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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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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
230 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
88 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
98 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
825 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
167 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
150 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
317 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
462 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
330 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
2k 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
128 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
56 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
75 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
196 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
39 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
116 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
87 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
366 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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1answer
65 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
74 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
36 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
355 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
101 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
19 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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1answer
83 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
76 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
141 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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1answer
347 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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1answer
675 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
786 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
144 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
220 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
237 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
107 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
376 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 ...
4
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1answer
308 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
123 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
2k 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: ...
2
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1answer
1k 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
665 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
416 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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1answer
187 views

Halving a floating point time series and it's effect on noise levels

Why does halving a floating point time series (audio) cause the noise shown in the spectogram to change? From having a look, all the steps from getting the first half of X, stft() and ...
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0answers
161 views

i want to mute a particular frequency in real time audio

i want to mute a particular frequency range in real time audio for eg: in audio signal frequency above 3 kHz should be muted for specific interval of time.
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0answers
71 views

Non-uniform resampling - spacing between output samples given speed factors

I'm working on an implementation of time-varying resampling in python. I've got the windowed sinc interpolation itself working nicely, but can't figure out where exactly to sample my input data for ...
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0answers
34 views

Is it possible to detect that sound source is playback of a recording?

I have been trying to find an article where it is described how real-time (or close to real-time) signal processing from a microphone is able to detect that an audio source is playback of a recording ...
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1answer
78 views

Basic sampling excercise - unsure of rationale informing np.arange parameters for time array

I'm extremely new to DSP, so I apologise if it's a very basic question. I'm having issues understanding the following exercise: "Create an array of 201 samples containing a sin wave with a frequency ...
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2answers
96 views

How to minimize peak-to-peak amplitude of periodic waveform through FFT phase shifts?

For one of my music projects, I'm playing back periodic audio signals (by looping single periods). Unfortunately, one of my waveforms sounds too quiet (even at maximum volume). I'm trying to use FFT ...
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2answers
639 views

Removing noise from PCM file [closed]

I want to implement basic noise reduction. I'm recording audio using a microphone mounted on a drone. I would like to filter the audio file from drone noise. What's the best way to do it ? My first ...
0
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1answer
93 views

Audio texture analysis for visualization

I'm making a complex music visualizer as a personal project. I've already determined that I will be using a series of modified Goertzel algorithms for frequency detection to place the frequencies on a ...