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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96 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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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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1answer
7k 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 ...
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632 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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79 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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81 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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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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89 views

How timbre shifting is done?

I've recently came across two programs - Morphvox, VCSdiamond that are able to preform pitch shift, but also timbre shift. As far as I know the timbre is nothing but the amplitude of the harmonics in ...
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138 views

Extracting audio segments based on detected onsets using libROSA

I have a couple of .wav sound files with very similar percussive signals of ~60ms duration. I can identify their onset times using libROSA's onset detection quite well. I would now like to extract the ...
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1answer
165 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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227 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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192 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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431 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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1answer
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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390 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
849 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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23 views

Standard audio format using DWT for compression

Is there any standard audio format that employs discrete wavelet transform to compress signals?
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1answer
72 views

How to decode a binary Signal in a wav file into binary - sent here from stackoverflow

I'm doing a puzzle that gets me to call a phone number which produces a load of beeping sounds. Putting this into audacity as a spectrogram produces this: Zoomed in it looks like this: This looks ...
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1answer
29 views

Will a 2-recording approach work to remove ambient sound to isolate insect activity noise?

I am recording insect noise - buzzing, rustling in flowers, munching, etc. with a lav mic on a long skinny stick. My new goal is to remove ambient sound like planes, distant cars, etc. Poor results (...
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35 views

Finding "Max Change" in a Audio Signal

I'm working on studying an endangered language and have the following problem: while I already have labeled exactly what the speakers I worked with have said at given timesteps on the word level and I ...
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60 views

Blow sound recognition without using machine learning

I am building an experience in Unity that will recognize when a user blows into the microphone. I have a working implementation that recognizes sounds above a certain loudness threshold, but I want to ...
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1answer
100 views

how to handle different durations of audio data?

I am new to signal preprocessing, I read about mel_spectrograms, MFCC's. Now I want to apply it and use the CNN model, But the data which I have for practice is having audio of different durations, ...
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2answers
176 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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52 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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289 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 ...
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1answer
422 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
202 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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128 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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1answer
2k 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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140 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
411 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
1k 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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110 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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193 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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142 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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265 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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1answer
1k 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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264 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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198 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
513 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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113 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
43 views

Why is this ring-buffer jumping?

Central problem (Abstract): A simple implementation of a two-buffer ring-buffer jumps after filling up completely. What am I missing? About: ADC to DAC feedthrough STM32L476RG NUCLEO as MCU Intended ...
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63 views

Why is voice clipping happening after Dynamic Range Compressor (MATLAB) when applied in real time on chunks of data?

I have been working on real-time audio signals on smartphones (iOS and Android) and my goal is to make voice amplitude stable in a certain range even if a person is talking closer to the mic of the ...
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1answer
33 views

Remove a full song from audio recording

I have some audio recordings at 16KHz, which only contain music (BenSound Adventure: https://www.youtube.com/watch?v=0H8JTsG1Jtk). Since I know the music and have a separate wav file, I was wondering ...
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1answer
49 views

Audio, estimate codec's compression ratio only by content

I am about music audios. These can be compressed by different formats with different degree. (I.e. I don't mean the dynamic compression). After being compressed by a large degree (say 96kbps mp3) the ...
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1answer
48 views

Time warping / frequency modulation natural examples?

"Time warping" defined as $$ x(t) \rightarrow x(t - \tau(t)) $$ where $|\tau'(t)| < 1$ ($=1$ is time stopping, $=0$ is time shift, $>1$ is time reversal). Where can this occur? A ...
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1answer
35 views

Differentiate betwee sibilant "sssh" voice sounds and instruments like hi-hat?

How would I differentiate betwee sibilant "sssh" voice sounds in a music track and a similar sounding instrument sounds like hi-hat or cymbals?
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1answer
54 views

Balance Fader in Digital Amplifiers

What is the purpose of Balance / Fader in a digital Audio amplifier? In a signal flow what is the most apt position to place balance and fader(for example after before a compressor unit or at the ...
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1answer
41 views

Why does my zoom audio, when there are nearby microphones, seem to usually take an open loop format?

The signal from the speaker on device A gets into the microphone from device B, then the amplifier and then the speaker from device B, back into device A. The result is a sort of runaway signal that ...
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50 views

Short Time Fourier Transform has different frequencies than Fourier Transform?

The reason we do the STFT is so that we can analyse for short segments of time how much of the components in the frequencies of the FT are present. However, it may be possible that completely ...

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