Questions tagged [audio-processing]
The audio-processing tag has no usage guidance.
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Audio Classification
Most of the paper about audio classification has worked like this they have specific sounds class wise and the did classification. What if the audio files i have collected have two different sounds at ...
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How to Eliminate Clicking Sounds in WAV File Generated from NFM Demodulation using FFTW?
I am working on a Digital Signal Processing (DSP) project where I am processing signals received from an SDR (Software Defined Radio) device. I am using the SoapySDR library to interface with the SDR ...
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How to obtain an impulse response from an exponential sine sweep (23kHz/50kHz)
I am currently working on my master thesis and have so far generated a broadband ESS to obtain frequency response and impulse response.
To generate the ESS as an excitation signal I use the following ...
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How to merge audio better?
I try to use a denoising algorithm to process an audio in streaming mode.
What I've done is spliting the audio into 0.5s long segments and use denoising algorithm respectively.Then use np.cat or np....
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request pointers on beginner DSP board for making an EQ simulating hearing aid prescription with TV sound
I want to make an equalizer set to similar frequency cut-boost according to the results from the graph from hearing test, where an audiometrist will play tones of various frequencies at various ...
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BPM Detection using Matlab xcorr function
Objective: Finding the BPM of a short drum loop.
Approach:
...
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How do you handle zeros when logarithmizing? [duplicate]
I'm trying to implement a procedure from a paper.
There the authors use two equations to transform a fft transformed music audio file to a cent-spectogram.
Equation 1:
$$ f_{cent} = 1200 \log_{2} \...
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What is the audible level for digital audio dB units?
What is the audible level for digital audio dB units?
I was looking at a soundwave from an mp3 file. I am using the "Sonic Visualizer" app available on Linux. The amplitude is between ~ -0.1,...
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Dominant Speaker Selection in a Proximity Cluster of Smartphones with Arbitrary Positions (Not static)
I am working on an algorithm for dominant speaker selection in a closed proximity cluster of smartphones (iOS and Android) with arbitrary positions. The smartphones act as clients, and one of them ...
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Anonymize / Obfuscate speech when doing audio classification
Let me preface that I am new to audio processing and audio analysis ;) (I asked the same question on reddit, I wanted to increase it's reach))
I am trying to classify specific events (like a gong or ...
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Plotting a Manually made Spectrogram with python
I'm following a guide about signal processing, but since I'm a fresher to the domain, the guide just stops at a point where only a function that could return the spectrogram values is written. So, my ...
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What should be a sensible range for the various librosa parameters to extract features from a very high sample rate (2MHz) audio signal?
I am trying to implement a deep learning classification solution for (1 second long) audio signals recorded at a very high sample rate, i.e. 2 MHz. All the material and tutorials that I could find ...
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Smoothing clicks when scrubbing/seeking through the audio
I have a buffer of N samples from which I'm playing a loop of L samples (L <= N) starting from sample S. The reading position is calculated as (S + P) % N where P is current playhead position which ...
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Effect of changing sampling rate of an audio signal
What will happen when sampling rate of an audio signal is changed? Does it changes signals duration or signals total sample or both?
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Resource recommendations to learn audio processing
I'm looking for some good resources to learn audio processing for a machine learning task based on classification of users as either 'COVID-19 positive' or 'COVID-19 negative' based on their cough ...
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How to produce a causal, linear phase filter kernel with an arbitrary magnitude frequency response
I have been reading The Scientist and Engineer's Guide to DSP to learn how to create filter kernels with an arbitrary frequency response (I design the magnitude response by hand). The method proposed ...
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How to choose PCM Buffer Size?
I am new to MEMS digital microphone world. Currently, we are using built in PDM to PCM converter available in microcontroller.
The problem that is troubling me is how to choose the PCM buffer size, ...
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Is it possible to work with DSD audio on MATLAB?
Just for starters: I am very new to audio processing
I was looking for audio data that is in 1-bit format and I stumbled upon DSD. From what I was able to read it is old but kind of controversial ...
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Custom DFT filter adding odd sound to audio
I am attempting to design a custom filter for equalizing an audio track using the DFT. I am newish at this, but my understanding from the DSP Guide is that you can do this with the knowledge that ...
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confusion understanding/interpreting power spectrum of audio in MATLAB?
Please find attached snapshot. I am confused regarding power spectrum in MATLAB as according to google definition in attached snapshot"the distribution of the energy of a waveform among its ...
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How can sound waves be modeled in a manner that distinguishes individual voices but also recognizes words?
More specifically, I'm curious how we can represent sound waves in a way that would both distinguish between individual voices and also recognize, e.g. that recordings of different people saying the ...
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DSP microchip in Phonak products
Does anyone know which chips Phonak uses in their products to run the DSP algorithms?
Thanks.
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How to export raw audio from audacity?
Request to Experts:
I am a DSP novice. I would like to experiment with some methods for low pass filtering in the time domain.
To that purpose, I would like to generate a tone in audacity and export ...
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Need to make audio human listenable
I wrote some golang code which implements 3Blue1Brown's Hilbert's Curve: Is infinite math useful? ... give the code an image and it returns back its audio equivalent ( inverse Fourier Transform ) ......
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How would I figure out the frequency with which a recording of vibrato oscillates? [duplicate]
I'm not sure what the correct signal processing tool for this task is. My file is in m4a format. I'm comfortable using Python (arbitrary library(ies)) or MATLAB, but don't need code, just advice on an ...
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Which audio smoothing algorithm has the lowest latency?
I am working on an audio smoothing project. My inputs are from a record scratching MIDI controller, and I am trying to produce a smooth output (the raw data creates noisy audio).
I'm using a simple ...
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Modulating and transmitting three different voice signals (in MATLAB)
I want to record, modulate and transmit three different voice signals. I will be doing this in MATLAB.
Please can you help me figure out how can I do this?
Or give me a block diagram that can show ...
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Studio Monitor's Cone not making sound after playing loud music through it
I was playing some very loud pop music through a studio monitor (ADAM A8X), and after about 5 seconds the sound stopped coming out of the cone and only came through the speaker. After shutting it off ...
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Sliding 2D Fast Fourier Transform
How to implement sliding 2D FFT for audio data (shown in the link - R1-chirp)?
I have audio data for an array of 1024 microphone which record audio (for 2 seconds) from a sound source. Microphones are ...
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construct R data frame from audio file of sounds recorded from game app
I have an audio file (u8, 128kbps) of the sounds produced while playing a game app and clips of its three distinct sounds.
The file consists of sequences of these, separated by silences, indicating ...
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How to normalize(or other) the audio data so that the same labels with the similar characteristics from different records?
I am trying to detect swallows from recordings taken from hospital. I manually labelled the recordings on the Praat. Now the valid labels are silence, swallows and nonswallows(noise, enviromenment ...
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Identify abrupt changes in an audio waveform
I have an audio file which was created by substituting multiple speech segments into an original utterance.
I want to find the timestamps where such substitutions happened, so I guess that one thing I ...
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Is adding a mean point between every two points a valid way of upsampling?
Suppose we need to double the sampling rate of an audio signal.
Is adding a mean point between every two points a valid way of upsampling? If no, then why not?
For example, in images, pixels may be ...
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AES encryption of an audio file
I have an audio file that I encrypted and decrypted using the AES algorithm.
After the decryption is complete I output the audio file to a new file.
I wanted to calculate the root mean square between ...
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Comparing realtime guitar sound input to a .wav file
I am currently working on a guitar training progam that compares guitar input from a guitar to a song as you play, to give a score of how well you're playing. The reference song is converted from MIDI ...
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What are the step by step actions taken in order to convert PDM data to PCM data?
I would like to convert PDM (Pulse Density Modulation) microphone data into PCM (Pulse Code Modulation) data. I understand that ...
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What causes the mismatch between a calculated sine wave and a measured sine wave?
Note: this question has been edited to refer to a sample that does not use dithering.
I recently asked my first question on here, about what I saw in an uncompressed audio file written out as integers....
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Examples of audio files that are challenging to cross-fade
What can be recommended as challenging tests for cross-fading/mixing of music?
I am looking for examples of music-titles that exhibit severe volume dips when cross-faded or mixed.
Ideally answers ...
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Preprocessing lung sound data: replication of paper method
I'm really new to signal processing research. I'm trying to replicate
Kirill Kochetov and Andrey Filchenkov. 2021. Generative Adversarial Networks for Respiratory Sound Augmentation. In Proceedings ...
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How do I recognize the most important key or changing points in an audio file?
I would like to analyse a wav audio file and determine the key passages of the song using python.
To do this, I have so far first analysed the root mean square and then formed the derivative, which ...
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What highpass filter is implemented in Audacity?
When I selected highpass filter effect in Audacity (an open-source audio editor), it lets me choose the cutoff frequency and the roll-off but it doesn't specify which filter type is used (IIR ...
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How to detect "buzz" components in audio by frequency domain analysis
I am trying to detect the amplitude, frequency and phase of the dominant "buzz" in an audio signal. By "buzz" I mean a periodic signal with sharp edges. For example, an impulse ...
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Setting ERBSpace for Filtering Audio (Audio Classification)
I have been creating different representations of audio as inputs to neural networks. I set my ERBSpace lower and upperbound to (20hz,20000hz) for the cochleagrams. Works without a hitch, and the ...
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Find the position of a short music track in a longer track
I have a long music track (2 hours) which consists of many smaller tracks (1 min each in avg). How to programmatically or using a service to determine the time position of a particular small track in ...
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Am I using FIR filters correctly for audio filtering?
I'm new to DSP and I'm interested primarily in the implementation of audio filters. I've been reading about FIR filters as they seemed like a good place to start but now I'm stuck. I've written some ...
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Order of operations when a real audio signal becomes complex
I have a real audio signal $x[n]$, and I'd like to apply a frequency shift by $f$ and then envelope it by $\cos(\omega n + \phi)$.
If I wanted to do the frequency shift in the ideal sense, from posts ...
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Log of Filterbank Energies
In common literature, when generating spectrograms, mel-spectrograms, and cochleagrams, the log of the resulting filterbank energies is taken. Why is this done? I notice that my convolutional neural ...
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How can I best quantitatively compare multiple 'de-bleeding' attempts?
Let $\mathit{S}$ and $\mathit{T}$ be audio tracks, and suppose that $\mathit{S}$ is the superposition of
An audio signal $x(t)$ of interest, and
A signal that is audibly quieter than, but otherwise ...
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Audio smooth / enhance algorithm, software
Years ago I was working with a sound editor program called Sony Sound Forge. It had this filter of smooth/enhance which I liked and used a lot, but now I can't find such filter in Adobe Audition or ...
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