Questions tagged [audio-processing]

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22 views

Is there an industry standard audio data processing library for C/C++? [closed]

I'm new to audio signal processing and want to work towards using DSP to develop effect/synthesizer plugins. Are there any industry standards I should know about going into this type of project? ...
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30 views

Why do wav files have the same initial values? [closed]

I have three wav files recorded with different devices. All of them are mono sound and 16 bit per sample. One of them has sampling frequency of 22050 Hz while the other twos' is 44100 Hz. When I open ...
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1answer
29 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
41 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
27 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
48 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
38 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
12 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
16 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
75 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
61 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
62 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
1k 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
22 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
139 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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43 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
18 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
25 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
835 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
111 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
42 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
16 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
87 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
96 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
178 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
21 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
17 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
184 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
48 views

Yet another fft question - a vocoder

Lets assume a pipeline X: ...
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1answer
48 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
243 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
33 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
47 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
98 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 ...
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1answer
98 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
113 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
45 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
10 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 ...
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3answers
98 views

Using different algorithms/formulas to create spectrograms

On the internet, I have seen people use Matplotlib/Python and short-time fourier transformations to create spectrograms. Are their other formula/algorithms to create spectrograms? And are they ...
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4answers
222 views

What are some alternatives to Spectrogram charts? [closed]

Spectrogram renders of audio are limited in their representation of audio. For example, one could not use a Spectrogram to re-render the original audio. What are some alternatives to Spectrogram ...
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1answer
23 views

Is it possible to have high energy compaction in audio signal? [closed]

We all know that the energy compactness in image can be achieved in high scale.But in case of audio signal is it possible to have such high energy compactness?
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1answer
36 views

From pressure to SPL : microphone calibration using sound level meter calibrator and sound level meter

I have a microphone + signal conditioner with adjustable gain and filter. I capture amplified signal using oscilloscope. I'm using this signal to compute equivalent continuous sound level With signal ...
2
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1answer
143 views

How to interpret these MP3 spectrums?

Recently, I found I had this song duplicated in my music library, but when listening to them I noticed one version was of noticeably higher quality than the other despite both claiming to have a ...
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0answers
21 views

Blind source separation from microphone array

I was following this article and I wanted to "borrow" their Idea for my own work. The difference is that I am using a microphone tetrahedral array and not a binaural microphone. I have two different ...
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1answer
72 views

Distort voice to hide talker identity as much as possible

I need to develop an app which makes speech recognition. Because I need to be able to constantly improve the recognition capability, I would like to train the engine with real recordings and match the ...
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2answers
306 views

audio spectrogram normalization

I am performing language classification from audio signals using mel-spectrograms as my inputs for a ResNet. It works well as long as all of my audio data from different languages is from the same ...
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0answers
31 views

Comparing two audio (Phone operator messages)

I have this two wavs. Same audible audio differente data. The audios are phone operator messages, so they are always the same. Wav:http://instaud.io/3Ezu Wav:http://instaud.io/3Ezw Using this ...
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0answers
31 views

What is the difference in the algorithm for noise reduction in SoX and Audacity?

I've looked at this question, and I understand that SoX works way FFT of the noise, then subtracting that from the FFT audio, and then resynthesizing using FFT, but I don't quite understand how this ...