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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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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 ...
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214 views

Are there other filter models besides IIR and FIR for processing digital signals?

I am very familiar with IIR and FIR filters for processing digital signals (audio is my main interest). I am keen to know if there are alternative filters based on different mathematical models to IIR ...
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How to know if a audio file is real lossless using its spectrogram?

I need to know if a certain audio file is real lossless and not a transcode (i.e. lossy converted in lossless, but lossy at the end). I was told there was a way to do this by creating a spectrogram of ...
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Acoustic feedback detection algorithm

I am woking on an sound processing application where I have limited processing resources. I need to be able to detect acoustic feedback and then signal that such an event occurred. I am not concerned ...
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1answer
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SNR calculation in audio reconstruction and what is an acceptable value

I have the original signal, a .wav file and by blind source separation, i have reconstructed the signal. How can I calculate the SNR in this case? or is it good to use any other performance parameters?...
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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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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. ...
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1answer
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Spectrograms for neural nets

General Question Given an audio file, say a 16-bit wav, what are some standard methods to preprocess a spectrogram of this wav so that it may be fed into a neural net? Context In Lee et al's 2009 ...
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Best beginners guide to sound/audio

I'm trying to learn audio programming, but am struggling with the basics of audio. Audio programming is new to me (I'm a C/C++ developer with over 30 years of experience), but I don't have any ...
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1answer
851 views

What is the difference between PSOLA and TDHS time-scaling or pitch-shifting?

What is the difference (if any) between PSOLA (Pitch Synchronous Overlap and Add) and TDHS (time domain harmonic scaling) time-pitch modification algorithms?
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Detection of wobbling sound with rising notes

In an audio file I would like to find calls of a specific bird specie, one that makes wobbling sound with rising notes. Can you recommend what would be the best audio feature to calculate in time or ...
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Measuring the “frequency stability” of a signal

I'm thinking about ways to automatically identify and analyze lots of musical instrument samples, and would like to find the part of the sample that's the most stable in frequency, without any ...
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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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How does Adobe After Effects generate its “audio spectrum” effect?

I'm trying to replicate the "audio spectrum" effect from Adobe After Effects. An example can be seen in this video: Obviously, it has to be some variant of a fourier transform, but I've tried taking ...
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1answer
308 views

Algorithm for identifying beat grid in music with non-constant tempo

Are there any algorithms for automatically fitting a beat grid to a song (tempo matching) when the song has a slowly changing tempo or BPM? The output would be a vector of onsets. Here's a picture ...
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Filter away sinusoidal noise properly

I have a stereo music signal corrupted by strong sinusoidal noise that varies over time. Here is the spectrogram of Left channel I plotted with Matlab. As you can see there are 3 or 4 strong ...
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3answers
105 views

Classifing audio signals to detect fault

The basic idea i'm working on is automatic fault detection using audio signals captured from Motor. I have like set of sample audio signal which are recorded when there is no fault and with fault at ...
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1answer
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From MFCC to Machine learning. What are the steps?

I was able to get a dataset with MFCC coefficients. However, depending on the length of my sound file I get a different sized matrix. As in, 13 (13 MFCC coefficients) by XXX, where XXX will vary ...
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Understanding audio downsampling code (JavaScript) and adapting to a different ratio

I inherited responsibility for some audio downsampling code that I don't fully understand: ...
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1answer
776 views

Calculating Beam pattern, Directivity of microphone array

As answered in Relationship of Fourier Transform and Spatial Directivity Pattern the relation between directivity pattern and weights is given. But I am bit confused, I have Beamformer weights (...
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1answer
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Audio Steganography using Phase Encoding Technique

I referred the below papers to understand the working of Phase Encoding,but the explanation is not proper.Just a few steps have been given in brief. ijcsi.org www.ijarcce.com Can someone please ...
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Onset Detection Peak Picking

I have a guitar music signal and I want to detect the onset points via energy. This is the signal example: This is the energy plot: My question is what would be the best method defining the ...
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Note Recognition Software

I'm wondering if anyone knows of any open source solutions to note recognition. I'm planning on building a program for online sheet music, which recognizes your notes, and tries to tell you what you ...
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1answer
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Convolving With half-Hanning

The goal is to get a 200ms decaying delay of an audio signal while preserving the sharpness of attacks to mimick human perception of sound. The paper I'm following convolves a 200ms half-Hanning ...
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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 ...
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3answers
341 views

Voice recovery with high-noise, low-gain

I have a recorded interview with two speakers, in which one voice's gain was much higher than the other (by three orders of magnitude). Increasing the gain does not help, because speaker 2's voice is ...
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6answers
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Find Short Clip of Audio Within Longer Clip Of Audio

I need help figuring out a way to automatically synchronize two audio files that are both different recordings of the same source. More info below. This example is from a concert, where there is only ...
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1answer
221 views

Combining time domain and frequency domain data for BPM detection in real time

I would like to write a BPM detector. Does either of the domains provide more information for this type of analysis? What exactly is being looked for in the data to detect onset for BPM detection in ...
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1answer
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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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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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1answer
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Demodulating FSK audio in Python

I am trying to write a Python script that can demodulate an FSK modulated audio file and return the data encoded in the audio. The data being transmitted is GPS NMEA strings which are embedded as the ...
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Verifying MFCC Final Result

I have created a Octave code for MFCC calculation for speech recognition, which takes a wav file as input and then gives the MFCC as the output. But the final results are a bit confusing, so I am ...
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1answer
638 views

$1/n$ octave complex smoothing

An excellent answer to this post explains how to do $1/n$ octave energy smoothing and mentions complex smoothing can be done as well, but it's tricky business because of phase wrapping. How is phase ...
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STFT overlap add method - What am i doing wrong?

i am trying to implement a short time fourier transform of an audio signal to do some filtering (subtract another stft signal using spectral subtraction). I am using a hamming window function with 50% ...
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1answer
200 views

Teager's Energy Operator

Does anyone know how to find information on the derivation (rather than the definition and application) of Teager's Energy Operator, both for the continuous and sampled case. I have Kaiser's early ...
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0answers
587 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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4answers
813 views

Noise reduction using multiple recordings of the same signal

I have two (audio) recordings of one analog signal, I used two separate devices (smart-phones) recording from two different locations. Each recording device is pretty crappy, lot's of electrical hum ...
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3answers
575 views

Signal processing for audio and speech

I have started learning about signals and I am interested in sound signals. There are some questions that I need to resolve. For example in music, difference of notes like 'sol' and 'la' are about ...
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Why audio clipping produces harmonics?

Today I was talking to my collegue about clipping and he said that clipping produces harmonics in frequency domain as he showed me the clipped signal frequency response. However when I asked him why ...
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2answers
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Eliminate Signal A from Signal B

i have 2 audio signals, both available in time and frequency domain: A: Noise B: Noise(A with minor variations) + Additive sounds (C) i want to eliminate Signal A from B to leave only C. C is for ...
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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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Is it accurate to say in digital audio that, when a fader is down, then its value is “-$\infty$”?

Is it accurate to say in digital audio that, when a fader is down, then its value is "-$\infty$"? What has confused me is, whether it's really $-\infty$ or some lower bound of the amplitude. it could ...
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2answers
213 views

Are there equations for placing sounds in the stereo field using delays?

I'm thinking of the typical stereo widener (haas, binaural) effects here, those that extend from the typical pan knob to create "out of the headphones" -experience. These processors can be automated ...
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5answers
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generating spectrograms in python with less noise

I'm trying to do something similar to Shazam (paper here) in Python. Basically the idea is to use FFT with sliding window to transform both WAV files and recorded audio to the same spectral ...
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333 views

Relation between samplingrate and frequency

I am working on Fourier Transformation, and applying this for recognizing an audioclip. I have a 9 second long audio clip of a guitar strumming an A-Minor. The audioclip has a sampling rate of 44100 ...
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5answers
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Pitch Detection: avoiding frequency doubling / halving

I am working on a tuning app. I have so far tried 3 different libraries with a number of different algorithms. However, when I compare to other tuner apps, I seem to be getting frequency doubling on ...
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2answers
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What audio effects, filters, distortions, etc. create a “vinyl” effect?

I want to make any song sound like it's being played on a gramophone record in the 1920s. What specific filters would I apply to make this happen? I'm looking for technical details, not some magical ...
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2answers
628 views

Generating inaudible sound waves using MATLAB

I want to generate inaudible sound waves (above 18kHz) with 48kHz sampling rate using MATLAB. Generating a sound wave with 18kHz frequency makes it inaudible. I've inserted the data in the 18kHz ...
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Sound absorption coefficients for frequencies higher than 8kHz and lower than 80Hz?

I am building a matlab simulation that applies room compensation to a 3D audio rendering system, and I'm searching for the frequency dependent sound absorption coefficients of various materials in ...
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1answer
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How to do continuous signal processing (i.e without windowing)?

I'm working in speech recognition research, and I wonder if there is a way to analyse sound just like how ears do (i.e. without windowing), so for example to take some feature continuously with an ...