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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62 views

Generating band-limited noise with a flat spectrum between two given frequencies [on hold]

I am hoping to a) make a record attempt to the cassette with the highest spectral entropy and b) make it sound good. So to make sure I get BOTH A and B OK I hoped to generate noise between 100 and ...
2
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2answers
96 views

What is spectral entropy?

Can anyone explain what spectral entropy is? Does noise with a restricted bandwidth have the same spectral entropy as white noise?
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1answer
18 views

Applying a mathematical function to a 16-bit Python array

I am using scipy.io.wavfile to read a WAV file and playing it with pyaudio. Before the numpy array is written to a string and output through pyaudio, I would like to be able to apply some sort of ...
2
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0answers
39 views
+50

5.1 Rear To 5.1 Side mixing matrix

I am searching for a matrix definition for converting a audio signal (5.1 with rear) to a audio signal (5.1 with side). Currently I am using the definitions from: ...
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0answers
10 views

Frame-level autocorrelation coefficients with lags bigger than frame size

I am reading the paper "Automatic Generation of Social Tags for Music Recommendation" and struggling with understanding this: The features we use include [...] 176 autocorrelation coefficients ...
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1answer
19 views

Why does speeding up audio files with VLC Player not match with doing the same thing with RPG Maker or Audacity?

When I speed up 150% an audio file with VLC Player, why doesn't it match the higher-pitched version I get when I speed the same file 150% with RPG Maker VX Ace's Sound Test? When I speed up with VLC ...
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1answer
47 views

Controlling one signal with another one (Rocket engine effect - Subtractive synthesis of sound)

I programming some sound effects in Java and exporting them into .wav files. Currently, I am trying to program a rocket engine sound effect. I want to do it in the following way: The sound of a ...
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2answers
48 views

Noise removal from audio using FFT/Spectral editing

I have a noisy recording with some letters spoken out loud. I've noticed that the noise is easily recognizable when looking at a spectrogram of the audio. Is there anyway to set a power/frequency ...
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2answers
21 views

MIDI out from freeware sequencer to PD or Max/MSP

I'm looking for a way to send MIDI from a cheap or free MIDI sequencer to Pure Data or Max MSP so I can do realtime MIDI processing in PD/MAX. So I guess I have a couple questions: What is a good ...
2
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4answers
103 views

Why every musical instrument sounds different than any other while listening?

This is soft question.I am not expert in music. We know that English language has 26 letters (which are again grouped into vowels and consonants) , while language of music is same all over the world ...
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1answer
21 views

What are the units of amplitude of my ultrasonic recording equipment

I'm not familiar with the topic of sound. I have ultrasonic audio recordings sampled at 250kHz (16bit) in WAV format. The equipment used to record them is linked here: ...
1
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1answer
43 views

FIR Filter Design - obtaining FIR coefficients from frequency response

I would like to obtain FIR filter (low-pass) coefficients from a given frequency response by using C++ to develop algorithms. I have read at http://iowahills.com/B2PolynomialFIRFilters.html, that, if ...
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2answers
41 views

Authoritative DSP book/reference material on Psychoacoustics

I am an auto didactic Electrical Engineer currently pursuing a goal to design a custom application targeted at deeply embedded linux(think BBB or even Raspberrypi-2) type device. My DSP background is ...
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1answer
25 views

Invert soundsc function in Matlab

Does anyone know what is the inverse function of soundsc in matlab. I'm playing a sound with a my PC microphone using soundsc, ...
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0answers
29 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
66 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 ...
0
votes
1answer
71 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 ...
0
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2answers
44 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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2answers
70 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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0answers
30 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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0answers
29 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 ...
0
votes
1answer
52 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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0answers
32 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
48 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 ...
0
votes
1answer
43 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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0answers
39 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 ...
-1
votes
1answer
41 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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0answers
20 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
41 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 ...
0
votes
1answer
32 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
78 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 ...
0
votes
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 ...
0
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0answers
25 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
13 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 ...
0
votes
1answer
44 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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vote
2answers
70 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
votes
1answer
52 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
37 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
votes
1answer
94 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 ...
0
votes
3answers
92 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 ...
0
votes
1answer
34 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 ...
1
vote
3answers
156 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? ...
0
votes
2answers
50 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 ...
0
votes
1answer
65 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
votes
2answers
143 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 ...
0
votes
0answers
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: ...
0
votes
1answer
82 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 ...
0
votes
2answers
92 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 ...
0
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0answers
72 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 ...
-1
votes
1answer
53 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 ...