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.
140
questions
4
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1answer
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Compensating Loudspeaker frequency response in an audio signal
I have been working on a project in which I was required to work on the audio signals recorded from the loudspeaker kept in front of a filter. So, to simply explain it:
$$\boxed{\rm LoudSpeaker} \...
11
votes
2answers
3k views
How to deal with low fundamental when using AMDF for pitch extraction?
I am using the Average Magnitude Difference Function to estimate the fundamental frequency of a quasi-periodic audio signal. The AMDF is defined as
$$ D_n = \frac{1}{N-n}\sum_{k=n}^{N-1}|S_k - S_{k-n}...
65
votes
7answers
104k 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 ...
31
votes
5answers
18k 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 ...
54
votes
10answers
38k 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 "...
12
votes
3answers
13k 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 ...
20
votes
4answers
26k 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 ...
22
votes
3answers
11k views
Tips for improving pitch detection
I'm working on a simple web app that allows the user to tune his/her guitar. I'm a real beginner in signal processing, so please don't judge me too harshly if my question is inappropriate.
So, I ...
0
votes
1answer
466 views
How can I find SNR, PEAQ, and ODG values by comparing two audios?
I have two audios, A and B. B is the noisy version of A. I want to know how much B is different from A.
Can I do this by calculating signal to noise ratio (SNR), Objective difference grade (ODG), and ...
46
votes
5answers
63k 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 ...
16
votes
2answers
12k 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 ...
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 ...
35
votes
9answers
13k 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 ...
31
votes
4answers
42k 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 ...
13
votes
4answers
22k 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
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 ...
5
votes
3answers
8k views
Generalized Cross Correlation
What is the definition of Generalized Cross Correlation?
I need it for estimating the delay (time offset) between two audio signals. The time offset between the two signals might be as large as 500ms....
4
votes
1answer
2k views
Use MATLAB to Restore a Signal from a Given Degraded Signal Using Tikhonov Regularization
Anyone could share how to develop an application in MATLAB to restore the signal from a given degraded signal using Tikhonov regularization i.e restoring the signal $f$ via solving
$$
\min || g - f ...
7
votes
5answers
10k views
Equal power crossfade
I am working on an audio system where I need to implement an equal power crossfade between two signals over a specified number of samples. Initially I implemented a linear fade but this creates a dip ...
4
votes
2answers
5k views
Audio time stretching, without pitch shifting
This might be a Sound Design question, or a StackOverflow question since I am attempting to do this with Java.
I would like to play back a sound at the same pitch, but stretched out in time.
My ...
44
votes
6answers
48k 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 ...
7
votes
3answers
6k views
Reconstruction of Audio Signal from its Absolute Spectrogram
I have the absolute Spectrogram of an audio signals.
I lost the phase data of the Spectogram because of various processing applied on the original spectrogram of the signal.
I'm trying to reconstruct ...
9
votes
2answers
17k 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?
8
votes
2answers
9k views
Help implementing audio dynamic range compression
I am trying to implement audio dynamic range compression in JavaScript (not using web audio API).
There is a lot of articles for sound technicians, and some high level documentation, but I couldn't ...
6
votes
4answers
3k views
How to recognize patterns in a noisy audio frequency spectrum?
I have an audio spectrum (FFT) that looks like this:
...
10
votes
2answers
872 views
Filtering of streaming audio
Given something like 103 data points ($N=103$), a DFT will give back 103 frequency values. Then to do something like filtering the high frequencies involves setting the high frequency values from the ...
5
votes
4answers
8k views
How Can I Resample a Signal with an Arbitrary Factor (For Example - 128000Hz to 16000.1Hz) in MATLAB?
I need to simulate the sampling of a continuous (fsCtu = 128000Hz), acoustic signal with two microphones that have a slight offset in sampling rate (fsMic1 = 16000, fsMic2 = 16000.1) in Matlab. What ...
8
votes
2answers
2k views
Fast pitch recognition
I need to detect pitch (measure signal frequency) while the musicians play music, giving a warning if they are out of tune, but music happens to be a bit too fast for FFT (Fast Fourier Transform).
...
4
votes
2answers
3k views
Estimating the time offset (delay) between two audio signals in real-time
I'm looking for an algorithm which can accurately estimate the time-offset/delay between two audio signals in real-time. One audio signal is a loudspeaker signal and the other signal is the microphone ...
1
vote
1answer
239 views
Precise Centre frequency of an All-pole digital filter
I'm not an engineer and have essentially taught myself all I know - this present problem is giving me some problems.
I have an all-pole filter (a gammatone filter - impulse response is essentially a ...
0
votes
1answer
2k views
Specific Frequency Resolution
I have an audio signal
SampleRate Fs: 44100 Hz
TotalSamples: 94144 samples
Duration t: 2.1348 s
The frequency resolution is given by ...
5
votes
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 ...
0
votes
2answers
4k views
Wav To Spectrogram, Back To Wav
Currently, I'm writing a Python script, which should do the following:
read an audio file respectively a wav file via scipy.io.wavfile.read().
calculate the spectrogram of given wav file.
write the ...
54
votes
4answers
25k 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 ...
17
votes
4answers
31k 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) ...
16
votes
2answers
11k 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 ...
15
votes
2answers
8k 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 ...
8
votes
1answer
2k views
Coloring a waveform with spectral centroid or by other means
Update 8
Unhappy of having to upload tracks to the service and looking at the new release of RekordBox 3 I've decided to take another look for an offline approach and a finer resolution :D
Sounds ...
8
votes
5answers
4k views
Piano pitch detection
I'm working on a piano tuning program and part of it requires real-time pitch detection. Here is the scheme I have so far which works to some degree but could probably use some refinement.
I'm ...
6
votes
4answers
1k views
Monotonic, symmetrical soft-clipping polynomial
Okay, I intend to answer this question myself, but I would like other folks to pipe in on it. The issues or requirements are this:
Input to soft-clipper is $x(t)$, output is $y(t)$.
Output $y(t)$ ...
8
votes
1answer
8k views
Speech comparison algorithm for rating on similarities
I am trying to compare 2 speech samples and rate them on similarities. Think of someone trying to repeat a phrase, and then comparing those 2 audio files.
I started by implementing the MFCC (http://...
11
votes
3answers
14k 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|}...
5
votes
3answers
10k views
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 ...
5
votes
5answers
320 views
Remove a known wav file from recorded file
I looked at many similar questions but still cannot find an answer to my problem.
I'm playing an audio file N, while recording a man talking (b).
The problem is that the microphone gets the played ...
15
votes
1answer
8k 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
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 ...
11
votes
1answer
5k 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 ...
7
votes
1answer
5k views
SPL values from FFT of microphone signal
I am trying to get a frequency response vs dB SPL plot for recorded sounds. I have a microphone with a flat response, a sensitivity of 4mV/Pa, and 1 Pa = 94dB.
To get the dB SPL, I have the following ...
12
votes
3answers
22k 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: ...
6
votes
3answers
8k views
How to change speed of audio samples without changing pitch? [duplicate]
This question may be more appropriate on music stack exchange but since sound is a branch of physics and this is more of a technical question relating to sound waves than a musical question I asked it ...
