If I want to use 18-bit ADC and DAC for an audio project how do I estimate how powerful CPU will I need? How to choose a good sampling rate for CD quality?
Standard CD sample frequency is 44.1 kHz.
The processor has to have a bit width, speed and memory sufficient to accomplish your goals.
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So, let's say 44kSPS? But how to estimate which processor? Also do I need amplification between the ADC/DAC + CPU ?
If you want help, provide basic information about your project.
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I don't exactly know but simply playing (uncompressed) audio doesn't require THAT much processing power. CD audio (16-bits, 44.1kHz, 2-channels) has a bitrate of 1411kbps (or 176kB per second). With18-bits you'll probably need another whole byte so that's 50% more data.
Of course you have to read from the memory and write to the DAC. And if there is any multitasking or "advanced processing" or decoding of compressed audio, you'll need a buffer to write-out the data smoothly.
CDs use 44,100 samples per second. You can probably "get-away" with going lower but you might be sacrificing some sound quality (depending on how low you go).
Nyquist theory says the signal can't be higher than half the sample rate (22,050 Hz for CD), and you generally have a low-pass "smoothing filter" and filters are imperfect (not a "brick wall") and 22,050 allows the actual-filtered audio to go all the way to the "traditional" human hearing limit of 20kHz.
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Why not say 44.1k SPS - because that's what it is!
No: that bit is all digital - it is just at logic levels.
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Do I need any filtering before the ADC and after the DAC?
You haven't given any details of your project, so no one can even guess.
Try it and see.
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Yes, sorry! I want to make a recording device with Arduino, just as a hobby! Play a bit with the filters, A/D, D/A. So, I need to record sounds, store them and when I want to play them. Also, I was amazed by this device: https://www.youtube.com/watch?v=4Wi9soy2nGQ
Sounds like you will learn by doing, which is fine. If the audio quality is poor, investigate the use of filters.
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Between the microphone and the ADC, as I found from google search, I need a low pass-filter. I believe something like [0, 44.1KHz]. Of course the cutoff frequency will be a bit more. Do I need anything else?
I am asking, because as far as the DAC is concerned, I found this article: Equalizing Techniques Flatten DAC Frequency Response | Analog Devices which uses many components. And I am worrying, does also ADC need anything more?
Perhaps not. To learn more about the topic, look up "sampling theorem", "signal aliasing" and "Nyquist limit".
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I have found this: Second-order Active Low Pass Filter Circuit from here: https://www.electronics-tutorials.ws/filter/filter_5.html My question is how do I supply them with voltage? Do I need +10 Volt and -10 Volt? I will use a 3.3 Volt Arduino. Also my power supply of the whole circuit will be 0-5 Volts. The audio input signal will be AC. So I believe I need this audio op-amp (rail-to-rail) TLV2462 ? do you see any problem here? Which capacitor (C1 or C2) should I keep stable and find the value of the other according to the cut-off frequency?
Not yet. So far these ruminations are completely theoretical, with no actual problem to be solved.
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Ok, the Arduino works on 0-3.3 Volt! So, maybe passive filter is better...I mean there is no idea for amplification...
Which has nothing at all to do with the audio signal.
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Why is that?
I'm confident that you will eventually figure that out, once you have the equipment.
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As far as I understand the LowPass filter between the microphone and the A/D will be of the same characteristics as the LowPassReconstruction filter after the D/A? Am I correct?
Both filters are options to consider rather carefully.
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