify an LTFAT demo script that records and plays back streaming audio simultaneously. The objective is to read in the block to a vector, and take the DFT of the block samples, and ...
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2answers
37 views

Optimization of execution time of audio capture call in c/c++ (ALSA)

I am working on a Audio Deep Learning Project. In this C++ project I need to capture real-time audio and then do pre-processing and the pre-processed output is fed to the Deep Learning Network. I am ...
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21 views

DOA estimation with reverberation and directional interference

I have a 3D 4 microphone array. I wish to estimate the DOA of the source. I am working on a scenario with directional interference and reverberations. The first approach that comes to mind is using ...
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0answers
50 views

Beamformer implementation methods

I'm currently reading articles about the different type of beamformers. This spatial filtering is gonna be used for acoustic purposes, to focus the beam on the person we need to hear talking in a ...
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2answers
41 views

How to re-program/modify sound frequencies without DSP processor?

I'm developing a mobile application that controls Bluetooth sound-speaker functions, which includes an equalizer to modify the output audio frequencies. The audio processing module on the speaker ...
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1answer
56 views

Modeling audio effects, reverse engineering

Okay so I'm relatively new to signal processing and don't know the specific terminology to help narrow my search, but I'm looking for advice in the form of books, papers, references, etc that can help ...
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29 views

Audio in Image,Graph or WaveForm Representation

Our task is to feed the audio data into the deep learning tensor flow model in the form of image representation(or graphical,waveform). Question is what is the best way of representing audio in image ...
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1answer
34 views

Volume increase of a magnitude spectrogram

Let's say I have an audio waveform and I perform an STFT on the waveform and then take the absolute value to get the magnitude spectrogram. Is there a transformation that I can make to that ...
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1answer
60 views

Is it possible for Matlab to read and decode an mp3 file?

I have run the following Matlab code and I got a signal as shown in the Figure below. ...
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1answer
75 views

How convert Given DBFS into DB in iOS?

I am getting DBFS value of an audio ranging from -160 to 1. I am looking for to create a SPL meter in iOS to show actual DB values of audio capture through mic in device. Using recorder class, I'm ...
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1answer
35 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
51 views

The sound of modulated audio signal

As I was learned in signal processing, modulation took the signal spectrum to higher frequency range, so I made an experiment for audio file. The audio file "handel.ogg" plays a small part of music "...
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0answers
120 views

How to work with portaudio?

I am working in a project to record audio and process in real-time in C.For starting, I wanted to check how it works. I downloaded the pa_stable_v190600_20161030.tgz from here. I ran ,/configure and ...
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0answers
14 views

audio quality evaluation of Linear prediction resynthesized signal

Does anybody try the audio quality of linear prediction resynthesized signal using a music file? Using the frame length of 50 or 20 ms. And have you ever tried the objective evaluation using the ...
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0answers
17 views

ICA School Project

I have been stuck on my school project that consists of independent component analysis. My code will run, but only 50-60% of the time it unmixes the signals. Another problem I have is the amplitudes ...