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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3
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1answer
319 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 ...
12
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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, ...
5
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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 ...
7
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2answers
820 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 ...
3
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1answer
172 views

Estimating harmonic loudness

I'm working with data that looks something like this: ...
7
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1answer
306 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 ...
4
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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 ...
5
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4answers
6k 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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0answers
77 views

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.(...
7
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2answers
864 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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6answers
43k views

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 ...
3
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1answer
257 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 ...
15
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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 ...
7
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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 ...
5
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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 ...
16
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1answer
4k views

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 ...
3
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1answer
475 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?
33
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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 ...
8
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4answers
594 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 ...
11
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2answers
2k views

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
5k views

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 ...
11
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4answers
11k views

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
4k views

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
2k views

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 ...
12
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2answers
1k views

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 ...
3
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1answer
797 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 ...
3
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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
1k views

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 ...
9
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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 ...
26
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5answers
15k 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 ...
9
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2answers
558 views

Pattern recognition for temporal data

I'm trying to detect and classify non-speech sounds. Currently, I'm using a series of moving overlapped power spectrums from training sounds as the features I am looking for. When I do analysis, I'm ...
47
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4answers
21k views

Analogous Library to OpenCV for Audio Processing / Analysis

I understand OpenCV is the de facto library for programming image processing in C/C++; I'm wondering if there is a C or C++ library like that for audio processing. I basically want to filter raw waves ...
4
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2answers
2k views

Whats the optimal window function to use for analyzing real-time data samples?

Say you wanted to run a X point FFT on the last X audio samples that were played. The problem being, using a normal hann window function would place emphasis on the "middle" of the audio sample. ...
15
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4answers
4k views

Lower frequencies stronger in FFT?

I'm calculating FFT from microphone input. I notice that lower frequencies always seems to have more power (higher dB) than higher frequencies. I cut the data into frames of 24576 bytes (4096*6). ...
12
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2answers
266 views

Subtracting sensors output that have mutual inductnace between them

Background: The software that i use to anaylze my signals is matlab. I have two audio signals that were recorded using two magnetic sensors. Lets call one sensor A and the other one B. A and B have ...
9
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1answer
239 views

How to segment phone call audio into silence/non silence?

My problem is that I don't know the energy of the background noise, so I can't just threshold the energy. The processing is done in real time, and I have about 500msec to decide. Ideally, I'd want ...
4
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1answer
503 views

Why did my sine wave turn into a square wave when written to a WAV file in Octave?

I am trying to use Octave to generate a pure sine wave. The code for the same is as follows: x = 10.*sin(2*pi*(300/16000)*(0:1:400)); The sampling rate is ...
6
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4answers
2k views

How to recognize patterns in a noisy audio frequency spectrum?

I have an audio spectrum (FFT) that looks like this: ...
27
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4answers
34k views

Hilbert transform to compute signal envelope?

I've heard that the Hilbert transform can be used to calculate the envelope of a signal. How does this work? And how is this "Hilbert envelope" different from the envelope one gets by simply ...
10
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1answer
4k views

Applying A-weighting

I want to calculate the loudness of a given signal and I've come across this thread. I'm slightly confused about the A-Weighting aspect of it. I can't post images, but I think this site has the ...
5
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2answers
261 views

Is there any way to make a sampling rate reduction in the frequency domain?

Is there any way to take advantage of the frequency domain to make a clean/nice sampling rate reduction effect ? and how ? Basically, deducted from my tests: if I set to 0 the magnitude uppon a ...
11
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3answers
19k views

Autocorrelation in audio analysis

I'm reading up on Autocorrelation, but I'm not sure I understand exactly how it works and what output I should expect. Am I right in thinking that I should input my signal to the AC function and have ...
7
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1answer
377 views

MP3 Decoding Discrepancy

Is it possible for two different decoders to output two different sample results for the same MP3? I'm decoding an mp3 on Matlab and it produces two columns (presumably one for each channel). When I ...
14
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2answers
5k views

What methods can be used to identify and remove echo from an audio system?

Background I am designing a system that will have a single small microphone and speakers for use in a phone type setting. Easiest example I can give is a Skype conversation where you are using your ...
3
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1answer
421 views

Am I handling offline FFT correctly?

I need some help clarifying FFTs and what they represent. I have a buffer containing compressed audio. Due to limitations, I can't handle the full uncompressed audio but can decompress small segments ...
14
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1answer
7k views

Getting loudness of a track with RMS

I'm trying to calculate the loudness of an audio track I have stored in a buffer. The buffer contains PCM data of the signal and I want to get how 'loud' it is by using Root Mean Squared. I assume I ...
12
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1answer
529 views

Recognizing math functions within songs

I'm new to DSP, and just discovered this StackExchange, so apologies if this isn't the right place to post this question. Is there a resource that describes genres in a more mathematical terms? For ...
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4answers
4k views

Complex conjugate and IFFT

I asked a question over on stack overflow. I'm having a slight problem however. As suggested by Paul R I am mirroring my lower $n/2$ bins into the upper $n/2$ bins. I have a few questions however. ...