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.

214 questions with no upvoted or accepted answers
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14
votes
2answers
357 views

When does the laugh track kick in?

One of my weekend projects has brought me into the deep waters of signal processing. As with all my code projects that require some heavy-duty math, I'm more than happy to tinker my way to a solution ...
4
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1answer
3k views

Demodulating FSK audio in Python

I am trying to write a Python script that can demodulate an FSK modulated audio file and return the data encoded in the audio. The data being transmitted is GPS NMEA strings which are embedded as the ...
4
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0answers
3k views

Verifying MFCC Final Result

I have created a Octave code for MFCC calculation for speech recognition, which takes a wav file as input and then gives the MFCC as the output. But the final results are a bit confusing, so I am ...
4
votes
1answer
591 views

$1/n$ octave complex smoothing

An excellent answer to this post explains how to do $1/n$ octave energy smoothing and mentions complex smoothing can be done as well, but it's tricky business because of phase wrapping. How is phase ...
4
votes
1answer
1k views

STFT overlap add method - What am i doing wrong?

i am trying to implement a short time fourier transform of an audio signal to do some filtering (subtract another stft signal using spectral subtraction). I am using a hamming window function with 50% ...
4
votes
1answer
177 views

Teager's Energy Operator

Does anyone know how to find information on the derivation (rather than the definition and application) of Teager's Energy Operator, both for the continuous and sampled case. I have Kaiser's early ...
4
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0answers
573 views

What is EAQUAL ? How does it relate to PEAQ?

This wiki post on wiki.hydrogenaudio.org says that it is very similar to PEAQ and also a part of ITU-R BS.1387 recommendation ? How reliable it is for determining audio quality. Are there any other ...
3
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0answers
404 views

Why are the frequencies associated with the STFT bins not representative of the energy?

I'm experimenting with a 30 Hz sine wave and I want to extract the frequencies of this, and have implemented the algorithm to do so. This gives a correct reading, well, gives 29 Hz. I'm now wanting ...
3
votes
0answers
8k views

numpy.correlate and autocorrelation; audio signal

I want to calculate delay between an input and an output audio signal of my audio processing system. The input and output signals are available as signed 16 bit integers. To try out, I tried the ...
3
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0answers
330 views

choosing windowing function for audio signal processing

Currently I am working on chord recognition, and get confused on several aspect. One of them is choosing windowing for reduce the spectral leakage. from what I read, hanning/hamming function is a ...
3
votes
1answer
737 views

identification of sounds within audio signals

I am looking for methods to identify sounds within audio files. I was hoping to find some ready-made user-friendly software to do this but haven't found it yet. I am familiar with Matlab and some C++...
2
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1answer
61 views

How to interpret these MP3 spectrums?

Recently, I found I had this song duplicated in my music library, but when listening to them I noticed one version was of noticeably higher quality than the other despite both claiming to have a ...
2
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0answers
19 views

Is it possible to create a causal filter with a positively-sloped phase response?

I am currently prototyping a set of noise-cancelling headphones that utilize active EQ to modify the noise-cancelling signal, allowing the user to modify the frequencies that they hear or don't hear. ...
2
votes
3answers
85 views

How can I convert audio to a series of notes using pitch detection?

I have a good pitch detection system set up, and I would like to return a series of notes given an array of audio samples. My current approach is as follows: I have a moving window across the audio ...
2
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0answers
77 views

How do audiobook players manage to quickly speedup/slowdown playback speed without using more memory?

Many audiobook player apps on Android have a reading speed slider which lets us choose the reading speed e.g. 0.5x 0.75x 1.00x 1.25x 1.50x. This is time-stretching ...
2
votes
0answers
130 views

Frequency translation with FFT

I'm new to DSP and I would like to make an SDR receiver. For this I want to shift bins in the frequency domain in order to bring to 0Hz any portion of the base band spectrum (I know that there are ...
2
votes
1answer
84 views

Introductory resources to mathematical properties of “dynamically recalculable” filters in audio/musical equalizers?

Introductory resources to mathematical properties of "dynamically recalculable" filters in audio/musical equalizers? Particularly, I want to understand What mathematical features (e.g. monotonicity ...
2
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0answers
232 views

What are F0 counters in Speech signals? What is represented by F0?

I can't find any dedicated definition for Frequency F0 counters. Can anyone tell me what does F0 exactly mean, I am confused about that.
2
votes
0answers
121 views

Is it possible to “equalise” a signal by deconvolving the impulse response of the room in which it is to be played?

I am using a test sweep with a flat power spectrum and linear group delay (Optimized Aoshima's Time Stretched Pulse) to measure a room's frequency response. Having obtained the impulse response of the ...
2
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0answers
64 views

Finding specifications for digital signal processors (DSPs) for given audio application

Note: This topic has been moved to https://electronics.stackexchange.com/posts/345289 since it fits better, please post any answers/comments there. I'm finding difficulties choosing an adequate ...
2
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0answers
118 views

Speaker diarization tool for acoustic analysis

I'm researching certain acoustic prosodic features during live clinical interactions. Because of the setting, the recordings always have cross channel bleed, and in some cases everything is recorded ...
2
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0answers
129 views

Room Impulse Response Inverse Problem

This article here talks about a room impulse response generator which takes the source and destination coordinates, sound speed, sampling rate, room dimension and wall reflection coefficients as input ...
2
votes
1answer
919 views

What does Fast and Slow Time Weighting signify for real-time audio signal processing?

I have a streaming audio signal that I am recording in real time. To make the explanation easy, say the signal that I receive is calibrated and has a sampling frequency, $F_s = 44100\textrm{ Hz}$. I ...
2
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0answers
135 views

Voice Biometrics (aka Speaker Recognition) - Standard accuracy rate? State of the art tech?

I'm working on a voice biometrics project and am looking for some guidance on two key issues: First, does anyone know if a standard exists for voice biometric system accuracy? I have been unable to ...
2
votes
1answer
291 views

How to make sense of audio file's $\textrm{dBFS}$?

I'm trying to build a song visualizer using the FFT method. When I analyze the audio, I get the $\textrm{dBFS}$ value for each frequency range. From my understanding, $\textrm{dBFS}$ is in the range ...
2
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0answers
891 views

detect what is noise in audio signal

I'm trying to filter noise from an audio signal. The signal was and is going to be recorded in a city environment which always contains some white/pink noise. At this moment I'm using two options to ...
2
votes
1answer
592 views

Applying DTW on MFCC frames

I want to use MFCC (Mel-frequency cepstral coefficients) and DTW (dynamic time warping) algorithms to make a speech comparison script. I calculated MFCC and got 13 coefficients for each frame. How ...
2
votes
0answers
106 views

Can you apply an audio filter/effect to only part of an audio signal and not get audio glitches?

I'm entertaining the idea of making a music-creating application. I'm a total newb at signal processing. The question that concerns me most right now is, if I apply a filter/effect such as convolution,...
2
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0answers
174 views

Alternative to scipy maximum_filter to find peaks?

I'm using scipy maximum_filter together with binary_erosion to find peaks in my spectrum. I just wondering if there are any ...
2
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0answers
95 views

Low-rank approximation of the STFT matrix : good audio compression technique?

I recently heard about this interesting math tool: With a matrix $A$, compute the SVD ($ A = U \Sigma ^tV$ factorization, where $\Sigma$ is a diagonal matrix containing the "singular values") Keep ...
2
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0answers
237 views

1/f noise generation

Recently I was reading a paper about the pink noise generation. But I don't have any background of this field. I am confused with some part in the paper. First, the Voss's algorithm generated a ...
2
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0answers
239 views

MFCC help: Same sound, different coefficients?

Problem: I'm getting totally different MFCC coefficients for identical and near identical sounds like several finger snaps or several desk taps. What am I missing? I thought since the timbre doesn't ...
2
votes
0answers
180 views

Speech features that can be inverted?

I am trying to figure out speech features that can be inverted to recover the audio back with no loss or as less loss as possible. I am currently using the power spectrum which is completely ...
2
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0answers
501 views

Audio watermark embedding text attack resistant

I'm developing an audio watermarking algorithm which embeds text (binary form) in DWT-DCT coefficients. The goal of the algorithm is to hide song information (like artist, song, album,...). Everything ...
2
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0answers
111 views

Complex Cepstral Transform and Audio Watermarking

I am doing research in audio watermarking and have come across the paper that uses the complex transform to embed the watermark in order to improve the robustness and inaudibility of the embedded ...
1
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2answers
36 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 ...
1
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0answers
26 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 ...
1
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0answers
16 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 ...
1
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1answer
44 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 ...
1
vote
1answer
44 views

Distort voice to hide talker identity as much as possible

I need to develop an app which makes speech recognition. Because I need to be able to constantly improve the recognition capability, I would like to train the engine with real recordings and match the ...
1
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0answers
53 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 ...
1
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0answers
25 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 ...
1
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0answers
73 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 ...
1
vote
1answer
105 views

Change durration/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 funciton in my ...
1
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0answers
33 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 ...
1
vote
1answer
33 views

Similarity in signal plots but dissimilarity in sensed content

I have two audio signals that look incredibly similar when plotted. However, when I play them, they sound quite different. The only difference I can even see in their plots is that one signal looks ...
1
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0answers
44 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 ...
1
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2answers
82 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 ...
1
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0answers
45 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: ...
1
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0answers
57 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 ...