Noise-cancellation with undirected microphones and different amplifiers

Question

I have the following setup https://sketchfab.com/show/7e2912f5f8794a7b96ef3ac5930e090a (It's a 3d viewer, use your mouse to view all angles)

The box has two nondirectional electret microphones(black dots). On the ground there are some elements falling down like water or similar(symbolized by the sphere) and creating noises. On top, someone is speaking in the box. Distances are roughly accurate, so the mouth is pretty close.

Inside the box there are two different amplifiers(but the same electret microphones) with two different amplification circuits(the mouth-one is louder in general and has some normalization circuitry integrated. Long story short, I can record this into a raw audio file with 44100 Hz, 16Bit and Stereo, while the left channel is the upper, the right channel is the lower microphone amplifier output.

Goal is to - even though the electret microphones are not directed and even though there are different amplifiers - subtract the lower microphone(facing the ground) from the upper microphone(facing the speaker) to have noise cancellation.

I tried(With Datei being the raw-filename). This includes a high or low pass filter and a routine to put the final result back into a raw mono file (%s.neu.raw)

The problem is - well - undefinable distortion. I can hear my voice but it's not bearable at all. If you need a sample I can upload one.

EDIT: New code.

static void *substractf( char *Datei)
{
  char ergebnis[80];                                                  
  sprintf(ergebnis,"%s.neu.raw",Datei);
  FILE* ausgabe = fopen(ergebnis, "wb");
  FILE* f = fopen(Datei, "rb");                    
  if (f == NULL)
    return;
  double g = 0.1;
  double RC = 1.0/(1215*2*3.14);
  double dt = 1.0/44100;
  double alpha = dt/(RC+dt);
  double noise_gain = 18.0;
  double voice_gain = 1.0;
  struct {
    uint8_t noise_lsb;
    int8_t  noise_msb;
    uint8_t voice_lsb;
    int8_t  voice_msb;
  } sample;  

  while (fread(&sample, sizeof sample, 1, f) == 1) 
  {
    int16_t noise_source = sample.noise_msb * 256 + sample.noise_lsb;
    int16_t voice_source = sample.voice_msb * 256 + sample.voice_lsb;
    double signal, difference_voice_noise;            
    difference_voice_noise = voice_gain*voice_source - noise_gain*noise_source;
    signal = (1.0 - alpha)*signal + alpha*difference_voice_noise;  
    putc((char) ( (signed)signal       & 0xff),ausgabe);
    putc((char) (((signed)signal >> 8) & 0xff),ausgabe);  
  }   
  fclose(f);                             
  fclose(ausgabe);  
  char output[300];                                 
  sprintf(output,"rm -frv \"%s\"",Datei);
  system(output);
}

while(wo !=EOF) { wo = getc(f); is a small problem. wo needs to be immediately tested for EOF after the getc(). Doing the test beforehand let's wo equal EOF for 1 loop. — chux - Reinstate Monica
2 ideas: 1) use double instead of float. Well A/D converted sounds is on the order of 20-22 bits fidelity, (float has 23) but various mathematical noise can be noticed at times. 2) Instead of subtracting, add. Sometimes the mic's wires are crossed and you need to undo that in software. — chux - Reinstate Monica
2 more ideas: 1)u se r-g*l-r ( or r+g*l) instead of r - fileter_l. (do the filtering After the subtraction) Use struct { int16_t l,r; } sample; while (fread(sample, sizeof sample, f) > 0) { to simplify all the byte splicing you are doing. — chux - Reinstate Monica

r3mainer r3mainer · Accepted Answer · 2013-10-30T03:44:44

Your code doesn't take differences of path length into consideration.

The path difference d₂ – d₁ between the sound source and the two mics corresponds to a time delay of (d₂ – d₁) / v, where v is the speed of sound (330 m/s).

Illustration of path difference with two microphones

Suppose d₂ – d₁ is equal to 10 cm. In this case, any sound wave whose frequency is a multiple of 3300 Hz (i.e., whose period is a multiple of (0.10/330) seconds) will be at exactly the same phase at both microphones. This is how you want things to be at all frequencies.

However, a sound wave at an odd multiple of half that frequency (1650 Hz, 4950 Hz, 8250 Hz, etc.) will have changed in phase by 180° by the time it reaches the second mic. As a result, your subtraction operation will actually have the opposite effect — you'll be boosting these frequencies instead of making them quieter.

The end result will be similar to what you get if you push all the alternate sliders on a graphic equaliser in opposite directions. This is probably what you're experiencing now.

Try estimating the length of this path difference and delaying the samples in one channel by a corresponding amount. At a sampling rate of 44100 Hz, one centimetre corresponds to about 0.75 samples. If the sound source is moving around, then things get a bit complicated. You'll have to find a way of estimating the path difference dynamically from the audio signals themselves.

Noise-cancellation with undirected microphones and different amplifiers

2 Answers