Questions tagged [audio-processing]

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Why HMM transition probabilities plays no significant role in speech modeling

I'm reading a thesis, it mainly uses a GMM-HMM to tackle the acoustic unit discovery task. It proposes a delicate Bayesian framework on top of the GMM-HMM. In page 15 it says Notice that we have not ...
sicheng mao's user avatar
2 votes
2 answers
303 views

Real-Time, efficient Loudness Normalization

I'm working on a Bluetooth speaker system where the same music file results in different SPL up to 10 dBA when played through different devices, such as iPhones and Android smartphones. The system ...
DSP novice's user avatar
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32 views

Estimate the number of words in audio record

I'm trying to estimate the number of words in a record I tried to extract the Short term energy(STE) feature, denoise it, then count the number of peaks in the STE, it gives a very close result to the ...
ahmad alghadban's user avatar
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1 answer
41 views

Is there a Mcu/ DSP IC which can help me convert wav to MP3 (44.1 Khz sampling) in real time and store it into an embedded SD card?

I have built a Bluetooth speaker using an ESP32. It can stream BLE audio and play MP3/WAV from an embedded SD card. I want to record 44.1Khz res audio from it but unfortunately the ESP32 is not ...
hrithik singla's user avatar
0 votes
1 answer
82 views

How to convert digital value from MEMS mic to dB

I have built 3x3 microphone array using 24bit ICS-52000 microphones. I want to convert digital values to decibels. I'm getting data via serial port into MATLAB and I used this code to convert those ...
erysvh's user avatar
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1 vote
3 answers
98 views

How are the results of a fourier-transform interpolated to cover the entire frequency range?

Say we have an audio signal and we do a Fourier transform with block size 2048. Now the outputs correspond to 1024 different frequencies. A spectrum analyzer however doesn't just display 1024 discrete ...
Potheker's user avatar
5 votes
1 answer
91 views

Removing a noise pulse from a signal

I have a sampled audio signal that is corrupted with a noise pulse that occurs at regular intervals. The pulse is in the form of several periods of a sine wave, and I believe the frequency of that ...
gschro's user avatar
  • 103
2 votes
1 answer
166 views

How to interpret FFT output?

I want to perform a spectrum analysis of my own voice using the code below. ...
walter's user avatar
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2 votes
2 answers
104 views

Laplace transform of this simple parallel RLC circuit? (For audio speaker simulation ...)

SPEAKER AS RLC CIRCUIT I read this article here which demonstrates a simulation of a speaker as a simple RLC circuit where the RLC components are in parallel: MY GOAL I am interested in creating a ...
mike's user avatar
  • 523
3 votes
1 answer
353 views

Identifying incorrectly sampled recordings and finding their actual sampling rate

I have a dataset that contains a lot of recordings sampled incorrectly. This incorrect resampling could be explained in two ways: A recording S, which was recorded with a sampling rate of 16kHz, was ...
thanatoz's user avatar
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3 votes
1 answer
151 views

How do amateur (ham) radio noise reduction filters work?

Amateur radio receivers often have several types of audio filters, and a common one is simply called "Noise Reduction." Compared to notch filters, impulse filters, and adjustable bandpass ...
gschro's user avatar
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76 views

Using python to remove Gaussian noise and a high frequency signal from a audio file

I have an audio file where people are talking, and Gaussian noise and high-frequency signals have been added to the file. My goal is to remove the noise from this file using Python. For this ...
Joe's user avatar
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1 vote
0 answers
42 views

In "Mel spectrogram" or "Mel filterbanks", what does Mel mean and why is it often capitalized? [closed]

In "Mel spectrogram" or "Mel filterbanks", what does Mel mean and why is it capitalized ? It doesn't seem to be the name of a person.
f10w's user avatar
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3 votes
1 answer
123 views

Audio EQ Cookbook Stability

I'm implementing 6 cascaded biquad peaking filters to make a 6-band EQ. In the system I want to calculate the coefficients on the fly (it's just gain thats changing, center frequency and BW aren't). I ...
Michael Beckwith's user avatar
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0 answers
30 views

Isn't the MFCC feature good for real time applications?

I am interested in the audio classification problem. You can think of the problem as ML, but my question is from a signal processing perspective. I am questioning the classification performance of ...
Yalçın Cenik's user avatar
1 vote
0 answers
35 views

is there a relation between video and audio

"I would like to extract 'n' frames and obtain the corresponding spectrogram for those frames. I'm unsure if there is a specific duration, such as 20 frames equating to 15 milliseconds or ...
Bala Sai Sigireddy's user avatar
2 votes
2 answers
667 views

Algorithm to "serialize" impulse responses

New user to dsp.stackexchange here. I'd like to know what's a quick and effective way to take 2 impulse responses and create one that's equivalent to both of these "in series" for audio ...
delt's user avatar
  • 129
2 votes
1 answer
248 views

Change pitch of voice

I am programming a voice changer (in c++). I need only to change the pitch of the voice. Unfortunately, I absolutely lack theoretical knowledge and hence I could not find information on the Internet :(...
ONamaeWa's user avatar
0 votes
0 answers
17 views

Understanding Feature Extraction in Automatic Speaker Verification (ASV)

I'm just starting out in the domain of automatic speaker verification, speaker recognition research. Based on the literature I've reviewed, I've summarized my understanding of feature extraction in ...
Hung Dinh Xuan's user avatar
2 votes
0 answers
165 views

Detecting the piano notes using python

I am trying to build a small application that is very similar to simply piano, where I am trying to detect the piano notes, my program will be listening and when a key gets pressed it will estimate ...
Sanju Reddy's user avatar
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0 answers
43 views

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 ...
Mudassir Ahmad's user avatar
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0 answers
77 views

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 ...
SignalWhisperer's user avatar
0 votes
1 answer
104 views

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 ...
Hannes Broschk's user avatar
1 vote
1 answer
115 views

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....
Killuaisaack's user avatar
1 vote
1 answer
70 views

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 ...
Kunal Vaidya's user avatar
2 votes
1 answer
160 views

BPM Detection using Matlab xcorr function

Objective: Finding the BPM of a short drum loop. Approach: ...
ChalupaBatmac's user avatar
0 votes
0 answers
31 views

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} \...
Rocket's user avatar
  • 25
5 votes
4 answers
2k views

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,...
user2183336's user avatar
2 votes
1 answer
41 views

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 ...
Khubaib Ahmad's user avatar
2 votes
2 answers
158 views

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 ...
Hurricane's user avatar
1 vote
1 answer
478 views

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 ...
OctoCat's user avatar
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0 votes
0 answers
64 views

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 ...
phydev's user avatar
  • 1
2 votes
0 answers
33 views

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 ...
Vladyslav Lytvynenko's user avatar
0 votes
2 answers
207 views

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?
chaos24's user avatar
0 votes
1 answer
56 views

What is limited this FXLMS performance

This is simulation data flow: ...
mohammadsdtmnd's user avatar
3 votes
3 answers
236 views

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 ...
Anonymous's user avatar
0 votes
1 answer
138 views

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 ...
Rob Allsopp's user avatar
1 vote
1 answer
245 views

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, ...
Tom's user avatar
  • 11
0 votes
0 answers
53 views

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 ...
Sahil Sharma's user avatar
2 votes
1 answer
99 views

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 ...
Rob Allsopp's user avatar
0 votes
2 answers
200 views

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 ...
DSP_CS's user avatar
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0 votes
1 answer
87 views

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 ...
Mikayla Eckel Cifrese's user avatar
1 vote
1 answer
59 views

DSP microchip in Phonak products

Does anyone know which chips Phonak uses in their products to run the DSP algorithms? Thanks.
Pavel_47's user avatar
-2 votes
1 answer
127 views

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 ...
Ritesh Singh's user avatar
1 vote
1 answer
78 views

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 ) ......
Scott Stensland's user avatar
0 votes
0 answers
30 views

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 ...
user67681's user avatar
0 votes
0 answers
82 views

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 ...
The Impossible Squish's user avatar
0 votes
0 answers
27 views

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 ...
Sajjad Ali's user avatar
0 votes
1 answer
58 views

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 ...
Mason Wang's user avatar
1 vote
0 answers
60 views

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 ...
Engineer's user avatar
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