Basic audio record queue and playback

[Bretcj7]

Looking for documentation for how to use a few filters and use the audio queue and playback. Specifically to output left channel and right channel separate as my pcb I made has two mems microphones

I’m trying to follow the GUI designer but it’s not for newbies like myself

Here is my code: (pseudo code)

Setup()
AudioInputI2S i2sIN;
AudioOutputI2S i2sOUT;
AudioFilterBiquad biquadINNER;
AudioFilterBiquad biquadOUTER;
AudioRecordQueue queueInInner;
AudioRecordQueue queueInOuter;
AudioPlayQueue queueOut;
AudioConnection pathCord0(i2sIN,0,biquadINNER,0);
AudioConnection pathCord3(i2sIN,1,biquadOUTER,0);
AudioConnection patchCord1(biquadINNER, 0, queueInInner, 0); // This is where we swap channels if necessary
AudioConnection patchCord2(biquadOUTER, 0, queueInOuter, 0); // Swap these two 0,1 to swap audio channels
AudioConnection patchCord9(queueOut,0,i2sOUT,0);
AudioConnection patchCord10(queueOut,0,i2sOUT,1);
AudioControlSGTL5000 sgtl5000_1;


Loop()
if (queueInInner.available() >= 2 && queueInOuter.available() >= 2) { //input buffer is filled, unload it please
memcpy(bufferOuter, queueInOuter.readBuffer(), 256);
memcpy(bufferInner, queueInInner.readBuffer(), 256);
memcpy(bufferAverage+(256*buffer_ptr), bufferInner, 256);

/*if (filterSwitch == 1) {
my_adaptive_filter(bufferInner, bufferInnerLeftover, bufferOuter, bufferOuterLeftover, errorout, aweights,10000, BufferLen, AFiltLen);
}*/
memcpy(errorout, bufferInner, 256);

int16_t * playerptr = queueOut.getBuffer();
memcpy(playerptr,errorout,256);
memcpy(bufferAverage+(256*buffer_ptr++), errorout, 256);
queueOut.playBuffer();
queueInOuter.freeBuffer();
queueInInner.freeBuffer();
}

 

[PaulStoffregen]

I’m trying to follow the GUI designer but it’s not for newbies like myself

We have a 31 page tutorial to help you get started with the GUI design tool. It also comes with a 45 minute full walkthrough video. If you get stuck anywhere in the tutorial, you can watch how we did it in the video!

However, the queue objects are not simple beginner material. They give you access to the raw audio data, which means you have to carefully craft your program to handle the rapid data streams of the audio. Usually this should be avoided, just let the audio library do all the audio data processing.

If your goal is simply to use 2 MEMS mics, apply filters, and then output those signals elsewhere, you can do all that in the audio library without use of the queues to transfer the data to your program. Do yourself a favor and just let the audio library do the heavy lifting of processing all that data. It's designed to be fairly simple to use that way.