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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28 views

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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17 views

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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2answers
81 views

Subtract audio to another audio

suppose I have a sample of a snare and that snare is used in a song,¿ there is an algorithm or tool that detects the sound of the snare and subtracts it, that is to say that cancels its phase or ...
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108 views

Generating a colored waveform of an audio file?

I would like to ask on what approaches could one attempt to generate a colored waveform out of an audio file. Below are two examples of waveform coloring by commercial products along an audio sample ...
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0answers
95 views

Additive Synthesis - clean playback of a spectrum frame

First of all - I have only a limited of Fourier transforms and the mathematics involved, so please pardon any potentially bad terminology or huge gaps in conceptual knowledge. I'm experimenting with ...
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0answers
39 views

What is the difference in the algorithm for noise reduction in SoX and Audacity?

I've looked at this question, and I understand that SoX works way FFT of the noise, then subtracting that from the FFT audio, and then resynthesizing using FFT, but I don't quite understand how this ...
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0answers
193 views

Implementing real-time variable fractional delay line in Matlab

I am attempting to implement a fractional delay line in Matlab which operates in real time, uses an object-oriented approach, and whose length can be modulated each time a sample is processed. I have ...
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1answer
439 views

Change duration/tempo of a .Wav file without introducing pops or pitch shift?

I am attempting to develop a similar function as Audacity's Effects>Change Tempo effect. Or more specifically: I am in need of a way to create a function in my ...
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0answers
41 views

Help in identifying low speed image transmission protocol over RF

In the seventies I worked for Muirhead on press fax machines that used to transmit pictures over radio and landline as I recall two FM systems were used a carrier 800 Hz modulated +- 150 Hz and 1900Hz ...
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0answers
56 views

Finding the input from the impulse response and output

I have $y,h,x$ which are all vectors. From $y[n]=x[n]*h[n]$ which is basically how I got $y[n]$. I also know $h[n]$. I put this through a Fourier transform. Let's assume that the capitalized ...
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2answers
117 views

Multiple different length Windows for high resolution realtime FFT

I am looking at trying to achieve a 'realtime' (as quick as possible to acquire, process and present data) FFT application to analyse audio. I have setup my app to acquire a set of samples, apply a ...
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0answers
63 views

Noise Reduction: How can I filter out saw or other tool noises?

I recently came across old recording from the 1930's. Unfortunately not only is the quality low, it's a bunch of carpenters talking as they saw things and make all sorts of other noises. I've tried: ...
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94 views

FIR Audio Equalizer Crackling Noise

I've implemented a few FIR filters into a song equalizer, but there seems to be a crackling that I'm unsure how to fix. Heres a video for reference: https://www.youtube.com/watch?v=IlzWRYpv0rk. Noise ...
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3answers
121 views

Modeling the human perception of noise

Add together some sinusoids with irrational or "nearly" irrational frequency ratios, and you get something pseudoperiodic. If there aren't too many, you can hear that it has only pitched components, ...
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0answers
2k views

Cocktail Party Problem — audio source separation

I am trying to solve the Cocktail Party Problem. I am trying to separate these two mixed audio files: mixed 1 [WAV] mixed 2 [WAV] into 2 separate audio files that contain the 2 original sources, ...
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0answers
113 views

Subtracting audio signal emitted - trying to use spectral subtraction to localize moving objects

I am a Software Engineer without much signal processing background and currently spending and experimenting to get use to it. My scenario: Assume a speaker and a microphone array. A speaker emits an ...
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3answers
177 views

Resample audio signal with a low factors

Currently, we need to handle a piece of an audio signal. We want to resample it with different factor rate (p/q) starting with 1.0002 to 0.9998 with a decrement of 0.000005. But Matlab function cannot ...
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1answer
47 views

How to recognize for small video sequence with a large sequence?

I'm working on the idea of recognizing a small video sequence (a sequence of 18 seconds) within a long one (a sequence of 15 minutes), the first idea that come to mind stats that I have to ...
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0answers
124 views

low-frequency amplitude with phase modulation of low Hz carrier

I am writing an FM/PM synthesizer, using Java. For the most part it works pretty well. What I am curious about is an attempt to use a low frequency tone as a carrier. By low frequency, I mean in the ...
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207 views

i want to mute a particular frequency in real time audio

i want to mute a particular frequency range in real time audio for eg: in audio signal frequency above 3 kHz should be muted for specific interval of time.
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0answers
34 views

Is it possible to detect that sound source is playback of a recording?

I have been trying to find an article where it is described how real-time (or close to real-time) signal processing from a microphone is able to detect that an audio source is playback of a recording ...
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0answers
88 views

Processing digital sound in Audiokit to sound like the 1960's

I am looking to imitate 70's analog equipments sound. I was wondering is there a way to create tape saturation / distortion inside Audiokit? I want to add that and randomized tape crackles and ...
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275 views

What is the algorithmic latency of STFT processing?

I was reading this paper, and in section 3 I got a bit confused. It claims that the algorithmic latency is equal to synthesis frame length. If the overlap between two consecutive frames is 50%, the ...
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1answer
1k views

Are there any signal processing libraries that implement TD-PSOLA?

For such a popular and old algorithm, I've had surprisingly little luck finding implementations. I've got it working in MATLAB, but am curious if any other languages have libraries containing it (...
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129 views

ICA for blind source separation clarification question

I'm currently trying to implement FastICA for blind source separation from scratch. The code below does not generate W, the umixing matrix, correctly. When I matrix multiplied the outputs ...
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0answers
216 views

What is advantage of Logarithmic Sampling?

Logarithmic Fourier Transformation This paper starts off in the abstract saying that the given audio signal is sampled logarithmically. We also know that the human ear has a sort of a filter bank ...
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176 views

What is logarithmic mapping in a filter bank such as the one in the human ear?

"The spacing of the filters in a filter bank used to make an ordinary spectrogram is linear, whereas an auditory filter bank has a spacing that corresponds to the way in which sinusoidal frequency ...
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94 views

Fastest way to subtract known music tracks from live videogame audio stream

I want to isolate video game "action" sounds from the video game soundtrack. Given all ~20 songs, I'd like to filter them out from any live audio stream during gameplay. What is a fast and robust way ...
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825 views

How does the size of a spectrogram relate to window length and frame rate?

I'm trying to understand the methodology from section 4 in this report about audio classification but I have some trouble understanding how they pre-process the data (dsp is not my area). $30$ ...
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0answers
493 views

removing noise from audio

I have an audio file, in the range of 4 kHz to 120 kHz. I want to remove the lines that are not harmonic and are recognized as environment noise. the energy of the signal in this interval for each ...
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0answers
331 views

What FPGA development tools recommended for building audio music synthesizers?

I want to experiment with FPGAs to build audio synthesizers. They would end up being manufactured in quantities of less than 100. I have very little knowledge of FPGAs, but I thought LabVIEW looks ...
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0answers
171 views

difference between logged power spectrum and power spectrum?

What is the difference between a logged power spectrum and power spectrum... Well the log part. So can a normal power spectrum be converted to a logged power spectrum just by taking the log of the ...
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0answers
103 views

Can MFCC consist any zero values?

After extracting audio features using some library, I just realized that sometimes it (MFCC) might or might not have zero values. At what time/event does MFCC could have zero values? Because it's (...
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86 views

Determining the probability of two signals, with added independent noise, being the same?

I have the first 100ms of two signals, each with a noise independent to that signal. I want to compare the two signals and determine the probability that these two signals are (minus noise) the same, ...
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1answer
481 views

Bandpass filtering an audio signal with MATLAB's $\tt kaiserord$: unknown new frequency component

I am trying to filter an audio signal with kaiserord in MATLAB, but I'm having a problem. Here is my code: ...
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152 views

Feature recognition within a single song

I want to write an algorithm in C++ which is capable of identifying specific features within a single song, e. g. the sound of a drum which is played 100 times during 5 min. State of the project So ...
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0answers
818 views

Detecting delay and sample drift in audio files (wav, mp3, etc.)

In my application, I have an audio recording with background music. I know which song is playing in the background and have a copy of the original song (wav, mp3, etc.). The goal is to align the ...
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250 views

I have 24-bit data in two's complement 32-bit integers. I need Q31 data

I have been trying to learn about Q31 format (as well as Q15), but I am having trouble understanding how it works. So my CODEC is giving me 24-bit two's complement data. I have converted this data ...
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193 views

Two way Audio crossover MATLAB code using Bessel

Could MATLAB's besself function be used in designing an Audio crossover (Tweeter and Woofer) with highpass and lowpass filters? Most likely the same with this ...
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370 views

Audio - TDOA and $\tt xcorr$, find the 'best' peak

I try to locate a mobile device using multilateration. The mobile device acts as the receiver and in the room I have three speakers. To calculate the position I need the time difference of arrival (...
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0answers
1k views

How to estimate THD?

I have a signal source that I can digitise and read. It produces mainly a 50Hz sine wave, but there is distortion as well. I'd like to estimate the total harmonic distortion for this signal. I've ...
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588 views

How do we detect presence of background noise in audio?

The problem I am facing to solve is detect background hissing or humming sound in any audio stream. The audio stream can be of any type ranging from movies to talk shows. The background noise is ...
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0answers
736 views

Can I use fpcalc to prove two audio files produce the same sound?

I'm using fingerprinting techniques to detect duplicate audio files on my music library. The first and easy step is to compare the file-size and if they match I use a md5 sum to compare them. The ...
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0answers
497 views

Adding two audio files at a predefined SNR level

I have two speech audio wav files. file1 is primary speech, file2 is supposed to be interference. I want to add them so that ...
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0answers
46 views

Is there a way to go from a set of (F/D)FT values to FIR coefficients?

I'm not yet well educated on the DSP subject, but I've initiated a project where I will do some audio filtering. My intuition tells me that there is a link between the coefficients of a FIR filter and ...
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0answers
559 views

How do I decode(demodulation) and get original data from audio sample data

Here I'm talking about the less-than-ideal situations, in other words, the audio sample data may have burrs or error bits in it. Encode and decode theory: I am using FSK to encode my original binary ...
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0answers
154 views

Averaging different frequencies

How does one average a group of frequencies? Frequencies taken for example using FFT. That is, given n frequencies over m time intervals (n $\leq$ m), how should one "interpolate" between those ...
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550 views

Difference between note based (MIDI) and sampled (FLAC, MP3, etc.) audio formats

I (somewhat) understand sampling and sampled formats, and their relation to the original (analog) signal, but do not quite relate it to note based (MIDI) formats. What actually is a note based audio ...
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131 views

Multirate filtering attenuates frequencies at overlap

I am trying to build a filter with different frequency bands in Matlab. I would like the filter to be as invisible as possible, but after the filtering, a great amount of frequencies are being ...
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16k views

Noise reduction on wave file

I have created a simple plot of waveforms using matplotlib of 2 wave files on python. Here is the code: ...