What in general, is the difference between a Audio ADC and an ADC, which has a similar sample rate (up to 100khz in my example) and a similar bit depth (12 bit in my example). I would like to use the MPC3208 in my circuit to convert analogue signals from a mic preamp to a digital signal. Do you think I can use it as a audio ADC? Subquestion: Those 12 bit should be enough when not doing expensive editing afterwards, right? I don't expect a superior HiFi sound, just one that sounds 'good enough' for consumers.


The important thing audio ADCs do is blocking higher frequency signals before sampling. An actual high quality audio ADC these days these days usually contains oversampling circuitry and digital filters since you cannot do the filtering job with analog filters alone at sufficient quality.

12bit of audio gives you 72dB of signal above quantization. You'll need a lot of noise shaping to make the quantization noise innocuous, not having any obvious relation to quiet signals. Good tape drives are in that range, and they degrade "graciously" both with regard to overdriving your signal and having it fade into the noise floor.

So don't expect things to "just work" with specs as confined as that.

  • Why don't analog filters provide a sufficient quality? I mean, there are for example low pass filters out there with a fairly steep slope!?
    – Ben
    Oct 11 '18 at 22:37
  • 1
    When you don't use oversampling, an analog filter for a 44.1 kHz ADC would have to provide 90 dB of attenuation over a range of just 2 kHz. This isn't just 'fairly steep', it's a vertical cliff. Filters this steep have significant drawbacks and it's hard/expensive to make a filter that won't audibly degrade the signal.
    – Hobbes
    Sep 24 '19 at 18:01

It depends on the application I guess. For example, a lot of speech processing (and transmission) nowadays is happening in the range that reaches up to 12 KHz or 16 KHz at maximum. Whether you intend to use it for speech or musical (or any other kind of audio) application you should make sure that aliasing will not occur (in considerable levels). This means that with MPC3208 you will have to take care of that. As both user25614 and user 26957 suggested you will either have to spend considerable amount of time to design a very good anti-aliasing filter or opt for high sampling rate and design a filter with reasonably low cut-off frequency.

In my opinion though, your biggest problem is the bit-depth here. 8-bits sound like Atari (this may be a matter of personal taste though as there are still people composing musical pieces in 8-bits) and, again, depending on your application 12 bits may still not be sufficient. Additionally, 12 bits ADC does not mean that you will end up with 12 bits of usable dynamic range. As the datasheet states there are also offset errors and gain errors (although they refer to DC accuracy they are still relevant to AC signals) and still we haven't included the noise of your circuit (and possibly from the anti-aliasing filter), and the possible instability of your reference signal.

I would suggest you use any available converter to you to downsample some material relevant or similar to your application and check if the quality is sufficient for you. The conversion will not necessarily provide identical results but can be a good "approximation"/estimation to what your converted signal will sound like in the case of a well designed circuit.


Audio ADCs of tolerable quality will not sample with a mere 100kHz. You need to significantly oversample in order to have reasonable leeway for digital filtering before downsampling to the final sampling frequency. Analog filters of comparable steepness and phase linearity would require so many stages that they would introduce an intolerable amount of noise.

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