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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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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
478 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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99 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
106 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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53 views

Derivative of a MFCC matrix

While reading this (see feature extraction section, subsection 4.1), I came across the following: The per-frame values for each coefficient are summarized across time using the following summary ...
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3answers
157 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
226 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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18 views

Can the Tonality index in psychoacoustics model be positive?

I have an issue with understanding the attached equation of Calculating the Tonality index. If you trace the equation, it appears that it is always less than 1 because the spectrum unpredictability (...
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2answers
252 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, ...
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1answer
540 views

How to convert NIST to RIFF wav file

I have a dataset of audios(namely TIDIGITS, collected at Texas Instruments, Inc.) and they are in .wav format. However, when I try to read them from python, I get the following error: ...
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0answers
94 views

Is it possible to “equalise” a signal by deconvolving the impulse response of the room in which it is to be played?

I am using a test sweep with a flat power spectrum and linear group delay (Optimized Aoshima's Time Stretched Pulse) to measure a room's frequency response. Having obtained the impulse response of the ...
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2answers
48 views

What should I do with FFT result array from microphone input? [duplicate]

I currently want to create a small pitch detector and I have to say that I am very new in this topic of signal processing. At the moment I implemented the FFt algorithm and I got an array of window ...
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2answers
62 views

Noise when dividing or multiplying signal in C

I'm trying to create a echo effect and have managed to add multiple delays but when i try to divide or multiply these numbers the signal is completely distorted and adds a lot of noise to the system. ...
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1answer
79 views

Add noise/signal to mel log spectrogram

I have a predicted mel log spectrogram $S$ and I want to add a noise signal $\eta$ to it. S = log(mel-filterbank(abs/real part of(stft(signal))) The naive way ...
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0answers
36 views

Relationships between $f(t)$ signal and $f(t) \times \sin(\omega t)$ signal

Let's assume that there is a signal $f(t)$, $0\le t \le T$. Then, is there any special meaning of the signal $g(t) =f(t) \times \sin(\omega t)$? Like AM transmission. What is shown/heard when we play ...
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1answer
93 views

audio filter designing

I know the basic filtering techniques available in signal processing.Does audio signal specifically have any different filtering method?.My application is to filter out the background noise in speech ...
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4answers
67 views

Circular buffer wrap around with chorus effect

I am developing an embedded DSP audio processor and am trying to implement a chorus effect. To my understanding, the chorus effect is multiple delayed versions of the original signal, where the delay ...
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1answer
129 views

Noise removal using adaptive noise cancellation algorithms in real time systems

I am newbie in signal processing. I want to remove noise from real time audio signal using adaptive noise cancellation(LMS&RLS) algorithms. I don't know how to choose reference signal(noise signal)...
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34 views

How to give tones to a voice?

I am developing a sing synthesizer application. I planned some steps for it. Parse letters to characters/syllables. Synthesize the letters/syllables to files using ...
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1answer
59 views

Getting a weired behavior when windowing an audio signal

I am processing audio signals using Essentia, an open-source library and tools for audio and music analysis, description and synthesis. Here is a waveform corresponding to a frame extracted from an ...
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0answers
50 views

How to build a software audio mixer - low level approach [duplicate]

my questions might come across different SE forums. But here it is. A little context : I'm currently building a midi sequencer, the architecture is this way (for now) I've got a ARM v7 + a PIC, the ...
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1answer
33 views

Is sparsity induced penalty in source separation “Entrywise matrix norms”?

I am reading this paper where they introduce norm penalties for source separation. In table 1, the $\log/ l_1$ type is $\sum_{g} log(\epsilon + \lVert H_{g} \rVert_1)$. I wonder this $\lVert H_{g} \...
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1answer
241 views

Machine Learning - any suggestions to solve Python rounding errors?

I am working with Python to isolate voiced segments from music using the Jamendo corpus for singing detection. Training a model, I break my audio into frames, and have a label (0,1) for each frame. ...
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1answer
22 views

Repair audio file [closed]

I have an audio recording, recorded using my smartphone. The audio format is .aac and is around 90 minutes long. The first 65 minutes of the audio file plays excellently, but then the audio is kinda ...
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1answer
78 views

FFT phase spectrum to suppress noise?

All that I heard so far about phase spectrum - that it is very hard to process. Are there any methods to do that, and is it possible to extract information about the noise from phase spectrum?
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0answers
18 views

Find which groups of ants are coexisted in the location from their signal profiles using NMF

I try to solve biological problem to tell which groups of ants are co-existed at this location by their sounds in specific time frame. My chosen framework is Non-negative Matrix Factorisation (NMF) ...
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0answers
42 views

Single sideband demodulation via phasing method

Math. I need the math! None of the articles show the math! Need help with demodulation part about "selecting either uppersideband or lowersideband". Modulation. Let's say I will get Lower sideband ...
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43 views

Sound and Vibration Signals- Frequency content

I am not sure if here is the right place to ask such question. I am missing my physics classes right now and maybe you guys can help me. Or give me some tips for researching the theme (biblio) I am ...
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1answer
62 views

How to find when a part of audio starts and ends in another audio?

I have two audio files in which a sentence is read (like singing a song) by two different people. So they have different lengths. They are just vocal, no instrument in it. A1: Audio File 1 A2: Audio ...
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1answer
56 views

FFT radix-2 in audio - resolution issues - logarithmic scale - sample rate VS buffer size

I would like to ask how to deal with audio DSP. Is FFT good at all to deal with audio? I am working on an app, that generates a single impulse, rout it to an EQ plugin, and the output of the plugin is ...
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1answer
106 views

Downsampling in MATLAB - why scaling filter coefficients?

Trying to undertsand how MATLAB does resampling, looking at toolbox/signal/resample.m. This is the part where (downsampling) filter coefficients are calculated: <...
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0answers
131 views

Spectrogram parameters for Audio Fingerprinting

I'm using the specgram() function in matplotlib library to get Spectrogram of songs for Audio Fingerprinting (for musics) and making unique hashes for each song (something like shazam) and im sure im ...
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53 views

For audio FFT, why timebins decreases, when frequency bins increases?

I am using librosa to read a sound file and matplotlib.pyplot.specgram to return its frequencies and time bins. I am using same ...
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373 views

Band pass filter

I am struggling for some time (3 days) with something thaat suppose to be simple (I think). I want to take an mp3 (or a wav, I can convert if easier) and split it to several parts with different ...
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1answer
230 views

how can I make a simple lowpass filter with this constraints? [closed]

I'm searching for a simple low pass filter code in C with this constraints a modulable frequency cutoff (how can I convert all these crazy coefficients into a frequency?) I could change with a slider....
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0answers
24 views

frequency signatures of musical instruments in an audio file

How can an audio signal be analyzed to identify the instruments in it? Let's say the audio contains drums and a guitar. The element that distinguishes these (or any) instruments is its timbre. Can ...
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1answer
378 views

MPEG-7 low-level audio descriptors

I'm working on an application which need to use the full set of 17 MPEG-7 low-level descriptors. Is there any library I can use for it? Preferably with a Python wrapper/API...
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1answer
387 views

How do I resample an audio signal without altering its pitch?

I'm working on concatenating videos again, this time with different playback speeds. The video stream part is easy enough, but the audio part is what's stumping me at the moment. Currently, I've ...
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1answer
101 views

Normalize speech features per dimensions or across dimensions

I am working on pipelines for audio processing (audio features that will go into a neural net to create some embeddings). We are using two features : Filter banks, or MFCC. I was wondering what was ...
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1answer
165 views

extracting MPEG-7 low-level audio descriptors using essentia

According to https://www.ntnu.edu/documents/1001201110/1266017954/DAFx-15_submission_43_v2.pdf, the audio processing toolbox Essentia includes the 100% of MPEG-7 low-level audio descriptors. However I ...
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1answer
98 views

Distinguish harmonic sounds from non-harmonic

I am trying to distinguish harmonic sounds in the range from $100\rm \ Hz$ to $1000\rm \ Hz$ in my audio recording. I attach the spectrogram ($y$-axis is freqeuncy and $x$-axis is time) of a typical ...
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0answers
72 views

low-frequency amplitude with phase modulation of low Hz carrier

I am writing an FM/PM synthesizer, using Java. For the most part it works pretty well. What I am curious about is an attempt to use a low frequency tone as a carrier. By low frequency, I mean in the ...
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1answer
276 views

How does it go about implementing pitch shifting without affecting the duration or speed on matlab? [closed]

I am creating a project that will record a sound sample from the world, for example, a bird's chirping. And then through pitch scaling, I could create different notes. I first tried to change the ...
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1answer
179 views

Algorithm for identifying beat grid in music with non-constant tempo

Are there any algorithms for automatically fitting a beat grid to a song (tempo matching) when the song has a slowly changing tempo or BPM? The output would be a vector of onsets. Here's a picture ...
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1answer
79 views

How can you 'perfectly' reconstruct phase for transients in a modified audio absolute spectrogram?

I am currently working on the absolute spectrogram of an input audio signal. I am using non-negative matrix factorization to separate out the rhythmic components (bass drum, snare drum and hi-hat) of ...
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1answer
1k views

Python, resampling tools dont work as expected

I've tried both using scikit.samplerate.resample and scipy.signal.resample. With neither of them can I achieve the result I want. Scipy.samplerate: ...
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1answer
393 views

Downsampling audio for use in Machine Learning

I'm trying to use the work (Neural Networks) done in this repo: https://github.com/jtkim-kaist/VAD It says this: Note: To apply this toolkit to other speech data, the speech data should be ...
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2answers
431 views

sound classification

Hello I am trying to do sound classification in matlab. I have different samples of sounds for 2 seconds. How can I proceed with that. The sounds I am using are churchbell, footsteps, trains, sirens ...
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2answers
250 views

Calculate algorithm's MIPS from cycles?

My audio DSP algorithm processes input sound file (45 seconds, 16 kHz, mono). After simulation is done, gprof profiler tells me: ...
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0answers
16 views

MFFC - Source moving towards the recorder in time

I'm doing my master thesis, it's about classifying ships using sonars. I'm using MFCC to extract the features. The question that I have is, since the ship will move towards the sonar in time, will I ...