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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1answer
1k views

What is the best way to represent audio visually? (x-post from UX)

Original Question: https://ux.stackexchange.com/q/23040/16006 I've only taken some basic signal analysis courses, so I might be missing some things. Purely theoretical question: What methods exist ...
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1answer
300 views

How to characterize the regularity of rhythm?

Again, my snoring detector: When I look at the rhythm of snoring vs the rhythm of talking in terms of peaks/events (which would represent snores) per hour, there's not much difference -- they range ...
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2answers
1k views

Understanding values saved in WAVE file

The values that one can find in a wave file, e.g. 0.0036 0.0026 0.0174 0.0050 0.0026 0.0108 0.0154 -0.0114 -0.0006 0.0115 ...
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0answers
585 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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1answer
1k views

Determining how similar audio is to human speech

While looking for an answer to this problem, I found this board so decided to cross post this question of mine from Stack Overflow. I am searching for a method of determining the similarity between ...
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2answers
892 views

differences in fundamental frequency/pitch between music and speech

I am developping a program, that extracts very expressive frames from a video (meaning: frames, which could summarize a video) for a university seminar. One of my ideas was, to analyze the audio data (...
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4answers
14k views

Scipy - Audio Processing

I am looking for good tools for Audio signal processing. e.g Speech & music analysis, automatic language identification, etc. Does the Scipy library provide functions for audio processing? Does ...
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3answers
4k views

Output of KISS FFT interpretation

I'm receiving PCM data trough socket connection in packets containing 320 samples. Sample rate of sound is 8000 samples per second. I am doing with it something like this: ...
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2answers
2k views

Audio signal comparison for automatic singing evaluation

I am trying to compare audio waveform from input waveform (A) to preset waveform (B)from in the device. I try to lookup everywhere. Most likely someone give such a source code as follow : ...
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4answers
7k views

Help with equations for exponential ADSR envelope

With application code, I have implemented a linear ADSR envelope for shaping the amplitude of an oscillator's output. The parameters for attack, decay and release duration as well as sustain level ...
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2answers
1k views

What bandpass filter design will yield the shortest impulse response?

Designing a simple 2nd order IIR bandpass Buterworth filter with a centre frequency of 500 Hz and a 1 octave bandwidth gives me the following frequency response ... Now, if I take the impulse ...
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1answer
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Speech Compression - In LPC how does the linear predictive filter work on a general level?

Hi I'm taking a multimedia systems course and I'm preparing for my exam on tuesday. I'm trying to get my head around LPC compression on a general level, but I'm having trouble with what is going on ...
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1answer
518 views

What are the basic categories of operations performed on audio signals to create interesting sounds?

I don't know if this question will make sense as I'm very new to dsp. In my limited foray into audio dsp I've come across: Frequency Modulation Amplitude Modulation Additive Synthesis Subtractive ...
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3answers
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Is there any free software that can recognize and classify sounds?

i am making a smart home automation software and i would like to know if there is any software that can recognize noises such as speech ,playing music,phone ringing etc The exact software i am looking ...
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3answers
17k views

Alias frequency Formula

I'm taking a multimedia systems class in my MSc Computer Science, and I'm having some trouble understanding the formula for the alias frequency - this could stem from my misunderstanding of the alias ...
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4answers
25k views

Libraries for Voice Activity Detection (Not Speech Recognition)

As follow up to my previous question I was wondering if there are any speech detection libraries in existence. By speech detection I mean passing in an audio buffer and getting back an index of where ...
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5answers
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Identify Where Singing Starts in a Voice Only Recording

I have a little karaoke-style app where a user sings 4 lines of a song, with a one second gap between each line. There is no backing music, so it's voice only, hopefully making the problem easier to ...
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1answer
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Help with algorithm for modulating oscillator pitch using LFO

I am developing a software emulation of an analog synthesizer. I am trying to modulate the pitch of an oscillator using an LFO. For each sample fed to the computer's sound system, I am calculating ...
3
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1answer
322 views

Real-time single speaker recognition

I'm developing an application for PC that records microphone data in real-time, and plays it back after applying some effects. I'd like to apply these effects only when a single known speaker is ...
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2answers
5k views

Low pass filtering on short int (16 bit PCM) samples

I am writing software for processing audio given as 16-bit PCM samples. The first stage of the processing involves calculating the energy (or total variation) in a certain frequency range (above a ...
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4answers
18k views

Efficiently calculating autocorrelation using FFTs

I'm trying to calculate an autocorrelation on a platform where the only accelerated primitive I have available is the (I)FFT. I'm having a problem though. I prototyped it in MATLAB. I am, however, ...
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1answer
63 views

I trigger recording when power of current sample is X dB greater than last Y samples, what is the terminology here?

Doing time domain processing of incoming audio. I sample the power level every 0.25s and keeping a running average of intensity over 5 samples. When the power level of the current sample is 10dB ...
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2answers
832 views

Designing a lowpass filter to minimize aliasing in pre-decimated streaming audio

I need to apply low-pass filter to PCM files. There are several methods such as FIR filters, IIR (butterworth-chebyshev..) filters but it seems to me applying Fast Fourier transform and eliminating ...
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1answer
172 views

Estimating harmonic loudness

I'm working with data that looks something like this: ...
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1answer
312 views

How do Phons relate to loudness?

Let's say I have the very same signal stored in 2 .wav files. And lets say I process each file using a function, and it is established that one file is 20 phons ...
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1answer
3k views

how important is the pre-emphasis stage in MFCC?

I wrote a simple MFCC routine, using all the steps usually suggested in the literature with the exception of the pre-emphasis stage. How important is this stage? Also, some authors suggest that using ...
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4answers
7k views

Calculate Phase angle between two signals i.e. digital Phase meter

I want to calculate the phase difference between the two channels(signals) of a stereo audio file.(uncompressed data i.e. PCM samples). I want to determine whether whether two signals are in phase or ...
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Calculate Phase angle between two signals [duplicate]

Possible Duplicate: Calculate Phase angle between two signals i.e. digital Phase meter I want to calculate the phase difference between the two channels(signals) of a stereo audio file.(...
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2answers
876 views

Guitar effects explanation books

Can you please suggest some books to get how guitar effects: distortion, overdrive, phaser and so on are achieved ?
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Simplest way of detecting where audio envelopes start and stop

Below is a signal which represents a recording of someone talking. I would like to create a series of smaller audio signals based on this. The idea being to detect when 'important' sound starts and ...
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1answer
258 views

Issue with AMR file structure

For developing an application I have to identify a region of sound. But I’m not sure if I have my basic knowledge right. I’m using AMR-NB sound clips to identify a sound region. According to my ...
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2answers
10k views

How to extract vocal part from stereo audio signal?

I'm now processing MP3 file and encounter this problem. My MP3 is stereo encoded. What I want to do is extract vocal part for further processing(whatever mode of output signals, mono or stereo are ...
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1answer
2k views

Synchronizing Two Audio Tracks

I would like to implement an algorithm that synchronizes two audio tracks. The two tracks are very similar; you can imagine they are two recordings of sound (not necessarily music) from two ...
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2answers
238 views

How do the English pronunciations of “tiu” and “tou” differ?

I'm creating an app (I'm a software developer) that for different sounds (interjections from the user) will do different things. For example if the user says: "Tiu", than it will do task A, if it ...
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1answer
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Find similar Music using FFT Spectrums

I have been experimenting some weeks to find a way to match/find similar Songs in a Library containing different Genres of Music. My first try was to Detect Features like Tempo or how much Bass there ...
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1answer
478 views

From a 44kHz audio signal, how could I extract 32 evenly spaced frequency bands?

Assume I have an array of floating points that contain the last 8192 samples played. How would I convert that array to 32 frequency bins that are spaced from 0Hz - Nqyst?
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9answers
12k views

Is there an algorithm for finding a frequency without DFT or FFT?

I was looking in the Android app store for a guitar tuner. I found a tuner app that claimed it was faster than other apps. It claimed it could find the frequency without using the DFT (I wish I still ...
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596 views

How to determine the “variability” in a frequency range?

This is a continuation of a previous question. I'm trying to analyze breathing and snoring sounds, and while I can fairly well detect snoring now, breathing is a bigger challenge. I've learned that ...
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2answers
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The current state of the art in audio signal interpolation

Three questions: What are all the metrics one can use to measure audio interpolation quality, objectively? (but also in terms of psychoacoustics if possible) By those metrics, what is the current ...
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1answer
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Creating a spectrogram

I've been trying to work out the logic for this task, and plan to use the KissFFT source package to perform the fast fourier transform. Please let me know if this looks about right: Allocate an FFT ...
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4answers
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Programming a Vocoder

I'm want to program my own vocoder synthesizer like "Songify" but I can't find a simple of vocoder algorithm. Maybe you can explain or tell where to look for information on how vocoder works.
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2answers
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short time energy function explanation

Will someone be able to explain short time energy function for me in a simple way explaining all the variables in the formula? The issue that I have is in the formula the summation varies m like, ...
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1answer
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using Short-Time Energy and zero crossing rate

I am trying to identify a sound region of a word from a wav file (44100Hz). My research gave me the idea that I must use "Short-Time Energy" and "zero crossing rate". But the problem is I am not clear ...
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2answers
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Parametrically degrade the speech to remove emotional content

I am happy to accept suggestions either in R or Matlab, but the code I present below is R-only. The audio file attached below is a short piece of conversation between two people. My goal is to ...
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1answer
799 views

Lip sync over IP

What are the most used techniques for lip sync (audio/video sync) for conferencing over IP (and related issues with playout buffering)? I could find a lot of paper about audio intrastream ...
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2answers
527 views

Strange results from my FM synthesis algorithm

It seems that I haven't studied enough about FM Synthesis. My current algorithm looks like this: ...
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2answers
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Detecting drum bpm in a noisy .wav file

I am looking for algorithm(s) to solve the following problem: Given a noisy .wav sound capture (some wind + friction noise on the microphone), how to detect the BPM of a soft drum beat? I have ...
9
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1answer
2k views

Frequency Modulation Synthesis Algorithm

Based on what I read, I've made an algorithm for FM sound synthesis. I'm not sure if I did it right. When creating a software synth instrument a function is used to generate an oscillator and a ...
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1answer
1k views

Discrete Fourier transform - finding the fundamental quickly?

First, I appologize because I'm a software developper and it's been a very long time I didn't dive into pure mathematics, so my question might seem dumb. I hope not. The context is pitch recognition ...
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16k views

How to create a sine wave generator that can smoothly transition between frequencies

I am able to write a basic sine wave generator for audio, but I want it to be able to smoothly transition from one frequency to another. If I just stop generating one frequency and immediately switch ...