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Voice Recording Using PIC microcontroller :

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nikhilraj

New Member
Hello everyone,

I am working on a project which should record the voice lively and simultaneously that voice will be sent to the local computer. For this I am using Microchip's "PIC24FJ128GA010" Microcontroller. After recording the analog voice will be converted to digital form and I am sending that converted data to the PC which is remotely connected (Wi-Fi) with this PIC.

I am using Explorer 16 board for this project. PIC can be embedded on the board. I have designed a separate circuit board for micro-phone which will record the voice. I am able to here the voice on the speaker if I give that output to the speaker.

So now the problem is that I am not able to send it through wi-fi. I am using ZG2100 wifi module and it is connected to the Explorer-16 board as using PIC-tail-connector. So I have tested the wifi connection and everything is fine. I think its the problem with sampling the recorded voice. As I am the new beginner I really dont know how to do that !!!

I want to know how I need to sample the recorded voice as the humans can listen to it ???

Can anyone please shed a light on this... I really appreciate your help..

Thank you very much for your consideration.

Thank you.
Nikhil Raj
 

nikhilraj

New Member
Thanks for your reply Colin :)

But that looks different !! I am not using any .WAV format file I can play RAW format directly. I am actually speaking on a microphone and that output of the mic circuit is connected to analog pin of PIC and the ADC will convert it to digital format (RAW). So this RAW data will be sent from PIC micro to local comp using ZG2100 Wi-Fi module. I have transmitted the recorded thing in to the system and I could successfully played it too. But I have set the auto sample rates... Problem is I dont know how exactly to set the sampling rate for this audio data after recording.

Can You please give some example codes so that I can get in to the idea of sampling.. PLease help :)

Thanks.
Nikhil
 

rumpfy

Active Member
Hi Niki,
In the 1930's paper by C.E.Shannon, of Bell Labs, titled; 'A mathematical theory of Communication', he showed how the sampling rate of an analog waveform needed to be at twice the highest frequency required to be transmitted.
In the telephone system, the bandwidth is in the 0.3 to 4.0 KHz range and therefore the sampllng rate needs to be 8 Kbits/sec. In modern phone systems the sampling rate for speech telephony is 8k bits/sec. For the bell system, there are 24 channels stacked to give a 1.44Mbits system. The CCITT system samples at the same rate and the 32 channel system runs at 2Mbit/sec.
In your project you need to sample at a rate of twice the highest frequency you wish to transfer through your system. Generally for speech, a bandwidth of say 6 to 7 Khz would be sufficient. Telephone quality would use a sampling rate of 8 Ksamples/ sec.
With audio CD's, the normal sampling rate is 44.1 KHz and this is slightly higher than twice the usual 20 to 20000 Hz bandwidth required for high quality audio. If you look at the various sampling rates for audio recordings made on say iTunes, you can select the sampling rate to suit your fidelity requirements.
hope this helps.
pr.
 
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