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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26 views

Filter signal1 from evolving frequency response of signal2

I'm a composer (of music) starting out with signal processing. I'm also reasonably proficient in C at this point. I'm not even sure if what I'm trying to do is possible, because I can't yet find an ...
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2answers
32 views

Decibels conversion into amplitude ratio

I want to convert db into amplitude ratio. I did some research but I only found online convertor, and I don't understand what's on Wikipedia very well... Here is a little of backstory : I am a ...
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1answer
66 views

What is the maximum frequency that can be captured by a given sampling rate?

Let's say that we want to digitally sample an analog signal and have a sampling frequency of 8000 samples/second. Usually the maximum frequency that can be captured is the half of the sampling ...
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2answers
36 views

Stitching together audio buffers with potential discontinuity at the boundary

I am writing a sound library for my c++ hobby game engine. The sound library mixes music and sound effects into a single buffer that is sent to the audio hardware. While mixing, any number of source ...
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26 views

short time energy matlab code for audio feature extraction

Is there code in matlab for short time energy feature extraction in matlab?? I want to extract audio feature (Short time energy) using matlab code
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27 views

Audio output distorted with any FFT Windowing other than rectangular

For an experiment i have a simple application that takes a quadrature signal from a radio and then does the following: Complex FFT -> Inverse FFT -> Cartesian to Polar -> audio out If I tune the radio ...
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25 views

What is the preferred method of aggregating audio FFT data given variable-size frequency bins?

I am coding a program which uses a FFT to process audio data. I need to aggregate the FFT output into three bins ("bass", "mid", and "treb") determined by two arbitrary frequency values. The end-goal ...
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1answer
50 views

efficient triangle to sine morphing?

I'm coding a synthesis external for MaxMSP in C. It's based on Peter Blasser's bounds and bounce concept ( http://petermopar.blogspot.co.uk/2014_04_01_archive.html )- basically triangle waves with ...
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16 views

Time frequency representation of DWT coefficients in Matlab

I am using discrete wavelet transform to analyze an audio file. I can calculate the dwt coefficients and the inverse dwt works fine. What I want to do now is to visualize how different coefficients ...
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2answers
42 views

Melody/Vocal extraction

I'm new to the whole DSP area, but I've built a program which plays music and does FFT to produce some cool visuals for the music in realtime. The FFT algorithm I chose was from a C# library called ...
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1answer
34 views

Average spectrum representation - SonicVisualiser

I try to have a signal (speech) represented in time and frequency domain (waveform, spectrum and spectrogram). I use Sonic Visualiser and for spectrum representation it shows an analysis at a given ...
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35 views

Relationship between values of frequency magnitude and values of time amplitude in an audio signal

I have an audio signal in wave format, with its time amplitude values located between -1 and 1. When I compute its fft (in respect to the whole signal) and then extract its absolute value to plot the ...
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1answer
29 views

MFCC from DFT (FFT)

I want to write an speaker recognition application (based on some password recorded few times for the authorized person). I've decided to do it with C++ and BASS library (http://www.un4seen.com/). I ...
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17 views

Formatting issue WAV - MP3?

I have two "sources" of audio: record WAV audio files (44.1kHz, 16bit, stereo) with Linux arecord and convert them with LameMP3Encoder to MP3 files (44.1kHz, 320kbps, 32bit**, CBR)*** record MP3 ...
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0answers
26 views

How to implement vector quantization after finding out the MFCC?

I am writing a code in MATLAB for speaker identification/recognition and I am planning to use MFCC coefficients and then implement the Vector quantization technique. I read upon VQ and know how it ...
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1answer
29 views

What interesting audio can be synthesized with ifft

I'm working on a presentation on audio synthesis for some fellow software engineers. I want to include some "fourier synthesis" creating time domain samples from frequency domain samples using ifft. ...
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2answers
74 views

Why does slowing down audio introduce aliasing?

I have an 8khz mono buffer of samples, and I am trying to manipulate the playback speed of it by modifying the number of samples in it. When slowing down (a speedScale value of 0.5 for example) then ...
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0answers
36 views

Algorithm used by mironsets

I was curious to know which algorithm(stepwise) is used by the mironsets function of the MIRToolbox library to find the onsets in the audio signal. I am talking about the default functionality when we ...
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20 views

Audio pattern detection

I could use some help on audio analysis. What I'm trying to do is find specific patterns in audio files. The thing is is that I don't have an exact audio reference, I'm just trying to find similar ...
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0answers
11 views

What is the best way to detect audio drops (audio level drops irregularly) in a video?

While recording a video, due to some problem in microphone (like low battery ) audio level go down inconsistently (voice____Voice__), I am trying to figure out an solution for this, is there any way ...
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1answer
35 views

Diagonal vs non diagonal processing for audio denoising

I am going through a technique for audio denoising by block thresholding. Here is the paper: http://www.di.ens.fr/~mallat/papiers/AudioBlock.pdf According to the definition in the paper for ...
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2answers
64 views

Distorted output of delay VST effect in C

I'm getting some odd results with some code I'm working on. Help me with the implementation of this delay effect? The code takes an input, adds delayed input to it and then passes it to the output ...
2
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1answer
45 views

method to detect frequency sweep in audio

I have search a bit to try to find a similar questions, but I have not been able to find any leads. There are techniques to programmatically detect steady tones in audio, such as the Goertzel ...
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2answers
34 views

Determining the 'amount' of speaking in a video

I'm working on a project to transcribe lecture videos. We are currently just using humans to do the transcriptions as we believe it is easier to transcribe than editing ASR, especially for technical ...
2
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1answer
87 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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3answers
78 views

How to add pitch-shifted version of signal to itself?

I want to write an algorithm to add pitch-shifted version of audio signal to itself (like Electro-Harmonix POG does). And I need to do it in realtime. How to implement it? My guess is: break signal ...
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1answer
32 views

Do I need an overlapping in STFT if I want to play the signal back?

I want to write an algorithm to implement a pitch-shifter, like in this article: http://www.guitarpitchshifter.com/matlab.html. And I need to do it in realtime, not with a signal already defined ...
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3answers
133 views

Why is telephone audio sampled at 8 kHz?

When did we decide to sample telephone at $8$ kHz? Has this always been the case? Why did we do that? Is it because higher bit rates can't be transferred as quick? And do these reasons still count? ...
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2answers
42 views

Playing back an audio file at an arbitrary rate with upsampling and downsampling

Background: My overall goal is to create an audio file player which can playback an audio file at an arbitrary rate. The rate can change continuously as the file plays. This means upsampling to ...
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42 views

Coding a Variable Delay with LFO (MATLAB/C)

I am trying to code a Variable Delay (chorus/flanger) effect in MATLAB using a paradigm that would be friendly for porting to a lower level language like C. What I currently have is a working ...
2
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2answers
141 views

Recognition of markes in long audio recording

I am about to do a lot of a recordings when performing tests. The thing is that there are some parts I would like to extract easily from my recording. Is there a way to play some special sequence of ...
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46 views

To what extent is it possible to to determine whether a phone call is local or long distance by analyzing the audio?

We have a recording of a phone call which contains some echo. We would like to determine the distance between the 2 people who spoke. The issue is explained here: ...
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1answer
70 views

Why are MFCCs of two equivalent signals completely different?

I have an audio file, of which I calculate 16 (actually 15 because I omit the first one) MFCCs in R. When I stream this file via VLC-Player and Icecast2-Server, receive it in Java (with the Player of ...
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2answers
69 views

Resample an array in real-time (application: pitch-bend)

I'm working on a sampler software (that plays .WAV files, when notes are played on a MIDI keyboard). In order to implement pitch-bend feature (i.e. you have a "pitch bend wheel" on the synth and you ...
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56 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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1answer
45 views

Feature extraction for sound recognition and classification

I am building an application that would "listen" to the microphone input, analyse it, and compare the analysis to a pre-analysed and pre-classified sound bank (small - maximum 20 sounds). It will then ...
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41 views

Detecting an echo using the autocorrelation of the cepstrum

I'm trying to implement an echo hiding steganography technique. I need a way to detect an echo in a audio segment. The documents I've been looking mention that I need to do an "autocorrelation of the ...
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29 views

detecting change of speaker in audio file

how can we detect whether the active speaker has changed ( either stopped or someone else started talking or many people started talking together ). I dont need to know who it was ( speaker ...
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2answers
66 views

can MFCC coefficients alone be used to do voice detection?

Aim of my project is to differentiate between cases of silence, single active speaker and multiple ( simultaneous ) active speakers using as basic ( but effective ) logic as possible. Is it possible ...
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1answer
50 views

Steganography - What techniques exist to hide an image in an audio file? [closed]

Can anyone mention me techniques to hide an image in an audio file ? Thanks
2
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1answer
90 views

Algorithm to pan audio

I'm looking for a way to pan the PCM audio I have. At the moment the audio is in a stereo format and I would like to pan it to the desired side (or more/less to a side). I've recently figured out how ...
0
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1answer
24 views

Objective, repeatable sound quality measurements and metrics

I am currently doing an electronics design for a basic 2.5W speaker amplifier for an embedded system based on COTS parts (e.g. codec and amp), similar to what one would find in a tablet. Aside from a ...
3
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2answers
99 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 ...
2
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2answers
110 views

How to know if a audio file is real lossless using its spectrogram?

I need to know if a certain audio file is real lossless and not a transcode (i.e. lossy converted in lossless, but lossy at the end). I was told there was a way to do this by creating a spectrogram of ...
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1answer
41 views

When do we normalize data in audio processing?

Lets assume we have a simple audio processing flow with preprocessing, feature-extraction and classification. Where and why do we normalize the data? I know the purpose of normalizing the data in ...
2
votes
1answer
66 views

Ear Test using MATLAB

I'm trying to test my ear frequency range by using MATLAB. The thing I could not understand is that I can hear the frequencies above 20 kHz. My sampling frequency is enough to satisfy the Nyquist ...
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1answer
64 views

MATLAB flanger…zipper noise, crackling, help with interpolation values?

i'm writing a batch script in MATLAB to flange. i'm getting some periodic noise, and i'm having trouble tracking down the source. it's definitely in the algorithm, because soundsc() will scale ...
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0answers
32 views

Common audio signal extraction from multiple similar audio files

Say you have two mono audio tracks recorded from a pair of matching omnidirectional microphones (call them A and B) that were, say, 20cm apart. The audio tracks were recorded at precisely the same ...
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3answers
50 views

Features for audio levels imbalance in films between speeh/music/effects

What would be the key features of an audio track or set of tracks (like 5.1) for identifying film audio tracks that have a clear unbalance between speech levels and sound effects levels (apart from ...
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6 views

How do we find the SNRseg if I have a block with 2D in time domain

I always get the answer -.0017 and it suppose to have some negative number around 30. I get the average for the plane voice then the average for the difference between plain and the encrypted voice ...