Talk:Analog-to-digital converter

Regarding musical application

This part should talk about the difference between sample rate and resolution regarding music recording and playback. Sample rate will increase the bandwith of the recording and nowadays 44kHz is too low. Theorist might call Nyquist theorem to the discussion, saying the normal person can't hear above 15-17kHz and that 44kHz is enough. The problem is that sound waves are produced mechanically by a speaker and if you simply dont send the hi-frequencies, it will vibrate differently, changing the sound. BTW, DVD is 96kHz. BTW, there is a musical producer that has detected a 95kHz frequency. The guys knew something was bad with his equipment, called in the tech guys, they didnt believe he could hear that high (experts usually hear 20kHz-22kHz) and made a series of test to see if the guy was really feeling it, and HE WAS!

Regarding resolution, 16bits are insufficient for today's standarts. Not only regarding the low SNR, but also regarding the depth, the dynamics of the music. Simply put, between 16bits and 20 or 24 bits, YOU WILL NOTICE the difference, even if you are not an expert. Once again, DVD quality is 24bits.

--- My two cents: we should isolate this discussion in another article, about psychoacoustics. This article would list scientific references about sound perception. I do know one or two articles about high and low frequency perception. Signal quantization is something, hearing quantized signals is another thing.

Now, I don't know what you mean when you say that DVD has 24bits and 96kHz. DVDs are MPEG2 streams, and not raw signals like CDs!!...

AFAIK, sampling rates and bit depths larger than 44100/16 are only required when the signals are going to be digitally processed. But this means "in consumer applications"... The real limits of human perception is a current area of research, an open question. -- nwerneck, 01 Dec 2005 20:25:49 -0200

The music perception is a highly controversial topic and not related to the ADC itself. So the question if the current CD audio standard is sufficient should not be part of the ADC article. If at all this would be a point with the CD standard or digital music part, but even there the controversial part is better avoided. 2A04:1C40:34C9:0:2139:39FF:F647:CF90 (talk) 20:03, 17 June 2022 (UTC)

Aliasing misrepresented

The aliasing section says the following: "The frequency of the aliased signal is the difference between the signal frequency and the sampling rate. For example, a 2 kHz sine wave being sampled at 1.5 kHz would be reconstructed as a 500 Hz sine wave. This problem is called aliasing."

This is a gross misrepresenation of the topic. It mentions sampling, but omits reconstruction. It fails to take into account spectral folding and image bands, and egregiously it fails to mention that *all* frequencies on the frequency axis fold into the band 0 - fs/2.

Please, someone with the knowledge, expertise, and time, rewrite this section, and any other parts similarly afflicted! — Preceding unsigned comment added by 63.230.166.220 (talk) 20:50, 28 June 2012 (UTC)

I don't see a problem with this example. Reconstruction does not come into play here because the error occurs during sampling. ~Kvng (talk) 18:22, 31 January 2019 (UTC)

Nonlinearity

In Analog-to-digital converter § Nonlinearity there is a claim that all ADCs have linearity issues. A 1-bit delta-sigma converter is designed to address this. I'm sure there are some residual non-linearities in supporting circuits but the core process is fundamentally linear. Should we soften this claim? In reviewing articles like this I've found a pattern of WP:NPOV issues from editors who can't seem to believe it is possible for digital systems to represent an analog signal accurately. ~Kvng (talk) 18:11, 31 January 2019 (UTC)

Unclear English: Delta Sigma

A sigma-delta ADC (also known as a delta-sigma ADC) oversamples the incoming signal by a large factor using a smaller number of bits than required are converted using a flash ADC and filters the desired signal band.

The second half of this sentence appears garbled and I don't know the subject matter well enough to correct it with any degree of confidence. 72.141.157.76 (talk) 01:13, 5 November 2022 (UTC)

The describtion is indeed not really understandable for someone who does not know an SD ADC. A simpler describtion would be preferred, but it is a really difficult topic. It is not just the languish part, but also the content that need to be simpler. Maybe the following could be a start:

A sigma-delta ADC (also known as a delta-sigma ADC) is based on noise shaping, over-sampling, digital filtering and decimation. The noise shaping part integrates the difference between the input and a feedback signal from a low resolution DAC (often 1 bit). The feedback signal is generated from a low resolution (often 1 bit) ADC that is reading the filtered (integrated) difference signal. The DAC data represent the input signal over a wide band. A digital low pass filter removes most of the higher frequency quantization noise and decimation reduces the data rate for the final result. So the high rate, low resolution stream is converted to a lower rate, higher resolution result.

Due to the noise shaping from the integrator much of the quantization noise is at high frequencies and thus removed by the digital filter. This way the oversampling gets more efficient (e.g. gain 1 bit from 2 fold oversampling) than normal simple averaging without noise shaping (1 bit from 4 fold oversampling).

High resolution ADCs usually have a higher order filter instead of a simple integrator for noise shaping and this way gain even more bits from oversampling. Ulrich67 (talk) 20:23, 13 January 2023 (UTC)

Improvement implemented by Ulrich67 ~Kvng (talk) 16:21, 20 May 2024 (UTC)