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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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6
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2answers
10k views

Can you turn a square wave into a sine wave using a low pass filter?

And if it could, would it make the sound of the square wave thinner than before because of losing its harmonics?
6
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2answers
735 views

Difference between jpg compression and mp3 compression

My understanding of jpg compression is that, if done repeatedly, any image will eventually be reduced to a single flat color, due to the way it deals with neighboring pixels. However, years ago I was ...
6
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3answers
4k views

Algorithm to pan audio

I'm looking for a way to pan the PCM audio I have. At the moment the audio is in a stereo format and I would like to pan it to the desired side (or more/less to a side). I've recently figured out how ...
6
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4answers
8k views

Equal power crossfade

I am working on an audio system where I need to implement an equal power crossfade between two signals over a specified number of samples. Initially I implemented a linear fade but this creates a dip ...
6
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3answers
2k views

Common methods of transmitting digital data through an extremely noisy audio signal?

I am developing software for transmitting data from a computer to a mobile device using audio. Specifically, from the computer's speaker to the device's microphone. (I am developing both the sender ...
6
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1answer
2k views

Audio fingerprint questions

I am developing an application that requires audio fingerprints. I have been reading a lot of articles and PDFs, now i think i have gotten myself confused. Based on my present understanding, I have ...
6
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1answer
234 views

Morphing between two FFTs

I'm a musician and I'm making a script that takes a wave file and snaps each of its frequencies from the fourier transforms to the nearest musical harmonic. Thanks to help from another question I ...
6
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4answers
2k views

How to recognize patterns in a noisy audio frequency spectrum?

I have an audio spectrum (FFT) that looks like this: ...
6
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1answer
151 views

Bandwidth expansion by computing missing harmonics in music?

In old recordings or highly compressed audio files (lossy), the higher frequencies are lost, and we hear a distinctive muffling effect. (Just like if the song was played through a wall) Is it ...
6
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1answer
2k views

How do I construct input to neural network from audio signals?

Input: Microphone recordings of digits from 0 to 9 from different speakers. Output: The digit from 0 to 9. I am doing this for fun. So first I will train my neural network using some samples and ...
6
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1answer
4k views

SPL values from FFT of microphone signal

I am trying to get a frequency response vs dB SPL plot for recorded sounds. I have a microphone with a flat response, a sensitivity of 4mV/Pa, and 1 Pa = 94dB. To get the dB SPL, I have the following ...
5
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4answers
7k views

Calculate Phase angle between two signals i.e. digital Phase meter

I want to calculate the phase difference between the two channels(signals) of a stereo audio file.(uncompressed data i.e. PCM samples). I want to determine whether whether two signals are in phase or ...
5
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3answers
7k views

Generalized Cross Correlation

What is the definition of Generalized Cross Correlation? I need it for estimating the delay (time offset) between two audio signals. The time offset between the two signals might be as large as 500ms....
5
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3answers
690 views

Digital delay effect: avoiding clicks on delay time change

I'm implementing a digital audio delay effect using a simple circular delay line. Changing the delay time by making the delay line longer or shorter introduces discontinuities in the audio signal, ...
5
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2answers
13k views

What is spectral entropy?

Can anyone explain what spectral entropy is? Does noise with a restricted bandwidth have the same spectral entropy as white noise?
5
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2answers
268 views

Is there any way to make a sampling rate reduction in the frequency domain?

Is there any way to take advantage of the frequency domain to make a clean/nice sampling rate reduction effect ? and how ? Basically, deducted from my tests: if I set to 0 the magnitude uppon a ...
5
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4answers
870 views

Monotonic, symmetrical soft-clipping polynomial

Okay, I intend to answer this question myself, but I would like other folks to pipe in on it. The issues or requirements are this: Input to soft-clipper is $x(t)$, output is $y(t)$. Output $y(t)$ ...
5
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2answers
240 views

How do the English pronunciations of “tiu” and “tou” differ?

I'm creating an app (I'm a software developer) that for different sounds (interjections from the user) will do different things. For example if the user says: "Tiu", than it will do task A, if it ...
5
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2answers
4k views

How to stitch together MFCCs from multiple frames?

I am trying to record a sound two times and compare the two recordings. I am playing exactly the same sound both times (for example, play a small .wav file twice and record it each time), so I hoped ...
5
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3answers
5k views

How to change speed of audio samples without changing pitch? [duplicate]

This question may be more appropriate on music stack exchange but since sound is a branch of physics and this is more of a technical question relating to sound waves than a musical question I asked it ...
5
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5answers
272 views

Remove a known wav file from recorded file

I looked at many similar questions but still cannot find an answer to my problem. I'm playing an audio file N, while recording a man talking (b). The problem is that the microphone gets the played ...
5
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2answers
1k views

Calculating bpm from machine sound (not music)

I have a hydraulic breaker on a excavator that hammers with a dynamic bpm. I have recorded this sound so that it is in a mp3-file and would like to determine the momentary bpm for the length of the ...
5
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2answers
900 views

How to prevent “zipping” effect on a modulated fractional audio delay line? (Flanger)

I am implementing a Flanger using a fractional delay line. I am modulating the length of the delay line using a sin function. The delay line already uses linear interpolation to compute the delay ...
5
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3answers
10k views

Is there any free software that can recognize and classify sounds?

i am making a smart home automation software and i would like to know if there is any software that can recognize noises such as speech ,playing music,phone ringing etc The exact software i am looking ...
5
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1answer
63 views

I trigger recording when power of current sample is X dB greater than last Y samples, what is the terminology here?

Doing time domain processing of incoming audio. I sample the power level every 0.25s and keeping a running average of intensity over 5 samples. When the power level of the current sample is 10dB ...
5
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1answer
4k views

Voice Audio Detection algorithm

I have to detect speech intervals in large pre-recorded files in my project. I think, there won't be so much noise in the background (audio will be recordered in the room or even in the studio), but ...
5
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1answer
982 views

The Bode frequency shifter

Anyone know if the Bode frequency shifter has been studied somewhere. I just used an audio plug-in called EchoBode from SonicCharge and this is claimed to use a frequency (pitch) shifting algorithm ...
5
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1answer
4k views

Signal filtering using Python

I'm trying to create an application using python that is capable of recording an audio signal and detecting short glitches in the signal. For example, I could be recording a 1000Hz sine wave and ...
5
votes
1answer
177 views

How to create a sine wave that transitions through frequencies and returns to phase

For an auditory psychological experiment, I need to create a sine wave, such that at a given point the frequency of the wave will rise smoothly, and descend back to base frequency. I need it to return ...
5
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2answers
297 views

Audio DSP HW Recommendation

I'm very new to DSP hardware, so I am taking it upon myself to learn it with regard to Audio and Acoustic Signal Processing. I've learned quite a bit of theory in a graduate DSP course, but absolutely ...
5
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2answers
384 views

how to extract telephone audio only from an mp3

I have mp3s of radio talk shows in which the host interviews people on the phone. The host uses a microphone, and the interviewee uses either a landline phone or a cell phone. I want to save just the ...
4
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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 ...
4
votes
3answers
13k views

python; audio; detecting silence in audio signal

I have real-time audio application written in python, and I have question about detecting silence. What's the most efficient way to distinguish silence from non-silence, in an input audio signal? By ...
4
votes
3answers
5k views

Reconstruction of Audio Signal from its Absolute Spectrogram

I have the absolute Spectrogram of an audio signals. I lost the phase data of the Spectogram because of various processing applied on the original spectrogram of the signal. I'm trying to reconstruct ...
4
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5answers
2k views

Does it make sense to have complex numbers representing real-world audio signals?

In this course (Coursera: Audio Signal Processing for Music Applications) the professor uses an example of obtaining the DFT of a complex sinusoid: \begin{align} x_2[n]&=e^{\jmath\left(2\pi f_0 n+...
4
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1answer
512 views

Why did my sine wave turn into a square wave when written to a WAV file in Octave?

I am trying to use Octave to generate a pure sine wave. The code for the same is as follows: x = 10.*sin(2*pi*(300/16000)*(0:1:400)); The sampling rate is ...
4
votes
2answers
297 views

How can I measure the harmonicity of an audio spectrum?

I would like to estimate the "harmonicity" (harmonics-to-noise ratio ?) of an audio signal from its spectrum. What kind of algorithm can I use?
4
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2answers
3k views

Estimating the time offset (delay) between two audio signals in real-time

I'm looking for an algorithm which can accurately estimate the time-offset/delay between two audio signals in real-time. One audio signal is a loudspeaker signal and the other signal is the microphone ...
4
votes
2answers
381 views

Sum of sines sound wave using microcontroller

I need to produce a signal which contains only several frequencies, for example, 17KHz, 17.2 kHz,17.6 kHz. The components I'm using are microcontroller (PIC16F1783 by microchip) which works at 80 kHz ...
4
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2answers
2k views

Extracting audio track to determine the song beat

I'm completely new to this field. Perhaps I'm calling it by a wrong name, but I want to analyze an audio track and retrieve a timestamp to beat mapping. Is there an open source 3rd party library ...
4
votes
1answer
478 views

RMS values above 0 dbFS?

I wonder why several RMS meters I'm using for music production show RMS values above 0. For example, when using an input signal with a constant DC of 1 (0 dbFS), I get these results, although ...
4
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2answers
1k views

What's the impact of aliasing in the time domain?

I've been studying digital audio and come across something I can't understand. There appears to be something like a consensus (among those capable of understanding such things) that the impact of ...
4
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3answers
2k views

Are there libraries for extraction of sound wave features?

I am looking help for my project for which I need C++ (or any other language) libraries useful for extraction of sound wave features like frequency, loudness, pitch and orientation. Using these ...
4
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1answer
1k views

Smoothed Square Wave

new here and probably pretty unexperienced compared to the rest of you. This should be simple enough but just wanted some clarification. I'm trying to model an analog square wave oscillator with C. ...
4
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2answers
1k views

Low Pass Filter Artifacts

I have created a Low Pass filter (I am trying to understand the topic of DSP, and it's hard.. But moving along with samples, books and a nice post on Stack Overflow). I am saving the 'output' to a ...
4
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1answer
1k views

Compensating Loudspeaker frequency response in an audio signal

I have been working on a project in which I was required to work on the audio signals recorded from the loudspeaker kept in front of a filter. So, to simply explain it: $$\boxed{\rm LoudSpeaker} \...
4
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2answers
186 views

Processing signals with phantom fundamentals

Is there a formula to calculate the frequency of a phantom fundamental based on the harmonics, and would this formula be applied directly to the energy bins of an FFT?
4
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3answers
317 views

Harmonic errors in acoustic instruments

If a note has a fundamental frequency of 100Hz, then its second harmonic is located at 200Hz and its tenth harmonic is located at 1000Hz. But for acoustic instruments, the harmonics will not occur at ...
4
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2answers
895 views

differences in fundamental frequency/pitch between music and speech

I am developping a program, that extracts very expressive frames from a video (meaning: frames, which could summarize a video) for a university seminar. One of my ideas was, to analyze the audio data (...
4
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2answers
277 views

Estimating effect of filter on headroom

I am working with beamforming algorithms for speaker arrays. One of the problems we face is that we need to estimate the maximum amplitude of our test signals after we applied FIR/IIR filters to them. ...