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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Digital Distortion effect algorithm

I read the DAFX book by Udo Zölzer about the distortion effect at pages 124-125 and it says that suitable simulation of the distortion is given by the function: $$f(x)=\frac{x}{|x|}\left(1-e^{x^2/|x|}...
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5answers
886 views

Mathematical question that comes out of using bilinear transform

So this is related to the Cookbook and I tried solving it maybe two decades ago, gave up, and was reminded of the unsolved problem. But it's pretty damn straight forward, but I still got slogged down ...
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1answer
574 views

What are the basic categories of operations performed on audio signals to create interesting sounds?

I don't know if this question will make sense as I'm very new to dsp. In my limited foray into audio dsp I've come across: Frequency Modulation Amplitude Modulation Additive Synthesis Subtractive ...
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1answer
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Creating a spectrogram

I've been trying to work out the logic for this task, and plan to use the KissFFT source package to perform the fast fourier transform. Please let me know if this looks about right: Allocate an FFT ...
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1answer
872 views

Ultrasonic speaker calibration and emitting calibrated signals

I'm trying to calibrate an ultrasonic speaker with the aim of emitting predictable signals, but alas I keep running into trouble, probably due to my lack of DSP-fu. A little background I want to be ...
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812 views

Filtering of streaming audio

Given something like 103 data points ($N=103$), a DFT will give back 103 frequency values. Then to do something like filtering the high frequencies involves setting the high frequency values from the ...
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4k views

How to check the FFT results of a sine wave?

I have been given an audio file (sine wave) 1000Hz as an input to my FFT algorithm. I have got 8192 power spectrum samples in a array. What is the best and easiest way to check whether my output is ...
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Why is telephone audio sampled at 8 kHz?

When did we decide to sample telephone at $8$ kHz? Has this always been the case? Why did we do that? Is it because higher bit rates can't be transferred as quick? And do these reasons still count? ...
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2answers
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How to analyze audio input/spectrum correctly

I am quite new to Java and Signal Processing, but I was assigned a project dealing in audio processing. My topic given to me is a game that has a component of analyzing a song (any given song) and ...
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3answers
24k views

Alias frequency Formula

I'm taking a multimedia systems class in my MSc Computer Science, and I'm having some trouble understanding the formula for the alias frequency - this could stem from my misunderstanding of the alias ...
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1answer
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1/n octave smoothing

Given a frequency response obtained with FFT, I would like to apply a 1/n octave smoothing. What filter should I be using and how? Maybe someone could point to a good reference (a paper or book on the ...
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Low pass filtering on short int (16 bit PCM) samples

I am writing software for processing audio given as 16-bit PCM samples. The first stage of the processing involves calculating the energy (or total variation) in a certain frequency range (above a ...
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1answer
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Measuring time delay of audio signals

Before anyone yells at me I fully realize that this question has been asked numerous times. I assure you that I've read through the existing questions and answers but I am still confused about part of ...
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Logarithmic fourier transform(LFT) on audio signal

I am trying to analyze the musics as possible as precisely. Of course I tried FFT, but got some problems. I found low frequencies have very low resolution than human's hearing. I tried very long time ...
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1answer
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Discrete Fourier transform - finding the fundamental quickly?

First, I appologize because I'm a software developper and it's been a very long time I didn't dive into pure mathematics, so my question might seem dumb. I hope not. The context is pitch recognition ...
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Noise reduction from very noisy audio

I am trying to write an algorithm that would automatically segment a piece of audio with bird calls recordings. My input data are 1 minute-long wave files and on the output I would like to get ...
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2answers
618 views

Pattern recognition for temporal data

I'm trying to detect and classify non-speech sounds. Currently, I'm using a series of moving overlapped power spectrums from training sounds as the features I am looking for. When I do analysis, I'm ...
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3k views

Synchronizing Two Audio Tracks

I would like to implement an algorithm that synchronizes two audio tracks. The two tracks are very similar; you can imagine they are two recordings of sound (not necessarily music) from two ...
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318 views

How to segment phone call audio into silence/non silence?

My problem is that I don't know the energy of the background noise, so I can't just threshold the energy. The processing is done in real time, and I have about 500msec to decide. Ideally, I'd want ...
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How do I go about detecting whistles, pops and other sounds in live audio input?

I've read many questions on SO, and frankly, each one of them is not describing any particular way to go about it. Some say "do FFT" and some say "zero crossing" etc. But I've only gone as far as ...
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1answer
362 views

Recognizing data clustered into shapes

I am working on a project in Python to detect and classify some bird song, and I have found myself in a position where I need to convert a wave file into frequency vs. time data. This hasn't been too ...
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1answer
2k views

Frequency Modulation Synthesis Algorithm

Based on what I read, I've made an algorithm for FM sound synthesis. I'm not sure if I did it right. When creating a software synth instrument a function is used to generate an oscillator and a ...
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1answer
724 views

Detect human speech in realtime audio on mobile phones

I am looking to develop an Android app. As part of the functionality the app would require to randomly sample 3-5 seconds of audio and classify it as containing human speech or not. I understand that ...
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472 views

Can you represent an audio signal in any other domains besides time and frequency?

Maybe a stupid question, but if there is a time-domain representation of audio signal, and also frequency-domain, so is there any other domain that signal can be represented in?
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What is spectral entropy?

Can anyone explain what spectral entropy is? Does noise with a restricted bandwidth have the same spectral entropy as white noise?
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Speech comparison algorithm for rating on similarities

I am trying to compare 2 speech samples and rate them on similarities. Think of someone trying to repeat a phrase, and then comparing those 2 audio files. I started by implementing the MFCC (http://...
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5answers
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Piano pitch detection

I'm working on a piano tuning program and part of it requires real-time pitch detection. Here is the scheme I have so far which works to some degree but could probably use some refinement. I'm ...
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2answers
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Fast pitch recognition

I need to detect pitch (measure signal frequency) while the musicians play music, giving a warning if they are out of tune, but music happens to be a bit too fast for FFT (Fast Fourier Transform). ...
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Audio normalization

I have a recording in pcm format and I want to do some simple analysis. I have some questions about what is normalization. As far I understand it is to get all the amplitudes between a range i.e. [-1,...
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4answers
634 views

How to determine the “variability” in a frequency range?

This is a continuation of a previous question. I'm trying to analyze breathing and snoring sounds, and while I can fairly well detect snoring now, breathing is a bigger challenge. I've learned that if ...
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5k 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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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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“Subtracting” one audio stream from another audio stream

I am a long time software engineer but have practically zero experience with signal/audio processing. I am interested in learning about signal processing via a use-case we have for one of our audio ...
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PC-to-PC data communication over Sound FSK / PSK / DSSS

For fun and work I've to implement a java based application that can Encode text or data to sound and transmit using PC's speaker and on the receiving side is a microphone with decoder software. I ...
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Help with algorithm for modulating oscillator pitch using LFO

I am developing a software emulation of an analog synthesizer. I am trying to modulate the pitch of an oscillator using an LFO. For each sample fed to the computer's sound system, I am calculating ...
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Help implementing audio dynamic range compression

I am trying to implement audio dynamic range compression in JavaScript (not using web audio API). There is a lot of articles for sound technicians, and some high level documentation, but I couldn't ...
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1answer
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Coloring a waveform with spectral centroid or by other means

Update 8 Unhappy of having to upload tracks to the service and looking at the new release of RekordBox 3 I've decided to take another look for an offline approach and a finer resolution :D Sounds ...
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4answers
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What are Chunks when recording a voice signal

I am having trouble understanding the example program for audio recording , it comes with PyAudio module. I am using an external Audio Card to record the sound , obviously, the voice should be ...
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Algorithm to pan audio

I'm looking for a way to pan the PCM audio I have. At the moment the audio is in a stereo format and I would like to pan it to the desired side (or more/less to a side). I've recently figured out how ...
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3answers
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Common methods of transmitting digital data through an extremely noisy audio signal?

I am developing software for transmitting data from a computer to a mobile device using audio. Specifically, from the computer's speaker to the device's microphone. (I am developing both the sender ...
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1answer
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Detecting changes between voice and music

I have a bunch of audio files that have both speech and music parts (think radio). For instance, a given file might consist of the following sections: ...
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1answer
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Prerequisites for audio signal processing

I started a DSP course online and quickly discovered that as much as some people say that it is possible to avoid the more advanced mathematics of signal processing, this didn't seem the case. It ...
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Speech Compression - In LPC how does the linear predictive filter work on a general level?

Hi I'm taking a multimedia systems course and I'm preparing for my exam on tuesday. I'm trying to get my head around LPC compression on a general level, but I'm having trouble with what is going on ...
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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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Does it make sense to have complex numbers representing real-world audio signals?

In this course (Coursera: Audio Signal Processing for Music Applications) the professor uses an example of obtaining the DFT of a complex sinusoid: \begin{align} x_2[n]&=e^{\jmath\left(2\pi f_0 n+...
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1answer
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Library to differentiate people by their voice timbre

Say there are 3 people in the room. Each of them have a unique voice timbre. I want to recognize people by their timbre and know who's speaking.
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How to change volume of a PCM 16 bit signed audio?

I know I can multiply samples, then clip but perceived volume is non-linear for humans. Can you please help with a formula.
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928 views

Guitar effects explanation books

Can you please suggest some books to get how guitar effects: distortion, overdrive, phaser and so on are achieved ?
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Loudness of PCM stream

I'm receiving PCM samples through stream which is one channel, 8000 samples per second. I figured out way of finding out loudness by calculating 'force' of signal by adding abs value of every next ...
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420 views

MP3 Decoding Discrepancy

Is it possible for two different decoders to output two different sample results for the same MP3? I'm decoding an mp3 on Matlab and it produces two columns (presumably one for each channel). When I ...

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