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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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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
103 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
74 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
174 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
60 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
53 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
34 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
279 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
89 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
71 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
59 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
117 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
146 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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1answer
262 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
477 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
453 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
111 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
197 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
169 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
85 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
306 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
213 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
93 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
561 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
507 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
283 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
122 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
127 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
56 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
28 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
68 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
81 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
492 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 ...
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1answer
79 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 ...
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1answer
176 views

Maximum dB SPL from a digital microphone

Basic sanity-check question: if I have a digital microphone with a sensitivity of -26 dBFS (say, this one), does this mean that the max dB SPL it can handle is 120 dB SPL? My thinking is that if the ...
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1answer
68 views

Does the sound volume affects the MP3 file size?

Let's say I have an X music track in some lossless format (wav?). If I convert it to MP3 format with its original volume and get the file "A.mp3". Then I edit the lossless source and reduce the ...
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2answers
81 views

Approximate the change in perceived gain from applying transfer function to audio signal

I have an audio signal that I am applying an analogue transfer function $H(s)$ to. I would like to approximate the change in gain ($\Delta \text{ Gain}$) as perceived by the listener for an arbitrary ...
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1answer
2k views

How to get the volume level from PCM audio data?

I have a decoded PCM audio data obtained from the Web Audio API. I'd like to generate a chart that represents the volume level (perceived by humans) over time. ...
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1answer
121 views

How do I deepen my voice (I'm a female) to sound like a man in a video for a school project? [closed]

Some extra info. that I think might help or make things clearer: The video has several other people in it also talking (I don't want to change their voices). I have some experience in editing ...
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3answers
138 views

Features Used for Instrument Recognition?

I'm building a neural network for instrument recognition based on the IRMAS data set. However, I'm a bit at a loss for what features to use. MFCC seems to be quite popular but extracting this per ...
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2answers
1k views

What does “Lag” mean in terms of Cross Correlation?

I am working on a program using the Essentia library for audio analysis. I've implemented the Cross Correlation algorithm (here: http://essentia.upf.edu/documentation/reference/std_CrossCorrelation....
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2answers
123 views

Filter away sinusoidal noise properly

I have a stereo music signal corrupted by strong sinusoidal noise that varies over time. Here is the spectrogram of Left channel I plotted with Matlab. As you can see there are 3 or 4 strong ...
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1answer
81 views

How to keep the same power (sonority) between pure sines?

I have a list of sine waves in 1/3 Octave (63,80,100...1k,1.25k,etc) that I want to reproduce with the same power. Right now I just have a global gain that applies to all this freqs, but of course the ...
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1answer
124 views

Calculating gain coefficients for comb filters in Moorer's algorithm

Could someone explain how Moorer calculated the gain coefficients for the comb filters in his reverb? In his book, Udo Zölzer's layed out this equation: $$g = 10 ^ {-3 \left(\mathrm{Reverb \ Time} \...
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1answer
52 views

How to correctly classify voice from other sound events?

I have a huge labeled dataset of several thousand sound events, including human voice, dish washing, things falling to the ground, among others. I need to report when a human voice event takes place....
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0answers
65 views

Processing digital sound in Audiokit to sound like the 1960's

I am looking to imitate 70's analog equipments sound. I was wondering is there a way to create tape saturation / distortion inside Audiokit? I want to add that and randomized tape crackles and ...
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3answers
940 views

Get frequency and amplitude from an audio sample [closed]

I'm wondering how can i get the frequency and the amplitude from a 16 bits sample. I have actually an array of 16 bits signed int and I would like to get the frequency and the amplitude of each sample....
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1answer
622 views

How to convert dB back to manitude and then to rectangular format [duplicate]

I am experimenting with audio (wav files) using Short-Time Fourier Transform (STFT) in Python using scipy.signal.stft. As I ...