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2answers
62 views

Recognizing bodily noises (ie, not spoken words)

I was pointed here by a chain of stackoverflow posts. I'm a software developer looking to make a module that recognizes noises like snorts and coughs. Here was my original question on stackoverflow ...
1
vote
0answers
32 views

How to synchronise two audio files using a marker

I'm working on a project on which we use WebRTC and the RecordRTC library to record audio files. I would like to be able to process the audio files server-side to ensure a good synchronisation. To ...
1
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1answer
30 views

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 ...
1
vote
1answer
126 views

Feature extraction for sound classification

I'm trying to extract features from a sound file and classify the sound as belonging to a particular category (eg : dog bark, vehicle engine e.t.c). I'd like some clarity on the following things : 1) ...
17
votes
4answers
1k 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 ...
-1
votes
1answer
58 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 ...
1
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1answer
49 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 ...
0
votes
1answer
117 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 ...
0
votes
2answers
115 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: ...
0
votes
1answer
42 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 ...
1
vote
1answer
40 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 ...
0
votes
1answer
31 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. ...
0
votes
1answer
37 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 ...
0
votes
1answer
51 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 ...
0
votes
1answer
84 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 ...
0
votes
2answers
55 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 ...
0
votes
3answers
84 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
0
votes
1answer
31 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 ...
0
votes
1answer
206 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 ...
0
votes
0answers
52 views

AMARI ERROR in Matlab ( FASTICA Comparison)

I am now doing a project comparing non-linearities in FASTICA and the best way to compare them is by using AMARI error. Could I ask how to perform Amari Error after fastICA in Matlab?I tried to search ...
0
votes
0answers
14 views

Doubt A weighting curve's phon value

Some sources say A weighting curve is inverted from 40 dBSPL, i.e 40 phon, while some sources say it's inverted from 30 dBSPL. Which is true? PS: For example, this says 40 and this says 30 Someone ...
1
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1answer
111 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 ...
0
votes
1answer
54 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.
0
votes
1answer
36 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 ...
2
votes
2answers
116 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. ...
1
vote
1answer
89 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 ...
1
vote
1answer
76 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 ...
0
votes
2answers
93 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 ...
-2
votes
1answer
74 views
1
vote
1answer
1k 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 ...
1
vote
1answer
131 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 ...
0
votes
1answer
80 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 ...
1
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1answer
72 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.
1
vote
1answer
102 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 ...
1
vote
1answer
93 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 ...
3
votes
2answers
73 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 ...
0
votes
0answers
42 views

Localization using sound

I am using cross correlation for sound localization. I have a four microphone array (arranged linearly). My problem is, sometimes the time diff found by cross correlation between a given pair of mics ...
0
votes
1answer
132 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 ...
0
votes
1answer
44 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 ...
0
votes
2answers
69 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 ...
1
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2answers
130 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 ...
0
votes
1answer
41 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 ...
2
votes
1answer
1k 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 ...
0
votes
1answer
126 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 ...
0
votes
3answers
56 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 ...
1
vote
1answer
114 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 ...
3
votes
1answer
177 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 ...
2
votes
2answers
100 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 ...
-1
votes
2answers
209 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 ...
0
votes
1answer
158 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 ...