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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18 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
21 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
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2answers
65 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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0answers
24 views

Mind helping me understand what this matlab code is doing? is this a PSD? or FFT?

I'm trying to learn how to get a useful frequency response of a single plucked string on a guitar. I have some old code that I'm working off of... It works in that i get a spectral response, but i ...
0
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1answer
43 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
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1answer
27 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
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1answer
32 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
3answers
64 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
35 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 ...
-1
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0answers
9 views

embed a low-bitrate signal in a g.722 audio stream

I would like to embed a small code (<16 bits) in an audio stream such that it can be recovered after g.722 encoding. It is not particularly important that the code be inaudible, as long as the ...
0
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1answer
28 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
49 views

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

What does the graph of audio signal tells? what are negative samples?
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0answers
36 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
9 views

Segment audio signals + calculate the variances

I don't know any thing about Matlab, but I need to use it to analyze data as a part of my current project. So, could you please help me to figure how to do the following ? First, I need to segment ...
0
votes
0answers
40 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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votes
1answer
24 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
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1answer
37 views

Help me understanding this source code of a Phaser

I tried to understand the following source code of a Phaser: ...
0
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1answer
40 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
38 views

Harmonizer Effect

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
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1answer
36 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
89 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
50 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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0answers
30 views

0 accuracy on test data with 24 MFCCs

I have designed a classifier for "bols"(strokes) of the tabla, using the MFCC coefficients 2-13 (ignoring the first coefficient.) at the onsets. When I do a cross validation, I get approximately 46% ...
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3answers
66 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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vote
1answer
66 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
53 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
44 views

How to organize MFCC features?

I have computed 12d-MFCC features on 3 consecutive audio frames, say: ...
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vote
2answers
46 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 ...
0
votes
1answer
37 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
votes
2answers
47 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 ...
0
votes
2answers
31 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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votes
1answer
44 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 ...
0
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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 ...
0
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1answer
44 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
26 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
77 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
vote
1answer
49 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
vote
1answer
51 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 ...
0
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0answers
45 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 ...
0
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1answer
55 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?
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2answers
238 views

Audio signal processing

I am trying to code audio signal processing software in java. Purpose: It will use audio signal processing library to filter out noise from the raw PCM based input (.wav) of heartbeat recording. ...
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5answers
176 views

Pitch Detection: avoiding frequency doubling / halving

I am working on a tuning app. I have so far tried 3 different libraries with a number of different algorithms. However, when I compare to other tuner apps, I seem to be getting frequency doubling on ...
0
votes
1answer
55 views

Generation of noise floor file for estimating signal-to-noise ratio of a wave file

I am trying to estimate the signal-to-noise ratio of a wave file (The wave file is a recording of a conversation in a room). I found examples where the Matlab scripts compare the wave file data to a ...
8
votes
3answers
1k views

Why do we choose 44.1 kHz as recording sampling rate

People's ear can hear sound whose frequencies range from 20 Hz to 20 kHz. Based on the Nyquist theorem, recording rate should be at least 40 kHz, is it the reason choosing 44.1 kHz?
2
votes
0answers
56 views

How do I construct input to neural network from audio signals?

Input: Microphone recordings of digits from 0 to 9 from different speakers. Output: The digit from 0 to 9. I am doing this for fun. So first I will train my neural network using some samples and ...
0
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0answers
34 views

Less Fidelity Than HRTF

Is there a known or accepted simple reference model of the receiver pattern for a typical human listener? I am aware of the existence of the Head Related Transfer Function, and that is emphatically ...
-1
votes
1answer
47 views

phase amplitude transform as a solution to reverberation

I am trying to use generalized cross correlation with phase transform (GCC_PHAT) on some audio processing indoor, with the existence of low noise. I ve read here that by canceling the magnitude ...
0
votes
1answer
185 views

python; audio; detecting silence in audio signal

I have real-time audio application written in python, and I have question about detecting silence. What's the most efficient way to distinguish silence from non-silence, in an input audio signal? By ...
0
votes
1answer
49 views

Long burst of high amplitude noise removal

I have a setup with an embedded DSPIC. This is used to link multiple audio inputs to the outputs and changed between them either when told to, or when one is active. This changing takes place on a ...
0
votes
2answers
106 views

Generating a high SNR sinusoide with lut on a DSP

I'm generating a sinus by using a lut method on a DSPic33f. My sample rate is 48 kHz, so I saved 12000 of the first value (unsigned int, 16 bits) and use trigonometric formulas to calculate the other ...