Questions tagged [audio-processing]

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24 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
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 ...
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3answers
77 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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0answers
32 views

How to convert an audio signal processed by smartphone to PCM format using Android Studio? The ideia is to apply the FFT algorithm

I'm developing an app for Android that records an audio when we press "Record", play it when we press "Play" button, and shows it's spectrum of frequencies for this audio signal recorded, when we ...
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29 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
32 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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0answers
18 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
42 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
38 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
43 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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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 ...
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1answer
30 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
44 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
49 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
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!
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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 "...
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0answers
62 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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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 ...
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0answers
82 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
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1answer
66 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 ...
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2answers
98 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 ...
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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 ...
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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 ...
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0answers
24 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 ...
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0answers
188 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 ...
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0answers
47 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 ...
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0answers
22 views

Temporally synchronize two different or more real time digital audio signals

Are there any DSP techniques to temporally synchronize two different or more real-time digital audio signals without timestamp information?
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0answers
35 views

How to Synchronize multiple real time audio streams before mixing them?

I have been working on a VoIP solution in which I have to mix multiple audio streams coming from various sources. I have been facing muffled sound after the mixing of streams. I have implemented ASW ...
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4answers
858 views

Noise reduction using multiple recordings of the same signal

I have two (audio) recordings of one analog signal, I used two separate devices (smart-phones) recording from two different locations. Each recording device is pretty crappy, lot's of electrical hum ...
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2answers
166 views

how to remove noise from heart sounds?

i have .wav files acquired through putting android cell phone MIC directly on chest of person.For feature extraction, i have to made them noise free. These recordings were taken in rooms with no noise ...
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1answer
45 views

Cone of Confusion

In sound perception, humans adapt by tilting the head or relying on other location cues such as visual input when the source of the sound lies on the cone of confusion. On this cone, no time and level ...
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0answers
17 views

Hardware that Mixes Two Digital Audio Source into One

Is there any off-the-shelf hardware that extracts two digital audio streaming source from two digital audio stream(like USB+SPDIF), mixes, and transmits in single digital stream real-time(if the ...
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1answer
184 views

How can I reduce background noise from recorded audio conversation and separate speakers in python?

I have a recording of an audio conversation of two persons. I want to remove the background noise from audio and then also separate both persons' audio separately in python. I have gone through some ...
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1answer
209 views

Convolution between two vectors. Length and normalization

I have an RIR vector $h[n]$ with $N$ samples and an audio source $x[n]$ with $M$ samples. I wish to simulate a 5 seconds audio segment with $x[n]$ randomly located within (timewise). Using MATLABs <...
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1answer
231 views

Preprocess sound for voice recognition in Pocket Sphinx on Raspberry Pi 3

I want to improve my voice recognition system on a Raspberry Pi 3 running Pocket Spinx. It's pretty accurate (limited recognition vocabulary) but is terrible when other noise is around. Is there any ...
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2answers
23 views

Sudden loudness changes producing unwanted noises?

What's that effect called? Is that one of the reasons for having attack and release parameters when implementing synths? And is there a generic method to avoid such sound artifacts? Example: I have ...
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1answer
19 views

Arbitrary shaped RIR generator

I wish to simulate some sound sources convolved with a Room Impulse Response (RIR) and impinging on a spherical microphone array. I could find a couple of generators online (such as this one), though ...
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0answers
248 views

splitting real voice based on words using python or other program

Recently, I have a research project that need me to cut a large number of wav files containing the voice from movies, youtube videos or songs based on words for statistical analysis. The sounds are ...
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0answers
17 views

De-Doppler a signal: Audio Timestretch Factor controlled by function f(x)?

Timestretching usually is constant. Few programs offer bézier handles and such to control the lead-in and lead-out. But that's all. I need better. So the audio I have was in fact recorded in ...
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1answer
50 views

Yet another fft question - a vocoder

Lets assume a pipeline X: ...
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1answer
52 views

Attenuate the low frequencies in a speech wave

I am designing an FIR filter to attenuate the frequencies below 300Hz. I designed the filter using the ...
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1answer
350 views

Removing breaths using python

My wife is trying her hand at making audio books and spends a lot of time cleaning up some voice actors breaths in their "finished" audio. We have tried some filters in Adobe Audition and it helps but ...
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1answer
34 views

Need some references on sound quality evaluation and syncing

I have to do a project where I'm receiving the same sound signal from different channels (with different offsets), but they may not be consistent all the time. At any given time, I should decide which ...
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1answer
49 views

Audio Processing Glossary

I'm new to audio processing and I hope here be a good place to ask about the definition of terms in this context. My question is what is difference between sound, audio, voice, speech and tone, from ...
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2answers
103 views

Number of microphone limitation for Spherical Harmonics based DOA estimation

I am following this book "Theory and Applications of Spherical Microphone Array Processing", as well as this paper on DOA estimation "Eigenbeam-ESPRIT for DOA-Vector Estimation". My dataset is ...
2
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1answer
142 views

Overlap Add - which length to use?

I am doing realtime audio. Let's say I have a signal $x(n)$ which is passed in blocks of 512 (siobuffer) to the program filter/FFT method. Within this block I want to convolve the signal with the ...
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0answers
146 views

How to find frequency and corresponding time for an audio signal?

I want to read in a sound file and get the frequency at each time interval of the signal. I thought that the best way to approach this is by applying FFT to small chunks of the signal. Because FFT ...
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0answers
48 views

Robust custom watermark algorithms?

I'm doing research on image and audio watermarking, and so far I've read some literature in order to get basic introduction and good initial understanding of this topic. Right now I'm reading book "...
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0answers
13 views

Why can ARSS not recreate audio from a spectrogram without sounding muffled?

Why do ARSS and all audio to spectrogram then back to audio program's create audio files that sound muffled? I know their is some physics and math with the short-time Fourier transform and etc. I ...