Questions tagged [audio-processing]

The tag has no usage guidance.

Filter by
Sorted by
Tagged with
0
votes
0answers
28 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 ...
0
votes
1answer
29 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 ...
0
votes
0answers
14 views

How to convert audio buffer stream (of type '_io.BytesIO') to audio data? [migrated]

I'm trying to access the audio of a YouTube video to perform some operation on it. The goal is to capture the audio data and perform the desired operation rather than downloading audio file and then ...
2
votes
1answer
47 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 ...
0
votes
1answer
49 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:...
0
votes
0answers
12 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 ...
0
votes
1answer
31 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, ...
0
votes
2answers
42 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. ...
1
vote
0answers
22 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 ...
0
votes
1answer
30 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 ...
0
votes
0answers
26 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
votes
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 ...
0
votes
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 ...
0
votes
2answers
99 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 ...
0
votes
0answers
25 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/...
0
votes
1answer
38 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 ...
0
votes
1answer
26 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 ...
0
votes
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 ...
0
votes
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 ...
0
votes
0answers
38 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 ...
0
votes
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 ...
0
votes
0answers
19 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 ...
1
vote
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. ...
0
votes
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 ...
0
votes
0answers
31 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 ...
0
votes
1answer
19 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 ...
2
votes
3answers
89 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 ...
0
votes
0answers
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 ...
0
votes
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 ...
0
votes
0answers
20 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 ...
3
votes
0answers
48 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 ...
0
votes
2answers
39 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 ...
0
votes
1answer
51 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 ...
0
votes
0answers
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 ...
0
votes
1answer
33 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 ...
0
votes
1answer
59 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. ...
-1
votes
1answer
66 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 ...
0
votes
1answer
34 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!
0
votes
1answer
49 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 "...
2
votes
0answers
100 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 ...
0
votes
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 ...
1
vote
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 ...
0
votes
0answers
83 views

extracting mfccs after beamforming

I have a deep network for sound classification with mfccs as features. I wish to go through the process of beamforming before applying the classification. The original network uses ...
3
votes
1answer
74 views

Count heart beat from audio capture using smartphone

I am working on a project where I am required to perform heart rate BPM calculation from an audio signal captured using smartphone microphone / hands-free. Constraint is not to use ML. So I am left to ...
0
votes
2answers
137 views

Sine wave amplitude modulation - Analysis using Audacity

I am trying amplitude modulation in C++ on a sine wave which is a 1000 Hz sine wave sampled at 48000 Hz. The original sine wave is like this (In Audacity) - When I basically double the amplitude for ...
5
votes
3answers
2k views

How do impulse response guitar amp simulators work?

I am wondering how impulse response guitar amp simulators/modelers work. I thought it was a matter of convolving a signal of recorded impulse response in time-space with a guitar sample. I tried to ...
0
votes
1answer
14 views

simulating directional interference to evaluate model performance

I have a model for audio processing. The model receives as input, a source, convolved with a room impulse response (RIR). I would like to test the performance of the model against inputs containing ...
0
votes
0answers
25 views

How to understand the sound parameters so that it can be reproduced in Matlab

I am reading a paper called "Mouse Auditory Brainstem Response Testing". In order to make the experiment myself I would like to create the stimulus sound using matlab. When describing the sound (in ...
1
vote
0answers
219 views

Problem using pitch shifting with TD-PSOLA and formant preservation

I tried out this implementation of TD-PSOLA (https://github.com/sannawag/TD-PSOLA) and even though I know how the algorithm works, I can't understand why I get NO audible differences in the output ...
0
votes
0answers
51 views

Audio Activity Detection and Classification

I am starting a new project. Actually, my real intention is to learn speech recognition but for warming up I want to improve in audio signal processing. In my project, I aim to record sounds except ...