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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23 views

Noise reduction on wave file

I have created a simple plot of waveforms using matplotlib of 2 wave files on python. Here is the code: ...
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1answer
27 views

How does spatially white noise differ from diffuse noise

In noise reduction, having read that directivity index is used to reduce diffuse noise and white noise gain reduces spatially white noise, I would like to know the difference between both the noise ...
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1answer
28 views

playing audio continously in matlab

I am in my way to design a function generator using matlab,the point which made me stuck is the challenge of continuous waveform generation code snippets of my attempt ...
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0answers
25 views

Dynamic range compression in the frequency domain?

I'm trying to do frequency-band specific dynamic range compression (like multi-band compression) and was thinking of whether dynamic range compression is doable in the frequency domain or whether it's ...
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0answers
15 views

Matching two video conversations

I have two video where two person having conversation. They begin recording different times. I want to match these two videos' frame numbers, i.e. By matching audio. How can I do that?
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0answers
30 views

DSP Library IIR and FIR filters

Me and a coworker are currently in the process of making an audio dsp library. We are wondering what some good FIR and IIR filters to use were. I know ALOT of them but what do you guys think in ...
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1answer
75 views

How to simplify the Fourier Series Using an Approximation?

I have a signal, $f(t)$. I know a function that can be used to generate this signal, such that I can determine its Fourier series. I want to express this Fourier series in simpler terms so that the ...
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2answers
52 views

How to generate realistic echo to sounds?

I guess the most straightforward way of adding an artificial echo to a sound signal is to add its time-delayed version to the original signal. However, I am not sure how realistic that would sound ...
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1answer
37 views

How to fix audio clipping in Adobe Audition?

I am editing audio from video that was captured from the sound board. The audio was recorded too loud since someone turned up the board and and the bass and vocals sound scratchy. I have brought the ...
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1answer
20 views

Cancelling the acoustic echo that occurs with a variable delay and volume

I am trying to build a software that records audio to a server and then broadcasts that audio into the same room that the microphone is present. This means that the audio signal is uploaded to the ...
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1answer
70 views

How to interpret these different Fourier analysis of this audio signal?

This is my first dive in DSP. I would like to familiarize myself with frequency analysis. I have two audio tracks which should be digitized at 16bit-44.1kHz and 24bit-192kHz (music, presented as a ...
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0answers
30 views

Detailed Analysis of Phase Delay of an Audio Signal

I write this question as a new reference point because it is difficult to find something exhaustive regarding this topic. I'm talking about the calculus and interpretation of the phase delay of a ...
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0answers
17 views

Should I normalize my input vector when using a matched filter?

I am using a matched filter to recognize an audio sample within a noisy input stream by cross correlating my sample with the input array and checking if the maximum crosses over my threshold. ...
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1answer
43 views

Using Chebyshev polynomials in wavetables

How are Chebyshev polynomials used in wavetables for audio synthesis? I dont understand how to store these inside wavetables? How does the synthesized audio result in matching to the correct index ...
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0answers
27 views

Advice for creating differential microphone array

For a school project I'm required to create a differential microphone array that can filter out background ambience, so a clear voice signal can be recorded in noisy environments. The plan was to use ...
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0answers
37 views

How to calculate the frequency shift of FMCW transmitted signal and received signal using Matlab?

I generate a FMCW chirp signal between 18k-20kHz using Matlab, then sound the signal through the speaker in laptop, then record the reflected signal through the Mic in the laptop. What I want to do is ...
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0answers
26 views

Invertible multi-resolution representation of audio/speech signals

What are the invertible (preferably linear) representations of an audio/speech signal which captures the essence of signal in different resolutions and details. Basically, I am looking for ...
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0answers
19 views

Representing and compartmentalizing FFT of audio sample into user friendly spectrum (need direction)

I am using the Apple accelerate framework to do some audio FFT analysis. Here is what I have so far and I want to make sure that my methodology is correct and I am indeed displaying appropriate data ...
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0answers
9 views

What's horming frame in AMR-WB code?

I find a lot of word to talking about "homing frame" in the AMRWB spec and code. So what it is?
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3answers
68 views

EQ - Calculate the power per Octave of a signal through FFT

I'm working on a audio Equalizer, and I'm not sure if the way I obtain the power per octave of my signal is good. Here is the code I have : (I use the third octave bands, which is initialized earlier ...
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0answers
29 views

Use SNR for Wav signals

I have implemented an audio steganography program in MATLAB. One way to evaluate your steganography method is to use SNR. The formula I found for SNR calculation is mentioned below. The problem is, ...
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1answer
39 views

Adaptive volume normalization on a stream of PCM data

I am writing a program that receives a continual stream of PCM data from various sources. Some sources have different volumes than others. The PCM data is always a live continuous stream of 16 not ...
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1answer
34 views

Division by zero in Levinson–Durbin recursion?

I am doing a LPC analysis of a speech signal using the autocorrelation method. To calculate the LPCs I am using the Levinson–Durbin recursion. In my literature the error is initialized with the first ...
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2answers
32 views

phase, magnitude, audio file and information

I've been doing some thinking lately. I know that for image files a larger portion of information is contained in the phase of the signal. How does that go for the audio? Does the same hold or is the ...
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0answers
22 views

How to detect impulse noises in audio recording?

How do you detect impulse noises in audio recordings of speech? By "impulse noise" I am thinking about noises like: Typing on a keyboard A door slamming Somebody bumping into a table Tapping on a ...
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0answers
24 views

How can I convert a FM discriminated signal to binary data using software

I currently have FM discriminated data in a .wav file. How can I convert it to binary data (+0.5 == high bit, -0.5 == low bit) given a baud rate.
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1answer
50 views

Possible to implement simple IIR filter with greater than 6 db/octave slope?

I am using the following IIR code as a low-pass filter in a software project, and it works great. However, I would like the slope of the filter to be much steeper as the one here seems to be very ...
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1answer
67 views

Determining The Sparsity or Density of a Chroma Vector

I have a range of songs where I have took an 'unfolded' chromagram computed from a series of short-term windows. Specifically, for one song, I have one vector which contains the amplitudes of the ...
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0answers
43 views
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28 views

Goertzel filter Poles on unity circle influence of a behavior of a signal at high frequency

I'm trying to apply a goertzel filter on my sound data of 20500 Hz with sampling rate of 44100[samples/sec]. My bit duration which is passed to the goertzel function is returning a very low magnitude ...
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0answers
21 views

Can software distinguish between different musical instruments

If possible, I would like to write a little script that can identify different musical instruments from just a recording. Apart from variations of amplitude and frequency over time (which can be ...
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1answer
52 views

How to emulate human sound recognition

A person can recognise a large variety of sounds and fairly subtle differences in sound, even if they have only heard the sound once. Is it possible to recreate this with software, and how? ...
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2answers
103 views

Which filter for an audio equalizer

I’m new to DSP and I have to do an audio equalizer in c++. I did a lot of research about it and tried some stuff in the last month, but I’m a little overwhelmed with all those informations and it’s ...
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0answers
28 views

How are sounds turned into phonemes

In the field of speech recognition, part of the process is to take a voice signal and break it up into constituent phonemes. For example, the English word hear is ...
2
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1answer
35 views

stretch factor in SOLA time stretch methods

I notice that in the general Synchronous Overlap and Add (SOLA) algorithm, we need to obtain the best overlapping start index using, for example, cross-correlation. But won't that start index change ...
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0answers
32 views

goertzel or bandpass?

I try to decode freqs of 17k & 17.8k which carry on a sinus of 15ms duration. I try to decode them with a goertzel filter. It seems fine when one of the filters show a magnitude that is around ...
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2answers
121 views

What is spectral entropy?

Can anyone explain what spectral entropy is? Does noise with a restricted bandwidth have the same spectral entropy as white noise?
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1answer
34 views

Applying a mathematical function to a 16-bit Python array

I am using scipy.io.wavfile to read a WAV file and playing it with pyaudio. Before the numpy array is written to a string and output through pyaudio, I would like to be able to apply some sort of ...
3
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1answer
83 views

5.1 Rear To 5.1 Side mixing matrix

I am searching for a matrix definition for converting a audio signal (5.1 with rear) to a audio signal (5.1 with side). Currently I am using the definitions from: ...
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0answers
16 views

Frame-level autocorrelation coefficients with lags bigger than frame size

I am reading the paper "Automatic Generation of Social Tags for Music Recommendation" and struggling with understanding this: The features we use include [...] 176 autocorrelation coefficients ...
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1answer
38 views

Why does speeding up audio files with VLC Player not match with doing the same thing with RPG Maker or Audacity?

When I speed up 150% an audio file with VLC Player, why doesn't it match the higher-pitched version I get when I speed the same file 150% with RPG Maker VX Ace's Sound Test? When I speed up with VLC ...
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1answer
62 views

Controlling one signal with another one (Rocket engine effect - Subtractive synthesis of sound)

I programming some sound effects in Java and exporting them into .wav files. Currently, I am trying to program a rocket engine sound effect. I want to do it in the following way: The sound of a ...
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2answers
153 views

Noise removal from audio using FFT/Spectral editing

I have a noisy recording with some letters spoken out loud. I've noticed that the noise is easily recognizable when looking at a spectrogram of the audio. Is there anyway to set a power/frequency ...
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2answers
27 views

MIDI out from freeware sequencer to PD or Max/MSP

I'm looking for a way to send MIDI from a cheap or free MIDI sequencer to Pure Data or Max MSP so I can do realtime MIDI processing in PD/MAX. So I guess I have a couple questions: What is a good ...
3
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4answers
138 views

Why every musical instrument sounds different than any other while listening?

This is soft question.I am not expert in music. We know that English language has 26 letters (which are again grouped into vowels and consonants) , while language of music is same all over the world ...
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1answer
24 views

What are the units of amplitude of my ultrasonic recording equipment

I'm not familiar with the topic of sound. I have ultrasonic audio recordings sampled at 250kHz (16bit) in WAV format. The equipment used to record them is linked here: ...
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1answer
64 views

FIR Filter Design - obtaining FIR coefficients from frequency response

I would like to obtain FIR filter (low-pass) coefficients from a given frequency response by using C++ to develop algorithms. I have read at http://iowahills.com/B2PolynomialFIRFilters.html, that, if ...
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2answers
70 views

Authoritative DSP book/reference material on Psychoacoustics

I am an auto didactic Electrical Engineer currently pursuing a goal to design a custom application targeted at deeply embedded linux(think BBB or even Raspberrypi-2) type device. My DSP background is ...
0
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1answer
42 views

Invert soundsc function in Matlab

Does anyone know what is the inverse function of soundsc in matlab. I'm playing a sound with a my PC microphone using soundsc, ...
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0answers
29 views

Filter signal1 from evolving frequency response of signal2

I'm a composer (of music) starting out with signal processing. I'm also reasonably proficient in C at this point. I'm not even sure if what I'm trying to do is possible, because I can't yet find an ...