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Say we have a 12-bit ADC with 30 MHz sampling frequency. I want to perform several DSP operations on output digital signal so that resulting signal would have increased accuracy to 14 bits and output sampling frequency would become 2 MHz. In nutshell I want a DSP block that converts a 12-bit ADC with 30 MHz Sampling rate to a 14-bit ADC with 2 MHz sampling. Even a slight tip for me to know where to look for would be helpful. (This project would be implemented on a FPGA.)

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    $\begingroup$ hm, sounds like this isn't the perfect project for you. But, attempting to help you nevertheless: something that changes the sampling rate of a signal is called a resampler; the special case where the output rate is a fraction of the input rate is a decimator. $\endgroup$ – Marcus Müller Jan 19 at 14:16
  • $\begingroup$ Also, you never accepted an answer to your previous question, is that intentional? $\endgroup$ – Marcus Müller Jan 19 at 14:17
  • $\begingroup$ Thank you marcus. I would research decimator implementations. $\endgroup$ – Krsh Jan 19 at 16:06
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    $\begingroup$ No, i found the answer myself and forgot to check the question i asked. $\endgroup$ – Krsh Jan 19 at 16:06
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With typical sampling (not noise shaped) you get half a bit for every halving of the sampling rate (with proper filtering of the out of band noise prior to downsampling). So a decimate by two for example consists of a half band filter followed by a down-sample by two which is done simply by selecting every other sample. Assuming white noise, the half band filter removed half the noise or 3 dB, and the the SNR of ADC’s goes as 6 dB/bit.

Therefore, if you went from 30 MHz to 7.5 MHz with a decimate by four, you would gain a full bit or 12 bits extended to 13 bits assuming noise is limited by quantization noise and the quantization noise process is sufficiently white as typically assumed and approximated.

It’s should be straight forward to see now how to go from 12 bits to 14 bits through successive decimations. (And the filtering and decimation can be done in one block each or distributed which gives flexibility in the filter designs used).

An additional caution when working with real hardware is to refer to the expected Effective Number of Bits (ENOB) for the actual ADC used and in what conditions of sampling, input frequency etc as the number of bits to be achieved and improved on, and pay attention to actual spur levels (as given by the Spurious Free Dynamic Range, SFDR) that will ultimately limit the SNR since decimation will do nothing to reduce the power level of in band spurs. Often these spurs are below the noise level due to quantization in which case dynamic range improvement can be gained through decimation.

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  • $\begingroup$ Thanks a lot for your comprehensive answer.Is there any resource that you would recommend for further study? $\endgroup$ – Krsh Jan 19 at 20:17
  • $\begingroup$ Hi @Krsh - If you look under my name here I have several other answers related to this with more detail. As far as further study with multirate sampling I would recommend fred harris' book Multirate Signal Processing. I don't know your background but assuming you have already studies Signals and Systems and basic Digital Signal Processing you should be well equipped. I also teach DSP courses online through the Boston IEEE that you can watch for at this link (I have a Python course starting next week!) ieeeboston.org/2021-courses $\endgroup$ – Dan Boschen Jan 20 at 3:13
  • $\begingroup$ Thanks a lot.Yes i have the background you assumed but the problems that i solved in university usually consisted of just filtering then downsampling the signal and never needed to use this sort of problems.Thanks a lot for the mentioned courses. $\endgroup$ – Krsh Jan 21 at 7:21

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