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Hello Hitesh, using a microcontroller and other things depend on the resolution of the media as well. For example, in case you just need an asynchronous serial interface which will be used to connect with the Bluetooth module and an ADC (Analog to Digital Converter), you can choose any 8-bit microcontroller host as well. This happens when the quality needed ranges from 8-bit resolution to 12-bit resolution. However, in case the requirement is 16-bit resolution to get high-fidelity, then it is better to use the microcontroller along with an external ADC to get a better result. For external ADC, you can choose from different options available as mostly the features will be beneficial. As for the 8-bit resolution, I would suggest you get your microcontroller from Microchip’s PIC18 products. I have used it personally and it really works well as it reduces the workload that falls on the CPU too. This is because it comes with the Direct Memory Access feature to simplify the internal transfer which is rare in 8-bit microcontrollers. I hope this helped you.

 Hi Hitesh, this is a complex process that you are asking to explain. However, I will try to be as explicit as possible and hope that you get the points clearly. First of all, I would like to say that Noah has answered really well but there is another parameter that needs to be followed; that is,the type of audio signal. Depending on the audio signal the sampling rate can differ. For example, In case of speech, 8 ksamples/s will become the standardised rate. If the frequency of the audio signal ranges somewhere between 20Hz and 20kHz, then the standard sampling frequency rate will be 44.1kHz. Also, where there is a limited bandwidth on an audio signal, then the Nyquist sampling rate/theorem can be applied. The theorem is like fs>= 2 fm. (In such cases, usually fs is considered to be greater than 2 fmax (maximum frequency) when it comes to audio signals as it is a law pass filter that makes the interpolation easier. Lastly, the audio signal can also be compressed in MP3 form through the MPEG standards as it improves the transmission rate as well as saves a lot of storage.   So, the main point is that converting audio signals to digital data depends on the type of signal you are converting. I hope I made my point clear and that it will help you in sorting your project further.

Hi Hitesh, I see you have already received answers and from my knowledge, I don’t have anything to add additionally. I would just like to say that give your best but don’t feel disheartened if the result doesn’t turn out the way you want as it often happens during the experiments. If you come across a time when you need help, consult an expert maybe. It helps in many ways, especially when you need to clear your mind with reasoned and logical answers. You can get a consultation from here too, when you come across a hurdle. https://www.bostondissertationhelp.com/ Or maybe search for similar websites who can provide reliable help in your country. Good luck, Buddy!