I am trying to make a basic VoIP implementation and I am stuck obtaining sound samples from the mic and playing the read data on a computer. I tried to sample the voice from the microphone and send it to the serial terminal using a very minimal code:

void setup() {

void loop() {
    int sensorValue = analogRead(A0);

The delayMicroseconds(122) causes the sample rate to be around 8192 Hz. I also tried with a sample rate of 1000 Hz. The data plot obtained looks good, however, when I try to play it with the MATLAB command

sound(csvread('samples.csv'), 8192);

I do not really hear what it should be (I just hear random noise). I am attaching a graph of the obtained data and my microphone circuit. (I also tried subtracting the average from the array so the values are centered around zero, but no use).

Data Obtained:

enter image description here

Microphone Circuit:

enter image description here



Your signal looks like it doesn't have much dynamic range. You may want to look into amplifying it before it reaches the arduino's ADC.

The AVR based Arduinos have a 10 bit ADC -- 0 to 1024. At 5 volts Vref that is

\$ \frac{5volts}{ 1024 counts} = 4.9 \frac{mVolt }{ count}\$

Your signal ranges from ~150 to ~225. This is 75 counts of range or 7% of the ADC's resolution. Your input signal varies by about 366mV peak to peak.

It should vary by about 2.5v peak to peak.


Matlab's documentation on audio files states the following:

The MATLAB sound and soundsc functions support only single- or double-precision values between -1 and 1.

Your arduino code is printing values from 0-1024. Integers. You will need to normalize them and remove the DC bias: make the signal have an average of 0, and a range of [-1 1].

input = double(csvread('samples.csv')); input = input - 512; % convert the [0 to 1024] range to [-512 to 512] input = input ./ 512; % convert the [-512 to 512] range to [-1 to 1] sound(input, 8192);


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