Questions tagged [audio]

Audio, or in terms of signal processing, an audio signal is an analog or digital representation of sound, typically as an electrical voltage.

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30 views

Audio Processing Used in Voip Mobile Apps

I am new to VoIP apps and am learning about the various audio processing that are applied in such apps. I have looked at Linphone and ...
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1answer
50 views

About speech preprocessing in mobile phones

I understand that cell phones transmit speech signals in digital form through 2G/3G or other standards. But I could not find anything about the signal processing method lying in between the analog ...
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2answers
66 views

Is it accurate to say in digital audio that, when a fader is down, then its value is “-$\infty$”?

Is it accurate to say in digital audio that, when a fader is down, then its value is "-$\infty$"? What has confused me is, whether it's really $-\infty$ or some lower bound of the amplitude. it could ...
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0answers
812 views

Cocktail Party Problem — audio source separation

I am trying to solve the Cocktail Party Problem. I am trying to separate these two mixed audio files: mixed 1 [WAV] mixed 2 [WAV] into 2 separate audio files that contain the 2 original sources, ...
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0answers
64 views

Digital vs Analog music [closed]

What are the different constraints for mastering audio content for either Vinyl or CD? By the way, I'd like to know if there're any papers or documents, with a comparison between different digital ...
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1answer
131 views

How to delay an audio signal by 1 microsecond? [duplicate]

I have an audio signal sampled at 44100Hz. I want to delay this signal by 1 microsecond (and maybe even less) for a steganographic purpose. Now if I delay it one sample, the corresponding time delay ...
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1answer
66 views

How many psychoacoustical critical bands are there?

I understand that the human ear works as a band-pass filter and I have doubt regarding the number of such bandpass filters in the filter bank. I came across these two possible explanations of how it ...
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2answers
328 views

Software to extract frequency spectrum data from an audio file

I've been trying to learn how computers recognize musical notes, and I found apps like spectrogram which shows the frequency spectrum of a sound, but what I need is the data which is plotted or the ...
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1answer
165 views

Hardware requirements for ambient active noise cancallation system using adaptive LMS

Background I'm working on a hobby project to reduce ambient low-frequency noise generated from HVAC equipment within a small area of a room. The noise is predictable and concentrated between 55Hz and ...
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2answers
201 views

how to calculate the maximum obtainable directivity for a microphone array?

Introduction The relationship between the maximum obtainable directivity and aperture of an antenna is described by: $$ D = \frac{4\pi A_e}{\lambda^2} $$ Where $D$ is the directivity, $A_e$ the ...
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0answers
92 views

Subtracting audio signal emitted - trying to use spectral subtraction to localize moving objects

I am a Software Engineer without much signal processing background and currently spending and experimenting to get use to it. My scenario: Assume a speaker and a microphone array. A speaker emits an ...
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0answers
62 views

Given a log-plot of frequency-magnitude-phase how to apply it as an EQ curve to a signal?

I have several hundred data points that represents an EQ correction curve, where each point contains frequency, magnitude, and phase (-1..+1). The frequencies are distributed exponentially, not ...
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3answers
121 views

Resample audio signal with a low factors

Currently, we need to handle a piece of an audio signal. We want to resample it with different factor rate (p/q) starting with 1.0002 to 0.9998 with a decrement of 0.000005. But Matlab function cannot ...
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1answer
46 views

How to generate such sounds as in example? [closed]

I found these files on Internet: 1.wav 2.wav 3.wav 4.wav How to generate such sounds? Geno Chen wrote "Acoustic Grand Piano (Instrument #0)" What to do next?
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2answers
160 views

Audio Steganography: Inaudible Audio Watermarking for Source Identification

What is a relatively inaudible audio watermarking technique to overlay a broadband audio signal on top of playing music without analysing the music for low-bitrate source identification? Ideally ...
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1answer
107 views

How is temporal pre masking possible?

I was just learning about psychoacoustic audio data compression and came across some sources suggesting that a strong tonal component (masker) that occurs within a certain short timespan after a ...
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1answer
178 views

No cross-correlation between same signal recorded using two different microphones

I've recorded the same audio signal using two microphones located at the same approximate location. However, when checking the cross-correlation using MATLAB (using the xcorr function) it is ...
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1answer
84 views

Introductory resources to mathematical properties of “dynamically recalculable” filters in audio/musical equalizers?

Introductory resources to mathematical properties of "dynamically recalculable" filters in audio/musical equalizers? Particularly, I want to understand What mathematical features (e.g. monotonicity ...
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1answer
195 views

Where to begin? - Multi source polyphonic pitch detection

I'm quite interested in real time pitch tracking. I have used mcleod pitch method, Yin and other methods to perform real time pitch tracking for a monophonic audio track. I know that Polyphonic ...
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1answer
79 views

nearest neighbor implementation matlab

I am trying to implement this research paper: I understand everything in the paper except the part about the nearest neighbors calculation (section 3). The only MATLAB function I can think of that ...
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1answer
3k views

Calculating the SNR of Audio Signal (Recommended Libraries)

Are there any open source packages or libraries available which can be useful in calculating the SNR(signal to noise ratio) of an audio signal. The input will be just an audio signal and I have to ...
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0answers
235 views

What are F0 counters in Speech signals? What is represented by F0?

I can't find any dedicated definition for Frequency F0 counters. Can anyone tell me what does F0 exactly mean, I am confused about that.
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3answers
248 views

Decreasing Sample Rate in DFT (FFT) for Audio Analysis

I'm working with audio data analysis through FFT algorithm. My example audio is a sine wave at 440 Hz and 44100 Hz sampling rate. FFT methods in programming, like ...
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1answer
43 views

How to recognize for small video sequence with a large sequence?

I'm working on the idea of recognizing a small video sequence (a sequence of 18 seconds) within a long one (a sequence of 15 minutes), the first idea that come to mind stats that I have to ...
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2answers
2k views

What are some python packages I can use to cut audio files

I have a project in which I have a batch of audio files and I need to remove the audio in it from say time 2secs to 5secs i.e. I have a set duration to be removed. How can I do this in python?
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1answer
200 views

Morphing between two FFTs

I'm a musician and I'm making a script that takes a wave file and snaps each of its frequencies from the fourier transforms to the nearest musical harmonic. Thanks to help from another question I ...
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1answer
79 views

DSP using audio IO from PC using C++

This is my first question on dsp. I am looking for something very simple here and I am sure there are libraries that will do pitch recognition and FFT but my requirement is very simple. I have a PC ...
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0answers
84 views

Generalized Sidelobe Cancelling performance

I'm tinkering with different adaptive beam-forming algorithms for a research project in which I want to use a Uniform Rectangular microphones Array (URA) to isolate speech in a room. I am determining ...
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1answer
698 views

How to simulate analog VU meter response to audio?

I'd like to simulate the movement of an analog VU meter dial as shown below and often found on stereos and other music monitoring equipment. I just learned through this article by Rod Elliot (ESP) ...
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1answer
148 views

Audio transfer function modelling with pink noise + signal

I'm trying to model a transfer function for a noisy audio system, specifically to measure delayed system response. Before I can confidently apply control, I need to verify that I can exert control ...
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3answers
135 views

Detect the beginning of an increasing signal

After denoising and cleaning, I get amplitude signals like this (y-axis: dB): On bottom left of each of these 3 graphs, you can see a noise floor (nearly "horizontal line"). This noise floor ...
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3answers
256 views

How to calculate arbitrary phase shift in discrete signal?

When I have digital audio signal, represented in samples, I can easily phase-shift the signal by 180° by simply inverting the value of each sample. But what if I want to phase shift an arbitrary ...
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2answers
403 views

Amplitude ranges for different bit-depths

In Python, I am using the library Soundfile (which uses the library libsndfile) for digital sound processing. I am working with different sound files coming from different internet databases, and ...
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2answers
296 views

How to edit audios so that they have the same length

I have a dataset of audio files, which differ on their lengths by a couple of milliseconds. Some are little shorter than 1s and some are little longer than 1s. They all have the same sampling rate, ...
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1answer
1k views

How to convert NIST to RIFF wav file

I have a dataset of audios(namely TIDIGITS, collected at Texas Instruments, Inc.) and they are in .wav format. However, when I try to read them from python, I get the following error: ...
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0answers
122 views

Is it possible to “equalise” a signal by deconvolving the impulse response of the room in which it is to be played?

I am using a test sweep with a flat power spectrum and linear group delay (Optimized Aoshima's Time Stretched Pulse) to measure a room's frequency response. Having obtained the impulse response of the ...
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2answers
54 views

What should I do with FFT result array from microphone input? [duplicate]

I currently want to create a small pitch detector and I have to say that I am very new in this topic of signal processing. At the moment I implemented the FFt algorithm and I got an array of window ...
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2answers
70 views

Noise when dividing or multiplying signal in C

I'm trying to create a echo effect and have managed to add multiple delays but when i try to divide or multiply these numbers the signal is completely distorted and adds a lot of noise to the system. ...
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1answer
166 views

Add noise/signal to mel log spectrogram

I have a predicted mel log spectrogram $S$ and I want to add a noise signal $\eta$ to it. S = log(mel-filterbank(abs/real part of(stft(signal))) The naive way ...
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0answers
38 views

Relationships between $f(t)$ signal and $f(t) \times \sin(\omega t)$ signal

Let's assume that there is a signal $f(t)$, $0\le t \le T$. Then, is there any special meaning of the signal $g(t) =f(t) \times \sin(\omega t)$? Like AM transmission. What is shown/heard when we play ...
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1answer
112 views

audio filter designing

I know the basic filtering techniques available in signal processing.Does audio signal specifically have any different filtering method?.My application is to filter out the background noise in speech ...
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4answers
78 views

Circular buffer wrap around with chorus effect

I am developing an embedded DSP audio processor and am trying to implement a chorus effect. To my understanding, the chorus effect is multiple delayed versions of the original signal, where the delay ...
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1answer
299 views

Noise removal using adaptive noise cancellation algorithms in real time systems

I am newbie in signal processing. I want to remove noise from real time audio signal using adaptive noise cancellation(LMS&RLS) algorithms. I don't know how to choose reference signal(noise signal)...
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1answer
63 views

Getting a weired behavior when windowing an audio signal

I am processing audio signals using Essentia, an open-source library and tools for audio and music analysis, description and synthesis. Here is a waveform corresponding to a frame extracted from an ...
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0answers
64 views

How to build a software audio mixer - low level approach [duplicate]

my questions might come across different SE forums. But here it is. A little context : I'm currently building a midi sequencer, the architecture is this way (for now) I've got a ARM v7 + a PIC, the ...
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1answer
35 views

Is sparsity induced penalty in source separation “Entrywise matrix norms”?

I am reading this paper where they introduce norm penalties for source separation. In table 1, the $\log/ l_1$ type is $\sum_{g} log(\epsilon + \lVert H_{g} \rVert_1)$. I wonder this $\lVert H_{g} \...
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1answer
333 views

Machine Learning - any suggestions to solve Python rounding errors?

I am working with Python to isolate voiced segments from music using the Jamendo corpus for singing detection. Training a model, I break my audio into frames, and have a label (0,1) for each frame. ...
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1answer
99 views

FFT phase spectrum to suppress noise?

All that I heard so far about phase spectrum - that it is very hard to process. Are there any methods to do that, and is it possible to extract information about the noise from phase spectrum?
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0answers
19 views

Find which groups of ants are coexisted in the location from their signal profiles using NMF

I try to solve biological problem to tell which groups of ants are co-existed at this location by their sounds in specific time frame. My chosen framework is Non-negative Matrix Factorisation (NMF) ...
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1answer
81 views

How to find when a part of audio starts and ends in another audio?

I have two audio files in which a sentence is read (like singing a song) by two different people. So they have different lengths. They are just vocal, no instrument in it. A1: Audio File 1 A2: Audio ...