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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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2answers
353 views

Relation between samplingrate and frequency

I am working on Fourier Transformation, and applying this for recognizing an audioclip. I have a 9 second long audio clip of a guitar strumming an A-Minor. The audioclip has a sampling rate of 44100 ...
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5answers
2k views

Pitch Detection: avoiding frequency doubling / halving

I am working on a tuning app. I have so far tried 3 different libraries with a number of different algorithms. However, when I compare to other tuner apps, I seem to be getting frequency doubling on ...
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What audio effects, filters, distortions, etc. create a “vinyl” effect?

I want to make any song sound like it's being played on a gramophone record in the 1920s. What specific filters would I apply to make this happen? I'm looking for technical details, not some magical ...
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657 views

Generating inaudible sound waves using MATLAB

I want to generate inaudible sound waves (above 18kHz) with 48kHz sampling rate using MATLAB. Generating a sound wave with 18kHz frequency makes it inaudible. I've inserted the data in the 18kHz ...
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384 views

Sound absorption coefficients for frequencies higher than 8kHz and lower than 80Hz?

I am building a matlab simulation that applies room compensation to a 3D audio rendering system, and I'm searching for the frequency dependent sound absorption coefficients of various materials in ...
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1answer
89 views

How to do continuous signal processing (i.e without windowing)?

I'm working in speech recognition research, and I wonder if there is a way to analyse sound just like how ears do (i.e. without windowing), so for example to take some feature continuously with an ...
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4answers
429 views

Why every musical instrument sounds different than any other while listening?

This is soft question.I am not expert in music. We know that English language has 26 letters (which are again grouped into vowels and consonants) , while language of music is same all over the world ...
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1answer
264 views

How to add more cowbell?

Remember that Christopher Walken skit on Saturday Night live where he told the band “it needs more cowbell” https://en.wikipedia.org/wiki/More_cowbell ? This site automated it http://www.morecowbell....
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1answer
1k views

How to make pop-free, crisp two-beep ringtone?

I hope this ultra-noob question is not entirely out of place here. I'm trying to make a super simple ringtone: 0.5s silence + 1s "pure tone" + 0.5s silence + 1s "pure tone" + 0.5s silence. By "pure ...
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2answers
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Self studying, getting a quality spectrogram

So I've been self studying signal processing, and been trying to use Python to get quality spectrograms, quality spectrogram like the one that would be produced with audacity etc. The steps I have ...
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1answer
803 views

Lip sync over IP

What are the most used techniques for lip sync (audio/video sync) for conferencing over IP (and related issues with playout buffering)? I could find a lot of paper about audio intrastream ...
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4answers
119 views

Can you recommend an efficient encoding algorithm for sound with long periods of silence?

As a side personal project, I'm coding an Android application that I plan to use to record the sounds of our first baby. I'm kind of new to the world of Android development and sound processing, so ...
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1k views

Signal Processing/FFT gives very high magnitudes for low frequencies

The script essentially pulls PCM data from the sound card and stores them in a buffer. Then I apply fft and graph it. I am looking through this blog post for some examples and ideas on how to ...
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2answers
41k views

Removing noise from audio using Fourier transform in Matlab

I'm trying to remove noise from an audio file. This audio file contains speech as well as constant pink noise. I know that I have to use the Fourier transform to convert to the frequency domain then ...
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2answers
128 views

What is the theoretic difference in the ways to compute the spectrogram?

I have noticed that the spectrogram for a audio signal is often calculated in different ways when the result from the DFT is transformed into real numbers; some tend to compute $|k|$ or $|k|^2$ while ...
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2answers
327 views

Linear predictive model convolution

Accidentally asked this question in the general area and was told to ask here, so... I've been trying to develop a lightweight, relatively-fast-to-decode sound compression format for use in my gaming ...
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1answer
277 views

Stereo sound separation into three sources

Whenever we listen stereo songs, one part of the audio like voice hear at the center of the head, and other parts are nearby left ear and right ear. If I represent center part by C, and other parts by ...
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1answer
891 views

Common Filter Types for Audio Applications

I'm beginning to write a basic "virtual analog" synthesizer and want to implement a lowpass with variable cutoff frequency and resonance. Does anyone have insight into filter types commonly used to ...
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2answers
2k views

Upsampling PCM audio: from 6 kHz to 8kHz

What would a straight-forward way to convert an audio payload (PCM 16bit) from 6 kHz to 8kHz? I understand that this is an interpolation problem. I fear that a linear interpolation would introduce ...
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2answers
2k views

Audio signal dither and noise shaping

I'm trying to get a handle on the importance of the error feedback term in noise shaping operation in typical audio dither algorithms. I'm thinking in terms of four signals. The original signal, the ...
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2answers
3k views

Hilbert transform filter for audio applications: Using IIR half-band parallel all pass structure

Does anyone have any experience designing wideband IIR Hilbert transform filters with audio applications in mind? I am using the filter for single side band modulation audio effects discussed in this ...
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1answer
98 views

Definition of DSP Terms

can someone define/explain briefly (or with some detail-> up to you) the following terms: Block Processing? and what are the differences between it and Sample Processing? (removed) What is a 3D ...
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1answer
7k views

MFCC feature vector from wav file

I am trying to implement a spoken language identifier from audio files, using Neural Network. I need to generate one feature vector for each audio file. From what I have read the best features (for ...
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4answers
2k views

How to calculate an azimuth of a sound source in stereo signal

Let say we have a stereo recording of only one source. What is a correct algorithm to calculate its azimuth using inter channel time difference and inter channel level difference?
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2answers
200 views

“Flanging” in frequency domain?

Flanging is defined as a mix of two identical signals where one signal is delayed in time by a small and gradually changing period, around 10-20 milliseconds. Since delay in the time domain is ...
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1answer
1k views

Emulate tape delay

I'm trying to emulate a tape delay in a VST plugin, but I'm not sure how to handle the delay changing? I want it to sound like a tape is actually fast forwarding/rewinding. Most digital delays I've ...
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2answers
1k views

How to identify useful elements of audio recording and ignore lulls?

I assume this is a relatively simple DSP task, but I'm having trouble finding any information on how to approach my problem. Hopefully someone with more experience and brain power can help me out: I ...
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1answer
3k views

Remove a frequency and all its multiple (harmonics)

In a soundfile, I'd like to remove a frequency and all its harmonics. What is the best solution in your opinion ? Do a loop with 20 notch filters, one after another. // f is the fundamental ...
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2answers
2k views

what does musical noise sound like and is there a sample clip for it?

Musical residual noise is created when a noisy signal passes through a noise reduction algorithm. For instance, please see the paper: Esch, T; Vary, P: "Efficient Musical Noise Suppression for Speech ...
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1answer
1k views

pitch vs playback rate

Let's say I have an audio sample, whose pitch is known to be C4 (don't wanna go in detail here, let's say I just sampled a tuned piano playing the C4 key). The rules of the game are: I can only ...
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2answers
7k views

algorithm for feature extraction from audio signals

I am trying to build a model for speaker identification, and I understand that the first step is to extract the features from the audio signals that are in my database. I have done quite a bit of ...
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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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2answers
275 views

Which parameters should I take care of while converting .au to .wav for Genre Classification

I am learning to develop Music Genre Classification models. I have a dataset in which files are in ".au" format, which I need to convert into ".wav". I used SoX (Sound eXchange) to convert it. With a ...
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1answer
466 views

how to correct minBLEP sawtooth oscillator DC bias

In my implementation of minBLEP, overlapping minBLEPs result in a DC bias as the frequency increases. This is because each minBLEP correction (i.e. the minBLEP minus the aliased step) has a DC bias. ...
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1answer
160 views

stretch factor in SOLA time stretch methods

I notice that in the general Synchronous Overlap and Add (SOLA) algorithm, we need to obtain the best overlapping start index using, for example, cross-correlation. But won't that start index change ...
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1answer
442 views

5.1 Rear To 5.1 Side mixing matrix

I am searching for a matrix definition for converting a audio signal (5.1 with rear) to a audio signal (5.1 with side). Currently I am using the definitions from: https://msdn.microsoft.com/en-us/...
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1answer
520 views

FIR Filter Design - obtaining FIR coefficients from frequency response

I would like to obtain FIR filter (low-pass) coefficients from a given frequency response by using C++ to develop algorithms. I have read at http://iowahills.com/B2PolynomialFIRFilters.html, that, if ...
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1answer
152 views

Codec Output has “Wavy” Amplitude on Chirp Input

I am sending a chirp that sweeps form 20Hz to 500Hz to an audio codec, CS42L52, connected to an MCU, STM32F405, and I am receiving a "wavy" amplitude output. There is no processing on the MCU, I pass ...
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1answer
4k views

How to model Tape Saturation (Audio DSP)?

I'm looking for info that would help me to build a Tape Saturation process. I'm working with the WebAudio API. The API provides a number of DSP nodes that perform basic processes, such as a Convolver ...
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2answers
1k views

How does Songify work?

Does anyone know how does the "Songify" app algorithm work? There are 2 inputs: speech (or amateur singing), and underlying music. Somehow the speech is altered and mixed into the music. Which DSP ...
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1answer
835 views

Equal loudness curves: any good numerical approximation equations?

What might be, if any, some good numerical approximation functions to the ISO 226 family of human perceptual equal loudness curves? Is there a polynomial fit of reasonably low degree?
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2answers
868 views

change the frequency component of a spectrum in an audio file

I am working on an audio dynamic range compressor using Matlab. After calculating the FFT of the audio file, I use several bandpass filters to amplify specific bands separately. What I want to do then ...
3
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1answer
333 views

Real-time single speaker recognition

I'm developing an application for PC that records microphone data in real-time, and plays it back after applying some effects. I'd like to apply these effects only when a single known speaker is ...
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1answer
484 views

From a 44kHz audio signal, how could I extract 32 evenly spaced frequency bands?

Assume I have an array of floating points that contain the last 8192 samples played. How would I convert that array to 32 frequency bins that are spaced from 0Hz - Nqyst?
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1answer
429 views

Am I handling offline FFT correctly?

I need some help clarifying FFTs and what they represent. I have a buffer containing compressed audio. Due to limitations, I can't handle the full uncompressed audio but can decompress small segments ...
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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 ...
3
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1answer
229 views

How can I calculate the entropy of a signal that's not independent from itself?

I really don't know how to word this better, sorry if that title is confusing. Basically, I'm wanting to use the noise from an AM radio audio signal to generate random numbers. To do this effectively,...
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1answer
358 views

Detecting a low frequency cue in audio recording

I am hoping to find a linux program or library for a language like perl or python (open to ...
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3answers
401 views

What is the physical significance of the Fourier coefficients in the DFT of audio signals and how can they be best displayed in a spectrogram?

I am trying to display the spectrogram of an input audio signal (a newcomer to all things spectral here). The amplitudes of my audio signal can be assumed to be normalized between -1 and +1. Suppose ...
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1answer
181 views

Acoustical Design considerations for Active Noise Cancellation Project for Air vent

I am working on ''Active noise cancellation for air vent'' project. I am using NI myRIO1900, Audio Technica ATR 3350 omni directional microphones and Tevion Loudspeakers. I have done the programming ...