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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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 ...
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1answer
28 views

Adjust Matlab Gui parameter while playing an audio [on hold]

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 ...
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2answers
25 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 ...
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2answers
82 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 ...
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1answer
53 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 ...
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13 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 ...
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0answers
59 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 ...
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3answers
63 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) ...
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1answer
86 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 $V_i(k)$ ...
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2answers
62 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: ...
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2answers
36 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 ...
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1answer
63 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 ...
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2answers
76 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 ...
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1answer
25 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 ...
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1answer
74 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 ...
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0answers
44 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 ...
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3answers
127 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 ...
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1answer
79 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 ...
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1answer
35 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 ...
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1answer
45 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 ...
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4answers
98 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 ...
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1answer
49 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 ...
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1answer
40 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 ...
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1answer
56 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?
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40 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 ...
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0answers
47 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. ...
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1answer
40 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 ...
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1answer
46 views

Help me understanding this source code of a Phaser

I tried to understand the following source code of a Phaser: ...
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1answer
54 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 ...
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1answer
40 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 ...
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2answers
58 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 ...
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1answer
130 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 ...
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2answers
71 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 ...
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3answers
73 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 ...
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1answer
84 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"). ...
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2answers
56 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 ...
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2answers
55 views

How to organize MFCC features?

I have computed 12d-MFCC features on 3 consecutive audio frames, say: ...
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2answers
48 views

Signal corrupted using peak (biquad) filter at low frequencies

I'm trying to process an audio stream through a series of peak filters at static frequencies that range from 0 to nyquist. If I start the range from anything below about 25Hz, the audio gets corrupted ...
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1answer
52 views

Wav file scale factor

I seek a scaled audio file format for my dsp output. I'd like to produce "scaled wav files", when I've done my pre processing. Eg. if the amplitude is 15.5 (of whatever unit), you should be able to ...
0
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2answers
55 views

How to transform large set of audio samples using FFT?

Well, I am new to DSP so I apologize for the basic question. I am trying to transform audio 16bit PCM (which I store in normalized 0.0-1.0 range) samples into frequency domain using fft. I ...
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2answers
32 views

Re-recording audio vs. convoluting old audio data - differences?

I want to transform a studio audio dataset into a smartphone audio dataset (used for a speech recognition application). Would there be any difference in the resulting audio files using one of the ...
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1answer
47 views

Can someone please help me out with audio, sampling frequency question

I really having a hard time to tackle the following question. It is asked to load a ".mat" file using load file.mat. And the result is two variables: Fs varaible ...
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1answer
24 views

Available Online Recordings for Acoustic Echo Identifiablity

I am trying to add a few samples for a publication on identification of clean signal from the one which has echo on it. For this purpose I need data set that includes playback of an already available ...
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1answer
46 views

Segmenting a sound file before each of its attacks

Like the title says... I want to split audio file where a single instrument is played (not so fast)... I just need to detect the attack phase of the waveform to know where a played note begins. Can ...
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0answers
27 views

Where is the energy goes in ANC (Active noise control)?

I just learned there is ANC technology in signal/audio processing domain. By my understanding, if two audio signal, whose phase is diff by 180, added the result will be silent. But from the physical ...
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0answers
95 views

how to comparing two audio signals?

I know this question has been asked in so many places but I can't still figure out how to solve this. I have two audio signals of breathing. One is a healthy breathing and it is saved inside the ...
1
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1answer
53 views

matching the mouth shape of an audio file(lip sync)

I am working on something like cartoon, but the character's mouth shape is automatically detected and matched by the word they are saying. my guess is to use speech recognize algorithm to analyze ...
1
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1answer
54 views

How to obtain mean Power Spectrum from two audio channels?

I have a source of noise in a duct, on either side of the noise source I have a microphone, which register the pressure, so that I get two pressure signals: $p_1$ and $p_2$. I want to get the acoustic ...
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0answers
46 views

multipath propagation

i recorded a tone between two devices and applied cross correlation on the recorded and the sent signals. I want to know if there is a multipath effect on my result (on MATLAB). what can i use for ...
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1answer
58 views

relationship between energy and sampling rate

Excuse my silly question, but i really want to know does changing the sampling rate affects the energy (bandwidth) of a signal? therefore improves cross correlation output?