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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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 ...
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2answers
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% ...
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1answer
245 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 ...
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0answers
623 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 ...
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0answers
1k 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 ...
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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 ...
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1answer
773 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 ...
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0answers
31 views

Efficient Convolution Without I/O Delay (Gardner 1995) implementation

I am trying to perform real-time convolution on an audio stream using a fairly large FIR for convolutional reverb. I found a great paper explaining how to do just that: Gardner, 1995. But, I'm new to ...
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1answer
98 views

Real-time mixing of two audio clips while only modifying the data of one

Problem I want to mix two audio sources in real-time: game audio and microphone audio. The audio is provided at 44 100 Hz in the form of vectors of 1024 numbers between $-1$ and $1$ (PCM data). The ...
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0answers
49 views

Beamformer implementation methods

I'm currently reading articles about the different type of beamformers. This spatial filtering is gonna be used for acoustic purposes, to focus the beam on the person we need to hear talking in a ...
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0answers
52 views

Audio sharpening

I'm working on a project where the purpose is to enhance FFT peaks in a signal (musical instruments), by other words, reduce non-harmonic components and enhance the identified FFT peaks. So far I'm ...
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0answers
183 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 ...
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0answers
418 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 ...
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0answers
9k 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 ...
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0answers
372 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 ...
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1answer
814 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++...
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33 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. ...
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3answers
167 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 ...
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0answers
104 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 ...
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0answers
195 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
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1answer
95 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 ...
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1answer
333 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.
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1answer
161 views

How can you 'perfectly' reconstruct phase for transients in a modified audio absolute spectrogram?

I am currently working on the absolute spectrogram of an input audio signal. I am using non-negative matrix factorization to separate out the rhythmic components (bass drum, snare drum and hi-hat) of ...
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0answers
125 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 ...
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0answers
149 views

Gain of resonating filters

I'm trying to make a music instrument based on a resonator with a high Q, such as Ableton's Corpus and Ableton's Resonator (I think the former is based on a simple 2nd order resonant filter while the ...
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0answers
139 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 ...
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1answer
341 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 ...
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1answer
799 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 ...
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0answers
107 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,...
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0answers
188 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 ...
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0answers
119 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 ...
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0answers
248 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 ...
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0answers
252 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 ...
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0answers
195 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 ...
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0answers
510 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 ...
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0answers
112 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 ...
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1answer
876 views

Adaptive volume normalization on a stream of PCM data

I am writing a program that receives a continual stream of PCM data from various sources. Some sources have different volumes than others. The PCM data is always a live continuous stream of 16 not ...
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0answers
15 views

Order of operations when creating a spectrogram

Signals noob here! I'm dipping my toes into your world by trying to recreate the Shazam algorithm, and I have a quick practical question: When creating a spectrogram, is it better to take in a whole ...
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0answers
33 views

The size of an FIR filter for PDM-PCM conversion

I have been trying to understand how the conversion from PDM to PCM happens so that I could interface with a microphone. I have been doing some research online and I found that the most common way to ...
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0answers
69 views

Problem with natural logarithmic function in fixed point notation

I am facing a problem with natural logarithmic function in fixed-point notation. Let's say $$x = 0.54, \qquad\text{then}\qquad \ln(x) = -0.616186, \qquad\text{a negative number} \tag{1}$$ Then in ...
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0answers
20 views

Why are state-space filters good for time-varying systems?

"The state-space filter representation is especially powerful for time-varying linear systems." Why is this the case? What is the explanation behind the claim?
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59 views

Real-time audio processing FIR vs IIR

My project consists in filtering an audio signal from the microphone, in order to remove noise, and find the time distance between two peeks with the highest precision possible. I understood that an ...
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0answers
28 views

“Wandering” Whine in Converted VHS tapes

I am attempting to clean up the audio from several VHS tapes that were dubbed to DVDs and then ripped. My tool of choice has been audacity, but I am open to other options. I have two questions: In ...
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1answer
35 views

Digital Audio Mixing Fundamental. What will be the data rate of mixed audio?

I have voice data at 128 kbps coming from three different sources. I'm looking for mixing all three voices together and sending it back to speakers. What methodologies are there for mixing digital ...
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0answers
25 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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0answers
17 views

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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0answers
18 views

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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42 views

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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0answers
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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0answers
18 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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