9
votes
Accepted
Why is each window/frame overlapping?
Why is each window/frame overlapping?
Windowing is a means to stationarize signals. Inside a small enough window, you can expect that the properties of the signal chunk do not vary too fast. And you ...
- 31.7k
9
votes
What are i-vectors and x-vectors in the context of Speech Recognition?
The i-vectors and x-vectors share the ability to represent speech utterance in a compact way (as a vector of fixed size, regardless of length of the utterance).
The extraction algorithms of i-vectors ...
- 231
8
votes
Accepted
Signal processing for audio and speech
In line with a previous similar question here are my suggestions:
There are so many nice books but I believe you should first have a look at the science of sound from Rossing for getting the most ...
- 28k
7
votes
Why is each window/frame overlapping?
More overlap means you end up with more windows (of a given length) per second of audio. More windows (of a given length) requires more FFTs which requires more MACs or FLOPs which generally requires ...
- 35.2k
7
votes
Accepted
Identify abrupt changes in an audio waveform
Synchrosqueezed Wavelet Transform is an option. I have developed a complete algorithm for this task, which scores 100% train accuracy and 86% test accuracy with 0.05 sec tolerance, without machine ...
- 8,815
7
votes
What is the explanation that filter EQ changes "Ah" to "O"?
Expanding a bit on Tim Wescott's answer:
Vowels are made by a combination of an excitation and a resonant filter. The excitation comes from the glottis in your throat. The speed of that vibration ...
- 42.6k
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 ...
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5
votes
Negative Signal to Noise Ratio
Signal to Noise ratio (SNR) is a ratio of powers, hence it's always greater than or equal to zero.
On the other hand, very commonly, SNR is expressed in decibel (dB) notation:
$$ \text{SNR}_{\text{dB}}...
- 28k
5
votes
Converting speech audio to telephone audio
Band-pass filtering with cut-off frequencies of 300 Hz and 3400 Hz should result in a good approximation. Try with a Chebychev filter or order not more than 6.
Then you may need to downsample your ...
- 4,986
4
votes
Signal processing for audio and speech
But which features of signals reveal differences between piano and guitar
Look at things like timbre; basically, most things when excited to oscillate will not only produce a single tone, but a set ...
- 29.6k
4
votes
Cepstrum calculus disambiguation
A common technique for computing the inverse fft is to invert the imaginary part of the input array, perform a forward fft and then invert the imaginary part of the output array. In the case of the ...
- 548
4
votes
What is a "pitch period"?
The pitch period of a perfectly periodic function, $x(t)$, is the smallest positive value $P>0$ such that
$$ x(t+P) = x(t) \qquad \forall t \in \mathbb{R} $$
Now, simply because a function is ...
- 20.1k
4
votes
Identify abrupt changes in an audio waveform
This is more meat-and-potatoes.
An oft-used algorithm in audio is the pitch detector, which of course is attempting to accurately determine the instantaneous pitch of a note. The note is a waveform ...
- 20.1k
4
votes
Identify abrupt changes in an audio waveform
One domain where continuous (complex) wavelets (CWT) are especially efficient is when you expect the 1D signals to be piecewise regular. When a signal is $C^\alpha$ by pieces, then the $\alpha$ ...
- 31.7k
4
votes
Accepted
What is the explanation that filter EQ changes "Ah" to "O"?
What is the explanation of this phenomenon?
Your vocal cords have a very raspy sort of vibration -- they generate spectral components at the fundamental, and at many many harmonics going up from ...
- 11.8k
3
votes
Signal processing for audio and speech
Here is a series of tutorial videos on Speech and Audio Processing by Professor E. Ambikairajah; about 1 hour each. They can serve as a basis to focus on more specific topic.
Speech and Audio ...
- 31.7k
3
votes
Calculation of SNR
I have a suggestion for you.
Compute the mean power of the signal at the input of the filter without any signal applied at the input. You should only receive noise. This will be the noise power $ P_{...
- 677
3
votes
Accepted
Logarithmic Spacing
Let's note $x_k$ and $x_{k+1}$ two successive samples. In the usual case of uniform sampling, the spacing between two successive samples is independent of $k$ and is given by $T$, the sampling period.
...
- 947
3
votes
Parseval's Theorem for discrete series
You are right, but you have mistaken the DTFTs...
Considering the Parseval's relation for real discrete-time sequences to be:
$$ \sum_{n} x[n]y[n] = \frac{1}{2\pi} \int_{-\pi}^{\pi} X(e^{j\omega})Y(...
- 28k
3
votes
Accepted
How to edit audios so that they have the same length
ImageMagick is a great tool to edit multiple images in the command line. Thanks to your question, I just discovered SoX, or Sound eXchange:
SoX is a cross-platform (Windows, Linux, MacOS X, etc.) ...
- 31.7k
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 ...
- 28k
3
votes
Mel Cepstral Distortion
Ok so here is what I found. The distance is dependent on the way that the mfccs are calculated. This makes sense to me and also explains why cepstral mean normalization affects the values of the MCD.
...
- 81
3
votes
Is there any way that we can perform speech recognition without using Fourier transforms?
All recognition tasks (doesn't even have to be speech recognition) are reductions of a very high-dimensional signal (your speech recording's dimension is the number of audio samples!) to a low-...
- 29.6k
3
votes
Problem using pitch shifting with TD-PSOLA and formant preservation
I believe you're correct, it's an issue with the overlap-add process. The code doesn't correctly calculate the triangular windows for lower pitches.
In short, the windows are too big.
It produces ...
- 185
3
votes
Conceptually confused by LPC for speech: Do we synthesize by the inverse filter (FIR)?
Given speech samples, the LPC computation yields linear prediction coefficients $a_1,\ldots,a_p$. These describe a dependence model for a few samples. It is assumed that for short-enough collections ...
- 616
3
votes
Accepted
Is this statement correct from DSP aspect?
It's almost a matter of philosophy, i.e., difficult to argue hard facts.
On the one hand all the features you mention can be extracted from the raw signals. So in theory the network should be able to ...
- 2,413
3
votes
Accepted
Why are males' pitch lower than females?
Males usually have deeper voices than females because they have longer vocal folds.
Yes
Short and relaxed vocal folds produce a deep voice, long and tense vocal folds produce a high-pitched voice.
...
- 42.6k
3
votes
Accepted
Male and female voice spectrum
in my opinion and limited experience, you can't just compare spectra of male and female voices to identify which is which.
2 decades ago i worked on two products called PurePitch and PitchDoctor that ...
- 20.1k
3
votes
Identify abrupt changes in an audio waveform
I would point on 2 approaches:
Parametric Model
You build a parametric model for the data and then apply simple change detection methods for the paraments. For instance, you may build a Kalman Filter ...
- 19.3k
Only top scored, non community-wiki answers of a minimum length are eligible