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So currently I am attempting to pick up audio signals in an arduino via the analog input pins and an electret microphone circuit.

The device I am using for audio capturing is the sparkfun sound detector [It wouldnt allow me to post more than 2 links and the pictures I've uploaded after the edits are more important so I apologize for taking the link for this item out]

I have ordered pins to solder onto this board, but at the moment, I have it wired up as shown in the attached picture (which may be the issue?)

The problem is, the audio output from the sound-detector board gives me a constant value on the arduino, whether there is sound coming into the microphone or not. The only time the value changes is when I unplug the sound-detector board from the power source, at which point I get values of 0.

The envelope detector output, however, works correctly. In a quiet environment, I get very low voltage values, and when speaking into the microphone, I get varying voltage values depending on how loudly I am speaking into the microphone.

However, for my application, I need raw audio-data and not the audio signal envelope.

Does anyone know what the issue could be, the code is very simply and shown below.

float i;
void setup()
{
  Serial.begin(9600);
}

void loop() {
  // send the value of analog input 0:
  Serial.println(analogRead(A0));
  // wait a bit for the analog-to-digital converter
  // to stabilize after the last reading:
}

enter image description here

3/16 EDIT: Sampling Rate Code

So some people were mentioning how I am only achieving a sampling rate of around 150 hz or 500 hz using the analogread function (Unless I misunderstood)

Below is some code I am using to measure the amount of time that passes between an analogread sample;

// Arrays to save our results in
unsigned long start_times[100];
unsigned long stop_times[100];
unsigned long values[100];

// Setup the serial port and pin 2
void setup() {
  Serial.begin(9600);
}


void loop() {  
  unsigned int i;

  // capture the values to memory
  for(i=0;i<100;i++) {
    start_times[i] = micros();
    values[i] = analogRead(A0);
    stop_times[i] = micros();
  }

  // print out the results
  Serial.println("\n\n--- Results ---"); 
  for(i=0;i<100;i++) {
    Serial.print(values[i]);
    Serial.print(" elapse = ");
    Serial.print(stop_times[i] - start_times[i]);
    Serial.print(" us\n");
  }
  delay(6000);

the print-out results of the above code shows an average time of about 116 micro-seconds;

1/116-uS is around 8.6kHz, am I misunderstanding something here? I understand that the Serial.print command in the first block of code is slowing the sampling down significantly, if that is what users here were initially trying to convey; However, if I sample and store the analog-input values in an array and plot them at some later point, without using any serial command, the audio output should be relatively well via the sound-detector and that plotted signal should resemble a fairly decent audio-signal, correct?

8.6kHz is what I was aiming to achieve as far as sampling rate goes, will the audio output really be this distorted?

3/17 EDIT: Audio-Output MATLAB Plot

Well, here is a quick Matlab plot I made of the audio signal (I simply took 1000 sampled values and plotted them against a time vector)

This is actually looking pretty decent, maybe the non-soldered wires are a big issue

If i'm missing anything else that might be effecting this i'm sure someone here will catch it.

enter image description here

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  • \$\begingroup\$ Considering the amount of time it will take to 'print' the sample value to the serial port at a baud rate of 9600, that delay(2) is redundant. \$\endgroup\$
    – brhans
    Mar 16, 2017 at 19:52
  • \$\begingroup\$ Yup your sample rate is under 500hz. What is your signal level? \$\endgroup\$
    – Trevor_G
    Mar 16, 2017 at 19:54
  • \$\begingroup\$ Took out the delay(2); The signal is just me speaking into the microphone Trevor, I'm sorry I don't understand your question. \$\endgroup\$ Mar 16, 2017 at 19:58
  • \$\begingroup\$ I mean do you have a scope... have you looked at what the signal level is out of the sparkfun board when you talk into the mic? \$\endgroup\$
    – Trevor_G
    Mar 16, 2017 at 20:00
  • \$\begingroup\$ Oh, sorry. I am printing out the values and viewing them on the serial monitor. I get constant values of around 350-360, which converted to voltage is around 1.6 volts; I am also graphing this using the processing software which runs on java and I get the same values. I am assuming this is just noise \$\endgroup\$ Mar 16, 2017 at 20:03

1 Answer 1

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You can't reach your stated goals the way you are trying to go.

  1. You appear to have the "envelope" output connected to the Arduino, but say you want the wave rather than the envelope. Connect the "audio" output to the Arduino.

  2. Sampling with the Arduino the way you are, you will never get anything that looks like digitized audio out of the comport. You can get (maybe) 300 samples per second the way you have that written. That allows a total audio bandwidth of less than 150Hz. That is around the highest tones of a subwoofer. Not much speech to be had there.


If you connect the "audio" output to your analog input on the Arduino, then it should show values around 500 when it is quiet. The detector board has the audio "floating" on 2.5VDC, which is half of the Arduino input range - half of 1024 is 512.

You picture actually shows no connection between the detector and the analog in on the Arduino.

You can solder the wires to the detector. That would work better than "poke 'em through and pray."


You've updated your question.

Capturing and then transmitting all in one block is better.

It still won't get you very far.

The Arduino has 2 kilobytes of RAM. So, after less than one fourth of a second, the RAM is full. Transmitting that block of data over the serial port will take while, and you can't sample and transmit data at the same time if you want to sample as fast as possible.

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  • \$\begingroup\$ Ya, you need at least like 32kHz sampling rates preferably at 12-bit resolution for general audio, maybe 8kHz at 10-bits can work for basic speech \$\endgroup\$
    – vicatcu
    Mar 16, 2017 at 21:08
  • \$\begingroup\$ JRE, I understand that the detector board isnt properly connected to the arduino, this picture was just to show the wires going through the holes in that manner and the sites' opinion on whether that might be the issue; I guess I do not understand the differences between baud rate, sampling rate, and the arduino yun's default sampling rate. I will have to dig further into that material. \$\endgroup\$ Mar 16, 2017 at 21:52
  • \$\begingroup\$ Also, best case arduino sampling rate at full 10-bit resolution is about 125/13 kHz = 9.6kHz, FYI... and that is using continuous sampling mode, which means using an ISR implementation, not just analogRead In a loop. \$\endgroup\$
    – vicatcu
    Mar 16, 2017 at 22:25
  • \$\begingroup\$ Guys, please correct me if I am wrong, but I believe the default analogread function is taking in samples at around 8k; I will edit the main post to show you why I believe so \$\endgroup\$ Mar 16, 2017 at 23:19
  • \$\begingroup\$ @vicatcu: You can do just fine with 8 bits and an 8kHz sampling rate. Telephone systems used just exactly that for a very long time. Checkout ISDN. \$\endgroup\$
    – JRE
    Mar 17, 2017 at 6:37

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