Hi! This is Jooyoung Kim, mixing engineer and music producer.
The paper I submitted to JASA is currently under review.
The assigned editor is a renowned scholar in the field of acoustic signal processing at Tsinghua University. I’m more nervous than I was during my college exams..
By the way, today we will talk about the aliasing and sampling theorem – the basic knowledge for mastering audio.
Let’s start!!
Aliasing
Aliasing is a phenomenon caused by converting analog data to digital data. You can see the same picture; the left one looks normal, but the right one appears distorted. The right image was made by resizing left image in low resolution.
Why does it happen?
Nyquist-Shannon Sampling Theorem
But the courageous Mr. Nyquist says in a strong tone that this is not the case…chill guy..
Nyquist laid the foundation for sampling theory through his research on telegraphy in 1924 and 1928.
Claude E. Shannon expanded on this in his 1949 paper, ‘Communication in the Presence of Noise,’ which solidified the modern sampling theory. If you are curious about what kind of paper was actually written, I have included a link so please take a look..
- Certain Factors Affecting Telegragh Speed(1924, Nyquist)
- Certain Topics in Telegraph Transmission Theory(1928, Nyquist)
- Communication in the presence of noise(1949, Shannon)
For those working in music and audio, the finer details aren’t crucial, so I’ll focus on the key points.
1) Only frequencies up to half the sampling rate can be accurately represented (this is known as the Nyquist frequency).
2) The original source with a value higher than the Nyquist frequency is expressed as a value below the Nyquist frequency when sampled (this is the aliasing phenomenon).
3) Therefore, if you filter it in the high-pass to filter out this noise, it becomes clean (this is called the anti-aliasing filter, The original Nyquist paper made telegraph transmission easier through this filtering).
The blocky artifact shown in the first image is also a form of aliasing caused by sampling errors. Since it samples space, it is a bit different from the sound of sampling time.
An ADC (Analog-to-Digital Converter) chip performs these tasks!
High-end ADCs and DACs improve the precision of this process.. Still, Lavry Gold converters are extremely expensive… Maybe due to low demand?
The first part of this video shows how aliasing sounds when you lower the sample rate. Using a filter before this plugin to remove frequencies near the Nyquist limit will significantly reduce aliasing noise.
That’s all for today.
In the next article, I will talk about oversampling and upsampling.
See you in the next post!
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