Audio, or in terms of signal processing, an audio signal is an analog or digital representation of sound, typically as an electrical voltage.

learn more… | top users | synonyms

0
votes
0answers
12 views

Detecting an echo using the autocorrelation of the cepstrum

I'm trying to implement an echo hiding steganography technique. I need a way to detect an echo in a audio segment. The documents I've been looking mention that I need to do an "autocorrelation of the ...
0
votes
0answers
26 views

detecting change of speaker in audio file

how can we detect whether the active speaker has changed ( either stopped or someone else started talking or many people started talking together ). I dont need to know who it was ( speaker ...
-1
votes
2answers
53 views

can MFCC coefficients alone be used to do voice detection?

Aim of my project is to differentiate between cases of silence, single active speaker and multiple ( simultaneous ) active speakers using as basic ( but effective ) logic as possible. Is it possible ...
0
votes
1answer
45 views

Steganography - What techniques exist to hide an image in an audio file? [closed]

Can anyone mention me techniques to hide an image in an audio file ? Thanks
2
votes
1answer
67 views

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 ...
0
votes
1answer
21 views

Objective, repeatable sound quality measurements and metrics

I am currently doing an electronics design for a basic 2.5W speaker amplifier for an embedded system based on COTS parts (e.g. codec and amp), similar to what one would find in a tablet. Aside from a ...
3
votes
2answers
74 views

How to identify useful elements of audio recording and ignore lulls?

I assume this is a relatively simple DSP task, but I'm having trouble finding any information on how to approach my problem. Hopefully someone with more experience and brain power can help me out: I ...
2
votes
2answers
80 views

How to know if a audio file is real lossless using its spectrogram?

I need to know if a certain audio file is real lossless and not a transcode (i.e. lossy converted in lossless, but lossy at the end). I was told there was a way to do this by creating a spectrogram of ...
0
votes
1answer
33 views

When do we normalize data in audio processing?

Lets assume we have a simple audio processing flow with preprocessing, feature-extraction and classification. Where and why do we normalize the data? I know the purpose of normalizing the data in ...
2
votes
1answer
60 views

Ear Test using MATLAB

I'm trying to test my ear frequency range by using MATLAB. The thing I could not understand is that I can hear the frequencies above 20 kHz. My sampling frequency is enough to satisfy the Nyquist ...
0
votes
1answer
52 views

MATLAB flanger…zipper noise, crackling, help with interpolation values?

i'm writing a batch script in MATLAB to flange. i'm getting some periodic noise, and i'm having trouble tracking down the source. it's definitely in the algorithm, because soundsc() will scale ...
0
votes
0answers
26 views

Common audio signal extraction from multiple similar audio files

Say you have two mono audio tracks recorded from a pair of matching omnidirectional microphones (call them A and B) that were, say, 20cm apart. The audio tracks were recorded at precisely the same ...
0
votes
3answers
45 views

Features for audio levels imbalance in films between speeh/music/effects

What would be the key features of an audio track or set of tracks (like 5.1) for identifying film audio tracks that have a clear unbalance between speech levels and sound effects levels (apart from ...
0
votes
0answers
6 views

How do we find the SNRseg if I have a block with 2D in time domain

I always get the answer -.0017 and it suppose to have some negative number around 30. I get the average for the plane voice then the average for the difference between plain and the encrypted voice ...
1
vote
1answer
42 views

Adjust Matlab Gui parameter while playing an audio [closed]

I would like to design a very simple MATLAB Gui program to adjust the magnitude of an audio while it is being played via a slider (slider1). However, the program ...
1
vote
2answers
34 views

How to select the center frequency and filter type for 5-band digital audio equalizer?

I have a small team project: use Matlab implement a Gui 5-band digital equalizer using FIR and IIR filters (2 cases) - The sample rate of all audio files is standard audio 44.1kHz. Coding is not my ...
2
votes
2answers
114 views

Upsampling PCM audio: from 6 kHz to 8kHz

What would a straight-forward way to convert an audio payload (PCM 16bit) from 6 kHz to 8kHz? I understand that this is an interpolation problem. I fear that a linear interpolation would introduce ...
0
votes
1answer
61 views

Allpass filter feedforward/feedback design and code

I want to design/implement a simple feedforward/feedback all-pass filter, and I am having some troubles and questions with it. From Shroeder design (based on J. Smith article here): I came up ...
0
votes
0answers
15 views

clarification on difference between features and descriptors for audio files

I'm working on a project aiming at recognizing musical instruments from a given sound. To do so, I need to extract many statistics from the sound. I base my work on ...
0
votes
0answers
66 views

bandpass FIR filter

I need to make simple bandpass audio filter. Now I use this simple C++ class: http://www.cardinalpeak.com/blog/a-c-class-to-implement-low-pass-high-pass-and-band-pass-filters It works good and cut ...
0
votes
3answers
69 views

detect impact from digital audio

I've started a hobby project which is to detect and count the number of impacts (for example someone clapping, bouncing a ball or hitting a boxing bag). I've been reading and playing around (in java) ...
0
votes
1answer
89 views

Psudocode implementation of true envelope

I am trying to implement the algorithm described in this paper, I'll quote the relevant portion: http://recherche.ircam.fr/equipes/analyse-synthese/roebel/paper/trueenv_dafx2005.pdf Let $V_i(k)$ ...
1
vote
2answers
100 views

FFT calculation suitable, Wav File, Java

If I have a wav file with this characteristics: 16bits (2 Bytes per Sample) 32000 Sampling Rate 60 Seconds 2 Channels then I have: ...
0
votes
2answers
37 views

Acoustic record and store devices

I am working on a project involving analysis of bird sounds and I want to record these sounds in slightly remote locations. It is enough if the device can only record and store them (maybe in an SD ...
1
vote
1answer
83 views

Beat Detection with Arduino

I'm trying to make a robot dance on several songs using Arduino. I need to understand how to get the rhythm from the songs, something like detecting the beats, calculating the bpm.. I'm not expert but ...
1
vote
2answers
81 views

Questions about Active Noise Cancellation

Being new to ANC, this is my understanding of how ANC works: Ambient noise (n) is entering a geographically small zone. To suppress the noise, an interfering (corrective) sound is injected ...
0
votes
1answer
29 views

Record and playback has echo/feedback

I didn't mess around with the code here, it was straight from their website, I just want to see if it would work. However, whenever I run it, it has feedback in it, sometimes I can't hear anything at ...
0
votes
1answer
86 views

Complex numbers in ifft of an MMSE amplitude estimator

I am trying to reconstruct a signal from a noisy speech using an MMSE algorithm proposed long time ago by Ephraim and Malah (1984). After going through the algorithm, I got a matrix A which represents ...
0
votes
0answers
50 views

How to, efficiently, identify telephone number from the key pressed data audio file?

A very common homework question. You are given an audio file(.wav) which has the recording of the keys pressed during a telephone call. You have to identify the telephone number from this audio file ...
2
votes
3answers
144 views

Is magnitude information enough to reconstruct an audio signal

I have used an MMSE STSA estimator to obtain the magnitude of an audio signal. The original signal is combined with white noise and I used an algorithm given in an old research paper by Ephraim and ...
0
votes
1answer
110 views

Uses of peaks in audio auto-correlation

Do you know of any algorithms for audio processing that make use of peaks in auto-correlation function? I am currently involved in research on mapping methodology for dynamic algorithms. I came ...
0
votes
1answer
36 views

resolving clipping audio issues

I have implemented a pre-emphasis filter by the following (pseudo) code: a = 0.5; s1[0] = s[0]; for (n = 1; n < N; n++) { s1[n] = (s[n-1] * a) + s[n] } The ...
0
votes
1answer
51 views

How can I enforce aliasing with Audacity?

I would like to hear how aliasing sounds like. How can I get this with Audacity? (I was told that I should adjust the Fourier coefficients in Audacity. I have no idea where I can do so / how I should ...
2
votes
5answers
122 views

Find Short Clip of Audio Within Longer Clip Of Audio

I need help figuring out a way to automatically synchronize two audio files that are both different recordings of the same source. More info below. This example is from a concert, where there is ...
-1
votes
1answer
54 views

Confusion when applying Hamming window to signal

I am writing some code which takes a signal of length $n$, and processes it in 20 ms chunks with 5 ms of overlap... So basically I have a buffer of 20 ms worth of sampled data, and I am trying to ...
0
votes
1answer
51 views

Audio processing — difference between block and sample-by-sample?

If a system operates on a signal with a time-domain IIR or FIR, why would an acquisition system chunk the audio into powers of two? I can understand filling a buffer by a power of two for an FFT ...
-1
votes
1answer
63 views

What does the graph of audio signal tells? what are negative samples? [closed]

What does the graph of audio signal tells? what are negative samples?
0
votes
0answers
42 views

What are the known ways of changing an audio signal that are not audible to humans?

For instance, I believe we cannot detect global phase shifts. And of course frequencies above ~20kHz or below ~20Hz. I've heard of "frequency masking", but don't know much about it and would ...
0
votes
0answers
48 views

Filter for audio sampler

An audio sampler (in "music" terms because I'm not experienced in dsp) is a software which loads an audio file and allows you to play it at different pitches by slowing it down and speeding it up. ...
2
votes
1answer
46 views

Codec Output has “Wavy” Amplitude on Chirp Input

I am sending a chirp that sweeps form 20Hz to 500Hz to an audio codec, CS42L52, connected to an MCU, STM32F405, and I am receiving a "wavy" amplitude output. There is no processing on the MCU, I pass ...
0
votes
1answer
48 views

Help me understanding this source code of a Phaser

I tried to understand the following source code of a Phaser: ...
0
votes
1answer
87 views

Biquad Filter Transfer Function Frequency Response

$H(z) = G\cdot\displaystyle\frac{2b_0\left(1-z^{-2}\right)}{1-2a_1z^{-1}-2a_2z^{-2}} + 1$ I have a data sheet with the above transfer function, I need to calculate its frequency response in code. I ...
0
votes
1answer
56 views

Harmonizer Effect [closed]

I know that to build Harmonizer I need pitch shift the signal in differents ways, my question is how know for what scale/factor/notes/frequency I need pitch shift to create harmony? For example if i ...
0
votes
2answers
60 views

At what time stretching ration would an audiophile notice a wrong pitch

I have some low quality audio from my native tongue on a VHS. I am using this audio to dub HQ released of the same program (from another language source). The video frames/second ratio is 25:23.98 . I ...
1
vote
1answer
156 views

Why these harmonics always present?

I am using Audio Precision 2700 to test some test vectors. I have connected resistor banks as class - d power amp's load, and tap from this resistor using a XLR cable to the AP. When my power amp ...
0
votes
2answers
93 views

What is the difference between pitch detection and Onset detection?

According to wikipedia Onset detection: Onset refers to the beginning of a musical note or other sound, in which the amplitude rises from zero to an initial peak. It is related to (but ...
0
votes
3answers
77 views

Choosing the best measure for machine learning on a Spectrogram

I am working on an application to classify Talas(a piece of music with onsets). For that, I have the spectrogram of a Tala, which I want to compare with the test Tala, or compare 2 Talas. For this I ...
1
vote
1answer
96 views

Audio Recognition and Learning

I'm a beginner with absolutely no experience in signal processing, but it has been a long-term goal of mine to do something that mixes music (a hobby of mine) and computer science (my "expertise"). ...
0
votes
2answers
58 views

Determine the time length of audio training samples?

Currently I am trying to use MFCC feature for audio training in order to do human voice classification (can later distinguish the person who speaks). I have problems to determine the proper time ...
0
votes
2answers
64 views

How to organize MFCC features?

I have computed 12d-MFCC features on 3 consecutive audio frames, say: ...