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0
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2answers
29 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
72 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
28 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
130 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
31 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
12 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
vote
1answer
98 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
35 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
106 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
72 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
70 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
89 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
71 views
1
vote
1answer
539 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
92 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
60 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
vote
1answer
63 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
92 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
87 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
72 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
34 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
114 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
43 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
62 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 ...
0
votes
2answers
106 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
36 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
363 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
109 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
55 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
108 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
173 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
91 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
177 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
134 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 ...
0
votes
1answer
535 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 ...
0
votes
0answers
25 views

Midi Pitch Bend merge

I have a patented midi controller in development which graduates the pitch bend function of a keyboard over a two octave guitar neck using wired frets. If the pitch bend range is set to +or- 12 ...
0
votes
1answer
58 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 ? ...
5
votes
3answers
584 views

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 ...
0
votes
0answers
87 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 ...
1
vote
2answers
70 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 ...
3
votes
1answer
554 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?
1
vote
2answers
69 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 ...
3
votes
1answer
220 views

How to make pop-free, crisp two-beep ringtone?

I hope this ultra-noob question is not entirely out of place here. I'm trying to make a super simple ringtone: 0.5s silence + 1s "pure tone" + 0.5s silence + 1s "pure tone" + 0.5s silence. By "pure ...
1
vote
1answer
308 views

What is the DTW result (applied to the MFCCs of an audio file) proportional to?

I have computed the MFCCs for two audio files and applied the DTW to both sets of coeficients for both files. Now each such pair the two files is described by one single value (computed by the DTW). ...
1
vote
0answers
42 views

Length of frames for calculating cepstral coefficients

For audio classification we are using the cepstral coefficients and their derivatives. Which I calculate using the Matlab libary voicebox. I am looking for some literature for how to choose the frame ...
1
vote
0answers
93 views

How to convert a sound file to any other data format which can be manipulated

I know this is very basic question and i am new to this site. But can anybody help me with my question. How to convert a sound file to other data format say using some type of transform (FFT or DFT) ...
1
vote
1answer
165 views

How to decide the number of cepstral coefficients?

For audio classification we are using cepstral coefficients, but how should I decide the number of coefficients?
3
votes
1answer
273 views

Ultrasonic data transfer

I am trying to send a short amount of data (10-15 characters) from one mobile device to another via (inaudible) sound within 1-2 seconds. I'd like to be able to deal with normal external noises ...
0
votes
0answers
210 views

Getting sound pressure level (SPL) for a frequency band from Welch's derived PSD

I'm using Welch's method (pwelch, in octave) to get the PSD of a recorded sound signal (i.e. pressure level vs. time). I then took the PSD data and took the sum of the amplitude in each octave band ...