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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16 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 ...
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2answers
38 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: ...
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1answer
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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
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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 ...
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1answer
31 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, ...
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1answer
28 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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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 ...
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2answers
65 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 ...
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1answer
29 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 ...
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1answer
58 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
25 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?
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5answers
262 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
36 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 ...
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3answers
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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
62 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
15 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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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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30 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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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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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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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, ...
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4answers
821 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 ...
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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 ...
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1answer
48 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 ...
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1answer
20 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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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 ...
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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
83 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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26 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
195 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 ...
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1answer
39 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 ...
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1answer
44 views

What determines peaks in FFT?

I ran FFT on three audio files and found that the results for some have more peaks than the other. Could anyone give me any conceptual explanation as to what determines these peaks? Below are plots of ...
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2answers
60 views

Parallel equalizer, working for boosts but not cuts

I am trying to replicate the behaviour of a parallel equalizer. The approach is to use a parallel sum of bandpass filters like so: $Response = 1 + bandpass1 * (gain1 - 1) + bandpass2 * (gain2 - 1) + ...
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Hardware that Mixes Two Digital Audio Source into One

Is there any off-the-shelf hardware that extracts two digital audio streaming source from two digital audio stream(like USB+SPDIF), mixes, and transmits in single digital stream real-time(if the ...
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17 views

How to use DTW for keyword spotting in human speech?

I am trying to use Dynamic Time Wrapping (DTW) + KNN to detect word occurrences in human speech audio. I am to implement the same approach suggested in this DSP StackExchange post, but I am seeing ...
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1answer
58 views

How to compute key of a song?

I'm looking for a precise way to compute the key of a song. BPM would be cool too. Analog is ok. I tried to approach it from the DFT angle, but got stuck and can't really put a finger on what's the ...
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38 views

audio processing for audio files quick listening

I need to speed the listening task. I want to listen only 1 second every three seconds. Could you give me some advice? Many thanks! Best regards, Sergio
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1answer
77 views

Biquad high-pass filter not working as expected

I'm trying to implement a high-pass filter using the the coefficients described in RBJ's well-known EQ cookbook (see https://www.w3.org/2011/audio/audio-eq-cookbook.html). However, when I run white ...
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1answer
122 views

Preprocess sound for voice recognition in Pocket Sphinx on Raspberry Pi 3

I want to improve my voice recognition system on a Raspberry Pi 3 running Pocket Spinx. It's pretty accurate (limited recognition vocabulary) but is terrible when other noise is around. Is there any ...
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2answers
20 views

Sudden loudness changes producing unwanted noises?

What's that effect called? Is that one of the reasons for having attack and release parameters when implementing synths? And is there a generic method to avoid such sound artifacts? Example: I have ...
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2answers
77 views

How to reconstruct a sound from magnitude spectrogram?

I have an audio magnitude spectrogram but I don't have the phase, try to randomize the phases of each container and then make a reverse fourier, but only pure noise is heard How can I reconstruct the ...
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2answers
41 views

Can I slow down an audio file by halving the sampling rate?

A question came up, and as a layman I am not sure how to appropriately find an answer; bear with me. Given a 2 minute audio file, say a .wav file sampled at 44,100 Hz, I am curious if it is possible ...
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Granular synthesis of a periodic waveform - How to derive grain start and end point

I understand to find the length of the grain I need to perform an FFT over the source to find the fundamental harmonic, I can do that and find an accurate grain length, but now I need to find a start ...
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1answer
44 views

Are Total Harmonic Distortion and Noise (THD+N) and THD correlated in practice?

I've read about THD and THD+N due to interest in audio quality. Some audio devices (ej. speakers) report %THD or %THD+N. %THD+N is greater than %THD. I've found that sometimes the difference is huge, ...
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1answer
40 views

Does FFT magnitude of a sound drop to 1/10 of its value if sound level is decreased by 20dB?

I understand decibel = 20 * log (magnitude/reference value). To truly appreciate the meaning of this, if I have a single frequency tone that corresponds to a certain FFT magnitude shown on a PC that's ...
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De-Doppler a signal: Audio Timestretch Factor controlled by function f(x)?

Timestretching usually is constant. Few programs offer bézier handles and such to control the lead-in and lead-out. But that's all. I need better. So the audio I have was in fact recorded in ...
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25 views

Applying binary mask to audio signal

I am trying to implement a simple VAD algorithm (a weighted ES VAD method from http://ceur-ws.org/Vol-2259/aics_17.pdf) and have been able to derive a mask that is either a 1 if speech conditions for ...
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2answers
98 views

Half Window Functions and Formulas

I am looking at implementing different window functions in an application I am making. I have found the formulas for all major symmetrical windows, however I would like to implement half windows. ...
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1answer
48 views

Attenuate the low frequencies in a speech wave

I am designing an FIR filter to attenuate the frequencies below 300Hz. I designed the filter using the ...
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2answers
96 views

Phase information of a signal when creating a spectrogram

Sorry for the uninformed question but I need some help understanding this. I'm trying to understand what phase information is and it would be very helpful if someone could correct my understanding of ...