Bits, hertz, shaped dithering …


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Bits, hertz, shaped dithering …

bits

What is behind these concepts?

bits

When developing the standard for CD Audio (CD Audio), 44 kHz, 16-bit, and 2-channel (ie stereo) settings were adopted. Why exactly so many? What is the reason for this choice, and also why are attempts being made to increase these values ​​to, say, 96 kHz and 24 or even 32 bits …

Let’s first find out with the sampling resolution, that is, with the bitness. You happen to have to choose between the numbers 16, 24 and 32. The middle values ​​would of course be more convenient in terms of sound, but too unpleasant for use in digital technology (a very controversial statement, since that many ADCs have 11 or 12 bit digital output (status approx.).

What is this parameter responsible for? Simply put, for dynamic range. The volume range played simultaneously is from the maximum amplitude (0 decibels) to the lowest that the resolution can transmit, for example, approximately minus 93 decibels for 16-bit audio. Interestingly, this is strongly related to the noise level of the soundtrack. In principle, for 16-bit audio, it is quite possible to transmit signals with a power of -120 dB, however, these signals will be difficult to apply in practice due to such a fundamental concept as sampling noise …. The fact is that when taking digital values, we always make mistakes, rounding the true analog value to the closest possible digital value. The smallest possible error is zero, but at most we are wrong with half of the last bit (bit, hereinafter, the term least significant bit will be abbreviated MB). This error gives us the so-called sampling noise, a random discrepancy between the digitized signal and the original. This noise is constant and has a maximum amplitude equal to half of the least significant bit. This can be considered as random values ​​mixed in the digital signal. This is sometimes called rounding noise or quantization noise (which is a more accurate name since encoding the amplitude is called quantization, and sampling is the process of converting a continuous signal into a discrete sequence (pulses) – approx . comp.).

Let’s dwell in more detail on what is meant by signal power, measured in bits. The strongest signal in digital audio processing is generally taken as 0 dB, this corresponds to all bits set to 1. If the most significant bit (hereinafter SB) is set to zero, the resulting digital value will be half, which corresponds to a loss level of 6 decibels (10 * log (2) = 6). Therefore, by zeroing the most significant bits to the least significant, we will decrease the signal level by six decibels. It is clear that the minimum signal level (one in the least significant bit and all other digits are zeros) is (N-1) * 6 dB, where N is the digit capacity of the sample. For 16 digits, the weakest signal level is 90 decibels.

When we say “half the least significant bit”, we do not mean -90/2, but half a step to the next bit, that is, another 3 decibels less, minus 93 decibels.

We return to the choice of scanning resolution. As already mentioned, digitization introduces noise at the level of the middle of the least significant bit, which means that a 16-bit digitized record constantly makes noise at minus 93 decibels. It can transmit signals and is quieter, but the noise is still -93 dB. On this basis, the dynamic range of digital sound is determined: where the signal-to-noise ratio is transformed into noise / signal (there is more noise than the useful signal), the edge of this range is at the bottom. Therefore, the main criterion for digitizing is the amount of noise. Can we afford a recovered signal? The answer to this question depends in part on how much noise was on the original track. An important conclusion: if we digitize something with a noise level of less than 80 decibels, there is absolutely no reason to digitize it to more than 16 bits, since, for one thing, the noise of -93 dB adds very little to the existing one. Huge (comparatively) noise of -80 dB and, on the other hand, quieter than -80 dB on the phonogram itself, the noise / signal already starts, and there is simply no need to digitize and transmit said signal.


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Author: R. Arias

R. Arias is the author of this article and has extensive experience for more than 30 years as a recording engineer and audio specialist, as well as more than 20 years of experience creating algorithms related to audio and video. Linkedin