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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60
votes
6answers
26k views

If humans can only hear up to 20 kHz frequency sound, why is music audio sampled at 44.1 kHz?

I read in some places that music is mostly sampled at 44.1 kHz whereas we can only hear up to 20 kHz. Why is it?
58
votes
7answers
90k views

How do I implement cross-correlation to prove two audio files are similar?

I have to do cross correlation of two audio file to prove they are similar. I have taken the FFT of the two audio files and have their power spectrum values in separate arrays. How should I proceed ...
49
votes
4answers
22k 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 ...
49
votes
10answers
31k views

Algorithm(s) to mix audio signals without clipping

I'd like to mix two or more PCM audio channels (eg recorded samples) digitally in an acoustically-faithful manner, preferably in near-real-time (meaning little or no peek-ahead). The physically "...
43
votes
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 ...
41
votes
5answers
52k views

What is the difference between phase delay and group delay?

I'm studying some DSP and I'm having trouble understanding the difference between phase delay and group delay. It seems to me that they both measure the delay time of sinusoids passed through a ...
33
votes
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 ...
32
votes
4answers
10k views

What are good ways to detect signal clipping in a recording?

Given a recording I need to detect whether any clipping has occurred. Can I safely conclude there was clipping if any (one) sample reaches the maximum sample value, or should I look for a series of ...
27
votes
5answers
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 ...
27
votes
6answers
2k views

Calculating the PDF of a waveform from its samples

A while ago I was trying different ways to draw digital waveforms, and one of the things I tried was, instead of the standard silhouette of the amplitude envelope, to display it more like an ...
27
votes
4answers
35k 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 ...
21
votes
5answers
2k views

What are the audio equivalents of images like “Lena”, “Mandrill”, and “Cameraman”?

I don't know what to call these images, but they're very often used in the image processing world as standard examples to show the output of a particular operation. Here's a link to the page ...
20
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6answers
12k views

Why do we choose 44.1 kHz as recording sampling rate?

Peoples' ears can hear sound whose frequencies range from 20 Hz to 20 kHz. Based on the Nyquist theorem, the recording rate should be at least 40 kHz. Is it the reason for choosing 44.1 ...
19
votes
6answers
770 views

What does one need to know in order to build a software synthesizer?

For someone who has some experience in software development, loves his synths and knows the [very] basics of digital audio representation, what are the first steps he should follow in order to program ...
19
votes
4answers
21k views

Reconstruction of audio signal from Spectrogram

I have a set of songs for which I extracted the magnitude spectrogram using a Hamming Window with 50% overlap. After extracting the spectrogram, I did some dimensionality reduction using Principal ...
18
votes
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 ...
18
votes
3answers
3k views

Show others how I hear myself

Sooo .. I've been thinking about this stuff. We all know that we sound different from what we hear of our own voice. It is easy to find out how others hear us by recording oneself and listen to it. ...
16
votes
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 ...
16
votes
3answers
18k views

human speech noise filter

Does anyone know of a filter to attenuate non-speech? I am writing speech recognition software and would like to filter out everything but human speech. This would include background noise, noise ...
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). ...
15
votes
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 ...
14
votes
4answers
26k views

Feature extraction for sound classification

I'm trying to extract features from a sound file and classify the sound as belonging to a particular category (eg : dog bark, vehicle engine e.t.c). I'd like some clarity on the following things : 1) ...
14
votes
2answers
6k 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 ...
14
votes
4answers
13k views

Most efficient way to extract frequency components of audio signal with near-human accuracy in real time

I'm trying to work out how (if possible) to extract the frequency components of an arbitrary audio sample (typically music) in an FFT-like manner, but in my research on the FFT algorithm, I'm learning ...
14
votes
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 ...
14
votes
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 ...
14
votes
5answers
2k views

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 ...
14
votes
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 ...
14
votes
1answer
1k views

Estimating onset time of a tone burst in noise?

What techniques might one use to estimate the onset time of a sinusoidal tone burst in a noisy signal? Assume the tone burst has a known fixed frequency (but unknown phase) and a very sharp rise ...
14
votes
2answers
378 views

When does the laugh track kick in?

One of my weekend projects has brought me into the deep waters of signal processing. As with all my code projects that require some heavy-duty math, I'm more than happy to tinker my way to a solution ...
13
votes
2answers
2k views

Learning roadmap for the audio signal processing beginner

I would like to start learning audio signal processing. There are numerous books online and academic papers all of which seem to skip the fundamentals of the topic. I would like to know a rough ...
12
votes
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, ...
12
votes
2answers
10k views

Add odd/even harmonics to signal?

How do I add odd or even harmonics to a floating point signal? Do I have to use tanh or sin? What I'm trying to do is achieve some very simple distortion effects, but I'm having a hard time finding ...
12
votes
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 ...
12
votes
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 ...
12
votes
3answers
2k views

Transmit data through sound between 2 computers (very close distance)

I'm writing an example on transmitting data through sound betwwen 2 computers. Some requirements: The distance is very close , i.e the 2 computers are basically adjacent to each other Very little ...
12
votes
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 ...
12
votes
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 ...
12
votes
3answers
17k views

How do I compare 2 audio files, returning a percentage of difference

Is there a way to compare 2 audio files, whereby a percentage of difference will be returned? I am using C# VS08, .net framework 3.5. Edited: difference in the sound (like for eg. Audio 1: ...
12
votes
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 ...
11
votes
5answers
6k views

When to consider double (64 bit) floating point for Audio

When synthesizing and processing audio on modern processors, when would one consider using something other than single precision (32 bit) floating point? Obviously, the audio coming in and going out ...
11
votes
3answers
20k 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 ...
11
votes
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.
11
votes
3answers
11k views

How to resample audio using FFT or DFT

I'm down sampling voice audio by first performing an FFT, then only taking the parts of the result that I need, and then performing an inverse FFT. However, it's only working properly when I'm using ...
11
votes
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 ...
10
votes
4answers
446 views

Audio output quality

When a digital device (PC, portable media player, etc.) plays a digital audio file (ogg, mp3, flac, etc.) the audio signal output is always the same, regardless of the device type/brand, am I right? ...
10
votes
3answers
11k views

Digital Distortion effect algorithm

I read the DAFX book by Udo Zölzer about the distortion effect at pages 124-125 and it says that suitable simulation of the distortion is given by the function: $$f(x)=\frac{x}{|x|}\left(1-e^{x^2/|x|}...
10
votes
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 ...
10
votes
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 ...
10
votes
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 ...