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.
1,162
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27 views
How can I append time series data into a finite array with a vanishing point?
I'm not sure what this is called in DSP terms, but I'm looking for a way I can append a new time series data point to an finite array and have it decay slowly over time. The best way I can visualize ...
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0answers
8 views
What features capture the semantics of a sentence from a gien audio?
I have been thinking lately and looking up on what could be the possible features that capture the semantics of an audio and came across MFCC: MFCC features represent phonemes (distinct units of sound)...
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1answer
28 views
FastICA: what happens when we have more source than channel?
I am reading the original article proposing FastICA and I have a couple of question on point not covered in the article. Both answering me or providing a source will be very appreciated.
I want to ...
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1answer
46 views
How to normalize frequency response charts
I want write a simplistic program that can transform a high-quality voice recording in a way that it sounds like it was recorded using a specific low-quality microphone.
I have obtained the mic's FR ...
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1answer
31 views
Is it possible for Matlab to read and decode an mp3 file?
I have run the following Matlab code and I got a signal as shown in the Figure below.
...
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1answer
36 views
Can an audio recording tell whether it is made in a large or small room?
The conversation of one or more people will be analyzed. I think need to look for something similar to reverberation or echo in an audio signal. How can this be implemented?
Update: small room is ...
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1answer
77 views
Is there any science to explain the “weight” or “3D depth” of hardware audio vs software that some people claim?
Is there any science to explain the "weight" or "3D depth" of hardware audio vs software that some people claim?
There's, I think, strong subjective evidence to suggest that the phenomenon is true. ...
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1answer
24 views
Pitch detection using Cepstrum in matlab
What i am trying to do is to detect the pitch of a given sound signal, so for that purpose, i wanted to find cepstrum of a given signal.
This is my code in MATLAB:
...
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1answer
42 views
How convert Given DBFS into DB in iOS?
I am getting DBFS value of an audio ranging from -160 to 1.
I am looking for to create a SPL meter in iOS to show actual DB values of audio capture through mic in device.
Using recorder class, I'm ...
1
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1answer
23 views
Digital Audio Mixing Fundamental. What will be the data rate of mixed audio?
I have voice data at 128 kbps coming from three different sources. I'm looking for mixing all three voices together and sending it back to speakers. What methodologies are there for mixing digital ...
0
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1answer
79 views
Which (Fourier transform?) processing method to use for time variant audio processing? (newbie)
Edit:
I've since been able to understand the how the relvant parts of Fourier transform work and how they relate to my problem / what I want to do.
And have come to the conclusion that Fourier ...
0
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1answer
27 views
Hydrophone sensitivity rating
If I use a hydrophone with the sensitivity given as: "-200 dB re 1V/μPa".
Question: How should I interpret this? Is it the voltage given at -200 dB SPL or??
All the Best!
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1answer
36 views
Shazam / SoundHound Music Analysis - Recommendation for Reading
I need to explore some area around music frequency analysis.
A good example is the way Shazam / SoundHound app works (mapping frequency domain, comparing, creating some vector map, handling noises and ...
3
votes
0answers
36 views
Audio sharpening
I'm working on a project where the purpose is to enhance FFT peaks in a signal (musical instruments), by other words, reduce non-harmonic components and enhance the identified FFT peaks. So far I'm ...
3
votes
1answer
44 views
Given 1 recording of my voice, how can I process it to sound like 2 independent recordings of my voice played in unison?
I have one recording of my voice $x[n]$ singing a song. One thing I could do is independently record me singing the song again to obtain $y[n]$. Then, $z[n] = \frac{1}{2}\left(x[n] + y[n]\right)$ ...
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1answer
15 views
script to check zero crossings to check if microphones phases are aligned
I have 'n' microphones placed in square fashion and I want to make sure that all the channels must be aligned in time exactly the same if the signal is equidistant from all the 'n' microphones i.e. in ...
0
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1answer
33 views
generate a repeated square pulse train in python
Generate a pulse train with -2dBFS pulses of 20 sample duration every 8 seconds. I have referred here to generate a pulse train but not able to repeat it. How can I repeat this audio signal every 8 ...
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0answers
26 views
Mixing signal with noise in frequency space (STFT)
I am working on a preprocessing function which is supposed to mix two audio signals clean and noise in the context of neural ...
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0answers
24 views
STFT to spectrogram
I would like to know whether I am correct in my understanding of going from STFT to a spectrogram. My goal is to convert a spectrogram back to a wav file.
If I have my STFT:
...
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0answers
22 views
What are the missing steps in the derivation of this equation?
I am trying to understand the model of a piano hammer described here. The relevant excerpt is:
As I understand it, they are saying in (12a) that if the hammer spring's uncompressed end's y position ...
0
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2answers
41 views
How to fix frequency drift when producing an arbitrary length signal by repeatedly looping over a single-period waveform?
I have an audio application that produces an arbitrary length signal by repeatedly looping over a single-period waveform. In pseudo-code:
...
0
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1answer
32 views
How to measure the “correctness” of a pink noise generator?
I've implemented a pink noise generator for audio applications. How can I measure how close of an ideal pink noise is my output?
I do not have access to Matlab or Mathematica. I would prefer a ...
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1answer
49 views
audio classification based only on first few seconds
I record audio notes on my recorder, and each belongs to a certain category ("work", "parenting", "miscellaneous", etc.). At the beginning of each recording, I say the name of the category, then stay ...
0
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1answer
38 views
How to chose a kernel function for a FIR low pass filter in audio processing?
As the answer of a previous question, I was suggested to use a FIR filter to act as a low-pass filter to reconstruct the envelope of a sound wave (edited, emphasis mine):
As I said in a comment, ...
2
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1answer
31 views
Synchronising audio in realtime without knowledge of original signal
Was doubting whether this is the right SE forum to ask my (elaborate) question, so don’t hesitate to let me know if it’s not. Also, if anyone knows of a better way to title the question, feel free to ...
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0answers
17 views
ICA School Project
I have been stuck on my school project that consists of independent component analysis.
My code will run, but only 50-60% of the time it unmixes the signals. Another problem I have is the amplitudes ...
0
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2answers
73 views
Envelope reconstruction algorithm for audio processing
I need to reconstruct the envelope of a sound.
Audio data are professionally-recorded natural sounds (speech, bird songs) with very little noise. I would prefer working in the time domain rather ...
0
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1answer
32 views
Detect playback position in swept sine sweep
I am wondering what the most precise method is to detect the playback sample position of a sine sweep? The real playback sample position is unknown and an DSP with constraint computing capabilities ...
3
votes
1answer
61 views
Count heart beat from audio capture using smartphone
I am working on a project where I am required to perform heart rate BPM calculation from an audio signal captured using smartphone microphone / hands-free. Constraint is not to use ML. So I am left to ...
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1answer
27 views
Phase response in a filter for an Audio Application
Should I be concerned with the group phase delay which a filter produces? Especially in an audio application? Should I aim at using a linear phase delay filter for example?
3
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5answers
284 views
What Are The Semantics Of Wav-File Sample Values?
The title is the question. To make this concrete, assume I have a standard 16-bit single channel wav file, with (although it doesn't matter) a 16000 Hz sampling rate.
Each sample magnitude will be a ...
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2answers
66 views
Sine wave amplitude modulation - Analysis using Audacity
I am trying amplitude modulation in C++ on a sine wave which is a 1000 Hz sine wave sampled at 48000 Hz. The original sine wave is like this (In Audacity) -
When I basically double the amplitude for ...
5
votes
3answers
1k views
How do impulse response guitar amp simulators work?
I am wondering how impulse response guitar amp simulators/modelers work. I thought it was a matter of convolving a signal of recorded impulse response in time-space with a guitar sample.
I tried to ...
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4answers
64 views
concatenating short noise sequences
I'm trying to merge 2 identical audio files. The Audio file is nothing more than white noise generated using MATLAB commands
...
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0answers
17 views
Why can I not record ambisonics in Unity with Google Resonance?
I've set up Unity (2018.4.0f1 LTS) with the Google Resonance project and I'm attempting to record an ambisonic environment with the Resonance Audio Listener component - however on hitting Record, ...
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0answers
22 views
How to understand the sound parameters so that it can be reproduced in Matlab
I am reading a paper called "Mouse Auditory Brainstem Response Testing".
In order to make the experiment myself I would like to create the stimulus sound using matlab.
When describing the sound (in ...
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0answers
32 views
FFT frequency-magnitude not correct when a positive gain EQ point is applied
I have a measurement system, which applies a log sine sweep as stimulus to a DUT and records the return signal.
I then extract the impulse response, and perform an FFT on this to examing phase and ...
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0answers
35 views
What is the exact compression that is happening from the first sequence of raw audio waveform data to the second sequence? [closed]
The first sequence of audio samples is recorded in JavaScript from the browser in mono, PCM format, 32-bit, and 96,000 Hz.
I wrote this audio file as a blob to a server through a JavaScript FormData ...
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0answers
17 views
Digital signal processing how audio wave and function chage with respect to each other
If an audio signal is f(t) and if i play same audio with 2× speed then what will function of 2× speed signal f(2t) or f(t/2) and why?
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0answers
16 views
Is there a way to filter out low volume content in an audio sample?
Imagine you are listening to a digital file that contains some random speech.
Now, while you are listening to it, you also hear in the background (spatially) some low volume content, either noise, ...
3
votes
4answers
836 views
Noise reduction using multiple recordings of the same signal
I have two (audio) recordings of one analog signal, I used two separate devices (smart-phones) recording from two different locations.
Each recording device is pretty crappy, lot's of electrical hum ...
0
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1answer
41 views
How to synchronize audio and video streaming for not fixed FPS streaming?
It's a 3-stage process.
In the first stage, a video (with audio) is recorded at a fixed 25 FPS rate. In the second stage, it's demuxed in two different channels and the video is processed. In the ...
1
vote
1answer
54 views
Feature Extraction of Insect Sounds
First of all, forgive me if I sound stupid. I am still learning Audio Processing.
So I am trying to make a machine learning model for detecting a specific insect through sound.
Here is a sample raw ...
0
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1answer
37 views
Generating spectrogram for band-pass filtered wav file python
I have some .wav files of heart sounds. I want to apply band-pass filter [20Hx 100 Hz]. Then generating PCG images size of 560x420 pixels and RGB format. How could i do this?
Any help will be highly ...
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0answers
24 views
Using a Multirate FFT system to analyse audio
I am trying to implement a multirate fft system to try and achieve frequency windowed analysis of audio.
Say I have managed to get an impulse response of a system. This impulse may be say one ...
0
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0answers
16 views
Is it possible to restore audio that has been tonally modified by a phase vocoder?
My objective is to apply some process to the modified audio that allows me to obtain the original audio. I think it can be solved through machine learning techniques. I don't know if there is any ...
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2answers
113 views
how to remove noise from heart sounds?
i have .wav files acquired through putting android cell phone MIC directly on chest of person.For feature extraction, i have to made them noise free. These recordings were taken in rooms with no noise ...
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0answers
27 views
Finding frequencies(and bandwidth) that works best audio in a certain medium
Complete DSP and coding newbie(learning python). I am trying to find frequencies that can pass through well in a musical instrument tube. I am trying to find audio frequencies(audible and inaudible ...
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3answers
212 views
Low-Pass Filtering on Small Chunks of Audio
I've got a script (Python) that chops 44.1 kHz realtime audio into 10 ms chunks, processes each chunk, and stitches the chunks back together with no added artifacts. I'm trying to create a low-pass ...
0
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1answer
43 views
Cone of Confusion
In sound perception, humans adapt by tilting the head or relying on other location cues such as visual input when the source of the sound lies on the cone of confusion. On this cone, no time and level ...
