13
votes
Accepted
Multi-channel audio upsampling interpolation
Does cubic interpolation (or any other) have any advantages over linear for the specific case of audio?
You'd use neither for audio. The reason is simple: The signal models you typically assume for ...
- 25.6k
11
votes
Accepted
How do impulse response guitar amp simulators work?
When talking about modeling, there are two things that usually get modeled: 1. the guitar amp, and 2. the speaker cabinet. Only the latter is modeled by an impulse response, which means that the ...
- 79.2k
9
votes
Accepted
Noise reduction using multiple recordings of the same signal
Is it possible to reconstruct the original pure signal?
No, that is information-theoretical impossible. Also, that signal doesn't exist, probably, to begin with ;)
However, you can definitely ...
- 25.6k
9
votes
Accepted
Can Principal Component Analysis (PCA) Solve the Cocktail Party Problem?
The Cocktail Party Problem is a Blind Source Separation (BSS) problem.
Given a linear mixture of signals:
$$ \boldsymbol{y} \left[ n \right] = A \boldsymbol{x} \left[ n \right] $$
We're trying to ...
- 39k
8
votes
Accepted
Finding Reference Audio Signal in Test Audio Signal and Cropping Accordingly
If you have a reference signal you want to find in a different signal then your model matches almost perfectly (Up to the environment the signal to be found is in) to Matched Filter.
So basically you ...
- 39k
8
votes
How do impulse response guitar amp simulators work?
If you're an EE student, you will have encountered the term LTI System (or you certainly will soon enough!): A system that, no matter the absolute time, outputs, given the same input, the same output; ...
- 25.6k
7
votes
Realtime sample rate conversion
The consumption time and transmission time is identical: One second of data is still one second of data regardless of sampling time. However if the transmitter and receiver are not synchronized then ...
- 36k
6
votes
Understanding the frequency domain
In simple terms, Image shown here speaks for itself.
Before speaking about Fourier Transform black magic lets understand idea behind it. Work of the Mathematician Joseph Fourier demonstrated any ...
- 181
6
votes
Accepted
Audio Processing Glossary
My own interpretation:
Sound: a mechanical wave that propagates through the air or water.
Audio: sound in the 20 Hz to 20 kHz range; in other words, sound that is (at least in theory) audible to a ...
- 13.5k
6
votes
What is the principle behind my voice echoing perpetually when 2 separate devices are used for playback and recording on video calls?
On each individual device, the speaker output can get subtracted from the microphone before it gets sent to other locations. This prevents others from hearing themselves through your microphone. When ...
- 261
5
votes
Accepted
Modelling Unwanted Signal in a LMS Adaptive Filter
The LMS and many of the variants of Adaptive Filters (In the Linear System context) work in the following settings (Intuitive):
You have access to 2 signals.
One signal is the result of the other one ...
- 39k
5
votes
Does 1 kHz sine tone means $\sin(2(1000)\pi t)$ or $\sin(2(500)\pi t)$?
The trigonometric functions "do not know" what a Hertz is and they do not care either. The only thing they know is that a full circle is $2 \pi$ radians. Whether this circle concludes in days, hours, ...
- 10k
5
votes
Multi-channel audio upsampling interpolation
Audio signals
An audio special-purpose analog-to-digital converter (ADC) normally has an internal or external analog low-pass filter and samples the analog filtered signal at a multiple of the target ...
- 12.3k
5
votes
Accepted
Actual Sample Rate slightly off?
If you calculate the error, 44102/44100 is only about 45 parts per million. That is well within operating tolerance of many crystal oscillators used in consumer equipment to generate audio sampling ...
- 1,441
4
votes
Accepted
What is the difference between undersampling and oversampling in analog to digital conversion ?
Good question as you can actually undersample and oversample at the same time! See my "DSP Puzzle" question on that specifically here:
How do you simultaneously undersample and oversample?
To best ...
- 36k
4
votes
Accepted
Compute group delay of an audio file from STFT
The problem is that the stft function is splitting the signal up into different windows. That means that the signal from time $n$ to $n+N_{w}-1$ is multiplied by
$$ ...
- 21.7k
4
votes
Realtime sample rate conversion
The usual answer is that that you must convert by integer ratios, therefore 44.1 kHz to 48 kHz requires integer up and down conversions. Since this has been repeated in DSP text books for at least the ...
- 721
4
votes
Analog-Digital Dilemma on Audio Signal Processing
the predominant stage of the speaker is the audio signal processing stage.
That's a questionable statement. Transducers, enclosure design, physical management of diffraction and dispersion are ...
- 30.6k
4
votes
Why mixing two sines sounds like a triangle?
I tried looking at what your libraries output as just a sine wave by modifying your code:
...
- 21.7k
3
votes
What is the best input for de-noising autoencoder for sound data?
As you mentioned MFCC features are one of the best features to represent audio as it captures both the time and frequency variations in the audio clip.You can get more details about MFCCS features in ...
3
votes
Audio Quality Assessment
Audio quality assessment is one of the most critical pieces of audio coding and enhancing applications. The task requires an accurate and objective (mathematical) modeling of human auditory system ...
- 26.5k
3
votes
Multi-channel audio upsampling interpolation
Sorry MM, I'm agreeing with Havakok on this one: A time domain interpolation solution should do just as well, practically speaking, and be significantly cheaper in terms of computation. (Assuming ...
- 6,875
3
votes
Noise reduction using multiple recordings of the same signal
As Stanley Pawlukiewicz said: even under ideal circumstance, you can gain 3 dB of SNR per doubling of recordings. I.e., to increase SNR by, say, 15 dB, you'd need to average $$ 2^{\frac{15}{3}} = 2^{5}...
- 1,169
3
votes
Modeling audio effects, reverse engineering
Now is there a general method to analyze and determine the exact function of that effect
No, there can't be. Effects can (and will be) arbitrary, non-linear, memory-affected mappings.
You will have ...
- 25.6k
3
votes
What is difference between frequency and pitch?in nutshell and simple words?
Frequency is mathematically defined as the number of cycles per second. So it is a more strict word mathematically. It is represented numerically by the unit called Hertz.
$f=1/T$, where $T$ ...
- 41
3
votes
Sampling a low-frequency sound doesn't work at some sampling rates
Could be a few things
You may be borderline clipping. Your sine wave has an amplitude of 1, which is just at the edge of clipping (depending on how its rendered). Try it with an amplitude of 0.5
Your ...
- 30.6k
3
votes
Accepted
What is the principle behind my voice echoing perpetually when 2 separate devices are used for playback and recording on video calls?
The Explanation from @ScienceGeyser provides a good explanation to the phenomenon.
There are two more things to address the question on how is this phenomenon avoided.
The feedback read by the ...
- 1,463
3
votes
Audio classification without deep learning
From what you've mentioned it looks like the task is for environmental sound event detection. I think that the best starting point for you is to check the DCASE challenge (Detection and Classification ...
- 10.4k
Only top scored, non community-wiki answers of a minimum length are eligible