Questions tagged [audio]

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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2answers
601 views

Measuring microphone Signal to Noise ratio (SNR) with A-weighting

Foreword: I have already read Applying A-weighting and user @endolith's A-weighting code I would like to measure the SNR of a microphone with the same method applied by the main manufacturers, that ...
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2answers
101 views

Modeling the human perception of noise

Add together some sinusoids with irrational or "nearly" irrational frequency ratios, and you get something pseudoperiodic. If there aren't too many, you can hear that it has only pitched components, ...
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1answer
127 views

Find filter coefficients to model a device using its measurement

I am trying to realize a digital filter that has the same freq. response of an existing speaker. I have fed an audio sine sweep to the speaker and measured the speaker output, both at 48kHz. Then I ...
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3answers
385 views

What is the proper way to implement a real time spectrogram?

I am trying to implement a real time spectrogram and I am not sure that I am implementing it the right way. First I decided to use Matlab, I was taking a short recording of lets say 0.1 seconds and ...
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1answer
109 views

Extracting feature from audio track using DWT

Our teacher gave us an assignment: Write a program to transform an audio track to time - frequency domain, and create a vector include at least 4 features extracted from transformed output signal. I ...
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1answer
67 views

4-phase FIR LPF in MATLAB

I'm trying to create a low pass FIR filter using the coefficients I found in ITU 1770-3 on page 18. This is the last piece of the puzzle for an audio loudness metering algorithm I'm working on. I'm ...
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1answer
100 views

Frequency detection in audio and what parameter or set of parameters help/limit frequency detection

I'm trying to learn audio and voice/speech processing techniques out of interest. I intend to learn, experiment with algorithms and design, implementation of such systems, frameworks using programming ...
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1answer
127 views

Separate two sinusoids with very close frequencies (5 Hz difference) in a 44.1 Khz audio signal, with good time-domain resolution [closed]

Here is an audio signal which is a mix of (WAV file available here): a 2000 Hz sinusoid, beginning at 1.00 sec, without fade-in/fade-out a 2005 Hz sinusoid, beginning at 1.031 sec, with a slow fade-...
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2answers
135 views

Recommended resources for noise reduction

I know the question What Resources Are Recommended for an Introduction to Signal Processing (DSP)? and I have already read some general DSP books, such as Rick Lyons's "Understanding DSP," and also ...
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0answers
139 views

Frequency translation with FFT

I'm new to DSP and I would like to make an SDR receiver. For this I want to shift bins in the frequency domain in order to bring to 0Hz any portion of the base band spectrum (I know that there are ...
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1answer
247 views

Convert 96 Khz to 48 Khz audio: is this simple downsampling method ok?

I have an audio signal x sampled at 96 Khz. I have already lowpassed it with a cutoff somewhere between 20 Khz and 22 Khz, so there should be nothing left of ...
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2answers
159 views

Any way to reverse engineer professional audio algorithms or those in e.g. rack units?

Any way to reverse engineer professional audio algorithms or those in e.g. rack units? In order to study how they're implemented? Not suggesting to steal ideas, but as education.
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79 views

harmonic algorithm for extending compressed bandwidth of audio file

I just found this site and feel at home :) I've been playin around with python and some compression ideas and was wondering if there's anything like what I am trying to figure out... I've noticed ...
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1answer
100 views

What is the name of this very simple spectral subtraction technique?

Let $S = X + N$ be the sum of two audio signals $X$ and $N$ which are both stationnary (let's think X is a constant volume 440 Hz sinusoid and N is constant volume noise). If the sum S has a -20 dB ...
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2answers
57 views

How can I calculate what a recording would sound like if it was played through my speakers and mic?

I am trying to isolate the sound of someone speaking while music plays back in a noisy environment. I have multiple speakers and a microphone in my system. I am able to play back just music and ...
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1answer
99 views

What does it take to design a 4th order band-pass filter with -24dB/Octave?

I would like to implement in software the same isolators that the Pioneer EFX-500 features: The manual says the following about the isolators: High-performance 3-band Isolator Function [Fourth-order ...
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1answer
34 views

Audio Processing Used in Voip Mobile Apps

I am new to VoIP apps and am learning about the various audio processing that are applied in such apps. I have looked at Linphone and ...
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1answer
52 views

About speech preprocessing in mobile phones

I understand that cell phones transmit speech signals in digital form through 2G/3G or other standards. But I could not find anything about the signal processing method lying in between the analog ...
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2answers
67 views

Is it accurate to say in digital audio that, when a fader is down, then its value is “-$\infty$”?

Is it accurate to say in digital audio that, when a fader is down, then its value is "-$\infty$"? What has confused me is, whether it's really $-\infty$ or some lower bound of the amplitude. it could ...
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0answers
938 views

Cocktail Party Problem — audio source separation

I am trying to solve the Cocktail Party Problem. I am trying to separate these two mixed audio files: mixed 1 [WAV] mixed 2 [WAV] into 2 separate audio files that contain the 2 original sources, ...
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0answers
64 views

Digital vs Analog music [closed]

What are the different constraints for mastering audio content for either Vinyl or CD? By the way, I'd like to know if there're any papers or documents, with a comparison between different digital ...
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1answer
138 views

How to delay an audio signal by 1 microsecond? [duplicate]

I have an audio signal sampled at 44100Hz. I want to delay this signal by 1 microsecond (and maybe even less) for a steganographic purpose. Now if I delay it one sample, the corresponding time delay ...
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1answer
74 views

How many psychoacoustical critical bands are there?

I understand that the human ear works as a band-pass filter and I have doubt regarding the number of such bandpass filters in the filter bank. I came across these two possible explanations of how it ...
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2answers
372 views

Software to extract frequency spectrum data from an audio file

I've been trying to learn how computers recognize musical notes, and I found apps like spectrogram which shows the frequency spectrum of a sound, but what I need is the data which is plotted or the ...
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1answer
190 views

Hardware requirements for ambient active noise cancallation system using adaptive LMS

Background I'm working on a hobby project to reduce ambient low-frequency noise generated from HVAC equipment within a small area of a room. The noise is predictable and concentrated between 55Hz and ...
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2answers
214 views

how to calculate the maximum obtainable directivity for a microphone array?

Introduction The relationship between the maximum obtainable directivity and aperture of an antenna is described by: $$ D = \frac{4\pi A_e}{\lambda^2} $$ Where $D$ is the directivity, $A_e$ the ...
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0answers
97 views

Subtracting audio signal emitted - trying to use spectral subtraction to localize moving objects

I am a Software Engineer without much signal processing background and currently spending and experimenting to get use to it. My scenario: Assume a speaker and a microphone array. A speaker emits an ...
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0answers
65 views

Given a log-plot of frequency-magnitude-phase how to apply it as an EQ curve to a signal?

I have several hundred data points that represents an EQ correction curve, where each point contains frequency, magnitude, and phase (-1..+1). The frequencies are distributed exponentially, not ...
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3answers
131 views

Resample audio signal with a low factors

Currently, we need to handle a piece of an audio signal. We want to resample it with different factor rate (p/q) starting with 1.0002 to 0.9998 with a decrement of 0.000005. But Matlab function cannot ...
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1answer
46 views

How to generate such sounds as in example? [closed]

I found these files on Internet: 1.wav 2.wav 3.wav 4.wav How to generate such sounds? Geno Chen wrote "Acoustic Grand Piano (Instrument #0)" What to do next?
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2answers
176 views

Audio Steganography: Inaudible Audio Watermarking for Source Identification

What is a relatively inaudible audio watermarking technique to overlay a broadband audio signal on top of playing music without analysing the music for low-bitrate source identification? Ideally ...
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1answer
118 views

How is temporal pre masking possible?

I was just learning about psychoacoustic audio data compression and came across some sources suggesting that a strong tonal component (masker) that occurs within a certain short timespan after a ...
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1answer
193 views

No cross-correlation between same signal recorded using two different microphones

I've recorded the same audio signal using two microphones located at the same approximate location. However, when checking the cross-correlation using MATLAB (using the xcorr function) it is ...
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1answer
87 views

Introductory resources to mathematical properties of “dynamically recalculable” filters in audio/musical equalizers?

Introductory resources to mathematical properties of "dynamically recalculable" filters in audio/musical equalizers? Particularly, I want to understand What mathematical features (e.g. monotonicity ...
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1answer
225 views

Where to begin? - Multi source polyphonic pitch detection

I'm quite interested in real time pitch tracking. I have used mcleod pitch method, Yin and other methods to perform real time pitch tracking for a monophonic audio track. I know that Polyphonic ...
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1answer
84 views

nearest neighbor implementation matlab

I am trying to implement this research paper: I understand everything in the paper except the part about the nearest neighbors calculation (section 3). The only MATLAB function I can think of that ...
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1answer
4k views

Calculating the SNR of Audio Signal (Recommended Libraries)

Are there any open source packages or libraries available which can be useful in calculating the SNR(signal to noise ratio) of an audio signal. The input will be just an audio signal and I have to ...
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0answers
257 views

What are F0 counters in Speech signals? What is represented by F0?

I can't find any dedicated definition for Frequency F0 counters. Can anyone tell me what does F0 exactly mean, I am confused about that.
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3answers
318 views

Decreasing Sample Rate in DFT (FFT) for Audio Analysis

I'm working with audio data analysis through FFT algorithm. My example audio is a sine wave at 440 Hz and 44100 Hz sampling rate. FFT methods in programming, like ...
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1answer
44 views

How to recognize for small video sequence with a large sequence?

I'm working on the idea of recognizing a small video sequence (a sequence of 18 seconds) within a long one (a sequence of 15 minutes), the first idea that come to mind stats that I have to ...
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2answers
3k views

What are some python packages I can use to cut audio files

I have a project in which I have a batch of audio files and I need to remove the audio in it from say time 2secs to 5secs i.e. I have a set duration to be removed. How can I do this in python?
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1answer
214 views

Morphing between two FFTs

I'm a musician and I'm making a script that takes a wave file and snaps each of its frequencies from the fourier transforms to the nearest musical harmonic. Thanks to help from another question I ...
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1answer
84 views

DSP using audio IO from PC using C++

This is my first question on dsp. I am looking for something very simple here and I am sure there are libraries that will do pitch recognition and FFT but my requirement is very simple. I have a PC ...
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0answers
93 views

Generalized Sidelobe Cancelling performance

I'm tinkering with different adaptive beam-forming algorithms for a research project in which I want to use a Uniform Rectangular microphones Array (URA) to isolate speech in a room. I am determining ...
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1answer
769 views

How to simulate analog VU meter response to audio?

I'd like to simulate the movement of an analog VU meter dial as shown below and often found on stereos and other music monitoring equipment. I just learned through this article by Rod Elliot (ESP) ...
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1answer
157 views

Audio transfer function modelling with pink noise + signal

I'm trying to model a transfer function for a noisy audio system, specifically to measure delayed system response. Before I can confidently apply control, I need to verify that I can exert control ...
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3answers
140 views

Detect the beginning of an increasing signal

After denoising and cleaning, I get amplitude signals like this (y-axis: dB): On bottom left of each of these 3 graphs, you can see a noise floor (nearly "horizontal line"). This noise floor ...
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3answers
289 views

How to calculate arbitrary phase shift in discrete signal?

When I have digital audio signal, represented in samples, I can easily phase-shift the signal by 180° by simply inverting the value of each sample. But what if I want to phase shift an arbitrary ...
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2answers
440 views

Amplitude ranges for different bit-depths

In Python, I am using the library Soundfile (which uses the library libsndfile) for digital sound processing. I am working with different sound files coming from different internet databases, and ...
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2answers
312 views

How to edit audios so that they have the same length

I have a dataset of audio files, which differ on their lengths by a couple of milliseconds. Some are little shorter than 1s and some are little longer than 1s. They all have the same sampling rate, ...