Questions tagged [sound]

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How can I connect multiple microphones to my computer (more than 4) and process the outputs independently? [closed]

I am trying to create something like a heat map for sounds originating in a classroom to study the discussions inside the classroom. For this I am instaling directional microphones in various parts of ...
Nishant's user avatar
  • 11
22 votes
5 answers
3k views

What are the audio equivalents of images like "Lena", "Mandrill", and "Cameraman"?

I don't know what to call these images, but they're very often used in the image processing world as standard examples to show the output of a particular operation. Here's a link to the page ...
aardvarkk's user avatar
  • 321
-1 votes
1 answer
608 views

how to read full input signal with STFT in matlab

Please help! I am trying to figure out why in the code i have below my input signal is being cropped but in the code I have a full signal created. The white noise created is 5000 samples for the right ...
Asau7610's user avatar
1 vote
1 answer
104 views

What does it mean for audio to be 'sped up'? How can I undo it?

I very simply have a .wav audio file, but there is a note with it that says "This audio has been sped up by 10x". I am wondering, what does it mean exactly from a signal processing perspective for an ...
TheGrapeBeyond's user avatar
0 votes
1 answer
7k views

how to apply Short-time fourier transform to code in matlab

Please Help, I have a white noise signal that I created and need help to apply the Short-time fourier transform to my code so that it can do the bandpass filtering. I am trying to put it in the code ...
Asau7610's user avatar
0 votes
2 answers
1k views

Android: fundamental frequency

I want to find the fundamental frequency for human voice in an Android Application. I want to use autocorrelation method. My code is this: ...
user3582433's user avatar
0 votes
1 answer
342 views

How to Concatenate white noise signal and applying bandpass filter matlab

I created a white noise signal. I am trying to use the array I have created with the white noise to play in 2 channels. When the left channel I was trying to have 5000 samples and the right will have ...
Asau7610's user avatar
2 votes
1 answer
412 views

Time-varying waveform

My situation is as follows: i am trying to generate a waveform the hard way, by constructing the samples one by one and then saving the result to a .wav file using Python. When the frequency is ...
alansammarone's user avatar
0 votes
1 answer
117 views

How to store the signal in a variable and plot the signal in MATLAB

I am trying to store the signal I have created in a single variable and graph the signal in two subplots in one single graph, one for left channel, one for right channel. ...
Asau7610's user avatar
0 votes
1 answer
1k views

How to add repeated beeps to sound created with a pause in between each beep in MATLAB

I have 2 sounds that I created and trying to make them beeps. Maybe 4 or 5 repeated beeps but have a pause in between each beep but make it still part of the signal. I think I have it so that the ...
Asau7610's user avatar
2 votes
2 answers
2k views

Creating 2D array to play audio in left and right channel in matlab

I'm new to matlab sound processing and I am trying to create a 2D array for the sound code below. I have to make the sound have 400 samples(i.e 10ms) at 20hz, but having trouble getting the code to ...
Asau7610's user avatar
1 vote
1 answer
656 views

Resonant peak frequencies for phaser effect

I have a conventional phaser setup with allpass filters and feedback. I can calculate the peaks frequencies of the phaser when the allpass delays are set to 1 (then they work as simple one sample ...
teadrinker's user avatar
1 vote
2 answers
614 views

Separating excitation from filtering for sound sythesis

From my days at college I rememeber that LPC splits a signal into a series of slowly changing filter coefficents and an audio-rate residual error signal, such that the original signal gets reproduced ...
Martin Drautzburg's user avatar
0 votes
3 answers
473 views

How to delay a signal by D degrees

I read in a paper that a sound signal can be delayed by D degrees. How can such a thing be possible? Isn't sound signals expressed only in terms of the time axis? Thanks in advance
JMFS's user avatar
  • 336
0 votes
1 answer
101 views

what is standard acoustic volume

In Audio Experts book, it's given as Sound pressure meters must be calibrated at factory to a standard acoustic volume. So what doth this ...
nmxprime's user avatar
  • 207
2 votes
1 answer
5k views

Time-frequency representation of sound signal using Matlab

I am currently doing a project on sound signal processing by using MATLAB. I have a problem regarding creating the command line so that I can get a graph containing frequency by its time. But this ...
ashley mraz's user avatar
1 vote
1 answer
207 views

Fourier Transform for frequency analysis

I applied Fourier transform on a sound (PCM generated digitally with sin function) and frequency detection is accurate. However, if I play the sound and record it, and then apply a Fourier transform ...
user3388324's user avatar
0 votes
1 answer
620 views

Q value for a high pass filter

How to calculate the Q value for a high pass filter ? Ps: I'm a beginner in signal processing, so please put it in simple terms.
Srini Vas's user avatar
0 votes
1 answer
53 views

Frequency filtering - Audio

Sound are digitalized in computers as an archive with its intensity amplitude at the given time. I want to ask a very simple question about frequency filtering. How it's done, if there's no frequency ...
Mr Zantos's user avatar
2 votes
2 answers
334 views

Need help with FFT results

I am playing around with FFT and cant seem to get the expected results. I have a sound clip that i recorded from a mic placed on an analogue clock. So it has a sound every second. ...
user1348051's user avatar
3 votes
1 answer
2k views

Colored noises: Stationary or non-stationary?

I know for sure that white noise is considered as a stationary sound, but is that true for the rest of them? http://en.wikipedia.org/wiki/Colors_of_noise Also, can we meet some of these sounds in ...
bibliobibuli's user avatar
0 votes
2 answers
990 views

Pitch class profiling

Currently, I am developing a piece of software for chord recognition. There are FFT and pitch class profiling on it. I tried using the pitch class profile proposed by Takuya Fujishima, but the result ...
wendy0402's user avatar
  • 131
0 votes
2 answers
435 views

Why do we follow Shannon's Theorem in audio sampling?

My professor told me that in sampling a signal, is not usual to respect the Shannon theorem because the interpolation formula contains sines and fractions which are difficult to calculate with a ...
fortea's user avatar
  • 103
-1 votes
1 answer
110 views

difficulty in understanding this code [closed]

...
zeee's user avatar
  • 9
1 vote
1 answer
4k views

Separating two sound file frequencies using bandpass filter matlab

can someone please help me? I have a project to implement a superheterodyne transmitter and receiver without using matlab functions that modulate. The first part which I can't seem to get right is the ...
Noha's user avatar
  • 11
2 votes
1 answer
943 views

How to compare 2 matrices of MFCCs, keeping the time information

I have a corpus of several thousand sounds. Now I pick a sound sample S, and I want to find in my corpus the sound S' that is the most similar to S. The application is for music, for a composition ...
sebpiq's user avatar
  • 283
0 votes
2 answers
486 views

How to decipher this Microphone sensitivity Chart for Choosing a SPL meter

I have a mic having sensitivity chart as below which has relative dB on y-axis showing 1V/0.1Pa what does this convey how to read my mic's sensitivity a document on microphone sensitivity from ...
kakeh's user avatar
  • 365
1 vote
2 answers
462 views

Speech Enhancement Libraries

Is there any speech enhancement algorithm library? I need to process a sound to voice recognition, and I want to remove noise from audio source.
Victor's user avatar
  • 111
1 vote
3 answers
6k views

Easiest Beat Tracking Algorithim

I want a relatively easy to implement but relatively reliable algorithm for detecting beat locations in an audio file.What I am aiming to do is get the location of the closest beat to a particular ...
KillaKem's user avatar
  • 326
2 votes
1 answer
260 views

Detecting presence of music in ambient sound

I am hoping to create an application that would listen to ambient sound and detect if music is being played. It is not important to identify the music being played; just detecting that some music is ...
Abhilash's user avatar
  • 121
3 votes
2 answers
142 views

Acoustic measure of a specific room

I would like to make an acoustic measure of one room, so I can compare it with a acoustic measure of some other room. Then compare these data. So when I take an acoustic measure of the first room ...
Ivo Patrick Tudor Weiss's user avatar
0 votes
1 answer
1k views

Why only take 205 sample when using a sample rate of 10khz or 8khz?

I've often seen i my DSP book, that when they record a sound, they use a sample rate at 10khz or 8 khz, which makes sense since the fs is 2 times bigger than the sampling signals max freq. What i don'...
Guest's user avatar
  • 11
0 votes
1 answer
171 views

Sound card for recording audio

I need to record sound using 2 (or 4, eventually) microphones and then calculate the time delay of arrival of sound at the different microphone pairs. But first, I need a microphone array. And since I'...
Kevin Martin Jose's user avatar
0 votes
2 answers
399 views

Measure and simulate non-linear distortion

We can capture frequency response, phase response and reverb in recording of impulse response. Then we can use it to simulate audio system and/or space by convolving it with dry audio signal. Is ...
adiblol's user avatar
  • 93
1 vote
2 answers
849 views

Audio interface or External sound card?

I want to create a microphone array that can be connected to my laptop. Previous discussions in stack exchange and other forums suggested me to get a USB audio interface. I would want to do a lot of ...
Kevin Martin Jose's user avatar
0 votes
1 answer
61 views

Algorithm to change an already sampled guitarstring to sound like it was a bit damped by the hand [closed]

I'm trying to simulate the damping of the hand on a guitarstring on an already recorded/sampled open guitarstring sound. I've been trying to use low pass filter and had a moving frequency range but ...
Morgan's user avatar
  • 101
8 votes
1 answer
25k views

Why should one use windowing functions for FFT?

So I just revised my pitch calculation algorithm using a harmonic product spectrum algorithm. I was just curious about why this explanation of Harmonic Product Spectrum states that you need to ...
Skylion's user avatar
  • 291
0 votes
1 answer
488 views

Modelling propagation of sound wave by particle simulation

How might one model the propagation of a sound wave through a 3D environment with obstacles in it? My initial thought would be to use freely moving particles that repel one another; with sufficiently ...
P i's user avatar
  • 1,319
0 votes
3 answers
111 views

Auditory receptive field mapping

The Wikipedia page on sound localisation states: Localization accuracy is 1 degree for sources in front of the listener and 15 degrees for sources to the sides. Humans can discern interaural ...
P i's user avatar
  • 1,319
1 vote
1 answer
767 views

Comparing two sound files in embedded-systems (C language)

what would be the most efficient ("cheap" in terms of system resources etc.) way to compare 2 sound files? Here is the scenario. Sound A comes as an input. There is sound B stored in memory. We need ...
Stpn's user avatar
  • 113
3 votes
1 answer
295 views

How to add more cowbell?

Remember that Christopher Walken skit on Saturday Night live where he told the band “it needs more cowbell” https://en.wikipedia.org/wiki/More_cowbell ? This site automated it http://www.morecowbell....
SpeedCoder5's user avatar
3 votes
2 answers
191 views

Two generated tones creates a side tone - how to avoid the side tone

I'm trying to create two tones at different frequencies. So I have one tone at 15000 Hz and one at 15400 Hz using Audacity. The problem i'm running into is there is a third tone created at a much ...
user2610470's user avatar
1 vote
2 answers
2k views

Direction of sound using microphone array

I want to estimate the direction of arrival of sound (angle in a 2d plane) using a microphone array. I'm hoping to follow a time-delay based method using a 4 mic array. The method requires me to find ...
Kevin Martin Jose's user avatar
0 votes
1 answer
2k views

How to convert text into wave [closed]

I am trying to transfer text data via sound wave between two android phones. I am a newbie and would like to know how to proceed with it. I am stuck at the very first step on how to modulate text data ...
Perk's user avatar
  • 1
0 votes
2 answers
2k views

voice pattern matching algorithm

My idea is to find a word in wav/mp3 files. It is just one word. I have a database with a few hundred songs. I am looking for pattern matching (this is not an exact pattern) it can be an word i ...
Gilad's user avatar
  • 215
0 votes
1 answer
66 views

fliter doesn't filter

In this code I try to decompose music from singer sound with wfilters but when I hear it it is still mixed where Iam I wrong ? ...
Gambour's user avatar
  • 33
0 votes
2 answers
598 views

What's the best method to extract vocals from music?

I want to achieve something like this https://soundcloud.com/user7421656/sonicworx-isolate-preview . I've researched it and a few papers mention the use of "descriptors" on the audio and use ...
user1530335's user avatar
1 vote
2 answers
387 views

Measuring how "non-noisy" a sound signal is

Given a sound clip (whose source is an FM radio signal) I'd like to measure how different from background noise (the "in-between channels" noise) the sound clip is, so that I can compare the "non-...
haroba's user avatar
  • 203
15 votes
4 answers
30k views

How to compute dBFS?

I asked myself how to compute dBFS (dB full scale) from a value of sample between 1 and -1? and in general?
JustGoscha's user avatar
1 vote
2 answers
164 views

Generating Music Tones, How to handle when tones overlap? (16bit samples) [duplicate]

If i add the two tones then the result if two peaks are added together will be greater then the the sample max (2^15). For the life of me I couldn't think of the right words to google. Think i have ...
user1530335's user avatar