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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Remove a frequency and all its multiple (harmonics)

In a soundfile, I'd like to remove a frequency and all its harmonics. What is the best solution in your opinion ? Do a loop with 20 notch filters, one after another. // f is the fundamental ...
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2answers
3k views

Which filter for an audio equalizer

I’m new to DSP and I have to do an audio equalizer in c++. I did a lot of research about it and tried some stuff in the last month, but I’m a little overwhelmed with all those informations and it’s ...
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1answer
522 views

What are best practices to compute an audio spectrogram?

The spectrogram if generally defined with the squared magnitude of the fft. However, in lots of implementation, it seems that people just use the magnitude without square. Moreover an audio signal is ...
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1answer
781 views

What features describe audio signals? (Besides frequency and amplitude)

I recorded sounds with a microphone and I try to distinguish them in my Java program. The frequency works quite good, but if I look at the fourier transforms it seems like there should be more ...
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1answer
138 views

4-phase FIR LPF in MATLAB

I'm trying to create a low pass FIR filter using the coefficients I found in ITU 1770-3 on page 18. This is the last piece of the puzzle for an audio loudness metering algorithm I'm working on. I'm ...
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2answers
2k views

IQ FM Demodulation of Wow and Flutter Signal in Software

Background: I have a recording device that exhibits Wow and Flutter, or periodic frequency deviation. I use this device to record a calibrated 3.15kHz test tone from a tape film to generate a .wav ...
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3answers
3k views

Explanation of binning in frequency analysis post-FFT

I am creating an Android app which records sound for t seconds at a sampling rate of 44.1kHz with a buffer size of 8192 (16 bit mono samples). I need to plot a ...
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1answer
475 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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493 views

inverse FFT when only a few time points are needed

I am developing C++ code to do some modeling that will take as input a parametric model of the complex frequency response (amplitude, phase) and produce as output an impulse response, e.g. using the ...
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3answers
2k views

Convolution and Deconvolution in C

I understand the theoretical foundations of convolution, but now that I'm trying to program it I'm having some issues conceptually. Say I have two blocks of 64 audio samples each. I also have access ...
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1answer
849 views

MPEG-7 low-level audio descriptors

I'm working on an application which need to use the full set of 17 MPEG-7 low-level descriptors. Is there any library I can use for it? Preferably with a Python wrapper/API...
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1answer
55 views

Removing background noise algorithm

What is the algorithm for removing background noise from an audio clip of human speech? Say I have a 10 second audio clip of somebody talking, and there is background noise of them tapping on their ...
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478 views

Audio output quality

When a digital device (PC, portable media player, etc.) plays a digital audio file (ogg, mp3, flac, etc.) the audio signal output is always the same, regardless of the device type/brand, am I right? ...
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Non-negative matrix factorization for audio separation - why does it work?

Non-Negative Matrix Factorization aims to factorize a matrix $\mathbf V$ into the product of two matrices, $\mathbf V = \mathbf W\mathbf H$, where $\mathbf W$ represents a set of basis vectors and $\...
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4answers
4k views

Complex conjugate and IFFT

I asked a question over on stack overflow. I'm having a slight problem however. As suggested by Paul R I am mirroring my lower $n/2$ bins into the upper $n/2$ bins. I have a few questions however. ...
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2answers
3k views

High audio frequencies in television broadcast

I would like to know if in television broadcast high end audio frequencies are transmitted e.g. frequencies higher than 14 kHz. What is the cut off audio frequency of transmission in television ...
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1k views

Subtracting Original Audio Signal

here is my scenario: one speaker is sending a chirp, and a microphone is recording it as well as all following echoes. The function of the chirp is already know. My goal: subtract the original chirp ...
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418 views

Why are the frequencies associated with the STFT bins not representative of the energy?

I'm experimenting with a 30 Hz sine wave and I want to extract the frequencies of this, and have implemented the algorithm to do so. This gives a correct reading, well, gives 29 Hz. I'm now wanting ...
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1answer
287 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
924 views

How to mix 8bit/8kHz unsigned linear PCM data in AppsScript?

I am having the most difficult time mixing simple unsigned 8-bit/8kHz linear PCM audio samples in JavaScript (AppsScript). I've tried all the basic maths listed here, employing basic number arrays. ...
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1answer
585 views

Why are MFCCs of two equivalent signals completely different?

I have an audio file, of which I calculate 16 (actually 15 because I omit the first one) MFCCs in R. When I stream this file via VLC-Player and Icecast2-Server, receive it in Java (with the Player of ...
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1answer
731 views

Division by zero in Levinson–Durbin recursion?

I am doing a LPC analysis of a speech signal using the autocorrelation method. To calculate the LPCs I am using the Levinson–Durbin recursion. In my literature the error is initialized with the first ...
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1answer
286 views

How to build a realtime simple audio spectogram? [closed]

I'm trying to build a spectogram view of history time of 4 seconds and I'm stuck at the part where you load/draw FFT vectors into the 2D bitmap. I just don't know how to do that, but i got the FFT in ...
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2answers
422 views

Extending Goertzel algorithm to 24 kHz, 32 kHz and 48 kHz

I'm learning to implement Goertzel's algorithm to detect DTMF tones from recorded wave files. I got one implemented in python from here. It supports audio sampled at 8 kHz and 16 kHz. I would like to ...
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1answer
129 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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1answer
133 views

Multichannel input/output evaluation board

I'm working on a project at school and the first step is to choose a dsp/evaluation board but I'm having trouble. The project is to create a system that can be used for a variety of audio applications ...
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0answers
86 views

Determining the probability of two signals, with added independent noise, being the same?

I have the first 100ms of two signals, each with a noise independent to that signal. I want to compare the two signals and determine the probability that these two signals are (minus noise) the same, ...
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115 views

Envelope reconstruction algorithm for audio processing

I need to reconstruct the envelope of a sound. Audio data are professionally-recorded natural sounds (speech, bird songs) with very little noise. I would prefer working in the time domain rather ...
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1answer
5k views

Noise removal from audio using FFT/Spectral editing

I have a noisy recording with some letters spoken out loud. I've noticed that the noise is easily recognizable when looking at a spectrogram of the audio. Is there anyway to set a power/frequency ...
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4answers
2k views

Convolution and windowing using a buffer - how do I do overlap add?

I'm reading chunks of audio signal (1024 samples) using a buffer, applying a Hanning window, doing an FFT on it, then reading an Impulse Response, doing its FFT and then multiplying the two (...
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1answer
621 views

Filter is too large

Hi I'm trying to separate sound from music using wfilters and heard each part separately using its name scale3 function ...
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1answer
128 views

What is the name of this very simple spectral subtraction technique?

Let $S = X + N$ be the sum of two audio signals $X$ and $N$ which are both stationnary (let's think X is a constant volume 440 Hz sinusoid and N is constant volume noise). If the sum S has a -20 dB ...
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2answers
521 views

Audio Separation of .wav signal [duplicate]

What I am attempting to do is take a signal (.wav audio file) which contains calls from different Bat species. Here is a graph representation of the signal: What I have done so far is: 1) Emphasise ...
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805 views

Distribution of energy in frequency bands

On my wav-file (music), I do a short-time Fourier transform (STFT). I have spectrum, which spans from $0\textrm{ Hz}$ to $f_s/2$. I took out the range between $1$ and $1000\textrm{ Hz}$, which I ...

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